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2534 Commits

Author SHA1 Message Date
Leif Madsen
60a568bb28 Use autotagged externals
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.6.0.13-rc1@209433 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-28 14:28:47 +00:00
Leif Madsen
d053633e50 Importing files for 1.6.0.13-rc1 release.
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.6.0.13-rc1@209432 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-28 14:28:42 +00:00
Leif Madsen
de99ea4d23 Creating tag for the release of asterisk-1.6.0.13-rc1
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.6.0.13-rc1@209420 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-28 14:23:20 +00:00
Kevin P. Fleming
d060351d79 Correct error in backport of latest app_fax fixes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@209394 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-28 12:01:36 +00:00
Tilghman Lesher
e106e7a1cf Merged revisions 209317 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r209317 | tilghman | 2009-07-27 19:14:12 -0500 (Mon, 27 Jul 2009) | 9 lines
  
  Merged revisions 209315 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r209315 | tilghman | 2009-07-27 19:12:03 -0500 (Mon, 27 Jul 2009) | 2 lines
    
    Publish French extra sounds
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@209325 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-28 00:19:38 +00:00
Kevin P. Fleming
d7f3974c8d Merged revisions 209279 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r209279 | kpfleming | 2009-07-27 16:43:36 -0500 (Mon, 27 Jul 2009) | 7 lines
  
  Cleanup T.38 negotiation changes.
  
  Convert LOG_NOTICE messages about T.38 negotiation in debug level 1 messages,
  clean up some looping logic, and correct an improper use of ast_free() for 
  freeing an ast_frame.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@209280 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-27 21:44:08 +00:00
Kevin P. Fleming
dcefa607e5 Merged revisions 209256 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r209256 | kpfleming | 2009-07-27 16:21:43 -0500 (Mon, 27 Jul 2009) | 10 lines
  
  Make T.38 switchover in ReceiveFAX synchronous.
  
  In receive mode, if the channel that ReceiveFAX is running on supports T.38,
  we should *always* attempt to switch T.38, rather than listening for an incoming
  CNG tone and only triggering on that. The channel may be using a low-bitrate
  codec that distorts the CNG tone, the sending FAX endpoint may not send CNG
  at all, or there could be a variety of other reasons that we don't detect it,
  but in all those cases if T.38 is available we certainly want to use it.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@209259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-27 21:22:22 +00:00
Mark Michelson
af177fd104 Blocked revisions 209235 via svnmerge
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  r209235 | mmichelson | 2009-07-27 15:54:54 -0500 (Mon, 27 Jul 2009) | 5 lines
  
  Gracefully handle malformed RTP text packets.
  
  AST-2009-004
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@209236 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-27 20:57:19 +00:00
David Brooks
6d177693e0 Merged revisions 209098 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r209098 | dbrooks | 2009-07-27 11:33:50 -0500 (Mon, 27 Jul 2009) | 6 lines
  
  Fixing typos. Replaces "recieved" with "received" and "initilize" with "initialize"
  
  (closes issue #15571)
  Reported by: alecdavis
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@209221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-27 20:23:33 +00:00
Mark Michelson
7f81823913 Merged revisions 209197 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r209197 | mmichelson | 2009-07-27 15:11:42 -0500 (Mon, 27 Jul 2009) | 9 lines
  
  Honor channel's music class when using realtime music on hold.
  
  (closes issue #15051)
  Reported by: alexh
  Patches:
        15051.patch uploaded by mmichelson (license 60)
  Tested by: alexh
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@209198 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-27 20:16:28 +00:00
Mark Michelson
881cdb00e5 Merged revisions 209132 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r209132 | mmichelson | 2009-07-27 12:50:04 -0500 (Mon, 27 Jul 2009) | 24 lines
  
  Merged revisions 209131 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul 2009) | 18 lines
    
    Allow for UDPTL to use only even-numbered ports if desired.
    
    There are some VoIP providers out there that will not accept SDP
    offers with odd numbered UDPTL ports. While it is my personal opinion
    that these VoIP providers are misinterpreting RFC 2327, it really is
    not a big deal to play along with their silly little games. Of course,
    since restricting UDPTL ports to only even numbers reduces the range
    of available ports by half, so the option to use only even port numbers
    is off by default. A user can enable the behavior by setting
    use_even_ports=yes in udptl.conf.
    
    (closes issue #15182)
    Reported by: CGMChris
    Patches:
          15182.patch uploaded by mmichelson (license 60)
    Tested by: CGMChris
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@209133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-27 17:52:36 +00:00
David Brooks
7351db2bd1 Just replacing typos "recieved" with "received".
From issue #15360, forgot to apply to trunk and other branches.



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@209061 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-27 16:06:10 +00:00
Kevin P. Fleming
a1cb9549be Merged revisions 209056 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r209056 | kpfleming | 2009-07-27 10:38:59 -0500 (Mon, 27 Jul 2009) | 10 lines
  
  Restore explicit export of ASTCFLAGS/ASTLDFLAGS and underscore-variants to sub-makes.
  
  During the recent Makefile improvements I made, it seemed the 'make' was
  automatically carrying down the ASTCFLAGS/ASTLDFLAGS settings to sub-makes,
  so I removed the explict export of them. However, there are some circumstances
  where make does this, and some where it does not, so I've brought them back
  to ensure they are always exported. I also removed an extraneous double setting
  of _ASTLDFLAGS on *BSD platforms.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@209057 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-27 15:39:44 +00:00
Jeff Peeler
ab117ba86d Merged revisions 208924 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r208924 | jpeeler | 2009-07-26 20:20:37 -0500 (Sun, 26 Jul 2009) | 9 lines
  
  Merged revisions 208923 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r208923 | jpeeler | 2009-07-26 20:18:31 -0500 (Sun, 26 Jul 2009) | 2 lines
    
    Fix logic errors from 208746
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208925 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-27 01:21:54 +00:00
Michiel van Baak
420cfcc16a Blocked revisions 208886 via svnmerge
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  r208886 | mvanbaak | 2009-07-26 16:00:52 +0200 (Sun, 26 Jul 2009) | 2 lines
  
  add OpenBSD to the install_prereq script
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-26 14:01:55 +00:00
Michiel van Baak
aa3928f100 Blocked revisions 208848 via svnmerge
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  r208848 | mvanbaak | 2009-07-25 14:28:38 +0200 (Sat, 25 Jul 2009) | 2 lines
  
  libxml2-dev is needed as well by default.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208851 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-25 12:29:44 +00:00
Michiel van Baak
ed6555a37c Blocked revisions 208813 via svnmerge
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  r208813 | mvanbaak | 2009-07-25 14:03:25 +0200 (Sat, 25 Jul 2009) | 10 lines
  
  add default alias reload to run module reload.
  
  Requiring 'module reload' to reload everything, including
  core etc makes russell very unhappy.
  
  The default configuration already loads the 'friendly' aliases template.
  Added 'reload=module reload' to that template.
  
  Also removed the comment in main/cli.c that reload should come back.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208814 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-25 12:06:45 +00:00
Jeff Peeler
bf0a2c9fa5 Merged revisions 208749 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r208749 | jpeeler | 2009-07-25 01:23:18 -0500 (Sat, 25 Jul 2009) | 13 lines
  
  Merged revisions 208746 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r208746 | jpeeler | 2009-07-25 01:19:50 -0500 (Sat, 25 Jul 2009) | 7 lines
    
    Fix compiling under dev-mode with gcc 4.4.0.
    
    Mostly trivial changes, but I did not know of any other way to fix the
    "dereferencing type-punned pointer will break strict-aliasing rules" error
    without creating a tmp variable in chan_skinny.
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208752 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-25 06:24:47 +00:00
Mark Michelson
a564e9b944 Blocked revisions 208622 via svnmerge
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  r208622 | mmichelson | 2009-07-24 14:24:28 -0500 (Fri, 24 Jul 2009) | 16 lines
  
  Don't impose an arbitrary limit on member lines in queues.conf
  
  I know what some of you are thinking: "UGH! Mark, why are you using
  ast_strdup and ast_free for the string when you can just use ast_strdupa
  and let the memory free itself?! Have the bats been chewing on your brain
  again?"
  
  Based on past experiences, I don't like using ast_strdupa inside a loop.
  It's a good way to potentially exhaust stack space. Also, since this only
  happens when reloading queues, I don't think that heap allocations and
  frees are going to be a huge problem.
  
  (closes issue #15559)
  Reported by: amorsen
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208655 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-24 19:33:54 +00:00
Russell Bryant
765d10231b Merged revisions 208593 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r208593 | russell | 2009-07-24 13:42:32 -0500 (Fri, 24 Jul 2009) | 14 lines
  
  Merged revisions 208592 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r208592 | russell | 2009-07-24 13:38:24 -0500 (Fri, 24 Jul 2009) | 7 lines
    
    Do not log an ERROR if autoservice_stop() returns -1.
    
    This does not indicate an error.  A return of -1 just means that the channel
    has been hung up.
    
    (reported in #asterisk-dev)
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208594 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-24 18:49:43 +00:00
Mark Michelson
6aa63436ab Merged revisions 208588 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r208588 | mmichelson | 2009-07-24 13:31:04 -0500 (Fri, 24 Jul 2009) | 16 lines
  
  Merged revisions 208587 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r208587 | mmichelson | 2009-07-24 13:26:50 -0500 (Fri, 24 Jul 2009) | 10 lines
    
    Only send a BYE when hanging up a channel that is up.
    
    For cases where Asterisk sends an INVITE and receives a non 2XX final
    response, Asterisk would follow the INVITE transaction by immediately
    sending a BYE, which was unnecessary.
    
    (closes issue #14575)
    Reported by: chris-mac
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-24 18:31:35 +00:00
Kevin P. Fleming
f43a65fd21 Merged revisions 208548 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r208548 | kpfleming | 2009-07-24 10:02:53 -0500 (Fri, 24 Jul 2009) | 8 lines
  
  Resolve a T.38 negotiation issue left over from the udptl-updates merge.
  
  The udptl-updates branch that was merged yesterday failed to properly send back
  T.38 SDP responses with the correct error correction mode, if the incoming SDP
  from the other end caused us to change error correction modes. This patch
  corrects that situation.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208549 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-24 15:04:31 +00:00
Michiel van Baak
84671eef56 Blocked revisions 208542 via svnmerge
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  r208542 | mvanbaak | 2009-07-24 16:35:49 +0200 (Fri, 24 Jul 2009) | 13 lines
  
  use aptitude for debian based systems
  
  The function to check wether we need to install packages was using
  dpkg-query which was gives wrong output on Debian 5
  
  Also, the apt-get has been replaced with aptitude because aptitude
  is now the preferred way to handle packages on Debian
  
  (closes issue #15570)
  Reported by: mvanbaak
  Patches:
        2009072400_installprereq-aptitude.diff uploaded by mvanbaak (license 7)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208543 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-24 14:36:32 +00:00
Kevin P. Fleming
560384cf09 Use correct formatting for T.38 change note in UPGRADE.txt
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208502 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-23 22:32:02 +00:00
Kevin P. Fleming
791d4f0478 Merged revisions 208464 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r208464 | kpfleming | 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) | 46 lines
  
  Rework of T.38 negotiation and UDPTL API to address interoperability problems
  
  Over the past couple of months, a number of issues with Asterisk
  negotiating (and successfully completing) T.38 sessions with various
  endpoints have been found. This patch attempts to address many of
  them, primarily focused around ensuring that the endpoints'
  MaxDatagram size is honored, and in addition by ensuring that T.38
  session parameter negotiation is performed correctly according to the
  ITU T.38 Recommendation.
  
  The major changes here are:
  
  1) T.38 applications in Asterisk (app_fax) only generate/receive IFP
  packets, they do not ever work with UDPTL packets. As a result of
  this, they cannot be allowed to generate packets that would overflow
  the other endpoints' MaxDatagram size after the UDPTL stack adds any
  error correction information. With this patch, the application is told
  the maximum *IFP* size it can generate, based on a calculation using
  the far end MaxDatagram size and the active error correction mode on
  the T.38 session. The same is true for sending *our* MaxDatagram size
  to the remote endpoint; it is computed from the value that the
  application says it can accept (for a single IFP packet) combined with
  the active error correction mode.
  
  2) All treatment of T.38 session parameters as 'capabilities' in
  chan_sip has been removed; these parameters are not at all like
  audio/video stream capabilities. There are strict rules to follow for
  computing an answer to a T.38 offer, and chan_sip now follows those
  rules, using the desired parameters from the application (or channel)
  that wants to accept the T.38 negotiation.
  
  3) chan_sip now stores and forwards ast_control_t38_parameters
  structures for tracking 'our' and 'their' T.38 session parameters;
  this greatly simplifies negotiation, especially for pass-through
  calls.
  
  4) Since T.38 negotiation without specifying parameters or receiving
  the final negotiated parameters is not very worthwhile, the
  AST_CONTROL_T38 control frame has been removed. A note has been added
  to UPGRADE.txt about this removal, since any out-of-tree applications
  that use it will no longer function properly until they are upgraded
  to use AST_CONTROL_T38_PARAMETERS.
  
  Review: https://reviewboard.asterisk.org/r/310/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208468 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-23 22:14:29 +00:00
Mark Michelson
db6c757a3d Merged revisions 208388 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r208388 | mmichelson | 2009-07-23 14:34:49 -0500 (Thu, 23 Jul 2009) | 24 lines
  
  Merged revisions 208386 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r208386 | mmichelson | 2009-07-23 14:24:21 -0500 (Thu, 23 Jul 2009) | 17 lines
    
    Fix a problem where a 491 response could be sent out of dialog.
    
    This generalizes the fix for issue 13849. The initial fix corrected the
    problem that Asterisk would reply with a 491 if a reinvite were received
    from an endpoint and we had not yet received an ACK from that endpoint
    for the initial INVITE it had sent us. This expansion also allows Asterisk
    to appropriately handle an INVITE with authorization credentials if Asterisk
    had not received an ACK from the previous transaction in which Asterisk had
    responded to an unauthorized INVITE with a 407.
    
    (closes issue #14239)
    Reported by: klaus3000
    Patches:
          14239.patch uploaded by mmichelson (license 60)
    Tested by: klaus3000
    	  
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208389 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-23 19:35:27 +00:00
Jeff Peeler
8d49fb4502 Merged revisions 208383 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r208383 | jpeeler | 2009-07-23 14:21:50 -0500 (Thu, 23 Jul 2009) | 12 lines
  
  Merged revisions 208380 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r208380 | jpeeler | 2009-07-23 14:19:53 -0500 (Thu, 23 Jul 2009) | 6 lines
    
    Only set the priindication setting when not performing a reload
    
    (closes issue #14696)
    Reported by: fdecher
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-23 19:23:33 +00:00
Mark Michelson
a9ad08042d Merged revisions 208314 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r208314 | mmichelson | 2009-07-23 11:29:37 -0500 (Thu, 23 Jul 2009) | 9 lines
  
  Merged revisions 208312 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r208312 | mmichelson | 2009-07-23 11:29:18 -0500 (Thu, 23 Jul 2009) | 3 lines
    
    Remove inaccurate XXX comment.
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-23 16:30:00 +00:00
Jeff Peeler
0bf989f1c0 Blocked revisions 208267 via svnmerge
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  r208267 | jpeeler | 2009-07-23 10:59:44 -0500 (Thu, 23 Jul 2009) | 13 lines
  
  Fix sending of interface identifier unconditionally in sig_pri
  
  The wrong logic was being used in chan_dahdi to convert a sig_pri_chan
  to the proper libpri channel number. The most significant bit must only
  be set only when trunk groups are being used.
  
  (closes issue #15452)
  Reported by: alecdavis
  Patches:
        bug15452.patch uploaded by jpeeler (license 325)
  Tested by: alecdavis
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-23 16:00:54 +00:00
Mark Michelson
0a6ccac217 Merged revisions 208263 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r208263 | mmichelson | 2009-07-23 10:46:34 -0500 (Thu, 23 Jul 2009) | 15 lines
  
  Merged revisions 208262 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r208262 | mmichelson | 2009-07-23 10:43:07 -0500 (Thu, 23 Jul 2009) | 8 lines
    
    Properly handle 183 responses which do not contain an SDP.
    
    (closes issue #15442)
    Reported by: ffloimair
    Patches:
          15442.patch uploaded by mmichelson (license 60)
    Tested by: tkarl, ffloimair
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208264 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-23 15:47:36 +00:00
Jeff Peeler
444475f1cc Blocked revisions 208155 via svnmerge
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  r208155 | jpeeler | 2009-07-22 17:42:33 -0500 (Wed, 22 Jul 2009) | 5 lines
  
  Reset the fax buffers back to default settings regardless of signaling in use -
  Pointed out by Matt F.
  Also in the case of not using a signaling module, set the law back to the
  default as well.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208157 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-22 22:46:59 +00:00
Jason Parker
fbe102e8f1 Blocked revisions 208113 via svnmerge
........
  r208113 | qwell | 2009-07-22 16:43:57 -0500 (Wed, 22 Jul 2009) | 9 lines
  
  Restore an int declaration on PPC platforms.
  
  This x is one crafty little bugger...
  It was used for 2 different things (one of which was only done on PPC) in 1.4.
  One of the uses were removed in trunk, and with it went the declaration.
  
  (closes issue #14038)
  Reported by: ffloimair
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208114 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-22 21:44:53 +00:00
Jeff Peeler
5c18c82487 Blocked revisions 207950 via svnmerge
........
  r207950 | jpeeler | 2009-07-21 17:51:47 -0500 (Tue, 21 Jul 2009) | 7 lines
  
  Do not dial digits when none were specified for sig_pri based calls
  
  (closes issue #15524)
  Reported by: elguero
  Patches:
        pri-sig-no-dest-set.patch uploaded by elguero (license 37)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-21 22:53:09 +00:00
Tilghman Lesher
7d73264772 Merged revisions 207946 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r207946 | tilghman | 2009-07-21 17:45:32 -0500 (Tue, 21 Jul 2009) | 15 lines
  
  Merged revisions 207945 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r207945 | tilghman | 2009-07-21 17:38:54 -0500 (Tue, 21 Jul 2009) | 8 lines
    
    Force an error if a blank is passed to QUOTE (because the documentation states the argument is not optional).
    This change makes URIENCODE and QUOTE behave similarly, since the documentation
    states that the argument is not optional, for both.
    (closes issue #15439)
     Reported by: pkempgen
     Patches: 
           20090706__issue15439.diff.txt uploaded by tilghman (license 14)
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-21 22:47:31 +00:00
Jeff Peeler
c62d1b0b58 Blocked revisions 207902 via svnmerge
........
  r207902 | jpeeler | 2009-07-21 17:02:25 -0500 (Tue, 21 Jul 2009) | 2 lines
  
  Fix my_is_off_hook to check rxbits only for FXS signaling
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207903 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-21 22:03:17 +00:00
Jeff Peeler
4ab9bff204 Merged revisions 207854 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r207854 | jpeeler | 2009-07-21 15:26:02 -0500 (Tue, 21 Jul 2009) | 16 lines
  
  Merged revisions 207827 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r207827 | jpeeler | 2009-07-21 15:16:55 -0500 (Tue, 21 Jul 2009) | 9 lines
    
    Wait for wink before dialing when using E&M wink signaling
    
    There was already code for other signaling types in dahdi_handle_event to
    handle dialing if a dial operation dial string was present. Simply add
    SIG_EMWINK to the list.
    
    (closes issue #14434)
    Reported by: araasch
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207860 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-21 20:27:47 +00:00
Jeff Peeler
ab6510ebf1 Revert r207636, this approach could potentially block for an unacceptable
amount of time.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-21 17:11:21 +00:00
Mark Michelson
51101e961b Merged revisions 207723 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r207723 | mmichelson | 2009-07-21 09:29:40 -0500 (Tue, 21 Jul 2009) | 11 lines
  
  Merged revisions 207714 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r207714 | mmichelson | 2009-07-21 09:26:00 -0500 (Tue, 21 Jul 2009) | 5 lines
    
    Document default timeout for AMI originations.
    
    AST-224
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207725 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-21 14:30:44 +00:00
Kevin P. Fleming
69255bd210 Merged revisions 207680 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r207680 | kpfleming | 2009-07-21 08:28:04 -0500 (Tue, 21 Jul 2009) | 18 lines
  
  Merged revisions 207647 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul 2009) | 12 lines
    
    Ensure that user-provided CFLAGS and LDFLAGS are honored.
    
    This commit changes the build system so that user-provided flags (in ASTCFLAGS
    and ASTLDFLAGS) are supplied to the compiler/linker *after* all flags provided
    by the build system itself, so that the user can effectively override the
    build system's flags if desired. In addition, ASTCFLAGS and ASTLDFLAGS can now
    be provided *either* in the environment before running 'make', or as variable
    assignments on the 'make' command line. As a result, the use of COPTS and LDOPTS
    is no longer necessary, so they are no longer documented, but are still supported
    so as not to break existing build systems that supply them when building Asterisk.
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207683 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-21 13:39:44 +00:00
Jeff Peeler
55a51b9194 Wait for wink before dialing when using E&M wink signaling
This patch adds a new dahdi_wait function to specifically wait for the wink
event. If the wink is not eventually received the channel is hung up. 

(closes issue #14434)
Reported by: araasch
Patches:
      emwinkmod uploaded by araasch (license 693)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207636 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-21 04:38:57 +00:00
Mark Michelson
935f33e481 Merged revisions 207424 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r207424 | mmichelson | 2009-07-20 14:48:12 -0500 (Mon, 20 Jul 2009) | 39 lines
  
  Merged revisions 207423 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r207423 | mmichelson | 2009-07-20 14:39:59 -0500 (Mon, 20 Jul 2009) | 33 lines
    
    Answer video SDP offers properly when videosupport is not enabled.
    
    Copied from Review board:
    
    In issue 12434, the reporter describes a situation in which audio and video 
    is offered on the call, but because videosupport is disabled in sip.conf, 
    Asterisk gives no response at all to the video offer. According to RFC 3264, 
    all media offers should have a corresponding answer. For offers we do not 
    intend to actually reply to with meaningful values, we should still reply 
    with the port for the media stream set to 0.
    
    In this patch, we take note of what types of media have been offered and 
    save the information on the sip_pvt. The SDP in the response will take into 
    account whether media was offered. If we are not otherwise going to answer 
    a media offer, we will insert an appropriate m= line with the port set to 0.
    
    It is important to note that this patch is pretty much a bandage being 
    applied to a broken bone. The patch *only* helps for situations where video 
    is offered but videosupport is disabled and when udptl_pt is disabled but 
    T.38 is offered. Asterisk is not guaranteed to respond to every media offer. 
    Notable cases are when multiple streams of the same type are offered. 
    The 2 media stream limit is still present with this patch, too.
    
    In trunk and the 1.6.X branches, things will be a bit different since Asterisk 
    also supports text in SDPs as well.
    
    (closes issue #12434)
    Reported by: mnnojd
    
    Review: https://reviewboard.asterisk.org/r/311
    Review: https://reviewboard.asterisk.org/r/313
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207425 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-20 19:55:28 +00:00
Russell Bryant
784e78e526 Merged revisions 207361 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r207361 | russell | 2009-07-20 11:36:15 -0500 (Mon, 20 Jul 2009) | 16 lines
  
  Merged revisions 207360 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r207360 | russell | 2009-07-20 11:26:24 -0500 (Mon, 20 Jul 2009) | 9 lines
    
    Only do the chan->fdno check in ast_read() in a developer build.
    
    I changed this check to only happen in a dev-mode build.  I also added a
    comment explaining what is going on.  I also made it so that detection of
    this situation does not affect ast_read() operation.
    
    (closes issue #14723)
    Reported by: seadweller
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207362 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-20 16:37:54 +00:00
Richard Mudgett
c305a6d0a3 Merged revisions 145293,158010 from
https://origsvn.digium.com/svn/asterisk/branches/1.4
to make merging easier.  These changes are already on trunk.

................
  r145293 | rmudgett | 2008-09-30 18:55:24 -0500 (Tue, 30 Sep 2008) | 54 lines

  channels/chan_misdn.c
  channels/misdn/isdn_lib.c
  *  Miscellaneous other fixes from trunk to make merging easier later.

  ........
  r145200 | rmudgett | 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines

  *  Miscellaneous formatting changes to make v1.4 and trunk
  more merge compatible in the mISDN area.

  channels/chan_misdn.c
  *  Eliminated redundant code in cb_events() EVENT_SETUP

  ........
  r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) | 9 lines

  improved helptext of misdn_set_opt.
  ........
  r142181 | rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line

  Cleaned up comment

  ........
  r138738 | rmudgett | 2008-08-18 16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines

  channels/chan_misdn.c
  *  Made bearer2str() use allowed_bearers_array[]
  *  Made use the causes.h defines instead of hardcoded numbers.
  *  Made use Asterisk presentation indicator values if either of the
  mISDN presentation or screen options are negative.
  *  Updated the misdn_set_opt application option descriptions.
  *  Renamed the awkward Caller ID presentation misdn_set_opt
  application option value not_screened to restricted.
  Deprecated the not_screened option value.

  channels/misdn/isdn_lib.c
  *  Made use the causes.h defines instead of hardcoded numbers.
  *  Fixed some spelling errors and typos.
  *  Added all defined facility code strings to fac2str().

  channels/misdn/isdn_lib.h
  *  Added doxygen comments to struct misdn_bchannel.

  channels/misdn/isdn_lib_intern.h
  *  Added doxygen comments to struct misdn_stack.

  channels/misdn_config.c
  configs/misdn.conf.sample
  *  Updated the mISDN presentation and screen parameter descriptions.

  doc/misdn.txt (doc/tex/misdn.tex)
  *  Updated the misdn_set_opt application option descriptions.
  *  Fixed some spelling errors and typos.
................
  r158010 | rmudgett | 2008-11-19 19:46:09 -0600 (Wed, 19 Nov 2008) | 9 lines

  Merged revision 157977 from
  https://origsvn.digium.com/svn/asterisk/team/group/issue8824

  ........
  Fixes JIRA ABE-1726

  The dial extension could be empty if you are using MISDN_KEYPAD
  to control ISDN provider features.
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207286 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-18 01:35:06 +00:00
Jeff Peeler
08fb833859 Merged revisions 207156 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r207156 | jpeeler | 2009-07-17 14:37:38 -0500 (Fri, 17 Jul 2009) | 14 lines
  
  Merged revisions 207155 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r207155 | jpeeler | 2009-07-17 14:36:19 -0500 (Fri, 17 Jul 2009) | 7 lines
    
    Fix format specifier to print out an unsigned long long.
    
    Yep, it's even ifdefed out code. But it made it to the RR list...
    
    (closes issue #14726)
    Reported by: lmadsen
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207157 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-17 19:38:54 +00:00
Jeff Peeler
21526941cb Merged revisions 207095 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r207095 | jpeeler | 2009-07-17 14:16:35 -0500 (Fri, 17 Jul 2009) | 2 lines
  
  Update some missing allowed options for overlapdial
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-17 19:17:33 +00:00
David Vossel
5f6fa4990f Merged revisions 207029 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r207029 | dvossel | 2009-07-17 12:51:44 -0500 (Fri, 17 Jul 2009) | 6 lines
  
  sip option flags handled incorrectly
  
  (closes issue #15376)
  Reported by: Takehiko Ooshima
  Tested by: dvossel, Takehiko_Ooshima
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207032 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-17 17:53:50 +00:00
Jeff Peeler
e577092f8e Blocked revisions 206998 via svnmerge
........
  r206998 | jpeeler | 2009-07-17 12:02:44 -0500 (Fri, 17 Jul 2009) | 14 lines
  
  Fix segfault in sig_analog when using callwaiting, respect callwaiting options
  
  Sig_analog handles allocating the sub channel for callwaiting, so no longer try
  to do it in chan_dahdi. Modified analog_alloc_sub to only mark the sub as
  allocated upon success of the alloc_sub callback, which was responsible for the
  segfault. Also, the callwaiting and callwaitingcallerid options were being
  unconditionally set to true. Now, the options are properly set from
  chan_dahdi.conf.
  
  (closes issue #15508)
  Reported by: elguero
  Tested by: elguero
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@206999 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-17 17:03:37 +00:00
David Vossel
263df0044d Merged revisions 206939 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r206939 | dvossel | 2009-07-17 11:13:22 -0500 (Fri, 17 Jul 2009) | 20 lines
  
  Merged revisions 206938 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r206938 | dvossel | 2009-07-17 11:05:06 -0500 (Fri, 17 Jul 2009) | 14 lines
    
    SIP incorrect From: header information when callpres is prohib
    
    Some ITSP make use of the "Anonymous" display name to detect a
    requirement to withhold caller id across the PSTN. This does
    not work if the display name is "Unknown".
    
    (closes issue #14465)
    Reported by: Nick_Lewis
    Patches:
          chan_sip.c-callerpres.patch uploaded by Nick (license 657)
          chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license 671)
    Tested by: Nick_Lewis, dvossel
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@206947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-17 16:18:49 +00:00
David Vossel
b66607d448 Blocked revisions 206877 via svnmerge
........
  r206877 | dvossel | 2009-07-16 16:45:14 -0500 (Thu, 16 Jul 2009) | 6 lines
  
  TIMEOUT(absolute) returned negative value.
  
  (closes issue #15513)
  Reported by: ys
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@206880 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-16 21:47:12 +00:00
David Vossel
b400eb240e Merged revisions 206873 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r206873 | dvossel | 2009-07-16 16:33:51 -0500 (Thu, 16 Jul 2009) | 12 lines
  
  Merged revisions 206872 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r206872 | dvossel | 2009-07-16 16:33:19 -0500 (Thu, 16 Jul 2009) | 6 lines
    
    error in iax.conf related IP-based access control
    
    (closes issue #15518)
    Reported by: pkempgen
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@206876 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-16 21:35:50 +00:00
David Vossel
7304dedfaf Merged revisions 206868 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r206868 | dvossel | 2009-07-16 16:25:22 -0500 (Thu, 16 Jul 2009) | 14 lines
  
  Merged revisions 206867 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r206867 | dvossel | 2009-07-16 16:24:16 -0500 (Thu, 16 Jul 2009) | 8 lines
    
    avoid segfault caused by user error
    
    If the CALLERPRES() dialplan function is set to nothing,
    a segfault occurs.  This is user error to begin with, but
    I'd rather see a cli warning message than have Asterisk
    crash on me.
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@206871 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-16 21:27:49 +00:00
Tilghman Lesher
5d94f8e6b9 Merged revisions 206808 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r206808 | tilghman | 2009-07-16 11:51:05 -0500 (Thu, 16 Jul 2009) | 13 lines
  
  Merged revisions 206807 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r206807 | tilghman | 2009-07-16 11:27:35 -0500 (Thu, 16 Jul 2009) | 6 lines
    
    Fix a memory leak.
    (closes issue #15517)
     Reported by: adomjan
     Patches: 
           func_realtime.c-ast_variable_destroy.diff uploaded by adomjan (license 487)
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@206809 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-16 16:52:52 +00:00
David Vossel
0faed3d459 Merged revisions 206768 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r206768 | dvossel | 2009-07-15 17:04:13 -0500 (Wed, 15 Jul 2009) | 8 lines
  
  Session timer were not activated if Supported header field in INVITE had both "timer" and other options.
  
  (closes issue #15403)
  Reported by: makoto
  Patches:
        sip-session-timer.patch uploaded by makoto (license
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@206775 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-15 22:06:36 +00:00
Jeff Peeler
a562b6afa9 Blocked revisions 206767 via svnmerge
........
  r206767 | jpeeler | 2009-07-15 17:02:55 -0500 (Wed, 15 Jul 2009) | 10 lines
  
  The dialing flag was mistakingly removed from sig_pri.
  
  This readds the proper setting of the flag and is really a continuation of
  r205731. The flag was being set properly in sig_analog, but use of the 
  newly added set_dialing callback allowed for some simplification in
  chan_dahdi.
  
  (closes issue #15486)
  Reported by: rmudgett
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@206769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-15 22:04:44 +00:00
Richard Mudgett
57f664c8f4 Merged revisions 206707 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r206707 | rmudgett | 2009-07-15 16:14:41 -0500 (Wed, 15 Jul 2009) | 33 lines
  
  Merged revisions 206706 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ................
    r206706 | rmudgett | 2009-07-15 15:44:55 -0500 (Wed, 15 Jul 2009) | 26 lines
    
    Merged revision 206700 from
    https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
    
    ..........
      Fixed chan_misdn crash because mISDNuser library is not thread safe.
    
      With Asterisk the mISDNuser library is driven by two threads concurrently:
      1. channels/misdn/isdn_lib.c::manager_event_handler()
      2. channels/misdn/isdn_lib.c::misdn_lib_isdn_event_catcher()
    
      Calls into the library are done concurrently and recursively from
      isdn_lib.c.
    
      Both threads can fiddle with the master/child layer3_proc_t lists.  One
      thread may traverse the list when the other interrupts it and then removes
      the list element which the first thread was currently handling.  This is
      exactly what caused the crash.  About 60 calls were needed to a Gigaset
      CX475 before it occurred once.
    
      This patch adds locking when calling into the mISDNuser library.
      This also fixes some cb_log calls with wrong port parameter.
    
      JIRA ABE-1913
          Patches: misdn-locking.patch (Modified with mostly cosmetic changes)
    ..........
  ................
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@206762 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-15 21:34:28 +00:00
David Vossel
f84624e23d Merged revisions 206702 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r206702 | dvossel | 2009-07-15 15:20:01 -0500 (Wed, 15 Jul 2009) | 10 lines
  
  callerid(num) is wrong when username is missing 
  
  A domain only sip uri <sip:123.123.123.123> would return
  123.123.123.123 as callid num.  Now, if the username is
  missing from a uri, the callerid num field is left empty.
  
  (closes issue #15476)
  Reported by: viraptor
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@206705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-15 20:21:34 +00:00
Sean Bright
d5a7745520 Merged revisions 206636 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r206636 | seanbright | 2009-07-15 12:00:24 -0400 (Wed, 15 Jul 2009) | 9 lines
  
  Merged revisions 206635 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r206635 | seanbright | 2009-07-15 11:57:51 -0400 (Wed, 15 Jul 2009) | 1 line
    
    Only print debug info in codec_dahdi if we are asking for it.
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@206637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-15 16:02:04 +00:00
Jeff Peeler
6468311645 Blocked revisions 206566 via svnmerge
........
  r206566 | jpeeler | 2009-07-14 15:01:10 -0500 (Tue, 14 Jul 2009) | 8 lines
  
  Restore some missing functionality to sig_analog.
  
  The main purpose of this commit is to restore missing functionality present in 
  the ss_thread before all the sig related work was done. Two of the biggest
  missing things were distinctive ring detection and cid handling for V23.
  fxsoffhookstate and associated mwi variables have been moved inside sig_analog
  as they were not being set properly as well.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@206600 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-14 20:36:44 +00:00
Tilghman Lesher
78917685dd Recorded merge of revisions 206567 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r206567 | tilghman | 2009-07-14 15:14:45 -0500 (Tue, 14 Jul 2009) | 6 lines
  
  Document all meetme realtime fields, and in the process, make some field lengths more consistent.
  (closes issue #15493)
   Reported by: lasko
   Patches: 
         meetme.diff uploaded by lasko (license 833)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@206585 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-14 20:22:27 +00:00
Richard Mudgett
587c202b8c Merged revisions 206489 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r206489 | rmudgett | 2009-07-14 12:01:48 -0500 (Tue, 14 Jul 2009) | 35 lines
  
  Merged revisions 206487 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r206487 | rmudgett | 2009-07-14 11:44:47 -0500 (Tue, 14 Jul 2009) | 28 lines
    
    Fixes several call transfer issues with chan_misdn.
    
    *  issue #14355 - Crash if attempt to transfer a call to an application.
    Masquerade the other pair of the four asterisk channels involved in the
    two calls.  The held call already must be a bridged call (not an
    applicaton) or it would have been rejected.
    
    *  issue #14692 - Held calls are not automatically cleared after transfer.
    Allow the core to initate disconnect of held calls to the ISDN port.  This
    also fixes a similar case where the party on hold hangs up before being
    transferred or taken off hold.
    
    *  JIRA ABE-1903 - Orphaned held calls left in music-on-hold.
    Do not simply block passing the hangup event on held calls to asterisk
    core.
    
    *  Fixed to allow held calls to be transferred to ringing calls.
    Previously, held calls could only be transferred to connected calls.
    *  Eliminated unused call states to simplify hangup code.
    *  Eliminated most uses of "holded" because it is not a word.
    
    (closes issue #14355)
    (closes issue #14692)
    Reported by: sodom
    Patches:
          misdn_xfer_v14_r205839.patch uploaded by rmudgett (license 664)
    Tested by: rmudgett
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@206555 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-14 18:17:15 +00:00
Russell Bryant
f54c70ea66 Merged revisions 206386 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r206386 | russell | 2009-07-14 09:51:44 -0500 (Tue, 14 Jul 2009) | 20 lines
  
  Merged revisions 206385 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ................
    r206385 | russell | 2009-07-14 09:48:00 -0500 (Tue, 14 Jul 2009) | 13 lines
    
    Merged revisions 206384 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.2
    
    ........
      r206384 | russell | 2009-07-14 09:45:47 -0500 (Tue, 14 Jul 2009) | 6 lines
      
      Ensure apathetic replies are sent out on the proper socket.
      
      chan_iax2 supports multiple address bindings.  The send_apathetic_reply()
      function did not attempt to send its response on the same socket that the
      incoming message came in on.
    ........
  ................
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@206387 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-14 14:54:47 +00:00
Richard Mudgett
3d3e165752 Merged revisions 206341 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r206341 | rmudgett | 2009-07-13 19:48:59 -0500 (Mon, 13 Jul 2009) | 11 lines
  
  Merged revisions 206284 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r206284 | rmudgett | 2009-07-13 19:17:28 -0500 (Mon, 13 Jul 2009) | 4 lines
    
    Fix some memory leaks in chan_misdn.
    
    JIRA ABE-1911
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@206369 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-14 01:25:27 +00:00
David Vossel
23705acc5e Merged revisions 205985 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r205985 | dvossel | 2009-07-10 16:42:10 -0500 (Fri, 10 Jul 2009) | 16 lines
  
  SIP register not using peer's outbound proxy
  
  If callbackextension is defined for a peer it successfully causes
  a registration to occur, but the registration ignores the
  outboundproxy settings for the peer.  This patch allows the
  peer to be passed to obproxy_get() in transmit_register().
  
  (closes issue #14344)
  Reported by: Nick_Lewis
  Patches:
        callbackextension_peer_trunk.diff uploaded by dvossel (license 671)
  Tested by: dvossel
  
  Review: https://reviewboard.asterisk.org/r/294/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@206017 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-10 22:50:51 +00:00
Kevin P. Fleming
61753d8e16 Merged revisions 205939 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r205939 | kpfleming | 2009-07-10 13:44:09 -0500 (Fri, 10 Jul 2009) | 1 line
  
  Update comments about the level of T.38 support in Asterisk.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205940 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-10 18:44:52 +00:00
Mark Michelson
d2c214e042 Fix build.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205880 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-10 17:44:34 +00:00
Mark Michelson
b3c7b4fa2d Merged revisions 205878 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r205878 | mmichelson | 2009-07-10 12:39:57 -0500 (Fri, 10 Jul 2009) | 30 lines
  
  Merged revisions 205877 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ................
    r205877 | mmichelson | 2009-07-10 12:39:13 -0500 (Fri, 10 Jul 2009) | 23 lines
    
    Merged revisions 205776 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/trunk
    
    ................
      r205776 | mmichelson | 2009-07-10 10:56:45 -0500 (Fri, 10 Jul 2009) | 16 lines
      
      Merged revisions 205775 via svnmerge from 
      https://origsvn.digium.com/svn/asterisk/branches/1.4
      
      ........
        r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul 2009) | 10 lines
        
        Ensure that outbound NOTIFY requests are properly routed through stateful proxies.
        
        With this change, we make note of Record-Route headers present in any SUBSCRIBE
        request that we receive so that our outbound NOTIFY requests will have the proper
        Route headers in them.
        
        (closes issue #14725)
        Reported by: ibc
      ........
    ................
  ................
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205879 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-10 17:42:19 +00:00
David Vossel
6e6557cb04 Merged revisions 205840 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r205840 | dvossel | 2009-07-10 11:42:04 -0500 (Fri, 10 Jul 2009) | 37 lines
  
  Merged revisions 205804 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r205804 | dvossel | 2009-07-10 11:23:59 -0500 (Fri, 10 Jul 2009) | 31 lines
    
    SIP registration auth loop caused by stale nonce
    
    If an endpoint sends two registration requests in a very short
    period of time with the same nonce, both receive 401 responses
    from Asterisk, each with a different nonce (the second 401
    containing the current nonce and the first one being stale).
    If the endpoint responds to the first 401, it does not match
    the current nonce so Asterisk sends a third 401 with a newly
    generated nonce (which updates the current nonce)... Now if
    the endpoint responds to the second 401, it does not match the
    current nonce either and Asterisk sends a fourth 401 with a
    newly generated nonce... This loop goes on and on.
    
    There appears to be a simple fix for this.  If the nonce from
    the request does not match our nonce, but is a good response
    to a previous nonce, instead of sending a 401 with a newly
    generated nonce, use the current one instead.  This breaks
    the loop as the nonce is not updated until a response is
    received. Additional logic has been added to make sure no
    nonce can be responded to twice though.
    
    (closes issue #15102)
    Reported by: Jamuel
    Patches:
          patch-bug_0015102 uploaded by Jamuel (license 809)
          nonce_sip.diff uploaded by dvossel (license 671)
    Tested by: Jamuel
    
    Review: https://reviewboard.asterisk.org/r/289/
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205843 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-10 16:48:56 +00:00
Mark Michelson
966a316fac Merged revisions 205776 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r205776 | mmichelson | 2009-07-10 10:56:45 -0500 (Fri, 10 Jul 2009) | 16 lines
  
  Merged revisions 205775 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul 2009) | 10 lines
    
    Ensure that outbound NOTIFY requests are properly routed through stateful proxies.
    
    With this change, we make note of Record-Route headers present in any SUBSCRIBE
    request that we receive so that our outbound NOTIFY requests will have the proper
    Route headers in them.
    
    (closes issue #14725)
    Reported by: ibc
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205777 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-10 15:57:08 +00:00
Kevin P. Fleming
2e5761d3cd Merged revisions 205770 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r205770 | kpfleming | 2009-07-10 10:28:11 -0500 (Fri, 10 Jul 2009) | 12 lines
  
  Fix some remaining T.38 negotiation problems in app_fax.
  
  Revision 205696 did not quite fix all the issues with the T.38 negotiation
  changes and app_fax; this patch corrects them, along with a couple of other
  minor issues.
  
  (closes issue #15480)
  Reported by: dimas
  Patches:
        test2-15480.patch uploaded by dimas (license 88)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205771 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-10 15:35:50 +00:00
Richard Mudgett
35dbf93676 Merged revisions 205728 via svn merge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r205728 | rmudgett | 2009-07-09 18:37:53 -0500 (Thu, 09 Jul 2009) | 21 lines
  
  No audio on calls from Asterisk to various ISDN devices until DTMF sent by caller.
  
  Add missing clearing of the dialing flag when the ISDN call is CONNECTED.
  (i.e. When libpri generates the event PRI_EVENT_ANSWER.)
  
  (closes issue #15420)
  Reported by: scottbmilne
  Patches:
        bug15420-1.4.25.1-diff2.txt uploaded by alecdavis (license 585)
  Tested by: scottbmilne, alecdavis
  
  (closes issue #15416)
  Reported by: avinoash
  
  (closes issue #15389)
  Reported by: alecdavis
  
  This patch should also fix the following issue:
  (issue #15205)
  Reported by: vinsik
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205729 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-09 23:46:22 +00:00
Kevin P. Fleming
b2e3c3e436 Merged revisions 205696 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r205696 | kpfleming | 2009-07-09 16:20:23 -0500 (Thu, 09 Jul 2009) | 16 lines
  
  Repair ability of SendFAX/ReceiveFAX to respond to T.38 switchover.
  
  Recent changes in T.38 negotiation in Asterisk caused these applications to
  not respond when the other endpoint initiated a switchover to T.38; this
  resulted in the T.38 switchover failing, and the FAX attempt to be made
  using an audio connection, instead of T.38 (which would usually cause the
  FAX to fail completely).
  
  This patch corrects this problem, and the applications will now correctly
  respond to the T.38 switchover request. In addition, the response will include
  the appopriate T.38 session parameters based on what the other end offered
  and what our end is capable of.
  
  (closes issue #14849)
  Reported by: afosorio
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205697 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-09 21:26:00 +00:00
David Vossel
d1fb490d7a Merged revisions 205600 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r205600 | dvossel | 2009-07-09 11:19:09 -0500 (Thu, 09 Jul 2009) | 9 lines
  
  Merged revisions 205599 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r205599 | dvossel | 2009-07-09 11:18:09 -0500 (Thu, 09 Jul 2009) | 2 lines
    
    Changing ast_samp2tv to not use floating point.
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205608 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-09 16:21:10 +00:00
David Vossel
f22cf5c484 Merged revisions 205479 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r205479 | dvossel | 2009-07-08 18:19:09 -0500 (Wed, 08 Jul 2009) | 16 lines
  
  Merged revisions 205471 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009) | 10 lines
    
    Fixes 8khz assumptions
    
    Many calculations assume 8khz is the codec rate. This
    is not always the case.  This patch only addresses chan_iax.c
    and res_rtp_asterisk.c, but I am sure there are other areas
    that make this assumption as well.
    
    Review: https://reviewboard.asterisk.org/r/306/
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205597 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-09 15:57:28 +00:00
Michiel van Baak
2e426f028e Blocked revisions 205562 via svnmerge
........
  r205562 | mvanbaak | 2009-07-09 16:10:01 +0200 (Thu, 09 Jul 2009) | 2 lines
  
  make this compile again under devmode
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205563 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-09 14:11:32 +00:00
Michiel van Baak
1ab060a770 Merged revisions 205532 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r205532 | mvanbaak | 2009-07-09 10:31:24 +0200 (Thu, 09 Jul 2009) | 5 lines
  
  pthread_self returns a pthread_t which is not an unsigned int on all
  pthread implementations. Casting it to an unsigned int fixes compiler warnings.
  
  Tested on OpenBSD and Linux both 32 and 64 bit
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205533 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-09 08:32:42 +00:00
David Vossel
308c62cf59 Merged revisions 205412 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r205412 | dvossel | 2009-07-08 17:15:06 -0500 (Wed, 08 Jul 2009) | 12 lines
  
  Merged revisions 205409 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009) | 6 lines
    
    moving ast_devstate_to_extenstate to pbx.c from devicestate.c
    
    ast_devstate_to_extenstate belongs in pbx.c.  This change
    fixes a compile time error with chan_vpb as well.
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205415 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-08 22:17:08 +00:00
Mark Michelson
f05e03dc9d Merged revisions 205350 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r205350 | mmichelson | 2009-07-08 14:26:55 -0500 (Wed, 08 Jul 2009) | 20 lines
  
  Merged revisions 205349 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r205349 | mmichelson | 2009-07-08 14:26:13 -0500 (Wed, 08 Jul 2009) | 14 lines
    
    Prevent phantom calls to queue members.
    
    If a caller were to hang up while a periodic announcement or position
    were being said, the return value for those functions would incorrectly
    indicate that the caller was still in the queue. With these changes,
    the problem does not occur.
    
    (closes issue #14631)
    Reported by: latinsud
    Patches:
          queue_announce_ghost_call2.diff uploaded by latinsud (license 745)
    	  (with small modification from me)
  ........
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2009-07-08 19:27:26 +00:00
Jason Parker
8fab346c00 Merged revisions 205291 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r205291 | qwell | 2009-07-08 13:19:46 -0500 (Wed, 08 Jul 2009) | 9 lines
  
  Merged revisions 205288 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r205288 | qwell | 2009-07-08 13:19:03 -0500 (Wed, 08 Jul 2009) | 1 line
    
    Update config.guess and config.sub from the savannah.gnu.org git repo.
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205296 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-08 18:20:45 +00:00
Tilghman Lesher
30f969fd6f oops, fixing build
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205224 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-08 17:01:02 +00:00
David Vossel
dedd3645ab Merged revisions 205216 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r205216 | dvossel | 2009-07-08 11:54:24 -0500 (Wed, 08 Jul 2009) | 17 lines
  
  Merged revisions 205215 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r205215 | dvossel | 2009-07-08 11:53:40 -0500 (Wed, 08 Jul 2009) | 10 lines
    
    ast_samp2tv needs floating point for 16khz audio
    
    In ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate is 16000.
    The .5 is currently stripped off because we don't calculate
    using floating points.  This causes madness with 16khz audio.
    
    (issue ABE-1899)
    
    Review: https://reviewboard.asterisk.org/r/305/
  ........
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2009-07-08 16:56:56 +00:00
Tilghman Lesher
177484b13d Merged revisions 205196 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r205196 | tilghman | 2009-07-08 11:27:50 -0500 (Wed, 08 Jul 2009) | 9 lines
  
  Merged revisions 205188 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r205188 | tilghman | 2009-07-08 11:26:15 -0500 (Wed, 08 Jul 2009) | 2 lines
    
    Add redirection warnings for the invalid language codes previously removed.
  ........
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2009-07-08 16:28:41 +00:00
Russell Bryant
c897ec65f0 Merged revisions 205151 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r205151 | russell | 2009-07-08 10:56:28 -0500 (Wed, 08 Jul 2009) | 2 lines
  
  Use tabs instead of spaces for indentation.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205152 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-08 15:56:55 +00:00
Russell Bryant
d38c8395b4 Merged revisions 205120 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r205120 | russell | 2009-07-08 10:17:19 -0500 (Wed, 08 Jul 2009) | 16 lines
  
  Move OpenSSL initialization to a single place, make library usage thread-safe.
  
  While doing some reading about OpenSSL, I noticed a couple of things that
  needed to be improved with our usage of OpenSSL.
  
  1) We had initialization of the library done in multiple modules.  This has now
     been moved to a core function that gets executed during Asterisk startup.
     We already link OpenSSL into the core for TCP/TLS functionality, so this
     was the most logical place to do it.
  
  2) OpenSSL is not thread-safe by default.  However, making it thread safe is
     very easy.  We just have to provide a couple of callbacks.  One callback
     returns a thread ID.  The other handles locking.  For more information,
     start with the "Is OpenSSL thread-safe?" question on the FAQ page of
     openssl.org.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205139 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-08 15:22:43 +00:00
David Vossel
6547190dd4 SIP Dialog ref counting
This patch adds reference counting for sip dialogs into 1.6.0.
When proc_session_timer() is called from the scheduler thread
it has no guarantee the session timer's dialog won't be freed
from underneath it.  Now the session timer holds a reference
to the dialog, preventing it from being destroyed during the
middle of proc_session_timer().

(closes issue #13623)
Reported by: Nik Soggia

Review: https://reviewboard.asterisk.org/r/302/



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205117 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-08 14:35:57 +00:00
Tilghman Lesher
c3b8c5ead3 Restore Hungarian (mistakenly removed during merge)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@204980 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-06 15:17:50 +00:00
Kevin P. Fleming
148695e367 Merged revisions 204948 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r204948 | kpfleming | 2009-07-06 08:38:29 -0500 (Mon, 06 Jul 2009) | 7 lines
  
  Improve handling of AST_CONTROL_T38 and AST_CONTROL_T38_PARAMETERS for non-T.38-capable channels.
  
  This change allows applications that request T.38 negotiation on a channel that
  does not support it to get the proper indication that it is not supported, rather
  than thinking that negotiation was started when it was not.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@204949 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-06 13:39:51 +00:00
Richard Mudgett
0a0144c4a0 Merged revisions 204835 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r204835 | rmudgett | 2009-07-02 17:01:28 -0500 (Thu, 02 Jul 2009) | 17 lines
  
  Merged revisions 204834 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r204834 | rmudgett | 2009-07-02 16:59:43 -0500 (Thu, 02 Jul 2009) | 10 lines
    
    Removed confusing warning message "Got Busy in Connected State"
    
    If an incoming mISDN call is answered with the Answer application and a
    subsequent Dial gets a busy endpoint then it is valid for that already
    connected channel to get the busy indication.  Asterisk will play the busy
    tones until the dialplan plays something else or hangs up the call.
    
    (closes issue #11974)
    Reported by: fvdb
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@204836 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-02 22:03:25 +00:00
David Vossel
d7306096a8 Merged revisions 204710 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r204710 | dvossel | 2009-07-02 11:03:44 -0500 (Thu, 02 Jul 2009) | 21 lines
  
  Merged revisions 204681 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009) | 14 lines
    
    Improved mapping of extension states from combined device states.
    
    This fixes a few issues with incorrect extension states and adds
    a cli command, core show device2extenstate, to display all possible
    state mappings.
    
    (closes issue #15413)
    Reported by: legart
    Patches:
          exten_helper.diff uploaded by dvossel (license 671)
    Tested by: dvossel, legart, amilcar
    
    Review: https://reviewboard.asterisk.org/r/301/
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@204754 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-02 18:07:35 +00:00
David Vossel
8d1643655b removes fake dialog_unref and dialog_ref function calls.
dialog_unref() and dialog_ref() in 1.6.0 where only place holders
for reference counting once it was implemented.  The functions
did nothing but return the pointer on ref and NULL on unref.  These
calls have been removed to make way for a patch that actually does
dialog ref counting.



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@204652 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-01 18:48:50 +00:00
Tilghman Lesher
c21a9fec34 Merged revisions 204563 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r204563 | tilghman | 2009-06-30 15:41:04 -0500 (Tue, 30 Jun 2009) | 13 lines
  
  Merged revisions 204556 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r204556 | tilghman | 2009-06-30 15:23:51 -0500 (Tue, 30 Jun 2009) | 6 lines
    
    More incorrect language codes, plus ensuring that regionalizations use the specified language, and not English for grammar.
    (closes issue #15022)
     Reported by: greenfieldtech
     Patches: 
           20090519__issue15022.diff.txt uploaded by tilghman (license 14)
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@204581 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-30 21:21:43 +00:00
Jason Parker
59d5bd11ce Merged revisions 204475 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r204475 | qwell | 2009-06-30 13:48:35 -0500 (Tue, 30 Jun 2009) | 9 lines
  
  Merged revisions 204474 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r204474 | qwell | 2009-06-30 13:47:06 -0500 (Tue, 30 Jun 2009) | 1 line
    
    Fix ast_say_counted_noun to correctly handle Polish.  Fix a comment typo in passing.
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@204476 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-30 18:50:50 +00:00
Tilghman Lesher
f2097d9072 Recorded merge of revisions 204470 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r204470 | tilghman | 2009-06-30 13:36:24 -0500 (Tue, 30 Jun 2009) | 18 lines
  
  Recorded merge of revisions 204469 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r204469 | tilghman | 2009-06-30 13:23:35 -0500 (Tue, 30 Jun 2009) | 11 lines
    
    "tw" is the language specification for Twi (from Ghana) not Taiwanese.
    (closes issue #15346)
     Reported by: volivier
     Patches: 
           20090617__issue15346__1.4.diff.txt uploaded by tilghman (license 14)
           20090617__issue15346__trunk.diff.txt uploaded by tilghman (license 14)
           20090617__issue15346__1.6.0.diff.txt uploaded by tilghman (license 14)
           20090617__issue15346__1.6.1.diff.txt uploaded by tilghman (license 14)
           20090617__issue15346__1.6.2.diff.txt uploaded by tilghman (license 14)
     Tested by: volivier
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@204471 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-30 18:44:16 +00:00
Mark Michelson
ae065d0125 Merged revisions 204301 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r204301 | mmichelson | 2009-06-29 17:50:35 -0500 (Mon, 29 Jun 2009) | 15 lines
  
  Merged revisions 204300 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r204300 | mmichelson | 2009-06-29 17:45:34 -0500 (Mon, 29 Jun 2009) | 9 lines
    
    Add error message so that it is clear why a SIP peer was not processed when
    a DNS lookup fails on a host or outboundproxy.
    
    (closes issue #13432)
    Reported by: p_lindheimer
    Patches:
          outboundproxy.patch uploaded by p (license 558)
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@204302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-29 22:52:39 +00:00
Mark Michelson
0889af49c6 Merged revisions 204247 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r204247 | mmichelson | 2009-06-29 16:48:54 -0500 (Mon, 29 Jun 2009) | 32 lines
  
  Merged revisions 204243,204246 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r204243 | mmichelson | 2009-06-29 16:23:43 -0500 (Mon, 29 Jun 2009) | 22 lines
    
    Fix a problem where chan_sip would ignore "old" but valid responses.
    
    chan_sip has had a problem for quite a long time that would manifest when
    Asterisk would send multiple SIP responses on the same dialog before receiving
    a response. The problem occurred because chan_sip only kept track of the highest
    outgoing sequence number used on the dialog. If Asterisk sent two requests out,
    and a response arrived for the first request sent, then Asterisk would ignore
    the response. The result was that Asterisk would continue retransmitting the
    requests and ignoring the responses until the maximum number of retransmissions
    had been reached.
    
    The fix here is to rearrange the code a bit so that instead of simply comparing
    the sequence number of the response to our latest outgoing sequence number, we
    walk our list of outstanding packets and determine if there is a match. If there is,
    we continue. If not, then we ignore the response.
    
    In doing this, I found a few completely useless variables that I have now removed.
    
    (closes issue #11231)
    Reported by: flefoll

    Review: https://reviewboard.asterisk.org/r/298
  ........
    r204246 | mmichelson | 2009-06-29 16:37:05 -0500 (Mon, 29 Jun 2009) | 3 lines
    
    Fix build oops.
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@204248 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-29 21:51:27 +00:00
Mark Michelson
0b144b325d Blocked revisions 204013 via svnmerge
................
  r204013 | mmichelson | 2009-06-29 10:04:39 -0500 (Mon, 29 Jun 2009) | 11 lines
  
  Blocked revisions 204012 via svnmerge
  
  ........
    r204012 | mmichelson | 2009-06-29 10:04:17 -0500 (Mon, 29 Jun 2009) | 6 lines
    
    Place unlock of mutex in an else block so that it does not get unlocked twice.
    
    (closes issue #15400)
    Reported by: aragon
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@204015 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-29 15:06:38 +00:00
Richard Mudgett
4730b56a50 Merged revisions 203909 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r203909 | rmudgett | 2009-06-26 20:07:52 -0500 (Fri, 26 Jun 2009) | 23 lines
  
  Merged revisions 203908 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r203908 | rmudgett | 2009-06-26 19:55:12 -0500 (Fri, 26 Jun 2009) | 16 lines
    
    The ISDN CPE side should not exclusively pick B channels normally.
    
    Before this patch, Asterisk unconditionally picked B channels exclusively
    on the CPE side and normally allowed alternative B channels on the network
    side.  Now Asterisk does the opposite.
    
    Reasons for the CPE side to normally not pick B channels exclusively:
    *  For CPE point-to-multipoint mode (i.e. phone side), the CPE side does
    not have enough information to exclusively pick B channels.  (There may be
    other devices on the line.)
    *  Q.931 gives preference to the network side picking B channels.
    *  Some telcos require the CPE side to not pick B channels exclusively.
    
    (closes issue #14383)
    Reported by: mbrancaleoni
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@203910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-27 01:14:18 +00:00
Jeff Peeler
21af4c4acb Merged revisions 203853 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r203853 | jpeeler | 2009-06-26 17:11:31 -0500 (Fri, 26 Jun 2009) | 12 lines
  
  Merged revisions 203848 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r203848 | jpeeler | 2009-06-26 17:09:19 -0500 (Fri, 26 Jun 2009) | 5 lines
    
    Make sure to recreate the dahdi pseudo channel after dahdi restart
    
    (closes issue #14477)
    Reported by: timking
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@203855 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-26 22:12:48 +00:00
Russell Bryant
fb2261d5a2 Merged revisions 203802 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r203802 | russell | 2009-06-26 16:21:48 -0500 (Fri, 26 Jun 2009) | 22 lines
  
  Merged revisions 203785 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r203785 | russell | 2009-06-26 16:16:39 -0500 (Fri, 26 Jun 2009) | 15 lines
    
    Don't fast forward past the end of a message.
    
    This is nice change for users of the voicemail application.  If someone gets a
    little carried away with fast forwarding through a message, they can easily
    get to the end and accidentally exit the voicemail application by hitting the
    fast forward key during the following prompt.
    
    This adds some safety by not allowing a fast forward past the end of a message.
    
    (closes issue #14554)
    Reported by: lacoursj
    Patches:
          21761.patch uploaded by lacoursj (license 707)
    Tested by: lacoursj
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@203818 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-26 21:25:52 +00:00
Russell Bryant
3be09ad7e9 Merged revisions 203779 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r203779 | russell | 2009-06-26 15:45:00 -0500 (Fri, 26 Jun 2009) | 5 lines
  
  Ensure the TCP read buffer is fully initialized before handling each packet.
  
  (closes issue #14452)
  Reported by: umberto71
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@203780 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-26 20:46:55 +00:00
David Brooks
3a8efb7b9f Merged revisions 203721 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r203721 | dbrooks | 2009-06-26 15:13:51 -0500 (Fri, 26 Jun 2009) | 16 lines
  
  Fixing voicemail's error in checking max silence vs min message length
  
  Max silence was represented in milliseconds, yet vmminsecs (minmessage) was represented
  as seconds.
  
  Also, the inequality was reversed. The warning, if triggered, was "Max silence should 
  be less than minmessage or you may get empty messages", which should have been logged 
  if max silence was greater than minmessage, but the check was for less than.
  
  Also, conforming if statement to coding guidelines.
  
  closes issue #15331)
  Reported by: markd
  
  Review: https://reviewboard.asterisk.org/r/293/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@203722 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-26 20:16:24 +00:00
Jeff Peeler
cb69985027 reverse whitespace change 203711 that was based on looking at sig_analog (which has about a 1000 line indentation change that is not worth doing here)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@203717 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-26 19:54:51 +00:00
David Vossel
9fa99fc5b7 Merged revisions 203710 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r203710 | dvossel | 2009-06-26 14:47:11 -0500 (Fri, 26 Jun 2009) | 7 lines
  
  moving debug message from level 0 to 1.
  
  (closes issue #15404)
  Reported by: leobrown
  Patches:
        iax_codec_debug.patch uploaded by leobrown (license 541)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@203716 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-26 19:49:33 +00:00
Jeff Peeler
a184348797 whitespace fix
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@203711 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-26 19:47:45 +00:00
Joshua Colp
fc33f7b57e Merged revisions 203699 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r203699 | file | 2009-06-26 16:27:24 -0300 (Fri, 26 Jun 2009) | 2 lines
  
  Improve T.38 negotiation by exchanging session parameters between application and channel.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@203701 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-26 19:29:02 +00:00
Jeff Peeler
7f0af9cf29 Merged revisions 203672 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r203672 | jpeeler | 2009-06-26 14:03:25 -0500 (Fri, 26 Jun 2009) | 16 lines
  
  Check if polarityonanswerdelay has elapsed before setting a channel as answered
  after a polarity reversal.
  
  Previously on a polarity switch event chan_dahdi would set the channel
  immediately as answered. This would cause problems if a polarity reversal
  occurred when the line was picked up as the dial would not have yet occurred. 
  Now if the polarity reversal occurs before delay has elapsed after coming off
  hook or an answer, it is ignored. Also, some refactoring was done in
  _handle_event.
  
  (closes issue #13917)
  Reported by: alecdavis
  Patches:
        chan_dahdi.bug13917.feb09.diff2.txt uploaded by alecdavis (license 585)
  Tested by: alecdavis
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@203698 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-26 19:25:51 +00:00
David Vossel
4406d80ba5 Merged revisions 203444 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r203444 | dvossel | 2009-06-25 16:45:32 -0500 (Thu, 25 Jun 2009) | 4 lines
  
  fixes a few redundant conditions
  
  (issue #15269)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@203447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-25 21:47:47 +00:00
Jeff Peeler
ea02e59776 Blocked revisions 203402 via svnmerge
........
  r203402 | jpeeler | 2009-06-25 16:22:12 -0500 (Thu, 25 Jun 2009) | 2 lines
  
  Remove some unnecessary code and update sample config file with respect to GR-303.
........


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2009-06-25 21:22:59 +00:00
Terry Wilson
2f088dcbe3 Merged revisions 203381 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r203381 | twilson | 2009-06-25 16:15:11 -0500 (Thu, 25 Jun 2009) | 11 lines
  
  Merged revisions 203380 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r203380 | twilson | 2009-06-25 16:13:10 -0500 (Thu, 25 Jun 2009) | 4 lines
    
    I didn't see that Mark already fixed the underlying issue!
    
    Yay for removing useless code.
  ........
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2009-06-25 21:18:21 +00:00
Russell Bryant
2c6cf6910f Merged revisions 203376 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r203376 | russell | 2009-06-25 16:04:55 -0500 (Thu, 25 Jun 2009) | 16 lines
  
  Merged revisions 203375 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r203375 | russell | 2009-06-25 16:02:18 -0500 (Thu, 25 Jun 2009) | 9 lines
    
    Fix a case where CDR answer time could be before the start time involving parking.
    
    (closes issue #13794)
    Reported by: davidw
    Patches:
          13794.patch uploaded by murf (license 17)
          13794.patch.160 uploaded by murf (license 17)
    Tested by: murf, dbrooks
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@203377 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-25 21:06:06 +00:00
Terry Wilson
ac944ba945 Blocked revisions 203338 via svnmerge
................
  r203338 | twilson | 2009-06-25 15:25:39 -0500 (Thu, 25 Jun 2009) | 9 lines
  
  Merged revisions 203311 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r203311 | twilson | 2009-06-25 15:09:15 -0500 (Thu, 25 Jun 2009) | 2 lines
    
    Don't try to free NULL
  ........
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2009-06-25 20:28:10 +00:00
Jeff Peeler
11467d7f0d Blocked revisions 203304 via svnmerge
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  r203304 | jpeeler | 2009-06-25 14:54:12 -0500 (Thu, 25 Jun 2009) | 6 lines
  
  New signaling module to handle PRI/BRI operations in chan_dahdi
  
  This merge splits the PRI/BRI signaling logic out of chan_dahdi.c into
  sig_pri.c. Functionality in theory should not change (mostly). A few trivial
  changes were made in sig_analog with verbose messages and commenting.
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2009-06-25 19:55:25 +00:00
Jason Parker
907bccd997 Blocked revisions 203258 via svnmerge
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  r203258 | qwell | 2009-06-25 14:22:46 -0500 (Thu, 25 Jun 2009) | 10 lines
  
  Unmute when we get a dtmfup (we muted on dtmfdown) event.
  
  This would occasionally cause one-way audio when using hardware DTMF detection.
  
  (closes issue #14761)
  Reported by: tzafrir
  Patches:
        v1-14761.patch uploaded by dimas (license 88)
  Tested by: tzafrir, dimas
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2009-06-25 19:27:23 +00:00
Mark Michelson
49bf75e7b2 Merged revisions 203231 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r203231 | mmichelson | 2009-06-25 13:53:12 -0500 (Thu, 25 Jun 2009) | 8 lines
  
  Blocked revisions 203230 via svnmerge
  
  ........
    r203230 | mmichelson | 2009-06-25 13:52:22 -0500 (Thu, 25 Jun 2009) | 3 lines
    
    Prevent false positives when freeing a NULL pointer with MALLOC_DEBUG enabled.
  ........
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2009-06-25 18:54:15 +00:00
Russell Bryant
8ee2d538bd Merged revisions 203116 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r203116 | russell | 2009-06-25 11:04:10 -0500 (Thu, 25 Jun 2009) | 18 lines
  
  Merged revisions 203115 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r203115 | russell | 2009-06-25 11:02:16 -0500 (Thu, 25 Jun 2009) | 11 lines
    
    Resolve a crash related to a T.38 reinvite race condition.
    
    This change resolves a crash observed locally during some T.38 testing.
    A call was set up using a call file, and when the T.38 reinvite came in,
    the channel state was still AST_STATE_DOWN.  The reason is explained by
    a comment in the code that previously lived in the handling of
    AST_STATE_RINGING.  This change modifies the logic to handle the same
    race condition for any channel state that is not UP.
    
    (closes ABE-1895)
  ........
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2009-06-25 16:05:36 +00:00
Richard Mudgett
db7b94fafb Merged revisions 203037 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r203037 | rmudgett | 2009-06-24 16:08:55 -0500 (Wed, 24 Jun 2009) | 15 lines
  
  Merged revisions 203036 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r203036 | rmudgett | 2009-06-24 16:01:43 -0500 (Wed, 24 Jun 2009) | 8 lines
    
    Improved chan_dahdi.conf pritimer error checking.
    
    Valid format is: pritimer=timer_name,timer_value
    
    *  Fixed segfault if the ',' is missing.
    *  Completely check the range returned by pri_timer2idx() to prevent
    possible access outside array bounds.
  ........
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2009-06-24 21:18:48 +00:00
Mark Michelson
a228e74faa Merged revisions 202967 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r202967 | mmichelson | 2009-06-24 13:29:10 -0500 (Wed, 24 Jun 2009) | 9 lines
  
  Merged revisions 202966 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r202966 | mmichelson | 2009-06-24 13:28:47 -0500 (Wed, 24 Jun 2009) | 3 lines
    
    Use the handy UNLINK macro instead of hand-coding the same thing in-line.
  ........
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2009-06-24 18:29:43 +00:00
Joshua Colp
03914b9a4a Merged revisions 202925 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r202925 | file | 2009-06-24 15:08:17 -0300 (Wed, 24 Jun 2009) | 2 lines
  
  Ensure the default settings are applied for T.38 when we set it up for a peer.
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2009-06-24 18:09:31 +00:00
Matthew Fredrickson
d9ab6e17c1 Merged revisions 202761 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r202761 | mattf | 2009-06-23 17:08:43 -0500 (Tue, 23 Jun 2009) | 1 line

I could have sworn I committed this patch ages ago, but... bug fix with setting NAI properly on linksets in certain situations.
........


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2009-06-23 22:09:53 +00:00
Ryan Brindley
663c38c02e Merged revisions 202753 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r202753 | rbrindley | 2009-06-23 16:25:17 -0500 (Tue, 23 Jun 2009) | 9 lines
  
  If we delete the info, lets also delete the lines
  
  (closes issue #14509)
  Reported by: timeshell
  Patches:
        20090504__bug14509.diff.txt uploaded by tilghman (license 14)
  Tested by: awk, timeshell
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@202754 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-23 21:26:47 +00:00
David Vossel
cbdbf23bdc Merged revisions 202672 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r202672 | dvossel | 2009-06-23 11:31:30 -0500 (Tue, 23 Jun 2009) | 18 lines
  
  Merged revisions 202671 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r202671 | dvossel | 2009-06-23 11:28:46 -0500 (Tue, 23 Jun 2009) | 12 lines
    
    MWI NOTIFY contains a wrong URI if Asterisk listens to non-standard port and transport
    
    (closes issue #14659)
    Reported by: klaus3000
    Patches:
          patch_chan_sip_fixMWIuri_1.4.txt uploaded by klaus3000 (license 65)
          mwi_port-transport_trunk.diff uploaded by dvossel (license 671)
    Tested by: dvossel, klaus3000
    
    Review: https://reviewboard.asterisk.org/r/288/
  ........
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2009-06-23 16:40:15 +00:00
Mark Michelson
b535dda70c Merged revisions 202603 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r202603 | mmichelson | 2009-06-23 10:23:00 -0500 (Tue, 23 Jun 2009) | 8 lines
  
  Blocked revisions 202601 via svnmerge
  
  ........
    r202601 | mmichelson | 2009-06-23 10:22:35 -0500 (Tue, 23 Jun 2009) | 3 lines
    
    Fix more memory leaks that may result if rtp is not successfully allocated.
  ........
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2009-06-23 15:25:03 +00:00
Mark Michelson
436dce6109 Recorded merge of revisions 202574 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r202574 | mmichelson | 2009-06-23 10:11:47 -0500 (Tue, 23 Jun 2009) | 8 lines
  
  Blocked revisions 202572 via svnmerge
  
  ........
    r202572 | mmichelson | 2009-06-23 10:08:27 -0500 (Tue, 23 Jun 2009) | 3 lines
    
    Fix potential memory leak in chan_sip when video rtp is not allocated properly.
  ........
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2009-06-23 15:16:06 +00:00
Mark Michelson
9341e2212c Recorded merge of revisions 202574 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r202574 | mmichelson | 2009-06-23 10:11:47 -0500 (Tue, 23 Jun 2009) | 8 lines
  
  Blocked revisions 202572 via svnmerge
  
  ........
    r202572 | mmichelson | 2009-06-23 10:08:27 -0500 (Tue, 23 Jun 2009) | 3 lines
    
    Fix potential memory leak in chan_sip when video rtp is not allocated properly.
  ........
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2009-06-23 15:15:30 +00:00
Russell Bryant
a03fe391fb Merged revisions 202497 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r202497 | russell | 2009-06-22 15:11:04 -0500 (Mon, 22 Jun 2009) | 11 lines
  
  Merged revisions 202496 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r202496 | russell | 2009-06-22 15:08:53 -0500 (Mon, 22 Jun 2009) | 4 lines
    
    Report CallerID change during a masquerade.
    
    Reported by: markster
  ........
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2009-06-22 20:12:29 +00:00
Sean Bright
5e6fbf81c7 Merged revisions 202417 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r202417 | seanbright | 2009-06-22 12:09:50 -0400 (Mon, 22 Jun 2009) | 4 lines
  
  Fix lock usage in cdr_sqlite3_custom to avoid potential crashes during reload.
  
  Pointed out by Russell while working on the CEL branch.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@202471 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-22 16:30:58 +00:00
Russell Bryant
eca12f6e13 Merged revisions 202415 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r202415 | russell | 2009-06-22 11:05:08 -0500 (Mon, 22 Jun 2009) | 9 lines
  
  Merged revisions 202414 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r202414 | russell | 2009-06-22 11:00:00 -0500 (Mon, 22 Jun 2009) | 2 lines
    
    Make Polycom subscription type override check more explicit.
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@202416 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-22 16:06:43 +00:00
David Vossel
9f41725a79 Blocked revisions 202410 via svnmerge
........
  r202410 | dvossel | 2009-06-22 10:33:35 -0500 (Mon, 22 Jun 2009) | 5 lines
  
  attempting to load running modules
  
  Modules placed in the priority heap for loading were not properly removed from the linked list.  This resulted in some modules attempting to load twice.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@202413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-22 15:42:34 +00:00
Mark Michelson
25b0edc60a Merged revisions 202343 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r202343 | mmichelson | 2009-06-22 09:58:24 -0500 (Mon, 22 Jun 2009) | 36 lines
  
  Merged revisions 202341-202342 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r202341 | mmichelson | 2009-06-22 09:42:55 -0500 (Mon, 22 Jun 2009) | 26 lines
    
    Fix a situation in which Asterisk would not stop retransmitting 487s.
    
    If a CANCEL were received by Asterisk, we would send a 487 in response
    to the original INVITE and a 200 OK for the CANCEL. If there were a network
    hiccup which caused the 200 OK and the 487 to be lost, then the UA communicating
    with Asterisk may try to retransmit its CANCEL. Asterisk's response to this used
    to be to try sending another 487 to the canceled INVITE and another 200 OK to the
    CANCEL.
    
    The problem here is that the originally-sent 487 was sent "reliably" meaning that
    it will be retransmitted until it is received properly. So when we receive the second
    CANCEL it is likely that the first batch of 487s we sent is still going strong and
    reaches the UA. The result was that the second set of 487s would be retransmitted
    constantly until the maximum number of retries had been reached.
    
    The fix for this is that if we receive a second CANCEL for an INVITE, then we cancel
    the retransmission of the first set of 487s and start a second set. This causes the
    dialog to be terminated reasonably.
    
    (closes issue #14584)
    Reported by: klaus3000
    Patches:
          14584_v2.patch uploaded by mmichelson (license 60)
    Tested by: klaus3000
  ........
    r202342 | mmichelson | 2009-06-22 09:44:58 -0500 (Mon, 22 Jun 2009) | 3 lines
    
    Remove an extra debug line left from previous commit.
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@202344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-22 15:05:00 +00:00
Mark Michelson
03f46e7a81 Merged revisions 202337 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r202337 | mmichelson | 2009-06-22 09:35:09 -0500 (Mon, 22 Jun 2009) | 31 lines
  
  Merged revisions 202336 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r202336 | mmichelson | 2009-06-22 09:34:05 -0500 (Mon, 22 Jun 2009) | 25 lines
    
    Fix a possible infinite loop in SDP parsing during glare situation.
    
    There was a while loop in get_ip_and_port_from_sdp which was controlled
    by a call to get_sdp_iterate. The loop would exit either if what we were
    searching for was found or if the return was NULL. The problem is that
    get_sdp_iterate never returns NULL. This means that if what we were searching
    for was not present, the loop would run infinitely. This modification of the
    loop fixes the problem.
    
    (closes issue #15213)
    Reported by: schmidts
    
    (closes issue #15349)
    Reported by: samy
    
    (closes issue #14464)
    Reported by: pj
    
    (closes issue #15345)
    Reported by: aragon
    Patches:
          sip_inf_loop.patch uploaded by mmichelson (license 60)
    Tested by: aragon
  ........
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2009-06-22 14:35:35 +00:00
Russell Bryant
c364f6af66 Merged revisions 202262 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r202262 | russell | 2009-06-21 11:11:48 -0500 (Sun, 21 Jun 2009) | 2 lines
  
  Fix possibility of crashiness during reload in custom fields handling.
........


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2009-06-21 16:14:05 +00:00
Russell Bryant
a6429678a3 Merged revisions 202258 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r202258 | russell | 2009-06-21 11:00:23 -0500 (Sun, 21 Jun 2009) | 2 lines
  
  Standardize return values of load_config() so reload() doesn't report an error on success.
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2009-06-21 16:02:50 +00:00
Sean Bright
eef330e9d9 Merged revisions 202183 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r202183 | seanbright | 2009-06-20 15:09:47 -0400 (Sat, 20 Jun 2009) | 5 lines
  
  Fix version detection for API changes in spandsp.
  
  (closes issue #15355)
  Reported by: deuffy
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2009-06-20 19:14:20 +00:00
Matthew Nicholson
93017afcc8 Added deadlock protection to try_suggested_sip_codec in chan_sip.c.
Review: https://reviewboard.asterisk.org/r/287/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@202006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-19 21:07:53 +00:00
David Vossel
ac6ab2899d Merged revisions 201994 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r201994 | dvossel | 2009-06-19 15:24:37 -0500 (Fri, 19 Jun 2009) | 14 lines
  
  Merged revisions 201993 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r201993 | dvossel | 2009-06-19 15:22:02 -0500 (Fri, 19 Jun 2009) | 8 lines
    
    timestamp was being converted to host order as a short rather than a long
    
    (closes issue #15361)
    Reported by: ffloimair
    Patches:
          ts_issue.diff uploaded by dvossel (license 671)
  ........
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2009-06-19 20:27:31 +00:00
Matthew Nicholson
74bc7c2a28 Blocked revisions 201717 via svnmerge
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  r201717 | mnicholson | 2009-06-18 12:41:09 -0500 (Thu, 18 Jun 2009) | 4 lines
  
  Added deadlock protection to try_suggested_sip_codec in chan_sip.c.
  
  Review: https://reviewboard.asterisk.org/r/285/
........


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2009-06-19 17:42:13 +00:00
Tilghman Lesher
fca68adb95 Merged revisions 201829 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r201829 | tilghman | 2009-06-18 19:43:41 -0500 (Thu, 18 Jun 2009) | 13 lines
  
  Merged revisions 201828 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r201828 | tilghman | 2009-06-18 19:40:41 -0500 (Thu, 18 Jun 2009) | 6 lines
    
    If the "h" extension fails, give it another chance in main/pbx.c.
    If the "h" extension fails, give it another chance in main/pbx.c, when it
    returns from the bridge code.  Fixes an issue where the "h" extension may
    occasionally not fire, when a Dial is executed from a Macro.
    Debugged in #asterisk with user tompaw.
  ........
................


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2009-06-19 00:44:18 +00:00
Tilghman Lesher
287972bc55 Merged revisions 201783 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r201783 | tilghman | 2009-06-18 15:52:36 -0500 (Thu, 18 Jun 2009) | 6 lines
  
  One of the changes in 1.6.1 was to allow app_directory to use functionality
  within app_voicemail for directory functions.  It is therefore no longer
  necessary for app_directory to be linked against the ODBC libraries (and it
  never was necessary for app_directory to be linked against IMAP, though it
  was).
........


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2009-06-18 20:59:16 +00:00
David Vossel
986be7d2a2 Merged revisions 201678 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r201678 | dvossel | 2009-06-18 11:37:42 -0500 (Thu, 18 Jun 2009) | 11 lines
  
  fixes some memory leaks and redundant conditions
  
  (closes issue #15269)
  Reported by: contactmayankjain
  Patches:
        patch.txt uploaded by contactmayankjain (license 740)
        memory_leak_stuff.trunk.diff uploaded by dvossel (license 671)
  Tested by: contactmayankjain, dvossel
........


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2009-06-18 16:58:03 +00:00
Russell Bryant
07a26c8744 Merged revisions 201610 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r201610 | russell | 2009-06-18 10:27:10 -0500 (Thu, 18 Jun 2009) | 36 lines
  
  Merged revisions 201600 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r201600 | russell | 2009-06-18 10:24:31 -0500 (Thu, 18 Jun 2009) | 29 lines
    
    Fix memory corruption and leakage related reloads of non files mode MoH classes.
    
    For Music on Hold classes that are not files mode, meaning that we are executing
    an application that will feed us audio data, we use a thread to monitor the
    external application and read audio from it.  This thread also makes use of the
    MoH class object.  In the MoH class destructor, we used pthread_cancel() to ask
    the thread to exit.  Unfortunately, the code did not wait to ensure that the
    thread actually went away.  What needed to be done is a pthread_join() to ensure
    that the thread fully cleans up before we proceed.  By adding this one line, we
    resolve two significant problems:
    
      1) Since the thread was never joined, it never fully goes away.  So, on every
         reload of non-files mode MoH, an unused thread was sticking around.
    
      2) There was a race condition here where the application monitoring thread
         could still try to access the MoH class, even though the thread executing
         the MoH reload has already destroyed it.
    
    (issue #15109)
    Reported by: jvandal
    
    (issue #15123)
    Reported by: axisinternet
    
    (issue #15195)
    Reported by: amorsen
    
    (issue AST-208)
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@201612 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-18 15:32:37 +00:00
David Vossel
b47546614f Blocked revisions 201570 via svnmerge
........
  r201570 | dvossel | 2009-06-18 10:16:05 -0500 (Thu, 18 Jun 2009) | 11 lines
  
  parsing extension correctly from sip register lines
  
  If a transport type was specified, but no extension, parsing of the extension would return whatever was after the transport rather than defaulting to 's'.
  
  (closes issue #15111)
  Reported by: ffs
  Patches:
        chan_sip.c_register-parser.patch uploaded by ffs (license 730)
  Tested by: ffs, dvossel
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@201611 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-18 15:29:35 +00:00
Mark Michelson
82f2aa293d Merged revisions 201462 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r201462 | mmichelson | 2009-06-17 15:10:01 -0500 (Wed, 17 Jun 2009) | 12 lines
  
  Fix problem with no audio due to ignoring the SDP.
  
  A recent change to our SDP version comparison made audio not function
  on some calls. This was because of a test wherein we were trying to
  see if an unsigned value was less than 0. This is a dumb comparison
  and arguably the compiler should have warned about it. Alas, though,
  it slipped past. Now it's fixed by changing the variable to be a
  signed type.
  
  Found by several developers. Tested by mnicholson and dbrooks.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@201463 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-17 20:10:50 +00:00
Mark Michelson
3068fa75a2 Merged revisions 201458 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r201458 | mmichelson | 2009-06-17 15:04:12 -0500 (Wed, 17 Jun 2009) | 15 lines
  
  Merged revisions 201450 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r201450 | mmichelson | 2009-06-17 14:59:31 -0500 (Wed, 17 Jun 2009) | 9 lines
    
    Change the datastore traversal in ast_do_masquerade to use a safe list traversal.
    
    It is possible for datastore fixup functions to remove the datastore from the list
    and free it. In particular, the queue_transfer_fixup in app_queue does this. While
    I don't yet know of this causing any crashes, it certainly could.
    
    Found while discussing a separate issue with Brian Degenhardt.
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@201459 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-17 20:05:09 +00:00
David Vossel
48dd606532 Blocked revisions 201453 via svnmerge
........
  r201453 | dvossel | 2009-06-17 15:00:51 -0500 (Wed, 17 Jun 2009) | 3 lines
  
  ast_channel_datastore_alloc is no longer used. updating datastores.txt to reflect that.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@201457 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-17 20:02:03 +00:00
David Vossel
86eaa43257 Merged revisions 201445 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r201445 | dvossel | 2009-06-17 14:45:35 -0500 (Wed, 17 Jun 2009) | 25 lines
  
  Merged revisions 201423 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r201423 | dvossel | 2009-06-17 14:28:12 -0500 (Wed, 17 Jun 2009) | 19 lines
    
    StopMixMonitor race condition (not giving up file immediately)
    
    StopMixMonitor only indicates to the MixMonitor thread to stop
    writing to the file.  It does not guarantee that the recording's
    file handle is available to the dialplan immediately after execution.
    This results in a race condition.  To resolve this, the filestream
    pointer is placed in a datastore on the channel. When StopMixMonitor
    is called, the datastore is retrieved from the channel and the
    filestream is closed immediately before returning to the dialplan.
    Documentation indicating the use of StopMixMonitor to free files
    has been updated as well.
    
    (closes issue #15259)
    Reported by: travisghansen
    Tested by: dvossel
    
    Review: https://reviewboard.asterisk.org/r/283/
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@201449 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-17 19:55:44 +00:00
David Brooks
ca7b9b9fe4 Merged revisions 201381 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r201381 | dbrooks | 2009-06-17 14:15:07 -0500 (Wed, 17 Jun 2009) | 16 lines
  
  Merged revisions 201380 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r201380 | dbrooks | 2009-06-17 13:45:50 -0500 (Wed, 17 Jun 2009) | 9 lines
    
    Checks for NULL sip_pvt pointer in chan_sip.c->acf_channel_read()
    
    Zombie channels could be passed, and chan_sip.c wasn't checking for it.
    Could crash Asterisk. Now checking for NULL pointer.
    
    (closes issue #15330)
    Reported by: okrief
    Tested by: dbrooks
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@201443 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-17 19:35:23 +00:00
David Vossel
ae2ea7cb34 Blocked revisions 201344 via svnmerge
........
  r201344 | dvossel | 2009-06-17 10:20:26 -0500 (Wed, 17 Jun 2009) | 16 lines
  
  SIP registry ref count error
  
  During a sip reload, the list of sip_registry objects are
  supposed to be traversed, unlinked, and destroyed, but
  destruction never takes place due to a ref counting error.
  This causes a memory leak when registry items are removed
  from sip.conf and reloaded.  While the registries are removed
  from the global list, they are not removed from the scheduler.
  Because of this, SIP register attempts continue to be sent
  out for the item even though it may no longer be in the .conf.
  
  (closes issue #15295)
  Reported by: amorsen
  
  Review: https://reviewboard.asterisk.org/r/282/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@201366 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-17 15:39:49 +00:00
Kevin P. Fleming
af930c11b3 Merged revisions 201262 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r201262 | kpfleming | 2009-06-17 07:04:17 -0500 (Wed, 17 Jun 2009) | 15 lines
  
  Merged revisions 201261 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r201261 | kpfleming | 2009-06-17 07:03:25 -0500 (Wed, 17 Jun 2009) | 9 lines
    
    Correct AST_LIST_APPEND_LIST behavior when list to be appended is empty.
    
    When the list to be appended is empty, and the list to be appended to is *not*,
    AST_LIST_APPEND_LIST would actually cause the target list to become broken,
    and no longer have a pointer to its last entry. This patch fixes the problem.
    
    (reported by Stanislaw Pitucha on the asterisk-dev mailing list)
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@201263 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-17 12:05:05 +00:00
David Vossel
6cbe57b730 Merged revisions 201223 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r201223 | dvossel | 2009-06-16 17:29:30 -0500 (Tue, 16 Jun 2009) | 2 lines
  
  fix issue with build_contact introduced by the "SIP trasnport type issues" commit
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@201226 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-16 22:31:42 +00:00
Kevin P. Fleming
968108c25c Merged revisions 201056,201090 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r201056 | kpfleming | 2009-06-16 13:54:30 -0500 (Tue, 16 Jun 2009) | 18 lines
  
  Merged revisions 200991 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun 2009) | 11 lines
    
    Improve support for media paths that can generate multiple frames at once.
    
    There are various media paths in Asterisk (codec translators and UDPTL, primarily)
    that can generate more than one frame to be generated when the application calling
    them expects only a single frame. This patch addresses a number of those cases,
    at least the primary ones to solve the known problems. In addition it removes the
    broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API
    functions, and cleans up various code paths affected by these changes.
    
    https://reviewboard.asterisk.org/r/175/
  ........
................
  r201090 | kpfleming | 2009-06-16 14:27:12 -0500 (Tue, 16 Jun 2009) | 5 lines
  
  Another minor fix to compiler attribute checking.
  
  Defaulting to 'static' for the function scope was bad... so remove it.
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@201093 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-16 19:34:39 +00:00
David Vossel
c2d79c89bb Merged revisions 200946 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r200946 | dvossel | 2009-06-16 11:03:30 -0500 (Tue, 16 Jun 2009) | 32 lines
  
  SIP transport type issues
  
  What this patch addresses:
  1. ast_sip_ouraddrfor() by default binds to the UDP address/port
  reguardless if the sip->pvt is of type UDP or not.  Now when no
  remapping is required, ast_sip_ouraddrfor() checks the sip_pvt's
  transport type, attempting to set the address and port to the
  correct TCP/TLS bindings if necessary.
  2.  It is not necessary to send the port number in the Contact
  header unless the port is non-standard for the transport type.
  This patch fixes this and removes the todo note.
  3.  In sip_alloc(), the default dialog built always uses transport
  type UDP.  Now sip_alloc() looks at the sip_request (if present)
  and determines what transport type to use by default.
  4.  When changing the transport type of a sip_socket, the file
  descriptor must be set to -1 and in some cases the tcptls_session's
  ref count must be decremented and set to NULL.  I've encountered
  several issues associated with this process and have created a function,
  set_socket_transport(), to handle the setting of the socket type.
  
  
  (closes issue #13865)
  Reported by: st
  Patches:
        dont_add_port_if_tls.patch uploaded by Kristijan (license 753)
        13865.patch uploaded by mmichelson (license 60)
        tls_port_v5.patch uploaded by vrban (license 756)
        transport_issues.diff uploaded by dvossel (license 671)
  Tested by: mmichelson, Kristijan, vrban, jmacz, dvossel
  
  Review: https://reviewboard.asterisk.org/r/278/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@200992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-16 17:11:51 +00:00
Kevin P. Fleming
64cfe299bd Merged revisions 200985 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r200985 | kpfleming | 2009-06-16 11:32:36 -0500 (Tue, 16 Jun 2009) | 7 lines
  
  Fix problems with new compiler attribute checking in configure script.
  
  The last changes to ast_gcc_attribute.m4 caused some problems checking for
  various attributes, because the scope of the symbol the attribute is applied
  to can be important; this patch allows the scope to be specified for the check.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@200986 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-16 16:34:03 +00:00
Michiel van Baak
6ba01d613f Merged revisions 200943 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r200943 | mvanbaak | 2009-06-16 17:51:36 +0200 (Tue, 16 Jun 2009) | 9 lines
  
  add FILE_STORAGE to Voicemail Build Options
  
  Voicemail can only use one storage module at the moment.
  Because it's unclear that selecting one of the storage modules
  in menuselect will disable filesystem storage we now have
  a FILE_STORAGE option that conflicts with the other modules.
  
  (closes issue #15333)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@200945 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-16 16:02:08 +00:00
Kevin P. Fleming
95157288a2 Merged revisions 200764 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r200764 | kpfleming | 2009-06-15 20:28:08 -0500 (Mon, 15 Jun 2009) | 11 lines
  
  Ensure that configure-script testing for compiler attributes actually works.
  
  The configure script tests for compiler attributes didn't actually enable
  enough warnings or provide a proper test harness to determine whether the 
  compiler supports the attribute in question or not; this caused gcc 4.1 to
  report that it supports 'weakref', but it doesn't actually support it in the
  way that is needed for our optional API mechanism. The new configure script
  test will properly distinguish between full support and partial support
  for this attribute, among others.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@200767 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-16 01:33:30 +00:00
Kevin P. Fleming
c1cc00fae6 Merged revisions 200726 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r200726 | kpfleming | 2009-06-15 20:03:22 -0500 (Mon, 15 Jun 2009) | 6 lines
  
  Document the new automatic 'ignoresdpversion' behavior.
  
  Asterisk will now automatically ignore incorrect incoming SDP version numbers
  when necessary to complete a T.38 re-INVITE operation.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@200729 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-16 01:08:56 +00:00
Kevin P. Fleming
40757d599e Merged revisions 165180,200689 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r165180 | mnicholson | 2008-12-17 12:49:12 -0600 (Wed, 17 Dec 2008) | 14 lines
  
  This patch adds a new 'ignoresdpversion' option to sip.conf.  When this is
  enabled (either globally or for a specific peer), chan_sip will treat any SDP
  data it receives as new data and update the media stream accordingly.  By
  default, Asterisk will only modify the media stream if the SDP session version
  received is different from the current SDP session version.  This option is
  required to interoperate with devices that have non-standard SDP session
  version implementations (observed by toc on the bug tracker with Microsoft OCS
  which always uses 0 as the session version).
  
  http://reviewboard.digium.com/r/94/
  (closes issue #13958)
  Reported by: toc
  Tested by: toc
........
  r200689 | kpfleming | 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) | 12 lines
  
  Accept T.38 re-INVITE responses with invalid SDP versions.
  
  This commit changes the 'incoming SDP version' check logic a bit more; when
  'ignoresdpversion' is *not* set for a peer, if we initiate a re-INVITE to
  switch to T.38, we'll always accept the peer's SDP response, even if they
  don't properly increment the SDP version number as they should. If this situation
  occurs, a warning message will be generated suggesting that the peer's
  configuration be changed to include the 'ignoresdpversion' configuration option
  (although ideally they'd fix their SIP implementation to be RFC compliant).
  
  AST-221
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@200724 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-15 21:29:27 +00:00
Mark Michelson
5f0b3e489f Merged revisions 200514 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r200514 | mmichelson | 2009-06-15 10:22:11 -0500 (Mon, 15 Jun 2009) | 11 lines
  
  Merged revisions 200513 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r200513 | mmichelson | 2009-06-15 10:21:46 -0500 (Mon, 15 Jun 2009) | 5 lines
    
    Add INFO to our allowed methods so that endpoints know they may send it to us.
    
    AST-223
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@200515 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-15 15:22:34 +00:00
Mark Michelson
660bceff3c Merged revisions 200361 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r200361 | mmichelson | 2009-06-12 14:07:51 -0500 (Fri, 12 Jun 2009) | 16 lines
  
  Merged revisions 200360 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r200360 | mmichelson | 2009-06-12 14:06:41 -0500 (Fri, 12 Jun 2009) | 10 lines
    
    Suppress a warning message and give a better return code when generating
    inband ringing after a call is answered.
    
    (closes issue #15158)
    Reported by: madkins
    Patches:
          15158.patch uploaded by mmichelson (license 60)
    Tested by: madkins
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@200362 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-12 19:08:15 +00:00
Sean Bright
a5c9e82ebb Merged revisions 199781 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r199781 | seanbright | 2009-06-09 14:08:53 -0400 (Tue, 09 Jun 2009) | 2 lines
  
  Fix all of the parallel build warnings issued when running make -j#.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@200228 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-11 22:42:05 +00:00
Mark Michelson
bd9f6cf82d Merged revisions 200146 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r200146 | mmichelson | 2009-06-11 16:17:14 -0500 (Thu, 11 Jun 2009) | 5 lines
  
  Fix a crash due to a potentially NULL p->options.
  
  Thanks to mnicholson for pointing it out.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@200149 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-11 21:18:03 +00:00
Leif Madsen
6651984931 Merged revisions 200039 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r200039 | lmadsen | 2009-06-11 08:15:09 -0400 (Thu, 11 Jun 2009) | 8 lines
  
  Fix path for .flavor and .version
  
  (issue #14737)
  Reported by: davidw
  Patches:
        flavor.patch uploaded by davidw (license 780)
  Tested by: davidw
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@200040 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-11 12:16:12 +00:00
David Brooks
3516eaa7e0 Fixes the argument order in definition of new_find_extension().
In the definition of new_find_extension(), the arguments 'callerid' and
'label' were swapped. The prototype declaration and all calls to the
function are ordered 'callerid' then 'label', but the function itself
was ordered 'label' then 'callerid'.

(closes issue #15303)
Reported by: JimDickenson


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@199994 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-10 20:29:52 +00:00
Mark Michelson
7a3b46c789 The 1.6.0 branch was missing all invite_branch logic. It has now been added.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@199975 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-10 20:20:53 +00:00
Mark Michelson
8bb3dcacad Blocked revisions 199958 via svnmerge
........
  r199958 | mmichelson | 2009-06-10 15:15:48 -0500 (Wed, 10 Jun 2009) | 6 lines
  
  Only try to use the invite_branch on outgoing INVITEs with auth credentials.
  
  I have added a comment to the code to help ease understanding of the logic here
  as well.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@199967 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-10 20:18:58 +00:00
Sean Bright
1e91c9f3bc Merged revisions 199857 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r199857 | seanbright | 2009-06-10 12:10:23 -0400 (Wed, 10 Jun 2009) | 9 lines
  
  Merged revisions 199856 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r199856 | seanbright | 2009-06-10 12:08:35 -0400 (Wed, 10 Jun 2009) | 2 lines
    
    __WORDSIZE is not available on all platforms, so use sizeof(void *) instead.
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@199858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-10 16:13:14 +00:00
David Vossel
6cab2f47e6 Merged revisions 199818 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r199818 | dvossel | 2009-06-09 15:47:57 -0500 (Tue, 09 Jun 2009) | 11 lines
  
  CLI NOTIFY sending wrong transport type.
  
  SIP's cli NOTIFY command only used UDP rather than copying the transport type from the peer.
  
  (closes issue #15283)
  Reported by: jthurman
  Patches:
        sip-notify-tcp-svn199728.patch uploaded by jthurman (license 614)
  Tested by: jthurman, dvossel
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@199821 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-09 20:54:10 +00:00
David Vossel
36fe89c775 Blocked revisions 199743 via svnmerge
........
  r199743 | dvossel | 2009-06-09 11:22:04 -0500 (Tue, 09 Jun 2009) | 11 lines
  
  module load priority
  
  This patch adds the option to give a module a load priority. The value represents the order in which a module's load() function is initialized.  The lower the value, the higher the priority.  The value is only checked if the AST_MODFLAG_LOAD_ORDER flag is set.  If the AST_MODFLAG_LOAD_ORDER flag is not set, the value will never be read and the module will be given the lowest possible priority
  on load.  Since some modules are reliant on a timing interface, the timing modules have been given a high load priorty.
  
  (closes issue #15191)
  Reported by: alecdavis
  Tested by: dvossel
  
  Review: https://reviewboard.asterisk.org/r/262/
........


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2009-06-09 16:34:19 +00:00
Sean Bright
046332e03e Merged revisions 199630 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r199630 | seanbright | 2009-06-08 15:33:09 -0400 (Mon, 08 Jun 2009) | 32 lines
  
  Merged revisions 199626,199628 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun 2009) | 21 lines
    
    Increase the size of our thread stack on 64 bit processors.
    
    We were setting the stack size for each thread to 240KB regardless of
    architecture, which meant that in some scenarios we actually had less available
    stack space on 64 bit processors (pointers use 8 bytes instead of 4).  So now we
    calculate the stack size we reserve based on the platform's __WORDSIZE, which
    gives us:
    
         32 bit -> 240KB
         64 bit -> 496KB
        128 bit -> 1008KB (that's right, we're ready for 128 bit processors)
    
    Patch typed by me but written by several members of #asterisk-dev, including
    Kevin, Tilghman, and Qwell.
    
    (closes issue #14932)
    Reported by: jpiszcz
    Patches:
          06052009_issue14932.patch uploaded by seanbright (license 71)
    Tested by: seanbright
  ........
    r199628 | seanbright | 2009-06-08 15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines
    
    Fix a typo in the stack size calculation just introduced.
  ........
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2009-06-08 19:39:22 +00:00
Mark Michelson
c1bece3429 Blocked revisions 199588 via svnmerge
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  r199588 | mmichelson | 2009-06-08 12:32:04 -0500 (Mon, 08 Jun 2009) | 9 lines
  
  Fix a deadlock that could occur when setting rtp stats on SIP calls.
  
  (closes issue #15143)
  Reported by: cristiandimache
  Patches:
        15143.patch uploaded by mmichelson (license 60)
  Tested by: cristiandimache
........


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2009-06-08 17:33:17 +00:00
David Vossel
8e3df8bd1f Merged revisions 199298 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r199298 | dvossel | 2009-06-05 16:21:22 -0500 (Fri, 05 Jun 2009) | 21 lines
  
  Merged revisions 199297 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009) | 14 lines
    
    Fixes issue with hints giving unexpected results.
    
    Hints with two or more devices that include ONHOLD gave unexpected results.
    
    (closes issue #15057)
    Reported by: p_lindheimer
    Patches:
          onhold_trunk.diff uploaded by dvossel (license 671)
          pbx.c.1.4.patch uploaded by p (license 558)
          devicestate.c.trunk.patch uploaded by p (license 671)
    Tested by: p_lindheimer, dvossel
    
    Review: https://reviewboard.asterisk.org/r/254/
  ........
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2009-06-05 21:37:01 +00:00
Mark Michelson
29113e4ad5 Merged revisions 199227 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r199227 | mmichelson | 2009-06-05 08:51:08 -0500 (Fri, 05 Jun 2009) | 14 lines
  
  Correct "dahdi show channels" output when specifying a group.
  
  Since a DAHDI channel may belong to multiple groups, we need to use
  a bitwise and instead of equivalence to determine whether to display
  the channel information.
  
  
  (closes issue #15248)
  Reported by: gentian
  Patches:
        15248.patch uploaded by mmichelson (license 60)
  Tested by: gentian
........


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2009-06-05 13:51:26 +00:00
David Vossel
ea62e16ffe Merged revisions 199139 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r199139 | dvossel | 2009-06-04 14:10:16 -0500 (Thu, 04 Jun 2009) | 9 lines
  
  Merged revisions 199138 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r199138 | dvossel | 2009-06-04 14:00:15 -0500 (Thu, 04 Jun 2009) | 3 lines
    
    Additional updates to AST-2009-001
  ........
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2009-06-04 19:16:58 +00:00
Sean Bright
8340a34a4a Merged revisions 199051 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r199051 | seanbright | 2009-06-04 10:31:24 -0400 (Thu, 04 Jun 2009) | 47 lines
  
  Merged revisions 199022 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun 2009) | 40 lines
    
    Safely handle AMI connections/reload requests that occur during startup.
    
    During asterisk startup, a lock on the list of modules is obtained by the
    primary thread while each module is initialized.  Issue 13778 pointed out a
    problem with this approach, however.  Because the AMI is loaded before other
    modules, it is possible for a module reload to be issued by a connected client
    (via Action: Command), causing a deadlock.
    
    The resolution for 13778 was to move initialization of the manager to happen
    after the other modules had already been lodaded.  While this fixed this
    particular issue, it caused a problem for users (like FreePBX) who call AMI
    scripts via an #exec in a configuration file (See issue 15189).
    
    The solution I have come up with is to defer any reload requests that come in
    until after the server is fully booted.  When a call comes in to
    ast_module_reload (from wherever) before we are fully booted, the request is
    added to a queue of pending requests.  Once we are done booting up, we then
    execute these deferred requests in turn.
    
    Note that I have tried to make this a bit more intelligent in that it will not
    queue up more than 1 request for the same module to be reloaded, and if a
    general reload request comes in ('module reload') the queue is flushed and we
    only issue a single deferred reload for the entire system.
    
    As for how this will impact existing installations - Before 13778, a reload
    issued before module initialization was completed would result in a deadlock.
    After 13778, you simply couldn't connect to the manager during startup (which
    causes problems with #exec-that-calls-AMI configuration files).  I believe this
    is a good general purpose solution that won't negatively impact existing
    installations.
    
    (closes issue #15189)
    (closes issue #13778)
    Reported by: p_lindheimer
    Patches:
          06032009_15189_deferred_reloads.diff uploaded by seanbright (license 71)
    Tested by: p_lindheimer, seanbright
    
    Review: https://reviewboard.asterisk.org/r/272/
  ........
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2009-06-04 14:53:49 +00:00
Sean Bright
4b0840b16a Blocked revisions 198958 via svnmerge
................
  r198958 | seanbright | 2009-06-03 16:49:11 -0400 (Wed, 03 Jun 2009) | 17 lines
  
  Blocked revisions 198957 via svnmerge
  
  ........
    r198957 | seanbright | 2009-06-03 16:39:10 -0400 (Wed, 03 Jun 2009) | 11 lines
    
    Fix a possible crash in pbx_spool.
    
    We were trying to reference members of a struct that had previously been freed.
    This patch makes sure that we free the struct after it has been removed from
    the spooler queue.
    
    (closes issue #15072)
    Reported by: garlew
    Patches:
          spool.diff uploaded by garlew (license 376)
  ........
................


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2009-06-03 20:51:44 +00:00
David Vossel
9c6652d306 Merged revisions 198856 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r198856 | dvossel | 2009-06-02 16:17:49 -0500 (Tue, 02 Jun 2009) | 10 lines
  
  Generic call forward api, ast_call_forward()
  
  The function ast_call_forward() forwards a call to an extension specified in an ast_channel's call_forward string.  After an ast_channel is called, if the channel's call_forward string is set this function can be used to forward the call to a new channel and terminate the original one.  I have included this api call in both channel.c's ast_request_and_dial() and feature.c's feature_request_and_dial().  App_dial and app_queue already contain call forward logic specific for their application and options.
  
  (closes issue #13630)
  Reported by: festr
  
  Review: https://reviewboard.asterisk.org/r/271/
........


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2009-06-03 15:27:30 +00:00
David Vossel
d1b1506bc5 Merged revisions 198824 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r198824 | dvossel | 2009-06-02 12:55:35 -0500 (Tue, 02 Jun 2009) | 8 lines
  
  fixes issue with channels not going down after transfer
  
  Iax2 currently does not support native bridging if the timeoutms value is set.  We check for that in iax2_bridge, but then set timeoutms to 0 by default.  If the timeoutms is not provided it is set to -1. By setting timeoutms to 0 it is processed causing a bridging retry loop.
  
  (closes issue #15216)
  Reported by: oxymoron
  Tested by: dvossel
........


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2009-06-02 17:56:19 +00:00
Joshua Colp
bc1b330dec Merged revisions 198791 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r198791 | file | 2009-06-02 10:48:06 -0300 (Tue, 02 Jun 2009) | 5 lines
  
  Correct documentation for the register line, specifically where the domain should be specified.
  
  (closes issue #14367)
  Reported by: Nick_Lewis
........


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2009-06-02 13:49:24 +00:00
Tilghman Lesher
9a2dd22314 Merged revisions 198626 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r198626 | tilghman | 2009-06-01 13:40:35 -0500 (Mon, 01 Jun 2009) | 2 lines
  
  Add information for new meetme realtime fields
........


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2009-06-01 18:42:25 +00:00
Sean Bright
e9dea62cd7 Merged revisions 198375 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r198375 | seanbright | 2009-05-30 16:11:33 -0400 (Sat, 30 May 2009) | 13 lines
  
  Properly terminate the receive buffer before sending to iksemel.
  
  aji_io_recv takes the maximum number of bytes to read (instead of the total
  buffer size), so we have to subtract 1 from our buffer size.  Without this, when
  we receive packets that are larger than our buffer, iksemel will choke and
  things get wonky.
  
  (closes issue #15232)
  Reported by: lp0
  Patches:
        05302009_res_jabber.c.patch uploaded by seanbright (license 71)
  Tested by: seanbright, lp0
........


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2009-05-30 20:19:27 +00:00
Sean Bright
6b2d977b04 Merged revisions 198371 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r198371 | seanbright | 2009-05-30 15:38:58 -0400 (Sat, 30 May 2009) | 19 lines
  
  Merged revisions 198370 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r198370 | seanbright | 2009-05-30 15:36:20 -0400 (Sat, 30 May 2009) | 12 lines
    
    Properly terminate AMI JabberSend response messages.
    
    The response message (either Error or Success) needs an extra trailing \r\n
    after the fields to inform the client that the message is complete.
    
    (closes issue #14876)
    Reported by: srt
    Patches:
          05302009_1.4_res_jabber.c.diff uploaded by seanbright (license 71)
          asterisk_14876.patch uploaded by srt (license 378)
          trunk-14876-2.diff uploaded by phsultan (license 73)
  ........
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2009-05-30 19:40:16 +00:00
Russell Bryant
5a1f34576d Merged revisions 198312 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r198312 | russell | 2009-05-29 22:43:23 -0500 (Fri, 29 May 2009) | 12 lines
  
  Merged revisions 198311 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r198311 | russell | 2009-05-29 22:42:46 -0500 (Fri, 29 May 2009) | 5 lines
    
    Fix a crash that occurred when MWI SMDI messages expired.
    
    (closes issue #14561)
    Reported by: cmoss28
  ........
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2009-05-30 03:48:37 +00:00
Sean Bright
dfe2793610 Merged revisions 198285 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r198285 | seanbright | 2009-05-29 23:26:06 -0400 (Fri, 29 May 2009) | 15 lines
  
  Merged revisions 198251 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r198251 | seanbright | 2009-05-29 22:46:41 -0400 (Fri, 29 May 2009) | 8 lines
    
    Treat an empty FORWARD_CONTEXT the same way we treat a missing one.
    
    (closes issue #15056)
    Reported by: p_lindheimer
    Patches:
          05292009_bug15056.diff uploaded by seanbright (license 71)
    Tested by: p_lindheimer
  ........
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2009-05-30 03:27:31 +00:00
Jeff Peeler
9bbc25b099 Blocked revisions 198088 via svnmerge
........
  r198088 | jpeeler | 2009-05-29 14:19:51 -0500 (Fri, 29 May 2009) | 9 lines
  
  New signaling module to handle analog operations in chan_dahdi
  
  This branch splits all the analog signaling logic out of chan_dahdi.c into
  sig_analog.c. Functionality in theory should not change at all. As noted
  in the code, there is still some unused code remaining that will be cleaned
  up in a later commit.
  
  Review: https://reviewboard.asterisk.org/r/253/
........


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2009-05-29 19:53:47 +00:00
Matthew Nicholson
95fac13256 Merged revisions 198072 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r198072 | mnicholson | 2009-05-29 14:04:24 -0500 (Fri, 29 May 2009) | 21 lines
  
  Merged revisions 198068 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May 2009) | 15 lines
    
    Use AST_CDR_NOANSWER instead of AST_CDR_NULL as the default CDR disposition.
    
    This change also involves the addition of an AST_CDR_FLAG_ORIGINATED flag that is used on originated channels to distinguish: them from dialed channels.
    
    (closes issue #12946)
    Reported by: meral
    Patches:
          null-cdr2.diff uploaded by mnicholson (license 96)
    Tested by: mnicholson, dbrooks
    
    (closes issue #15122)
    Reported by: sum
    Tested by: sum
  ........
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2009-05-29 19:13:03 +00:00
Sean Bright
1f3bacb4c3 Merged revisions 198000 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r198000 | seanbright | 2009-05-29 14:15:15 -0400 (Fri, 29 May 2009) | 15 lines
  
  Merged revisions 197998 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r197998 | seanbright | 2009-05-29 14:14:12 -0400 (Fri, 29 May 2009) | 8 lines
    
    Fix 'make config' target for Slackware.
    
    There was a missing semi-colon after the echo statement in the Makefile that was
    causing problems for some users.  Fix suggested by reporter.
    
    (closes issue #15225)
    Reported by: pdavis
  ........
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2009-05-29 18:16:49 +00:00
Leif Madsen
9cb98c991c Update MixMonitor documentation.
Updated the MixMonitor documentation for the 'b' option so that
it is more obvious that you must not optimize awat the Local
channel when using this option.

(issue #14829)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@197896 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28 23:58:36 +00:00
Terry Wilson
951e170004 Blocked revisions 197738 via svnmerge
........
  r197738 | twilson | 2009-05-28 14:57:18 -0500 (Thu, 28 May 2009) | 19 lines
  
  Add Calendaring support for Asterisk
  
  This commit add Calendaring support to Asterisk for iCalendar, CalDAV, and MS
  Exchange calendars. Exchange support has only been tested on Exchange Server 2k3
  and does not support forms-based authentication at this time (patches *very*
  welcome). Exchange support is also currently missing the ability to return a
  list of a meting's attendees (again, patches are very, very welcome).
  
  Features include:
    Querying a calendar for events over a specific time range
    Checking a calendar's busy status via the dialplan
    Writing calendar events via the dialplan (CalDAV and Exchange only)
    Handling calendar event notifications through the dialplan
  
  (closes issue #14771)
  Tested by: lmadsen, twilson, Shivaprakash
  
  Review: https://reviewboard.asterisk.org/r/58
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2009-05-28 20:50:01 +00:00
Eliel C. Sardanons
93e30e3e23 Merged revisions 197621 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r197621 | eliel | 2009-05-28 12:01:48 -0400 (Thu, 28 May 2009) | 19 lines
  
  Merged revisions 197562 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r197562 | eliel | 2009-05-28 11:21:32 -0400 (Thu, 28 May 2009) | 13 lines
    
    Use the address we already know when reloading a peer with nat=yes.
    
    If we already have an address for a peer, and we are reloading the sip
    configuration, try to use that address to contact the peer, instead of
    getting it from the Contact.
    
    (closes issue #15194)
    Reported by: ibc
    Patches:
          sip.patch uploaded by eliel (license 64)
          Tested by: manwe
  ........
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2009-05-28 19:00:21 +00:00
Joshua Colp
b422316b51 Merged revisions 197697 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r197697 | file | 2009-05-28 15:45:11 -0300 (Thu, 28 May 2009) | 2 lines
  
  Fix a bug where the trunkmtu setting was not set to the default value of 1240 on load but was on reload.
........


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2009-05-28 18:46:42 +00:00
David Vossel
6c05ae7cc1 'iax show peer blah' now outputs whether or not peer 'blah' is in trunk mode or not.
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2009-05-28 16:06:05 +00:00
Mark Michelson
a66b938920 Merged revisions 197606 via svnmerge from
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................
  r197606 | mmichelson | 2009-05-28 10:32:19 -0500 (Thu, 28 May 2009) | 22 lines
  
  Recorded merge of revisions 197588 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r197588 | mmichelson | 2009-05-28 10:27:49 -0500 (Thu, 28 May 2009) | 16 lines
    
    Allow for media to arrive from an alternate source when responding to a reinvite with 491.
    
    When we receive a SIP reinvite, it is possible that we may not be able to process the
    reinvite immediately since we have also sent a reinvite out ourselves. The problem is
    that whoever sent us the reinvite may have also sent a reinvite out to another party,
    and that reinvite may have succeeded.
    
    As a result, even though we are not going to accept the reinvite we just received, it
    is important for us to not have problems if we suddenly start receiving RTP from a new
    source. The fix for this is to grab the media source information from the SDP of the
    reinvite that we receive. This information is passed to the RTP layer so that it will
    know about the alternate source for media.
    
    Review: https://reviewboard.asterisk.org/r/252
  ........
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2009-05-28 15:34:48 +00:00
Mark Michelson
c7731d3489 Merged revisions 197543 via svnmerge from
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................
  r197543 | mmichelson | 2009-05-28 09:58:06 -0500 (Thu, 28 May 2009) | 27 lines
  
  Merged revisions 197537 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May 2009) | 21 lines
    
    Add flags to chanspy audiohook so that audio stays in sync.
    
    There are two flags being added to the chanspy audiohook here. One
    is the pre-existing AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set,
    we ensure that the read and write slinfactories on the audiohook do
    not skew beyond a certain tolerance.
    
    In addition, there is a new audiohook flag added here,
    AST_AUDIOHOOK_SMALL_QUEUE. With this flag set, we do not allow for
    a slinfactory to build up a substantial amount of audio before 
    flushing it. For this particular issue, this means that the person 
    spying on the call will hear the conversations in real time with very 
    little delay in the audio.
    
    (closes issue #13745)
    Reported by: geoffs
    Patches:
          13745.patch uploaded by mmichelson (license 60)
    Tested by: snblitz
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@197544 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28 15:03:55 +00:00
Joshua Colp
5d28a72130 Merged revisions 197538 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r197538 | file | 2009-05-28 11:51:43 -0300 (Thu, 28 May 2009) | 5 lines
  
  Fix a bug in stringfields where it did not actually free the pools of memory.
  
  (closes issue #15074)
  Reported by: pj
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@197539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28 14:52:39 +00:00
Joshua Colp
cd950bcbf8 Merged revisions 197467 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r197467 | file | 2009-05-28 10:47:45 -0300 (Thu, 28 May 2009) | 15 lines
  
  Merged revisions 197466 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r197466 | file | 2009-05-28 10:44:58 -0300 (Thu, 28 May 2009) | 8 lines
    
    Fix a bug where the flag indicating the presence of rport would get overwritten by the nat setting.
    
    The presence of rport is now stored as a separate flag. Once the dialog is setup and authenticated
    (or it passes through unauthenticated) the proper nat flag is set.
    
    (closes issue #13823)
    Reported by: dimas
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@197469 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28 13:50:19 +00:00
Gavin Henry
4f54d18c57 issue #15155 and issue #15156 from trunk
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@197439 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28 11:40:15 +00:00
Sean Bright
e8e4bf2100 Merged revisions 197260 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r197260 | seanbright | 2009-05-27 16:08:16 -0400 (Wed, 27 May 2009) | 6 lines
  
  Use bash explicitly when calling build_tools/mkpkgconfig from the Makefile.
  
  Since we use bashisms in build_tools/mkpkgconfig, we should call on bash
  explicitly when running from the Makefile, otherwise we get errors during a
  'make install.'
  
  (closes issue #15209)
  Reported by: seandarcy
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@197261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-27 20:10:17 +00:00
Tilghman Lesher
ae41d65286 Recorded merge of revisions 197209 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r197209 | tilghman | 2009-05-27 14:20:56 -0500 (Wed, 27 May 2009) | 12 lines
  
  Recorded merge of revisions 197194 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r197194 | tilghman | 2009-05-27 14:09:42 -0500 (Wed, 27 May 2009) | 5 lines
    
    Use a different determinator on whether to print the delimiter, since leading fields may be blank.
    (closes issue #15208)
     Reported by: ramonpeek
     Patch by me, though inspired in part by a patch from ramonpeek
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@197244 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-27 19:29:48 +00:00
Jeff Peeler
76b9a6b5d8 Fix broken attended transfers
The bridge was terminating immediately after the attended transfer was 
completed. The problem was because upon reentering ast_channel_bridge
nexteventts was checked to see if it was set and if so could possibly
return AST_BRIDGE_COMPLETE.
  
(closes issue #15183)
Reported by: andrebarbosa
Tested by: andrebarbosa, tootai, loloski


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@197126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-27 17:09:25 +00:00
Sean Bright
9654401867 Merged revisions 197089 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r197089 | seanbright | 2009-05-27 12:07:57 -0400 (Wed, 27 May 2009) | 6 lines
  
  Fix references to /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf in
  the sample configuration files.
  
  (closes issue #15207)
  Reported by: seandarcy
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@197090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-27 16:11:14 +00:00
Leif Madsen
7fe3f9692a Change register format example to match wording.
This change does not affect any other 1.6 branches as they have
already been updated for other changes, which uses the word 'domain'
as I have here.

(closes issue #15204)
Reported by: okrief

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@197088 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-27 16:04:56 +00:00
David Vossel
2b5d7b9ce7 Fixes merge issue during r196454.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@197086 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-27 15:58:24 +00:00
Sean Bright
6aa494b8f2 Merged revisions 196988 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r196988 | seanbright | 2009-05-27 09:02:54 -0400 (Wed, 27 May 2009) | 9 lines
  
  Display an error message when chan_alsa fails to load due to a missing
  or inaccessible configuration file.
  
  Before this change, when chan_alsa failed to load due to a missing or
  inaccessible configuration file, no message would be displayed.  With this
  change, when chan_alsa fails to load due to a missing or inaccessible
  configuration file, a message will be displayed.
  
  (closes issue #14760)
  Reported by: Nick_Lewis
  Patches:
        chan_alsa.c-confload.patch uploaded by Nick (license 657)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@196989 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-27 13:04:10 +00:00
Russell Bryant
dd5e00ccac Merged revisions 196843 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r196843 | russell | 2009-05-26 13:20:57 -0500 (Tue, 26 May 2009) | 16 lines
  
  Merged revisions 196826 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r196826 | russell | 2009-05-26 13:14:36 -0500 (Tue, 26 May 2009) | 9 lines
    
    Resolve a file handle leak.
    
    The frames here should have always been freed.  However, out of luck, there was
    never any memory leaked.  However, after file streams became reference counted,
    this code would leak the file stream for the file being read.
    
    (closes issue #15181)
    Reported by: jkroon
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@196865 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-26 18:25:47 +00:00
Joshua Colp
6c1703877b Merged revisions 196721 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r196721 | file | 2009-05-26 10:43:13 -0300 (Tue, 26 May 2009) | 7 lines
  
  Fix a bug where the sip unregister CLI command did not completely unregister the peer.
  
  (closes issue #15118)
  Reported by: alecdavis
  Patches:
        chan_sip_unregister.diff2.txt uploaded by alecdavis (license 585)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@196722 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-26 13:44:47 +00:00
Joshua Colp
4e3fd1fc02 Merged revisions 196658 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r196658 | file | 2009-05-26 10:06:50 -0300 (Tue, 26 May 2009) | 14 lines
  
  Merged revisions 196657 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r196657 | file | 2009-05-26 10:06:09 -0300 (Tue, 26 May 2009) | 7 lines
    
    Remove some bash specific stuff from safe_asterisk.
    
    (closes issue #10812)
    Reported by: paravoid
    Patches:
          safe_asterisk_bashism.diff uploaded by tzafrir (license 46)
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@196659 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-26 13:07:29 +00:00
David Vossel
5249104890 Merged revisions 196416 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r196416 | dvossel | 2009-05-22 16:09:45 -0500 (Fri, 22 May 2009) | 19 lines
  
  SIP set outbound transport type from Registration
  
  In sip.conf the transport option allows for the configuration of what transport types (udp, tcp, and tls) a peer will accept, but only the first type listed was used for outbound connections.  This patch changes this.  Now the default transport type is only used until the peer registers.  When registration takes place the transport type is parsed out of the Contact header.  If the Contact header's transport type is equal to one that the peer supports, the peer's default transport type for outbound connections is set to match the Contact header's type.  If the Contact header's transport type is not present, then the peer's default transport type is set to match the one the peer registered with.  When a peer unregisters or the registration expires, the default transport type for that peer is reset.
  
  (closes issue #12282)
  Reported by: rjain
  Patches:
        reg_patch_1.diff uploaded by dvossel (license 671)
  Tested by: dvossel
  
  (closes issue #14727)
  Reported by: pj
  Patches:
        reg_patch_3.diff uploaded by dvossel (license 671)
  Tested by: pj, dvossel
  
  Review: https://reviewboard.asterisk.org/r/249/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@196454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-22 22:51:09 +00:00
Joshua Colp
9495819c2e Merged revisions 196117 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r196117 | file | 2009-05-22 10:56:47 -0300 (Fri, 22 May 2009) | 12 lines
  
  Merged revisions 196116 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r196116 | file | 2009-05-22 10:54:17 -0300 (Fri, 22 May 2009) | 5 lines
    
    Fix a bug where using immediate with mISDN caused a cause code of 16 to get sent back instead of 1 if the 's' extension did not exist.
    
    (closes issue #12286)
    Reported by: lmamane
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@196118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-22 13:57:56 +00:00
David Vossel
be7700852b Merged revisions 195995 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r195995 | dvossel | 2009-05-21 14:11:49 -0500 (Thu, 21 May 2009) | 20 lines
  
  Merged revisions 195991 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r195991 | dvossel | 2009-05-21 14:04:56 -0500 (Thu, 21 May 2009) | 14 lines
    
    Sign problem calculating timestamp for iax frame leads to no audio on the receiving peer.
    
    There are rare cases in which a frame's delivery timestamp is slightly less than the iax2_pvt's offset.  This causes the pvt's timestamp to be a small negative number, but since the timestamp value is unsigned it looks like a huge positive number.  This patch checks for this negative case and sets the ms to zero.  A similar check is already done right below this one in the 'else' statement.
    
    (closes issue #15032)
    Reported by: guillecabeza
    Patches:
          chan_iax2.c.patch_timestamp uploaded by guillecabeza (license 380)
    Tested by: guillecabeza
    
    (closes issue #14216)
    Reported by: Andrey Sofronov
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@195997 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-21 19:12:55 +00:00
Matthew Nicholson
e5d7298863 Merged revisions 195882 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r195882 | mnicholson | 2009-05-21 10:33:55 -0500 (Thu, 21 May 2009) | 20 lines
  
  Merged revisions 195881 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r195881 | mnicholson | 2009-05-21 10:25:50 -0500 (Thu, 21 May 2009) | 13 lines
    
    This commit prevents cdr records with AST_CDR_FLAG_ANSLOCKED and AST_CDR_FLAG_LOCKED from being updated in certain cases.
    
    This is accomplished by adding two functions to update the answer time and disposition of calls that checks for the proper lock flags.  These functions are used in the ast_bridge_call() function so that ForkCDR(A) calls are respected.
    
    This patch also modifies the way ast_bridge_call() chooses the cdr record to base the bridged_cdr on.  Previously the first unlocked cdr record would be chosen, now instead the first cdr record is chosen and forked cdr records are moved to the bridge_cdr.  This allows the original cdr record and any forked cdr records to be properly updated with answer and end times.
    
    (closes issue #13797)
    Reported by: sh0t
    Tested by: sh0t
    
    (closes issue #14744)
    Reported by: deepesh
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@195894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-21 16:19:24 +00:00
Tilghman Lesher
7f5817fbcb Merged revisions 195839 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r195839 | tilghman | 2009-05-20 18:30:05 -0500 (Wed, 20 May 2009) | 3 lines
  
  If a variable had a blank value upon the initial setting, then it would do nothing.
  Identified by Dmitry Andrianov via private email, fixed by me.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@195840 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-20 23:31:09 +00:00
Joshua Colp
4c8cf3398e Merged revisions 195698 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r195698 | file | 2009-05-20 14:33:02 -0300 (Wed, 20 May 2009) | 12 lines
  
  Merged revisions 195688 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r195688 | file | 2009-05-20 14:30:25 -0300 (Wed, 20 May 2009) | 5 lines
    
    Fix some code that wrongly assumed a pointer would always be non-NULL when dealing with CDRs after a bridge.
    
    (closes issue #15079)
    Reported by: barryf
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@195702 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-20 17:33:49 +00:00
Joshua Colp
be7149e2c8 Merged revisions 195636 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r195636 | file | 2009-05-20 14:14:42 -0300 (Wed, 20 May 2009) | 12 lines
  
  Merged revisions 195635 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r195635 | file | 2009-05-20 14:14:00 -0300 (Wed, 20 May 2009) | 5 lines
    
    Fix a bug where the MeetMe option 'D' did not actually prompt for the pin.
    
    (closes issue #15050)
    Reported by: pmhaddad
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@195637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-20 17:15:23 +00:00
Tilghman Lesher
b3e97c0ac5 Merged revisions 195521 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r195521 | tilghman | 2009-05-19 15:16:01 -0500 (Tue, 19 May 2009) | 14 lines
  
  Merged revisions 195520 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r195520 | tilghman | 2009-05-19 15:12:20 -0500 (Tue, 19 May 2009) | 7 lines
    
    Ensure thread keys are initialized before attempting to access them.
    (closes issue #14889)
     Reported by: jaroth
     Patches: 
           app_voicemail.c.patch uploaded by msirota (license 758)
     Tested by: msirota, BlargMaN
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@195522 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-19 20:17:10 +00:00
Joshua Colp
0b6e79502e Merged revisions 195449 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r195449 | file | 2009-05-19 11:43:54 -0300 (Tue, 19 May 2009) | 14 lines
  
  Merged revisions 195448 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r195448 | file | 2009-05-19 11:41:45 -0300 (Tue, 19 May 2009) | 7 lines
    
    Fix a bug where direct RTP setup would partially occur even when disabled if the calling channel was answered.
    
    (issue #13545)
    Reported by: davidw
    (issue #14244)
    Reported by: mbnwa
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@195450 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-19 14:45:54 +00:00
Eliel C. Sardanons
91b11ad2a3 Merged revisions 195369 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r195369 | eliel | 2009-05-18 16:49:20 -0400 (Mon, 18 May 2009) | 8 lines
  
  Fix the CLI command 'manager show command' documentation and functionality.
  
  The CLI command 'manager show command' supports passing multiple action names in
  the same line, but it was not allowing that because of a incorrect check in the
  argumentes counter. Also the documentation was updated to show that this usage
  of the command is possible.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@195442 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-18 21:37:00 +00:00
Tilghman Lesher
4892be94a6 Recorded merge of revisions 195370 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r195370 | tilghman | 2009-05-18 15:52:33 -0500 (Mon, 18 May 2009) | 15 lines
  
  Recorded merge of revisions 195366 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r195366 | tilghman | 2009-05-18 15:24:13 -0500 (Mon, 18 May 2009) | 8 lines
    
    Add a similar dependency on SMDI for voicemail as already exists for ADSI.
    (closes issue #14846)
     Reported by: pj
     Patches: 
           20090413__bug14846__1.4.diff.txt uploaded by tilghman (license 14)
           20090507__issue14846__1.6.0.diff.txt uploaded by tilghman (license 14)
           20090507__issue14846__1.6.1.diff.txt uploaded by tilghman (license 14)
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@195371 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-18 20:53:37 +00:00
Tilghman Lesher
8d65e37144 Merged revisions 195320 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r195320 | tilghman | 2009-05-18 14:17:15 -0500 (Mon, 18 May 2009) | 9 lines
  
  Move the spawn of astcanary down, until after the call to daemon(3).
  This avoids possible conflicts with the internal implementation of
  daemon(3).
  (closes issue #15093)
   Reported by: tzafrir
   Patches: 
         20090513__issue15093__2.diff.txt uploaded by tilghman (license 14)
   Tested by: tzafrir
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@195357 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-18 19:27:24 +00:00
Mark Michelson
e759722828 Merged revisions 195316 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r195316 | mmichelson | 2009-05-18 13:58:26 -0500 (Mon, 18 May 2009) | 18 lines
  
  Fix externalivr's setvariable command so that it properly sets multiple variables.
  
  The command had a for loop that was guaranteed to only execute once since
  the continuation operation of the loop would set the input buffer NULL. I rewrote
  the loop so that its operation was more obvious, and it would set multiple variables
  correctly.
  
  I also reduced stack space required for the function, constified the input string,
  and modified the function so that it would not modify the input string while I was
  at it.
  
  (closes issue #15114)
  Reported by: chris-mac
  Patches:
        15114.patch uploaded by mmichelson (license 60)
  Tested by: chris-mac
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@195317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-18 19:00:24 +00:00
Joshua Colp
6bb5afcfab Merged revisions 195207 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r195207 | file | 2009-05-18 12:53:26 -0300 (Mon, 18 May 2009) | 14 lines
  
  Merged revisions 195206 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r195206 | file | 2009-05-18 12:51:22 -0300 (Mon, 18 May 2009) | 7 lines
    
    Fix a typo which caused loss of audio when using G729 in some scenarios with a smoother present.
    
    (closes issue #15105)
    Reported by: bamby
    Patches:
          process-vad-correctly.diff uploaded by bamby (license 430)
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@195208 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-18 15:54:26 +00:00
Eliel C. Sardanons
8bba1f73ce Merged revisions 195162 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r195162 | eliel | 2009-05-18 10:45:23 -0400 (Mon, 18 May 2009) | 9 lines
  
  Warn about the use of the application WaitExten() within a Macro().
  
  Update applications documentation to warn the user about the use of the
  WaitExten() application within a Macro(). Recommend the use of Read()
  instead.
  
  (closes issue #14444)
  Reported by: ewieling
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2009-05-18 15:19:27 +00:00
Joshua Colp
2baf6dd677 Merged revisions 195096 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r195096 | file | 2009-05-18 10:56:16 -0300 (Mon, 18 May 2009) | 12 lines
  
  Merged revisions 195095 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r195095 | file | 2009-05-18 10:53:39 -0300 (Mon, 18 May 2009) | 5 lines
    
    Fix a bug where the codecs of the called party leg were not properly sent back to the caller call leg when reinvited.
    
    (closes issue #13569)
    Reported by: bkw918
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@195097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-18 13:57:33 +00:00
Joshua Colp
bbcf0052fd Merged revisions 195089 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r195089 | file | 2009-05-18 10:36:17 -0300 (Mon, 18 May 2009) | 5 lines
  
  Fix a bug where specifying an empty outboundproxy would cause packets to get sent to ourself.
  
  (closes issue #15106)
  Reported by: timeshell
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2009-05-18 13:37:20 +00:00
Russell Bryant
37085ced64 Merged revisions 195021 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r195021 | russell | 2009-05-18 07:59:11 -0500 (Mon, 18 May 2009) | 12 lines
  
  Recorded merge of revisions 195020 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r195020 | russell | 2009-05-18 07:57:46 -0500 (Mon, 18 May 2009) | 5 lines
    
    Don't try to unlock a bogus channel.
    
    (closes issue #15144)
    Reported by: cristiandimache
  ........
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2009-05-18 13:01:16 +00:00
David Vossel
70e4d673f1 Merged revisions 194874 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r194874 | dvossel | 2009-05-15 17:44:44 -0500 (Fri, 15 May 2009) | 23 lines
  
  Merged revisions 194873 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r194873 | dvossel | 2009-05-15 17:43:13 -0500 (Fri, 15 May 2009) | 17 lines
    
    IAX2 REGAUTH loop
    
    IAX was not sending REGREJ to terminate invalid registrations.  Instead it sent another REGAUTH if the authentication challenge failed.  This caused a loop of REGREQ and REGAUTH frames.
    
    (Related to Security fix AST-2009-001)
    
    (closes issue #14867)
    Reported by: aragon
    Tested by: dvossel
    
    (closes issue #14717)
    Reported by: mobeck
    Patches:
          regauth_loop_update_patch.diff uploaded by dvossel (license 671)
    Tested by: dvossel
  ........
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2009-05-15 22:45:52 +00:00
David Vossel
134d5899c8 Merged revisions 194833 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r194833 | dvossel | 2009-05-15 15:52:12 -0500 (Fri, 15 May 2009) | 24 lines
  
  Merged revisions 194557,194685 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r194557 | dvossel | 2009-05-14 17:59:43 -0500 (Thu, 14 May 2009) | 10 lines
    
    IAX2 "Ghost" Channels
    
    There is a bug tracker issue where people are reporting "Ghost" channels in their 'iax2 show channels' output.  The confusion is caused by channels being listed as "(NONE)" with format "unknown".  These are not channels of coarse.  They are usually just pending registration or poke requests, but it is confusing output.  To help make sense of this I have added two columns to 'iax2 show channels'.  One shows the first message which started the transaction, and the second shows the last message sent by either side of the call.  This helps diagnose why the entry exists and why it may not go away.
    
    (closes issue #14207)
    Reported by: clive18
    
    Review: https://reviewboard.asterisk.org/r/246/
  ........
    r194685 | dvossel | 2009-05-15 10:40:37 -0500 (Fri, 15 May 2009) | 6 lines
    
    Update to previous IAX2 "Ghost" Channels patch.
    
    Fixed some comments made on reviewboard for the previous patch.
    
    (issue #14207)
  ........
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2009-05-15 21:08:26 +00:00
Russell Bryant
d067d7b06a Merged revisions 194765 via svnmerge from
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r194765 | russell | 2009-05-15 13:43:42 -0500 (Fri, 15 May 2009) | 10 lines

Merged revisions 194764 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r194764 | russell | 2009-05-15 13:43:18 -0500 (Fri, 15 May 2009) | 2 lines

Fix some spelling fail.

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2009-05-15 18:43:58 +00:00
Russell Bryant
26c5affb96 Merged revisions 194722 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r194722 | russell | 2009-05-15 12:59:08 -0500 (Fri, 15 May 2009) | 4 lines

Shuttle some bits around to address some gain issues with G.722.

(closes AST-209)

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2009-05-15 18:04:52 +00:00
Russell Bryant
ef334aff5f Merged revisions 194718 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r194718 | russell | 2009-05-15 12:37:12 -0500 (Fri, 15 May 2009) | 2 lines

Further simplify codec_g722 build.

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2009-05-15 17:37:29 +00:00
Russell Bryant
7981c09ed2 Merged revisions 194714 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r194714 | russell | 2009-05-15 12:24:39 -0500 (Fri, 15 May 2009) | 2 lines

Actually force running make for g722.

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2009-05-15 17:25:06 +00:00
Kevin P. Fleming
5aad6a76fd Merged revisions 194520 via svnmerge from
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  r194520 | kpfleming | 2009-05-14 17:26:02 -0500 (Thu, 14 May 2009) | 9 lines
  
  Merged revisions 194509 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r194509 | kpfleming | 2009-05-14 17:23:49 -0500 (Thu, 14 May 2009) | 1 line
    
    Update URL to Reviewboard
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2009-05-14 22:28:34 +00:00
Mark Michelson
bd0383c3be Merged revisions 194496 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r194496 | mmichelson | 2009-05-14 17:20:51 -0500 (Thu, 14 May 2009) | 30 lines
  
  Merged revisions 194484 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r194484 | mmichelson | 2009-05-14 17:17:55 -0500 (Thu, 14 May 2009) | 24 lines
    
    Fix a race condition where a reinvite could trigger a 482 response.
    
    The loop detection/spiral detection code in chan_sip used the owner
    channel's state as a criterion for determining if the incoming INVITE
    is a looped request. The problem with this is that the INVITE-handling
    code happens in a different thread than the thread that marks the owner
    channel as being up. As a result, if a reinvite were to come in very quickly,
    say from another Asterisk on the same LAN, it was possible for the reinvite
    to arrive before the owner channel had been set to the up state.
    
    This patch corrects the problem by using the invitestate of the sip_pvt
    instead, since that can be guaranteed to be set correctly by the time
    the reinvite arrives. Since there is a switch statement further in the
    INVITE-handling code, the AST_STATE_RINGING state also checks the invitestate
    of the sip_pvt in case we should actually be treating the channel as if it were
    up already.
    
    (closes issue #12215)
    Reported by: jpyle
    Patches:
          12215_confirmed.patch uploaded by mmichelson (license 60)
    Tested by: lmadsen
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@194504 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-14 22:22:44 +00:00
Joshua Colp
e946fa7eae Merged revisions 194434 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r194434 | file | 2009-05-14 14:05:33 -0300 (Thu, 14 May 2009) | 7 lines
  
  Fix a bug where the 'T' option to Meetme did not work.
  
  (closes issue #15031)
  Reported by: Stochastic
  (closes issue #13801)
  Reported by: justdave
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2009-05-14 17:06:23 +00:00
Mark Michelson
9e2182d974 Recorded merge of revisions 194357 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r194357 | mmichelson | 2009-05-13 14:42:51 -0500 (Wed, 13 May 2009) | 18 lines
  
  Blocked revisions 194356 via svnmerge
  
  ........
    r194356 | mmichelson | 2009-05-13 14:41:44 -0500 (Wed, 13 May 2009) | 13 lines
    
    Remove an extraneous unlocking operation from ast_channel_free.
    
    In the case that we could not remove the desired channel from the
    list of channels, there was an extra call to unlock the channel list.
    Since we unlock the list later on in the function anyway, this results
    in the list being unlocked twice yet only being locked once.
    
    (closes issue #15098)
    Reported by: tim_ringenbach
    Patches:
          remove_extra_unlock.diff uploaded by tim (license 540)
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@194358 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-13 19:50:04 +00:00
Joshua Colp
86e8f9c7e5 Merged revisions 194209 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r194209 | file | 2009-05-13 10:39:10 -0300 (Wed, 13 May 2009) | 18 lines
  
  Merged revisions 194208 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r194208 | file | 2009-05-13 10:38:01 -0300 (Wed, 13 May 2009) | 11 lines
    
    Fix RFC2833 issues with DTMF getting duplicated and with duration wrapping over.
    
    (closes issue #14815)
    Reported by: geoff2010
    Patches:
          v1-14815.patch uploaded by dimas (license 88)
    Tested by: geoff2010, file, dimas, ZX81, moliveras
    (closes issue #14460)
    Reported by: moliveras
    Tested by: moliveras
  ........
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2009-05-13 13:40:18 +00:00
Tilghman Lesher
b306d2e1ab Merged revisions 194138 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r194138 | tilghman | 2009-05-12 19:52:49 -0500 (Tue, 12 May 2009) | 14 lines
  
  Merged revisions 194137 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r194137 | tilghman | 2009-05-12 19:52:03 -0500 (Tue, 12 May 2009) | 7 lines
    
    Fix logic for how to proceed with a single digit extension.
    (closes issue #15091)
     Reported by: andrew
     Patches: 
           20090512__issue15091.diff.txt uploaded by tilghman (license 14)
     Tested by: andrew
  ........
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2009-05-13 00:54:07 +00:00
Matthew Nicholson
10a7f969fe Merged revisions 194057 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r194057 | mnicholson | 2009-05-12 17:32:13 -0500 (Tue, 12 May 2009) | 22 lines
  
  Merged revisions 194028 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r194028 | mnicholson | 2009-05-12 17:15:45 -0500 (Tue, 12 May 2009) | 16 lines
    
    This change modifies app_queue to properly generate CDR records in failure
    situations.
    
    This involves setting a proper cdr disposition coresponding to the given
    failure condition and ensuring the proper information is stored in the cdr
    record.
    
    (closes issue #13691)
    Reported by: dferrer
    Tested by: mnicholson
    
    (closes issue #13637)
    Reported by: atis
    Tested by: atis
  ........
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2009-05-12 23:03:58 +00:00
Mark Michelson
4a9497891c Merged revisions 193954 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r193954 | mmichelson | 2009-05-12 15:28:13 -0500 (Tue, 12 May 2009) | 18 lines
  
  Update spiral support in trunk and 1.6.X to match what is in 1.4.
  
  In 1.4, a SIP spiral is treated the same way as a call forward. This
  works much better than what is currently in trunk and 1.6.X. The code
  in trunk and 1.6.X did not create a new call to the recipient of the spiral,
  instead trying to continue the same call. In addition to just being plain
  wrong, this also had the side effect of only being able to spiral calls
  to other SIP channels.
  
  With this in place, as long as call forwards are honored, SIP spirals
  will work properly. This means that it will work for outbound calls
  made  by the Queue, Dial, and Page applications. For originated calls and
  spool calls, however, the spiral will not work properly until a generic
  call forward mechanism is introduced into Asterisk.
  
  (relates to issue #13630)
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2009-05-12 20:50:30 +00:00
Tilghman Lesher
538c562092 Merged revisions 193956 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r193956 | tilghman | 2009-05-12 15:40:22 -0500 (Tue, 12 May 2009) | 13 lines
  
  Merged revisions 193955 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r193955 | tilghman | 2009-05-12 15:39:21 -0500 (Tue, 12 May 2009) | 6 lines
    
    Avoid initializing routines if the authentication fails.  Fixes a crash (RR) issue.
    (closes issue #14508)
     Reported by: tiziano
     Patches: 
           20090221_2_wrongmailbox.diff.txt uploaded by tiziano (license 377)
  ........
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2009-05-12 20:42:09 +00:00
Mark Michelson
5a092bedca Blocked revisions 193886 via svnmerge
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  r193886 | mmichelson | 2009-05-12 13:20:14 -0500 (Tue, 12 May 2009) | 17 lines
  
  Blocked revisions 193880 via svnmerge
  
  ........
    r193880 | mmichelson | 2009-05-12 13:18:44 -0500 (Tue, 12 May 2009) | 12 lines
    
    Set the invitestate to INV_CANCELLED only if we are actually sending a SIP CANCEL.
    
    The problem was that the hangup code was setting the invitestate too early. The result of
    this was that we would always send a CANCEL request, even if it was not an appropriate
    time to do so (e.g. we have not yet received a provisional response for our INVITE).
    
    Note that this same fix had been applied to trunk and the 1.6.X branches starting with
    revision 155467. This is why you will see this revision being blocked from those places.
    
    AST-216
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2009-05-12 18:20:33 +00:00
Tilghman Lesher
b9d140e943 Merged revisions 193870 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r193870 | tilghman | 2009-05-12 12:29:33 -0500 (Tue, 12 May 2009) | 2 lines
  
  Convert a THREADSTORAGE object into a simple malloc'd object (as suggested by Russell on -dev)
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2009-05-12 17:47:16 +00:00
Tilghman Lesher
0569b32ac0 Recorded merge of revisions 193756 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r193756 | tilghman | 2009-05-11 17:50:47 -0500 (Mon, 11 May 2009) | 25 lines
  
  Recorded merge of revisions 193755 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r193755 | tilghman | 2009-05-11 17:48:20 -0500 (Mon, 11 May 2009) | 18 lines
    
    Move 300 bytes around on the stack, to make more room for an extension buffer.
    This allows more concurrent extensions to be copied for a single voicemail,
    without creating a possibility of upsetting existing users, where a dialplan
    could run out of stack space where it had run fine before.  Alternatively,
    we could have allocated off the heap, but that is a larger change and would
    have increased the chance for instability introduced by this change.
    
    This is really solved starting in 1.6.0.11, as the use of an ast_str buffer
    allows an unlimited number of extensions (up to available memory).  We
    additionally create a new warning message when the buffer length is exceeded,
    permitting administrators to see an issue after the fact, whereas previously
    the list was silently truncated.
    (closes issue #14739)
     Reported by: p_lindheimer
     Patches: 
           20090417__bug14739.diff.txt uploaded by tilghman (license 14)
     Tested by: p_lindheimer
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2009-05-11 23:23:15 +00:00
Richard Mudgett
d111483ab8 Merged revisions 193614 via svnmerge from
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  r193614 | rmudgett | 2009-05-11 14:11:29 -0500 (Mon, 11 May 2009) | 19 lines
  
  Merged revisions 193613 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r193613 | rmudgett | 2009-05-11 14:09:00 -0500 (Mon, 11 May 2009) | 12 lines
    
    Sent wrong message to clear a call we started if the other end has not responed yet.
    
    In the state MISDN_CALLING (i.e. SETUP was sent but no answer has arrived yet),
    it is not allowed to clear the call with RELEASE_COMPLETE.  It must be
    cleared with DISCONNECT.  A RELEASE_COMPLETE is only allowed as an answer
    to a SETUP.  (See Q.931 ch. 5.3.2, 5.3.2.a, 5.3.2.b)
    
    Patches:
        chan-misdn-ccstate7.patch uploaded by customer.
    
    JIRA ABE-1862
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2009-05-11 19:15:13 +00:00
Leif Madsen
156d70ea47 Recorded merge of revisions 193545 via svnmerge from
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  r193545 | lmadsen | 2009-05-11 14:01:44 -0400 (Mon, 11 May 2009) | 14 lines
  
  Recorded merge of revisions 193544 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r193544 | lmadsen | 2009-05-11 13:35:17 -0400 (Mon, 11 May 2009) | 7 lines
    
    Document CHANNEL(transfercapability) in CLI documentation.
    
    (issue #15073)
    Reported by: pkempgen
    Patches:
          20090511__issue15073.diff.txt uploaded by tilghman (license 14)
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2009-05-11 18:06:53 +00:00
David Vossel
80a37a02a6 Merged revisions 193387 via svnmerge from
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  r193387 | dvossel | 2009-05-08 15:32:51 -0500 (Fri, 08 May 2009) | 7 lines
  
  TCP not matching valid peer.
  
  find_peer() does not find a valid peer when using pvt->recv as the sockaddr_in argument.  Because of the way TCP works, the port number in pvt->recv is not what we're looking for at all.  There is currently only one place that find_peer searches for a peer using the sockaddr_in argument.  If the peer is not found after using pvt->recv (works for UDP since the port number will be correct), a temp sockaddr_in struct is made using the Contact header in the sip_request.  This has the correct port number in it.
  
  Review: http://reviewboard.digium.com/r/236/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@193449 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-08 21:32:25 +00:00
Sean Bright
3c62a045b7 Merged revisions 193274 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r193274 | seanbright | 2009-05-08 11:18:40 -0400 (Fri, 08 May 2009) | 2 lines
  
  Fix the spelling of UNAVAILABLE in func_devstate CLI completion.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@193334 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-08 15:36:14 +00:00
David Vossel
3c092fb35b Merged revisions 193263 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r193263 | dvossel | 2009-05-08 09:52:19 -0500 (Fri, 08 May 2009) | 15 lines
  
  Merged revisions 193262 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r193262 | dvossel | 2009-05-08 09:51:09 -0500 (Fri, 08 May 2009) | 9 lines
    
    "misdn show config" segfaults asterisk, if no MSN lists 
    
    (closes issue #14976)
    Reported by: alecdavis
    Patches:
          misdn_config.diff.txt uploaded by alecdavis (license 585)
    Tested by: alecdavis, FabienToune
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@193264 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-08 14:53:28 +00:00
Kevin P. Fleming
61714bdefc Merged revisions 193194 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r193194 | kpfleming | 2009-05-08 09:06:15 -0500 (Fri, 08 May 2009) | 13 lines
  
  Merged revisions 193193 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r193193 | kpfleming | 2009-05-08 09:03:28 -0500 (Fri, 08 May 2009) | 7 lines
    
    Make absolute paths for logger channels work properly
    
    (Note: This is not a new feature, it was previously undocumented and broken.)
    
    The Asterisk logger has a feature to support absolute pathnames for logger channels, but the code implementing the feature was broken. This has been fixed, and the absolute path feature is now documented in the sample logger.conf.
  ........
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2009-05-08 14:09:05 +00:00
Tilghman Lesher
46cb79f430 Merged revisions 193120 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r193120 | tilghman | 2009-05-07 18:42:28 -0500 (Thu, 07 May 2009) | 26 lines
  
  Merged revisions 193119 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r193119 | tilghman | 2009-05-07 18:41:11 -0500 (Thu, 07 May 2009) | 19 lines
    
    Fix Background within a Macro for FreePBX.
    If the single digit DTMF is an extension in the specified context, then
    go there and signal no DTMF.  Otherwise, we should exit with that DTMF.
    If we're in Macro, we'll exit and seek that DTMF as the beginning of an
    extension in the Macro's calling context.  If we're not in Macro, then
    we'll simply seek that extension in the calling context.  Previously,
    someone complained about the behavior as it related to the interior of a
    Gosub routine, and the fix (#14011) inadvertently broke FreePBX
    (#14940).  This change should fix both of these situations, but with the
    possible incompatibility that if a single digit extension does not exist
    (but a longer extension COULD have matched), it would have previously
    gone immediately to the "i" extension, but will now need to wait for a
    timeout.
    (closes issue #14940)
     Reported by: p_lindheimer
     Patches: 
           20090420__bug14940.diff.txt uploaded by tilghman (license 14)
     Tested by: p_lindheimer
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@193121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-07 23:43:23 +00:00
Richard Mudgett
e9cf423d60 Merged revisions 193077 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r193077 | rmudgett | 2009-05-07 17:24:04 -0500 (Thu, 07 May 2009) | 12 lines
  
  Merged revisions 193050 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r193050 | rmudgett | 2009-05-07 17:17:06 -0500 (Thu, 07 May 2009) | 5 lines
    
    Give a more helpful message when an incoming call's dialed extension does not match.
    
    Added the dialed extension and context to the chan_misdn messages warning
    that the dialed number cannot be matched in the dialplan.
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@193078 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-07 22:34:00 +00:00
Tilghman Lesher
6315384b63 Merged revisions 192938 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r192938 | tilghman | 2009-05-07 12:13:36 -0500 (Thu, 07 May 2009) | 6 lines
  
  Send DTMF frame before playing back audio.
  (closes issue #14858)
   Reported by: barryf
   Patches: 
         20090507__bug14858.diff.txt uploaded by tilghman (license 14)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@192939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-07 17:14:26 +00:00
Tilghman Lesher
2a0f0439e5 Merged revisions 192933 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r192933 | tilghman | 2009-05-07 11:43:56 -0500 (Thu, 07 May 2009) | 17 lines
  
  Merged revisions 192932 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r192932 | tilghman | 2009-05-07 11:29:08 -0500 (Thu, 07 May 2009) | 10 lines
    
    Eliminate repetition of fullcontact during reconstruction.
    If the fullcontact field appears in both the sippeers and the
    sipregs table, then during reconstruction of the field, it will
    otherwise be doubled.
    (closes issue #14754)
     Reported by: Alexei Gradinari
     Patches: 
           20090506__bug14754.diff.txt uploaded by tilghman (license 14)
     Tested by: lmadsen
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@192934 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-07 16:45:23 +00:00
Jeff Peeler
76c66014b8 Merged revisions 192861 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r192861 | jpeeler | 2009-05-06 17:17:27 -0500 (Wed, 06 May 2009) | 17 lines
  
  Merged revisions 192858 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r192858 | jpeeler | 2009-05-06 17:15:19 -0500 (Wed, 06 May 2009) | 10 lines
    
    Make ParkedCall application stop execution of the dialplan after hang up
    
    Just changed park_exec to always return non-zero. I really wasn't entirely sure
    at first if this was a bug. Decided it was since it would be surprising when 
    not using ParkedCall in the dialplan to hang up and have dialplan execution
    continue.
    
    (closes issue #14555)
    Reported by: francesco_r
  ........
................


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2009-05-06 22:18:30 +00:00
Jeff Peeler
4600d1a538 Blocked revisions 192853 via svnmerge
........
  r192853 | jpeeler | 2009-05-06 17:02:46 -0500 (Wed, 06 May 2009) | 2 lines
  
  If no extension was found in the pattern tree, don't crash.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@192856 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-06 22:06:02 +00:00
Matthew Fredrickson
dca288830a Merged revisions 190946 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
r190946 | mattf | 2009-04-28 17:05:05 -0500 (Tue, 28 Apr 2009) | 1 line

Make sure that we do not clear the down flag on the BRI during PTMP link transients.  Also fix no audio bug caused by big early audio patch.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@192811 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-06 17:41:58 +00:00
Joshua Colp
2f3e7752f2 Merged revisions 192634 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r192634 | file | 2009-05-06 10:34:35 -0300 (Wed, 06 May 2009) | 14 lines
  
  Merged revisions 192633 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r192633 | file | 2009-05-06 10:30:51 -0300 (Wed, 06 May 2009) | 7 lines
    
    Update some old logic to stop both begin and end DTMF frames from reaching the core if rfc2833 is not enabled.
    
    (closes issue #15036)
    Reported by: dimas
    Patches:
          v1-15036.patch uploaded by dimas (license 88)
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@192635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-06 13:36:19 +00:00
Sean Bright
b516cf9b81 Merged revisions 192525 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r192525 | seanbright | 2009-05-05 15:57:49 -0400 (Tue, 05 May 2009) | 18 lines
  
  Merged revisions 192524 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r192524 | seanbright | 2009-05-05 15:56:11 -0400 (Tue, 05 May 2009) | 11 lines
    
    Fix Javascript error when using astman.js in Internet Explorer.
    
    Internet Explorer (tested with 7.0) does not like trailing commas on constructs
    like object initializers, so get rid of them to avoid some errors.
    
    (closes issue #15026)
    Reported by: rajnishgiri
    Patches:
          bug15026.patch uploaded by seanbright (license 71)
    Tested by: seanbright
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@192526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-05 20:02:10 +00:00
Joshua Colp
7c39ce9105 Merged revisions 192462 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r192462 | file | 2009-05-05 15:23:58 -0300 (Tue, 05 May 2009) | 15 lines
  
  Merged revisions 192454 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r192454 | file | 2009-05-05 15:22:27 -0300 (Tue, 05 May 2009) | 8 lines
    
    Fix an incorrect assumption that certain values on the channel will always exist when they may not.
    
    The CDR code involved with bridges wrongly assumed that the currently executing application and data
    values will always exist. It is possible for this to be false when call forwarding is involved.
    
    (closes issue #14984)
    Reported by: gincantalupo
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@192467 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-05 18:25:12 +00:00
Joshua Colp
7080914d21 Merged revisions 192430 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r192430 | file | 2009-05-05 14:46:51 -0300 (Tue, 05 May 2009) | 12 lines
  
  Merged revisions 192429 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r192429 | file | 2009-05-05 14:43:30 -0300 (Tue, 05 May 2009) | 5 lines
    
    Fix a bug where the followme application would continue trying numbers after the caller hung up.
    
    (closes issue #13624)
    Reported by: sgenyuk
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@192431 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-05 17:48:41 +00:00
Joshua Colp
e088059f46 Merged revisions 192387 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r192387 | file | 2009-05-05 11:22:47 -0300 (Tue, 05 May 2009) | 10 lines
  
  Fix a bug with setting t38pt_udptl at the user or peer level.
  
  If an incoming call authenticated as a user or peer and t38pt_udptl was
  not set to yes in general then no UDPTL session would be present and any
  T38 related things would fail. This commit changes it so that if after
  authenticating T38 is enabled but no UDPTL session is present one will be
  created.
  
  (issue AST-215)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@192397 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-05 14:25:15 +00:00
Kevin P. Fleming
3f94c3bd6b Merged revisions 192357 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r192357 | kpfleming | 2009-05-05 15:18:21 +0200 (Tue, 05 May 2009) | 5 lines
  
  Correct some flaws in the memory accounting code for stringfields and ao2 objects
  
  Under some conditions, the memory allocation for stringfields and ao2 objects would not have supplied valid file/function names for MALLOC_DEBUG tracking, so this commit corrects that.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@192358 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-05 13:23:27 +00:00
Kevin P. Fleming
10a2c0099e Merged revisions 192318 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r192318 | kpfleming | 2009-05-05 12:34:19 +0200 (Tue, 05 May 2009) | 5 lines
  
  Properly account for memory allocated for channels and datastores
  
  As in previous commits, when channels are allocated (with ast_channel_alloc) or datastores are allocated (with ast_datastore_alloc) properly account for the memory being owned by the caller, instead of the allocator function itself.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@192353 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-05 12:27:18 +00:00
Kevin P. Fleming
60280df797 Merged revisions 192279 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r192279 | kpfleming | 2009-05-05 10:51:06 +0200 (Tue, 05 May 2009) | 5 lines
  
  Ensure that string pools allocated to hold stringfields are properly accounted in MALLOC_DEBUG mode
  
  This commit modifies the stringfield pool allocator to remember the 'owner' of the stringfield manager the pool is being allocated for, and ensures that pools allocated in the future when fields are populated are owned by that file/function.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@192280 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-05 09:02:19 +00:00
David Vossel
efe35d94c1 Merged revisions 192214 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r192214 | dvossel | 2009-05-04 17:44:51 -0500 (Mon, 04 May 2009) | 17 lines
  
  Merged revisions 192213 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r192213 | dvossel | 2009-05-04 17:37:31 -0500 (Mon, 04 May 2009) | 11 lines
    
    global mohinterpret setting is ignored
    
    mohinterpret and mohsuggest global variables were not copied over during build_users and build_peers.
    
    (closes issue #14728)
    Reported by: dimas
    Patches:
          v1-14728.patch uploaded by dimas (license 88)
    Tested by: dimas, dvossel
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@192215 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-04 22:47:20 +00:00
Tilghman Lesher
4b6075f430 Regen configure script
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@192174 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-04 19:32:35 +00:00
Tilghman Lesher
9817471013 Merged revisions 192132 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r192132 | tilghman | 2009-05-04 13:42:56 -0500 (Mon, 04 May 2009) | 6 lines
  
  Pass libraries in LIBS, not LDFLAGS.
  (closes issue #14671)
   Reported by: Chainsaw
   Patches: 
         asterisk-1.6.0.6-toolcheck-libs-not-ldflags.patch uploaded by Chainsaw (license 723)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@192133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-04 18:44:08 +00:00
Leif Madsen
cd1b968ec1 Commit the changes from rev 192094 in the 1.6.1 branch to the 1.6.0 branch.
Since the documentation system is different between <= 1.6.1 and >= 1.6.2
the changes to the documentation I made for issue 14801, I'm merging these
changes in manually. Additional changes will then be put into trunk and
merged down to 1.6.2 as those two versions share a similar documentation
system (XML).

(issue #14801)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@192095 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-04 17:39:30 +00:00
Kevin P. Fleming
de96aab0b8 Merged revisions 191955 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r191955 | kpfleming | 2009-05-04 11:57:36 +0200 (Mon, 04 May 2009) | 8 lines
  
  Ensure that by default only one console channel driver is loaded
  
  This configuration file was changed to ensure that only one console channel driver
  (chan_oss) is loaded by default, but the change would only work if chan_console
  was not built. Now it will work as expected; if chan_alsa or chan_console are built
  and installed, they will not be loaded unless explicity requested.
........


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2009-05-04 09:58:56 +00:00
Mark Michelson
c0519ce31d Blocked revisions 191781 via svnmerge
................
  r191781 | mmichelson | 2009-05-02 13:54:52 -0500 (Sat, 02 May 2009) | 16 lines
  
  Blocked revisions 191778 via svnmerge
  
  ........
    r191778 | mmichelson | 2009-05-02 13:48:20 -0500 (Sat, 02 May 2009) | 11 lines
    
    Fix a bug which resulted from the Hebrew voicemail commit.
    
    This fixes a case where a certain message could get played twice.
    
    (closes issue #13155)
    Reported by: greenfieldtech
    Patches:
          app_voicemail.c.multi-lang-patch uploaded by greenfieldtech (license 369)
    Tested by: greenfieldtech
  ........
................


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2009-05-02 18:55:17 +00:00
Kevin P. Fleming
52e5bad131 Merged revisions 191775 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r191775 | kpfleming | 2009-05-02 20:39:48 +0200 (Sat, 02 May 2009) | 5 lines
  
  Fix an error in queue_log file rotation optimization code
  
  This code was copy-and-pasted without properly changing references to event_rotate into queue_rotate, so under some conditions the log rotation would rotate queue_log even though it was not necessary.
........


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2009-05-02 18:44:05 +00:00
Sean Bright
ea6efbb4a2 Merged revisions 191700 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r191700 | seanbright | 2009-05-02 11:45:07 -0400 (Sat, 02 May 2009) | 1 line
  
  Update copyright year to 2009
........


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2009-05-02 15:51:40 +00:00
Mark Michelson
acf3ce5f18 Blocked revisions 191630 via svnmerge
................
  r191630 | mmichelson | 2009-05-02 05:46:02 -0500 (Sat, 02 May 2009) | 17 lines
  
  Blocked revisions 191628-191629 via svnmerge
  
  ........
    r191628 | mmichelson | 2009-05-02 05:21:00 -0500 (Sat, 02 May 2009) | 8 lines
    
    Move static buffers to outside for loops in app_chanspy.
    
    Similar to seanbright's commit 191422, this moves some static buffers
    to be defined outside of for loops since it is undefined if memory
    will be re-used or if the stack will grow with each iteration of the
    loop.
  ........
    r191629 | mmichelson | 2009-05-02 05:45:24 -0500 (Sat, 02 May 2009) | 3 lines
    
    Kevin has informed me that thi sort of thing is not necessary.
  ........
................


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2009-05-02 10:46:29 +00:00
Tilghman Lesher
26c5ebaa04 Merged revisions 191560 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r191560 | tilghman | 2009-05-01 15:01:21 -0500 (Fri, 01 May 2009) | 13 lines
  
  Merged revisions 191559 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r191559 | tilghman | 2009-05-01 15:00:23 -0500 (Fri, 01 May 2009) | 6 lines
    
    SIP Response 410 maps to cause code 22 (or 23), not 1.
    (closes issue #14993)
     Reported by: BigJimmy
     Patches: 
           causepatch uploaded by BigJimmy (license 371)
  ........
................


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2009-05-01 20:02:33 +00:00
Tilghman Lesher
10a9c43c9f Merged revisions 191494 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r191494 | tilghman | 2009-05-01 13:18:00 -0500 (Fri, 01 May 2009) | 4 lines
  
  Set debug message back to DEBUG level.
  (closes issue #15007)
   Reported by: hulber
........


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2009-05-01 18:31:04 +00:00
Jeff Peeler
7193a465d7 Merged revisions 191489 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r191489 | jpeeler | 2009-05-01 13:09:23 -0500 (Fri, 01 May 2009) | 15 lines
  
  Merged revisions 191488 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r191488 | jpeeler | 2009-05-01 12:40:46 -0500 (Fri, 01 May 2009) | 9 lines
    
    Fix DTMF not being sent to other side after a partial feature match
    
    This fixes a regression from commit 176701. The issue was that
    ast_generic_bridge never exited after the feature digit timeout had elapsed,
    which prevented the queued DTMF from being sent to the other side.
    
    This issue was reported to me directly.
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@191499 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-01 18:19:05 +00:00
Sean Bright
c87521c295 Fix a crash in app_queue with very long member lists.
A user reported via #asterisk that with very long lists of members, a crash
occurs in ast_strdupa, so just use a single buffer and ast_copy_string instead
of stack allocating copys of each interface name.

(Related to revision 191041 in branches/1.4)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@191424 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-01 15:58:40 +00:00
Tilghman Lesher
5e04e9d398 Merged revisions 191367 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r191367 | tilghman | 2009-04-30 12:40:58 -0500 (Thu, 30 Apr 2009) | 3 lines
  
  Detect eaccess (or euidaccess) before using it.
  Reported by Andrew Lindh via the -dev list.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@191368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-30 17:43:37 +00:00
Tilghman Lesher
1abc47f2c5 Merged revisions 191283 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r191283 | tilghman | 2009-04-30 01:47:13 -0500 (Thu, 30 Apr 2009) | 11 lines
  
  Change working directory to / under certain conditions.
  If backgrounding and no core will be produced, then changing the directory
  won't break anything; likewise, if the CWD isn't accessible by the current
  user, then a core wasn't possible anyway.
  (closes issue #14831)
   Reported by: chris-mac
   Patches: 
         20090428__bug14831.diff.txt uploaded by tilghman (license 14)
         20090430__bug14831.diff.txt uploaded by tilghman (license 14)
   Tested by: chris-mac
........


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2009-04-30 06:48:00 +00:00
Tilghman Lesher
9e50cbb6c2 Merged revisions 191219 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r191219 | tilghman | 2009-04-29 18:06:56 -0500 (Wed, 29 Apr 2009) | 2 lines
  
  Make H.323 compile with FDLEAK detection code enabled
........


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2009-04-29 23:14:51 +00:00
Jeff Peeler
c136637456 Blocked revisions 191213 via svnmerge
........
  r191213 | jpeeler | 2009-04-29 17:56:55 -0500 (Wed, 29 Apr 2009) | 1 line
  
  fix typos
........


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2009-04-29 22:57:48 +00:00
David Brooks
6a5c253517 Merged revisions 191136 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r191136 | dbrooks | 2009-04-29 13:32:58 -0500 (Wed, 29 Apr 2009) | 3 lines
  
  Removing crufty code that is no longer necessary. Code cleanup.
........


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2009-04-29 18:39:14 +00:00
David Brooks
458d2d5253 Blocked revisions 191116 via svnmerge
................
  r191116 | dbrooks | 2009-04-29 13:22:24 -0500 (Wed, 29 Apr 2009) | 14 lines
  
  Blocked revisions 191096 via svnmerge
  
  ........
    r191096 | dbrooks | 2009-04-29 13:07:59 -0500 (Wed, 29 Apr 2009) | 8 lines
    
    Patch to fix tab-completion crash on "remove extension"
    
    This patch simply removes some old code back before Asterisk used editline. 
    This fixes the crash that occurred when tab-completing "remove extension".
    
    (closes issue #14689)
    Reported by: isaacgal
  ........
................


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2009-04-29 18:24:24 +00:00
Russell Bryant
e2b2d16176 Revert revision 190576 after out of band discussion with transnexus.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@190986 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-29 07:37:04 +00:00
Tilghman Lesher
3598b08507 Merged revisions 190904 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r190904 | tilghman | 2009-04-28 12:31:43 -0500 (Tue, 28 Apr 2009) | 2 lines
  
  UniqueID column has a maximum size of 150
........


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2009-04-28 17:32:24 +00:00
Kevin P. Fleming
5d7dba2c9b Blocked revisions 190865 via svnmerge
........
  r190865 | kpfleming | 2009-04-28 09:15:47 -0500 (Tue, 28 Apr 2009) | 5 lines
  
  Build XML documention from *only* the source files that have docs in them
  
  Change the build process so that doc/core-en_US.xml is dependent solely on the source files that have documentation in them, not on all source files.
........


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2009-04-28 14:16:34 +00:00
Kevin P. Fleming
0e0f66d33c Merged revisions 190861 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r190861 | kpfleming | 2009-04-28 09:12:09 -0500 (Tue, 28 Apr 2009) | 5 lines
  
  Remove Makefile rules for bison and flex sources
  
  We never, ever want these files to processed automatically, because we store the output files in Subversion and users should never need to rebuild them.
........


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2009-04-28 14:12:53 +00:00
Jeff Peeler
e71fc5d141 Fix announcement of parking space after transfer to park.
Don't set the orig_chan_name in masq_park_call when the announcement is to be
played. This change was copied from all the other working 1.6 branches.

(closes issue #14863)
Reported by: noahisaac


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@190829 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-27 23:25:05 +00:00
Kevin P. Fleming
53843dfb82 Merged revisions 190725 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r190725 | kpfleming | 2009-04-27 14:30:54 -0500 (Mon, 27 Apr 2009) | 13 lines
  
  Merged revisions 190721 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r190721 | kpfleming | 2009-04-27 14:29:46 -0500 (Mon, 27 Apr 2009) | 7 lines
    
    Fix 'inconsistent line endings' when autoconf 2.63 is used
    
    Attempt to make configure script regeneration 'safe' using autoconf 2.63, which embeds a bare CR into the script, thus making Subversion complain about inconsistent line endings
    
    This commit changes the MIME type of the configure script to be 'binary' thus making Subversion no longer inspect line endings, and as a bonus 'svn diff' will no longer try to generate diff output for it, which is not generally useful anyway.
  ........
................


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2009-04-27 19:40:58 +00:00
Tilghman Lesher
960bc63346 Merged revisions 190726 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r190726 | tilghman | 2009-04-27 14:34:48 -0500 (Mon, 27 Apr 2009) | 4 lines
  
  Don't warn on pipe in the System call.
  (closes issue #14979)
   Reported by: pj
........


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2009-04-27 19:36:05 +00:00
Russell Bryant
3521d921a7 Blocked revisions 190663 via svnmerge
................
r190663 | russell | 2009-04-27 14:08:12 -0500 (Mon, 27 Apr 2009) | 22 lines

Merged revisions 190661-190662 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r190661 | russell | 2009-04-27 14:00:54 -0500 (Mon, 27 Apr 2009) | 9 lines

Resolve a crash in res_smdi when used with chan_dahdi.

When chan_dahdi goes to get an SMDI message, it provides no search criteria.
It just grabs the next message that arrives.  This code was written with the
SMDI dialplan functions in mind, since that is now the preferred method of
using SMDI.  However, this broke support of it being used from chan_dahdi.

(closes AST-212)

........
r190662 | russell | 2009-04-27 14:03:59 -0500 (Mon, 27 Apr 2009) | 2 lines

Fix a typo from 190661.

........

................


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2009-04-27 19:09:28 +00:00
Mark Michelson
2eb85214eb Blocked revisions 190626 via svnmerge
........
  r190626 | mmichelson | 2009-04-27 11:37:51 -0500 (Mon, 27 Apr 2009) | 14 lines
  
  Allow for a position to be specified when entering a queue.
  
  This would allow for one to add a caller to a specific place in the
  queue instead of just placing the caller in the back every time. To help
  facilitate some interesting manipulations, a new channel variable called
  QUEUEPOSITION has been added. When a caller is removed from a queue, his
  position in that queue is stored in the QUEUEPOSITION variable. One such
  strategy an administrator can employ is to allow for the removal of a caller
  from one queue followed by the insertion of the same caller into a separate
  queue in the same position.
  
  Review: http://reviewboard.digium.com/r/189
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@190627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-27 16:38:08 +00:00
Mark Michelson
d8775d20f2 Merged revisions 190622 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r190622 | mmichelson | 2009-04-27 11:26:14 -0500 (Mon, 27 Apr 2009) | 3 lines
  
  Update warning message to not have pipes and contain all options.
........


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2009-04-27 16:27:09 +00:00
Mark Michelson
ac0f48dc40 Blocked revisions 190577 via svnmerge
........
  r190577 | mmichelson | 2009-04-27 09:46:14 -0500 (Mon, 27 Apr 2009) | 6 lines
  
  Remove nonexistent option from sip.conf.sample.
  
  The option to choose which connected line header to
  use is not 'rpid_header' but 'sendrpid'
........


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2009-04-27 14:46:32 +00:00
TransNexus OSP Development
ab344908be 1. Fixed the issue caused by network ID.
2. Fixed the issue caused by without certificate files.
3. Fixed the issue caused by number portability parameters in user part of RURI.
4. Updated for OSP Toolkit 3.5.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@190576 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-27 06:14:17 +00:00
Tilghman Lesher
08c9634115 Merge of timedwrlock detection broke build.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@190419 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-23 22:21:06 +00:00
Russell Bryant
a92f2e2a77 Merged revisions 190357 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
r190357 | russell | 2009-04-23 16:13:07 -0500 (Thu, 23 Apr 2009) | 10 lines

Merged revisions 190356 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r190356 | russell | 2009-04-23 16:07:07 -0500 (Thu, 23 Apr 2009) | 2 lines

Remove a bogus ast_channel_unlock().

........

................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@190358 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-23 21:14:29 +00:00
Tilghman Lesher
8efbe92b1f Merged revisions 190352 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r190352 | tilghman | 2009-04-23 15:42:11 -0500 (Thu, 23 Apr 2009) | 7 lines
  
  Labels are sometimes (most of the time?) NULL for extensions.
  (closes issue #14895)
   Reported by: chris-mac
   Patches: 
         20090423__bug14895__2.diff.txt uploaded by tilghman (license 14)
   Tested by: lmadsen
........


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2009-04-23 20:42:58 +00:00
Joshua Colp
28b43b886f Merged revisions 190287 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r190287 | file | 2009-04-23 16:15:30 -0300 (Thu, 23 Apr 2009) | 13 lines
  
  Merged revisions 190286 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r190286 | file | 2009-04-23 16:13:18 -0300 (Thu, 23 Apr 2009) | 6 lines
    
    Fix a bug in chan_local glare hangup detection.
    
    If both sides of a Local channel were hung up at around the same time it was
    possible for one thread to destroy the local private structure and have the other thread
    immediately try to remove the already freed structure from the local channel list.
  ........
................


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2009-04-23 19:16:32 +00:00
Mark Michelson
fecac1654c Blocked revisions 190250 via svnmerge
........
  r190250 | mmichelson | 2009-04-23 12:45:35 -0500 (Thu, 23 Apr 2009) | 9 lines
  
  Fix reversed behavior of leavewhenempty option in queues.conf.
  
  (closes issue #14650)
  Reported by: alecdavis
  Patches:
        14650.patch uploaded by mmichelson (license 60)
  Tested by: mmichelson, lmadsen
........


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2009-04-23 17:46:06 +00:00
Tilghman Lesher
94b71d104f Merged revisions 190093 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r190093 | tilghman | 2009-04-22 16:38:15 -0500 (Wed, 22 Apr 2009) | 14 lines
  
  Merged revisions 190092 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r190092 | tilghman | 2009-04-22 16:35:03 -0500 (Wed, 22 Apr 2009) | 7 lines
    
    Detect availability of pthread_rwlock_timedwrlock() before using it.
    (closes issue #14930)
     Reported by: tilghman
     Patches: 
           20090420__bug14930.diff.txt uploaded by tilghman (license 14)
     Tested by: mvanbaak, tilghman
  ........
................


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2009-04-22 21:41:24 +00:00
Jeff Peeler
62f5546319 Merged revisions 190057 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r190057 | jpeeler | 2009-04-22 16:15:55 -0500 (Wed, 22 Apr 2009) | 9 lines
  
  Fix building of chan_h323 with gcc-3.3
  
  There seems to be a bug with old versions of g++ that doesn't allow a structure
  member to use the name list. Rename list member to group_list in ast_group_info
  and change the few places it is used.
  
  (closes issue #14790)
  Reported by: stuarth
........


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2009-04-22 21:17:29 +00:00
Jeff Peeler
7378553cda Merged revisions 189993 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r189993 | jpeeler | 2009-04-22 14:23:49 -0500 (Wed, 22 Apr 2009) | 18 lines
  
  Make chan_h323 respect packetization settings
  
  Previously, packetization settings were ignored and now they are not. A new
  config option 'autoframing' has been added to mirror the way chan_sip handles
  it. Turning on the autoframing option (available both as a global option or per
  peer) overrides the local settings with the remote packetization settings.
  Testing was performed with varying packetization levels with the following
  codecs: ulaw, alaw, gsm, and g729.
  
  (closes issue #12415)
  Reported by: pj
  Patches:
        2009012200_h323packetization.diff.txt uploaded by mvanbaak (license 7), 
        modified by me
........


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2009-04-22 19:33:07 +00:00
Russell Bryant
b44232f346 Merged revisions 189951 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r189951 | russell | 2009-04-22 11:56:43 -0500 (Wed, 22 Apr 2009) | 2 lines

Fix call parking callback.  Pipes -> Commas.

........


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2009-04-22 17:57:32 +00:00
Tilghman Lesher
87223fa20b Merged revisions 189911 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r189911 | tilghman | 2009-04-22 11:01:30 -0500 (Wed, 22 Apr 2009) | 7 lines
  
  Do not continue to receive DTMF, when the channel is hungup and about to be destroyed.
  (closes issue #14858)
   Reported by: barryf
   Patches: 
         20090421__bug14858.diff.txt uploaded by tilghman (license 14)
   Tested by: barryf
........


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2009-04-22 16:02:28 +00:00
Michiel van Baak
35cf6af3bc Merged revisions 189850 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r189850 | mvanbaak | 2009-04-22 16:30:47 +0200 (Wed, 22 Apr 2009) | 19 lines
  
  Merged revisions 189849 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r189849 | mvanbaak | 2009-04-22 16:29:28 +0200 (Wed, 22 Apr 2009) | 12 lines
    
    replace sed with tr to remove \r from downloaded file
    
    On some systems, sed does not recognize \r in the pattern the way it
    was used here.
    Use tr instead because this works the same across systems.
    
    (closes issue #14936)
    Reported by: leobrown
    Patches: 
          2009042201_14936.diff.txt uploaded by mvanbaak (license 7)
    	  Tested by: leobrown, mvanbaak
  ........
................


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2009-04-22 14:32:43 +00:00
Tilghman Lesher
8a52482a06 Merged revisions 189813 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r189813 | tilghman | 2009-04-22 01:33:08 -0500 (Wed, 22 Apr 2009) | 3 lines
  
  Detect liblua on SuSE, and add libm for linking for Fedora.
  (Reported via the -dev list, Subject: Compiling Asterisk with LUA)
........


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2009-04-22 06:35:30 +00:00
David Vossel
d9073982da Merged revisions 189771 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r189771 | dvossel | 2009-04-21 15:28:37 -0500 (Tue, 21 Apr 2009) | 11 lines
  
  Fixes segfault when switching UDP to TCP in sip.conf after reload.
  
  If transport in sip.conf is switched from UDP to TCP, Asterisk segfaults right after issuing a sip reload.  The problem is the socket type is changed to TCP but the fd may still be present for UDP.  Later, when the TCP session should be created or set using an existing one, it isn't because the old file descriptor is still present.  Now every time transport is changed during a sip.conf reload, the file descriptor is set to -1, signifying it must be created or found.
  
  (closes issue #14727)
  Reported by: pj
  Tested by: dvossel
  
  Review: http://reviewboard.digium.com/r/229/
........


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2009-04-21 20:41:41 +00:00
Doug Bailey
faedc64019 Merged revisions 189665 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r189665 | dbailey | 2009-04-21 10:54:16 -0500 (Tue, 21 Apr 2009) | 9 lines
  
  Merged revisions 189664 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r189664 | dbailey | 2009-04-21 10:52:13 -0500 (Tue, 21 Apr 2009) | 2 lines
    
    Remove daemon call on systems that do not support forking.
  ........
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2009-04-21 15:56:39 +00:00
Doug Bailey
a0c3c3dad9 Merged revisions 189629 via svnmerge from
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  r189629 | dbailey | 2009-04-21 09:28:04 -0500 (Tue, 21 Apr 2009) | 10 lines
  
  Merged revisions 189601 via svnmerge from 
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  ........
    r189601 | dbailey | 2009-04-21 09:00:55 -0500 (Tue, 21 Apr 2009) | 3 lines
    
    Add check in configure script to check for GLOB_NOMAGIC and GLOB_BRACE in glob.h 
    This allows config.c to compile when linked against uclibc that does not support these parameters
  ........
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2009-04-21 14:52:54 +00:00
Terry Wilson
207333f4fd Merged revisions 189516 via svnmerge from
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  r189516 | twilson | 2009-04-20 16:29:29 -0500 (Mon, 20 Apr 2009) | 9 lines
  
  Merged revisions 189465 via svnmerge from 
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  ........
    r189465 | twilson | 2009-04-20 16:10:27 -0500 (Mon, 20 Apr 2009) | 2 lines
    
    Update CDR appropriately when AST_CAUSE_NO_ANSWER is set
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2009-04-20 21:35:33 +00:00
Terry Wilson
4d71e7a511 Merged revisions 189495 via svnmerge from
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  r189495 | twilson | 2009-04-20 16:24:34 -0500 (Mon, 20 Apr 2009) | 9 lines
  
  Merged revisions 189463 via svnmerge from 
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  ........
    r189463 | twilson | 2009-04-20 16:00:52 -0500 (Mon, 20 Apr 2009) | 2 lines
    
    Don't treat a NOANSWER like a CHANUNAVAIL
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2009-04-20 21:32:38 +00:00
Sean Bright
c3a9db2bf6 Merged revisions 189464 via svnmerge from
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  r189464 | seanbright | 2009-04-20 17:09:59 -0400 (Mon, 20 Apr 2009) | 20 lines
  
  Merged revisions 189462 via svnmerge from 
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  ........
    r189462 | seanbright | 2009-04-20 16:58:39 -0400 (Mon, 20 Apr 2009) | 13 lines
    
    Properly handle @s within hints in AEL.
    
    AEL was not handling the case of a device hint containing an @ symbol, which
    caused parking hints (e.g. hint(park:exten@context)) to error out the parser.
    This patch makes AEL treat the @ the same way it treats colon and ampersand
    now, meaning the characters are included in verbatim.
    
    (closes issue #14941)
    Reported by: bpgoldsb
    Patches:
          bug14941.patch uploaded by seanbright (license 71)
    Tested by: bpgoldsb
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2009-04-20 21:31:36 +00:00
Sean Bright
b48cda78d3 Remove trunk-merge property (trunk-merged is still here)
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2009-04-20 21:21:49 +00:00
Doug Bailey
e193045198 Merged revisions 189419 via svnmerge from
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  r189419 | dbailey | 2009-04-20 14:28:16 -0500 (Mon, 20 Apr 2009) | 11 lines
  
  Merged revisions 189391 via svnmerge from 
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  ........
    r189391 | dbailey | 2009-04-20 14:10:56 -0500 (Mon, 20 Apr 2009) | 4 lines
    
    Clean up problem with manager implementation of mmap where it was not testing against MAP_FAILED response.
    Got rid of shadowed variable used in processign the mmap results. 
    Change test of mmap results to compare against MAP_FAILED
  ........
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2009-04-20 19:37:29 +00:00
Joshua Colp
0294285624 Blocked revisions 189350 via svnmerge
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  r189350 | file | 2009-04-20 14:05:15 -0300 (Mon, 20 Apr 2009) | 10 lines
  
  Fix a bug with non-UDP connections that caused dialogs to not get freed.
  
  This issue crept up because of a reference count issue on non-UDP based dialogs.
  The dialog reference count was increased when transmitting a packet reliably but never
  decreased. This caused the dialog structure to hang around despite being unlinked from
  the dialogs container.
  
  (closes issue #14919)
  Reported by: vrban
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2009-04-20 17:06:20 +00:00
Mark Michelson
ecc6d93b08 Merged revisions 189278 via svnmerge from
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  r189278 | mmichelson | 2009-04-20 09:05:27 -0500 (Mon, 20 Apr 2009) | 18 lines
  
  Merged revisions 189277 via svnmerge from 
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  ........
    r189277 | mmichelson | 2009-04-20 09:04:41 -0500 (Mon, 20 Apr 2009) | 12 lines
    
    Move the check for chan->fdno == -1 to after the zombie/hangup check.
    
    Many users were finding that their hung up channels were staying up and
    causing 100% CPU usage.
    
    (issue #14723)
    Reported by: seadweller
    Patches:
          14723_1-4-tip.patch uploaded by mmichelson (license 60)
    Tested by: falves11, bamby
  ........
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2009-04-20 14:05:53 +00:00
David Vossel
0eb7c1a0c7 Merged revisions 189204 via svnmerge from
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  r189204 | dvossel | 2009-04-17 20:28:45 -0500 (Fri, 17 Apr 2009) | 18 lines
  
  Merged revisions 189203 via svnmerge from 
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  ........
    r189203 | dvossel | 2009-04-17 20:27:19 -0500 (Fri, 17 Apr 2009) | 12 lines
    
    Fixed autologoff in agents.conf not working when agent logs in via AgentLogin app
    
    An agent logs in by calling an extension that calls the AgentLogin app.  In agents.conf ackcall=always is set, so when they get a call they have the choice to either acknowledge it or ignore it.  autologoff=10 is set as well, so if the agent ignores the call over 10sec one may assume that the agent should be logged out (and in this case hungup on as well), but this was not happening.
    
    (closes issue #14091)
    Reported by: evandro
    Patches:
          autologoff.diff uploaded by dvossel (license 671)
    
    Review: http://reviewboard.digium.com/r/225/
  ........
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2009-04-18 01:37:24 +00:00
Richard Mudgett
72a1f1a45c Merged revisions 189137 via svnmerge from
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  r189137 | rmudgett | 2009-04-17 16:48:10 -0500 (Fri, 17 Apr 2009) | 17 lines
  
  Merged revisions 188833,189134 via svnmerge from 
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  ........
    r188833 | rmudgett | 2009-04-16 16:37:58 -0500 (Thu, 16 Apr 2009) | 4 lines
    
    Only disable mISDN DSP if Asterisk DSP is enabled. Leave jitter setting alone.
    
    JIRA ABE-1835
  ........
    r189134 | rmudgett | 2009-04-17 16:27:55 -0500 (Fri, 17 Apr 2009) | 4 lines
    
    Modifed/added some debug messages.
    
    JIRA ABE-1835
  ........
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2009-04-17 21:51:57 +00:00
Mark Michelson
58705bdf54 Merged revisions 189097 via svnmerge from
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  r189097 | mmichelson | 2009-04-17 15:20:23 -0500 (Fri, 17 Apr 2009) | 13 lines
  
  Prevent a crash when SIP blonde transferring an unbridged call.
  
  If one attempts to use the attended transfer button on a SIP phone
  to transfer an unbridged call (such as a call to an IVR) but hangs
  up while the target of the transfer is still ringing, we need to not
  crash.
  
  The problem was that ast_hangup was called from outside the channel
  thread.
  
  AST-211
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2009-04-17 20:21:05 +00:00
Sean Bright
9765afd83e Merged revisions 189077 via svnmerge from
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  r189077 | seanbright | 2009-04-17 15:36:38 -0400 (Fri, 17 Apr 2009) | 1 line
  
  Fix copy/paste error with 'transmit silence' flag.
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2009-04-17 19:44:11 +00:00
Matthew Nicholson
73426fbc34 Merged revisions 189010 via svnmerge from
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  r189010 | mnicholson | 2009-04-17 10:44:18 -0500 (Fri, 17 Apr 2009) | 12 lines
  
  Merged revisions 189009 via svnmerge from 
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  ........
    r189009 | mnicholson | 2009-04-17 10:43:09 -0500 (Fri, 17 Apr 2009) | 5 lines
    
    Make Busy() application set the CDR disposition to BUSY.
    
    (closes issue #14306)
    Reported by: cristiandimache
  ........
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2009-04-17 17:33:36 +00:00
Joshua Colp
0bc53ef61e Merged revisions 188947 via svnmerge from
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  r188947 | file | 2009-04-17 11:44:56 -0300 (Fri, 17 Apr 2009) | 22 lines
  
  Merged revisions 188946 via svnmerge from 
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  ........
    r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) | 15 lines
    
    Fix a bug where a value used to create the channel name was bogus.
    
    This commit fixes the scenario where an incoming call is authenticated
    using a peer entry. Previously the channel name was created using either
    the username setting from the sip.conf entry or the IP address that the
    call came from. Now the channel name will be created using the peer name
    itself. This commit will not change the way the channel name is generated
    for users or friends.
    
    (closes issue #14256)
    Reported by: Nick_Lewis
    Patches:
          chan_sip.c-chname.patch uploaded by Nick (license 657)
    Tested by: Nick_Lewis, file
  ........
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2009-04-17 14:46:49 +00:00
Mark Michelson
4017eafb3e Blocked revisions 188942 via svnmerge
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  r188942 | mmichelson | 2009-04-17 09:33:50 -0500 (Fri, 17 Apr 2009) | 5 lines
  
  Fix a spacing issue that I claimed I would when I committed this code.
  
  Nothing major though.
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2009-04-17 14:33:59 +00:00
Joshua Colp
4ff1323d32 Merged revisions 188938 via svnmerge from
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  r188938 | file | 2009-04-17 11:26:53 -0300 (Fri, 17 Apr 2009) | 11 lines
  
  Merged revisions 188937 via svnmerge from 
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  ........
    r188937 | file | 2009-04-17 11:25:57 -0300 (Fri, 17 Apr 2009) | 4 lines
    
    Fix a situation where the DAHDI channel private structure lock was not unlocked when it should have been.
    
    (issue AST-210)
  ........
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2009-04-17 14:27:38 +00:00
Mark Michelson
7aacba571a Blocked revisions 188901 via svnmerge
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  r188901 | mmichelson | 2009-04-17 08:29:33 -0500 (Fri, 17 Apr 2009) | 28 lines
  
  Several fixes to the extenpatternmatchnew logic.
  
  1. Differentiate between literal characters in an extension
  and characters that should be treated as a pattern match. Prior to
  these fixes, an extension such as NNN would be treated as a pattern,
  rather than a literal string of N's.
  
  2. Fixed the logic used when matching an extension with a bracketed
  expression, such as 2[5-7]6.
  
  3. Removed all areas of code that were executed when NOT_NOW was
  #defined. The code in these areas had the potential to crash, for
  one thing, and the actual intent of these blocks seemed counterproductive.
  
  4. Fixed many many coding guidelines problems I encountered while looking
  through the corresponding code.
  
  5. Added failure cases and warning messages for when duplicate extensions
  are encountered.
  
  6. Miscellaneous fixes to incorrect or redundant statements.
  
  (closes issue #14615)
  Reported by: steinwej
  Tested by: mmichelson
  
  Review: http://reviewboard.digium.com/r/194/
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2009-04-17 13:30:42 +00:00
Tilghman Lesher
fdfaea10c0 Merged revisions 188836 via svnmerge from
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  r188836 | tilghman | 2009-04-16 16:57:37 -0500 (Thu, 16 Apr 2009) | 14 lines
  
  Merged revisions 188835 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r188835 | tilghman | 2009-04-16 16:41:13 -0500 (Thu, 16 Apr 2009) | 7 lines
    
    Only update realtime, if global option rtupdate != false
    (closes issue #14885)
     Reported by: deepesh
     Patches: 
           20090413__bug14885.diff.txt uploaded by tilghman (license 14)
     Tested by: deepesh
  ........
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2009-04-16 22:03:35 +00:00
Tilghman Lesher
5fad34cd42 Merged revisions 188774 via svnmerge from
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  r188774 | tilghman | 2009-04-16 16:03:31 -0500 (Thu, 16 Apr 2009) | 11 lines
  
  Merged revisions 188773 via svnmerge from 
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  ........
    r188773 | tilghman | 2009-04-16 16:02:29 -0500 (Thu, 16 Apr 2009) | 4 lines
    
    Umask should not be exported into global namespace.
    (closes issue #14912)
     Reported by: jcapp
  ........
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2009-04-16 21:04:53 +00:00
David Vossel
591eeaa505 Merged revisions 188647 via svnmerge from
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  r188647 | dvossel | 2009-04-15 17:10:04 -0500 (Wed, 15 Apr 2009) | 18 lines
  
  Merged revisions 188646 via svnmerge from 
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  ........
    r188646 | dvossel | 2009-04-15 17:08:40 -0500 (Wed, 15 Apr 2009) | 12 lines
    
    National prefix inserted even when caller ID not available
    
    When the caller ID is restricted, the expected behavior is for the caller id to be blank.  In chan_dahdi, the national prefix is placed onto the callers number even if its restricted (empty) causing the caller id to be the national prefix rather than blank.
    
    (closes issue #13207)
    Reported by: shawkris
    Patches:
          national_prefix.diff uploaded by dvossel (license 671)
    
    Review: http://reviewboard.digium.com/r/220/
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2009-04-15 22:11:48 +00:00
Mark Michelson
dccea3f82b Merged revisions 188585 via svnmerge from
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  r188585 | mmichelson | 2009-04-15 15:17:33 -0500 (Wed, 15 Apr 2009) | 13 lines
  
  Merged revisions 188582 via svnmerge from 
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  ........
    r188582 | mmichelson | 2009-04-15 15:04:20 -0500 (Wed, 15 Apr 2009) | 7 lines
    
    Update ast_readvideo_callback to match ast_readaudio_callback.
    
    This fixes potential refcount errors that may occur on ast_filestreams.
    
    AST-208
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2009-04-15 20:19:02 +00:00
Mark Michelson
072398b324 Blocked revisions 188544 via svnmerge
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  r188544 | mmichelson | 2009-04-15 10:24:50 -0500 (Wed, 15 Apr 2009) | 9 lines
  
  Make the cancellation of the dial timeout on a call forward optional.
  
  This introduces the 'z' option to app_dial. With it set, a call forward
  will cancel any timeout originally set for this instance of the Dial
  application.
  
  AST-207
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2009-04-15 16:14:49 +00:00
Jeff Peeler
80c84cd933 Blocked revisions 188515 via svnmerge
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  r188515 | jpeeler | 2009-04-15 09:57:20 -0500 (Wed, 15 Apr 2009) | 3 lines
  
  Don't try to do anything in pri_check_restart with service messages unless
  libpri supports it.
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Mark Michelson
0e5b39e00c Merged revisions 188470 via svnmerge from
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  r188470 | mmichelson | 2009-04-14 18:28:13 -0500 (Tue, 14 Apr 2009) | 3 lines
  
  Fix a couple of queue member reference leaks.
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2009-04-14 23:28:39 +00:00
Joshua Colp
a71333f009 Merged revisions 188413 via svnmerge from
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  r188413 | file | 2009-04-14 14:40:50 -0300 (Tue, 14 Apr 2009) | 5 lines
  
  Fix an incorrect clock rate when sending T140 text.
  
  (closes issue #14029)
  Reported by: epicac
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2009-04-14 17:42:11 +00:00
Jeff Peeler
f6d8375580 Blocked revisions 188378 via svnmerge
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  r188378 | jpeeler | 2009-04-14 11:49:12 -0500 (Tue, 14 Apr 2009) | 1 line
  
  change some capitalization
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Jeff Peeler
3abc73ab7f Blocked revisions 188342 via svnmerge
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  r188342 | jpeeler | 2009-04-14 10:54:16 -0500 (Tue, 14 Apr 2009) | 33 lines
  
  Add service maintenance message support
  
  This is the companion commit to libpri r732. Service messages are now supported
  for switch types 4ess/5ess. A new option service_message_support has been added
  to chan_dahdi.conf and is noted in the sample config file. The service message
  support is turned off by default. The current implementation relies on AstDB
  to keep track of channel state, which allows the statuses to be preserved
  across Asterisk restarts. Below is a description of the storage format.
  
  The state and reason for the service state are in the form <state>:<reason>,
  where:
  <state> ::= { 'O' }  // 'O' – Out Of Service
  <reason> ::= { '0' | '1' | '2' | '3' }, where:
  '0' – No reason (backwards compatibility)
  '1' – NEAR END
  '2' – FAR END
  '3' – both NEAR and FAR END
  
  The new CLI commands to handle channel service state are:
  pri service disable channel <chan>
  pri service enable channel <chan>
  
  Many people contributed to the development of this functionality. Because I
  entered at the very end I do not know the exact history. Special thanks to 
  all who moved the bug forward one way or another:
  cmaj, PCadach, markster, mattf, drmac, MikeJ, serge-v, murf, kanelbullar, Seb7,
  tilghman, lmadsen, and especially dhubbard (he answered lots of my questions
  and did a large portion of the work)
  
  (closes issue #3450)
  Reported by: cmaj
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Joshua Colp
eac8868bc6 Merged revisions 188247 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r188247 | file | 2009-04-14 10:14:21 -0300 (Tue, 14 Apr 2009) | 7 lines
  
  Fix a bug with the change I made yesterday to outbound proxy support.
  
  Per discussion with oej on IRC we need the actual IP address, not the
  outbound proxy IP address, in the sa field. This change matches the already
  existing code for all other uses of the outbound proxy setting.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@188248 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-14 13:16:05 +00:00
Tilghman Lesher
b7c591df10 Merged revisions 188210 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r188210 | tilghman | 2009-04-14 00:45:13 -0500 (Tue, 14 Apr 2009) | 2 lines
  
  As suggested by Russell, warn users when their dialplan arguments contain pipes, but not commas.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@188211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-14 05:45:55 +00:00
Tilghman Lesher
45c17d3e70 Merged revisions 188206 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r188206 | tilghman | 2009-04-14 00:27:53 -0500 (Tue, 14 Apr 2009) | 6 lines
  
  Application delimiter is ',', not '|'.
  (closes issue #14881)
   Reported by: stegro
   Patches: 
         smsq.patch uploaded by stegro (license 752)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@188207 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-14 05:28:33 +00:00
Mark Michelson
717ba5fdb9 Merged revisions 188102 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r188102 | mmichelson | 2009-04-13 14:31:48 -0500 (Mon, 13 Apr 2009) | 5 lines
  
  Fix another crash related to cached realtime music on hold.
  
  This was another off-by-one problem caused by moh_register.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@188103 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-13 19:32:43 +00:00
Joshua Colp
24f7a42dc5 Merged revisions 188067 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r188067 | file | 2009-04-13 13:28:06 -0300 (Mon, 13 Apr 2009) | 10 lines
  
  Fix a bug where using an outbound proxy would cause the local address to be 127.0.0.1.
  
  Copy the outbound proxy IP address into the SIP dialog structure as the IP address we will
  be sending to. This has to be done because the logic that determines what local IP address to use
  in the SIP messages is not aware of an outbound proxy being in place. It only knows what IP address
  we are sending to.
  
  (closes issue #12006)
  Reported by: mnicholson
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@188068 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-13 16:30:43 +00:00
Mark Michelson
c8e4eab26d Merged revisions 188032 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r188032 | mmichelson | 2009-04-13 09:17:56 -0500 (Mon, 13 Apr 2009) | 6 lines
  
  Set all queue variables on both the caller and member channels.
  
  This allows for the variables to be accessed if a member macro is run.
  Thanks to Grigoriy Puzankin for bringing this up on the -dev list.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@188037 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-13 14:20:14 +00:00
Jeff Peeler
6bf2b92729 Merged revisions 187906 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r187906 | jpeeler | 2009-04-10 15:26:46 -0500 (Fri, 10 Apr 2009) | 12 lines
  
  Fix module embedding for chan_h323.
  
  Include libchanh323.a in the modules.link file so that all the symbols can be
  resolved at link time.
  
  (closes issue #11966)
  Reported by: dome
  Patches:
        issue_11966.patch uploaded by kpfleming (license 421)
  Tested by: jpeeler
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@187912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-10 20:27:58 +00:00
Mark Michelson
c73e08259f Blocked revisions 187830 via svnmerge
........
  r187830 | mmichelson | 2009-04-10 13:56:52 -0500 (Fri, 10 Apr 2009) | 3 lines
  
  Indicating connected line or redirecting updates were missing a call to lock the local_pvt.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@187831 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-10 18:57:16 +00:00
Tilghman Lesher
5ce204bf7b Merged revisions 187764 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r187764 | tilghman | 2009-04-10 12:29:34 -0500 (Fri, 10 Apr 2009) | 9 lines
  
  Merged revisions 187763 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r187763 | tilghman | 2009-04-10 12:28:46 -0500 (Fri, 10 Apr 2009) | 2 lines
    
    Add lastms column to the contributed table designs
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@187765 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-10 17:30:11 +00:00
Kevin P. Fleming
bcfe1ecaed Merged revisions 187721 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r187721 | kpfleming | 2009-04-10 11:51:44 -0500 (Fri, 10 Apr 2009) | 5 lines
  
  clean up some patterns for files to remove
  
  add embedding support for bridge and test modules
........


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2009-04-10 16:52:45 +00:00
Mark Michelson
bad7592310 Blocked revisions 187714 via svnmerge
........
  r187714 | mmichelson | 2009-04-10 11:26:48 -0500 (Fri, 10 Apr 2009) | 3 lines
  
  ast_strdup failures aren't really failures if the original value was NULL.
........


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2009-04-10 16:27:02 +00:00
Mark Michelson
5becdf8cc9 Blocked revisions 187680 via svnmerge
........
  r187680 | mmichelson | 2009-04-10 11:06:22 -0500 (Fri, 10 Apr 2009) | 5 lines
  
  Don't let ast_channel_alloc fail if explicitly passed NULL cid_name or cid_number.
  
  This also fixes a small memory leak.
........


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2009-04-10 16:06:56 +00:00
Richard Mudgett
dffebb54e1 Blocked revisions 187634-187635 via svnmerge
........
  r187634 | rmudgett | 2009-04-10 09:50:42 -0500 (Fri, 10 Apr 2009) | 1 line
  
  Make chan_misdn_log() avoid generating the log message if logging is disabled.
........
  r187635 | rmudgett | 2009-04-10 09:53:59 -0500 (Fri, 10 Apr 2009) | 7 lines
  
  Miscellaneous minor changes to chan_misdn.
  
  * Miscellaneous spacing and comment changes.
  * Minor code rearangements.
  * Miscellaneous doxygen comments.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@187670 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-10 15:26:06 +00:00
Kevin P. Fleming
b146ea1ee5 Blocked revisions 187636 via svnmerge
........
  r187636 | kpfleming | 2009-04-10 10:11:16 -0500 (Fri, 10 Apr 2009) | 3 lines
  
  revert addition of LOG_SECURITY log channel; after further discussion, a much better solution will be used
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@187637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-10 15:12:21 +00:00
Mark Michelson
5ddfec2a21 Blocked revisions 187560 via svnmerge
........
  r187560 | mmichelson | 2009-04-09 16:06:26 -0500 (Thu, 09 Apr 2009) | 11 lines
  
  Add a new option, mwi_from, to sip.conf.
  
  This allows for you to change the From header for outgoing MWI
  NOTIFY requests. Prior to this, the best you could do was to
  set a callerid in the general section of sip.conf. The problem
  was that this was used for all outbound requests, not just
  MWI NOTIFY requests.
  
  AST-201
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@187561 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-09 21:06:40 +00:00
Mark Michelson
9145d4bbc8 Merge revision 187488 to 1.6.0.
A note to all of you. Don't block revisions in a branch if you actually
meant to merge them. Two very old revisions somehow didn't get merged into
1.6.0 and this change was dependent on those two old revisions. What should have
taken 2 minutes has now wasted about 30 minutes of my time :(



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@187555 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-09 20:14:33 +00:00
Mark Michelson
f10c9a61b3 Merged revisions 141810,141868 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r141810 | mmichelson | 2008-09-08 16:18:49 -0500 (Mon, 08 Sep 2008) | 22 lines
  
  Merged revisions 141809 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
  r141809 | mmichelson | 2008-09-08 16:10:10 -0500 (Mon, 08 Sep 2008) | 14 lines
  
  Fix pedantic mode of chan_sip to only check the
  remote tag of an endpoint once a dialog has
  been confirmed. Up until that point, it is possible
  and legal for the far-end to send provisional
  responses with a different To: tag each time. With
  this patch applied, these provisional messages
  will not cause a matching problem.
  
  (closes issue #11536)
  Reported by: ibc
  Patches:
        11536v2.patch uploaded by putnopvut (license 60)
  
  
  ........
................
  r141868 | mmichelson | 2008-09-08 17:14:40 -0500 (Mon, 08 Sep 2008) | 4 lines
  
  Um, apparently I didn't actually finish merging before committing.
  Bad bad bad
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@187554 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-09 20:09:13 +00:00
Mark Michelson
1f9a91a17b And another one, too.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@187553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-09 19:56:11 +00:00
Mark Michelson
667812ef7a Remove a revision from the trunk-blocked property since it should have been merged instead.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@187552 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-09 19:55:11 +00:00
Jeff Peeler
95760a36c1 Blocked revisions 187491 via svnmerge
........
  r187491 | jpeeler | 2009-04-09 14:10:02 -0500 (Thu, 09 Apr 2009) | 15 lines
  
  Add ability for dialplan execution to continue when caller hangs up.
  
  The F option to app_dial has been modified to accept no parameters and perform
  the above functionality. I don't see anywhere else that is doing function
  overloading, but this really is the best place for this operation because:
  
  - It makes it close to the 'g' option in the argument list which provides
  similar functionality.
  - The existing code to support the current F option provides a very
  convienient location to add this new feature.
  
  (closes issue #12381)
  Reported by: michael-fig
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@187492 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-09 19:10:28 +00:00
Tilghman Lesher
3d7bac20f6 typo
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@187490 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-09 19:03:50 +00:00
Tilghman Lesher
9a69733bc2 Merged revisions 187483 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r187483 | tilghman | 2009-04-09 13:40:01 -0500 (Thu, 09 Apr 2009) | 15 lines
  
  Merged revisions 187428 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r187428 | tilghman | 2009-04-09 13:08:20 -0500 (Thu, 09 Apr 2009) | 8 lines
    
    Race condition between ast_cli_command() and 'module unload' could cause a deadlock.
    Add lock timeouts to avoid this potential deadlock.
    (closes issue #14705)
     Reported by: jamessan
     Patches: 
           20090320__bug14705.diff.txt uploaded by tilghman (license 14)
     Tested by: jamessan
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@187485 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-09 18:52:58 +00:00
Mark Michelson
43036da2b4 Merged revisions 187421,187424 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r187421 | mmichelson | 2009-04-09 12:30:39 -0500 (Thu, 09 Apr 2009) | 21 lines
  
  Fix a crash in res_musiconhold when using cached realtime moh.
  
  The moh_register function links an mohclass and then immediately
  unrefs the class since the container now has a reference. The problem
  with using realtime music on hold is that the class is allocated,
  registered, and started in one fell swoop. The refcounting logic 
  resulted in the count being off by one. The same problem did not
  happen when using a static config because the allocation and registration
  of an mohclass is a separate operation from starting moh. This also did
  not affect non-cached realtime moh because the classes are not registered
  at all.
  
  I also have modified res_musiconhold to use the _t_ variants of the ao2_
  functions so that more info can be gleaned when attempting to trace the
  refcounts. I found this to be incredibly helpful for debugging this issue
  and there's no good reason to remove it.
  
  (closes issue #14661)
  Reported by: sum
........
  r187424 | mmichelson | 2009-04-09 12:34:39 -0500 (Thu, 09 Apr 2009) | 3 lines
  
  Use safe macro practices even though they really aren't necessary.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@187425 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-09 17:37:50 +00:00
Tilghman Lesher
92b4eb8d40 Merged revisions 187363 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r187363 | tilghman | 2009-04-09 11:39:43 -0500 (Thu, 09 Apr 2009) | 10 lines
  
  Merged revisions 187362 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r187362 | tilghman | 2009-04-09 11:38:37 -0500 (Thu, 09 Apr 2009) | 3 lines
    
    Permit zero-length text messages in SIP.
    (Related to an issue posted to the -users list, subject "AEL2, BASE64_DECODE and hexadecimal")
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@187364 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-09 16:40:29 +00:00
Tilghman Lesher
ec49f90f69 Merged revisions 187302 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r187302 | tilghman | 2009-04-08 23:59:05 -0500 (Wed, 08 Apr 2009) | 14 lines
  
  Merged revisions 187300-187301 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r187300 | tilghman | 2009-04-08 23:31:38 -0500 (Wed, 08 Apr 2009) | 3 lines
    
    Add debugging mode for diagnosing file descriptor leaks.
    (Related to issue #14625)
  ........
    r187301 | tilghman | 2009-04-08 23:32:40 -0500 (Wed, 08 Apr 2009) | 2 lines
    
    Oops, missed this file in the last commit.
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@187304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-09 05:16:06 +00:00
Kevin P. Fleming
d9b8b54a9a Blocked revisions 187269 via svnmerge
........
  r187269 | kpfleming | 2009-04-08 22:44:27 -0400 (Wed, 08 Apr 2009) | 5 lines
  
  add a dedicated log channel for modules to be able report security-related events, so that they can be fed into external processes for analysis and possible mitigation efforts
  
  (inspired by this evening's Toronto Asterisk Users Group meeting and previous dicussions amongst various community members)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@187270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-09 02:45:29 +00:00
Jeff Peeler
6a1c421db9 Blocked revisions 187211 via svnmerge
........
  r187211 | jpeeler | 2009-04-08 16:00:39 -0500 (Wed, 08 Apr 2009) | 20 lines
  
  Add timer for features so that backup bridge config can go away
  
  The biggest change done here was elimination of the backup_config for use with
  features. Previously, the bridging code upon detecting a feature would set the
  start time of the bridge to the start time of the feature. Then after the 
  feature had either expired or timed out the start time would be reset to the
  true bridge start time from the backup_config. Now, the time differences are
  calculated with respect to the newly added feature_start_time timeval instead.
  
  There should be no behavior changes from the previous functionality aside from
  the bridge timing being unaffected by either valid or partial feature matches.
  Previously the timing would be increased by the length of time configured for
  featuredigittimeout, which was probably never noticed.
  
  (closes issue #14503)
  Reported by: KNK
  Tested by: jpeeler
  
  Review: http://reviewboard.digium.com/r/179/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@187212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-08 21:01:33 +00:00
Mark Michelson
e933d93cab Blocked revisions 187138 via svnmerge
................
  r187138 | mmichelson | 2009-04-08 14:18:10 -0500 (Wed, 08 Apr 2009) | 13 lines
  
  Blocked revisions 187135 via svnmerge
  
  ........
    r187135 | mmichelson | 2009-04-08 14:16:49 -0500 (Wed, 08 Apr 2009) | 8 lines
    
    Fix a crash due to too few arguments to RetryDial.
    
    (closes issue #14852)
    Reported by: junky
    Patches:
          retry_fix.diff uploaded by junky (license 177)
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@187141 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-08 19:18:35 +00:00
Mark Michelson
3145a3c6b1 Merged revisions 187046 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r187046 | mmichelson | 2009-04-08 11:52:20 -0500 (Wed, 08 Apr 2009) | 16 lines
  
  Merged revisions 187045 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r187045 | mmichelson | 2009-04-08 11:52:03 -0500 (Wed, 08 Apr 2009) | 10 lines
    
    Fix a small logical error when loading moh classes.
    
    We were unconditionally incrementing the number of mohclasses
    registered. However, we should actually only increment if the
    call to moh_register was successful.
    
    While this probably has never caused problems, I noticed it
    and decided to fix it anyway.
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@187047 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-08 16:52:52 +00:00
Jason Parker
24ec838bbf That's how tired I was... I read misdn.conf. Should've been MGCP
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@186991 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-08 16:07:13 +00:00
Jason Parker
51affb6e81 Add missing line to CHANGES. Where did it go? Don't know!
Thanks to thehar for reporting this for me.  I noticed this at about 1:00am last night and just wanted to go to bed.

(closes issue #14853)
Reported by: thehar


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@186990 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-08 16:05:58 +00:00
Mark Michelson
82c0c7822e Merged revisions 186985 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r186985 | mmichelson | 2009-04-08 10:27:41 -0500 (Wed, 08 Apr 2009) | 30 lines
  
  Merged revisions 186984 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r186984 | mmichelson | 2009-04-08 10:26:46 -0500 (Wed, 08 Apr 2009) | 24 lines
    
    Make a couple of changes with regards to a new message printed in ast_read().
    
    "ast_read() called with no recorded file descriptor" is a new message added
    after a bug was discovered. Unfortunately, it seems there are a bunch of places
    that potentially make such calls to ast_read() and trigger this error message
    to be displayed. This commit does two things to help to make this message appear
    less.
    
    First, the message has been downgraded to a debug level message if dev mode is
    not enabled. The message means a lot more to developers than it does to end users,
    and so developers should take an effort to be sure to call ast_read only when
    a channel is ready to be read from. However, since this doesn't actually cause an
    error in operation and is not something a user can easily fix, we should not spam
    their console with these messages.
    
    Second, the message has been moved to after the check for any pending masquerades.
    ast_read() being called with no recorded file descriptor should not interfere with
    a masquerade taking place.
    
    This could be seen as a simple way of resolving issue #14723. However, I still want
    to try to clear out the existing ways of triggering this message, since I feel that
    would be a better resolution for the issue.
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@186986 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-08 15:28:22 +00:00
Mark Michelson
4ce04d6865 Merged revisions 186842 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r186842 | mmichelson | 2009-04-07 19:09:28 -0500 (Tue, 07 Apr 2009) | 14 lines
  
  Merged revisions 186841 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r186841 | mmichelson | 2009-04-07 19:09:04 -0500 (Tue, 07 Apr 2009) | 8 lines
    
    Fix a few typos of the word "frequency."
    
    (closes issue #14842)
    Reported by: jvandal
    Patches:
          frequency-typo.diff uploaded by jvandal (license 413)
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@186843 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-08 00:09:54 +00:00
Mark Michelson
f30f904099 Merged revisions 186837 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r186837 | mmichelson | 2009-04-07 19:01:49 -0500 (Tue, 07 Apr 2009) | 7 lines
  
  Fix bad merge from fix for issue 13867.
  
  (closes issue #14686)
  Reported by: davidw
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@186838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-08 00:02:19 +00:00
Mark Michelson
d3b10cb2a5 Merged revisions 186833 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r186833 | mmichelson | 2009-04-07 18:50:56 -0500 (Tue, 07 Apr 2009) | 15 lines
  
  Merged revisions 186832 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r186832 | mmichelson | 2009-04-07 18:49:49 -0500 (Tue, 07 Apr 2009) | 8 lines
    
    Set the AST_FEATURE_WARNING_ACTIVE flag when a p2p bridge returns AST_BRIDGE_RETRY.
    
    Without this flag set, warning sounds will not be properly played to either party
    of the bridge.
    
    (closes issue #14845)
    Reported by: adomjan
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@186834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-07 23:51:25 +00:00
Tilghman Lesher
cfe5f3e0fc Merged revisions 186799 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r186799 | tilghman | 2009-04-07 17:23:46 -0500 (Tue, 07 Apr 2009) | 10 lines
  
  Merged revisions 186775 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r186775 | tilghman | 2009-04-07 17:16:50 -0500 (Tue, 07 Apr 2009) | 3 lines
    
    Fix Macro documentation to match current (and intended) behavior.
    (See -dev mailing list)
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@186805 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-07 22:30:27 +00:00
Mark Michelson
47387bc563 Merged revisions 186720 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r186720 | mmichelson | 2009-04-07 15:46:18 -0500 (Tue, 07 Apr 2009) | 12 lines
  
  Merged revisions 186719 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r186719 | mmichelson | 2009-04-07 15:43:49 -0500 (Tue, 07 Apr 2009) | 6 lines
    
    Ensure that \r\n is printed after the ActionID in an OriginateResponse.
    
    (closes issue #14847)
    Reported by: kobaz
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@186721 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-07 20:49:54 +00:00
Mark Michelson
44550166d8 Blocked revisions 186620 via svnmerge
........
  r186620 | mmichelson | 2009-04-06 11:06:25 -0500 (Mon, 06 Apr 2009) | 3 lines
  
  Silly svn. These files didn't get merged over in the merge of the issue8824 branch.
........


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2009-04-06 16:07:25 +00:00
Mark Michelson
1a2133f46a Blocked revisions 186566 via svnmerge
................
  r186566 | mmichelson | 2009-04-06 08:57:39 -0500 (Mon, 06 Apr 2009) | 8 lines
  
  Blocked revisions 186565 via svnmerge
  
  ........
    r186565 | mmichelson | 2009-04-06 08:54:41 -0500 (Mon, 06 Apr 2009) | 3 lines
    
    Revert commit 186445 because it causes the build to fail when IMAP_STORAGE is used.
  ........
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2009-04-06 13:58:04 +00:00
Mark Michelson
047e77ad67 Blocked revisions 186525 via svnmerge
........
  r186525 | mmichelson | 2009-04-03 17:41:46 -0500 (Fri, 03 Apr 2009) | 22 lines
  
  This commit introduces COLP/CONP and Redirecting party information into Asterisk.
  
  The channel drivers which have been most heavily tested with these enhancements are
  chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be
  introduced in a later commit. chan_skinny has code added to it here, but according
  to user pj, the support on chan_skinny is not working as of now. This will be fixed in
  a later commit.
  
  A special thanks goes out to bugtracker user gareth for getting the ball rolling and
  providing the initial support for this work. Without his initial work on this, this would
  not have been nearly as painless as it was.
  
  This functionality has been tested by Digium's product quality department, as well as a
  customer site running thousands of calls every day. In addition, many many many many bugtracker
  users have tested this, too.
  
  (closes issue #8824)
  Reported by: gareth
  
  Review: http://reviewboard.digium.com/r/201
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@186526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-03 22:43:19 +00:00
Mark Michelson
0af241cd79 Remove an invalid call to free memory.
A bad merge from trunk to 1.6.0 meant freeing memory that
should not be freed. In trunk, pkt->data is an ast_str, but
in 1.6.0, it is allocated in the same chunk of memory as the
sip_pkt. This only affects 1.6.0.

(closes issue #14819)
Reported by: cwolff09



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@186517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-03 21:27:52 +00:00
Kevin P. Fleming
01cbad2bd6 Merged revisions 186461 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r186461 | kpfleming | 2009-04-03 15:20:01 -0500 (Fri, 03 Apr 2009) | 11 lines
  
  Merged revisions 186458 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r186458 | kpfleming | 2009-04-03 15:19:20 -0500 (Fri, 03 Apr 2009) | 5 lines
    
    Fix a bug where DAHDI/Zaptel channels would not properly switch formats when requested
    
    Don't offer AST_FORMAT_SLINEAR on DAHDI/Zaptel channels... while it could provide a slight performance benefit, the translation core in Asterisk has some flaws when a channel driver offers multiple raw formats. this fix is much simpler than fixing the translation core to solve that issue (although that will be done later).
  ........
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2009-04-03 20:20:52 +00:00
Tilghman Lesher
89c6baed8c Merged revisions 186444 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r186444 | tilghman | 2009-04-03 14:30:34 -0500 (Fri, 03 Apr 2009) | 14 lines
  
  Merged revisions 186415 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r186415 | tilghman | 2009-04-03 14:06:58 -0500 (Fri, 03 Apr 2009) | 7 lines
    
    Distinguish in a sent email between simple sends and forwards.
    (closes issue #11678)
     Reported by: jamessan
     Patches: 
           20090330__bug11678.diff.txt uploaded by tilghman (license 14)
     Tested by: tilghman, lmadsen
  ........
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2009-04-03 19:58:21 +00:00
David Vossel
c248c0cca2 Merged revisions 186379 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r186379 | dvossel | 2009-04-03 11:29:47 -0500 (Fri, 03 Apr 2009) | 4 lines
  
  audio_audiohook_write_list() did not correctly update sample size after ast_translate.
  
  audio_audiohook_write_list() did not take into account that the sample size may change after translation depending on if the original frame is is 8khz or 16khz.  the sample size is now updated after translating to reflect this possibility.  This caused the audio on the receiving end to sound terrible.  Thanks to jcolp and mmichelson for helping me work this out.

  (issue AST-197)
........


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2009-04-03 16:36:44 +00:00
Joshua Colp
01dc8d8ea9 Merged revisions 186321 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r186321 | file | 2009-04-03 12:52:50 -0300 (Fri, 03 Apr 2009) | 12 lines
  
  Merged revisions 186320 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r186320 | file | 2009-04-03 12:48:56 -0300 (Fri, 03 Apr 2009) | 5 lines
    
    Fix a problem with the crypto variable definitions not actually being defined properly.
    
    (closes issue #14804)
    Reported by: jvandal
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@186322 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-03 15:53:32 +00:00
Mark Michelson
968ae201df Merged revisions 186286 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r186286 | mmichelson | 2009-04-03 09:32:05 -0500 (Fri, 03 Apr 2009) | 20 lines
  
  Fix the ability to retrieve voicemail messages from IMAP.
  
  A recent change made interactive vm_states no longer get
  added to the list of vm_states and instead get stored in
  thread-local storage.
  
  In trunk and all the 1.6.X branches, the problem is that
  when we search for messages in a voicemail box, we would
  attempt to update the appropriate vm_state struct by directly
  searching in the list of vm_states instead of using the
  get_vm_state_by_imap_user function. This meant we could not
  find the interactive vm_state that we wanted.
  
  (closes issue #14685)
  Reported by: BlargMaN
  Patches:
        14685.patch uploaded by mmichelson (license 60)
  Tested by: BlargMaN, qualleyiv, mmichelson
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@186287 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-03 14:32:37 +00:00
Russell Bryant
e379077e6f Merged revisions 186230 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
r186230 | russell | 2009-04-02 21:03:48 -0500 (Thu, 02 Apr 2009) | 29 lines

Merged revisions 186229 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r186229 | russell | 2009-04-02 20:57:44 -0500 (Thu, 02 Apr 2009) | 21 lines

Fix a memory leak in cdr_radius.

I came across this while doing some testing of my ast_channel_ao2 branch.
After running a test overnight that generated over 5 million calls, Asterisk
had taken up about 1 GB of my system memory.  So, I re-ran the test with
MALLOC_DEBUG turned on.  However, it showed no leaks in Asterisk during the
test, even though Asterisk was still consuming it somehow.

Instead, I turned to valgrind, which when run with --leak-check=full, told
me exactly where the leak came from, which was from allocations inside the
radiusclient-ng library.  This explains why MALLOC_DEBUG did not report it.

After a bit of analysis, I found that we were leaking a little bit of memory
every time a CDR record was passed to cdr_radius.

I don't actually have a radius server set up to receive CDR records.  However,
I always have my development systems compile and install all modules.  In
addition to making sure there are not build errors across modules, always
loading modules helps find bugs like this, too, so it is strongly recommend for
all developers.

........

................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@186231 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-03 02:05:29 +00:00
Mark Michelson
00e9f4ca9a Merged revisions 186175 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r186175 | mmichelson | 2009-04-02 16:56:21 -0500 (Thu, 02 Apr 2009) | 11 lines
  
  Merged revisions 186174 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r186174 | mmichelson | 2009-04-02 16:55:34 -0500 (Thu, 02 Apr 2009) | 5 lines
    
    Fix instructions in one-step parking comment to make more sense.
    
    Changed a capital K to a lowercase k.
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@186176 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-02 21:58:37 +00:00
Kevin P. Fleming
04670fdac8 Merged revisions 186101 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r186101 | kpfleming | 2009-04-02 12:26:07 -0500 (Thu, 02 Apr 2009) | 9 lines
  
  Merged revisions 186081 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r186081 | kpfleming | 2009-04-02 12:21:29 -0500 (Thu, 02 Apr 2009) | 3 lines
    
    ensure that the buffer passed to DAHDI_SET_BUFINFO is fully initialized
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@186105 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-02 17:27:11 +00:00
Tilghman Lesher
5225fc08dc Merged revisions 186060 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r186060 | tilghman | 2009-04-02 12:10:28 -0500 (Thu, 02 Apr 2009) | 16 lines
  
  Merged revisions 186059 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ................
    r186059 | tilghman | 2009-04-02 12:09:13 -0500 (Thu, 02 Apr 2009) | 9 lines
    
    Merged revisions 186056 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.2
    
    ........
      r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02 Apr 2009) | 2 lines
      
      Fix for AST-2009-003
    ........
  ................
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@186061 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-02 17:12:40 +00:00
Kevin P. Fleming
ee5f6af8c6 Merged revisions 164602 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r164602 | russell | 2008-12-16 08:17:45 -0600 (Tue, 16 Dec 2008) | 7 lines
  
  Fix usage of the DAHDI_VMWI ioctl.
  
  (closes issue #14090)
  Reported by: alecdavis
  Patches:
        chan_dahdi.VMWI_ioctl.diff.txt uploaded by alecdavis (license 585)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@185960 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-02 13:57:07 +00:00
Kevin P. Fleming
7f8d648dc8 this fix should not be blocked from this branch... will merge in a moment
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@185959 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-02 13:56:30 +00:00
Kevin P. Fleming
901b525432 Merged revisions 185953 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r185953 | kpfleming | 2009-04-02 08:51:44 -0500 (Thu, 02 Apr 2009) | 11 lines
  
  Merged revisions 185952 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r185952 | kpfleming | 2009-04-02 08:43:43 -0500 (Thu, 02 Apr 2009) | 5 lines
    
    the DAHDI_GETCONF, DAHDI_SETCONF and DAHDI_GET_PARAMS ioctls were recently corrected to show that they do, in fact, read data from userspace as part of their work. due to this fix, valgrind now reports a number of cases where chan_dahdi passed an uninitialized (or partially) buffer to these ioctls, which could lead to unexpected behavior.
    
    this patch corrects chan_dahdi to ensure that buffers passed to these ioctls are always fully initialized.
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@185954 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-02 13:52:35 +00:00
David Vossel
5e9150506c Merged revisions 185846 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r185846 | dvossel | 2009-04-01 14:03:32 -0500 (Wed, 01 Apr 2009) | 16 lines
  
  Merged revisions 185845 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r185845 | dvossel | 2009-04-01 14:02:00 -0500 (Wed, 01 Apr 2009) | 10 lines
    
    Fixes issue with dropped calles due to re-Invite glare and re-Invites never executing after a 491
    
    Acknowledgement for 491 responses were never being processed because it didn't match our pending invite's seqno.  Since the ACK was never processed, the 491 frame would continue to be retransmitted until eventually the call was dropped due to max retries.  Now during a pending invite, if we receive another invite, we send an 491 and hold on to that glare invite's seqno in the "glareinvite" variable for that sip_pvt struct.  When ACK's are received, we first check to see if it is in response to our pending invite, if not we check to see if it is in response to a glare invite.  In this case, it is in response to the glare invite and must be dealt with or the call is dropped.  I've changed the wait time for resending the re-Invite after receving a 491 response to comply with RFC 3261.  Before this patch the scheduled re-Invite would only change a flag indicating that the re-Invite should be sent out, now it actually sends it out as well. 
    
    (closes issue #12013)
    Reported by: alx
    
    Review: http://reviewboard.digium.com/r/213/
  ........
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2009-04-01 19:05:27 +00:00
Mark Michelson
d4f9141500 Blocked revisions 185777 via svnmerge
........
  r185777 | mmichelson | 2009-04-01 08:59:34 -0500 (Wed, 01 Apr 2009) | 5 lines
  
  Address Russell's comments regarding rev 185704.
  
  Use ast_debug and ast_softhangup_nolock.
........


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2009-04-01 14:00:21 +00:00
Russell Bryant
e0b7350f8a Merged revisions 185772 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
r185772 | russell | 2009-04-01 08:48:26 -0500 (Wed, 01 Apr 2009) | 14 lines

Merged revisions 185771 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r185771 | russell | 2009-04-01 08:47:30 -0500 (Wed, 01 Apr 2009) | 6 lines

Fix a case where DTMF could bypass audiohooks.

This change fixes a situation where an audiohook that wants DTMF would not
actually get it.  This is in the code path where we end DTMF digit length
emulation while handling a NULL frame.

........

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2009-04-01 13:49:26 +00:00
Mark Michelson
96da477709 Blocked revisions 185704 via svnmerge
........
  r185704 | mmichelson | 2009-03-31 19:39:01 -0500 (Tue, 31 Mar 2009) | 8 lines
  
  Allow the AMI Hangup command to accept a Cause header.
  
  (closes issue #14695)
  Reported by: mneuhauser
  Patches:
        cause-for-hangup-manager-action.patch uploaded by mneuhauser (license 425)
........


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2009-04-01 00:41:24 +00:00
Kevin P. Fleming
562fb46389 Merged revisions 185664 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r185664 | kpfleming | 2009-03-31 17:35:07 -0500 (Tue, 31 Mar 2009) | 1 line
  
  ignore copied (generated) file
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2009-03-31 22:37:38 +00:00
Mark Michelson
a64e0f70ea Merged revisions 185600 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r185600 | mmichelson | 2009-03-31 17:02:48 -0500 (Tue, 31 Mar 2009) | 12 lines
  
  Merged revisions 185599 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r185599 | mmichelson | 2009-03-31 17:00:01 -0500 (Tue, 31 Mar 2009) | 6 lines
    
    Fix crash that would occur if an empty member was specified in queues.conf.
    
    (closes issue #14796)
    Reported by: pida
  ........
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2009-03-31 22:03:55 +00:00
Kevin P. Fleming
72f3dfafbc Blocked revisions 185581 via svnmerge
........
  r185581 | kpfleming | 2009-03-31 16:29:50 -0500 (Tue, 31 Mar 2009) | 19 lines
  
  Optimizations to the stringfields API
  
  This patch provides a number of optimizations to the stringfields API, focused around saving (not wasting) memory whenever possible. Thanks to Mark Michelson for inspiring this work and coming up with the first two optimizations that are represented here:
  
  Changes:
  
  - Cleanup of some code, fix incorrect doxygen comments
  
  - When a field is emptied or replaced with a new allocation, decrease the amount of 'active' space in the pool it was held in; if that pool reaches zero active space, and is not the current pool, then free it as it is no longer in use
  
  - When allocating a pool, try to allocate a size that will fit in a 'standard' malloc() allocation without wasting space
  
  - When allocating space for a field, store the amount of space in the two bytes immediately preceding the field; this eliminates the need to call strlen() on the field when overwriting it, and more importantly it 'remembers' the amount of space the field has available, even if a shorter string has been stored in it since it was allocated
  
  - Don't automatically double the size of each successive pool allocated; it's wasteful
  
  http://reviewboard.digium.com/r/165/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@185591 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-31 21:31:58 +00:00
Mark Michelson
411feb27b2 Blocked revisions 185532 via svnmerge
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  r185532 | mmichelson | 2009-03-31 15:56:46 -0500 (Tue, 31 Mar 2009) | 8 lines
  
  Blocked revisions 185531 via svnmerge
  
  ........
    r185531 | mmichelson | 2009-03-31 15:55:47 -0500 (Tue, 31 Mar 2009) | 3 lines
    
    Use AST_SCHED_DEL_SPINLOCK instead of manually using the logic.
  ........
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2009-03-31 20:59:06 +00:00
Mark Michelson
a89b373c43 Merged revisions 185469 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r185469 | mmichelson | 2009-03-31 14:46:18 -0500 (Tue, 31 Mar 2009) | 14 lines
  
  Merged revisions 185468 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r185468 | mmichelson | 2009-03-31 14:45:30 -0500 (Tue, 31 Mar 2009) | 8 lines
    
    Fix Russian voicemail intro to say the word "messages" properly.
    
    (closes issue #14736)
    Reported by: chappell
    Patches:
          voicemail_no_messages.diff uploaded by chappell (license 8)
  ........
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2009-03-31 19:46:52 +00:00
Mark Michelson
eabc30725a Blocked revisions 185299 via svnmerge
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  r185299 | mmichelson | 2009-03-31 10:34:29 -0500 (Tue, 31 Mar 2009) | 15 lines
  
  Blocked revisions 185298 via svnmerge
  
  ........
    r185298 | mmichelson | 2009-03-31 10:34:05 -0500 (Tue, 31 Mar 2009) | 10 lines
    
    Fix some state_interface stuff that was in trunk but not in the backport to 1.4.
    
    Issue #14359 was fixed between the time that I posted the review of the backport
    of the state interface change for 1.4. This merges the changes from that issue
    back into 1.4.
    
    (closes issue #14359)
    Reported by: francesco_r
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2009-03-31 19:39:52 +00:00
David Brooks
bf746fcb47 Merged revisions 185363 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r185363 | dbrooks | 2009-03-31 11:46:57 -0500 (Tue, 31 Mar 2009) | 44 lines
  
  Merged revisions 185362 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 Mar 2009) | 35 lines
    
    Fix incorrect parsing in chan_gtalk when xmpp contains extra whitespaces
    
    To drill into the xmpp to find the capabilities between channels, chan_gtalk 
    calls iks_child() and iks_next(). iks_child() and iks_next() are functions in 
    the iksemel xml parsing library that traverse xml nodes. The bug here is that 
    both iks_child() and iks_next() will return the next iks_struct node 
    *regardless* of type. chan_gtalk expects the next node to be of type IKS_TAG, 
    which in most cases, it is, but in this case (a call being made from the 
    Empathy IM client), there exists iks_struct nodes which are not IKS_TAG data 
    (they are extraneous whitespaces), and chan_gtalk doesn't handle that case, 
    so capabilities don't match, and a call cannot be made.
    
    iks_first_tag() and iks_next_tag(), on the other hand, will not return the 
    very next iks_struct, but will check to see if the next iks_struct is of 
    type IKS_TAG. If it isn't, it will be skipped, and the next struct of type 
    IKS_TAG it finds will be returned. This assures that chan_gtalk will find 
    the iks_struct it is looking for.
    
    This fix simply changes all calls to iks_child() and iks_next() to become 
    calls to iks_first_tag() and iks_next_tag(), which resolves the capability 
    matching.
    
    The following is a payload listing from Empathy, which, due to the extraneous 
    whitespace, will not be parsed correctly by iksemel:
    
    <iq from='dbrooksjab@235-22-24-10/Telepathy' to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'> <session xmlns='http://www.google.com/session' initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate' id='1837267342'> <description xmlns='http://www.google.com/session/phone'> <payload-type clockrate='16000' name='speex' id='96'/>
     <payload-type clockrate='8000' name='PCMA' id='8'/>
     <payload-type clockrate='8000' name='PCMU' id='0'/>
     <payload-type clockrate='90000' name='MPA' id='97'/>
     <payload-type clockrate='16000' name='SIREN' id='98'/>
     <payload-type clockrate='8000' name='telephone-event' id='99'/>
    </description>
    </session>
    </iq>
  
  Review: http://reviewboard.digium.com/r/181/
  ........
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2009-03-31 17:36:07 +00:00
Russell Bryant
cd84ad1fc0 Merged revisions 185261 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r185261 | russell | 2009-03-31 09:53:45 -0500 (Tue, 31 Mar 2009) | 5 lines

Don't free() an astobj2 object.

(closes issue #14672)
Reported by: makoto

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2009-03-31 14:56:27 +00:00
Joshua Colp
d1709254d4 Merged revisions 185197 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r185197 | file | 2009-03-31 11:07:36 -0300 (Tue, 31 Mar 2009) | 15 lines
  
  Merged revisions 185196 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r185196 | file | 2009-03-31 11:06:39 -0300 (Tue, 31 Mar 2009) | 8 lines
    
    Fix crash when moving audiohooks between channels.
    
    Handle the scenario where we are called to move audiohooks between channels
    and the source channel does not actually have any on it.
    
    (closes issue #14734)
    Reported by: corruptor
  ........
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2009-03-31 14:08:42 +00:00
Richard Mudgett
ae65af4244 Merged revisions 185123 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r185123 | rmudgett | 2009-03-30 15:42:14 -0500 (Mon, 30 Mar 2009) | 9 lines
  
  Merged revisions 185121 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r185121 | rmudgett | 2009-03-30 15:40:11 -0500 (Mon, 30 Mar 2009) | 1 line
    
    Update the channel allocation method documentation.
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2009-03-30 20:48:08 +00:00
Richard Mudgett
29aa5c71a5 Merged revisions 185122 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r185122 | rmudgett | 2009-03-30 15:41:24 -0500 (Mon, 30 Mar 2009) | 26 lines
  
  Merged revisions 185120 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r185120 | rmudgett | 2009-03-30 15:38:11 -0500 (Mon, 30 Mar 2009) | 19 lines
    
    Make chan_misdn BRI TE side normally defer channel selection to the NT side.
    
    Channel allocation collisions are not handled by chan_misdn very well.
    This patch simply avoids the problem for BRI only.
    
    For PRI, allocation collisions are still possible but less likely since
    there are simply more channels available and each end could use a different
    allocation strategy.
    
    misdn.conf options available:
    te_choose_channel - Use to force the TE side to allocate channels.
    method - Specify the channel allocation strategy.
    
    (closes issue #13488)
    Reported by: Christian_Pinedo
    Patches:
          isdn_lib.patch.txt uploaded by crich
    Tested by: crich, siepkes, festr
  ........
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2009-03-30 20:46:24 +00:00
Mark Michelson
6ee6d5edac Merged revisions 185072 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r185072 | mmichelson | 2009-03-30 11:26:48 -0500 (Mon, 30 Mar 2009) | 45 lines
  
  Merged revisions 185031 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r185031 | mmichelson | 2009-03-30 11:17:35 -0500 (Mon, 30 Mar 2009) | 39 lines
    
    Fix queue weight behavior so that calls in low-weight queues are not inappropriately blocked.
    
    (This is copied and pasted from the review request I made for this patch)
    
    Asterisk has some odd behavior when queue weights are used. The current logic used when
    potentially calling a queue member is:
    
    If the member we are going to call is part of another queue and _that other queue has any 
    callers in it_ and has a higher weight than the queue we are calling from, then don't try 
    to contact that member. The issue here is what I have marked with underscores. If the 
    higher-weighted queue has any callers in it at all, then the queue member will be unreachable 
    from the lower-weighted queue. This has the potential to be really really bad if using a 
    queue strategy, such as leastrecent or fewestcalls, with the potential to call the same 
    member repeatedly.
    
    The fix proposed by garychen on issue 13220 is very simple and, as far as I can see, works 
    well for this situation. With this set of changes, the logic used becomes:
    
    If the member we are going to call is part of another queue, the other queue has a higher 
    weight than the queue we are calling from, and the higher weight queue has at least as many 
    callers as available members, then do not try to contact the queue member. If the higher 
    weighted queue has fewer callers than available members, then there is no reason to deny 
    the call to this member since the other queue can afford to spare a member.
    
    Since the fix involved writing a generic function for determining the number of available 
    members in the queue, I also modified the is_our_turn function to make use of the new 
    num_available_members function to determine if it is our turn to try calling a member. There 
    is one small behavior change. Before writing this patch, if you had autofill disabled, then 
    if you were the head caller in a queue, you would automatically be told that it was your 
    turn to try calling a member. This did not take into account whether there were actually any 
    queue members available to take the call. Now we actually make sure there is at least one 
    member available to take the call if autofill is disabled.
    
    (closes issue #13220)
    Reported by: garychen
    
    Review: http://reviewboard.digium.com/r/202/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@185087 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-30 16:40:26 +00:00
Mark Michelson
1acb4947a9 Blocked revisions 184986 via svnmerge
................
  r184986 | mmichelson | 2009-03-30 10:25:04 -0500 (Mon, 30 Mar 2009) | 27 lines
  
  Blocked revisions 184980 via svnmerge
  
  ........
    r184980 | mmichelson | 2009-03-30 10:23:59 -0500 (Mon, 30 Mar 2009) | 22 lines
    
    Backport state interface changes to app_queue from trunk.
    
    After several issues raised on the Asterisk bugtracker against
    the 1.4 branch were determined to be fixable with the state interface
    change available in the 1.6.X series, it finally came time to just
    suck it up and backport the change.
    
    For a detailed explanation of what this change entails, the original
    trunk commit for this feature may be found here:
    
    http://svn.digium.com/view/asterisk?view=revision&revision=97203
    
    In addition, the details for the use of this change to fix the problems
    stated in issue #12970 may be found in the review request I made for
    this change. It is linked below.
    
    (closes issue #12970)
    Reported by: edugs15
    
    Review: http://reviewboard.digium.com/r/116
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184989 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-30 15:25:44 +00:00
Joshua Colp
1c3ca72745 Merged revisions 184948 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r184948 | file | 2009-03-30 11:37:47 -0300 (Mon, 30 Mar 2009) | 21 lines
  
  Merged revisions 184947 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r184947 | file | 2009-03-30 11:35:47 -0300 (Mon, 30 Mar 2009) | 14 lines
    
    Improve our handling of T38 in the initial INVITE from a device.
    
    We now answer with matching media streams to what is requested. If an INVITE
    is received with both a T38 and RTP media stream this means we answer with both.
    For any outgoing calls created as a result of this inbound one no T38 is requested
    in the initial INVITE. Instead if we start receiving udptl packets we trigger a
    reinvite on the outbound side.
    
    (closes issue #12437)
    Reported by: marsosa
    Tested by: pinga-fogo, okrief, file, afu
    
    Review: http://reviewboard.digium.com/r/208/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184949 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-30 14:39:32 +00:00
Russell Bryant
87be216c37 Merged revisions 184910 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r184910 | russell | 2009-03-30 08:55:44 -0500 (Mon, 30 Mar 2009) | 4 lines

Fix build error when chan_h323 is not being built.

(reported by cai1982 in #asterisk-dev)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184911 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-30 13:56:57 +00:00
Russell Bryant
2b719265a4 Merged revisions 184843 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r184843 | russell | 2009-03-29 00:52:20 -0500 (Sun, 29 Mar 2009) | 13 lines

Merged revisions 184842 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r184842 | russell | 2009-03-29 00:51:55 -0500 (Sun, 29 Mar 2009) | 5 lines

Ensure targs variable is fully initialized.

(closes issue #14758)
Reported by: tim_ringenbach

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184844 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-29 05:52:42 +00:00
Russell Bryant
f505cb1c08 Merged revisions 184838 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r184838 | russell | 2009-03-29 00:32:04 -0500 (Sun, 29 Mar 2009) | 8 lines

Simplify chan_h323 build to not require a second run of "make".

(closes issue #14715)
Reported by: jthurman
Patches:
      h323-makefile-1.6.0.7-rc2.patch uploaded by jthurman (license 614)
      h323-makefile-1.6.1.0-rc3.patch uploaded by jthurman (license 614)
      h323-makefile-1.6.2.0-beta1.patch uploaded by jthurman (license 614)
Tested by: tzafrir, russell

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2009-03-29 05:43:28 +00:00
Kevin P. Fleming
14c3631493 Blocked revisions 184762 via svnmerge
........
  r184762 | kpfleming | 2009-03-27 14:10:32 -0500 (Fri, 27 Mar 2009) | 12 lines
  
  Improve timing interface to remember which provider provided a timer
  
  The ability to load/unload timing interfaces is nice, but it means that when a timer is allocated, it may come from provider A, but later provider B becomes the 'preferred' provider. If this happens, all timer API calls on the timer that was provided by provider A will actually be handed to provider B, which will say WTF and return an error.
  
  This patch changes the timer API to include a pointer to the provider of the timer handle so that future operations on the timer will be forwarded to the proper provider.
  
  (closes issue #14697)
  Reported by: moy
  
  Review: http://reviewboard.digium.com/r/211/
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2009-03-27 19:11:28 +00:00
Russell Bryant
73096f42bc Merged revisions 184726 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r184726 | russell | 2009-03-27 13:04:43 -0500 (Fri, 27 Mar 2009) | 2 lines

Use ast_random() instead of rand() to ensure we use the best RNG available.

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2009-03-27 18:06:20 +00:00
Joshua Colp
80e628503b Merged revisions 184673 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r184673 | file | 2009-03-27 12:46:46 -0300 (Fri, 27 Mar 2009) | 7 lines
  
  Fix speech structure leak in the AGI speech recognition integration.
  
  The AGI dialplan applications did not destroy the speech structure automatically
  if it was not destroyed by the running AGI script. They will now do this.
  
  (issue LUMENVOX-15) 
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2009-03-27 15:48:16 +00:00
Joshua Colp
28b8ea89dd Merged revisions 184566 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r184566 | file | 2009-03-27 10:15:26 -0300 (Fri, 27 Mar 2009) | 16 lines
  
  Merged revisions 184565 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r184565 | file | 2009-03-27 10:06:45 -0300 (Fri, 27 Mar 2009) | 9 lines
    
    Fix an issue where nat=yes would not always take effect for the RTP session on outgoing calls.
    
    If calls were placed using an IP address or hostname the global nat setting was copied over
    but was not set on the RTP session itself. This caused the RTP stack to not perform symmetric RTP
    actions.
    
    (closes issue #14546)
    Reported by: acunningham
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2009-03-27 13:20:10 +00:00
Russell Bryant
0710549b88 Merged revisions 184515 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r184515 | russell | 2009-03-26 20:40:28 -0500 (Thu, 26 Mar 2009) | 2 lines

Don't act surprised if we get a -1 indication.

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2009-03-27 01:43:13 +00:00
Kevin P. Fleming
8acfb27cbb Merged revisions 184448 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r184448 | kpfleming | 2009-03-26 17:18:14 -0500 (Thu, 26 Mar 2009) | 9 lines
  
  Merged revisions 184447 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r184447 | kpfleming | 2009-03-26 17:17:32 -0500 (Thu, 26 Mar 2009) | 3 lines
    
    use new, improved 8kHz prompts
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2009-03-26 22:18:39 +00:00
David Vossel
56961c9df1 Merged revisions 184389 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r184389 | dvossel | 2009-03-26 16:09:37 -0500 (Thu, 26 Mar 2009) | 14 lines
  
  Merged revisions 184388 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r184388 | dvossel | 2009-03-26 16:07:32 -0500 (Thu, 26 Mar 2009) | 8 lines
    
    pri loop TestClient/TestServer fails: server SEND DTMF 8
    
    app_test was failing when sending the last DTMF digit, 8, because of the 100ms pause issued after DTMF is sent.  During this pause the other side would hang up causing the test to look like it failed. Now the other side waits a second before hanging up.
    
    (closes issue #12442)
    Reported by: tzafrir
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2009-03-26 21:17:44 +00:00
Joshua Colp
b2ca3bfb1e Merged revisions 184280 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r184280 | file | 2009-03-25 16:22:06 -0300 (Wed, 25 Mar 2009) | 5 lines
  
  Fix issue with a T38 reinvite being sent even if not configured to do so.
  
  If we receive a T38 request negotiate control frame we should only attempt to do so
  if the option is enabled on the dialog.
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2009-03-25 19:23:59 +00:00
Eliel C. Sardanons
6028684603 Merged revisions 184220 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r184220 | eliel | 2009-03-25 10:38:19 -0400 (Wed, 25 Mar 2009) | 19 lines
  
  Merged revisions 184188 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r184188 | eliel | 2009-03-25 10:12:54 -0400 (Wed, 25 Mar 2009) | 13 lines
    
    Avoid destroying the CLI line when moving the cursor backward and trying to autocomplete.
    
    When moving the cursor backward and pressing TAB to autocomplete, a NULL is put
    in the line and we are loosing what we have already wrote after the actual
    cursor position.
    
    (closes issue #14373)
    Reported by: eliel
    Patches:
          asterisk.c.patch uploaded by eliel (license 64)
          Tested by: lmadsen
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184222 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-25 14:52:21 +00:00
Russell Bryant
ce37a27180 Merged revisions 184147 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r184147 | russell | 2009-03-24 20:42:10 -0500 (Tue, 24 Mar 2009) | 5 lines
  
  Fix build issues on Mac OSX.
  
  (closes issue #14714)
  Reported by: ygor
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2009-03-25 01:48:17 +00:00
Mark Michelson
ff9f868d0f Merged revisions 184079 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r184079 | mmichelson | 2009-03-24 17:40:39 -0500 (Tue, 24 Mar 2009) | 15 lines
  
  Merged revisions 184078 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r184078 | mmichelson | 2009-03-24 17:34:45 -0500 (Tue, 24 Mar 2009) | 9 lines
    
    Change NULL pointer check to be ast_strlen_zero.
    
    The 'digit' variable is guaranteed to be non-NULL, so the if
    statement could never evaluate true. Changing to ast_strlen_zero
    makes the logic correct.
    
    This was found while reviewing ast_channel_ao2 code review.
  ........
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2009-03-24 22:41:37 +00:00
Russell Bryant
668adab048 Merged revisions 184037 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r184037 | russell | 2009-03-24 16:40:44 -0500 (Tue, 24 Mar 2009) | 6 lines

Exclude slin16, siren7, and siren14 from bandwidth=low and =medium

The default codec configuration for chan_iax2 is bandwidth=low.  I noticed
slin16 being negotiated as the codec in some test calls, but that no longer
happens after this change.

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2009-03-24 21:43:35 +00:00
Tilghman Lesher
b0021508ec Merged revisions 183914 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r183914 | tilghman | 2009-03-24 10:26:42 -0500 (Tue, 24 Mar 2009) | 10 lines
  
  Merged revisions 183913 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r183913 | tilghman | 2009-03-24 10:25:42 -0500 (Tue, 24 Mar 2009) | 3 lines
    
    Additionally note that the operator option needs an 'o' extension.
    (Related to issue #14731)
  ........
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2009-03-24 15:27:34 +00:00
Tilghman Lesher
14879eb45e Merged revisions 183865 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r183865 | tilghman | 2009-03-23 18:28:20 -0500 (Mon, 23 Mar 2009) | 2 lines
  
  Allow browsers to cache images and other static content.
  (This is a regression over 1.4)
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2009-03-23 23:29:51 +00:00
Mark Michelson
799189d232 Merged revisions 183766 via svnmerge from
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  r183766 | mmichelson | 2009-03-23 13:58:03 -0500 (Mon, 23 Mar 2009) | 13 lines
  
  Merged revisions 183700 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r183700 | mmichelson | 2009-03-23 12:59:28 -0500 (Mon, 23 Mar 2009) | 7 lines
    
    Fix a memory leak in res_monitor.c
    
    The only way that this leak would occur is if Monitor were started
    using the Manager interface and no File: header were given. Discovered
    while reviewing the ast_channel_ao2 review request.
  ........
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2009-03-23 18:59:02 +00:00
Leif Madsen
f9d542aab8 Merged revisions 183701 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r183701 | lmadsen | 2009-03-23 14:06:40 -0400 (Mon, 23 Mar 2009) | 7 lines
  
  Fixes a documentation error introduced during the CLI cleanup at AstriDevCon 2008.
  
  (closes issue #14655)
  Reported by: ulogic
  Patches:
        chan_dahdi.patch uploaded by ulogic (license 728)
  Tested by: lmadsen
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2009-03-23 18:11:12 +00:00
Russell Bryant
b95912e533 Merged revisions 183560 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r183560 | russell | 2009-03-20 12:00:58 -0500 (Fri, 20 Mar 2009) | 10 lines

Merged revisions 183559 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r183559 | russell | 2009-03-20 11:53:25 -0500 (Fri, 20 Mar 2009) | 2 lines

Fix a crash in IAX2 registration handling found during load testing with dvossel.

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2009-03-20 17:04:48 +00:00
Mark Michelson
6a00eba9cb Blocked revisions 183553-183555 via svnmerge
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  r183553 | mmichelson | 2009-03-20 11:19:53 -0500 (Fri, 20 Mar 2009) | 5 lines
  
  Add some missing symbols to main/asterisk.exports
  
  Hey! chan_sip.so loads now!
........
  r183554 | mmichelson | 2009-03-20 11:24:20 -0500 (Fri, 20 Mar 2009) | 3 lines
  
  Remove symbols I just added to main/asterisk.exports and instead rename the functions.
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  r183555 | mmichelson | 2009-03-20 11:25:17 -0500 (Fri, 20 Mar 2009) | 3 lines
  
  Fix chan_sip so it builds.
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2009-03-20 16:32:07 +00:00
Eliel C. Sardanons
e0fc5443b8 Blocked revisions 183511 via svnmerge
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  r183511 | eliel | 2009-03-20 08:12:49 -0400 (Fri, 20 Mar 2009) | 2 lines
  
  Remove duplicate <description> inside the xml documentation.
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2009-03-20 12:23:00 +00:00
David Vossel
90b97eddc4 Merged revisions 183436 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r183436 | dvossel | 2009-03-19 15:30:39 -0500 (Thu, 19 Mar 2009) | 13 lines
  
  Merged revisions 183386 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r183386 | dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009) | 6 lines
    
    Cleaning up a few things in detect disconnect patch
    
    Initialized ast_call_feature in detect_disconnect to avoid accessing uninitialized memory.  Cleaned up /param tags in features.h.  No longer send dynamic features in ast_feature_detect. 
    
    issue #11583
  ........
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2009-03-19 20:32:20 +00:00
Tilghman Lesher
84ab8b9963 Merged revisions 183321 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r183321 | tilghman | 2009-03-19 14:17:31 -0500 (Thu, 19 Mar 2009) | 15 lines
  
  Merged revisions 183319 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r183319 | tilghman | 2009-03-19 14:15:33 -0500 (Thu, 19 Mar 2009) | 8 lines
    
    Delay signalling progress until a PRI channel really signals progress.
    (closes issue #13034)
     Reported by: klaus3000
     Patches: 
           20090316__bug13034.diff.txt uploaded by tilghman (license 14)
           patch_trunk_183progress_klaus3000.txt uploaded by klaus3000 (license 65)
     Tested by: klaus3000
  ........
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2009-03-19 19:18:36 +00:00
Mark Michelson
bce77e2674 Merged revisions 183244 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r183244 | mmichelson | 2009-03-19 13:10:34 -0500 (Thu, 19 Mar 2009) | 16 lines
  
  Fix a memory leak associated with queues.
  
  For every attempt that app_queue made to place an outbound call to a queue member,
  we would allocate a queue_end_bridge structure. When the bridge for the call had
  completed, we would free the structure. Unfortunately not all call attempts actually
  end up bridged to a member, so we need to be more selective of when to allocate
  the structure. With this change, the allocation occurs in an area where we can
  guarantee that the call will be bridged.
  
  (closes issue #14680)
  Reported by: caspy
  Patches:
        14680.patch uploaded by mmichelson (license 60)
  Tested by: caspy
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2009-03-19 18:11:16 +00:00
Russell Bryant
70eaf71dfe Merged revisions 183242 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r183242 | russell | 2009-03-19 13:00:15 -0500 (Thu, 19 Mar 2009) | 10 lines

Merged revisions 183241 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r183241 | russell | 2009-03-19 12:52:52 -0500 (Thu, 19 Mar 2009) | 2 lines

Remove the use of RTLD_NOLOAD, as it is not behaving like expected.

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2009-03-19 18:08:02 +00:00
David Vossel
9628d62ffa Merged revisions 183172 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r183172 | dvossel | 2009-03-19 11:28:33 -0500 (Thu, 19 Mar 2009) | 20 lines
  
  Merged revisions 183126 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) | 17 lines
    
    Allow disconnect feature before a call is bridged
    
    feature.conf has a disconnect option.  By default this option is set to '*', but it could be anything.  If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else.  This is because features are unavailable until bridging takes place.  The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different.  This patch allows features to be detected from outside of the bridge, but not operated on.  In this case, the disconnect feature can be detected before briding and handled outside of features.c.
    
    (closes issue #11583)
    Reported by: sobomax
    Patches:
    	patch-apps__app_dial.c uploaded by sobomax (license 359)
    	11583.latest-patch uploaded by murf (license 17)
    	detect_disconnect.diff uploaded by dvossel (license 671)
    Tested by: sobomax, dvossel
    Review: http://reviewboard.digium.com/r/195/
  ........
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2009-03-19 17:10:09 +00:00
Mark Michelson
2099b522d5 Merged revisions 183117 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r183117 | mmichelson | 2009-03-19 11:07:54 -0500 (Thu, 19 Mar 2009) | 20 lines
  
  Merged revisions 183115 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r183115 | mmichelson | 2009-03-19 11:04:02 -0500 (Thu, 19 Mar 2009) | 14 lines
    
    Fix an issue where cancelled outgoing SIP calls would erroneously report the device as "in use."
    
    A user was having an issue where if an outgoing SIP call was canceled, the SIP device
    would remain in use if we had not received any response to the initial INVITE we sent out.
    The SIP device would remain in use until the autocongestion timer was exhausted.
    
    I tracked down the cause of this to be the section of code I am removing here. I asked several
    people what the purpose of this code was meant to be, but no one could give me any sort of
    answer as to why this was here. The person who was having this issue has been using this patch
    for several months and it has stopped the problems they have had.
    
    AST-196
  ........
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2009-03-19 16:08:47 +00:00
Joshua Colp
369b1b702a Merged revisions 183108 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r183108 | file | 2009-03-19 12:37:23 -0300 (Thu, 19 Mar 2009) | 11 lines
  
  Improve our triggering of a T38 switchover internally when triggered by a received reinvite.
  
  Previously we reached across the channel bridge to get the other party's SIP dialog
  structure in order to trigger an outgoing reinvite. This is extremely dangerous to do
  and only works if bridged to another SIP channel. This patch changes this to use the
  T38 control frame method of requesting a switchover. This change also causes the SIP
  channel driver to propogate back whether the switchover worked or not instead of blindly
  accepting the incoming T38 reinvite.
  
  Review: http://reviewboard.digium.com/r/200/
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2009-03-19 15:40:46 +00:00
Joshua Colp
07b411528b Merged revisions 183057 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r183057 | file | 2009-03-18 19:22:56 -0300 (Wed, 18 Mar 2009) | 6 lines
  
  Fix an issue where a T38 control frame would get dropped.
  
  If two channels were bridged together using a generic bridge the T38
  control frame would get passed up instead of being indicated on the
  other channel.
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2009-03-18 22:26:18 +00:00
Jeff Peeler
03bf1455dd Merged revisions 183028 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r183028 | jpeeler | 2009-03-18 16:18:27 -0500 (Wed, 18 Mar 2009) | 4 lines
  
  Add some code removed by mistake from commit 182722 that works around a file
  descriptor leak in versions of PWLib prior to 1.12.0.
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2009-03-18 21:19:00 +00:00
Russell Bryant
e047ec4d72 Merged revisions 182847 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009) | 52 lines

Merged revisions 182810 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines

Fix cases where the internal poll() was not being used when it needed to be.

We have seen a number of problems caused by poll() not working properly on 
Mac OSX.  If you search around, you'll find a number of references to using 
select() instead of poll() to work around these issues.  In Asterisk, we've 
had poll.c which implements poll() using select() internally.  However, we 
were still getting reports of problems.

vadim investigated a bit and realized that at least on his system, even 
though we were compiling in poll.o, the system poll() was still being used.  
So, the primary purpose of this patch is to ensure that we're using the 
internal poll() when we want it to be used.

The changes are:

1) Remove logic for when internal poll should be used from the Makefile.  
   Instead, put it in the configure script.  The logic in the configure 
   script is the same as it was in the Makefile.  Ideally, we would have 
   a functionality test for the problem, but that's not actually possible, 
   since we would have to be able to run an application on the _target_ 
   system to test poll() behavior.

2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT
   is not defined.

3) Change uses of poll() throughout the source tree to ast_poll().  I feel 
   that it is good practice to give the API call a new name when we are 
   changing its behavior and not using the system version directly in all cases.
   So, normally, ast_poll() is just redefined to poll().  On systems where 
   AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll().

4) Change poll() in main/poll.c to be ast_internal_poll().

It's worth noting that any code that still uses poll() directly will work fine 
(if they worked fine before).  So, for example, out of tree modules that are 
using poll() will not stop working or anything.  However, for modules to work 
properly on Mac OSX, ast_poll() needs to be used.

(closes issue #13404)
Reported by: agalbraith
Tested by: russell, vadim

http://reviewboard.digium.com/r/198/

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2009-03-18 14:24:27 +00:00
Jeff Peeler
3e6a72ae73 Merged revisions 182722 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r182722 | jpeeler | 2009-03-17 15:47:31 -0500 (Tue, 17 Mar 2009) | 15 lines
  
  Allow H.323 Plus library to be used in addition to the OpenH323 library
  
  Chan_h323 can now be compiled against both the previously supported versions of
  OpenH323 as well as the current H.323 Plus (version 1.20.2). The configure
  script has been modified to look in the default install location of h323 to
  hopefully help avoid using the environment variables OPENH323DIR and PWLIBDIR.
  Also, the CLI command "h323 show version" has been added which indicates which
  version of h323 is in use.
  
  (closes issue #11261)
  Reported by: vhatz
  Patches:
        asterisk-1.6.0.6-h323plus.patch uploaded by jthurman (license 614)
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2009-03-17 20:51:06 +00:00
Russell Bryant
24fc141a78 Merged revisions 182553 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r182553 | russell | 2009-03-17 10:22:12 -0500 (Tue, 17 Mar 2009) | 5 lines

Tweak the handling of the frame list inside of ast_answer().

This does not change any behavior, but moves the frames from the local frame
list back to the channel read queue using an O(n) algorithm instead of O(n^2).

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2009-03-17 15:27:27 +00:00
Kevin P. Fleming
4c88838302 Merged revisions 182530 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r182530 | kpfleming | 2009-03-17 09:59:33 -0500 (Tue, 17 Mar 2009) | 2 lines
  
  correct logic flaw in ast_answer() changes in r182525
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2009-03-17 15:00:29 +00:00
Kevin P. Fleming
c0219aa890 Merged revisions 182525 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r182525 | kpfleming | 2009-03-17 09:38:11 -0500 (Tue, 17 Mar 2009) | 11 lines
  
  Improve behavior of ast_answer() to not lose incoming frames
  
  ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations.
  
  When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames.
  
  This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller.
  
  http://reviewboard.digium.com/r/196/
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2009-03-17 14:39:16 +00:00
Sean Bright
23e0e35b13 Blocked revisions 182521 via svnmerge
........
  r182521 | seanbright | 2009-03-17 10:24:53 -0400 (Tue, 17 Mar 2009) | 3 lines
  
  Don't include a space before the optional extra text that may follow a help
  string.
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2009-03-17 14:36:35 +00:00
Tilghman Lesher
24e0d6d170 Merged revisions 182450 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r182450 | tilghman | 2009-03-17 00:51:54 -0500 (Tue, 17 Mar 2009) | 14 lines
  
  Merged revisions 182449 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r182449 | tilghman | 2009-03-17 00:50:52 -0500 (Tue, 17 Mar 2009) | 7 lines
    
    Fix race in astdb
    The underlying db1 implementation does not fully isolate the pages retrieved
    from astdb, so the lock protecting accesses needs to be extended until the
    copy from the shared memory structure is done.
    (closes issue #14682)
     Reported by: makoto
  ........
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2009-03-17 05:53:17 +00:00
Richard Mudgett
94542bd0b9 Blocked revisions 182408 via svnmerge
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  r182408 | rmudgett | 2009-03-16 20:54:53 -0500 (Mon, 16 Mar 2009) | 8 lines
  
  OPENR2 uses an incorrect string value if the extension delimiter is not present.
  
  *  Fixed OPENR2 using an incorrect string value if the extension
  delimiter is not present in the Dial() function.  This was fixed for
  SS7 and PRI in trunk -r172400.
  *  Made OPENR2 stripmsd behavior the same as the SS7, PRI, and others.
  *  Removed trailing whitespace that appeared with OPENR2.
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2009-03-17 02:09:23 +00:00
Russell Bryant
d9b9a2eeed Blocked revisions 182362 via svnmerge
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r182362 | russell | 2009-03-16 15:53:21 -0500 (Mon, 16 Mar 2009) | 2 lines

Update UPGRADE.txt and CHANGES for 1.6.3

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2009-03-16 20:53:47 +00:00
Russell Bryant
c8a75d957c Blocked revisions 182355 via svnmerge
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r182355 | russell | 2009-03-16 15:35:58 -0500 (Mon, 16 Mar 2009) | 29 lines

Add MFC/R2 support for chan_dahdi.

This commit introduces official support for R2 signaling in chan_dahdi.  The
modifications to chan_dahdi, and the supporting library, LibOpenR2, were both
written by Moises Silva.

Many users are using this code, or a variant of it, in Asterisk 1.2, 1.4 and 1.6
in Brazil, México and Argentina. An unknown number of users (but at least 1) 
are using it in each of the following countries: Colombia, Nepal, Thailand, 
Venezuela, Perú, and probably others.

To use this code, LibOpenR2 must be installed from http://www.libopenr2.org/.
Information about configuration can be found in configs/chan_dahdi.conf.sample.

The code committed is the most up to date version, which was being maintained
in svn/asterisk/team/moy/mfcr2/.

I would also like to include a Thank You to the many others that tested this
code beyond those listed in this commit message.  These are the names that I
could find in the mantis issue.

(closes issue #12509)
Reported by: moy
Patches:
      chan_zap-mfr2.patch uploaded by moy (license 222)
Tested by: moy, korihor, viniciusfontes, Skarmeth, loloski, asbestoshead, titogarrido, heliocoelhojr, konsultex, ncorrare, ecarruda, rtorresduque, PTorres, ychen

Review: http://reviewboard.digium.com/r/40/

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2009-03-16 20:37:53 +00:00
David Vossel
a8cf0b048b Merged revisions 182282 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r182282 | dvossel | 2009-03-16 12:49:58 -0500 (Mon, 16 Mar 2009) | 13 lines
  
  Merged revisions 182281 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r182281 | dvossel | 2009-03-16 12:47:42 -0500 (Mon, 16 Mar 2009) | 7 lines
    
    Randomize IAX2 encryption padding
    
    The 16-32 byte random padding at the beginning of an encrypted IAX2 frame turns out to not be all that random at all.  This patch calls ast_random to fill the padding buffer with random data.  The padding is randomized at the beginning of every encrypted call and for every encrypted retransmit frame.
    
    Review: http://reviewboard.digium.com/r/193/
  ........
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2009-03-16 17:52:28 +00:00
Tilghman Lesher
6a284f3de6 Merged revisions 182278 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r182278 | tilghman | 2009-03-16 12:33:38 -0500 (Mon, 16 Mar 2009) | 7 lines
  
  Fix an off-by-one error in the FILE() function, and extend FILE()'s length parameter to work like variable substitution.
  Previously, FILE() returned one less character than specified, due to the
  terminating NULL.  Both the offset and length parameters now behave
  identically to the way variable substitution offsets and lengths also work.
  (closes issue #14670)
   Reported by: BMC
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2009-03-16 17:36:51 +00:00
Tilghman Lesher
7faf648dab Merged revisions 182211 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r182211 | tilghman | 2009-03-16 10:50:55 -0500 (Mon, 16 Mar 2009) | 14 lines
  
  Merged revisions 182208 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r182208 | tilghman | 2009-03-16 10:39:15 -0500 (Mon, 16 Mar 2009) | 7 lines
    
    Fixup glare detection, to fix a memory leak of a local pvt structure.
    (closes issue #14656)
     Reported by: caspy
     Patches: 
           20090313__bug14656__2.diff.txt uploaded by tilghman (license 14)
     Tested by: caspy
  ........
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2009-03-16 16:09:11 +00:00
Joshua Colp
461e421402 Merged revisions 182171 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r182171 | file | 2009-03-16 10:58:24 -0300 (Mon, 16 Mar 2009) | 7 lines
  
  Fix a memory leak in the ast_answer / __ast_answer API call.
  
  For a channel that is not yet answered this API call will wait
  until a voice frame is received on the channel before returning.
  It does this by waiting for frames on the channel and reading them
  in. The frames read in were not freed when they should have been.
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2009-03-16 13:59:26 +00:00
Mark Michelson
00cfcfaa62 Merged revisions 182121 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r182121 | mmichelson | 2009-03-13 16:26:20 -0500 (Fri, 13 Mar 2009) | 6 lines
  
  Change faulty comparison used when announcing average hold minutes and seconds
  
  (closes issue #14227)
  Reported by: caspy
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2009-03-13 21:26:47 +00:00
Mark Michelson
118e535c0b Blocked revisions 182071 via svnmerge
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  r182071 | mmichelson | 2009-03-13 12:49:01 -0500 (Fri, 13 Mar 2009) | 3 lines
  
  Remove ast_ prefix from functions which are not public.
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2009-03-13 17:49:28 +00:00
Mark Michelson
bfd1393138 Merged revisions 182029 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r182029 | mmichelson | 2009-03-13 12:26:43 -0500 (Fri, 13 Mar 2009) | 41 lines
  
  Merged revisions 181990 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r181990 | mmichelson | 2009-03-13 12:12:32 -0500 (Fri, 13 Mar 2009) | 35 lines
    
    Check the DYNAMIC_FEATURES of both the chan and peer when interpreting DTMF.
    
    Dynamic features defined in the applicationmap section of features.conf allow
    one to specify whether the caller, callee, or both have the ability to use the
    feature. The documentation in the features.conf.sample file could be interpreted
    to mean that one only needs to set the DYNAMIC_FEATURES channel variable on the
    calling channel in order to allow for the callee to be able to use the features
    which he should have permission to use. However, the DYNAMIC_FEATURES variable
    would only be read from the channel of the participant that pressed the DTMF
    sequence to activate the feature. The result of this was that the callee was
    unable to use dynamic features unless the dialplan writer had taken measures
    to be sure that the DYNAMIC_FEATURES variable was set on the callee's channel.
    
    This commit changes the behavior of ast_feature_interpret to concatenate the
    values of DYNAMIC_FEATURES from both parties involved in the bridge. The features
    themselves determine who has permission to use them, so there is no reason to believe
    that one side of the bridge could gain the ability to perform an action that they
    should not have the ability to perform.
    
    Kevin Fleming pointed out on the asterisk-users list that the typical way that this
    was worked around in the past was by setting _DYNAMIC_FEATURES on the calling channel
    so that the value would be inherited by the called channel. While this works, the
    documentation alone is not enough to figure out why this is necessary for the callee
    to be able to use dynamic features. In this particular case, changing the code to match
    the documentation is safe, easy, and will generally make things easier for people for
    future installations.
    
    This bug was originally reported on the asterisk-users list by David Ruggles.
    
    (closes issue #14657)
    Reported by: mmichelson
    Patches:
          14657.patch uploaded by mmichelson (license 60)
  ........
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2009-03-13 17:33:29 +00:00
Joshua Colp
ee6dcca4f2 Merged revisions 182022 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r182022 | file | 2009-03-13 14:25:09 -0300 (Fri, 13 Mar 2009) | 7 lines
  
  Fix an issue with requesting a T38 reinvite before the call is answered.
  
  The code responsible for sending the T38 reinvite did not check if an INVITE was
  already being handled. This caused things to get confused and the call to fail.
  The code now defers sending the T38 reinvite until the current INVITE is done being
  handled.

  (issue AST-191)
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2009-03-13 17:28:14 +00:00
Kevin P. Fleming
606d637467 Blocked revisions 181985 via svnmerge
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  r181985 | kpfleming | 2009-03-13 11:55:38 -0500 (Fri, 13 Mar 2009) | 1 line
  
  improve a bit of suboptimal code
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2009-03-13 16:57:22 +00:00
Mark Michelson
82d3dc386d Merged revisions 181846 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r181846 | mmichelson | 2009-03-12 16:43:51 -0500 (Thu, 12 Mar 2009) | 3 lines
  
  Run the macro on the queue member's channel when he answers, not the caller's channel.
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2009-03-12 21:44:57 +00:00
Mark Michelson
504ae23462 Merged revisions 181769 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r181769 | mmichelson | 2009-03-12 13:30:58 -0500 (Thu, 12 Mar 2009) | 28 lines
  
  Merged revisions 181768 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r181768 | mmichelson | 2009-03-12 13:29:48 -0500 (Thu, 12 Mar 2009) | 22 lines
    
    Properly send a 487 on an INVITE we have not responded to if we receive a BYE.
    
    If we receive an INVITE from an endpoint and then later receive a BYE from that
    same endpoint before we have sent a final response for the INVITE, then we need
    to respond to the INVITE with a 487. 
    
    There was logic in the code prior to this commit which seemed to exist solely to 
    handle this situation, but there was one condition in an if statement which 
    was incorrect. The only way we would send a 487 was if the sip_pvt had no owner
    channel. This made no sense since we created the owner channel when we received
    the INVITE, meaning that the majority of the time we would never send the 487.
    The 487 being sent should not rely on whether we have created a channel. Its
    delivery should be dependent on the current state of the initial INVITE transaction.
    With this commit, that logic is now correctly in place.
    
    (closes issue #14149)
    Reported by: legranjl
    Patches:
          14149.patch uploaded by mmichelson (license 60)
    Tested by: legranjl
  ........
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2009-03-12 18:32:38 +00:00
Tilghman Lesher
a3ff3e1c1f Merged revisions 181731 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r181731 | tilghman | 2009-03-12 12:32:13 -0500 (Thu, 12 Mar 2009) | 9 lines
  
  Adjust translation table column widths based upon the translation times.
  Previously, only 5 columns were displayed, and if a translation time exceeded
  99,999 useconds, it would be displayed as 0, instead of its actual time.
  (closes issue #14532)
   Reported by: pj
   Patches: 
         20090311__bug14532.diff.txt uploaded by tilghman (license 14)
   Tested by: pj
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2009-03-12 17:58:25 +00:00
Joshua Colp
6774078cbb Merged revisions 181665 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r181665 | file | 2009-03-12 13:56:58 -0300 (Thu, 12 Mar 2009) | 9 lines
  
  Merged revisions 181664 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r181664 | file | 2009-03-12 13:56:20 -0300 (Thu, 12 Mar 2009) | 2 lines
    
    Fix incorrect usage of strncasecmp... I really meant to use strcasecmp.
  ........
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2009-03-12 16:57:43 +00:00
Joshua Colp
f62853dc12 Merged revisions 181661 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r181661 | file | 2009-03-12 13:53:52 -0300 (Thu, 12 Mar 2009) | 19 lines
  
  Merged revisions 181659-181660 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r181659 | file | 2009-03-12 13:50:37 -0300 (Thu, 12 Mar 2009) | 8 lines
    
    Fix another scenario where depending on configuration the stream would not get read.
    
    For custom commands we don't know whether the audio is coming from a stream or not
    so we are going to have to read the data despite no channels.
    
    (closes issue #14416)
    Reported by: caspy
  ........
    r181660 | file | 2009-03-12 13:52:45 -0300 (Thu, 12 Mar 2009) | 2 lines
    
    Fix logic flaw in previous commit.
  ........
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2009-03-12 16:54:35 +00:00
Joshua Colp
ef6ff7199a Merged revisions 181656 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r181656 | file | 2009-03-12 13:32:20 -0300 (Thu, 12 Mar 2009) | 17 lines
  
  Merged revisions 181655 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r181655 | file | 2009-03-12 13:29:19 -0300 (Thu, 12 Mar 2009) | 10 lines
    
    Fix issue with streaming MOH failing if nobody is listening.
    
    When a music class is setup to actually provide music on hold
    from a stream we need to constantly read audio from it since it
    will constantly be providing audio. This is now done despite there
    being no channels listening to it.
    
    (closes issue #14416)
    Reported by: caspy
  ........
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2009-03-12 16:33:47 +00:00
Joshua Colp
41f3b9eb4a Merged revisions 181612 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r181612 | file | 2009-03-12 10:24:12 -0300 (Thu, 12 Mar 2009) | 5 lines
  
  Fix crash when sleep and retries argument was not given to RetryDial application.
  
  (closes issue #14647)
  Reported by: sherpya
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2009-03-12 13:27:36 +00:00
Richard Mudgett
13ba3f9128 Merged revisions 181542 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r181542 | rmudgett | 2009-03-11 20:00:29 -0500 (Wed, 11 Mar 2009) | 1 line
  
  Use the correct branch integrated property when generating the version string
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2009-03-12 01:04:18 +00:00
Michiel van Baak
72353ade99 Merged revisions 181499 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

Provide correct hint to debug SIP trouble in the default config
(closes issue #14646)
Reported by: strk


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181509 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-11 23:19:08 +00:00
Russell Bryant
ed7ecdba6e Blocked revisions 181465 via svnmerge
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r181465 | russell | 2009-03-11 17:25:57 -0500 (Wed, 11 Mar 2009) | 2 lines

Make handling of the BRIDGE_PLAY_SOUND variable thread-safe.

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2009-03-11 22:26:30 +00:00
Jason Parker
9afc536f86 Merged revisions 181444 via svnmerge from
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  r181444 | qwell | 2009-03-11 17:20:13 -0500 (Wed, 11 Mar 2009) | 11 lines
  
  Merged revisions 181436 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r181436 | qwell | 2009-03-11 17:18:42 -0500 (Wed, 11 Mar 2009) | 4 lines
    
    Allow prefix to set localstatedir (when used and different from the default).
    
    This is similar to the /etc change that was made for the non-FreeBSD case.
  ........
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2009-03-11 22:22:07 +00:00
Russell Bryant
cefe73c073 Merged revisions 181428 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r181428 | russell | 2009-03-11 17:14:55 -0500 (Wed, 11 Mar 2009) | 2 lines

Make handling of the BRIDGEPVTCALLID variable thread-safe.

........


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2009-03-11 22:15:40 +00:00
Russell Bryant
91eb9b614b Merged revisions 181424 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r181424 | russell | 2009-03-11 16:49:29 -0500 (Wed, 11 Mar 2009) | 17 lines

Merged revisions 181423 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r181423 | russell | 2009-03-11 16:42:58 -0500 (Wed, 11 Mar 2009) | 9 lines

Make code that updates BRIDGEPEER variable thread-safe.

It is not safe to read the name field of an ast_channel without the channel
locked.  This patch fixes some places in channel.c where this was being done,
and lead to crashes related to masquerades.

(closes issue #14623)
Reported by: guillecabeza

........

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2009-03-11 21:55:24 +00:00
David Vossel
000375fcce Merged revisions 181371 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r181371 | dvossel | 2009-03-11 12:34:57 -0500 (Wed, 11 Mar 2009) | 17 lines
  
  Merged revisions 181340 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r181340 | dvossel | 2009-03-11 12:25:31 -0500 (Wed, 11 Mar 2009) | 11 lines
    
    encrypted IAX2 during packet loss causes decryption to fail on retransmitted frames
    
    If an iax channel is encrypted, and a retransmit frame is sent, that packet's iseqno is updated while it is encrypted.  This causes the entire frame to be corrupted.  When the corrupted frame is sent, the other side decrypts it and sends a VNAK back because the decrypted frame doesn't make any sense.  When we get the VNAK, we look through the sent queue and send the same corrupted frame causing a loop.  To fix this, encrypted frames requiring retransmission are decrypted, updated, then re-encrypted.  Since key-rotation may change the key held by the pvt struct, the keys used for encryption/decryption are held within the iax_frame to guarantee they remain correct.
    
    (closes issue #14607)
    Reported by: stevenla
    Tested by: dvossel
    
    Review: http://reviewboard.digium.com/r/192/
  ........
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2009-03-11 17:37:25 +00:00
Joshua Colp
ed9ddfd8a5 Merged revisions 181345 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r181345 | file | 2009-03-11 14:26:40 -0300 (Wed, 11 Mar 2009) | 21 lines
  
  Merged revisions 181328 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r181328 | file | 2009-03-11 14:22:52 -0300 (Wed, 11 Mar 2009) | 14 lines
    
    Fix issue where an attended transfer could not be completed under a rare scenario.
    
    When completing an attended transfer chan_sip does a check to make sure the extension
    in the URI portion of the Refer-To header is a local valid extension. We don't actually
    need to check this since we know for sure the other channel is already up and talking to
    the extension. Some devices do not put the extension in the Refer-To header either, which
    can cause the extension check to fail. We now no longer do this check if it is an attended
    transfer.
    
    (closes issue #14628)
    Reported by: sverre
    Patches:
          14628.diff uploaded by file (license 11)
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181352 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-11 17:28:12 +00:00
Joshua Colp
be3fc819ab Merged revisions 181296 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r181296 | file | 2009-03-11 13:40:48 -0300 (Wed, 11 Mar 2009) | 16 lines
  
  Merged revisions 181295 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r181295 | file | 2009-03-11 13:36:50 -0300 (Wed, 11 Mar 2009) | 9 lines
    
    Fix a problem with inband DTMF detection on outgoing SIP calls when dtmfmode=auto.
    
    When dtmfmode was set to auto the inband DTMF detector was not setup
    on outgoing SIP calls. This caused inband DTMF detection to fail.
    The inband DTMF detector is now setup for both dtmfmode inband and auto.
    
    (closes issue #13713)
    Reported by: makoto
  ........
................


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2009-03-11 16:42:21 +00:00
Russell Bryant
1644f0da92 Blocked revisions 181292 via svnmerge
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r181292 | russell | 2009-03-11 11:19:38 -0500 (Wed, 11 Mar 2009) | 2 lines

Replace contents of this doc with a pointer to its new home

........


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2009-03-11 16:20:15 +00:00
Jeff Peeler
95c0766715 add missing header file
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-11 15:54:46 +00:00
Mark Michelson
9f263bf684 Blocked revisions 181244 via svnmerge
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  r181244 | mmichelson | 2009-03-11 09:28:40 -0500 (Wed, 11 Mar 2009) | 11 lines
  
  Fix segfault when dialing a typo'd queue
  
  If trying to dial a non-existent queue, there would
  be a segfault when attempting to access q->weight, even
  though q was NULL. This problem was introduced during
  the queue-reset merge and thus only affects trunk.
  
  (closes issue #14643)
  Reported by: alecdavis
........


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2009-03-11 14:29:42 +00:00
Joshua Colp
cbaceb5473 Blocked revisions 181210 via svnmerge
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  r181210 | file | 2009-03-11 10:44:42 -0300 (Wed, 11 Mar 2009) | 3 lines
  
  Don't play the "you are about to be placed into the conference" and "the leader has left the conference" sounds if the quiet
  option is enabled. (reported by Vadim Lebedev on the asterisk-dev list)
........


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2009-03-11 13:45:45 +00:00
Jeff Peeler
b9ae2618b6 Fix merge oops from 181137
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181178 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-11 04:27:25 +00:00
Jeff Peeler
368b57494b Merged revisions 181135 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r181135 | jpeeler | 2009-03-10 23:06:44 -0500 (Tue, 10 Mar 2009) | 20 lines
  
  Fix malloc debug macros to work properly with h323.
  
  The main problem here was that cstdlib was undefining free thereby causing the
  proper debug macros to not be used. ast_h323.cxx has been changed to call
  ast_free instead to avoid the issue. 
  
  A few other issues were addressed:
  - There were a few instances of functions improperly passing ast_free instead
  of ast_free_ptr.
  - Some clean up was done to avoid the debug macros intentionally being redefined.
  (copied below from Kevin's commit, appreciate the help)
  - disable astmm.h from doing anything when STANDALONE is defined, which is used
  by the tools in the utils/ directory that use parts of Asterisk header files in
  hackish ways; also ensure that utils/extconf.c and utils/conf2ael.c are
  compiled with STANDALONE defined.
  
  (closes issue #13593)
  Reported by: pj
........


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2009-03-11 04:17:26 +00:00
Russell Bryant
72ba7ac7da Blocked revisions 181099 via svnmerge
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r181099 | russell | 2009-03-10 21:25:24 -0500 (Tue, 10 Mar 2009) | 2 lines

tabs to spaces

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2009-03-11 02:26:07 +00:00
Mark Michelson
d90ef47592 Merged revisions 181032-181033 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r181032 | mmichelson | 2009-03-10 19:46:47 -0500 (Tue, 10 Mar 2009) | 19 lines
  
  Merged revisions 181029,181031 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r181029 | mmichelson | 2009-03-10 19:30:26 -0500 (Tue, 10 Mar 2009) | 9 lines
    
    Fix incorrect tag checking on transfers when pedantic=yes is enabled.
    
    (closes issue #14611)
    Reported by: klaus3000
    Patches:
          patch_chan_sip_attended_transfer_1.4.23.txt uploaded by klaus3000 (license 65)
    Tested by: klaus3000
  ........
    r181031 | mmichelson | 2009-03-10 19:32:40 -0500 (Tue, 10 Mar 2009) | 3 lines
    
    Remove unused variables.
  ........
................
  r181033 | mmichelson | 2009-03-10 19:49:00 -0500 (Tue, 10 Mar 2009) | 3 lines
  
  Add missing comment that quotes RFC 3891
................


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2009-03-11 00:52:22 +00:00
Jason Parker
a993cd8adc Merged revisions 180944 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r180944 | qwell | 2009-03-10 17:03:41 -0500 (Tue, 10 Mar 2009) | 9 lines
  
  Merged revisions 180941 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r180941 | qwell | 2009-03-10 17:02:18 -0500 (Tue, 10 Mar 2009) | 1 line
    
    Make things happier when using autoconf 2.62+
  ........
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2009-03-10 22:05:35 +00:00
Russell Bryant
c9fa4f5d4e Blocked revisions 180942 via svnmerge
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r180942 | russell | 2009-03-10 17:03:16 -0500 (Tue, 10 Mar 2009) | 2 lines

Add some notes on getting in contact with the dev community

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2009-03-10 22:03:27 +00:00
Russell Bryant
d1ce051582 Blocked revisions 180938 via svnmerge
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r180938 | russell | 2009-03-10 16:55:49 -0500 (Tue, 10 Mar 2009) | 2 lines

Remove difficulty and language specifiers

........


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2009-03-10 21:56:03 +00:00
Russell Bryant
f36c3147d0 Blocked revisions 180935 via svnmerge
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r180935 | russell | 2009-03-10 16:45:54 -0500 (Tue, 10 Mar 2009) | 2 lines

Expand upon documentation of manager event project

........


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2009-03-10 21:46:19 +00:00
Michiel van Baak
f2e79dcf81 Blocked revisions 180898 via svnmerge
........
  r180898 | mvanbaak | 2009-03-10 22:15:29 +0100 (Tue, 10 Mar 2009) | 10 lines
  
  list the move of the astvarrundir from /var/run to /var/run/asterisk
  (actually its $(localstatedir)/run/asterisk
  
  Makes setups with asterisk as non-root easier to manage because you can
  setup permissions on this dir instead of touching a file and setting 
  permissions on that.
  Files that come to mind are asterisk.pid and asterisk.ctl socket.
  
  Prodded by and ok @russell
........


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2009-03-10 21:16:43 +00:00
Russell Bryant
d45b336720 Blocked revisions 180862 via svnmerge
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r180862 | russell | 2009-03-10 14:36:21 -0500 (Tue, 10 Mar 2009) | 1 line

add more projects
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2009-03-10 19:36:42 +00:00
Russell Bryant
90e3195b9c Blocked revisions 180859 via svnmerge
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r180859 | russell | 2009-03-10 14:23:41 -0500 (Tue, 10 Mar 2009) | 1 line

add more project ideas
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@180860 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-10 19:24:59 +00:00
Joshua Colp
f604257f31 Merged revisions 180800 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r180800 | file | 2009-03-10 11:40:38 -0300 (Tue, 10 Mar 2009) | 5 lines
  
  Reset the thread local string buffer when handling the UserEvent action.
  
  (closes issue #14593)
  Reported by: JimDickenson
........


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2009-03-10 14:41:33 +00:00
Joshua Colp
c6837adabc If a port is specified when dialing a peer then use it.
(closes issue #14626)
Reported by: acunningham


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@180799 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-10 13:32:58 +00:00
Russell Bryant
d375364194 Blocked revisions 180750 via svnmerge
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r180750 | russell | 2009-03-09 17:00:42 -0500 (Mon, 09 Mar 2009) | 4 lines

Add current mentors list, and first pass on a project list broken out of "PineMango"

I will work on adding projects that have been sent to be via email tomorrow.

........


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2009-03-09 22:01:16 +00:00
Jeff Peeler
d18b4d4f92 Blocked revisions 180719 via svnmerge
........
  r180719 | jpeeler | 2009-03-09 15:58:17 -0500 (Mon, 09 Mar 2009) | 16 lines
  
  Add Doxygen documentation for API changes from 1.6.0 to 1.6.1
  
  Copied from my review board description:
  This is a continuation of the API changes documentation started for describing
  changes between releases. Most of the API changes were pretty simple needing
  only to be brought to attention via the new "Asterisk API Changes" list.
  However, if you see anything that needs further explanation feel free to
  supplement what is there. The current method of documenting is to add (in the
  header file): \version <ver number> <description of changes> and then to add
  the function to the change list in doxyref.h on the AstAPIChanges page. I also
  made sure all the functions that were newly added were tagged with \since
  1.6.1. I think this is a good habit to start both for the historical aspect as
  well as for the future ability to easily add a "New Asterisk API" page.
  
  Review: http://reviewboard.digium.com/r/190/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@180720 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-09 20:59:34 +00:00
Joshua Colp
b332b085ee Ensure that the new outgoing dialog to a peer is able to set the socket details, even if the default is present.
(closes issue #14480)
Reported by: jon


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@180718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-09 20:14:05 +00:00
Russell Bryant
2d4de30630 Blocked revisions 180684 via svnmerge
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r180684 | russell | 2009-03-09 09:14:34 -0500 (Mon, 09 Mar 2009) | 2 lines

Add skeleton for GSoC ideas list

........


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2009-03-09 14:14:56 +00:00
Russell Bryant
0dd22e88ce Blocked revisions 180641 via svnmerge
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r180641 | russell | 2009-03-07 09:36:00 -0600 (Sat, 07 Mar 2009) | 7 lines

Make some minor updates to the doxygen configuration

- add bridges directory to be processed
- add some res/ subdirs
- alphabetize subdirs
- use consistent indentation

........


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2009-03-07 15:36:27 +00:00
Mark Michelson
a2985ff7c3 Merged revisions 180579 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r180579 | mmichelson | 2009-03-06 12:25:44 -0600 (Fri, 06 Mar 2009) | 9 lines
  
  Merged revisions 180567 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r180567 | mmichelson | 2009-03-06 12:23:09 -0600 (Fri, 06 Mar 2009) | 2 lines
    
    Make compilation succeed in dev-mode when IMAP storage is enabled.
  ........
................


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2009-03-06 18:26:28 +00:00
David Vossel
f94e5b2d7c Merged revisions 180534 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r180534 | dvossel | 2009-03-06 11:26:38 -0600 (Fri, 06 Mar 2009) | 15 lines
  
  Merged revisions 180532 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r180532 | dvossel | 2009-03-06 11:19:55 -0600 (Fri, 06 Mar 2009) | 9 lines
    
    Fix handling of backreferences for ENUM lookups
    
    enum.c did not handle regex backtraces correctly.  The '\1' in the regex is a backreference that requires a pattern match to be inserted.  The way the code used to work is that it would find the backreference and insert the entire input string minus the '+'.  This is incorrect.  The regexec() function takes in a variable called pmatch which is an array of structs containing the start and end indexes for each backreference substring.  The original code actually passed the pmatch array pointer into regexec but never did anything with it.  Now when a backtrace is found, the backtrace number is looked up in the pmatch array and the correct substring is inserted.
    
    (closes issue #14576)
    Reported by: chris-mac
    Review: http://reviewboard.digium.com/r/187/
  ........
................


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2009-03-06 17:28:52 +00:00
Mark Michelson
5d055cb8b2 Merged revisions 180465 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r180465 | mmichelson | 2009-03-05 17:26:58 -0600 (Thu, 05 Mar 2009) | 22 lines
  
  Merged revisions 180464 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r180464 | mmichelson | 2009-03-05 17:26:11 -0600 (Thu, 05 Mar 2009) | 16 lines
    
    [IMAP] Fix message retrieval issues when identical mailbox names were defined in separate contexts.
    
    There was a fix put in a while back so that an X-Asterisk-VM-Context message header was
    added to stored IMAP voicemails. This would allow for us to differentiate if the same
    mailbox name was used in multiple contexts. The problem still left was that not all places
    where messages were retrieved actually attempted to use this header for information when
    retrieving messages. This commit fixes that so that MWI and message retrieval from VoiceMailMain
    work as expected.
    
    (closes issue #13853)
    Reported by: vicks1
    Patches:
          13853_v2.patch uploaded by mmichelson (license 60)
    Tested by: lmadsen
  ........
................


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2009-03-05 23:28:01 +00:00
Mark Michelson
7810bed6b8 Merged revisions 180383 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r180383 | mmichelson | 2009-03-05 13:14:14 -0600 (Thu, 05 Mar 2009) | 31 lines
  
  Merged revisions 180380 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r180380 | mmichelson | 2009-03-05 12:58:48 -0600 (Thu, 05 Mar 2009) | 25 lines
    
    Fix broken mailbox parsing when searchcontexts option is enabled.
    
    When using the searchcontexts option in voicemail.conf, the code
    made the assumption that all mailbox names defined were unique across
    all contexts. However, the code did nothing to actually enforce this
    assumption, nor did it do anything to alert a user that he may have
    created an ambiguity in his voicemail.conf file by defining the same
    mailbox name in multiple contexts.
    
    With this change, we now will issue a nice long warning if searchcontexts
    is on and we encounter the same mailbox name in multiple contexts and ignore
    any duplicates after the first box. Whether searchcontexts is enabled or not,
    if we come across a duplicate mailbox in the same context, then we will issue
    a warning and ignore the duplicated mailbox. I have also added a small note
    to voicemail.conf.sample in the explanation for searchcontexts explaining
    that you cannot define the same mailbox in multiple contexts if you have
    enabled the option.
    
    (closes issue #14599)
    Reported by: lmadsen
    Patches:
          14599.patch uploaded by mmichelson (license 60) (with slight modification)
    Tested by: lmadsen
  ........
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2009-03-05 19:21:23 +00:00
Michiel van Baak
19f4a4ab25 Blocked revisions 180382 via svnmerge
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  r180382 | mvanbaak | 2009-03-05 20:05:20 +0100 (Thu, 05 Mar 2009) | 2 lines
  
  Make sure we terminate the first s| command so we can actually produce correct files.
........


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2009-03-05 19:15:14 +00:00
Kevin P. Fleming
2a877c8fcb Merged revisions 180373 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r180373 | kpfleming | 2009-03-05 12:29:38 -0600 (Thu, 05 Mar 2009) | 15 lines
  
  Merged revisions 180372 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar 2009) | 9 lines
    
    Fix problems when RTP packet frame size is changed
    
    During some code analysis, I found that calling ast_rtp_codec_setpref() on an ast_rtp session does not work as expected; it does not adjust the smoother that may on the RTP session, in fact it summarily drops it, even if it has data in it, even if the current format's framing size has not changed. This is not good.
    
    This patch changes this behavior, so that if the packetization size for the current format changes, any existing smoother is safely updated to use the new size, and if no smoother was present, one is created. A new API call for smoothers, ast_smoother_reconfigure(), was required to implement these changes.
    
    Review: http://reviewboard.digium.com/r/184/
  ........
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2009-03-05 18:36:31 +00:00
Joshua Colp
6dc1cc36c6 Blocked revisions 180369 via svnmerge
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  r180369 | file | 2009-03-05 14:18:27 -0400 (Thu, 05 Mar 2009) | 13 lines
  
  Merge phase 1 support for the new bridging architecture.
  
  This commit brings in the bridging core, bridging technologies,
  and the ConfBridge application.
  
  For usage information on the ConfBridge application please see
  the output of "core show application ConfBridge" from the CLI.
  
  For API documentation please see the doxygen page describing the
  architecture and the documentation for each API call.
  
  Review: http://reviewboard.digium.com/r/93/
........


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2009-03-05 18:19:06 +00:00
Russell Bryant
4becc9332a Blocked revisions 180261 via svnmerge
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r180261 | russell | 2009-03-04 15:01:05 -0600 (Wed, 04 Mar 2009) | 54 lines

Resolve object matching issues related to the removal of the sip_user object.

Previously, chan_sip had both sip_peer and sip_user objects in memory.  A
patch went in to remove sip_user to simplify the code, since everything
could be done with just sip_peer.  This patch resolves some regressions
found that were introduced by those changes.

This code comes from svn/asterisk/team/group/sip-object-matching/.

Here is a list of the changes that have been made:

1) When doing a match by name with the find_peer() function, make it much
   easier to specify which objects should be matched by having a parameter
   that specifies exactly which object types should be considered.  Also,
   update find_by_name() to handle this parameter.  Finally, update all
   code to use the new option values.

2) When looking up an object for an outbound request by name, consider
   peers only.  (create_addr())

3) Only match peers on an incoming registration request.

4) When doing authentication (except for SUBSCRIBE), look up users
   by name, instead of all objects by name.
   
5) When doing authentication (except for SUBSCRIBE), after looking for
   a user by name, look for a peer by IP address, instead of all objects
   by IP address.

6) When handling the SIP qualify CLI command or manager action, look for
   a peer by name, instead of any object by name.

7) When handling the SIP unregister CLI command, look for a peer by name,
   instead of any object by name.

9) In sip_do_debug_peer(), search for a peer by name, instead of any object
   by name.

9) When handling the SIPPEER() dialplan function, search for a peer by name,
   instead of any object by name.

10) In the following session timer related functions, st_get_se(),
    st_get_refresher(), and st_get_mode(), when looking for an object for a
    given sip_pvt using pvt->peername, look for a peer by name, instead of any
    object by name.

11) Fix build_peer() to properly handle the case where separate type=peer and
    type=user entries were specified in sip.conf.

(closes issue #14505)
Reported by: lmadsen

Review: http://reviewboard.digium.com/r/172/

........


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2009-03-04 21:03:28 +00:00
Joshua Colp
75d0d2389e Merged revisions 180195 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r180195 | file | 2009-03-04 15:24:59 -0400 (Wed, 04 Mar 2009) | 11 lines
  
  Merged revisions 180194 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r180194 | file | 2009-03-04 15:22:50 -0400 (Wed, 04 Mar 2009) | 4 lines
    
    Look for the number in a callerid string starting from the end. This way a value using <> can exist in the name portion.
    
    (issue #AST-194)
  ........
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2009-03-04 19:25:59 +00:00
Mark Michelson
85b1f7dcef Blocked revisions 180155 via svnmerge
........
  r180155 | mmichelson | 2009-03-04 11:03:32 -0600 (Wed, 04 Mar 2009) | 14 lines
  
  Allow for "magic" pickups to work when we wish to ignore the context
  
  When the subscription context for a call pickup subscription differs
  from the context of the call pickup target, there's not an easy way
  to divine what context should be used for the pickup. The way to work
  around this is to use PICKUPMARK as the context for the pickup.
  
  This has been documented in the sip.conf.sample file
  
  (ABE-1708)
  
  closes issue #14567
  submitted by: alecdavis
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2009-03-04 17:15:11 +00:00
Joshua Colp
0c8d1667b9 Merged revisions 180120 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r180120 | file | 2009-03-04 10:39:28 -0400 (Wed, 04 Mar 2009) | 7 lines
  
  Remove duplicate 'k' and 'K' Dial options.
  
  (closes issue #14601)
  Reported by: alecdavis
  Patches:
        app_dial.optionk.diff.txt uploaded by alecdavis (license 585)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@180121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-04 14:40:12 +00:00
Steve Murphy
bbd996fbc0 Blocked revisions 180079 via svnmerge
........
  r180079 | murf | 2009-03-03 16:35:26 -0700 (Tue, 03 Mar 2009) | 1 line
  
  My bad! left check_expr2 in the ALL_UTILS list by mistake. Already done to 1.6.x
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@180081 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-03 23:40:14 +00:00
David Vossel
9cad0b7e22 Merged revisions 180032 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r180032 | dvossel | 2009-03-03 17:21:18 -0600 (Tue, 03 Mar 2009) | 14 lines
  
  app_read does not break from prompt loop with user terminated empty string
  
  In app.c, ast_app_getdata is called to stream the prompts and receive DTMF input.  If ast_app_getdata() receives an empty string caused by the user inputing the end of string character, in this case '#', it should break from the prompt loop and return to app_read, but instead it cycles through all the prompts.  I've added a return value for this special case in ast_readstring() which uses an enum I've delcared in apps.h.  This enum is now used as a return value for ast_app_getdata().
  
  (closes issue #14279)
  Reported by: Marquis
  Patches:
  	fix_app_read.patch uploaded by Marquis (license 32)
  	read-ampersanmd.patch2 uploaded by dvossel (license 671)
  Tested by: Marquis, dvossel
  Review: http://reviewboard.digium.com/r/177/
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2009-03-03 23:35:18 +00:00
Steve Murphy
ce5bbc3eb8 Merged revisions 179973 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r179973 | murf | 2009-03-03 15:12:02 -0700 (Tue, 03 Mar 2009) | 33 lines
  
  Merged revisions 179807 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  I had some work to do to port these changes to trunk; the 
  check_expr stuff hasn't been updated here for quite some
  time, it appears. I added some more tests to the check_expr2
  suite. I had to play around with the makefile a bit, etc.
  
  I added STANDALONE2 #ifdefs to ast_expr2.y so as not to
  conflict structure with aelparse.
  
  ........
    r179807 | murf | 2009-03-03 11:11:34 -0700 (Tue, 03 Mar 2009) | 19 lines
    
    These changes allow AEL to better check ${} constructs within $[...], that are concatenated with text.
    
    I modified and added rules in ast_expr2.fl to better handle
    the concatenations.
    
    I added some default routines to ast_expr2.y so the standalone would
    compile. It also looks like I haven't run this thru bison since 2.1, so
    it's good to get this updated.
    
    The Makefile has comments added now for check_expr2 and check_expr to
    explain what they are for, and how to run them. 
    
    The testexpr2s stuff has been removed, in favor of check_expr2.
    
    expr2.testinput has been updated to include the two expressions
    that inspired these changes (from mcnobody on #asterisk this morning)
    The regression has been run and all looks well.
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@180058 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-03 23:26:54 +00:00
Mark Michelson
d24e3e05bc Merged revisions 180007 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r180007 | mmichelson | 2009-03-03 16:49:07 -0600 (Tue, 03 Mar 2009) | 22 lines
  
  Merged revisions 180006 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r180006 | mmichelson | 2009-03-03 16:48:18 -0600 (Tue, 03 Mar 2009) | 17 lines
    
    Clarify some documentation of queues.conf.sample
    
    It had always been possible to explicitly specify a "blank"
    value for a sound file in queues.conf and have no sound played
    back. The problem with this is that it would result in some ugly
    CLI warnings from file.c.
    
    This commit introduces a check when playing a file in app_queue
    to see if the name of the file is zero-length and return early if
    that is the case. Also, the ability to specify the blank sound
    files in queues.conf is now mentioned more clearly in queues.conf.sample
    
    (closes issue #14227)
    Reported by: caspy
  ........
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2009-03-03 22:49:32 +00:00
David Vossel
6767bd053d Blocked revisions 179972 via svnmerge
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  r179972 | dvossel | 2009-03-03 16:01:24 -0600 (Tue, 03 Mar 2009) | 13 lines
  
  app_meetme not setting filename and fileformat correctly for realtime
  
  When app_meetme finds a realtime conference, it doesn't get the filename and fileformat correctly when 'r' is set.  Now app_meetme first checks to see if fileformat and filename are declared in the db, if they're not it checks the .conf file, if its not declared there either it then uses defaults. 
  
  (closes issue #14545)
  Reported by: dalbaech
  Patches:
  	app_meetme-realtime5.patch uploaded by dvossel (license 671)
  	Realtime_Conference_Record_workaround.txt uploaded by dalbaech (license 705)
  Tested by: dvossel, dalbaech
  Review: http://reviewboard.digium.com/r/180/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@180005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-03 22:25:13 +00:00
Mark Michelson
3cb51f0127 Fix a memory leak when updating a realtime member field.
This was discovered while looking at issue #14353



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179971 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-03 21:43:09 +00:00
Mark Michelson
408082d376 Blocked revisions 179937 via svnmerge
........
  r179937 | mmichelson | 2009-03-03 14:59:16 -0600 (Tue, 03 Mar 2009) | 20 lines
  
  Add documentation for timing modules used in Asterisk
  
  This document specifies the timing modules available in Asterisk beginning
  with Asterisk 1.6.1. The document goes into detail about the differences
  between each and gives a general overview of what timing is used for in
  Asterisk. There is also a section which can be used to help customize
  your setup or to troubleshoot timing issues you may have.
  
  I also added messages to the DAHDI timing test used in res_timing_dahdi.c
  that points to this new documentation if people experience problems.
  
  Big thanks to all who contributed comments on this.
  
  (closes issue #14490)
  Reported by: mmichelson
  Patches:
        timing.txt uploaded by mmichelson (license 60)
  
  Review: http://reviewboard.digium.com/r/164/
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2009-03-03 21:00:16 +00:00
Russell Bryant
220d2b601f Blocked revisions 179903 via svnmerge
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r179903 | bmd | 2009-03-03 14:02:20 -0600 (Tue, 03 Mar 2009) | 1 line

fix a leaked channel lock (and future deadlock) when we try to pick up our own channel
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2009-03-03 20:07:30 +00:00
Joshua Colp
807cb98467 Merged revisions 179841 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r179841 | file | 2009-03-03 14:28:46 -0400 (Tue, 03 Mar 2009) | 16 lines
  
  Merged revisions 179840 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r179840 | file | 2009-03-03 14:27:09 -0400 (Tue, 03 Mar 2009) | 9 lines
    
    Do not assume that the bridge_cdr is still attached to the channel when the 'h' exten is finished executing.
    
    It is possible for a masquerade operation to occur when the 'h' exten is operating. This operation moves
    the CDR records around causing the bridge_cdr to no longer exist on the channel where it is expected to.
    We can not safely modify it afterwards because of this, so don't even try.
    
    (closes issue #14564)
    Reported by: meric
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179842 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-03 18:29:34 +00:00
Mark Michelson
bde7a0435b Blocked revisions 179745 via svnmerge
........
  r179745 | mmichelson | 2009-03-03 11:03:47 -0600 (Tue, 03 Mar 2009) | 8 lines
  
  Convert pbx_spool to use string fields instead of statically-sized buffers.
  
  In tests run after making this conversion, I noticed an approximate 85% 
  reduction in memory usage for call file processing.
  
  Review: http://reviewboard.digium.com/r/168/
........


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2009-03-03 17:04:17 +00:00
Russell Bryant
e4ae90e0cb Merged revisions 179742 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
r179742 | russell | 2009-03-03 10:47:28 -0600 (Tue, 03 Mar 2009) | 14 lines

Merged revisions 179741 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r179741 | russell | 2009-03-03 10:45:46 -0600 (Tue, 03 Mar 2009) | 6 lines

Ensure chan->fdno always gets reset to -1 after handling a channel fd event.

Since setting fdno to -1 had to be moved, a couple of other code paths that
do process an fd event return early and do not pass through the code path
where it was moved to.  So, set it to -1 in a few other places, too.

........

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2009-03-03 16:48:15 +00:00
Joshua Colp
5284efc9ce Merged revisions 179672 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r179672 | file | 2009-03-03 10:40:04 -0400 (Tue, 03 Mar 2009) | 10 lines
  
  Merged revisions 179671 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r179671 | file | 2009-03-03 10:38:09 -0400 (Tue, 03 Mar 2009) | 3 lines
    
    Move where fdno is set to the default value to *after* the read callback of the channel driver is called.
    We have to do this as the underlying channel driver may need the fdno value to determine what to read.
  ........
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2009-03-03 14:40:59 +00:00
Russell Bryant
1e4f2f5a1b Merged revisions 179609 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
r179609 | russell | 2009-03-03 07:54:41 -0600 (Tue, 03 Mar 2009) | 17 lines

Merged revisions 179608 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r179608 | russell | 2009-03-03 07:53:52 -0600 (Tue, 03 Mar 2009) | 9 lines

Make it easier to detect an improper call to ast_read().

When you call ast_waitfor() on a channel, the index into the channel fds array
that holds the file descriptor that poll() determines has input available is
stored in fdno.  This patch clears out this value after a call to ast_read()
and also reports errors if ast_read() is called without an fdno set.

From a discussion on the asterisk-dev list.

........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179610 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-03 13:55:34 +00:00
Jeff Peeler
f355d2180a Merged revisions 179537 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r179537 | jpeeler | 2009-03-02 18:01:51 -0600 (Mon, 02 Mar 2009) | 21 lines
  
  Merged revisions 179536 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r179536 | jpeeler | 2009-03-02 17:54:39 -0600 (Mon, 02 Mar 2009) | 15 lines
    
    Fix bridging regression from commit 176701
    
    This fixes a bad regression where the bridge would exit after an attended
    transfer was made. The problem was due to nexteventts getting set after the
    masquerade which caused the bridge to return AST_BRIDGE_COMPLETE.
    
    (closes issue #14315)
    Reported by: tim_ringenbach
  ........
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2009-03-03 00:03:36 +00:00
Russell Bryant
49d2383e12 Merged revisions 179533 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
r179533 | russell | 2009-03-02 17:36:38 -0600 (Mon, 02 Mar 2009) | 48 lines

Merged revisions 179532 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r179532 | russell | 2009-03-02 17:34:13 -0600 (Mon, 02 Mar 2009) | 40 lines

Move ast_waitfor() down to avoid the results of the API call becoming stale.

This call to ast_waitfor() was being done way too soon in this section of code.
Specifically, there was code in between the call to waitfor and the code that
uses the result that puts the channel in autoservice.  By putting the channel
in autoservice, the previous results of ast_waitfor() become meaningless,
as the autoservice thread will do it's own ast_waitfor() and ast_read()
on the channel.

So, when we came back out of autoservice and eventually hit the block of code
that calls ast_read() on the channel, there may not actually be any input on
the channel available.  Even though the previous call to ast_waitfor() in
app_meetme said there was input, the autoservice thread has since serviced
the channel for some period of time.

This bug manifested itself while dvossel was doing some testing of MeetMe in
Asterisk trunk.  He was using the timerfd timing module.  When the code hit
ast_read() erroneously, it determined that it must have been called because of
input on the timer fd, as chan->fdno was set to AST_TIMING_FD, since that was 
the cause of the last legitimate call to ast_read() done by autoservice.  

In this test, an IAX2 channel was calling into the MeetMe conference.  It was
_much_ more likely to be seen with an IAX2 channel because of the way audio
is handled.  Every audio frame that comes in results in a call to
ast_queue_frame(), which then uses ast_timer_enable_continuous() to notify
the channel thread that a frame is waiting to be handled.  So, the chances
of ast_waitfor() indicating that a channel needs servicing due to a timer
event on an IAX2 event is very high.

Finally, it is interesting to note that if a different timing interface was
being used, this bug would probably not be noticed.  When ast_read() is called
and erroneously thinks that there is a timer event to handle, it calls the
ast_timer_ack() function.  The pthread and dahdi timing modules handle the
ack() function being called when there is no event by simply ignoring it.
In the case of the timerfd module, it results in a read() on the timer fd
that will block forever, as there is no data to read.  This caused Asterisk
to lock up very quickly.

Thanks to dvossel and mmichelson for the fun debugging session.  :-)

........

................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179534 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-02 23:38:23 +00:00
Mark Michelson
f805a657a3 Merged revisions 151464 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r151464 | mmichelson | 2008-10-21 18:54:41 -0500 (Tue, 21 Oct 2008) | 11 lines
  
  Make the sip_standard_port function more granular by allowing separate
  type and port arguments. This is necessary because when building our From
  and Contact headers, we need to be absolutely sure that we are placing our
  source port there and not the peer's source port.
  
  (closes issue #12761)
  Reported by: asbestoshead
  Patches:
        patch-chan-sip-contact-port.txt uploaded by asbestoshead (license 455)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179473 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-02 23:15:51 +00:00
Tilghman Lesher
93bba53c69 Merged revisions 179469 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r179469 | tilghman | 2009-03-02 17:10:18 -0600 (Mon, 02 Mar 2009) | 17 lines
  
  Merged revisions 179468 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r179468 | tilghman | 2009-03-02 17:09:01 -0600 (Mon, 02 Mar 2009) | 10 lines
    
    When ending a recording with silence detection, remember to reduce the duration.
    The end of the recording is correspondingly trimmed, but the duration was not
    trimmed by the number of seconds trimmed, so the saved duration was necessarily
    longer than the actual soundfile duration.
    (closes issue #14406)
     Reported by: sasargen
     Patches: 
           20090226__bug14406.diff.txt uploaded by tilghman (license 14)
     Tested by: sasargen
  ........
................


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2009-03-02 23:11:24 +00:00
Russell Bryant
7091bad5a0 Blocked revisions 179465 via svnmerge
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r179465 | russell | 2009-03-02 17:06:16 -0600 (Mon, 02 Mar 2009) | 4 lines

Fix a reference leak in timerfd_set_rate().

(found during a debugging session with dvossel and mmichelson.)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-02 23:06:39 +00:00
Russell Bryant
9a6e93c561 Merged revisions 179462 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
r179462 | russell | 2009-03-02 17:00:30 -0600 (Mon, 02 Mar 2009) | 16 lines

Merged revisions 179461 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r179461 | russell | 2009-03-02 16:58:18 -0600 (Mon, 02 Mar 2009) | 8 lines

Ensure that only one thread is calling ast_settimeout() on a channel at a time.

For example, with an IAX2 channel, you can have both the channel thread and the
chan_iax2 processing threads calling this function, and doing so twice at the
same time is a bad thing.

(Found in a debugging session with dvossel and mmichelson)

........

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2009-03-02 23:02:49 +00:00
Jason Parker
98d73b0f65 Merged revisions 179396 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r179396 | qwell | 2009-03-02 14:16:51 -0600 (Mon, 02 Mar 2009) | 9 lines
  
  Merged revisions 179395 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r179395 | qwell | 2009-03-02 14:14:57 -0600 (Mon, 02 Mar 2009) | 1 line
    
    Remove several silly warnings in editline.  One about a broken preprocessor directive, and another about strlcpy/strlcat.

    (closes issue #14264)
    Reported by: dimas
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179402 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-02 20:17:55 +00:00
Tilghman Lesher
33d3f5ab4f KeepAlive application no longer exists, so fix gosub implementation to not use it.
(closes issue #14571)
 Reported by: zktech
 Patches: 
       20090302__bug14571.diff.txt uploaded by tilghman (license 14)
 Tested by: tilghman


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179363 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-02 17:58:20 +00:00
Tilghman Lesher
1a1831118b If cdr registration somehow succeeds without a config file, don't crash.
(closes issue #14563)
 Reported by: alerios


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179360 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-02 17:16:57 +00:00
Joshua Colp
bb4419c8f0 Blocked revisions 179323 via svnmerge
........
  r179323 | file | 2009-03-02 10:28:09 -0400 (Mon, 02 Mar 2009) | 5 lines
  
  Do not try to remove a registration scheduled item if the scheduler context has already been destroyed.
  
  (closes issue #14580)
  Reported by: alecdavis
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179324 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-02 14:28:56 +00:00
Joshua Colp
420a08e866 Blocked revisions 179291 via svnmerge
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  r179291 | file | 2009-03-02 10:13:45 -0400 (Mon, 02 Mar 2009) | 7 lines
  
  Fix issue where changing the volume of both directions of audio did not work.
  
  (closes issue #14574)
  Reported by: KNK
  Patches:
        audiohook_volume_fix.diff uploaded by KNK (license 545)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179292 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-02 14:14:11 +00:00
Mark Michelson
8d49f52408 Blocked revisions 179254 via svnmerge
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  r179254 | mmichelson | 2009-03-01 17:25:23 -0600 (Sun, 01 Mar 2009) | 5 lines
  
  Swap reversed timevals.
  
  This was pointed out by ScribbleJ in #asterisk-dev. Thanks very much, ScribbleJ!
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179255 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-01 23:27:03 +00:00
Mark Michelson
eb9e2f2c4e Add error checking when updating the "paused" field of a realtime queue member.
This code already existed in trunk and 1.6.1, but was not in 1.6.0 prior to
this commit.


(closes issue #14338)
Reported by: fiddur
Patches:
      14338.patch uploaded by mmichelson (license 60)
Tested by: fiddur



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179222 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-01 22:07:09 +00:00
Mark Michelson
8c5803b286 Merged revisions 179219 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r179219 | mmichelson | 2009-03-01 15:45:08 -0600 (Sun, 01 Mar 2009) | 18 lines
  
  Properly free memory and remove scheduler entries when a transmission failure occurs.
  
  Previously, only the "data" field of the sip_pkt created during __sip_reliable_xmit 
  was freed when XMIT_FAILURE was returned by __sip_xmit. When retrans_pkt was called,
  this inevitably resulted in the reading and writing of freed memory.
  
  XMIT_FAILURE is a condition meaning that we don't want to attempt resending the packet
  at all. The proper action to take is to remove the scheduler entry we just created,
  free the packet's data as well as the packet itself, and unlink it from the list of
  packets on the sip_pvt structure.
  
  (closes issue #14455)
  Reported by: Nick_Lewis
  Patches:
        14455.patch uploaded by mmichelson (license 60)
  Tested by: Nick_Lewis
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2009-03-01 21:52:39 +00:00
Russell Bryant
10464df9cd Blocked revisions 179164 via svnmerge
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r179164 | russell | 2009-02-27 15:47:18 -0600 (Fri, 27 Feb 2009) | 2 lines

Mark res_ais as experimental, as the binary event format is subject to change.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179165 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-27 21:47:37 +00:00
Tilghman Lesher
43ab90e500 Merged revisions 179161 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r179161 | tilghman | 2009-02-27 15:32:13 -0600 (Fri, 27 Feb 2009) | 3 lines
  
  If config file is blank, don't load module.
  (Closes issue #14563)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179162 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-27 21:33:31 +00:00
Russell Bryant
2769f06e91 Blocked revisions 179154 via svnmerge
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r179154 | russell | 2009-02-27 15:23:12 -0600 (Fri, 27 Feb 2009) | 2 lines

Add a note about the ordering of entries in sip.conf in 1.6.1.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179159 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-27 21:24:06 +00:00
Michiel van Baak
0689aae55b Blocked revisions 179122 via svnmerge
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  r179122 | mvanbaak | 2009-02-27 21:34:00 +0100 (Fri, 27 Feb 2009) | 16 lines
  
  Add reload support to chan_skinny.
  
  Special thanks goes to DEA who had to redo this patch twice
  because we first put unload/load support in and later redid the way
  we configure devices and lines.
  
  (closes issue #10297)
  Reported by: DEA
  Patches:
        skinny-reload-trunkv2.diff uploaded by wedhorn (license 30)
        skinny-reload-trunk-v4.txt uploaded by DEA (license 3)
  	  With mods by me based on feedback from wedhorn and Russell and seanbright
  Tested by: DEA, mvanbaak, pj
  
  Review: http://reviewboard.digium.com/r/130/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179123 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-27 20:36:47 +00:00
Jason Parker
edceb47475 Merged revisions 179057 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r179057 | qwell | 2009-02-27 13:04:57 -0600 (Fri, 27 Feb 2009) | 8 lines
  
  Update documentation for DIALEDTIME and ANSWEREDTIME variables.
  
  (closes issue #14566)
  Reported by: klaus3000
  Patches:
        ANSWEREDTIME-1.4-patch.txt uploaded by klaus3000 (license 65)
        ANSWEREDTIME-trunk-patch.txt uploaded by klaus3000 (license 65)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179058 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-27 19:05:36 +00:00
Russell Bryant
7c0de63410 Blocked revisions 179021 via svnmerge
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r179021 | russell | 2009-02-27 09:51:56 -0600 (Fri, 27 Feb 2009) | 7 lines

Fix downloading SIREN7 and SIREN14 sound packages.

In passing, also fix downloading SLIN16 extra sound packages.

(closes issue #14565)
Reported by: jtodd

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179022 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-27 15:52:24 +00:00
Steve Murphy
f42c14308d Merged revisions 178986 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r178986 | murf | 2009-02-26 20:45:58 -0700 (Thu, 26 Feb 2009) | 26 lines
  
  Merged revisions 178956 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  In this case, it's just a matter of reducing the default timeouts from 2000
  to 1000 msec, as the max def feature digit timeout is no longer halved.
  
  ........
    r178956 | murf | 2009-02-26 14:27:32 -0700 (Thu, 26 Feb 2009) | 18 lines
    
    This change moves the default feature digit timeout to 1000 ms from the previous default of 500.
    
    As per bug 14515, a dev discussion arrived at a "mediated concensus" 
    of a default feature digit timeout of 1.0 sec. Some voted for 1300;
    ctooley thought 1500 for distracted phone users in phone booths; 
    kpfleming put his foot down at 1.0 sec. 
    
    Users who found the previous default max delay of 250 msec perfect,
    are welcome to override the new default. Notice that I said that
    250 msec was the default; wait a minute, you might say, the config
    file said it was 500 msec!; well, because of the bug fix for 14515,
    we found that 500 msec was actually enforcing a max of 250. The bug
    fix would restore 500 msec, but we felt even that was a bit tight
    for most users... 2000 msec was pushed earlier by mmichelson, so
    that reduces to 1000 msec after the bug fix. Enjoy!
  ........
................


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2009-02-27 03:52:31 +00:00
David Vossel
73af4eb39c Merged revisions 178871 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r178871 | dvossel | 2009-02-26 11:46:12 -0600 (Thu, 26 Feb 2009) | 6 lines
  
  IAX2 prune realtime, minor tweak to last fix
  
  A return statement was missing which caused unexpected cli output.
  
  issue #14479
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2009-02-26 17:50:00 +00:00
Steve Murphy
3d3b6b46c6 Blocked revisions 178870 via svnmerge
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  r178870 | murf | 2009-02-26 10:45:22 -0700 (Thu, 26 Feb 2009) | 1 line
  
  These small fixes prevent compiler warnings with ubuntu 8.10's gcc-4.3.2, which tend to break my dev-mode build. Not a problem in 1.6.x.
........


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2009-02-26 17:47:18 +00:00
Steve Murphy
547b76aa6a Merged revisions 178828 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r178828 | murf | 2009-02-26 10:22:11 -0700 (Thu, 26 Feb 2009) | 34 lines
  
  Merged revisions 178804 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r178804 | murf | 2009-02-26 10:09:03 -0700 (Thu, 26 Feb 2009) | 28 lines
    
    This patch prevents the feature detection timeout from being cut in half.
    
    Because the ast_channel_bridge() call will return 0 and pass
    a frame pointer for both DTMF_BEGIN and DTMF_END, the feature_timer
    field in hte config struct is getting decremented twice, which 
    effectively cuts the digittimeout in half. I added conditions
    to the if statement to only let DTMF_END frames to flow thru,
    which solved the problem. Also, when the frame pointer is null,
    let control flow thru-- this usually happens on timeouts. I added
    a comment to the code to explain what's going on and why.
    
    Many thanks to sodom for reporting this problem. Personnally, it always seemed
    like something was wrong with the featuredigittimeout, but I never
    could quite decide what... and was too busy to investigate.
    This bug forced the issue, and now we know.
    
    Sodom had other issues in 14515, but I couldn't reproduce them. If
    he still has problems, and wants to get them solved, he is welcome
    to reopen 14515.
    
    
    (closes issue #14515)
    Reported by: sodom
    Patches:
          14515.patch uploaded by murf (license 17)
    Tested by: murf, sodom
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178866 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-26 17:29:53 +00:00
Joshua Colp
75a4a954f2 Blocked revisions 178801 via svnmerge
........
  r178801 | file | 2009-02-26 12:42:36 -0400 (Thu, 26 Feb 2009) | 5 lines
  
  Fix an issue where the timer for file playback would not be stopped if DAHDI was not installed.
  
  (closes issue #14541)
  Reported by: grant
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-26 16:44:08 +00:00
David Vossel
e515340cc2 Merged revisions 178767 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r178767 | dvossel | 2009-02-26 09:50:22 -0600 (Thu, 26 Feb 2009) | 8 lines
  
  IAX2 prune realtime fix
  
  Iax2 prune realtime had issues.  If "iax2 prune realtime all" was called, it would appear like the command was successful, but in reality nothing happened.  This is because the reload that was supposed to take place checks the config files, sees no changes, and does nothing.  If there had been a change in the the config file, the realtime users would have been marked for deletion and everything would have been fine.  Now prune_users() and prune_peers() are called instead of reload_config() to prune all users/peers that are realtime.  These functions remove all users/peers with the rtfriend and delme flags set. iax2_prune_realtime() also lacked the code to properly delete a single friend.  For example. if iax2 prune realtime <friend> was called, only the peer instance would be removed. The user would still remain.  
  
  (closes issue #14479)
  Reported by: mousepad99
  Review: http://reviewboard.digium.com/r/176/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178768 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-26 16:01:05 +00:00
Joshua Colp
e79c813fae Blocked revisions 178764 via svnmerge
........
  r178764 | file | 2009-02-26 11:40:10 -0400 (Thu, 26 Feb 2009) | 5 lines
  
  Ensure there is a valid tone part before trying to play tones.
  
  (closes issue #14558)
  Reported by: alecdavis
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178765 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-26 15:42:33 +00:00
Russell Bryant
e1eba74ef4 Merged revisions 178509 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
r178509 | russell | 2009-02-25 06:45:30 -0600 (Wed, 25 Feb 2009) | 10 lines

Merged revisions 178508 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r178508 | russell | 2009-02-25 06:43:36 -0600 (Wed, 25 Feb 2009) | 2 lines

Update the copyright year for the main page of the doxygen documentation.

........

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2009-02-25 12:46:06 +00:00
Tilghman Lesher
229442f533 Merged revisions 178446 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r178446 | tilghman | 2009-02-24 17:27:23 -0600 (Tue, 24 Feb 2009) | 12 lines
  
  Merged revisions 178445 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r178445 | tilghman | 2009-02-24 17:25:24 -0600 (Tue, 24 Feb 2009) | 5 lines
    
    Add section about the #exec command in configuration files.
    (closes issue #14540)
     Reported by: jtodd
     Patch by: jtodd, with additional notes by tilghman (license 14) 
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-24 23:28:03 +00:00
Tilghman Lesher
41451ea689 Merged revisions 178381 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r178381 | tilghman | 2009-02-24 14:52:44 -0600 (Tue, 24 Feb 2009) | 2 lines
  
  Apparently, a void cast doesn't override warn_unused_result.
........


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2009-02-24 20:53:27 +00:00
Russell Bryant
239ad71be7 Merged revisions 178374 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r178374 | russell | 2009-02-24 14:39:57 -0600 (Tue, 24 Feb 2009) | 14 lines

Merged revisions 178373 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r178373 | russell | 2009-02-24 14:36:19 -0600 (Tue, 24 Feb 2009) | 6 lines

Only set dtmfcount on BEGIN, and ensure it gets reset to 0 properly.

(issue #14460)
Reported by: moliveras
Tested by: russell

........

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2009-02-24 20:43:16 +00:00
Tilghman Lesher
425201ca94 Merged revisions 178375 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r178375 | tilghman | 2009-02-24 14:40:02 -0600 (Tue, 24 Feb 2009) | 2 lines
  
  The 3 possible errors with pipe(2) are all impossible in this situation.
........


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2009-02-24 20:40:49 +00:00
Tilghman Lesher
31fd4d3573 Merged revisions 178342 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r178342 | tilghman | 2009-02-24 14:06:48 -0600 (Tue, 24 Feb 2009) | 2 lines
  
  Use a SIGPIPE to kill the process, instead of depending upon the astcanary process being inherited by init.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178343 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-24 20:07:36 +00:00
Terry Wilson
de2bbd8f13 Change include order to make compile on Centos 5 with DAHDI
If BIT_TYPES_DEFINED gets defined before linux/types.h is included, the
__s32 type doesn't get defined


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178306 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-24 18:05:03 +00:00
Tilghman Lesher
04df6ac55a Merged revisions 178303 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r178303 | tilghman | 2009-02-24 11:51:36 -0600 (Tue, 24 Feb 2009) | 7 lines
  
  Cause astcanary to exit if Asterisk exits abnormally and doesn't kill astcanary.
  Also, add some documentation supporting the use of astcanary.
  (closes issue #14538)
   Reported by: KNK
   Patches: 
         asterisk-1.6.x-astcanary.diff uploaded by KNK (license 545)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-24 17:53:16 +00:00
David Vossel
3454c77364 Blocked revisions 178300 via svnmerge
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  r178300 | dvossel | 2009-02-24 11:42:37 -0600 (Tue, 24 Feb 2009) | 14 lines
  
  Allows manager command to see if IAX link is trunked and encrypted. Displays what kind of encryption is enabled as well. 
  
  Manager command "iaxpeers" now shows if a link is trunked and encrypted.  Instead of encryption saying simply "yes" or "no", it now displays what type of encryption is enabled and if keyrotation is on or not.  
  
  (closes issue #14427)
  Reported by: snuffy
  Patches:
  	iax_show_trunks.diff uploaded by snuffy (license 35)
  	2009022200_iax2_show_trunkencryption.diff.txt uploaded by mvanbaak (license 7)
  Tested by: mvanbaak, dvossel, snuffy
........


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2009-02-24 17:43:33 +00:00
Joshua Colp
be2caea0fa Merged revisions 178213 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r178213 | file | 2009-02-24 11:18:38 -0400 (Tue, 24 Feb 2009) | 16 lines
  
  Merged revisions 178205 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9 lines
    
    Skip check for extension when subscribing for MWI.
    
    Since the remote side is not actually subscribing to a specific extension when
    subscribing for MWI just skip the check to see if the extension exists. They can't use it
    to specify the mailbox either since we require configuration of that in sip.conf
    
    (closes issue #14531)
    Reported by: festr
  ........
................


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2009-02-24 15:20:47 +00:00
Russell Bryant
07b9f97f48 Merged revisions 178142 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r178142 | russell | 2009-02-23 17:11:37 -0600 (Mon, 23 Feb 2009) | 22 lines

Merged revisions 178141 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r178141 | russell | 2009-02-23 17:09:01 -0600 (Mon, 23 Feb 2009) | 14 lines

Fix infinite DTMF when a BEGIN is received without an END.

This commit is related to rev 175124 of 1.4 where a previous attempt was made
to fix this problem.  The problem with the previous patch was that the inserted
code needed to go _before_ setting the lastrxts to the current timestamp.
Because those were the same, the dtmfcount variable was never decremented, and
so the END was never sent.

In passing, I removed the dtmfsamples variable which was completed unused.  I
also removed a redundant setting of the lastrxts variable.

(closes issue #14460)
Reported by: moliveras

........

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2009-02-23 23:17:30 +00:00
David Vossel
0a96d6557e Blocked revisions 178030 via svnmerge
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  r178030 | dvossel | 2009-02-23 11:59:55 -0600 (Mon, 23 Feb 2009) | 7 lines
  
  Changes the way keyrotation is enabled by default
  
  Key rotation was enabled by default by setting the global encryption method to IAX_ENCRYPT_KEYROTATE.  the problem with this is that if encryption is not enabled, and the encryption method is set to anything except 0, the peer appears to have encryption enabled when issuing a "iax2 show peers".  Rather than have the key rotation bit always set by default, it is now only set when an encryption method is enabled. 
  
  (closes issue #14523)
  Reported by: mvanbaak
........


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2009-02-23 20:49:36 +00:00
Michiel van Baak
8e0166f058 Blocked revisions 178061 via svnmerge
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  r178061 | mvanbaak | 2009-02-23 19:23:38 +0100 (Mon, 23 Feb 2009) | 3 lines
  
  update the new manager commands in chan_skinny to match
  chan_sip's headers. requested by oej.
........


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2009-02-23 18:25:23 +00:00
Michiel van Baak
7843bf769e Blocked revisions 178027 via svnmerge
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  r178027 | mvanbaak | 2009-02-23 18:48:32 +0100 (Mon, 23 Feb 2009) | 2 lines
  
  list the addition of the SKINNY manager actions in the CHANGES file.
........


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2009-02-23 17:49:32 +00:00
Russell Bryant
604dc1731d Blocked revisions 178022 via svnmerge
........
r178022 | russell | 2009-02-23 11:29:16 -0600 (Mon, 23 Feb 2009) | 6 lines

Fix a regression in scheduler entry ordering, and add a regression test for it.

(closes issue #14522)
Reported by: pj
Tested by: russell

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178023 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-23 17:29:47 +00:00
Michiel van Baak
0585d43744 Blocked revisions 177988 via svnmerge
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  r177988 | mvanbaak | 2009-02-23 00:04:37 +0100 (Mon, 23 Feb 2009) | 13 lines
  
  Add a couple of manager commands to chan_skinny
  
  Added:
  SKINNYdevices
  SKINNYshowdevice
  SKINNYlines
  SKINNYshowline
  
  (closes issue #14521)
  Reported by: mvanbaak
  
  Review: http://reviewboard.digium.com/r/170/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177989 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-22 23:07:13 +00:00
Michiel van Baak
cbe9195358 Blocked revisions 177913 via svnmerge
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  r177913 | mvanbaak | 2009-02-21 15:37:04 +0100 (Sat, 21 Feb 2009) | 7 lines
  
  add extra check for sysinfo/sysctl
  
  (closes issue #14513)
  Reported by: snuffy
  Patches:
        bug14513_fixsysinfo.diff uploaded by snuffy (license 35)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177914 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-21 14:40:10 +00:00
Russell Bryant
515e20e913 Blocked revisions 177855 via svnmerge
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r177855 | russell | 2009-02-21 07:17:47 -0600 (Sat, 21 Feb 2009) | 5 lines

Fix build issues on Solaris and OpenBSD.

(closes issue #14512)
Reported by: snuffy

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177859 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-21 13:18:45 +00:00
Michiel van Baak
eb1721b567 Blocked revisions 177852 via svnmerge
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  r177852 | mvanbaak | 2009-02-21 14:13:35 +0100 (Sat, 21 Feb 2009) | 18 lines
  
  set ASTVARRUNDIR=$(localstatedir)/run/asterisk as default path
  
  When running asterisk as non-root and without this patch the pidfile wants
  to go into /var/run/asterisk.pid. This directory is not writable for
  the non-root user and changing permissions is not an option.
  Putting it in /var/run/asterisk/asterisk.pid makes it possible
  to set permissions on the /var/run/asterisk dir so everything
  works as it should be.
  
  Patched committed is based on pabelanger's patch.
  
  (closes issue #13153)
  Reported by: pabelanger
  Patches:
        2009012900_bug13153-nonrootscripts.diff.txt uploaded by mvanbaak (license 7)
  
  Review: http://reviewboard.digium.com/r/139/
........


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2009-02-21 13:14:50 +00:00
Michiel van Baak
01ad529ce9 Blocked revisions 177849 via svnmerge
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  r177849 | mvanbaak | 2009-02-21 13:22:32 +0100 (Sat, 21 Feb 2009) | 2 lines
  
  make chan_sip.c compile on OpenBSD again.
........


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2009-02-21 12:40:36 +00:00
Tilghman Lesher
d6c817f2f1 Merged revisions 177787 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r177787 | tilghman | 2009-02-20 17:02:35 -0600 (Fri, 20 Feb 2009) | 16 lines
  
  Merged revisions 177786 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r177786 | tilghman | 2009-02-20 16:59:52 -0600 (Fri, 20 Feb 2009) | 9 lines
    
    Don't print the CR-NL combination when we aren't outputting to the manager.
    
    An embedded CR-NL in a CLI command screws up several AMI parsers that don't
    expect to see that combination in the middle of output.
    
    (Closes issue #14305)
    Reported by: martins
    Patch by: tilghman
  ........
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2009-02-20 23:04:33 +00:00
Dwayne M. Hubbard
e8073da399 Merged revisions 177699 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r177699 | dhubbard | 2009-02-20 14:29:00 -0600 (Fri, 20 Feb 2009) | 9 lines
  
  Make app_fax compatible with spandsp-0.0.6pre4
  
  Prior to spandsp-0.0.6pre4 the t30_stats_t structure used a pages_transferred
  integer to indicate the number of pages transferred (so far) during the fax
  session.  The spandsp-0.0.6pre4 release removed the pages_transferred integer
  and replaced it with two different integers - pages_tx and pages_rx.  This
  revision uses the new integers for spandsp-0.0.6pre4 while maintaining backwards
  compatibility for previous spandsp releases.
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2009-02-20 21:56:39 +00:00
Dwayne M. Hubbard
14bb8408db Merged revisions 148868 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r148868 | tilghman | 2008-10-14 10:15:45 -0500 (Tue, 14 Oct 2008) | 6 lines
  
  API differences in spandsp 0.0.6pre1 and higher
  (closes issue #13688)
   Reported by: irroot
   Patches: 
         app_fax-span6.patch uploaded by irroot (license 52) with minor modifications by me
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2009-02-20 21:41:54 +00:00
Tilghman Lesher
aac9d07849 Merged revisions 177732 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r177732 | tilghman | 2009-02-20 15:25:37 -0600 (Fri, 20 Feb 2009) | 10 lines
  
  Merged revisions 177701 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r177701 | tilghman | 2009-02-20 15:15:01 -0600 (Fri, 20 Feb 2009) | 3 lines
    
    This exception does not appear to still be true for Solaris 10, and OpenSolaris definitely needs it to be removed.
    Fixed for snuff-home on -dev channel.
  ........
................


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2009-02-20 21:31:50 +00:00
David Vossel
e938398385 Fixes issue with undefined audio codecs in chan_iax2
During iax2 call negotiation, supported codecs are passed in an Information Element containing a 2 byte field where each bit correlates to a specific codec.  In 1.6 only audio codec bits 0-12 and 15 are defined, leaving bits 13-14 undefined.  By default all bits are enabled unless specified otherwise.  Since its a 2 byte field and 13-14 are not defined, these bits are never turned off.  In trunk, bits 13-14 are defined, which means 1.6 is advertising support for codecs it does not have when talking to trunk.  I fixed this by adding #define for undefined audio codec bits.  These bits are then removed from iax2's full bandwidth capabilities.   

(closes issue #14283)
Reported by: jcovert


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177698 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-20 20:27:45 +00:00
Tilghman Lesher
36f41cee56 Merged revisions 177664 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r177664 | tilghman | 2009-02-20 11:29:51 -0600 (Fri, 20 Feb 2009) | 8 lines
  
  Allow semicolons to be escaped, when passing arguments to the System command.
  (closes issue #14231)
   Reported by: jcovert
   Patches: 
         20090113__bug14231__2.diff.txt uploaded by Corydon76 (license 14)
         corrected_20090113__bug14231__2.diff.txt uploaded by jcovert (license 551)
   Tested by: jcovert
........


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2009-02-20 17:42:16 +00:00
Tilghman Lesher
b2cc0480ac Merged revisions 177661 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r177661 | tilghman | 2009-02-20 11:22:19 -0600 (Fri, 20 Feb 2009) | 2 lines
  
  Oops, merge broke trunk
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2009-02-20 17:27:20 +00:00
Jeff Peeler
2eb96d9bd8 Blocked revisions 177624 via svnmerge
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  r177624 | jpeeler | 2009-02-19 18:35:53 -0600 (Thu, 19 Feb 2009) | 7 lines
  
  Set sip_request ast_str data to NULL so ast_str_copy allocates space properly
  in copy_request
  
  (issue #14478)
  Reported by: erik_dedecker
........


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2009-02-20 00:36:27 +00:00
Steve Murphy
7ea292e195 Merged revisions 177595 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
r177595 | murf | 2009-02-19 16:56:50 -0700 (Thu, 19 Feb 2009) | 32 lines

Merged revisions 177540 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

Trunk was already pretty 8-bit clean; but I'm still
removing the --full from the flex command so everything
is uniform.

........
  r177540 | murf | 2009-02-19 15:51:37 -0700 (Thu, 19 Feb 2009) | 21 lines
  
  This patch fixes a problem with 8-bit input to the ast_expr2 scanner.
  
  The real culprit was the --full argument to flex
  in the Makefile! This causes a 7-bit scanner to be
  generated.
  
  I reviewed the rules and found one rule where I needed
  to specifically include 8-bit chars for a token.
  
  I tested against the text supplied by ibercom, and 
  all looks very well.
  
  This has been there a surprisingly long time!
  
  
  (closes issue #14498)
  Reported by: ibercom
  Patches:
        14498.patch uploaded by murf (license 17)
  Tested by: murf
........

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2009-02-20 00:16:46 +00:00
Tilghman Lesher
8b4c344ca7 Merged revisions 177537 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r177537 | tilghman | 2009-02-19 16:33:00 -0600 (Thu, 19 Feb 2009) | 14 lines
  
  Merged revisions 177536 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r177536 | tilghman | 2009-02-19 16:26:01 -0600 (Thu, 19 Feb 2009) | 7 lines
    
    Fix up potential crashes, by reducing the sharing between interactive and non-interactive threads.
    (closes issue #14253)
     Reported by: Skavin
     Patches: 
           20090219__bug14253.diff.txt uploaded by Corydon76 (license 14)
     Tested by: Skavin
  ........
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2009-02-19 22:34:37 +00:00
Jeff Peeler
5b4fb69323 Remove invalid ast_free calls for static character arrays
(issue #14478)
Reported by: erik_dedecker



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177445 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-19 17:40:16 +00:00
Jeff Peeler
fff4e4f7cd Merged revisions 177387 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r177387 | jpeeler | 2009-02-19 10:45:02 -0600 (Thu, 19 Feb 2009) | 3 lines
  
  Fix another merge error from 176708
........


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2009-02-19 16:45:30 +00:00
Joshua Colp
d82a4d919c Merged revisions 177384 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r177384 | file | 2009-02-19 12:38:41 -0400 (Thu, 19 Feb 2009) | 10 lines
  
  Merged revisions 177383 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r177383 | file | 2009-02-19 12:37:25 -0400 (Thu, 19 Feb 2009) | 3 lines
    
    If we are able to create a speech structure unset the ERROR variable in case it was previously set.
    (issue #LUMENVOX-13)
  ........
................


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2009-02-19 16:39:28 +00:00
Jeff Peeler
676fb48b9b Merged revisions 177356 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r177356 | jpeeler | 2009-02-19 09:56:31 -0600 (Thu, 19 Feb 2009) | 4 lines
  
  Fix mismerge from revision 176708 pointed out by Kaloyan Kovachev on the
  asterisk-dev mailing list. Thanks!
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177357 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-19 15:57:16 +00:00
Steve Murphy
62bbd7a589 Merged revisions 177286 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r177286 | murf | 2009-02-18 16:50:57 -0700 (Wed, 18 Feb 2009) | 39 lines
  
  Merged revisions 177225 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r177225 | murf | 2009-02-18 15:43:14 -0700 (Wed, 18 Feb 2009) | 34 lines
    
    This patch fixes a regression of sorts that was introduced in 
    rev 24425.
    
    It basically fixes AST-190/ABE-1782.
    
    What was wrong: the user has 6000 extensions in one context; and
    then 6000 contexts, one per extension. The parser could only handle
    about 4893 of the 6000 extens in the single context.
    
    This was due to the regression I mentioned. To get rid of
    shift/reduce conflicts, Luigi set up right-recursive lists
    for globals, context elements, switch lists, and statements.
    Right recursive lists got rid of the warnings, but instead, they
    use up a tremendous amount of stack space when the lists are long.
    
    I saw this a few years back, and resolved not to fix it until
    someone complained. That day has arrived!
    
    After the changes were made, I ran the regression test suite,
    and there were no problems.
    
    I took the test case the user provided, and added 100,000 
    extensions to the single context, that already had 6,000 extens
    in it. (I'll see your 6, and raise you 100!) It takes a few minutes
    to read it all in, check it and generate code for it, but no
    problems.
    
    So, I think I can say that fundamentally, there are no longer
    any limits on the number of items you can place in contexts,
    statement blocks, switches, or globals, beyond your virt mem
    constraints.
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177289 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-19 00:05:07 +00:00
Kevin P. Fleming
b72c17951f Merged revisions 177229 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r177229 | kpfleming | 2009-02-18 17:09:58 -0600 (Wed, 18 Feb 2009) | 3 lines
  
  fix two very minor bugs: if anyone ever uses SLINEAR16 as a format in RTP, ensure that the samples are byte-swapped to network order if needed. also, when a smoother is operating on a format that has a sample rate other than 8000 samples per second, use the proper sample rate for computing delivery timestamps.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177231 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-18 23:15:11 +00:00
David Vossel
48cccb7858 Merged revisions 177226 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r177226 | dvossel | 2009-02-18 16:51:38 -0600 (Wed, 18 Feb 2009) | 9 lines
  
  Locking issue in action_bridge and bridge_exec
  
  action_bridge() and bridge_exec() both search for the channels to bridge to, and then immediately drop the lock.  Instead, they should hold the lock until the masquerade is complete.  This will guarantee the channel remains and prevent any other weirdness from occurring.  In action_bridge() some more weirdness comes into play.  Both channels are needlessly locked at the same time and perform the exact same logic.  It makes sense from a coding organizational standpoint, but could cause a theoretical deadlock so I split the code up.  There is an issue associated with this, but since its a rather complicated thing to reproduce I'm not certain this alone will close it.
  
  issue# 14296
  Review: http://reviewboard.digium.com/r/167/
........


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2009-02-18 22:55:56 +00:00
Jeff Peeler
9fdccc9760 Merged revisions 177162 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r177162 | jpeeler | 2009-02-18 14:11:57 -0600 (Wed, 18 Feb 2009) | 14 lines
  
  Modify h323 to build against PTLib as well as the older PWLib
  
  Several changes in PTLib have occurred requiring build time detection. Changes
  accounted for include the library name change, config option change, install
  location change, and a boolean type change which is handled by ast_ptlib.h.
  Also, the sed check has been modified to properly work with autoconf >= 2.62.
  
  (closes issue #14224)
  Reported by: bergolth
  Patches:
        asterisk-autoconf-sed.patch uploaded by bergolth (license 661)
        asterisk-pwlib-v3.patch uploaded by bergolth (license 661)
  Tested by: jpeeler
........


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2009-02-18 20:15:37 +00:00
Russell Bryant
ffa0eaebef Merged revisions 177101 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
r177101 | russell | 2009-02-18 13:12:49 -0600 (Wed, 18 Feb 2009) | 8 lines

Re-add 'o' option to MeetMe, reverting rev 62297.

Enabling this option by default proved to be a bad idea, as the talker detection
is not very reliable.  So, make it optional again, and off by default.

(issue #13801)
Reported by: justdave

........


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2009-02-18 19:25:03 +00:00
Tilghman Lesher
fece998d9c Merged revisions 177098 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r177098 | tilghman | 2009-02-18 13:05:15 -0600 (Wed, 18 Feb 2009) | 9 lines
  
  Merged revisions 177096 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r177096 | tilghman | 2009-02-18 12:30:38 -0600 (Wed, 18 Feb 2009) | 2 lines
    
    Document the return value of the update method (as requested on -dev list)
  ........
................


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2009-02-18 19:07:05 +00:00
Doug Bailey
010ec27cfd Merged revisions 177035 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r177035 | dbailey | 2009-02-18 11:24:07 -0600 (Wed, 18 Feb 2009) | 2 lines
  
  Fixed error where a check for an zero length, terminated string was needed.
........


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2009-02-18 17:26:59 +00:00
Joshua Colp
b2a0fc1723 Merged revisions 177005 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r177005 | file | 2009-02-18 13:11:52 -0400 (Wed, 18 Feb 2009) | 6 lines
  
  Fix ordering of output for a ChannelUpdate manager event.
  (closes issue #14497)
  Reported by: vinsik
  Patches:
        chan_update_fix-chan_sip.c.diff uploaded by vinsik (license 623)
........


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2009-02-18 17:13:25 +00:00
Doug Bailey
20dea27b4b Merged revisions 176948 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r176948 | dbailey | 2009-02-18 10:09:12 -0600 (Wed, 18 Feb 2009) | 2 lines
  
  Need to take into account the \0 terminator of the old string to determine the amount available.
........


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2009-02-18 16:30:15 +00:00
Steve Murphy
babd94b38e Merged revisions 176943 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
r176943 | murf | 2009-02-18 08:35:26 -0700 (Wed, 18 Feb 2009) | 45 lines


This patch fixes merge_contexts_and_delete so it does not deadlock when hints are present.

Reason: when I re-engineered the merge_and_delete func to
reduce its lock time, I failed to notice that the 
functions it calls still also do locking as before.
This leads to deadlocks on dialplan reloads, when
there are actually living, subscribed hints registered
in the system.

While the reporter come across this problem while using
AEL, I might note that these deadlocks should also happen
if extensions.conf were used.

Here I added these routines to pbx.c:

ast_add_extension_nolock
add_pri_lockopt
ast_add_extension2_lockopt
find_context
add_hint_nolock

All of the above routines are static and restricted
to be used only within pbx.c, and more specifically
within the merge_contexts_and_delete routine.

They are pretty much the same as their counterparts
except they don't lock contexts or hints.

Most of them now do the real work of their
name-alike, with optional locking via extra arguments,
and are called by their name-alike. The goal was to
have the original functions so they would behave
exactly as before.

Both PJ and I tested these fixes, and the deadlocking
problem is no longer encountered.

(closes issue #14357)
Reported by: pj
Patches:
      14357.diff uploaded by murf (license 17)
Tested by: pj, murf


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176944 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-18 15:49:10 +00:00
Russell Bryant
d245def8b7 Blocked revisions 176904 via svnmerge
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r176904 | russell | 2009-02-18 00:14:47 -0600 (Wed, 18 Feb 2009) | 2 lines

Add example code for a heap traversal.

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2009-02-18 06:15:02 +00:00
Russell Bryant
eed32932d2 Blocked revisions 176901 via svnmerge
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r176901 | russell | 2009-02-18 00:00:40 -0600 (Wed, 18 Feb 2009) | 9 lines

Fix a number of incorrect uses of strncpy().

The big problem here is that the 3rd argument provided in these uses of strncpy()
did not reserve a byte for the null terminator, leaving the potential for writing
one byte past the end of the buffer.

Aside from this, there were coding guidelines violations with regards to spacing,
as well as hard coded lengths being used instead of sizeof().

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2009-02-18 06:01:25 +00:00
Dwayne M. Hubbard
da66687b2e Blocked revisions 176869 via svnmerge
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  r176869 | dhubbard | 2009-02-17 20:55:12 -0600 (Tue, 17 Feb 2009) | 7 lines
  
  T38 faxdetect should jump to the 'fax' extension for incoming calls only
  
  The previous implementation of T38 faxdetect resulted in both sides of the
  call jumping to a fax extension when both sides had 't38pt_udptl=yes' and
  'faxdetect=yes' in sip.conf and a 'fax' extension in the current context.
  This revision will jump to a 'fax' extension on incoming calls only.
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2009-02-18 03:01:46 +00:00
Kevin P. Fleming
43a18739de Blocked revisions 176841 via svnmerge
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  r176841 | kpfleming | 2009-02-17 20:02:54 -0600 (Tue, 17 Feb 2009) | 3 lines
  
  suppress smoothers for Siren codecs as well as Speex and G.723.1
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2009-02-18 02:03:32 +00:00
Shaun Ruffell
5c5511cb65 Merged revisions 176760 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r176760 | sruffell | 2009-02-17 16:28:41 -0600 (Tue, 17 Feb 2009) | 10 lines
  
  Several changes to codec_dahdi to play nice with G723.
  
  This commit brings in the changes that were living out on the
  svn/asterisk/team/sruffell/asterisk-trunk-transcoder branch.  codec_dahdi.c now
  always uses signed linear as the simple codec so that a soft g729 codec will
  not end up being preferred to the hardware codec.  There are also changes to
  allow codec_dahdi.c to feed packets to the hardware in the native sample size of
  the codec.  This solves problems with choppy audio when using G723. 
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2009-02-18 00:15:13 +00:00
Shaun Ruffell
d9a8cc0be1 Backing out an improper merge from trunk.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176804 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 23:47:15 +00:00
Shaun Ruffell
1755f4d31f Merged revisions 176760 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r176760 | sruffell | 2009-02-17 16:28:41 -0600 (Tue, 17 Feb 2009) | 10 lines
  
  Several changes to codec_dahdi to play nice with G723.
  
  This commit brings in the changes that were living out on the
  svn/asterisk/team/sruffell/asterisk-trunk-transcoder branch.  codec_dahdi.c now
  always uses signed linear as the simple codec so that a soft g729 codec will
  not end up being preferred to the hardware codec.  There are also changes to
  allow codec_dahdi.c to feed packets to the hardware in the native sample size of
  the codec.  This solves problems with choppy audio when using G723. 
........


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2009-02-17 23:36:59 +00:00
Russell Bryant
5594829c86 Blocked revisions 176771 via svnmerge
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r176771 | russell | 2009-02-17 16:52:43 -0600 (Tue, 17 Feb 2009) | 2 lines

Remove a dependency that no longer exists.

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2009-02-17 22:53:33 +00:00
Jeff Peeler
46db811169 Merged revisions 176708 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r176708 | jpeeler | 2009-02-17 16:08:00 -0600 (Tue, 17 Feb 2009) | 23 lines
  
  Merged revisions 176701 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r176701 | jpeeler | 2009-02-17 15:54:34 -0600 (Tue, 17 Feb 2009) | 17 lines
    
    Modify bridging to properly evaluate DTMF after first warning is played
    
    The main problem is currently if the Dial flag L is used with a warning sound,
    DTMF is not evaluated after the first warning sound. To fix this, a flag has 
    been added in ast_generic_bridge for playing the warning which ensures that if
    a scheduled warning is missed, multiple warrnings are not played back (due to a
    feature evaluation or waiting for digits). ast_channel_bridge was modified to
    store the nexteventts in the ast_bridge_config structure as that information
    was lost every time ast_channel_bridge was reentered, causing a hangup due to
    incorrect time calculations.
    
    (closes issue #14315)
    Reported by: tim_ringenbach
    
    Reviewed on reviewboard:
    http://reviewboard.digium.com/r/163/
  ........
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2009-02-17 22:14:38 +00:00
Dwayne M. Hubbard
24ef2a922e Merged revisions 176705 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r176705 | dhubbard | 2009-02-17 15:59:38 -0600 (Tue, 17 Feb 2009) | 11 lines
  
  create a UDPTL structure in create_addr_from_peer() if it does not already exist for T38
  
  This is required to create a UDPTL structure in create_addr_from_peer() to handle the
  scenario where 't38pt_udptl=yes' is not defined in the [general] section of sip.conf but 
  is defined the peer's context.  I tested this patch by enabling t38pt_udptl in the 
  [general] section on one system and only enabling t38pt_udptl in a peer's context on
  the system sending a fax.  Without the patch, the sending system will fail to initiate
  T38 negotiation with the warning message, "No way to add SDP without an UDPTL structure".
  When this patch is applied the sending side will successfully initiate T38 negotiation.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176709 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 22:09:16 +00:00
Mark Michelson
ddee1048c7 Merged revisions 176697 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r176697 | mmichelson | 2009-02-17 15:40:09 -0600 (Tue, 17 Feb 2009) | 3 lines
  
  Clear up documentation of AST_FRIENDLY_OFFSET in frame.h
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2009-02-17 21:40:40 +00:00
Russell Bryant
603e58ad3a Blocked revisions 176666 via svnmerge
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r176666 | russell | 2009-02-17 15:22:40 -0600 (Tue, 17 Feb 2009) | 16 lines

Update the timing API to have better support for multiple timing interfaces.

1) Add module use count handling so that timing modules can be unloaded.

2) Implement unload_module() functions for the timing interface modules.

3) Allow multiple timing modules to be loaded, and use the one with the
   highest priority value.

4) Report which timing module is being use in the "timing test" CLI command.

(closes issue #14489)
Reported by: russell

Review: http://reviewboard.digium.com/r/162/

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176670 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 21:23:15 +00:00
Tilghman Lesher
f8b942b064 Merged revisions 176642 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r176642 | tilghman | 2009-02-17 15:14:18 -0600 (Tue, 17 Feb 2009) | 8 lines
  
  Prior to masquerade, move the group definitions to the channel performing the
  masq, so that the group count lingers past the bridge.
  (closes issue #14275)
   Reported by: kowalma
   Patches: 
         20090216__bug14275.diff.txt uploaded by Corydon76 (license 14)
   Tested by: kowalma
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2009-02-17 21:15:10 +00:00
Russell Bryant
dab9950656 Blocked revisions 176639 via svnmerge
........
r176639 | russell | 2009-02-17 15:04:08 -0600 (Tue, 17 Feb 2009) | 9 lines

Significantly improve scheduler performance under high load.

This patch changes the scheduler to use a max-heap to store pending scheduler
entries instead of a fully sorted doubly linked list.  When the number of
entries in the scheduler gets large, this will perform much better.  For much
more detailed information on this change, see the review request.

Review: http://reviewboard.digium.com/r/160/

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2009-02-17 21:05:07 +00:00
Russell Bryant
8c08a905f9 Blocked revisions 176635 via svnmerge
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r176635 | russell | 2009-02-17 14:56:26 -0600 (Tue, 17 Feb 2009) | 4 lines

Add a test module for the heap implementation.

Review: http://reviewboard.digium.com/r/160/

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2009-02-17 20:56:41 +00:00
Russell Bryant
ecd8f84421 Blocked revisions 176632 via svnmerge
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r176632 | russell | 2009-02-17 14:51:10 -0600 (Tue, 17 Feb 2009) | 8 lines

Add an implementation of the heap data structure.

A heap is a convenient data structure for implementing a priority queue.

Code from svn/asterisk/team/russell/heap/.

Review: http://reviewboard.digium.com/r/160/

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2009-02-17 20:51:28 +00:00
Russell Bryant
1a121a8737 Blocked revisions 176627 via svnmerge
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r176627 | russell | 2009-02-17 14:41:24 -0600 (Tue, 17 Feb 2009) | 37 lines

Merge a large set of updates to the Asterisk indications API.

This patch includes a number of changes to the indications API.  The primary
motivation for this work was to improve stability.  The object management
in this API was significantly flawed, and a number of trivial situations could
cause crashes.

The changes included are:

1) Remove the module res_indications.  This included the critical functionality
   that actually loaded the indications configuration.  I have seen many people
   have Asterisk problems because they accidentally did not have an
   indications.conf present and loaded.  Now, this code is in the core,
   and Asterisk will fail to start without indications configuration.

   There was one part of res_indications, the dialplan applications, which did
   belong in a module, and have been moved to a new module, app_playtones.

2) Object management has been significantly changed.  Tone zones are now
   managed using astobj2, and it is no longer possible to crash Asterisk by
   issuing a reload that destroys tone zones while they are in use.

3) The API documentation has been filled out.

4) The API has been updated to follow our naming conventions.

5) Various bits of code throughout the tree have been updated to account
   for the API update.

6) Configuration parsing has been mostly re-written.

7) "Code cleanup"

The code is from svn/asterisk/team/russell/indications/.

Review: http://reviewboard.digium.com/r/149/

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2009-02-17 20:41:48 +00:00
Russell Bryant
fabeae2b4a Blocked revisions 176557 via svnmerge
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r176557 | russell | 2009-02-17 11:33:38 -0600 (Tue, 17 Feb 2009) | 12 lines

Fix a race condition that caused device states to become incorrect for hints.

The problem here is that the hint processing code was subscribed to the wrong
event type.  So, it started processing state for a hint too soon, before the
device state cache had been updated.

Also, fix a similar bug in app_queue, as it was also subscribed to the wrong
event type.

(closes issue #14461)
Reported by: alecdavis

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2009-02-17 17:34:29 +00:00
Tilghman Lesher
dce3b59e28 Oops, merge broke 1.6.0.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176548 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 16:27:24 +00:00
Tilghman Lesher
0ab34a15f5 In 1.6.0, the tablename is stored in a variable.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176502 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 14:42:10 +00:00
Tilghman Lesher
d019ac6ef4 Merged revisions 176459 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r176459 | tilghman | 2009-02-16 19:58:39 -0600 (Mon, 16 Feb 2009) | 17 lines
  
  Merged revisions 176426 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r176426 | tilghman | 2009-02-16 18:49:22 -0600 (Mon, 16 Feb 2009) | 10 lines
    
    After a 'sip reload', qualifies for realtime peers weren't immediately
    restarted, instead waiting until the next registration.  We're now
    caching the qualify across a reload/restart and starting the qualify
    immediately upon loading the peer.
    (closes issue #14196)
     Reported by: pdf
     Patches: 
           20090120__bug14196_1.4.diff.txt uploaded by pdf (license 663)
     Tested by: pdf
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 02:04:58 +00:00
David Vossel
f62cf2d8e3 Merged revisions 176355 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r176355 | dvossel | 2009-02-16 17:33:55 -0600 (Mon, 16 Feb 2009) | 13 lines
  
  Merged revisions 176354 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r176354 | dvossel | 2009-02-16 17:30:52 -0600 (Mon, 16 Feb 2009) | 8 lines
    
    Fixes issue with AST_CONTROL_SRCUPDATE not being relayed correctly during bridging
    
    This should have been committed with rev176247, but I missed it.  srcupdate frames no longer break out of the native bridge, but are not being sent to the other call leg either.  This fixs that.
    
    issue #13749
  ........
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2009-02-16 23:46:48 +00:00
Kevin P. Fleming
f64c4404b3 Blocked revisions 176356 via svnmerge
........
  r176356 | kpfleming | 2009-02-16 17:37:37 -0600 (Mon, 16 Feb 2009) | 3 lines
  
  add support for Siren7 and Siren14 flavors of prompts and music on hold
........


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2009-02-16 23:38:01 +00:00
Kevin P. Fleming
f3ae0e00a7 Merged revisions 176255 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r176255 | kpfleming | 2009-02-16 15:45:54 -0600 (Mon, 16 Feb 2009) | 13 lines
  
  Merged revisions 176216 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r176216 | kpfleming | 2009-02-16 15:10:38 -0600 (Mon, 16 Feb 2009) | 3 lines
    
    fix a flaw in the ast_string_field_build() family of API calls; these functions made no attempt to reuse the space already allocated to a field, so every time the field was written it would allocate new space, leading to what appeared to be a memory leak.
  ........
    r176254 | kpfleming | 2009-02-16 15:41:46 -0600 (Mon, 16 Feb 2009) | 3 lines
  
    correct a logic error in the last stringfields commit... don't mark additional space as allocated if the string was built using already-allocated space
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176258 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-16 21:50:47 +00:00
Mark Michelson
d504f89a9d Merged revisions 176253 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r176253 | mmichelson | 2009-02-16 15:40:40 -0600 (Mon, 16 Feb 2009) | 24 lines
  
  Merged revisions 176249,176252 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r176249 | mmichelson | 2009-02-16 15:34:27 -0600 (Mon, 16 Feb 2009) | 14 lines
    
    Open the DAHDI pseudo device and set it to be nonblocking atomically
    
    Apparently on FreeBSD, attempting to set the O_NONBLOCKING flag separately
    from opening the file was causing an "inappropriate ioctl for device" error.
    While I cannot fathom why this would be happening, I certainly am not opposed
    to making the code a bit more compact/efficient if it also fixes a bug.
    
    (closes issue #14482)
    Reported by: ys
    Patches:
          meetme.patch uploaded by ys (license 281)
    Tested by: ys
  ........
    r176252 | mmichelson | 2009-02-16 15:39:21 -0600 (Mon, 16 Feb 2009) | 3 lines
    
    Remove unused variable and make dev-mode compilation happy
  ........
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2009-02-16 21:47:11 +00:00
David Vossel
af1b475948 Merged revisions 176248 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r176248 | dvossel | 2009-02-16 15:30:17 -0600 (Mon, 16 Feb 2009) | 11 lines
  
  Merged revisions 175597 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/trunk
  
  ........
    r175597 | dvossel | 2009-02-13 14:11:55 -0600 (Fri, 13 Feb 2009) | 4 lines
    
    Fixed iax2 key rotation backwards compatibility
    
    Turns key rotation back on by default.  Added bit into encryption IE to indicate whether or not key rotation is supported or not. If it is not supported then it is not enabled, which insures backwards compatibility.  This eliminates the need for the keyrotate option in iax.conf, so it has been removed.  
  ........
................


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2009-02-16 21:35:07 +00:00
Mark Michelson
1999cdee6c Merged revisions 176174 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r176174 | mmichelson | 2009-02-16 12:25:57 -0600 (Mon, 16 Feb 2009) | 11 lines
  
  Assist proper thread synchronization when stopping the logger thread.
  
  I was finding that on my dev box, occasionally attempting to "stop now" in
  trunk would cause Asterisk to hang. I traced this to the fact that the logger
  thread was waiting on a condition which had already been signalled. The logger
  thread also need to be sure to check the value of the close_logger_thread variable.
  
  The close_logger_thread variable is only checked when the list of logmessages is empty.
  This allows for the logger thread to print and free any pending messages before exiting.
........


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2009-02-16 18:33:05 +00:00
Russell Bryant
968aada889 Blocked revisions 176100 via svnmerge
........
r176100 | russell | 2009-02-16 11:09:24 -0600 (Mon, 16 Feb 2009) | 4 lines

Remove chan_features.

Review: http://reviewboard.digium.com/r/161/

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2009-02-16 17:09:46 +00:00
Tilghman Lesher
c114192b13 Eliminate mention of a variable which is only available in trunk.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176098 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-16 17:06:39 +00:00
Joshua Colp
d86ea77d16 Merged revisions 176030 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r176030 | file | 2009-02-16 11:36:19 -0400 (Mon, 16 Feb 2009) | 16 lines
  
  Merged revisions 176029 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r176029 | file | 2009-02-16 11:33:53 -0400 (Mon, 16 Feb 2009) | 9 lines
    
    Don't have the Via header stored as a stringfield as it can change often during the lifetime of a dialog.
    
    This issue crept up with subscriptions on the AA50. When an outgoing NOTIFY is sent a new branch value
    is created and the Via header is changed to reflect it. Since this was a stringfield a new spot in the
    pool was used for the value while the old was left untouched/unused. If the current pool was full a new
    pool was created. This would cause memory usage to increase steadily.
    
    (issue #AA50-2332)
  ........
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2009-02-16 15:37:09 +00:00
Michiel van Baak
25c01347d8 Merged revisions 175952 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r175952 | mvanbaak | 2009-02-16 01:26:59 +0100 (Mon, 16 Feb 2009) | 10 lines
  
  Merged revisions 175921 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r175921 | mvanbaak | 2009-02-16 00:37:03 +0100 (Mon, 16 Feb 2009) | 3 lines
    
    fix mis-spelling of the word registered.
    Reported by De_Mon on #asterisk-dev.
  ........
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2009-02-16 09:40:22 +00:00
Russell Bryant
55f3396445 Blocked revisions 175983 via svnmerge
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r175983 | russell | 2009-02-15 20:54:42 -0600 (Sun, 15 Feb 2009) | 2 lines

Make the causes array static, and remove the type name as it is not needed.

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2009-02-16 02:55:24 +00:00
Russell Bryant
854d364ce9 Blocked revisions 175882 via svnmerge
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r175882 | russell | 2009-02-15 15:27:33 -0600 (Sun, 15 Feb 2009) | 2 lines

Make ast_sched_report() and ast_sched_dump() thread safe.

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2009-02-15 21:28:23 +00:00
Russell Bryant
abf8b910f9 Blocked revisions 175829 via svnmerge
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r175829 | russell | 2009-02-15 14:56:27 -0600 (Sun, 15 Feb 2009) | 14 lines

Fix a number of problems with ast_sched_report().

1) It had numerous coding guidelines violations with regards to formatting.

2) It allocated memory using ast_calloc() that was never freed.

3) It didn't check for failure from the allocation.

4) It used sprintf() and strcat() to build the result, doing zero checking to
   prevent writing past the end of the provided buffer.

The function also lacks API documentation, but that has not been addressed in
this commit.

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2009-02-15 20:56:55 +00:00
Olle Johansson
cd8a69a565 Merged revisions 175827 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r175827 | oej | 2009-02-15 21:39:55 +0100 (Sön, 15 Feb 2009) | 10 lines

Merged revisions 175825 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r175825 | oej | 2009-02-15 21:33:17 +0100 (Sön, 15 Feb 2009) | 2 lines

format_ilbc does not depend on codec libraries and can therefore always be made. My mistake. Ursäkta!

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2009-02-15 20:53:31 +00:00
Olle Johansson
fc681ee891 Merged revisions 175801 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r175801 | oej | 2009-02-15 21:22:12 +0100 (Sön, 15 Feb 2009) | 10 lines

Merged revisions 175792 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r175792 | oej | 2009-02-15 21:20:21 +0100 (Sön, 15 Feb 2009) | 2 lines

Disable format_ilbc.so by default, like codec_ilbc.so

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2009-02-15 20:25:42 +00:00
Russell Bryant
36cbd1bf2f Blocked revisions 175623,175636 via svnmerge
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r175623 | russell | 2009-02-13 14:23:39 -0600 (Fri, 13 Feb 2009) | 1 line

add missing </para>
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r175636 | russell | 2009-02-13 14:26:49 -0600 (Fri, 13 Feb 2009) | 1 line

fix a few more XML documentation problems
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2009-02-13 20:27:28 +00:00
Mark Michelson
aa60558799 Blocked revisions 175591 via svnmerge
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  r175591 | mmichelson | 2009-02-13 13:49:38 -0600 (Fri, 13 Feb 2009) | 22 lines
  
  Merged revisions 175590 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r175590 | mmichelson | 2009-02-13 13:47:48 -0600 (Fri, 13 Feb 2009) | 16 lines
    
    Fix a potential crash situation when using IMAP voicemail
    
    If calling into VoiceMailMain when using IMAP storage, it was
    possible to crash Asterisk by hanging up the phone when prompted
    for a voicemail mailbox. This patch fixes the issue.
    
    While it may appear that this patch is superficial, it allows code
    execution to continue to the failure case just below the IMAP_STORAGE
    code block where this patch has been applied
    
    (closes issue #14473)
    Reported by: dwpaul
    Patches:
          voicemail_imap_crash_no_mailbox.patch uploaded by dwpaul (license 689)
  ........
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2009-02-13 19:51:25 +00:00
Joshua Colp
496e168b87 Merged revisions 175549 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r175549 | file | 2009-02-13 12:41:15 -0400 (Fri, 13 Feb 2009) | 4 lines
  
  Add an option to keep the recorded file upon hangup.
  (closes issue #14341)
  Reported by: fnordian
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2009-02-13 16:43:13 +00:00
Kevin P. Fleming
351eab03b2 Blocked revisions 175512 via svnmerge
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  r175512 | kpfleming | 2009-02-13 07:41:52 -0600 (Fri, 13 Feb 2009) | 3 lines
  
  document G.722.1/.1C support
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2009-02-13 13:42:39 +00:00
Kevin P. Fleming
b0722c20ca Blocked revisions 175508 via svnmerge
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  r175508 | kpfleming | 2009-02-13 07:35:24 -0600 (Fri, 13 Feb 2009) | 15 lines
  
  Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14)
  
  This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.
  
  Along the way, some related work was done:
  
  1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.
  
  2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.
  
  3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).
  
  Review: http://reviewboard.digium.com/r/158/
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2009-02-13 13:36:49 +00:00
Dwayne M. Hubbard
89c50d282d Blocked revisions 175475 via svnmerge
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  r175475 | dhubbard | 2009-02-12 22:22:35 -0600 (Thu, 12 Feb 2009) | 1 line
  
  add 'faxbuffers' configuration option information to CHANGES
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2009-02-13 04:25:12 +00:00
Dwayne M. Hubbard
145fff8f3b Blocked revisions 175411 via svnmerge
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  r175411 | dhubbard | 2009-02-12 18:13:38 -0600 (Thu, 12 Feb 2009) | 13 lines
  
  Add dynamic fax buffer configuration option to chan_dahdi.conf
  
  When the 'faxdetect' configuration option is used, one may also want to use
  the 'faxbuffers' configuration option in chan_dahdi.conf.  This option will
  dynamically use the configured 'faxbuffers' buffer policy on a channel for
  the life of the call following the detection of fax tones.  The faxbuffers
  buffer policy will be reverted during call teardown.
  
  An example use of 'faxbuffers' is below.  This example would switch to using
  6 buffers with a full buffer policy.
  
  faxbuffers=>6,full
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2009-02-13 03:46:55 +00:00
Russell Bryant
48ade8a53e Merged revisions 175368 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r175368 | russell | 2009-02-12 15:41:01 -0600 (Thu, 12 Feb 2009) | 2 lines

Remove useless string copy, and make sscanf safe again

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2009-02-12 21:41:20 +00:00
David Vossel
4ddeba5e16 Blocked revisions 175344 via svnmerge
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  r175344 | dvossel | 2009-02-12 15:27:11 -0600 (Thu, 12 Feb 2009) | 10 lines
  
  Adds force encryption option to iax.conf
  
  This patch adds forceencryption=yes as an iax.conf option.  When force encryption is enabled, no unencrypted connections are allowed.  This insures all connections are encrypted.  This is a new feature, so CHANGES and iax.conf.sample are updated as well.   
  
  (closes issue #13285)
  Reported by: sgofferj
  Tested by: russell
  Review: http://reviewboard.digium.com/r/150/
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2009-02-12 21:32:54 +00:00
Tilghman Lesher
96a87efb7f Merged revisions 175334 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r175334 | tilghman | 2009-02-12 15:25:14 -0600 (Thu, 12 Feb 2009) | 16 lines
  
  Merged revisions 175311 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r175311 | tilghman | 2009-02-12 15:19:40 -0600 (Thu, 12 Feb 2009) | 9 lines
    
    Fix crashes when receiving certain T.38 packets.  Also, increase the maximum
    size of T.38 packets and warn users when they try to set the limits above those
    maximums.
    (closes issue #13050)
     Reported by: schern
     Patches: 
           20090212__bug13050.diff.txt uploaded by Corydon76 (license 14)
     Tested by: schern
  ........
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2009-02-12 21:27:58 +00:00
Jeff Peeler
8008dca29d Fix mistake in merging conflict from 175299.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175301 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-12 20:59:09 +00:00
Jeff Peeler
0a72cfe440 Merged revisions 175298 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r175298 | jpeeler | 2009-02-12 14:48:56 -0600 (Thu, 12 Feb 2009) | 15 lines
  
  Merged revisions 175294 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r175294 | jpeeler | 2009-02-12 14:34:36 -0600 (Thu, 12 Feb 2009) | 9 lines
    
    Fix ParkedCall event information for From field in the case of a blind transfer
    
    If the parker information can not be obtained from the peer, try and see if
    the BLINDTRANSFER channel variable has been set. Previously, a blind transfer
    to the ParkAndAnnounce app would return nothing for the From.
    
    Closes AST-189
  ........
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2009-02-12 20:50:30 +00:00
Russell Bryant
869e8bc417 Merged revisions 175295 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r175295 | russell | 2009-02-12 14:45:47 -0600 (Thu, 12 Feb 2009) | 2 lines

Avoid using ast_strdupa() in a loop.

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2009-02-12 20:46:11 +00:00
Russell Bryant
ecd3f17267 Merged revisions 175255 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r175255 | russell | 2009-02-12 13:11:08 -0600 (Thu, 12 Feb 2009) | 4 lines

Don't enable something by default that has a dependency on something _not_ enabled by default.

menuselect was not happy with this.

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2009-02-12 19:11:46 +00:00
Kevin P. Fleming
2a60c89b53 Blocked revisions 175250 via svnmerge
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  r175250 | kpfleming | 2009-02-12 12:48:52 -0600 (Thu, 12 Feb 2009) | 1 line
  
  correct warning message to not refer specifically to DAHDI
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2009-02-12 18:50:22 +00:00
Jeff Peeler
17df1c9d13 Merged revisions 175188 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r175188 | jpeeler | 2009-02-12 12:00:11 -0600 (Thu, 12 Feb 2009) | 12 lines
  
  Merged revisions 175187 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r175187 | jpeeler | 2009-02-12 11:57:10 -0600 (Thu, 12 Feb 2009) | 6 lines
    
    Fix crash in event of failed attempt to transfer to parking
    
    The peer may not necessarily exist, such as in the case of a transfer to 
    ParkAndAnnounce. In this case don't try to play a sound to it.
  ........
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2009-02-12 18:00:49 +00:00
Russell Bryant
d79cc1e799 Merged revisions 175125 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r175125 | russell | 2009-02-12 10:57:25 -0600 (Thu, 12 Feb 2009) | 35 lines

Merged revisions 175124 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r175124 | russell | 2009-02-12 10:51:13 -0600 (Thu, 12 Feb 2009) | 27 lines

Don't send DTMF for infinite time if we do not receive an END event.

I thought that this was going to end up being a pretty gnarly fix, but it turns
out that there was actually already a configuration option in rtp.conf, 
dtmftimeout, that was intended to handle this situation.  However, in between 
Asterisk 1.2 and Asterisk 1.4, the code that processed the option got lost.
So, this commit brings it back to life.

The default timeout is 3 seconds.  However, it is worth noting that having
this be configurable at all is not really the recommended behavior in RFC 2833.
From Section 3.5 of RFC 2833:

      Limiting the time period of extending the tone is necessary
      to avoid that a tone "gets stuck". Regardless of the
      algorithm used, the tone SHOULD NOT be extended by more than
      three packet interarrival times. A slight extension of tone
      durations and shortening of pauses is generally harmless.

Three seconds will pretty much _always_ be far more than three packet 
interarrival times.  However, that behavior is not required, so I'm going to
leave it with our legacy behavior for now.

Code from svn/asterisk/team/russell/issue_14460

(closes issue #14460)
Reported by: moliveras

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2009-02-12 17:03:21 +00:00
Mark Michelson
90ef4eb33e Merged revisions 175121 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r175121 | mmichelson | 2009-02-12 10:28:06 -0600 (Thu, 12 Feb 2009) | 11 lines
  
  Make lock information for ao2_trylock be more useful and gnarly
  
  Core show locks information involving an ao2_trylock did not
  show the function that called ao2_trylock, but would instead
  show ao2_trylock as the source of the lock. This is not useful
  when trying to debug locking issues.
  
  One bizarre note is that this logic is already in 1.4 but somehow
  did not get merged to trunk or the 1.6.X branches.
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2009-02-12 16:33:24 +00:00
Philippe Sultan
4c8559e441 Merged revisions 175089 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r175089 | phsultan | 2009-02-12 15:25:03 +0100 (Thu, 12 Feb 2009) | 6 lines

Issue a warning message if our candidate's IP is the loopback address.

(closes issue #13985)
Reported by: jcovert
Tested by: phsultan

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2009-02-12 14:27:59 +00:00
Philippe Sultan
e28edbe89c Merged revisions 175058 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r175058 | phsultan | 2009-02-12 11:31:36 +0100 (Thu, 12 Feb 2009) | 20 lines

Merged revisions 175029 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r175029 | phsultan | 2009-02-12 11:16:21 +0100 (Thu, 12 Feb 2009) | 12 lines

Set the initiator attribute to lowercase in our replies when receiving calls.

This attribute contains a JID that identifies the initiator of the GoogleTalk
voice session. The GoogleTalk client discards Asterisk's replies if the 
initiator attribute contains uppercase characters.

(closes issue #13984)
Reported by: jcovert
Patches:
      chan_gtalk.2.patch uploaded by jcovert (license 551)
Tested by: jcovert

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2009-02-12 10:42:03 +00:00
Mark Michelson
b8356f9a94 Merged revisions 174948 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r174948 | mmichelson | 2009-02-11 17:03:08 -0600 (Wed, 11 Feb 2009) | 35 lines
  
  Fix odd "thank you" sound playing behavior in app_queue.c
  
  If someone has configured the queue to play an position or holdtime
  announcement, then it is odd and potentially unexpected to hear a 
  "Thank you for your patience" sound when no position or holdtime
  was actually announced.
  
  This fixes the announcement so that the "thanks" sound is only played
  in the case that a position or holdtime was actually announced.
  
  There is a way that the "thank you" sound can be played without a
  position or holdtime, and that is to set announce-frequency to a value
  but keep announce-position and announce-holdtime both turned off.
  
  (closes issue #14227)
  Reported by: caspy
  Patches:
        14227_v3.patch uploaded by putnopvut (license 60)
  Tested by: caspy
  ................


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2009-02-11 23:04:10 +00:00
Mark Michelson
a45ec0c30a Merged revisions 174945 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r174945 | mmichelson | 2009-02-11 16:41:01 -0600 (Wed, 11 Feb 2009) | 29 lines
  
  Fix 'd' option for app_dial and add new option to Answer application
  
  The 'd' option would not work for channel types which use RTP to transport
  DTMF digits. The only way to allow for this to work was to answer the channel
  if we saw that this option was enabled.
  
  I realized that this may cause issues with CDRs, specifically with giving false
  dispositions and answer times. I therefore modified ast_answer to take another
  parameter which would tell if the CDR should be marked answered.
  
  I also extended this to the Answer application so that the channel may be answered
  but not CDRified if desired.
  
  I also modified app_dictate and app_waitforsilence to only answer the channel if it
  is not already up, to help not allow for faulty CDR answer times.
  
  All of these changes are going into Asterisk trunk. For 1.6.0 and 1.6.1, however, all
  the changes except for the change to the Answer application will go in since we do
  not introduce new features into stable branches
  
  (closes issue #14164)
  Reported by: DennisD
  Patches:
        14164.patch uploaded by putnopvut (license 60)
  Tested by: putnopvut
  
  Review: http://reviewboard.digium.com/r/145
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2009-02-11 22:48:11 +00:00
Joshua Colp
d4b9afebd8 Blocked revisions 174844 via svnmerge
........
  r174844 | file | 2009-02-11 10:44:47 -0400 (Wed, 11 Feb 2009) | 10 lines
  
  Tell the device state core a change happened when a channel is freed but not a specific state.
  We need to do this because while we know that the freeing of the channel may cause something to become
  not in use we do not know this for sure. There may be another channel that is still up which would cause
  it to be in use.
  (closes issue #13238)
  Reported by: kowalma
  Patches:
        20090121__bug13238.diff.txt uploaded by Corydon76 (license 14)
  Tested by: alecdavis
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2009-02-11 14:45:24 +00:00
Mark Michelson
ae6b71dfb6 Merged revisions 174805 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r174805 | mmichelson | 2009-02-10 17:17:03 -0600 (Tue, 10 Feb 2009) | 11 lines

Fix potential for stack overflows in app_chanspy.c

When using the 'g' or 'e' options, the stack allocations that
were used could cause a stack overflow if a spyer stayed on the
line long enough without actually successfully spying on anyone.

The problem has been corrected by using static buffers and copying
the contents of the appropriate strings into them instead of using
functions like alloca or ast_strdupa


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2009-02-10 23:20:27 +00:00
Mark Michelson
0aaff466b7 Merged revisions 174764 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
r174764 | mmichelson | 2009-02-10 15:45:14 -0600 (Tue, 10 Feb 2009) | 21 lines

Fix an fd leak that would occur in HTTP AMI sessions

The explanation behind this fix is a bit complicated, and I've already
typed it up in the code as a huge comment inside of manager.c, so I'll
give the abridged version here.

We needed a way to separate action-specific data from session-specific data.
Unfortunately, the only way to maintain API compatibility and to not have to
change every single manager action was to rename the current mansession structure
and wrap it inside a new mansession structure which actually contains action-
specific data.

(closes issue #14364)
Reported by: awk
Patches:
      14364_better.patch uploaded by putnopvut (license 60)
Tested by: putnopvut

Review: http://reviewboard.digium.com/r/148/


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@174765 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-10 21:49:14 +00:00
Joshua Colp
346b766917 Merged revisions 174710 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r174710 | file | 2009-02-10 16:15:43 -0400 (Tue, 10 Feb 2009) | 4 lines
  
  Only decrease inringing count if above zero.
  (issue #13238)
  Reported by: kowalma
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@174711 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-10 20:16:57 +00:00
Matthew Nicholson
bd217cb067 Merged revisions 174584 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r174584 | mnicholson | 2009-02-10 12:16:31 -0600 (Tue, 10 Feb 2009) | 25 lines
  
  Merged revisions 174583 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r174583 | mnicholson | 2009-02-10 11:52:42 -0600 (Tue, 10 Feb 2009) | 18 lines
    
    Improve behavior of jitterbuffer when maxjitterbuffer is set.
    
    This change improves the way the jitterbuffer handles maxjitterbuffer and
    dramatically reduces the number of frames dropped when maxjitterbuffer is
    exceeded.  In the previous jitterbuffer, when maxjitterbuffer was exceeded, all
    new frames were dropped until the jitterbuffer is empty.  This change modifies
    the code to only drop frames until maxjitterbuffer is no longer exceeded.
    
    Also, previously when maxjitterbuffer was exceeded, dropped frames were not
    tracked causing stats for dropped frames to be incorrect, this change also
    addresses that problem.
    
    (closes issue #14044)
    Patches:
          bug14044-1.diff uploaded by mnicholson (license 96)
    Tested by: mnicholson
    Review: http://reviewboard.digium.com/r/144/
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@174596 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-10 18:19:45 +00:00
Joshua Colp
a58fd5da85 Blocked revisions 174580 via svnmerge
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  r174580 | file | 2009-02-10 13:48:29 -0400 (Tue, 10 Feb 2009) | 4 lines
  
  Set the type for the peer structure to be a peer as the default.
  (closes issue #14447)
  Reported by: triccyx
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@174581 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-10 17:49:01 +00:00
Joshua Colp
c558922e9c Merged revisions 174543 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r174543 | file | 2009-02-10 11:37:07 -0400 (Tue, 10 Feb 2009) | 6 lines
  
  Make the logic for inuse and inringing manipluation match that of 1.4. The old broken logic would reset the values back to 0 during certain scenarios causing the wrong state to be reported.
  (closes issue #14399)
  Reported by: caspy
  (issue #13238)
  Reported by: kowalma
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2009-02-10 15:39:00 +00:00
Steve Murphy
707028163d For some strange reason, I didn't think 1.6.0 needed
this fix. I was wrong. Here it is.



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@174439 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-10 05:06:43 +00:00
Steve Murphy
936bba2b0c Blocked revisions 174435 via svnmerge
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r174435 | murf | 2009-02-09 21:49:02 -0700 (Mon, 09 Feb 2009) | 8 lines

This patch removes the use of AST_PBX_KEEPALIVE
from app_rpt.c.


(closes issue #14435)
Reported by: D_McNaul


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2009-02-10 04:51:39 +00:00
Steve Murphy
be401817a6 Blocked revisions 174432 via svnmerge
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r174432 | murf | 2009-02-09 21:36:22 -0700 (Mon, 09 Feb 2009) | 3 lines

More intptr_t work.


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2009-02-10 04:40:26 +00:00
Steve Murphy
179c759324 Blocked revisions 174370 via svnmerge
................
  r174370 | murf | 2009-02-09 19:45:56 -0700 (Mon, 09 Feb 2009) | 10 lines
  
  Merged revisions 174369 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r174369 | murf | 2009-02-09 19:27:40 -0700 (Mon, 09 Feb 2009) | 5 lines
    
    This patch solves some compiler complaints
    in both 32 and 64-bit environments.
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@174371 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-10 03:15:57 +00:00
Mark Michelson
f4113354e4 Merged revisions 174327 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r174327 | mmichelson | 2009-02-09 11:27:32 -0600 (Mon, 09 Feb 2009) | 3 lines

Fix something I messed up in the merge I just did


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@174328 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-09 17:28:52 +00:00
Mark Michelson
0919e13437 Merged revisions 174301 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r174301 | mmichelson | 2009-02-09 11:20:55 -0600 (Mon, 09 Feb 2009) | 20 lines

Merged revisions 174282 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r174282 | mmichelson | 2009-02-09 11:11:05 -0600 (Mon, 09 Feb 2009) | 12 lines

Don't do an SRV lookup if a port is specified

RFC 3263 says to do A record lookups on a hostname
if a port has been specified, so that's what we're
going to do. See section 4.2.

(closes issue #14419)
Reported by: klaus3000
Patches:
      patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000 (license 65)


........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@174322 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-09 17:25:09 +00:00
Joshua Colp
1eb4c5d727 Merged revisions 174219 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r174219 | file | 2009-02-09 10:49:24 -0400 (Mon, 09 Feb 2009) | 11 lines
  
  Merged revisions 174218 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r174218 | file | 2009-02-09 10:48:21 -0400 (Mon, 09 Feb 2009) | 4 lines
    
    Don't overwrite our pointer to the music class when music on hold stops. We will use this if it starts again to see if we can resume the music where it left off.
    (closes issue #14407)
    Reported by: mostyn
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@174220 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-09 14:50:09 +00:00
Russell Bryant
40864afe1d Merged revisions 174149 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r174149 | russell | 2009-02-07 10:16:50 -0600 (Sat, 07 Feb 2009) | 10 lines

Merged revisions 174148 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r174148 | russell | 2009-02-07 10:15:07 -0600 (Sat, 07 Feb 2009) | 2 lines

Fix a race condition that could cause a crash.

........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@174151 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-07 16:17:20 +00:00
Dwayne M. Hubbard
24e312a999 Merged revisions 174084 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r174084 | dhubbard | 2009-02-06 17:51:56 -0600 (Fri, 06 Feb 2009) | 13 lines

Merged revisions 174082 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r174082 | dhubbard | 2009-02-06 17:36:03 -0600 (Fri, 06 Feb 2009) | 5 lines

check ast_strlen_zero() before calling ast_strdupa() in sip_uri_headers_cmp()
and sip_uri_params_cmp()

The reporter didn't actually upload a properly-formed patch, instead a 
modified chan_sip.c file was uploaded.  I created a patch to determine the
changes, then modified the suggested changes to create a proper fix.  The
summary above is a complete description of the changes.

(closes issue #13547)
Reported by: tecnoxarxa
Patches:
      chan_sip.c.gz uploaded by tecnoxarxa (license 258)
Tested by: tecnoxarxa

........

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------------------------------------------------------------------------


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@174085 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-06 23:59:42 +00:00
David Vossel
f42760444c Blocked revisions 174046 via svnmerge
........
  r174046 | dvossel | 2009-02-06 14:12:33 -0600 (Fri, 06 Feb 2009) | 12 lines
  
  Adds immediate yes/no option to iax.conf
  
  This is very similar to the DAHDI immediate=yes option.  When the phone is picked up, instead of giving a dialtone it connects directly to the "s" extension.  Changes where implemented in chan_iax2.c to directly connect to the "s" extension in the appropriate context when this option is enabled.  Examples explaining its use are added to iax2.conf.sample.  CHANGES has been updated as well. 
  
  (closes issue #14266)
  Reported by: jcovert
  Patches:
        chan_iax2.c.patch-trunk uploaded by jcovert (license 551)
        iax.conf.sample.patch uploaded by jcovert (license 551)
  Tested by: jcovert, dvossel
  Review: http://reviewboard.digium.com/r/143/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@174075 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-06 20:25:58 +00:00
Joshua Colp
e9df380e47 Merged revisions 174041 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r174041 | file | 2009-02-06 15:28:53 -0400 (Fri, 06 Feb 2009) | 4 lines
  
  Don't subscribe to a mailbox on pseudo channels. It is futile. This solves an issue where duplicated pseudo channels would cause a crash because the first one would unsubscribe and the next one would also try to unsubscribe the same subscription.
  (closes issue #14322)
  Reported by: amessina
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@174042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-06 19:29:41 +00:00
Joshua Colp
494e3efac7 Merged revisions 173974 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r173974 | file | 2009-02-06 13:18:35 -0400 (Fri, 06 Feb 2009) | 15 lines
  
  Merged revisions 173967-173968 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r173967 | file | 2009-02-06 13:14:15 -0400 (Fri, 06 Feb 2009) | 4 lines
    
    Some clients do not put the call-id for replaces at the beginning, so support it being anywhere in the string.
    (closes issue #14350)
    Reported by: fhackenberger
  ........
    r173968 | file | 2009-02-06 13:15:01 -0400 (Fri, 06 Feb 2009) | 2 lines
    
    Remove a debug message I put in by accident.
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@173986 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-06 17:21:02 +00:00
Matthew Nicholson
76e95662cf Merged revisions 173952 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r173952 | mnicholson | 2009-02-06 10:28:19 -0600 (Fri, 06 Feb 2009) | 14 lines
  
  Merged revisions 173917 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb 2009) | 7 lines
    
    Limit the addition of the Contact header in SIP responses according to various
    SIP RFCs.
    
    (closes issue #13602)
    Reported by: hjourdain
    Tested by: mnicholson
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@173963 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-06 16:33:05 +00:00
Joshua Colp
514d93d9a0 Blocked revisions 173902 via svnmerge
........
  r173902 | file | 2009-02-06 11:59:17 -0400 (Fri, 06 Feb 2009) | 4 lines
  
  Always detach and destroy the whisper and barge audiohooks. Additionally also allow an audiohook to be detached if it has not been attached.
  (closes issue #14414)
  Reported by: bluecrow76
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@173903 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-06 15:59:54 +00:00
Russell Bryant
c8571f59e7 Blocked revisions 173858 via svnmerge
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r173858 | russell | 2009-02-06 04:55:35 -0600 (Fri, 06 Feb 2009) | 13 lines

Add a common implementation of a scheduler context with a dedicated thread.

This commit expands the Asterisk scheduler API to include a common implementation
of a scheduler context being processed by a dedicated thread.  chan_iax2 has been
updated to use this new code.  Also, as a result, this resolves some race
conditions related to the previous chan_iax2 scheduler handling.

Related to rev 171452 which resolved the same issues in 1.4.

Code from team/russell/sched_thread2

Review: http://reviewboard.digium.com/r/129/

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@173859 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-06 10:56:23 +00:00
Russell Bryant
e3a21e6e01 Blocked revisions 173848 via svnmerge
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r173848 | russell | 2009-02-06 04:25:09 -0600 (Fri, 06 Feb 2009) | 2 lines

Resolve a memory leak that would occur on an invalid channel given to Action: Status

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2009-02-06 10:25:40 +00:00
Mark Michelson
d652d5c06a Merged revisions 173776 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r173776 | mmichelson | 2009-02-05 17:48:48 -0600 (Thu, 05 Feb 2009) | 14 lines

Update extensions.conf.sample to be correct.

In trunk, the only necessary change pointed out was that the call
to ChanIsAvail uses an option that has been removed.

For the 1.6.1 branch, however, it appears that the sample file is
badly in need of updating since there are |'s used all over the place
there. My tentative plan is just to copy trunk's sample config file
to those branches since the info there is most up-to-date and should
be correct for use in 1.6.1

Thanks to macli in #asterisk-dev for bringing this up


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@173777 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-05 23:51:13 +00:00
Mark Michelson
e7a195d88b Merged revisions 173773 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r173773 | mmichelson | 2009-02-05 17:28:19 -0600 (Thu, 05 Feb 2009) | 7 lines

Properly set "seen" and "unseen" flags when moving messages from the new to the old folder when using IMAP for voicemail storage

(closes issue #13905)
Reported by: jaroth
Patches:
      foldermove_v2.patch uploaded by jaroth (license 50)

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2009-02-05 23:30:58 +00:00
Jeff Peeler
4bd27c1d11 Merged revisions 173697 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r173697 | jpeeler | 2009-02-05 15:00:26 -0600 (Thu, 05 Feb 2009) | 18 lines
  
  Merged revisions 173696 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r173696 | jpeeler | 2009-02-05 14:47:51 -0600 (Thu, 05 Feb 2009) | 12 lines
    
    Add new configuration option to make shared IMAP mailboxes function as expected.
    
    The new option is "imapvmshareid" which is an ID to tag multiple mailboxes
    using the same IMAP storage location to function as one mailbox. This allows
    all messages to be retrieved for any user in the group. The patch alters the
    'X-Asterisk-VM-Extension' header that is responsible for matching voicemails
    for a given user.
    
    (closes issue #13673)
    Reported by: howardwilkinson
  ........
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2009-02-05 21:04:57 +00:00
Mark Michelson
a31c0961ab Merged revisions 173693 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r173693 | mmichelson | 2009-02-05 14:30:45 -0600 (Thu, 05 Feb 2009) | 20 lines

Merged revisions 173692 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r173692 | mmichelson | 2009-02-05 14:29:09 -0600 (Thu, 05 Feb 2009) | 12 lines

Fix situations where queue members could be autopaused unexpectedly

Specifically, this patch prevents us from autopausing members when
we receive a busy or congestion frame from them.

(closes issue #14376)
Reported by: fiddur
Patches:
      14376.patch uploaded by putnopvut (license 60)
Tested by: fiddur


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2009-02-05 20:34:44 +00:00
Mark Michelson
1efdb3072c Merged revisions 173593 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r173593 | mmichelson | 2009-02-05 12:48:55 -0600 (Thu, 05 Feb 2009) | 11 lines

Merged revisions 173592 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r173592 | mmichelson | 2009-02-05 12:47:24 -0600 (Thu, 05 Feb 2009) | 3 lines

Add some missing cleanup to app_mixmonitor


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2009-02-05 18:49:22 +00:00
Mark Michelson
32f3ed9929 Merged revisions 173589 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r173589 | mmichelson | 2009-02-05 12:34:06 -0600 (Thu, 05 Feb 2009) | 33 lines

Merged revisions 173559 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r173559 | mmichelson | 2009-02-05 11:34:33 -0600 (Thu, 05 Feb 2009) | 25 lines

Fix a problem where a channel pointer becomes invalid due to masquerading or hanging up.

app_mixmonitor runs its own thread to monitor the channel's activity and write the mixed
audio to a file. Since this thread runs independently of the channel, it is possible that
the mixmonitor thread's channel pointer will point to freed memory when the channel either
is masqueraded or hangs up (technically, both cases are hangups, but we need to handle the
cases slightly differently).

The solution for this is to employ a datastore, which has the nice benefit of allowing us 
to hook into channel masquerades and hangups and update our pointer as necessary. If this
looks familiar, this same technique is employed in app_chanspy. app_chanspy is a bit more
involved since it does a lot more operations on the channel that is being spied upon.

app_mixmonitor does have an extra touch that app_chanspy doesn't have, though. Since there
is a thread race between the channel's thread and the mixmonitor thread on a hangup, we em-
ploy a condition-and-boolean combination to ensure that the channel thread finishes with
our structure before the mixmonitor thread attempts to free it. No crashes!

(closes issue #14374)
Reported by: aragon
Patches:
	  14374.patch uploaded by putnopvut (license 60)
Tested by: aragon, putnopvut


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2009-02-05 18:39:35 +00:00
Jeff Peeler
d7d53fa278 fix WORKING_FORK detection
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@173554 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-05 16:23:58 +00:00
Tilghman Lesher
24dc2f1054 regenerate with bootstrap.sh
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@173548 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-05 00:11:35 +00:00
Jeff Peeler
1b4e2a5d6b I messed up and accidentally reverted the trunk-merged prop before committing 173546. Added it manually.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@173547 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-04 23:44:56 +00:00
Jeff Peeler
58d71a9470 Merged revisions 173500 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r173500 | jpeeler | 2009-02-04 15:17:53 -0600 (Wed, 04 Feb 2009) | 23 lines
  
  Merged revisions 173211 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r173211 | jpeeler | 2009-02-03 15:57:01 -0600 (Tue, 03 Feb 2009) | 17 lines
    
    Parking attempts made to one end of a bridge no longer will hang up due to a
    parking failure.
    
    Parking attempts made using either one-touch, or doing either a blind or 
    assisted transfer to the parking extension now keep up the bridge instead of
    hanging up the attempted parked party. Normal causes for the parking attempt
    to fail includes the specific specified extension (via PARKINGEXTEN) not being 
    available or if all the parking spaces are currently in use. To avoid having
    to reverse a masquerade park_space_reserve was made to provide foresight if
    a parking attempt will succeed and if so reserve the parking space.
    
    (closes issue #13494)
    Reported by: mdu113
    
    Reviewed by Russell: http://reviewboard.digium.com/r/133/
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@173546 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-04 23:38:54 +00:00
Mark Michelson
f94677ac39 Merged revisions 173507 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r173507 | mmichelson | 2009-02-04 16:16:19 -0600 (Wed, 04 Feb 2009) | 7 lines

Fix some areas where the incorrect interface was passed to ast_device_state

I swear it feels like I already did this once...

(closes issue #14359)
Reported by: francesco_r

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@173534 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-04 22:23:52 +00:00
Tilghman Lesher
8c1a726973 Merged revisions 173458 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r173458 | tilghman | 2009-02-04 12:48:06 -0600 (Wed, 04 Feb 2009) | 9 lines
  
  When using a socket as a FILE *, the stdio functions will sometimes try to do
  an fseek() on the stream, which is an invalid operation for a socket.  Turning
  off buffering explicitly lets the stdio functions know they cannot do this,
  thus avoiding a potential error.
  (closes issue #14400)
   Reported by: fnordian
   Patches: 
         tcptls.patch uploaded by fnordian (license 110)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@173460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-04 18:55:32 +00:00
Mark Michelson
826418a799 Merged revisions 173397 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r173397 | mmichelson | 2009-02-04 11:45:14 -0600 (Wed, 04 Feb 2009) | 11 lines

Merged revisions 173396 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r173396 | mmichelson | 2009-02-04 11:44:48 -0600 (Wed, 04 Feb 2009) | 3 lines

Revert my previous change because it was stupid


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2009-02-04 17:46:12 +00:00
Mark Michelson
e1136de5bf Merged revisions 173393 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r173393 | mmichelson | 2009-02-04 11:41:02 -0600 (Wed, 04 Feb 2009) | 11 lines

Merged revisions 173392 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r173392 | mmichelson | 2009-02-04 11:40:29 -0600 (Wed, 04 Feb 2009) | 3 lines

Add a missing unlock. Extremely unlikely to ever matter, but it's needed.


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2009-02-04 17:42:01 +00:00
Mark Michelson
a71ca0b0e3 Merged revisions 173354 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r173354 | mmichelson | 2009-02-04 09:30:12 -0600 (Wed, 04 Feb 2009) | 30 lines

Fix a problem where file playback would cause fds to remain open forever

The problem came from the fact that a frame read from a format interpreter
was not freed. Adding a call to ast_frfree fixed this. The explanation for
why this caused the problem is a bit complex, but here goes:

There was a problem in all versions of Asterisk where the embedded frame
of a filestream structure was referenced after the filestream was freed. This
was fixed by adding reference counting to the filestream structure. The refcount
would increase every time that a filestream's frame pointer was pointing to an
actual frame of data. When the frame was freed, the refcount would decrease. Once
the refcount reached 0, the filestream was freed, and as part of the operation,
the open files were closed as well.

Thus it becomes more clear why a missing ast_frfree would cause a reference leak
and cause the files to not be closed. You may ask then if there was a frame leak
before this patch. The answer to that is actually no! The filestream code was
"smart" enough to know that since the frame we received came from a format interpreter,
the frame had no malloced data and thus didn't need to be freed. Now, however, there
is cleanup that needs to be done when we finish with the frame, so we do need to
call ast_frfree on the frame to be sure that the refcount for the filestream is
decremented appropriately.

(closes issue #14384)
Reported by: fiddur
Patches:
      14384.patch uploaded by putnopvut (license 60)
Tested by: fiddur, putnopvut


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@173355 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-04 15:30:54 +00:00
Tilghman Lesher
3d71c38287 Merged revisions 173311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r173311 | tilghman | 2009-02-03 18:43:52 -0600 (Tue, 03 Feb 2009) | 10 lines
  
  Ensure that commas placed in the middle of extension character classes do not
  interfere with correct parsing of the extension.  Also, if an unterminated
  character class DOES make its way into the pbx core (through some other
  method), ensure that it does not crash Asterisk.
  (closes issue #14362)
   Reported by: Nick_Lewis
   Patches: 
         20090129__bug14362.diff.txt uploaded by Corydon76 (license 14)
   Tested by: Corydon76
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@173312 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-04 00:45:32 +00:00
David Vossel
1c73c18e80 Fixes issue with IAX2 transfer not handing of calls.
Fixes issue with IAX2 transfers not taking place.  As it was, a call that was being transfered would never be handed off correctly to the call ends because of how call numbers were stored in a hash table.  The hash table, "iax_peercallno_pvt", storing all the current call numbers did not take into account the complications associated with transferring a call, so a separate hash table was required.  This second hash table "iax_transfercallno_pvt" handles calls being transfered, once the call transfer is complete the call is removed from the transfer hash table and added to the peer hash table resuming normal operations. Addition functions were created to handle storing, removing, and comparing items in the iax_transfercallno_pvt table. 

(issue #13468)
Review: http://reviewboard.digium.com/r/140/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@173250 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-03 23:41:28 +00:00
Tilghman Lesher
0d05f4cd3b Merged revisions 173104 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r173104 | tilghman | 2009-02-02 18:24:52 -0600 (Mon, 02 Feb 2009) | 12 lines
  
  Merged revisions 173070 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r173070 | tilghman | 2009-02-02 18:15:59 -0600 (Mon, 02 Feb 2009) | 5 lines
    
    Add warning to standard config, that globals may be overridden by other
    dialplan configuration files.
    (closes issue #14388)
     Reported by: macli
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@173111 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-03 00:26:08 +00:00
Terry Wilson
6aa5e4f230 Merged revisions 173067 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r173067 | twilson | 2009-02-02 17:57:25 -0600 (Mon, 02 Feb 2009) | 9 lines
  
  Merged revisions 173066 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r173066 | twilson | 2009-02-02 17:48:06 -0600 (Mon, 02 Feb 2009) | 2 lines
    
    Fix a feature inheritance bug I added after code review
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@173068 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-02 23:59:46 +00:00
Leif Madsen
b7734870fc Merged revisions 172894 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r172894 | lmadsen | 2009-02-02 13:13:40 -0500 (Mon, 02 Feb 2009) | 7 lines
  
  Update the res_ldap.conf file with a better working example.
  
  (closes issue #13861)
  Reported by: scramatte
  Patches:
        __20080110-res_ldap.conf-2.patch uploaded by blitzrage (license 10)
  Tested by: jcovert
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@172896 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-02 18:15:51 +00:00
Steve Murphy
20c1047c23 Blocked revisions 172890 via svnmerge
........
  r172890 | murf | 2009-02-02 10:37:15 -0700 (Mon, 02 Feb 2009) | 41 lines
  
  This change allows the disconnect feature (as in "one-touch" in features.c)
  to be used within the dial app, before a call is bridged.
  
  Many thanks to sobomax for submitting this patch. 
  
  Quoting from bug 11582:
  
    "So the goal of the patch was to use the user configured feature code during the 
     call setup phase. The original ast_feature_interpret() function is not well suited 
     for this purpose as it uses much call bridge specific data and doesn't separate a 
     detection of feature from a feature handler call. So a new function ast_feature_detect() 
     has been extracted off the ast_feature_interpret() function but keeping the original 
     logic intact except some insignificant changes to locking.
  
    "Having created the ast_feature_detect() function the possibility to use feature detection 
     in almost any place of the asterisk code. So a call to this function has been added to 
     wait_for_answer() function of app_dial.so module. This code doesn't call the feature handler 
     however and uses old call leg disconnect logic to make the changes as small and simple as 
     possible to prevent unexpected problems. A disconnect feature currently is the only one 
     supported during call setup as other features as call parking and call transfer don't make much 
     sense during call setup. However if need in some of the features would arise it is much easier to 
     implement as the infrastructure changes are already in place with this patch."
  
  I have cleaned up the patch somewhat, and verified that the existing functionality is not
  harmed, and that the new functionality works. Terry has committed his stuff, and there were
  no conflicts (see 14274).
  
  (closes issue #11583)
  Reported by: sobomax
  Patches:
        patch-apps__app_dial.c uploaded by sobomax (license 359)
        patch-include__asterisk__features.h uploaded by sobomax (license 359)
        patch-res__res_features.c uploaded by sobomax (license 359)
        enable-features-during-call-setup.diff uploaded by sobomax (license 359)
        11583.newdiff uploaded by murf (license 17)
        enable-features-during-call-setup-1.diff uploaded by sobomax (license 359)
        11583.latest-patch uploaded by murf (license 17)
  Tested by: sobomax, murf
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@172892 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-02 17:40:02 +00:00
Tilghman Lesher
66d4b791c8 Merged revisions 172741 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r172741 | tilghman | 2009-01-31 20:44:23 -0600 (Sat, 31 Jan 2009) | 4 lines
  
  Blank argument crashes Asterisk
  (closes issue #14377)
   Reported by: amorsen
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@172742 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-01 02:45:05 +00:00
Tilghman Lesher
54068a791e Merged revisions 172706 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r172706 | tilghman | 2009-01-31 10:40:59 -0600 (Sat, 31 Jan 2009) | 7 lines
  
  Don't increment the loop, now that incrementing is taken care of by the
  decoder function.
  (closes issue #14363)
   Reported by: andrew53
   Patches: 
         func_strings_filter.patch uploaded by andrew53 (license 519)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@172707 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-31 16:41:48 +00:00
Terry Wilson
fc3b4491f6 Merged revisions 172581 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r172581 | twilson | 2009-01-30 15:50:03 -0600 (Fri, 30 Jan 2009) | 2 lines
  
  Remove incorret line from sample config
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@172637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-31 00:06:27 +00:00
Terry Wilson
af2b34cb56 Merged revisions 172580 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r172580 | twilson | 2009-01-30 15:29:12 -0600 (Fri, 30 Jan 2009) | 44 lines
  
  Merged revisions 172517 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) | 37 lines
    
    Fix feature inheritance with builtin features
    
    When using builtin features like parking and transfers, the AST_FEATURE_* flags
    would not be set correctly for all instances when either performing a builtin
    attended transfer, or parking a call and getting the timeout callback.  Also,
    there was no way on a per-call basis to specify what features someone should
    have on picking up a parked call (since that doesn't involve the Dial() command).
    There was a global option for setting whether or not all users who pickup a
    parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT,
    AUTOMON, or PARKCALL.
    
    This patch:
    1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the
    dialplan or with setvar in channels that support it.  This variable can be set
    to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the
    equivalent dial options), to set what features should be activated on this
    channel.  The patch moves the setting of the features datastores into the
    bridging code instead of app_dial to help facilitate this.
    
    2) adds global options parkedcallparking, parkedcallhangup, and
    parkedcallrecording to be similar to the parkedcalltransfers option for
    globally setting features.
    
    3) has builtin_atxfer call builtin_parkcall if being transfered to the parking
    extension since tracking everything through multiple masquerades, etc. is
    difficult and error-prone
    
    4) attempts to fix all cases of return calls from parking and completed builtin
    transfers not having the correct permissions
    (closes issue #14274)
    Reported by: aragon
    Patches: 
          fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396)
    Tested by: aragon, otherwiseguy
    
    Review http://reviewboard.digium.com/r/138/
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@172635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-30 23:58:31 +00:00
Mark Michelson
07adec4209 Merged revisions 172598 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r172598 | mmichelson | 2009-01-30 16:22:04 -0600 (Fri, 30 Jan 2009) | 3 lines

Fix redefinition of flag in channel.h


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@172604 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-30 22:23:11 +00:00
Tilghman Lesher
407d3d8861 Merged revisions 172441 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r172441 | tilghman | 2009-01-29 17:15:40 -0600 (Thu, 29 Jan 2009) | 16 lines
  
  Merged revisions 172438 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r172438 | tilghman | 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009) | 9 lines
    
    Lose the CAP_NET_ADMIN at every fork, instead of at startup.  Otherwise, if
    Asterisk runs as a non-root user and the administrator does a 'restart now',
    Asterisk loses the ability to set QOS on packets.
    (closes issue #14004)
     Reported by: nemo
     Patches: 
           20090105__bug14004.diff.txt uploaded by Corydon76 (license 14)
     Tested by: Corydon76
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@172503 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-29 23:47:00 +00:00
Richard Mudgett
12782036b0 Merged revisions 172400 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r172400 | rmudgett | 2009-01-29 14:38:34 -0600 (Thu, 29 Jan 2009) | 12 lines
  
  channels/chan_dahdi.c
  *  Added doxygen comments to the major dahdi structures.
  *  Fixed PRI and SS7 using an incorrect string value if the extension
  delimiter is not present in the Dial() function.
  *  Fixed SS7 not checking if the dialed extension is at least as long
  as the stripmsd option.
  *  Fixed PRI not handling unknown TON/NPI prefix letters correctly.
  *  Fixed some uninitialized string variables on FXS ports.
  
  configs/chan_dahdi.conf.sample
  *  Updated some documentation.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@172434 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-29 21:35:18 +00:00
Tilghman Lesher
4b4f6c1497 Merged revisions 172315 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r172315 | tilghman | 2009-01-29 10:48:25 -0600 (Thu, 29 Jan 2009) | 2 lines
  
  Better document mode=multirow, based upon a conversation with Jared.
........


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2009-01-29 16:49:27 +00:00
Leif Madsen
148390c562 Merged revisions 172271 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r172271 | lmadsen | 2009-01-29 08:47:27 -0500 (Thu, 29 Jan 2009) | 5 lines
  
  The realtime_pgsql.sql script is missing a couple of fields.
  closes issue #14339)
  Reported by: fiddur
  Patches:
        realtime_pgsql.sql.diff uploaded by fiddur (license 678)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@172273 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-29 13:51:48 +00:00
Olle Johansson
3873563d82 Merged revisions 172173 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r172173 | oej | 2009-01-29 10:18:01 +0100 (Tor, 29 Jan 2009) | 24 lines

Merged revisions 172169 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r172169 | oej | 2009-01-29 09:48:18 +0100 (Tor, 29 Jan 2009) | 16 lines

Make sure that we always add the hangupcause headers. In some cases, the owner was disconnected before we checked for the cause.
This patch implements a temporary storage in the pvt and use that instead.

The code is based on ideas from code from Adomjan in issue #13385 (Add support for Reason: header)
Thanks to Klaus Darillion for testing!

(closes issue #14294)
related to issue #13385

Reported by: klaus3000 and adomjan
Patches: 
      bug14294b.diff uploaded by oej (license 306)
      Based on 20080829_chan_sip.c-q850reason_header.patch uploaded by adomjan (license 487)
Tested by: oej, klaus3000


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2009-01-29 09:56:53 +00:00
Steve Murphy
491c4a9c68 Merged revisions 172063 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r172063 | murf | 2009-01-28 13:31:06 -0700 (Wed, 28 Jan 2009) | 52 lines
  
  Merged revisions 172030 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r172030 | murf | 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) | 46 lines
    
    This patch fixes h-exten running misbehavior in manager-redirected 
    situations.
    
    What it does:
    1. A new Flag value is defined in include/asterisk/channel.h,
     AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the
     bridge hangup exten code not to run the h-exten there (nor
     publish the bridge cdr there). It will done at the pbx-loop
     level instead.
    2. In the manager Redirect code, I set this flag on the channel
     if the channel has a non-null pbx pointer. I did the same for the
     second (chan2) channel, which gets run if name2 is set...
     and the first succeeds.
    3. I restored the ending of the cdr for the pbx loop h-exten
     running code. Don't know why it was removed in the first place.
    4. The first attempt at the fix for this bug was to place code
       directly in the async_goto routine, which was called from a
       large number of places, and could affect a large number of
       cases, so I tested that fix against a fair number of transfer
       scenarios, both with and without the patch. In the process,
       I saw that putting the fix in async_goto seemed not to affect
       any of the blind or attended scenarios, but still, I was
       was highly concerned that some other scenarios I had not tested
       might be negatively impacted, so I refined the patch to 
       its current scope, and jmls tested both. In the process, tho,
       I saw that blind xfers in one situation, when the one-touch
       blind-xfer feature is used by the peer, we got strange 
       h-exten behavior.  So, I  inserted code to swap CDRs and
       to set the HANGUP_DONT field, to get uniform behavior.
    5. I added code to the bridge to obey the HANGUP_DONT flag,
       skipping both publishing the bridge CDR, and running
       the h-exten; they will be done at the pbx-loop (higher)
       level instead.
    6. I removed all the debug logs from the patch before committing.
    7. I moved the AUTOLOOP set/reset in the h-exten code in res_features
       so it's only done if the h-exten is going to be run. A very
       minor performance improvement, but technically correct.
    
    
    (closes issue #14241)
    Reported by: jmls
    Patches:
          14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by murf (license 17)
    Tested by: murf, jmls
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@172065 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-28 20:41:45 +00:00
Tilghman Lesher
86a664056f Merged revisions 171964 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r171964 | tilghman | 2009-01-28 11:27:40 -0600 (Wed, 28 Jan 2009) | 9 lines
  
  Merged revisions 171963 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r171963 | tilghman | 2009-01-28 11:25:18 -0600 (Wed, 28 Jan 2009) | 2 lines
    
    Clarify log message (suggested by manxpower on #asterisk-dev)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@171965 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-28 17:28:37 +00:00
Olle Johansson
d5948d6eda Merged revisions 171838 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r171838 | oej | 2009-01-28 14:11:44 +0100 (Ons, 28 Jan 2009) | 10 lines

Merged revisions 171837 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r171837 | oej | 2009-01-28 14:07:27 +0100 (Ons, 28 Jan 2009) | 2 lines

Add a better explanation of the difference between the device namespace and the dialplan for newbies.

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2009-01-28 13:18:54 +00:00
David Vossel
2204a467be Blocked revisions 171757 via svnmerge
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  r171757 | dvossel | 2009-01-27 16:43:36 -0600 (Tue, 27 Jan 2009) | 7 lines
  
  Adding AES_ENCRYPT and AES_DECRYPT dialplan functions.  
  
  (closes issue #14301)
  Reported by: amorsen
  
  review: http://reviewboard.digium.com/r/128/
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2009-01-27 22:48:48 +00:00
Mark Michelson
ac8fa4c2c3 Merged revisions 171691 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r171691 | mmichelson | 2009-01-27 15:58:39 -0600 (Tue, 27 Jan 2009) | 47 lines

Merged revisions 171689 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r171689 | mmichelson | 2009-01-27 15:55:08 -0600 (Tue, 27 Jan 2009) | 39 lines

Fix devicestate problems for "always-on" agent channels

A revision to chan_agent attempted to "inherit" the device
state of the underlying channel in order to report the
device state of an agent channel more accurately.

The problem with the logic here is that it makes no sense to
use this for always-on agents. If the agent is logged in, then
to the underlying channel, the agent will always appear to be
"in use," no matter if the agent is on a call or not. The reason
is that to the underlying channel, the channel is currently in use
on a call to the AgentLogin application.

The most common cause that I found for this issue to occur was for
a SIP channel to be the underlying channel type for an Agent channel.
If the SIP phone re-registers, then the registration will cause the
device state core to query the device state of the SIP channel. Since the
SIP channel is in use, the Agent channel would also inherit this status.
Once the agent channel was set to "in use" there was no way that the device
state could change on that channel unless the agent logged out.

The solution for this problem is a bit different in 1.4 than it is in the
other branches. In 1.4, there will be a one-line fix to make sure that only
callback agents will inherit device state from their underlying channel type.
For the other branches of Asterisk, since callback support has been removed, there
is also no need for device state inheritance in chan_agent, so I will simply be
removing it from the code.

In addition, the 1.4 source is getting a new comment to help the next person who
edits chan_agent.c. I'm adding a comment that a agent_pvt's loginchan field may be
used to determine if the agent is a callback agent or not.

(closes issue #14173)
Reported by: nathan
Patches:
      14173.patch uploaded by putnopvut (license 60)
Tested by: nathan, aramirez


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2009-01-27 22:00:44 +00:00
Mark Michelson
0a9595289a Merged revisions 171622 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r171622 | mmichelson | 2009-01-27 14:11:30 -0600 (Tue, 27 Jan 2009) | 26 lines

Merged revisions 171621 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r171621 | mmichelson | 2009-01-27 14:06:01 -0600 (Tue, 27 Jan 2009) | 18 lines

Prevent a crash from occurring when a jitter buffer interpolated frame is
removed from a slinfactory

slinfactory used the "samples" field of an ast_frame in order to determine
the amount of data contained within the frame. In certain cases, such as
jitter buffer interpolated frames, the frame would have a non-zero value for
"samples" but have NULL "data"

This caused a problem when a memcpy call in ast_slinfactory_read would attempt
to access invalid memory. The solution in use here is to never feed frames into
the slinfactory if they have NULL "data"

(closes issue #13116)
Reported by: aragon
Patches:
      13116.diff uploaded by putnopvut (license 60)


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2009-01-27 20:13:17 +00:00
Mark Michelson
344dfe6f84 Merged revisions 171618 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r171618 | mmichelson | 2009-01-27 13:30:54 -0600 (Tue, 27 Jan 2009) | 24 lines

Fix queue crashes that would occur after the calling channel was masqueraded.

The data passed to the end_bridge_callback was assumed to be data which was
still stack'd. The problem was that with some call features, attended transfers
in particular, a new bridge thread is started once the feature completes, meaning
that when the end_bridge_callback is called, the end_bridge_callback_data was
invalid.

To fix this problem, there are two measures taken

1. Instead of pointing to stacked data, we now used heap-allocated data for
passing to the end_bridge_callback in app_queue
2. Since bridges can end multiple times on a single logical call, we wait until
the final bridge is broken to actually set any queue variables. This is accomplished
through reference-counting and the use of an end_bridge_callback_data_fixup function
in app_queue.c

(closes issue #14260)
Reported by: ccesario
Patches:
      14260.patch uploaded by putnopvut (license 60)
Tested by: ccesario


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2009-01-27 19:38:27 +00:00
Matthew Fredrickson
d970de93a6 Revert some changes that shouldn't have made it in
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@171595 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-27 16:15:40 +00:00
Matthew Fredrickson
38c977d6d0 Make sure we do not go into alarm on PTMP links with non persistent D-channels
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@171594 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-27 16:14:08 +00:00
Olle Johansson
a43c0e9ad7 Merged revisions 171528 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r171528 | oej | 2009-01-27 16:00:19 +0100 (Tis, 27 Jan 2009) | 23 lines

Solving the same issue, but a bit different in trunk...

Merged revisions 171527 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r171527 | oej | 2009-01-27 15:33:20 +0100 (Tis, 27 Jan 2009) | 13 lines

Use the same branch tag in CANCEL as in INVITE

Originally putnopvut implemented some changes in revision 142079 that according to the bug report seemed to have worked then, but somehow fails now.
I guess code, as humans, get old and forget stuff. Anyway, this bug caused CANCEL not to work with picky systems. 

Thanks Fredrik for pointing out where the bug in the SIP messaging was.

(closes issue #14346)
Reported by: oej
Patches: 
      bug14346.diff uploaded by oej (license 306)
Tested by: oej

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2009-01-27 15:13:10 +00:00
Olle Johansson
d23360562f Merged revisions 171326 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r171326 | oej | 2009-01-26 14:44:40 +0100 (MÃ¥n, 26 Jan 2009) | 17 lines

Merged revisions 171264 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r171264 | oej | 2009-01-26 13:51:53 +0100 (MÃ¥n, 26 Jan 2009) | 9 lines

Don't retransmit 401 on REGISTER requests when alwaysauthreject=yes

(closes issue #14284)
Reported by: klaus3000
Patches: 
      patch_chan_sip_unreliable_1.4.23_14284.txt uploaded by klaus3000 (license 65)
Tested by: klaus3000


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2009-01-26 14:02:52 +00:00
Tilghman Lesher
ecbccc1bf4 Merged revisions 171188 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r171188 | tilghman | 2009-01-25 17:58:00 -0600 (Sun, 25 Jan 2009) | 13 lines
  
  Merged revisions 171187 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r171187 | tilghman | 2009-01-25 17:44:01 -0600 (Sun, 25 Jan 2009) | 6 lines
    
    Correctly track the hookstate
    (closes issue #13686)
     Reported by: itiliti
     Patches: 
           20081013__bug13686.diff.txt uploaded by Corydon76 (license 14)
  ........
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2009-01-26 00:03:37 +00:00
Michiel van Baak
86566a7ff7 Blocked revisions 171081 via svnmerge
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  r171081 | mvanbaak | 2009-01-25 17:50:53 +0100 (Sun, 25 Jan 2009) | 2 lines
  
  dont segfault when a MWI event occurs on a line without a registered device
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2009-01-25 16:53:23 +00:00
Michiel van Baak
b5160951d2 Blocked revisions 171043 via svnmerge
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  r171043 | mvanbaak | 2009-01-25 15:35:17 +0100 (Sun, 25 Jan 2009) | 7 lines
  
  Make the sample skinny.conf work
  
  (closes issue #14325)
  Reported by: DEA
  Patches:
        skinny.conf.sample-trunk.txt uploaded by DEA (license 3)
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2009-01-25 14:37:07 +00:00
Sean Bright
ecaaf9ca05 Merged revisions 170980 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r170980 | seanbright | 2009-01-25 08:35:48 -0500 (Sun, 25 Jan 2009) | 16 lines
  
  Merged revisions 170979 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r170979 | seanbright | 2009-01-25 08:33:20 -0500 (Sun, 25 Jan 2009) | 9 lines
    
    Resolve a logic error that was causing Page() to crash when more than one
    channel was specified.
    
    (closes issue #14308)
    Reported by: bluefox
    Patches:
          20090124__bug14308.diff.txt uploaded by seanbright (license 71)
    Tested by: kc0bvu
  ........
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2009-01-25 13:38:38 +00:00
Russell Bryant
d6ff97e30d Merged revisions 170943 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r170943 | russell | 2009-01-24 20:49:30 -0600 (Sat, 24 Jan 2009) | 6 lines

Change ARRAY_LEN() to be more C++ safe.

When the second part of this macro is written as 0[a] instead of a[0], it will
force a failure if the macro is used on a C++ object that overloads the []
operator.

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2009-01-25 02:50:24 +00:00
Russell Bryant
05b0390b1a Blocked revisions 170902 via svnmerge
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r170902 | russell | 2009-01-24 13:33:15 -0600 (Sat, 24 Jan 2009) | 2 lines

Add a todo to finish the XML docs in this module

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2009-01-24 19:42:11 +00:00
Tilghman Lesher
f53607a8b4 Merged revisions 170837 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r170837 | tilghman | 2009-01-24 07:55:53 -0600 (Sat, 24 Jan 2009) | 9 lines
  
  Merged revisions 170836 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r170836 | tilghman | 2009-01-24 07:55:02 -0600 (Sat, 24 Jan 2009) | 2 lines
    
    Remove superfluous implementation note (closes issue #14319)
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2009-01-24 13:56:42 +00:00
Richard Mudgett
4348840980 Merged revisions 170794 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r170794 | rmudgett | 2009-01-23 17:10:34 -0600 (Fri, 23 Jan 2009) | 1 line
  
  Fix asterisk.pdf generation if branch name has an underscore in it.
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2009-01-23 23:52:12 +00:00
Russell Bryant
1809456ab1 Merged revisions 170790 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r170790 | russell | 2009-01-23 16:58:37 -0600 (Fri, 23 Jan 2009) | 2 lines

Don't blow up if a branch name has an underscore in it

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@170791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-23 22:59:03 +00:00
Mark Michelson
dab84b73bf Merged revisions 170720 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r170720 | mmichelson | 2009-01-23 14:56:07 -0600 (Fri, 23 Jan 2009) | 16 lines

Merged revisions 170719 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r170719 | mmichelson | 2009-01-23 14:55:26 -0600 (Fri, 23 Jan 2009) | 8 lines

Add notes to the idlecheck explanation in res_odbc.conf.sample

(closes issue #14319)
Reported by: klaus3000
Patches:
      patch_idlecheck_res_odbc.conf.sample.txt uploaded by klaus3000 (license 65)


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2009-01-23 20:56:32 +00:00
Mark Michelson
29a89da943 Merged revisions 170677 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r170677 | mmichelson | 2009-01-23 14:23:00 -0600 (Fri, 23 Jan 2009) | 22 lines

Merged revisions 170671 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r170671 | mmichelson | 2009-01-23 14:21:51 -0600 (Fri, 23 Jan 2009) | 14 lines

Update contrib/i18n.testsuite.conf to not use deprecated syntax

* Convert Wait,1 to Wait(1)
* Convert SetLanguage to Set(CHANNEL(language))
* Use 'n' for all priorities beyond the first

Also added test for Chinese numbers, too.

(closes issue #14320)
Reported by: dant
Patches:
      i18n.testsuite.conf.issue14320.v2.diff uploaded by dant (license 670)


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2009-01-23 20:24:18 +00:00
Joshua Colp
d59c77dc20 Merged revisions 170652 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r170652 | file | 2009-01-23 16:18:05 -0400 (Fri, 23 Jan 2009) | 11 lines
  
  Merged revisions 170648 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r170648 | file | 2009-01-23 16:16:39 -0400 (Fri, 23 Jan 2009) | 4 lines
    
    When a channel is answered make sure any indications currently playing stop. Usually the phone would do this but if the channel was already answered then they are being generated by Asterisk and we darn well need to stop them.
    (closes issue #14249)
    Reported by: RadicAlish
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@170659 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-23 20:19:37 +00:00
Tilghman Lesher
0698a4ee31 Merged revisions 170608 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r170608 | tilghman | 2009-01-23 13:25:10 -0600 (Fri, 23 Jan 2009) | 9 lines
  
  Merged revisions 170588 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r170588 | tilghman | 2009-01-23 13:20:44 -0600 (Fri, 23 Jan 2009) | 2 lines
    
    Additions to AST-2009-001
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2009-01-23 19:33:10 +00:00
Joshua Colp
d00bc0014d Merged revisions 170569 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r170569 | file | 2009-01-23 15:09:18 -0400 (Fri, 23 Jan 2009) | 11 lines
  
  Merged revisions 170568 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r170568 | file | 2009-01-23 15:06:54 -0400 (Fri, 23 Jan 2009) | 4 lines
    
    When a call is forwarded stop any active indications. The new channel will provide an indication, if need be, itself.
    (closes issue #14310)
    Reported by: RadicAlish
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@170570 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-23 19:10:00 +00:00
Joshua Colp
9382891b4e Merged revisions 170505 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r170505 | file | 2009-01-23 14:09:45 -0400 (Fri, 23 Jan 2009) | 11 lines
  
  Merged revisions 170504 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r170504 | file | 2009-01-23 14:04:08 -0400 (Fri, 23 Jan 2009) | 4 lines
    
    Use the on hold flag to see if the call is on hold or not. It is possible that our address for them will still be valid even though they are on hold.
    (closes issue #14295)
    Reported by: klaus3000
  ........
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2009-01-23 18:10:41 +00:00
Michiel van Baak
a9bff352f9 Merged revisions 170501 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r170501 | mvanbaak | 2009-01-23 18:46:02 +0100 (Fri, 23 Jan 2009) | 1 line
  
  let's use SENTINEL^Wa correct sentinel where needed
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2009-01-23 17:52:04 +00:00
Joshua Colp
a8294dfa4f Blocked revisions 170498 via svnmerge
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  r170498 | file | 2009-01-23 13:32:26 -0400 (Fri, 23 Jan 2009) | 4 lines
  
  Reset the ast_str used for escape substitution. We need to do this since it is a thread local variable that may contain the value of a previous substitution.
  (closes issue #14312)
  Reported by: pj
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2009-01-23 17:33:02 +00:00
Michiel van Baak
10f14fc12c Blocked revisions 170460 via svnmerge
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  r170460 | mvanbaak | 2009-01-23 17:57:07 +0100 (Fri, 23 Jan 2009) | 8 lines
  
  Dont clear the display of skinny phones when not needed.
  
  (closes issue #13182)
  Reported by: pj
  Patches:
        2009011901_dontcleardisplay.diff.txt uploaded by mvanbaak (license 7)
  Tested by: mvanbaak, pj
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2009-01-23 16:58:35 +00:00
Mark Michelson
d57e63c0f1 Merged revisions 170393 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r170393 | mmichelson | 2009-01-23 09:44:27 -0600 (Fri, 23 Jan 2009) | 36 lines

Merged revisions 170392 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r170392 | mmichelson | 2009-01-23 09:40:39 -0600 (Fri, 23 Jan 2009) | 28 lines

Fix broken call pickup

There was a subtle change in ast_do_masquerade which
resulted in failed attempts to pickup calls. The problem
was that the value of the AST_FLAG_OUTGOING flag was
copied from the clone to the original channel. In the case
of call pickup, this meant that the AST_FLAG_OUTGOING flag
ended up being cleared on the channel that was attempting
to execute the pickup.

Because this flag was not set, when ast_read came across
an answer frame, it ignored it. The result of this was that
the calling channel was never properly answered.

This fix changes the behavior in ast_do_masquerade to set
the flags on the original channel to the union of the flags
on the clone channel. This way, if the AST_FLAG_OUTGOING
flag is set on either of the two channels involved in the
masquerade, the resulting channel will have the flag set
as well.

(closes issue #14206)
Reported by: francesco_r
Patches:
      14206.patch uploaded by putnopvut (license 60)
Tested by: francesco_r, aragon, putnopvut


........

................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@170394 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-23 15:49:50 +00:00
Joshua Colp
ebdd169c19 Merged revisions 170240 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r170240 | file | 2009-01-22 16:04:39 -0400 (Thu, 22 Jan 2009) | 14 lines
  
  Merged revisions 170239 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r170239 | file | 2009-01-22 16:02:35 -0400 (Thu, 22 Jan 2009) | 7 lines
    
    Don't crash if RTCP is not enabled on an RTP structure but statistics are output.
    (closes issue #14234)
    Reported by: jcovert
    Patches:
          rtp.c.patch-1.6.0.3 uploaded by jcovert (license 551)
          rtp.c.patch-svn-165599 uploaded by jcovert (license 551)
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@170241 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-22 20:05:24 +00:00
Tilghman Lesher
7455e58b3f Merged revisions 170165 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r170165 | tilghman | 2009-01-22 11:19:28 -0600 (Thu, 22 Jan 2009) | 13 lines
  
  Merged revisions 170158 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r170158 | tilghman | 2009-01-22 11:18:07 -0600 (Thu, 22 Jan 2009) | 6 lines
    
    Allow global variables after substitution to be as long as other variables.
    (closes issue #14263)
     Reported by: markd
     Patches: 
           20090120__bug14263.diff.txt uploaded by Corydon76 (license 14)
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@170172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-22 17:20:45 +00:00
Joshua Colp
cb0c483da0 Merged revisions 170148 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r170148 | file | 2009-01-22 12:52:21 -0400 (Thu, 22 Jan 2009) | 11 lines
  
  Merged revisions 170147 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r170147 | file | 2009-01-22 12:50:54 -0400 (Thu, 22 Jan 2009) | 4 lines
    
    If we are unable to request a DAHDI pseudo channel and we are using the user introduction without review option make sure it gets unset so other code does not blindly assume a DAHDI pseudo channel exists.
    (closes issue #14282)
    Reported by: cheesegrits
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@170149 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-22 16:53:12 +00:00
Joshua Colp
1f843fc79d Merged revisions 170051 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r170051 | file | 2009-01-22 11:14:50 -0400 (Thu, 22 Jan 2009) | 13 lines
  
  Merged revisions 170050 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r170050 | file | 2009-01-22 11:13:56 -0400 (Thu, 22 Jan 2009) | 6 lines
    
    Do a string comparison instead of pointer comparison since some people specify the context they are actually in as an argument to get around some funkiness.
    (closes issue #14011)
    Reported by: dveiga
    Patches:
          pbx.c.patch uploaded by dveiga (license 665)
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@170052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-22 15:15:35 +00:00
Joshua Colp
7766df456f Merged revisions 170047 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r170047 | file | 2009-01-22 11:01:54 -0400 (Thu, 22 Jan 2009) | 4 lines
  
  Clear the autoloop flag when parsing and setting the context/extension/priority to go back to. When the channel executes a PBX again we want it to start out at the point we explicitly say and at that point it will not yet be doing autoloop.
  (closes issue #14304)
  Reported by: jcovert
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@170048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-22 15:03:56 +00:00
Tilghman Lesher
4deaa3d67f Merged revisions 169944 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r169944 | tilghman | 2009-01-21 18:44:52 -0600 (Wed, 21 Jan 2009) | 16 lines
  
  Merged revisions 169943 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r169943 | tilghman | 2009-01-21 18:43:31 -0600 (Wed, 21 Jan 2009) | 9 lines
    
    AST_RWLOCK_INIT_VALUE is always defined.  What we really wanted to ask is
    whether autoconf detected a static initializer value.  This fixes rwlocks
    on all such platforms (mainly, Mac OS X).
    (closes issue #13767)
     Reported by: jcovert
     Patches: 
           20090121__bug13767.diff.txt uploaded by Corydon76 (license 14)
     Tested by: jcovert, Corydon76
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@169945 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-22 00:45:38 +00:00
Joshua Colp
dbf61acd1f Merged revisions 169869 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r169869 | file | 2009-01-21 19:25:27 -0400 (Wed, 21 Jan 2009) | 11 lines
  
  Merged revisions 169867 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r169867 | file | 2009-01-21 19:20:47 -0400 (Wed, 21 Jan 2009) | 4 lines
    
    Read lock the contexts to maintain the locking order when we are notified that the state of a device has changed.
    (closes issue #13839)
    Reported by: mcallist
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@169870 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-21 23:27:19 +00:00
Michiel van Baak
9bcc0cbc44 Merged revisions 169793 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r169793 | mvanbaak | 2009-01-21 23:04:16 +0100 (Wed, 21 Jan 2009) | 2 lines
  
  remove duplicated sentence.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@169819 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-21 22:21:16 +00:00
Mark Michelson
06b1a253d6 Merged revisions 169794 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
r169794 | mmichelson | 2009-01-21 16:10:02 -0600 (Wed, 21 Jan 2009) | 17 lines

Fix a crash when saying certain numbers in Chinese

This commit fixes a crash that was occurring when attempting to
say a number between 10000 and 100000 due to dividing by 0.

This also removes some places where a "zero" is spoken when it
should not be.


(closes issue #14291)
Reported by: dant
Patches:
      say.c-14291.diff uploaded by dant (license 670)
Tested by: dant



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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@169795 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-21 22:11:11 +00:00
Tilghman Lesher
69e39ce8e9 Merged revisions 169723 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r169723 | tilghman | 2009-01-21 15:03:40 -0600 (Wed, 21 Jan 2009) | 15 lines
  
  Merged revisions 169722 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r169722 | tilghman | 2009-01-21 15:02:32 -0600 (Wed, 21 Jan 2009) | 8 lines
    
    Extra NULLs in the output cause some terminal types to abort in the middle of
    a color code, causing terminal weirdness.
    (closes issue #14130)
     Reported by: coolmig
     Patches: 
           20090121__bug14130.diff.txt uploaded by Corydon76 (license 14)
     Tested by: Corydon76, coolmig
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@169724 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-21 21:04:24 +00:00
Steve Murphy
11675088b5 Blocked revisions 169673 via svnmerge
........
  r169673 | murf | 2009-01-21 10:21:40 -0700 (Wed, 21 Jan 2009) | 14 lines
  
  This patch corrects a segfault reported in 14289,
  due to a null ptr being refd.
  
  Yes, seanbright is right in the bug comments, that is the fix.
  Sorry for this oversight; I guess my personal
  usage didn't have this happen!
  
  murf
  
  (closes issue #14289)
  Reported by: jamesgolovich
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@169675 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-21 17:43:54 +00:00
Russell Bryant
d213783bd5 Blocked revisions 169620 via svnmerge
........
r169620 | russell | 2009-01-21 04:26:07 -0600 (Wed, 21 Jan 2009) | 10 lines

Fix a regression in TCP support.

This patch fixes a problem that caused chan_sip to think that every open TCP session
was to a remote address of 0.0.0.0:0.

(closes issue #14287)
Reported by: jamesgolovich
Patches:
      bug-14287.diff.txt uploaded by jamesgolovich (license 176)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@169621 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-21 10:26:58 +00:00
Mark Michelson
827aab8b29 Merged revisions 169611 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
r169611 | mmichelson | 2009-01-20 18:33:32 -0600 (Tue, 20 Jan 2009) | 22 lines

Fix device state parsing issues for channel names with multiple slashes

The fix being applied is a bit different for trunk and the 1.6.X branches.
For trunk, we only wish to strip off the characters beyond the second slash
if the channel is a Local channel (i.e. we are removing the /n from the device
name). Other channel technologies with multiple slashes (e.g. DAHDI) need the
information after the second slash in order to get the proper device state
information.

In addition to this fix, the 1.6.X branches are receiving a much more important
fix as well. The problem in 1.6.X is that the member's device name was being directly
changed instead of having a copy changed. This meant that we would strip off the
second slash and trailing characters and then leave the member's device name like
that permanently thereafter.

(closes issue #14014)
Reported by: kebl0155
Patches:
      14014_number2.patch uploaded by putnopvut (license 60)
Tested by: kebl0155


........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@169612 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-21 00:35:05 +00:00
Terry Wilson
7cdc49f51d We don't support ~expiry in 1.6.0 register statements
This must have inadvertantly got merged in sometime as the code doesn't handle
this option, and configs/sip.conf.sample doesn't mention it as available. So
just remove it from the WARNING message


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@169558 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-20 20:13:18 +00:00
Terry Wilson
cc34381691 Merged revisions 169510 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r169510 | twilson | 2009-01-20 13:22:24 -0600 (Tue, 20 Jan 2009) | 7 lines
  
  Make a proper builtin attended transfer to parking work
  
  This is an ugly hack from 1.4 that allows the timeout callback from a parked
  call to use the right channel name for the callback when the park is done with
  a builtin attended transfer (that isn't completed early).  This hasn't ever
  worked in trunk and no one has complained yet, so eh.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@169541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-20 19:29:24 +00:00
Terry Wilson
5c9043b90b Merged revisions 169486 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r169486 | twilson | 2009-01-20 12:48:14 -0600 (Tue, 20 Jan 2009) | 13 lines
  
  Merged revisions 169485 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r169485 | twilson | 2009-01-20 12:40:56 -0600 (Tue, 20 Jan 2009) | 6 lines
    
    Don't play audio to the channel if we've masqueraded
    
    (closes issue #14066)
    Reported by: bluefox
    Tested by: otherwiseguy, bluefox
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@169487 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-20 18:58:34 +00:00
Tilghman Lesher
0a108b6c1c Merged revisions 169365 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r169365 | tilghman | 2009-01-19 14:05:52 -0600 (Mon, 19 Jan 2009) | 11 lines
  
  Merged revisions 169364 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r169364 | tilghman | 2009-01-19 13:49:25 -0600 (Mon, 19 Jan 2009) | 4 lines
    
    Truncate userevents at the end of a line, when the command exceeds the buffer.
    (closes issue #14278)
     Reported by: fnordian
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@169366 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-19 20:08:06 +00:00
Mark Michelson
a6789c917a Merged revisions 169211 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
r169211 | mmichelson | 2009-01-19 09:54:06 -0600 (Mon, 19 Jan 2009) | 21 lines

Merged revisions 169210 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r169210 | mmichelson | 2009-01-19 09:52:15 -0600 (Mon, 19 Jan 2009) | 13 lines

Prevent a crash in chan_local due to a potential NULL pointer dereference

Move the check for if both channels on a local_pvt have generators to below
where p->chan is checked for NULLity (NULLness?). This prevents a crash from
occurring if p->chan is NULL.

(closes issue #14189)
Reported by: sascha
Patches:
      14189.patch uploaded by putnopvut (license 60)
Tested by: sascha


........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@169212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-19 15:55:00 +00:00
Terry Wilson
5fef15d7e7 Blocked revisions 169080 via svnmerge
........
  r169080 | twilson | 2009-01-16 19:56:36 -0600 (Fri, 16 Jan 2009) | 8 lines
  
  Fix qualify for TCP peer
  
  (closes issue #14192)
  Reported by: pabelanger
  Patches: 
        asterisk-bug14192.diff.txt uploaded by jamesgolovich (license 176)
  Tested by: jamesgolovich
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@169081 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-17 01:57:40 +00:00
Terry Wilson
82e5da8bc4 Merged revisions 169044 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r169044 | twilson | 2009-01-16 18:03:39 -0600 (Fri, 16 Jan 2009) | 8 lines
  
  Fix port :0 added to SIP INVITE URI when outboundproxy used
  
  (closes issue #14233)
  Reported by: chris-mac
  Patches: 
        asterisk-bug14233.diff.txt uploaded by jamesgolovich (license 176)
  Tested by: jamesgolovich, chris-mac, otherwiseguy
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@169078 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-17 00:29:43 +00:00
Terry Wilson
d996191010 Merged revisions 168941 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r168941 | twilson | 2009-01-16 16:16:23 -0600 (Fri, 16 Jan 2009) | 19 lines
  
  Merged revisions 168716 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r168716 | twilson | 2009-01-15 12:22:49 -0600 (Thu, 15 Jan 2009) | 12 lines
    
    Convert call to park_call_full to masq_park_call_announce
    
    Since we removed the AST_PBX_KEEPALIVE return value, we need to use masqueraded
    parking, otherwise we will try to call ast_hangup() in __pbx_run() and in
    do_parking_thread() and then promptly crash.
    (closes issue #14215)
    	Reported by: waverly360	
    	Tested by: otherwiseguy
    (closes issue #14228)
    	Reported by: kobaz
    	Tested by: otherwiseguy
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@168980 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-16 22:52:03 +00:00
Mark Michelson
63c3ccd9b0 Merged revisions 168976 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
r168976 | mmichelson | 2009-01-16 16:43:09 -0600 (Fri, 16 Jan 2009) | 26 lines

Merged revisions 168975 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r168975 | mmichelson | 2009-01-16 16:42:13 -0600 (Fri, 16 Jan 2009) | 18 lines

Account for possible NULL pointer when we receive a 408 in response to a REGISTER

It may be that by the time we receive a reply to a REGISTER request, the attempt has
timed out and thus the registry structure pointed to by the corresponding sip_pvt has
gone away. This situation was handled properly for a 200 OK response, but the 408
case assumed that the sip_registry struct was non-NULL, thus potentially causing a crash

This commit fixes this assumption and prints out a message to the console if we should
receive a late 408 response to a REGISTER


(closes issue #14211)
Reported by: aborghi
Patches:
      14211.diff uploaded by putnopvut (license 60)
Tested by: aborghi


........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@168978 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-16 22:46:11 +00:00
Tilghman Lesher
b669fac978 Merged revisions 168832 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r168832 | tilghman | 2009-01-16 12:49:09 -0600 (Fri, 16 Jan 2009) | 13 lines
  
  Merged revisions 168828 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r168828 | tilghman | 2009-01-16 12:41:35 -0600 (Fri, 16 Jan 2009) | 6 lines
    
    Fix the conjugation of Russian and Ukrainian languages.
    (related to issue #12475)
     Reported by: chappell
     Patches: 
           vm_multilang.patch uploaded by chappell (license 8)
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@168835 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-16 18:53:48 +00:00
Steve Murphy
166ff7e96b Merged revisions 168746 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r168746 | murf | 2009-01-15 17:34:31 -0700 (Thu, 15 Jan 2009) | 20 lines
  
  Merged revisions 168745 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r168745 | murf | 2009-01-15 17:19:12 -0700 (Thu, 15 Jan 2009) | 14 lines
    
    This patch fixes a problem where a goto (or jump, in this case)
    fails a consistency check because it can't find a matching 
    extension. The problem was a missing instruction to end
    the range notation in the code where it converts the pattern
    into a regex and uses the regex code to determine the match.
    
    I tested using the AEL code the user supplied, and now,
    the consistency check passes.
    
    
    (closes issue #14141)
    Reported by: dimas
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@168747 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-16 00:38:36 +00:00
Steve Murphy
5c80480c83 Merged revisions 168737 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r168737 | murf | 2009-01-15 13:54:59 -0700 (Thu, 15 Jan 2009) | 16 lines
  
  This patch allows null args in ast_expr2 func calls, and fixes commas being converted to pipes, which was 1.4 type stuff.
  
  If the user says count=ENUMLOOKUP(${EXTEN},ALL,c,,enum.mydomain.tld);
  
  then it won't complain about the empty arg (c,,...) and fabled's patch
  won't let it swap the commas for pipes.
  
  Ran it thru my dialplan and no complaints.
  
  
  (closes issue #14169)
  Reported by: fabled
  Patches:
        function-argument-separator-fix.diff uploaded by fabled (license 448)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@168738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-15 21:14:31 +00:00
Kevin P. Fleming
e192b455f8 Blocked revisions 168734 via svnmerge
........
  r168734 | kpfleming | 2009-01-15 14:18:53 -0600 (Thu, 15 Jan 2009) | 5 lines
  
  remove the PBX_ODBC logic from the configure script, and add GENERIC_ODCB logic that includes copying the relevant LIB and INCLUDE data from either UnixODBC or iODBC, based on which was found; if both were found, prefer UnixODBC
  
  this stops modules from being linked against both sets of libraries on systems that have both installed
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@168736 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-15 20:20:45 +00:00
Olle Johansson
2414c56148 Merged revisions 168722 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r168722 | oej | 2009-01-15 19:47:14 +0100 (Tor, 15 Jan 2009) | 10 lines

Merged revisions 168721 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r168721 | oej | 2009-01-15 19:43:43 +0100 (Tor, 15 Jan 2009) | 2 lines

Meetme actually has realtime but wasn't documented

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2009-01-15 18:52:54 +00:00
Olle Johansson
51dc13cfb2 Merged revisions 168711 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r168711 | oej | 2009-01-15 18:55:53 +0100 (Tor, 15 Jan 2009) | 4 lines

Clarify some misunderstandings and make it even more clear that you can refer to a peer
in the register= line.


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@168717 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-15 18:23:14 +00:00
Olle Johansson
8633f3884f Merged revisions 168712 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r168712 | oej | 2009-01-15 19:08:59 +0100 (Tor, 15 Jan 2009) | 3 lines

Make sure that we have the same terminology in sip.conf.sample and the source code warning.
Thanks Nick Lewis for pointing this out in the bug tracker.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@168713 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-15 18:14:20 +00:00
Sean Bright
bd1a0134d6 Merged revisions 168705 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r168705 | seanbright | 2009-01-15 10:33:18 -0500 (Thu, 15 Jan 2009) | 11 lines
  
  Add a missing unlock and properly handle the 'maxusers' setting on MeetMe
  conferences.  We were using the 'user number' field to compare against the
  maximum allowed users, which works assuming users with lower user numbers
  didn't leave the conference.
  
  (closes issue #14117)
  Reported by: sergedevorop
  Patches:
        20090114__bug14117-2.diff.txt uploaded by seanbright (license 71)
  Tested by: sergedevorop
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2009-01-15 15:34:35 +00:00
Mark Michelson
b40ebef036 Merged revisions 168629 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r168629 | mmichelson | 2009-01-14 18:14:17 -0600 (Wed, 14 Jan 2009) | 24 lines

Merged revisions 168628 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r168628 | mmichelson | 2009-01-14 18:11:01 -0600 (Wed, 14 Jan 2009) | 16 lines

Fix some crashes from bad datastore handling in app_queue.c

* The queue_transfer_fixup function was searching for and removing
  the datastore from the incorrect channel, so this was fixed.

* Most datastore operations regarding the queue_transfer datastore
  were being done without the channel locked, so proper channel locking
  was added, too.

(closes issue #14086)
Reported by: ZX81
Patches:
      14086v2.patch uploaded by putnopvut (license 60)
Tested by: ZX81, festr


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2009-01-15 00:14:48 +00:00
Richard Mudgett
d572518549 Merged revisions 168623 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r168623 | rmudgett | 2009-01-14 15:51:06 -0600 (Wed, 14 Jan 2009) | 11 lines
  
  Merged revisions 168622 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r168622 | rmudgett | 2009-01-14 15:48:22 -0600 (Wed, 14 Jan 2009) | 4 lines
    
    *  Fixed create_process() allocation of process ID values.
    The allocated process IDs could overflow their respective
    NT and TE fields.  Affects outgoing calls.
  ........
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2009-01-14 21:54:31 +00:00
Steve Murphy
240cd8d0be Merged revisions 168613 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r168613 | murf | 2009-01-14 13:51:26 -0700 (Wed, 14 Jan 2009) | 9 lines
  
  Merged revisions 168608 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r168608 | murf | 2009-01-14 12:34:35 -0700 (Wed, 14 Jan 2009) | 1 line
    
    app_page was failing to compile in dev-mode on my gcc-4.2.4 system. This change gets rid of the warning.
  ........
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2009-01-14 21:21:58 +00:00
Sean Bright
eb079909bc Merged revisions 168615 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r168615 | seanbright | 2009-01-14 15:58:26 -0500 (Wed, 14 Jan 2009) | 16 lines
  
  Merged revisions 168614 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r168614 | seanbright | 2009-01-14 15:52:00 -0500 (Wed, 14 Jan 2009) | 9 lines
    
    Update autosupport script to supply info for both Zaptel and DAHDI in 1.4 and
    be sure to run dahdi_test in 1.6.x and trunk instead of zttest.
    
    (closes issue #14132)
    Reported by: dsedivec
    Patches:
          asterisk-1.4-autosupport.patch uploaded by dsedivec (license 638)
          asterisk-trunk-autosupport.patch uploaded by dsedivec (license 638)
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@168616 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-14 20:59:33 +00:00
Richard Mudgett
5d29635bc5 Fix merge error caused by merging -r132883 and -r121770 from
https://origsvn.digium.com/svn/asterisk/trunk out of order.

(closes issue #13788)
Reported by: IgorG

(closes issue #13491)
Reported by: avalentin


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@168607 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-14 19:20:29 +00:00
Tilghman Lesher
ca54c11b79 Merged revisions 168604 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r168604 | tilghman | 2009-01-14 13:11:14 -0600 (Wed, 14 Jan 2009) | 14 lines
  
  Merged revisions 168603 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r168603 | tilghman | 2009-01-14 13:02:55 -0600 (Wed, 14 Jan 2009) | 7 lines
    
    Don't read into a buffer without first checking if a value is beyond the end.
    (closes issue #13600)
     Reported by: atis
     Patches: 
           20090106__bug13600.diff.txt uploaded by Corydon76 (license 14)
     Tested by: atis
  ........
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2009-01-14 19:11:56 +00:00
Terry Wilson
cee2699a5d Merged revisions 168594 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r168594 | twilson | 2009-01-13 20:00:40 -0600 (Tue, 13 Jan 2009) | 27 lines
  
  Merged revisions 168593 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r168593 | twilson | 2009-01-13 19:27:18 -0600 (Tue, 13 Jan 2009) | 20 lines
    
    Don't overflow when paging more than 128 extensions
    
    The number of available slots for calls in app_page was hardcoded to 128.
    Proper bounds checking was not in place to enforce this limit, so if more than
    128 extensions were passed to the Page() app, Asterisk would crash.  This patch
    instead dynamically allocates memory for the ast_dial structures and removes
    the (non-functional) arbitrary limit.
    
    This issue would have special importance to anyone who is dynamically creating
    the argument passed to the Page application and allowing more than 128
    extensions to be added by an outside user via some external interface.
    
    The patch posted by a_villacis was slightly modified for some coding guidelines
    and other cleanups.  Thanks, a_villacis!
    (closes issue #14217)
    Reported by: a_villacis
    Patches: 
          20080912-asterisk-app_page-fix-buffer-overflow.patch uploaded by a (license 660)
    Tested by: otherwiseguy
  ........
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2009-01-14 02:06:19 +00:00
Terry Wilson
a33a19b699 Merged revisions 168588 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r168588 | twilson | 2009-01-13 17:05:43 -0600 (Tue, 13 Jan 2009) | 5 lines
  
  Fully overwrite a same-named file when uploading
  
  (closes issue #14190)
  Reported by: timking
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@168589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-13 23:11:19 +00:00
Terry Wilson
5c805d776b Blocked revisions 168585 via svnmerge
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  r168585 | twilson | 2009-01-13 17:00:27 -0600 (Tue, 13 Jan 2009) | 8 lines
  
  Add option to hide console connect messages
  
  (closes issue #14222)
  Reported by: jamesgolovich
  Patches: 
        asterisk-hideconnect.diff.txt uploaded by jamesgolovich (license 176)
  Tested by: otherwiseguy
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@168586 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-13 23:01:25 +00:00
Terry Wilson
00647c1f8f Merged revisions 168578 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r168578 | twilson | 2009-01-13 16:22:34 -0600 (Tue, 13 Jan 2009) | 14 lines
  
  Merged revisions 168551 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r168551 | twilson | 2009-01-13 12:34:14 -0600 (Tue, 13 Jan 2009) | 7 lines
    
    Don't pass a value with a side effect to a macro
    
    (closes issue #14176)
    Reported by: paraeco
    Patches: 
          chan_sip.c.diff uploaded by paraeco (license 658)
  ........
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2009-01-13 22:31:20 +00:00
Russell Bryant
1b1c2db6bd Merged revisions 168562 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r168562 | russell | 2009-01-13 13:22:13 -0600 (Tue, 13 Jan 2009) | 10 lines

Merged revisions 168561 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r168561 | russell | 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009) | 2 lines

Revert unnecessary indications API change from rev 122314

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2009-01-13 19:27:54 +00:00
Tilghman Lesher
931ee5d3cb Merged revisions 168547 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r168547 | tilghman | 2009-01-13 11:51:12 -0600 (Tue, 13 Jan 2009) | 13 lines
  
  Merged revisions 168546 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r168546 | tilghman | 2009-01-13 11:48:00 -0600 (Tue, 13 Jan 2009) | 6 lines
    
    If either conditional is NULL, don't try copying it.
    (closes issue #14226)
     Reported by: caspy
     Patches: 
           20090113__bug14226.diff.txt uploaded by Corydon76 (license 14)
  ........
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2009-01-13 17:51:58 +00:00
Dwayne M. Hubbard
2a361e4b33 Blocked revisions 168539 via svnmerge
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r168539 | dhubbard | 2009-01-13 10:02:13 -0600 (Tue, 13 Jan 2009) | 1 line

correct a CLI description
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@168540 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-13 16:07:48 +00:00
Tilghman Lesher
a4fd6ae437 Merged revisions 168526 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r168526 | tilghman | 2009-01-12 17:45:51 -0600 (Mon, 12 Jan 2009) | 12 lines
  
  Merged revisions 167095 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r167095 | tilghman | 2008-12-31 18:01:22 -0600 (Wed, 31 Dec 2008) | 5 lines
    
    Repeat attempts to write when we receive -EAGAIN from the driver, as detailed
    in the ALSA sample code (see http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html#a32)
    Reported by: Jerry Geis (via the -users list)
    Fixed by: me (license 14)
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@168527 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-12 23:46:55 +00:00
Jeff Peeler
d1e7ac2285 Merged revisions 168517 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r168517 | jpeeler | 2009-01-12 15:51:46 -0600 (Mon, 12 Jan 2009) | 12 lines
  
  Merged revisions 168516 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r168516 | jpeeler | 2009-01-12 15:42:34 -0600 (Mon, 12 Jan 2009) | 5 lines
    
    (closes issue #13881)
    Reported by: hoowa
    
    Update the app CDR field for AGI commands that are not executing an application via "exec".
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@168518 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-12 21:52:32 +00:00
Jeff Peeler
e8b5385c7e Merged revisions 168508 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r168508 | jpeeler | 2009-01-12 14:53:04 -0600 (Mon, 12 Jan 2009) | 15 lines
  
  Merged revisions 168507 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r168507 | jpeeler | 2009-01-12 14:26:22 -0600 (Mon, 12 Jan 2009) | 9 lines
    
    (closes issue #12269)
    Reported by: IgorG
    Tested by: denisgalvao
    
    This gits rid of the notion of an owning_app allowing the request and hangup to be initiated by different threads. Originating from an active agent channel requires this. The implementation primarily changes __login_exec to wait on a condition variable rather than a lock.
    
    Review: http://reviewboard.digium.com/r/35/
  ........
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2009-01-12 21:01:55 +00:00
Olle Johansson
eeac5f539e Merged revisions 168497 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r168497 | oej | 2009-01-12 17:31:27 +0100 (MÃ¥n, 12 Jan 2009) | 2 lines

Better to use the proper app name in the STATUS variable

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2009-01-12 16:55:11 +00:00
Mark Michelson
03a951f368 Merged revisions 168485 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r168485 | mmichelson | 2009-01-12 09:00:00 -0600 (Mon, 12 Jan 2009) | 13 lines

Merged revisions 168482 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r168482 | mmichelson | 2009-01-12 08:58:25 -0600 (Mon, 12 Jan 2009) | 5 lines

I am reverting the fix made in revision 168128 (and its upward merges)
after being contacted by Olle Johansson and being shown how this fix is
incorrect. Thanks to Olle for clearing this up for me.


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2009-01-12 15:00:31 +00:00
Russell Bryant
6f07826706 Merged revisions 168481 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r168481 | russell | 2009-01-12 08:57:49 -0600 (Mon, 12 Jan 2009) | 10 lines

Merged revisions 168480 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r168480 | russell | 2009-01-12 08:57:27 -0600 (Mon, 12 Jan 2009) | 2 lines

s/ringdance/ringcadence/ for Bulgaria

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2009-01-12 14:58:38 +00:00
Tilghman Lesher
8f41e52f44 Merged revisions 168334 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r168334 | tilghman | 2009-01-09 19:42:45 -0600 (Fri, 09 Jan 2009) | 2 lines
  
  sizeof for a stringfield is 4.  Kinda low for reconstructing a field value.
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2009-01-10 01:43:51 +00:00
Kevin P. Fleming
7f0d370d0d Merged revisions 168270 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r168270 | kpfleming | 2009-01-09 17:16:08 -0600 (Fri, 09 Jan 2009) | 9 lines
  
  Merged revisions 168267 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r168267 | kpfleming | 2009-01-09 17:12:29 -0600 (Fri, 09 Jan 2009) | 1 line
    
    update to use new sound file packages that include license files
  ........
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2009-01-09 23:17:14 +00:00
Richard Mudgett
6806bf4a69 Missed change from trunk -r168192 when resolving conflicts.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@168268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-09 23:14:57 +00:00
Richard Mudgett
16c8142b29 Merged revisions 168192 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r168192 | rmudgett | 2009-01-09 15:43:30 -0600 (Fri, 09 Jan 2009) | 10 lines
  
  Merged revisions 168191 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r168191 | rmudgett | 2009-01-09 15:28:42 -0600 (Fri, 09 Jan 2009) | 3 lines
    
    *  Fix for JIRA AST-175/ABE-1757
    *  Miscellaneous doxygen comments added.
  ........
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2009-01-09 22:34:54 +00:00
Russell Bryant
7cd060bb84 Merged revisions 168200 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r168200 | russell | 2009-01-09 16:21:05 -0600 (Fri, 09 Jan 2009) | 10 lines

Merged revisions 168198 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r168198 | russell | 2009-01-09 16:14:38 -0600 (Fri, 09 Jan 2009) | 2 lines

Make this compile for mvanbaak

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2009-01-09 22:22:29 +00:00
Mark Michelson
271c6c9529 Merged revisions 168193 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r168193 | mmichelson | 2009-01-09 15:53:26 -0600 (Fri, 09 Jan 2009) | 21 lines

Merged revisions 168128 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r168128 | mmichelson | 2009-01-09 14:08:04 -0600 (Fri, 09 Jan 2009) | 13 lines

Add check_via calls to more request handlers

INFO, NOTIFY, OPTIONS, REFER, and MESSAGE requests
were not checking the topmost Via to determine where
to send the response. Adding check_via calls to those
request handlers solves this.

(closes issue #13071)
Reported by: baron
Patches:
      check_via.patch uploaded by baron (license 531)
Tested by: baron

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2009-01-09 21:57:01 +00:00
Terry Wilson
3fc6826248 Merged revisions 168142 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r168142 | twilson | 2009-01-09 14:25:25 -0600 (Fri, 09 Jan 2009) | 7 lines
  
  Don't leak memory if phoneprov.conf does not exist
  
  (closes issue #14203)
  Reported by: jamesgolovich
  Patches: 
        asterisk-phoneprovleak.diff.txt uploaded by jamesgolovich (license 176)
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2009-01-09 20:28:16 +00:00
Tilghman Lesher
76c697c90b Merged revisions 168090 via svnmerge from
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  r168090 | tilghman | 2009-01-09 12:30:55 -0600 (Fri, 09 Jan 2009) | 3 lines
  
  When using ast_str with a non-ast_str-enabled API, we need to update the buffer
  or otherwise, we cannot use ast_str_strlen().
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2009-01-09 18:36:03 +00:00
Matthew Nicholson
05c6d88ab2 Merged revisions 168014 via svnmerge from
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  r168014 | mnicholson | 2009-01-09 10:32:34 -0600 (Fri, 09 Jan 2009) | 5 lines
  
  Use ast_safe_system() in logger.c instead of system()
  
  (closes issue #14194)
  Reported by: pabelanger
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2009-01-09 16:41:36 +00:00
Terry Wilson
da5ecd2598 Blocked revisions 167973 via svnmerge
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  r167973 | twilson | 2009-01-08 19:15:43 -0600 (Thu, 08 Jan 2009) | 2 lines
  
  Set ORIGINATE_STATUS instead of OUTGOING_STATUS to match the documentation
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2009-01-09 01:16:28 +00:00
Terry Wilson
7970bef2d2 Merged revisions 167935 via svnmerge from
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  r167935 | twilson | 2009-01-08 18:13:12 -0600 (Thu, 08 Jan 2009) | 2 lines
  
  Set peer context and exten values so MACRO_EXTEN and MACRO_CONTEXT will be set
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2009-01-09 00:37:01 +00:00
Tilghman Lesher
31d37012e0 Merged revisions 167894 via svnmerge from
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  r167894 | tilghman | 2009-01-08 16:37:20 -0600 (Thu, 08 Jan 2009) | 13 lines
  
  Merged revisions 167840 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
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    r167840 | tilghman | 2009-01-08 16:08:56 -0600 (Thu, 08 Jan 2009) | 6 lines
    
    Don't truncate database results at 255 chars.
    (closes issue #14069)
     Reported by: evandro
     Patches: 
           20081214__bug14069.diff.txt uploaded by Corydon76 (license 14)
  ........
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2009-01-08 22:42:23 +00:00
Kevin P. Fleming
cad23358ab Merged revisions 167720 via svnmerge from
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  r167720 | kpfleming | 2009-01-08 11:26:03 -0600 (Thu, 08 Jan 2009) | 9 lines
  
  Merged revisions 167714 via svnmerge from 
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    r167714 | kpfleming | 2009-01-08 11:24:21 -0600 (Thu, 08 Jan 2009) | 1 line
    
    remove an unnecessary argument to queue_request()
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2009-01-08 17:29:36 +00:00
Kevin P. Fleming
85ac907af2 Merged revisions 167700 via svnmerge from
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  r167700 | kpfleming | 2009-01-08 10:43:26 -0600 (Thu, 08 Jan 2009) | 12 lines
  
  Merged revisions 167620 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r167620 | kpfleming | 2009-01-07 17:32:21 -0600 (Wed, 07 Jan 2009) | 5 lines
    
    When a SIP request or response arrives for a dialog with an associated Asterisk channel, and the lock on that channel cannot be obtained because it is held by another thread, instead of dropping the request/response, queue it for later processing when the channel lock becomes available.
    
    http://reviewboard.digium.com/r/123/
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2009-01-08 17:22:40 +00:00
Leif Madsen
6e2db4dc27 Merged revisions 167662 via svnmerge from
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  r167662 | lmadsen | 2009-01-08 09:27:53 -0500 (Thu, 08 Jan 2009) | 1 line
  
  Oops... fix the fieldname I changed yesterday to be right.
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2009-01-08 14:31:37 +00:00
Russell Bryant
685eda5a54 Merged revisions 167569 via svnmerge from
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r167569 | russell | 2009-01-07 16:36:34 -0600 (Wed, 07 Jan 2009) | 10 lines

Merged revisions 167566 via svnmerge from 
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r167566 | russell | 2009-01-07 16:35:36 -0600 (Wed, 07 Jan 2009) | 2 lines

Fix the last couple of places where free() was improperly used directly.

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2009-01-07 22:37:14 +00:00
Russell Bryant
6232e6d659 Merged revisions 167555 via svnmerge from
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r167555 | russell | 2009-01-07 16:27:23 -0600 (Wed, 07 Jan 2009) | 10 lines

Merged revisions 167554 via svnmerge from 
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r167554 | russell | 2009-01-07 16:26:42 -0600 (Wed, 07 Jan 2009) | 2 lines

Don't fclose() the file early, the filestream destructor will handle it.

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Russell Bryant
bc8cd9f02c Merged revisions 167546 via svnmerge from
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r167546 | russell | 2009-01-07 16:20:31 -0600 (Wed, 07 Jan 2009) | 10 lines

Merged revisions 167545 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r167545 | russell | 2009-01-07 16:19:47 -0600 (Wed, 07 Jan 2009) | 2 lines

Only try to close the file if one was actually opened

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2009-01-07 22:21:02 +00:00
Russell Bryant
8c2a49c221 Merged revisions 167542 via svnmerge from
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r167542 | russell | 2009-01-07 16:05:29 -0600 (Wed, 07 Jan 2009) | 12 lines

Merged revisions 167541 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r167541 | russell | 2009-01-07 16:03:59 -0600 (Wed, 07 Jan 2009) | 4 lines

Don't use free() directly.  This caused a crash since ast_filestream is now an ao2 object.

Reported by JunK-Y on IRC, #asterisk-dev

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2009-01-07 22:06:18 +00:00
BJ Weschke
976f09e010 Merged revisions 167478 via svnmerge from
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  r167478 | bweschke | 2009-01-07 13:20:31 -0500 (Wed, 07 Jan 2009) | 7 lines
  
   Answer the channel if it has not already been answered and we've already found a valid profile for followme. 
   (closes issue #14140)
   Reported by: dimas
   Patches:
         14140.patch uploaded by dimas
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2009-01-07 18:30:39 +00:00
Leif Madsen
eb0179bd13 Merged revisions 167477 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r167477 | lmadsen | 2009-01-07 13:18:45 -0500 (Wed, 07 Jan 2009) | 8 lines
  
  Update queues.conf.sample documentation.
  
  Update the queues.conf.sample documentation to mention that you need to preload chan_local.so as well if you plan on using Local channels for queue members, and you're preloading pbx_config.so.
  
  
  (closes issue #14179)
  Reported by: CrashHD
  Tested by: CrashHD
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2009-01-07 18:25:40 +00:00
Russell Bryant
6899204595 Merged revisions 167442 via svnmerge from
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r167442 | russell | 2009-01-07 11:35:39 -0600 (Wed, 07 Jan 2009) | 12 lines

Merged revisions 167432 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r167432 | russell | 2009-01-07 11:29:53 -0600 (Wed, 07 Jan 2009) | 4 lines

Treat an empty string the same way as a NULL country argument.

In passing, simplify the handling of returning a default tone zone.

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2009-01-07 17:42:27 +00:00
Leif Madsen
4d763741d4 Merged revisions 167373 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r167373 | lmadsen | 2009-01-07 09:26:19 -0500 (Wed, 07 Jan 2009) | 1 line
  
  Update the sip-friends.sql file to use the non-deprecated 'defaultname' instead of 'username' and remove an extra comma that would cause the script to fail as-is
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2009-01-07 14:33:38 +00:00
Mark Michelson
b8b4ae6501 Merged revisions 167301 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r167301 | mmichelson | 2009-01-06 15:36:44 -0600 (Tue, 06 Jan 2009) | 16 lines

Merged revisions 167299 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r167299 | mmichelson | 2009-01-06 15:35:57 -0600 (Tue, 06 Jan 2009) | 8 lines

Use the correct variable when creating the format string

(closes issue #14177)
Reported by: nic_bellamy
Patches:
      asterisk-trunk-svn-r167242-ast_db_gettree.patch uploaded by nic (license 299)


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2009-01-06 21:38:19 +00:00
Tilghman Lesher
4e803082b3 Merged revisions 167265 via svnmerge from
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  r167265 | tilghman | 2009-01-06 15:02:33 -0600 (Tue, 06 Jan 2009) | 16 lines
  
  Merged revisions 167260 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ................
    r167260 | tilghman | 2009-01-06 14:48:05 -0600 (Tue, 06 Jan 2009) | 9 lines
    
    Merged revisions 167259 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.2
    
    ........
      r167259 | tilghman | 2009-01-06 14:44:03 -0600 (Tue, 06 Jan 2009) | 2 lines
      
      Security fix AST-2009-001.
    ........
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2009-01-06 21:07:20 +00:00
Mark Michelson
bfbf5afa62 Merged revisions 167180 via svnmerge from
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r167180 | mmichelson | 2009-01-05 10:59:36 -0600 (Mon, 05 Jan 2009) | 49 lines

Merged revisions 167179 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r167179 | mmichelson | 2009-01-05 10:51:59 -0600 (Mon, 05 Jan 2009) | 41 lines

A couple of changes to T.38 SDP attribute handling

There are some boolean attributes for T.38 such
as T38FaxFillBitRemoval, T38FaxTranscodingMMR, and
T38FaxTranscodingJBIG. By simply being present, we
should treat these as a "true" value. The current
code, however, was requiring a 1 or 0 as the value
of the attribute in order to parse it. This is due
to the fact that there are some T.38 endpoints and
gateways that also transmit this information
incorrectly. This patch follows the "be liberal in
what you accept and strict in what you send"
philosophy by accepting both the correctly- and 
incorrectly-formatted attributes, but only sending
information as it is supposed to be sent.

It was also discovered that a particular type of 
T.38 gateway sends some non-standard T.38 SDP
attributes. Instead of using T38FaxMaxDatagram
and T38MaxBitRate, it used T38MaxDatagram and
T38FaxMaxRate respectively. We now will properly
accept these attributes as well.

Note that there are a lot of patches cited in
the below commit message template. This is
because the person who submitted these patches is
an awesome person and wrote 1.4, 1.6.0, and 1.6.1
variants.

(closes issue #13976)
Reported by: linulin
Patches:
     chan_sip.c.1.4-update1.diff uploaded by arcivanov (license 648)
	 chan_sip.c.1.6.0-update1.diff uploaded by arcivanov (license 648)
	 chan_sip.c.1.6.1-update1.diff uploaded by arcivanov (license 648)
	 chan_sip.c.1.4-relaxedT38_update1.diff uploaded by arcivanov (license 648)
	 chan_sip.c.1.6.0-relaxedT38_update1.diff uploaded by arcivanov (license 648)
	 chan_sip.c.1.6.1-relaxedT38_update1.diff uploaded by arcivanov (license 648)
Tested by: arcivanov


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Tilghman Lesher
1687071eed Merged revisions 167176 via svnmerge from
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  r167176 | tilghman | 2009-01-05 10:44:47 -0600 (Mon, 05 Jan 2009) | 7 lines
  
  More clearly explain that quote marks are no longer necessary.
  (closes issue #13718)
   Reported by: davidw
   Patches: 
         20081020__bug13718.diff.txt uploaded by Corydon76 (license 14)
   Tested by: blitzrage
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2009-01-05 16:45:44 +00:00
Jeff Peeler
f37957c5b5 Blocked revisions 167125 via svnmerge
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r167125 | jpeeler | 2009-01-03 14:29:54 -0600 (Sat, 03 Jan 2009) | 3 lines

When parsing environment variable ASTERISK_PROMPT, make sure to proceed to the next character when a non format specifier is used (no %). Otherwise, the while loop looking for the null byte will never exit.


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2009-01-03 20:32:12 +00:00
Terry Wilson
228bf17ee7 Blocked revisions 167057 via svnmerge
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  r167057 | twilson | 2008-12-31 16:53:55 -0600 (Wed, 31 Dec 2008) | 2 lines
  
  Don't forget to free typename
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2008-12-31 22:55:50 +00:00
Mark Michelson
dc2b180c27 Blocked revisions 167021 via svnmerge
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r167021 | mmichelson | 2008-12-31 15:52:02 -0600 (Wed, 31 Dec 2008) | 4 lines

Change some incorrect syntax for pri set debug and correct
an off-by-one error in ss7 set debug command


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2008-12-31 21:53:40 +00:00
Tilghman Lesher
ac3a15778a Merged revisions 166954 via svnmerge from
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  r166954 | tilghman | 2008-12-31 13:34:28 -0600 (Wed, 31 Dec 2008) | 12 lines
  
  Merged revisions 166953 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
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    r166953 | tilghman | 2008-12-31 13:20:35 -0600 (Wed, 31 Dec 2008) | 5 lines
    
    Also inherit the musiconhold class.
    (Closes #14153)
    Reported by: Jerry Geis, via the users list.
    Patch by: me (license 14)
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Terry Wilson
f43bfb0ace Merged revisions 166908 via svnmerge from
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  r166908 | twilson | 2008-12-30 14:50:05 -0600 (Tue, 30 Dec 2008) | 2 lines
  
  Fix some svn:keywords
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Mark Michelson
788f3a8fa5 Merged revisions 166861 via svnmerge from
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r166861 | mmichelson | 2008-12-29 12:04:52 -0600 (Mon, 29 Dec 2008) | 14 lines

Update app_queue to deal with the removal of AST_PBX_KEEPALIVE

When placing a call to a queue which ran a gosub on the member's
channel, Asterisk would crash every time, stemming from the fact
that the member's channel was being hung up unexpectedly when the
Gosub completed. The necessary change was pretty much copied and
pasted from app_dial's similar changes made last week.

I also took the opportunity to change a LOG_DEBUG message in
app_dial to use ast_debug. I am guessing this was due to a direct
merge from 1.4 that was not corrected to use trunk's preferred
syntax.


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Eliel C. Sardanons
9c28554fb1 Blocked revisions 166823 via svnmerge
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  r166823 | eliel | 2008-12-28 10:36:25 -0500 (Sun, 28 Dec 2008) | 3 lines
  
  Fix a typo in the XML documentation of the AUDIOHOOK_INHERIT dialplan function.
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2008-12-28 15:44:09 +00:00
Russell Bryant
a80d937882 Merged revisions 166773 via svnmerge from
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r166773 | russell | 2008-12-28 09:15:14 -0600 (Sun, 28 Dec 2008) | 12 lines

Merged revisions 166772 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r166772 | russell | 2008-12-28 09:13:48 -0600 (Sun, 28 Dec 2008) | 4 lines

Use strncat() instead of an sprintf() in which source and target buffers overlap

http://lists.digium.com/pipermail/asterisk-dev/2008-December/035919.html

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Terry Wilson
ba8c827dd3 Blocked revisions 166731 via svnmerge
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  r166731 | twilson | 2008-12-24 09:10:42 -0600 (Wed, 24 Dec 2008) | 10 lines
  
  There is no section 22.2.2 in rfc 3261.  I believe 26.2.2 is what was meant:
  
        Note that in the SIPS URI scheme, transport is independent of TLS,
        and thus "sips:alice@atlanta.com;transport=tcp" and
        "sips:alice@atlanta.com;transport=sctp" are both valid (although
        note that UDP is not a valid transport for SIPS).  The use of
        "transport=tls" has consequently been deprecated, partly because
        it was specific to a single hop of the request.  This is a change
        since RFC 2543.
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Steve Murphy
5d34e0df03 Merged revisions 166665 via svnmerge from
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Due to non-symmetrical updating, I had some fairly
interesting conflicts to straighten out in this
release. The changes were such that I was compelled
to run thru all the same tests as trunk, which turned
up some problems, which I fixed. 

................
  r166665 | murf | 2008-12-23 11:13:49 -0700 (Tue, 23 Dec 2008) | 153 lines
  
  Merged revisions 166093 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  In order to merge this 1.4 patch into trunk,
  I had to resolve some conflicts and wait for
  Russell to make some changes to res_agi.
  I re-ran all the tests; 39 calls in all, and
  made fairly careful notes and comparisons: I
  don't want this to blow up some aspect of 
  asterisk; I completely removed the KEEPALIVE
  from the pbx.h decls. The first 3 scenarios
  involving feature park; feature xfer to 700;
  hookflash park to Park() app call all behave
  the same, don't appear to leave hung channels,
  and no crashes.
  
  ........
    r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) | 131 lines
    
    This merges the masqpark branch into 1.4
    
    These changes eliminate the need for (and use of)
    the KEEPALIVE return code in res_features.c;
    There are other places that use this result code
    for similar purposes at a higher level, these appear
    to be left alone in 1.4, but attacked in trunk.
    
    The reason these changes are being made in 1.4, is
    that parking ends a channel's life, in some situations,
    and the code in the bridge (and some other places),
    was not checking the result code properly, and dereferencing
    the channel pointer, which could lead to memory corruption
    and crashes.
    
    Calling the masq_park function eliminates this danger 
    in higher levels.
    
    A series of previous commits have replaced some parking calls
    with masq_park, but this patch puts them ALL to rest,
    (except one, purposely left alone because a masquerade
    is done anyway), and gets rid of the code that tests
    the KEEPALIVE result, and the NOHANGUP_PEER result codes.
    
    While bug 13820 inspired this work, this patch does
    not solve all the problems mentioned there.
    
    I have tested this patch (again) to make sure I have
    not introduced regressions. 
    
    Crashes that occurred when a parked party hung up
    while the parking party was listening to the numbers
    of the parking stall being assigned, is eliminated.
    
    These are the cases where parking code may be activated:
    
    1. Feature one touch (eg. *3)
    2. Feature blind xfer to parking lot (eg ##700)
    3. Run Park() app from dialplan (eg sip xfer to 700)
       (eg. dahdi hookflash xfer to 700)
    4. Run Park via manager.
    
    The interesting testing cases for parking are:
    I. A calls B, A parks B
        a. B hangs up while A is getting the numbers announced.
        b. B hangs up after A gets the announcement, but 
           before the parking time expires
        c. B waits, time expires, A is redialed,
           A answers, B and A are connected, after
           which, B hangs up.
        d. C picks up B while still in parking lot.
    
    II. A calls B, B parks A
        a. A hangs up while B is getting the numbers announced.
        b. A hangs up after B gets the announcement, but 
           before the parking time expires
        c. A waits, time expires, B is redialed,
           B answers, A and B are connected, after
           which, A hangs up.
        d. C picks up A while still in parking lot.
    
    Testing this throroughly involves acting all the permutations
    of I and II, in situations 1,2,3, and 4.
    
    Since I added a few more changes (ALL references to KEEPALIVE in the bridge
    code eliimated (I missed one earlier), I retested
    most of the above cases, and no crashes.
    
    H-extension weirdness.
    
    Current h-extension execution is not completely
    correct for several of the cases.
    
    For the case where A calls B, and A parks B, the
    'h' exten is run on A's channel as soon as the park
    is accomplished. This is expected behavior.
    
    But when A calls B, and B parks A, this will be
    current behavior:
    
    After B parks A, B is hung up by the system, and
    the 'h' (hangup) exten gets run, but the channel
    mentioned will be a derivative of A's...
    
    Thus, if A is DAHDI/1, and B is DAHDI/2,
    the h-extension will be run on channel
    Parked/DAHDI/1-1<ZOMBIE>, and the 
    start/answer/end info will be those 
    relating to Channel A.
    
    And, in the case where A is reconnected to
    B after the park time expires, when both parties
    hang up after the joyful reunion, no h-exten
    will be run at all.
    
    In the case where C picks up A from the 
    parking lot, when either A or C hang up,
    the h-exten will be run for the C channel.
    
    CDR's are a separate issue, and not addressed
    here.
    
    As to WHY this strange behavior occurs, 
    the answer lies in the procedure followed
    to accomplish handing over the channel
    to the parking manager thread. This procedure
    is called masquerading. In the process,
    a duplicate copy of the channel is created,
    and most of the active data is given to the
    new copy. The original channel gets its name
    changed to XXX<ZOMBIE> and keeps the PBX
    information for the sake of the original
    thread (preserving its role as a call 
    originator, if it had this role to begin
    with), while the new channel is without
    this info and becomes a call target (a
    "peer").
    
    In this case, the parking lot manager
    thread is handed the new (masqueraded)
    channel. It will not run an h-exten
    on the channel if it hangs up while
    in the parking lot. The h exten will
    be run on the original channel instead,
    in the original thread, after the bridge
    completes.
    
    See bug 13820 for our intentions as
    to how to clean up the h exten behavior.
  
  Review: http://reviewboard.digium.com/r/29/
  
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@166729 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-24 00:52:12 +00:00
Tilghman Lesher
6964b59097 Merged revisions 166696 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r166696 | tilghman | 2008-12-23 14:47:08 -0600 (Tue, 23 Dec 2008) | 7 lines
  
  Allow semicolons and extended characters in user-specified SIP headers.
  (closes issue #14110)
   Reported by: gork
   Patches: 
         20081222__bug14110__2.diff.txt uploaded by Corydon76 (license 14)
   Tested by: gork, putnopvut
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@166697 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-23 20:52:19 +00:00
Mark Michelson
4e5fd8376c Merged revisions 166569 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166569 | mmichelson | 2008-12-23 09:17:54 -0600 (Tue, 23 Dec 2008) | 20 lines

Merged revisions 166568 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r166568 | mmichelson | 2008-12-23 09:16:26 -0600 (Tue, 23 Dec 2008) | 12 lines

Fix a crash resulting from a datastore with inheritance but no duplicate callback

The fix for this is to simply set the newly created datastore's data pointer
to NULL if it is inherited but has no duplicate callback.

(closes issue #14113)
Reported by: francesco_r
Patches:
      14113.patch uploaded by putnopvut (license 60)
Tested by: francesco_r


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2008-12-23 15:18:43 +00:00
Tilghman Lesher
b6244cfc7b Merged revisions 166533 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r166533 | tilghman | 2008-12-22 22:32:15 -0600 (Mon, 22 Dec 2008) | 11 lines
  
  Merged revisions 166509 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r166509 | tilghman | 2008-12-22 22:05:25 -0600 (Mon, 22 Dec 2008) | 4 lines
    
    Use the integer form of condition for integer comparisons.
    (closes issue #14127)
     Reported by: andrew
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@166534 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-23 04:33:37 +00:00
Mark Michelson
cc9fa36fbc Merged revisions 166470 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166470 | mmichelson | 2008-12-22 17:25:34 -0600 (Mon, 22 Dec 2008) | 11 lines

Always use the value of the AGISIGHUP when running an AGI.

Prior to this patch, the value of AGISIGUP was not always
honored when set on a channel.

(closes issue #13711)
Reported by: fmueller
Patches:
      13711.patch uploaded by putnopvut (license 60)


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2008-12-22 23:27:03 +00:00
Mark Michelson
61bf090dbb Merged revisions 166382 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166382 | mmichelson | 2008-12-22 15:08:03 -0600 (Mon, 22 Dec 2008) | 44 lines

Merged revisions 166380 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r166380 | mmichelson | 2008-12-22 14:56:29 -0600 (Mon, 22 Dec 2008) | 36 lines

Fix a deadlock relating to channel locks and autoservice

It has been discovered that if a channel is locked prior
to a call to ast_autoservice_stop, then it is likely that
a deadlock will occur. The reason is that the call to 
ast_autoservice_stop has a check built into it to be sure
that the thread running autoservice is not currently trying
to manipulate the channel we are about to pull out of 
autoservice.

The autoservice thread, however, cannot advance beyond where
it currently is, though, because it is trying to acquire
the lock of the channel for which autoservice is attempting
to be stopped.

The gist of all this is that a channel MUST NOT be locked
when attempting to stop autoservice on the channel.

In this particular case, the channel was locked by a call
to ast_read. A call to ast_exists_extension led to autoservice
being started and stopped due to the existence of dialplan
switches.

It may be that there are future commits which handle the same
symptoms but in a different location, but based on my looks through
the code, it is very rare to see a construct such as this one.

(closes issue #14057)
Reported by: rtrauntvein
Patches:
      14057v3.patch uploaded by putnopvut (license 60)
Tested by: rtrauntvein

Review: http://reviewboard.digium.com/r/107/


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2008-12-22 21:55:59 +00:00
Russell Bryant
3bd710a626 Merged revisions 166436 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166436 | russell | 2008-12-22 15:45:28 -0600 (Mon, 22 Dec 2008) | 2 lines

Cosmetic change - don't mix struct initializer styles.

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2008-12-22 21:46:24 +00:00
Russell Bryant
d34c6ad79d Merged revisions 166377 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166377 | russell | 2008-12-22 14:26:48 -0600 (Mon, 22 Dec 2008) | 2 lines

Fix a bad typo.

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2008-12-22 20:27:59 +00:00
Russell Bryant
0b4abe2273 Merged revisions 166317 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166317 | russell | 2008-12-22 11:29:10 -0600 (Mon, 22 Dec 2008) | 10 lines

Merged revisions 166297 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r166297 | russell | 2008-12-22 11:22:56 -0600 (Mon, 22 Dec 2008) | 2 lines

Fix up timeout handling in ast_carefulwrite().

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@166319 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-22 17:30:05 +00:00
Russell Bryant
70416419ce Merged revisions 166282 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166282 | russell | 2008-12-22 11:09:36 -0600 (Mon, 22 Dec 2008) | 12 lines

Introduce ast_careful_fwrite() and use in AMI to prevent partial writes.

This patch introduces a function to do careful writes on a file stream which
will handle timeouts and partial writes.  It is currently used in AMI to
address the issue that has been reported.  However, there are probably a few
other places where this could be used.

(closes issue #13546)
Reported by: srt
Tested by: russell
http://reviewboard.digium.com/r/104/

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@166283 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-22 17:14:18 +00:00
Mark Michelson
2e8b3deab5 When merging the fix for issue #14118, I found that
the issue didn't affect 1.6.0, but in this case that's
not an especially good thing, because it means that
the fix for issue #13496 was not merged into 1.6.0 in
the first place. This commit kills two birds with one
stone by putting both fixes in the 1.6.0 branch



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@166278 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-22 16:30:25 +00:00
Russell Bryant
b19363ed78 Merged revisions 166273 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166273 | russell | 2008-12-22 10:10:40 -0600 (Mon, 22 Dec 2008) | 7 lines

Re-work ref count handling of MoH classes using astobj2 to resolve crashes.

(closes issue #13566)
Reported by: igorcarneiro
Tested by: russell
Review: http://reviewboard.digium.com/r/106/

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@166276 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-22 16:17:57 +00:00
Mark Michelson
c01f7c0af8 Back these changes out for now. The initial refcounting
for filestreams didn't get merged into 1.6.0. I'll have
to get that in first and then put this change in



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@166274 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-22 16:12:34 +00:00
Mark Michelson
3eb9bd8e4e Merged revisions 166267 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166267 | mmichelson | 2008-12-22 10:07:59 -0600 (Mon, 22 Dec 2008) | 17 lines

Fix a file playback crash and explicitly initialize values in func_timeout.c

A crash was brought up on the bugtracker. The first run through valgrind
was full of legitimate complaints of uninitialized values in func_timeout when
setting a response timeout. These were fixed but the crash persisted.

A second run through showed the real problem. The reference counting used
for filestreams was incorrect because there were some missing increments
when a frame was read from a format module.

(closes issue #14118)
Reported by: blitzrage
Patches:
      14118v2.patch uploaded by putnopvut (license 60)
Tested by: blitzrage


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@166271 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-22 16:10:14 +00:00
Joshua Colp
e6a7286d11 Blocked revisions 166268 via svnmerge
........
  r166268 | file | 2008-12-22 12:08:13 -0400 (Mon, 22 Dec 2008) | 7 lines
  
  Record the previous port in the temporary address structure so that the comparison does not treat the host as having changed even if it did not. This would have been uninitialized before and would have led to a baddddd port.
  (closes issue #13628)
  Reported by: pananix
  Patches:
        bug13628.patch uploaded by jpeeler (license 325)
  Tested by: file, blitzrage
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@166270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-22 16:09:36 +00:00
Russell Bryant
589ebf8421 Merged revisions 166258 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166258 | russell | 2008-12-22 08:16:54 -0600 (Mon, 22 Dec 2008) | 26 lines

Remove AST_PBX_KEEPALIVE usage from res_agi.

This patch removes the usage of AST_PBX_KEEPALIVE from res_agi.  The only usage
was for the AGI command, "asyncagi break".  This patch removes this feature.
Normally, a feature would not be removed like this.  However, this code is
broken and usage of it will result in a memory leak.

Usage of this feature will make the AGI code return a result of 
AST_PBX_KEEPALIVE.  The PBX handler assumes that another thread has assumed
ownership of the channel.  The channel thread will exit without destroying the
channel.  Unfortunately, _no_ thread has ownership of the channel at this
point.  There are a couple of serious problems here:

1) The only way to recover the caller is to issue a channel redirect.  This
   will work, but this will be done with a masquerade, and the old ast_channel
   structure will be lost.

2) Until the channel redirect happens, there is no code servicing the channel.
   That means nothing is reading audio or handling events coming from the
   channel.  This is very bad.

The recommended way to get this same "break" functionality is to issue the
redirect while the channel is still being handled by the AGI code.  That way,
there will be no memory leak, and there will be no period of time that the
channel is not being serviced.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@166259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-22 14:18:08 +00:00
Mark Michelson
bd050e9fae Merged revisions 166162 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166162 | mmichelson | 2008-12-19 17:45:00 -0600 (Fri, 19 Dec 2008) | 6 lines

Get rid of an extra space.

I don't know how this crept back in when I had already
fixed it earlier


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@166163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-19 23:45:26 +00:00
Mark Michelson
15ec36e3aa Adding missing file func_audiohookinherit.c
I don't know why this wasn't added to begin with. It's
here now though :)



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@166160 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-19 23:41:09 +00:00
Mark Michelson
eec3edde9f Merged revisions 166092,166095 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166092 | mmichelson | 2008-12-19 16:26:16 -0600 (Fri, 19 Dec 2008) | 28 lines

Adding a new dialplan function AUDIOHOOK_INHERIT

This function is being added as a method to allow for
an audiohook to move to a new channel during a channel
masquerade. The most obvious use for such a facility is
for MixMonitor when a transfer is performed. Prior to
the addition of this functionality, if a channel 
running MixMonitor was transferred by another party, then
the recording would stop once the transfer had completed.
By using AUDIOHOOK_INHERIT, you can make MixMonitor 
continue recording the call even after the transfer
has completed.

It has also been determined that since this is seen
by most as a bug fix and is not an invasive change,
this functionality will also be backported to 1.4 and
merged into the 1.6.0 branches, even though they are
feature-frozen.

(closes issue #13538)
Reported by: mbit
Patches:
      13538.patch uploaded by putnopvut (license 60)
	  Tested by: putnopvut

Review: http://reviewboard.digium.com/r/102/


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r166095 | mmichelson | 2008-12-19 16:40:57 -0600 (Fri, 19 Dec 2008) | 5 lines

Remove the verbatim tag from the author line

I could have sworn I already did that before, though...


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2008-12-19 23:04:07 +00:00
Eliel C. Sardanons
fccf7383df Blocked revisions 165954 via svnmerge
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  r165954 | eliel | 2008-12-19 16:20:46 -0200 (Fri, 19 Dec 2008) | 4 lines
  
  Fix the XML documentation for Record().
  <value> tags inside <variable> elements must have CDATA and no
  another XML node.
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2008-12-19 18:23:47 +00:00
Russell Bryant
21012610d2 Merged revisions 165890 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r165890 | russell | 2008-12-19 09:05:09 -0600 (Fri, 19 Dec 2008) | 17 lines

Merged revisions 165889 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r165889 | russell | 2008-12-19 09:03:02 -0600 (Fri, 19 Dec 2008) | 9 lines

Ensure that the chanspy datastore is fully initialized.

This patch resolved some random crash issues observed by a user on a BSD system

(closes issue #14111)
Reported by: ys
Patches:
      app_chanspy.c.diff uploaded by ys (license 281)

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2008-12-19 15:08:00 +00:00
Russell Bryant
202b8352ef Blocked revisions 165886 via svnmerge
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r165886 | russell | 2008-12-19 08:48:35 -0600 (Fri, 19 Dec 2008) | 2 lines

Disable some automatic links generated by doxygen.

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2008-12-19 14:48:51 +00:00
Russell Bryant
3c9aebde62 Blocked revisions 165883 via svnmerge
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r165883 | russell | 2008-12-19 08:42:51 -0600 (Fri, 19 Dec 2008) | 8 lines

Introduce commit message formatting guidelines.

This documents the recommended outline to use for commit message.  It also
covers information on special tags that can be used in commit messages, as well
as the template functionality that is available on bugs.digium.com.

Review: http://reviewboard.digium.com/r/96/

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2008-12-19 14:43:13 +00:00
Tilghman Lesher
8c22d38f3a Merged revisions 165797 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r165797 | tilghman | 2008-12-18 15:41:02 -0600 (Thu, 18 Dec 2008) | 15 lines
  
  Merged revisions 165767 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r165767 | tilghman | 2008-12-18 15:14:47 -0600 (Thu, 18 Dec 2008) | 8 lines
    
    Add mutexes around accesses to the IMAP library interface.  This prevents
    certain crashes, especially when shared mailboxes are used.
    (closes issue #13653)
     Reported by: howardwilkinson
     Patches: 
           asterisk-1.4.21.2-appvoicemail-sharedimap-lock.patch uploaded by howardwilkinson (license 590)
     Tested by: jpeeler
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2008-12-18 21:56:27 +00:00
Russell Bryant
4dcb02958e Merged revisions 165801 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r165801 | russell | 2008-12-18 15:44:47 -0600 (Thu, 18 Dec 2008) | 19 lines

Merged revisions 165796 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r165796 | russell | 2008-12-18 15:39:25 -0600 (Thu, 18 Dec 2008) | 11 lines

Make ast_carefulwrite() be more careful.

This patch handles some additional cases that could result in partial writes
to the file description.  This was done to address complaints about partial
writes on AMI.

(issue #13546) (more changes needed to address potential problems in 1.6)
Reported by: srt
Tested by: russell
Review: http://reviewboard.digium.com/r/99/

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2008-12-18 21:45:38 +00:00
Jeff Peeler
c3662a64c7 Blocked revisions 165798 via svnmerge
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r165798 | jpeeler | 2008-12-18 15:43:17 -0600 (Thu, 18 Dec 2008) | 6 lines

(closes issue #13993)
Reported by: mika

Add ActionID response to ping if sent with request.


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2008-12-18 21:44:05 +00:00
Joshua Colp
3bde2f6f89 Blocked revisions 165792 via svnmerge
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  r165792 | file | 2008-12-18 17:21:44 -0400 (Thu, 18 Dec 2008) | 6 lines
  
  Numerous documentation updates.
  (closes issue #13970)
  Reported by: pkempgen
  Patches:
        __20081217_cli_usage_fixes.patch.txt uploaded by blitzrage (license 10)
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2008-12-18 21:22:29 +00:00
Russell Bryant
ca1c37e47c Merged revisions 165723 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r165723 | russell | 2008-12-18 13:33:42 -0600 (Thu, 18 Dec 2008) | 14 lines

Remove the need for AST_PBX_KEEPALIVE with the GoSub option from Dial.

This is part of an effort to completely remove AST_PBX_KEEPALIVE and other
similar return codes from the source.  While this usage was perfectly safe,
there are others that are problematic.  Since we know ahead of time that
we do not want to PBX to destroy the channel, the PBX API has been changed
so that information can be provided as an argument, instead, thus removing
the need for the KEEPALIVE return value.

Further changes to get rid of KEEPALIVE and related code is being done by
murf.  There is a patch up for that on review 29.

Review: http://reviewboard.digium.com/r/98/

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2008-12-18 19:40:14 +00:00
Russell Bryant
9ca178a9be Merged revisions 165662 via svnmerge from
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r165662 | russell | 2008-12-18 12:54:47 -0600 (Thu, 18 Dec 2008) | 15 lines

Merged revisions 165661 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r165661 | russell | 2008-12-18 12:52:18 -0600 (Thu, 18 Dec 2008) | 7 lines

Set the process group ID on the MOH process so that all children will get killed

(closes issue #14099)
Reported by: caspy
Patches:
      res_musiconhold.c.patch.killpg.try2 uploaded by caspy (license 645)

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2008-12-18 18:56:32 +00:00
Tilghman Lesher
90d0e13f44 Merged revisions 165658 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r165658 | tilghman | 2008-12-18 12:36:48 -0600 (Thu, 18 Dec 2008) | 2 lines
  
  Fix 2 resource leaks and fix another pipe-to-comma conversion
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2008-12-18 18:46:23 +00:00
Joshua Colp
7333579bf9 Merged revisions 165599 via svnmerge from
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  r165599 | file | 2008-12-18 13:13:32 -0400 (Thu, 18 Dec 2008) | 11 lines
  
  Merged revisions 165591 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r165591 | file | 2008-12-18 13:11:42 -0400 (Thu, 18 Dec 2008) | 4 lines
    
    Only care about a compatible codec for early bridging if we are actually bridging to another channel. If we are not we actually want to bring the audio back to us.
    (closes issue #13545)
    Reported by: davidw
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Joshua Colp
4e3184c908 Blocked revisions 165538 via svnmerge
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  r165538 | file | 2008-12-18 12:11:08 -0400 (Thu, 18 Dec 2008) | 12 lines
  
  Blocked revisions 165537 via svnmerge
  
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    r165537 | file | 2008-12-18 12:07:37 -0400 (Thu, 18 Dec 2008) | 6 lines
    
    Do not crash if we are not passed in a followme id.
    (closes issue #14106)
    Reported by: ys
    Patches:
          app_followme.c.2.diff uploaded by ys (license 281)
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2008-12-18 16:11:49 +00:00
Eliel C. Sardanons
900e6b19b8 Blocked revisions 165502 via svnmerge
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  r165502 | eliel | 2008-12-18 13:25:15 -0200 (Thu, 18 Dec 2008) | 9 lines
  
  Remove duplicate code from the ast_str API. We now use __AST_STR_* to
  access 'struct ast_str' members, but this must only be used inside the API implementation.
  
  (closes issue #14098)
  Reported by: eliel
  Patches:
        ast_str.patch uploaded by eliel (license 64)
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2008-12-18 15:28:41 +00:00
Russell Bryant
dbfd0a3928 Blocked revisions 165433 via svnmerge
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r165433 | russell | 2008-12-18 07:33:34 -0600 (Thu, 18 Dec 2008) | 9 lines

Add a new application, Originate.

(closes issue #14075)
Reported by: rcasas
Patches:
      app_originate.c uploaded by rcasas (license 641), heavily modified by me
Tested by: russell
Review: http://reviewboard.digium.com/r/95/

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2008-12-18 13:34:02 +00:00
Mark Michelson
f4760ab030 Blocked revisions 165330 via svnmerge
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r165330 | mmichelson | 2008-12-17 15:46:19 -0600 (Wed, 17 Dec 2008) | 3 lines

Fix a refcount leak in res_odbc


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2008-12-17 21:48:00 +00:00
Tilghman Lesher
c4fc79c5c7 Oops, broke 1.6.0
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2008-12-17 21:29:32 +00:00
Mark Michelson
f9e08bac6d Merged revisions 165318 via svnmerge from
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r165318 | mmichelson | 2008-12-17 15:17:20 -0600 (Wed, 17 Dec 2008) | 15 lines

Merged revisions 165255 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r165255 | mmichelson | 2008-12-17 14:51:38 -0600 (Wed, 17 Dec 2008) | 7 lines

Fix some memory leaks found while looking at how realtime
configs are handled.

Also cleaned up some coding guidelines violations in app_realtime.c,
mostly related to spacing


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2008-12-17 21:22:10 +00:00
Tilghman Lesher
475abaedf1 Merged revisions 165319 via svnmerge from
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  r165319 | tilghman | 2008-12-17 15:18:57 -0600 (Wed, 17 Dec 2008) | 11 lines
  
  Merged revisions 165317 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r165317 | tilghman | 2008-12-17 15:14:37 -0600 (Wed, 17 Dec 2008) | 4 lines
    
    Reverse the fix from issue #6176 and add proper handling for that issue.
    (Closes issue #13962, closes issue #13363)
    Fixed by myself (license 14)
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2008-12-17 21:21:19 +00:00
Steve Murphy
f79f987366 Merged revisions 165254 via svnmerge from
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r165254 | murf | 2008-12-17 13:50:19 -0700 (Wed, 17 Dec 2008) | 5 lines

This patch is here committed to satisfy the
buildbot, who has a problem with the const.



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2008-12-17 20:59:42 +00:00
Terry Wilson
1665fe8b81 Merged revisions 165219 via svnmerge from
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  r165219 | twilson | 2008-12-17 13:55:10 -0600 (Wed, 17 Dec 2008) | 2 lines
  
  Polycom phones close the connection after reading a little bit of the firmware files, we should stop sending in that case. Also, make that case print out a debug statement instead of a scary WARNING.
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2008-12-17 20:00:11 +00:00
Joshua Colp
665ba5d64b Merged revisions 165216 via svnmerge from
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  r165216 | file | 2008-12-17 15:52:40 -0400 (Wed, 17 Dec 2008) | 4 lines
  
  Call proxy_update so that the IP address gets populated. Sending stuff to 0.0.0.0 is silly!
  (closes issue #14055)
  Reported by: chris-mac
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2008-12-17 19:53:42 +00:00
Mark Michelson
0ad1fc15cd Merged revisions 165142-165143 via svnmerge from
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r165142 | mmichelson | 2008-12-17 11:52:50 -0600 (Wed, 17 Dec 2008) | 10 lines

Use the create_vm_state_from_user function in a place where
it was not being used before. Also, I've moved the urgent
folder check in messagecount() up a bit so that the flow is
a bit better.

This was something I noticed while taking a look at issue
#13973, although I don't think this is the underlying cause
of the issue.


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r165143 | mmichelson | 2008-12-17 11:53:37 -0600 (Wed, 17 Dec 2008) | 3 lines

And actually assign the function to a pointer...


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2008-12-17 17:55:02 +00:00
Steve Murphy
6bddb621ce Merged revisions 165071 via svnmerge from
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r165071 | murf | 2008-12-16 22:04:56 -0700 (Tue, 16 Dec 2008) | 31 lines

A possibly "horrible fix" for a "horribly broken"
situation.

As stuff shifts around in the asterisk code, the
miscellaneous inclusions from the standalone stuff
gets broken. There's no easy fix for this situation.

I made sure that everything in utils builds without
problem ***AND*** that aelparse runs the regressions
correctly with the following make menuselect options
both on and off:

DONT_OPTIMIZE
DEBUG_THREADS
DEBUG_CHANNEL_LOCKS
MALLOC_DEBUG
MTX_PROFILE
DEBUG_SCHEDULER
DEBUG_THREADLOCALS
DETECT_DEADLOCKS
CHANNEL_TRACE

I think from now on, I'm going to #undef
all these features in the various utils native
files; I guess I could do the same for the
copied-in files, surrounded by STANDALONE ifdef.

A standalone isn't going to care about threads,
mutexes, etc.


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2008-12-17 05:35:48 +00:00
Mark Michelson
3377535a60 Merged revisions 164978 via svnmerge from
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r164978 | mmichelson | 2008-12-16 17:06:04 -0600 (Tue, 16 Dec 2008) | 15 lines

Merged revisions 164977 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r164977 | mmichelson | 2008-12-16 17:04:27 -0600 (Tue, 16 Dec 2008) | 7 lines

After looking through SIP registration code most of the day, this
is one of the few things I could find that was just plain wrong.
Even though it probably isn't possible for it to happen, it seems weird
to have code that checks if a pointer is NULL and then immediately dereferences
that pointer if it was NULL.


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2008-12-16 23:07:06 +00:00
Jeff Peeler
543315bd6b Merged revisions 164942 via svnmerge from
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r164942 | jpeeler | 2008-12-16 16:45:39 -0600 (Tue, 16 Dec 2008) | 6 lines

(closes issue #13669)
Reported by: pj

Delete file recording if recording terminated from a hangup.


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2008-12-16 22:51:55 +00:00
Russell Bryant
599dbbd2b7 Merged revisions 164882 via svnmerge from
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r164882 | russell | 2008-12-16 15:39:15 -0600 (Tue, 16 Dec 2008) | 17 lines

Merged revisions 164881 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r164881 | russell | 2008-12-16 15:38:29 -0600 (Tue, 16 Dec 2008) | 9 lines

Fix an issue where DEBUG_THREADS may erroneously report that a thread 
is exiting while holding a lock.

If the last lock attempt was a trylock, and it failed, it will still be in the
list of locks so that it can be reported.

(closes issue #13219)
Reported by: pj

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2008-12-16 21:39:45 +00:00
Russell Bryant
7e0ca3563f Merged revisions 164877 via svnmerge from
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r164877 | russell | 2008-12-16 15:12:49 -0600 (Tue, 16 Dec 2008) | 14 lines

Merged revisions 164876 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r164876 | russell | 2008-12-16 15:10:44 -0600 (Tue, 16 Dec 2008) | 6 lines

Do not dereference the channel if AST_PBX_KEEPALIVE has been returned.

This is a bug I noticed while looking at the code for app_macro.  This return code
means that another thread has assumed ownership of the channel and it can no longer
be touched.  (I hate this return code with a passion, by the way.)

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Russell Bryant
c411947f0a Blocked revisions 164821 via svnmerge
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r164821 | russell | 2008-12-16 14:49:25 -0600 (Tue, 16 Dec 2008) | 2 lines

Fix build issues on Linux after sysinfo related changes

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2008-12-16 20:49:47 +00:00
Joshua Colp
9329c0f105 Blocked revisions 164814 via svnmerge
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  r164814 | file | 2008-12-16 16:47:31 -0400 (Tue, 16 Dec 2008) | 2 lines
  
  Qualify trumps poke per lmadsen.
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2008-12-16 20:48:13 +00:00
Russell Bryant
c91f280aff Merged revisions 164807 via svnmerge from
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r164807 | russell | 2008-12-16 14:41:51 -0600 (Tue, 16 Dec 2008) | 17 lines

Merged revisions 164806 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r164806 | russell | 2008-12-16 14:35:25 -0600 (Tue, 16 Dec 2008) | 9 lines

Add "restart gracefully" to the AMI blacklist of CLI commands.  

"module unload" was already identified as a command that can not be used 
from the AMI.  "restart gracefully" effectively unloads all modules, and will 
run in to the same problems.

(closes issue #13894)
Reported by: kernelsensei

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2008-12-16 20:43:16 +00:00
Joshua Colp
38a107a673 Blocked revisions 164809 via svnmerge
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  r164809 | file | 2008-12-16 16:42:33 -0400 (Tue, 16 Dec 2008) | 4 lines
  
  Add configuration options for finer control over how Asterisk handles having to poke all peers at seemingly the same time.
  (closes issue #13217)
  Reported by: cervajs
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2008-12-16 20:43:06 +00:00
Michiel van Baak
bd8b830613 Blocked revisions 164802 via svnmerge
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  r164802 | mvanbaak | 2008-12-16 21:08:34 +0100 (Tue, 16 Dec 2008) | 10 lines
  
  introduce 'core show sysinfo' for systems that dont have the Linux-ish sysinfo stuff but do have sysctl.
  
  (closes issue #13433)
  Reported by: mvanbaak
  Patches:
        2008121300_sysinfosysctl.diff.txt uploaded by mvanbaak (license 7)
  	  with two free calls replaced with ast_free based on feedback on reviewboard
  Review:
        http://reviewboard.digium.com/r/91/
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2008-12-16 20:12:05 +00:00
Steve Murphy
cd1587ae99 Merged revisions 164801 via svnmerge from
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r164801 | murf | 2008-12-16 13:04:46 -0700 (Tue, 16 Dec 2008) | 36 lines

(closes issue #14076)
Reported by: toc
Tested by: murf

OK, Well this issue has had its share of flip-flopping.
I found the following:

1. the code in question, in ext_cmp1 in pbx.c, would not
allow two extensions that vary only by any dashes contained
within them, to be defined in the same context.

2. for input dialstrings, dashes are NOT ignored.
So, skipping them when sorting patterns seemed a bit silly.
Thus, you might declare ext 891 in a context, but
if you try dialing 8-9-1, it will NOT match 891.

So, I proposed to remove the code from ext_cmp1 to 
skip the spaces and dashes. Just kept us from 
declaring 891 and 8-9-1 in the same context,
forcing users to generate otherwise uselessly
obfuscated dialplan code to get the same effect.

Then, I tried out 1.4, and found that:

1. you can declare 891 and 8-9-1 in the
same context!

2. You can't define 891, and have 8-9-1 match
it! Nor can you define 8-9-1, and have 891
match it!

So, it appears that my proposal simply restores
the pbx to behaving as it did in 1.4.



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2008-12-16 20:08:39 +00:00
Tilghman Lesher
d1b141fb9b Merged revisions 164798 via svnmerge from
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  r164798 | tilghman | 2008-12-16 13:54:11 -0600 (Tue, 16 Dec 2008) | 4 lines
  
  Set up umask as a possible configuration option.
  (closes issue #13753)
   Reported by: irroot
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2008-12-16 19:55:38 +00:00
Russell Bryant
f83308aa28 Merged revisions 164737 via svnmerge from
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r164737 | russell | 2008-12-16 11:14:01 -0600 (Tue, 16 Dec 2008) | 22 lines

Merged revisions 164736 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r164736 | russell | 2008-12-16 11:06:29 -0600 (Tue, 16 Dec 2008) | 14 lines

Fix memory leak and invalid reporting issues with DEBUG_THREADLOCALS.

One issue was that the ast_mutex_* API was being used within the context of the
thread local data destructors.  We would go off and allocate more thread local data
while the pthread lib was in the middle of destroying it all.  This led to a memory 
leak.

Another issue was an invalid argument being provided to the the object_add
API call.

(closes issue #13678)
Reported by: ys
Tested by: Russell

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Joshua Colp
209eaf5032 Blocked revisions 164733 via svnmerge
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  r164733 | file | 2008-12-16 12:50:11 -0400 (Tue, 16 Dec 2008) | 6 lines
  
  Be more detailed about why the include did not get included.
  (closes issue #14071)
  Reported by: kshumard
  Patches:
        pbx_config.patch.improvederroroutput.txt uploaded by kshumard (license 92)
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Russell Bryant
4fa477f58c Merged revisions 164675 via svnmerge from
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r164675 | russell | 2008-12-16 10:00:29 -0600 (Tue, 16 Dec 2008) | 19 lines

Merged revisions 164672 via svnmerge from 
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r164672 | russell | 2008-12-16 09:56:37 -0600 (Tue, 16 Dec 2008) | 11 lines

Fix a memory leak related to the use of the "setvar" configuration option.

The problem was that these variables were being appended to the list of vars
on the sip_pvt every time a re-registration or re-subscription came in.
Since it's just a waste of memory to put them there unless the request was an
INVITE, then the fix is to check the request type before copying the vars.

(closes issue #14037)
Reported by: marvinek
Tested by: russell

........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@164676 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-16 16:02:22 +00:00
Joshua Colp
3c6b69d7a6 Merged revisions 164659 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r164659 | file | 2008-12-16 11:44:28 -0400 (Tue, 16 Dec 2008) | 4 lines
  
  When using externhost make sure the port gets set to the bindaddr port if one was not specified in the externhost value itself.
  (closes issue #13634)
  Reported by: performer
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2008-12-16 15:45:46 +00:00
Steve Murphy
218d7cebb1 Merged revisions 164648 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r164648 | murf | 2008-12-16 08:31:54 -0700 (Tue, 16 Dec 2008) | 13 lines

Merged revisions 164634 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r164634 | murf | 2008-12-16 08:15:58 -0700 (Tue, 16 Dec 2008) | 5 lines

I added a sentence to clarify why - and ' ' are ignored in patterns
as per bug 14076. Leif says he'll put some stuff about it in the
extensions.conf sample, etc.


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2008-12-16 15:37:46 +00:00
Russell Bryant
eba2ca0412 Merged revisions 164623 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r164623 | russell | 2008-12-16 09:00:27 -0600 (Tue, 16 Dec 2008) | 5 lines

Set MINIVM_ACCMESS_STATUS in all cases.  Also, remove a variable that was not needed.

(closes issue #14081)
Reported by: pkempgen

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2008-12-16 15:01:31 +00:00
Russell Bryant
cb3bf2b7e5 Merged revisions 164606 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r164606 | russell | 2008-12-16 08:31:02 -0600 (Tue, 16 Dec 2008) | 13 lines

Merged revisions 164605 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r164605 | russell | 2008-12-16 08:28:10 -0600 (Tue, 16 Dec 2008) | 5 lines

Don't try to change working directory if a directory was not configured.

(closes issue #14089)
Reported by: caspy

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2008-12-16 14:39:40 +00:00
Russell Bryant
4f181c60de Blocked revisions 164602 via svnmerge
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r164602 | russell | 2008-12-16 08:17:45 -0600 (Tue, 16 Dec 2008) | 7 lines

Fix usage of the DAHDI_VMWI ioctl.

(closes issue #14090)
Reported by: alecdavis
Patches:
      chan_dahdi.VMWI_ioctl.diff.txt uploaded by alecdavis (license 585)

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2008-12-16 14:18:07 +00:00
Russell Bryant
4946bd6c1f Blocked revisions 164525 via svnmerge
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r164525 | russell | 2008-12-15 16:25:46 -0600 (Mon, 15 Dec 2008) | 6 lines

Open a timer before loading configuration so that the trunking configuration option
will take effect.

(closes issue #14082)
Reported by: seandarcy

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2008-12-15 22:26:01 +00:00
Russell Bryant
ab1aa9a013 Blocked revisions 164522 via svnmerge
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r164522 | russell | 2008-12-15 16:22:43 -0600 (Mon, 15 Dec 2008) | 4 lines

Fix log message to refer to the generic timing interface, not DAHDI specifically

(inspired by issue #14082)

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2008-12-15 22:23:03 +00:00
Russell Bryant
920968e19a Blocked revisions 164519 via svnmerge
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r164519 | russell | 2008-12-15 15:53:30 -0600 (Mon, 15 Dec 2008) | 7 lines

Make sure we handle a uint32_t payload in ast_frdup()

(closes issue #14080)
Reported by: fnordian
Patches:
      frame.patch uploaded by fnordian (license 110)

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2008-12-15 21:54:52 +00:00
Steve Murphy
a1e8150599 Blocked revisions 164415 via svnmerge
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r164415 | murf | 2008-12-15 12:42:05 -0700 (Mon, 15 Dec 2008) | 16 lines

I was getting this warning during a compile
on a 64-bit machine running ubuntu server 8.10, 
and gcc-4.3.2:

   [CXXi] chan_vpb.ii -> chan_vpb.oo
cc1plus: warnings being treated as errors
In file included from /home/murf/asterisk/trunk/include/asterisk/utils.h:671,
                 from chan_vpb.cc:46:
/home/murf/asterisk/trunk/include/asterisk/strings.h: In function ‘char* ast_str_truncate(ast_str*, ssize_t)’:
/home/murf/asterisk/trunk/include/asterisk/strings.h:479: error: comparison between signed and unsigned integer expressions
make[1]: *** [chan_vpb.oo] Error 1
make: *** [channels] Error 2

which this fix silences


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2008-12-15 19:55:40 +00:00
Mark Michelson
2bfe78f74e Merged revisions 164423 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r164423 | mmichelson | 2008-12-15 13:53:29 -0600 (Mon, 15 Dec 2008) | 11 lines

Merged revisions 164422 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r164422 | mmichelson | 2008-12-15 13:53:08 -0600 (Mon, 15 Dec 2008) | 3 lines

Add the deadlock note to ast_spawn_extension as well


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2008-12-15 19:53:49 +00:00
Mark Michelson
f981298c64 Merged revisions 164419 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r164419 | mmichelson | 2008-12-15 13:51:24 -0600 (Mon, 15 Dec 2008) | 12 lines

Merged revisions 164416 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r164416 | mmichelson | 2008-12-15 13:45:07 -0600 (Mon, 15 Dec 2008) | 4 lines

Add notes to autoservice and pbx doxygen regarding a potential
deadlock scenario so that it is avoided in the future


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2008-12-15 19:51:53 +00:00
Tilghman Lesher
592262787b Merged revisions 164349 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r164349 | tilghman | 2008-12-15 12:09:58 -0600 (Mon, 15 Dec 2008) | 4 lines
  
  When querying for the structure of the CDR table, remove the schema, if it
  exists.
  (Closes issue #14058)
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2008-12-15 18:26:46 +00:00
Joshua Colp
004570ef12 Merged revisions 164351 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r164351 | file | 2008-12-15 14:12:24 -0400 (Mon, 15 Dec 2008) | 13 lines
  
  Merged revisions 164350 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r164350 | file | 2008-12-15 14:11:21 -0400 (Mon, 15 Dec 2008) | 6 lines
    
    Do not try to unlock a non-existant channel if the transfer fails.
    (closes issue #13800)
    Reported by: dwagner
    Patches:
          asterisk-1.4.22-chan-sip-nullp.patch uploaded by tweety (license 608)
  ........
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2008-12-15 18:13:13 +00:00
Joshua Colp
a469b317c6 Blocked revisions 164344 via svnmerge
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  r164344 | file | 2008-12-15 13:46:32 -0400 (Mon, 15 Dec 2008) | 10 lines
  
  Blocked revisions 164343 via svnmerge
  
  ........
    r164343 | file | 2008-12-15 13:43:59 -0400 (Mon, 15 Dec 2008) | 4 lines
    
    Use autoconf logic to determine whether the system has timersub or not. Do not blindly assume Solaris does not.
    (closes issue #13838)
    Reported by: ano
  ........
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2008-12-15 17:47:10 +00:00
Joshua Colp
2afe695cb9 Blocked revisions 164312 via svnmerge
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  r164312 | file | 2008-12-15 13:24:28 -0400 (Mon, 15 Dec 2008) | 4 lines
  
  Use ast_seekstream to return the file stream back to the beginning instead of directly seeking to zero. This is because some audio formats have headers at the front that need to be skipped, which will be done by the format module.
  (closes issue #14079)
  Reported by: elguero
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2008-12-15 17:25:35 +00:00
Russell Bryant
c80f103ed6 Blocked revisions 164309 via svnmerge
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r164309 | russell | 2008-12-15 11:21:38 -0600 (Mon, 15 Dec 2008) | 2 lines

Fix a couple more build issues related to ast_str_opaque

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2008-12-15 17:21:59 +00:00
Russell Bryant
28a97c18ee Merged revisions 164203 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r164203 | russell | 2008-12-15 08:40:24 -0600 (Mon, 15 Dec 2008) | 39 lines

Merged revisions 164201 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r164201 | russell | 2008-12-15 08:31:37 -0600 (Mon, 15 Dec 2008) | 31 lines

Handle a case where a call can be bridged to a channel that is still ringing.

The issue that was reported was about a case where a RINGING channel got 
redirected to an extension to pick up a call from parking.  Once the parked 
call got taken out of parking, it heard silence until the other side answered.  
Ideally, the caller that was parked would get a ringing indication.  This patch
fixes this case so that the caller receives ringback once it comes out of 
parking until the other side answers.

The fixes are:

 - Make sure we remember that a channel was an outgoing channel when doing 
   a masquerade.  This prevents an erroneous ast_answer() call on the channel,
   which causes a bogus 200 OK to be sent in the case of SIP.

 - Add some additional comments to explain related parts of code.

 - Update the handling of the ast_channel visible_indication field.  Storing 
   values that are not stateful is pointless.  Control frames that are events 
   or commands should be ignored.

 - When a bridge first starts, check to see if the peer channel needs to be 
   given ringing indication because the calling side is still ringing.

 - Rework ast_indicate_data() a bit for the sake of readability.

(closes issue #13747)
Reported by: davidw
Tested by: russell
Review: http://reviewboard.digium.com/r/90/

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2008-12-15 16:26:09 +00:00
Russell Bryant
8560f7fc20 Merged revisions 164272 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r164272 | russell | 2008-12-15 10:17:55 -0600 (Mon, 15 Dec 2008) | 8 lines

When a reload is issued, always process the configuration for dundi.conf.

The reason is that a reload can be used to refresh DNS lookups for defined peers.
Even if the config file hasn't changed, we want to process it for that purpose.

(closes issue #13776)
Reported by: kombjuder

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2008-12-15 16:19:00 +00:00
Mark Michelson
22ccedb314 Merged revisions 164270 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r164270 | mmichelson | 2008-12-15 10:16:47 -0600 (Mon, 15 Dec 2008) | 4 lines

Fix a compile warning and a logic error that could have been bad
for non-realtime queues


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2008-12-15 16:17:19 +00:00
Mark Michelson
fde29e8939 Merged revisions 164268 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r164268 | mmichelson | 2008-12-15 10:10:43 -0600 (Mon, 15 Dec 2008) | 17 lines

Fix up a few issues with regards to queues

* Fix reference counting used in the __queues_show function
* Add code to be sure that the "queue show" command does not
  print information for a realtime queue which has been deleted
  from the backend
* Add a missing unref to the realtime queue loading function for
  the case where a queue is in the module's container but has been
  deleted from the realtime backend

(closes issue #14033)
Reported by: cristiandimache
Patches:
      14033.patch uploaded by putnopvut (license 60)
Tested by: cristiandimache


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2008-12-15 16:12:09 +00:00
Joshua Colp
87c8ebd562 Merged revisions 164257 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r164257 | file | 2008-12-15 11:41:22 -0400 (Mon, 15 Dec 2008) | 4 lines
  
  Make app_fax compatible with newer versions of spandsp. This remains backwards compatible with earlier versions though so do not fret.
  (closes issue #14073)
  Reported by: seandarcy
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2008-12-15 15:48:22 +00:00
Joshua Colp
f04d178d28 Blocked revisions 164208 via svnmerge
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  r164208 | file | 2008-12-15 11:21:23 -0400 (Mon, 15 Dec 2008) | 2 lines
  
  Update to work with new ast_str changes.
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2008-12-15 15:21:49 +00:00
Joshua Colp
050bb72674 Blocked revisions 164205 via svnmerge
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  r164205 | file | 2008-12-15 11:06:38 -0400 (Mon, 15 Dec 2008) | 10 lines
  
  Blocked revisions 164204 via svnmerge
  
  ........
    r164204 | file | 2008-12-15 11:05:08 -0400 (Mon, 15 Dec 2008) | 4 lines
    
    Can we try not to assign an unsigned int to -1?
    (closes issue #14074)
    Reported by: wetwired
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2008-12-15 15:07:34 +00:00
Sean Bright
67265f048f Blocked revisions 163951-163952 via svnmerge
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  r163951 | seanbright | 2008-12-12 22:00:26 -0500 (Fri, 12 Dec 2008) | 1 line
  
  Use actual tables instead of ASCII art ones.
........
  r163952 | seanbright | 2008-12-12 22:03:15 -0500 (Fri, 12 Dec 2008) | 1 line
  
  This shouldn't have gotten commited.  We might want to generate this into a separate file instead of the version controlled one.
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2008-12-13 03:05:41 +00:00
Joshua Colp
8777c0ce95 Blocked revisions 163912 via svnmerge
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  r163912 | file | 2008-12-12 20:59:24 -0400 (Fri, 12 Dec 2008) | 2 lines
  
  Only detach and destroy the whisper audiohooks if they are actually in use.
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2008-12-13 01:00:41 +00:00
Terry Wilson
d56bef322c Merged revisions 163873 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r163873 | twilson | 2008-12-12 17:48:26 -0600 (Fri, 12 Dec 2008) | 6 lines
  
  When using realtime queues, app_queue wasn't updating the strategy if it was changed in the realtime backend.  This patch resolves the issue for almost all situations.  It is currently not supported to switch to the linear strategy via realtime since the ao2_container for members will have been set to have multiple buckets and therefore the members would be unordered.
  
  (closes issue #14034)
  Reported by: cristiandimache
  Tested by: otherwiseguy, cristiandimache
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2008-12-13 00:05:53 +00:00
Russell Bryant
20aa89180e Blocked revisions 163828 via svnmerge
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r163828 | russell | 2008-12-12 17:06:55 -0600 (Fri, 12 Dec 2008) | 2 lines

Add a note to indicate why this only supports one channel for now.

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2008-12-12 23:07:36 +00:00
Tilghman Lesher
cb692bccde Merged revisions 163762 via svnmerge from
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  r163762 | tilghman | 2008-12-12 16:04:26 -0600 (Fri, 12 Dec 2008) | 14 lines
  
  Merged revisions 163761 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r163761 | tilghman | 2008-12-12 16:03:10 -0600 (Fri, 12 Dec 2008) | 7 lines
    
    Simple fix for Ctrl-C not immediately exiting Asterisk, but also add a
    pointer inside editline to look back to asterisk.c, so others don't spend
    as much time as I did looking (in the wrong place) for the appropriate
    function.
    Reported by: ZX81, via the #asterisk-users channel
    Fixed by: me (license 14)
  ........
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Russell Bryant
32f63eb45e Blocked revisions 163716 via svnmerge
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r163716 | russell | 2008-12-12 14:12:23 -0600 (Fri, 12 Dec 2008) | 5 lines

Add a new CLI command, "channel redirect", which is similar in operation
to AMI Redirect.

Review: http://reviewboard.digium.com/r/89/

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2008-12-12 20:12:43 +00:00
Steve Murphy
076e16cf9f Merged revisions 163675 via svnmerge from
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  r163675 | murf | 2008-12-12 12:16:32 -0700 (Fri, 12 Dec 2008) | 1 line
  
  demote always-appearing debug message (for certain boards) to ast_debug lev 3 msg instead
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2008-12-12 19:40:04 +00:00
Russell Bryant
1b1801f155 Merged revisions 163670 via svnmerge from
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r163670 | russell | 2008-12-12 12:45:03 -0600 (Fri, 12 Dec 2008) | 6 lines

Rename a number of tcptls_session variables.  There are no functional changes here.

The name "ser" was used in a lot of places.  However, it is a relic from when
the struct was a server_instance, not a session_instance.  It was renamed since
it represents both a server or client connection.

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2008-12-12 18:52:14 +00:00
Russell Bryant
ab65296510 Merged revisions 163667 via svnmerge from
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r163667 | russell | 2008-12-12 12:33:27 -0600 (Fri, 12 Dec 2008) | 5 lines

Fix a small race condition in sip_tcp_locate().

We must increase the reference count on the tcptls_session _before_ unlocking
the thread list.

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2008-12-12 18:34:56 +00:00
Russell Bryant
a3a784988d Merged revisions 163642 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r163642 | russell | 2008-12-12 12:19:47 -0600 (Fri, 12 Dec 2008) | 7 lines

Resolve crashes when using SIP TCP/TLS with qualify.

The problem was a reference count error on the tcptls_session structure.

(closes issue #13989)
Reported by: Nugget

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Joshua Colp
934c3d072a Blocked revisions 163629 via svnmerge
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  r163629 | file | 2008-12-12 14:17:12 -0400 (Fri, 12 Dec 2008) | 4 lines
  
  When a device registers we need to unlink them (if linked) from the peers_by_ip container and link them back in since their IP address has changed. This would have manifested itself if you configured a new device (as type=peer), registered, and then tried to place a call from the device. Since the peer was not linked into the peers_by_ip container it would have never been found.
  (closes issue #13811)
  Reported by: pj
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2008-12-12 18:18:34 +00:00
Michiel van Baak
1953d171c5 Merged revisions 163612 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r163612 | mvanbaak | 2008-12-12 18:22:47 +0100 (Fri, 12 Dec 2008) | 7 lines
  
  Document default Monitor file location.
  
  (closes issue #14065)
  Reported by: kshumard
  Patches:
        res_monitor.documentation.patch.txt uploaded by kshumard (license 92)
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2008-12-12 17:24:31 +00:00
Michiel van Baak
31969af963 Blocked revisions 163582 via svnmerge
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  r163582 | mvanbaak | 2008-12-12 18:14:13 +0100 (Fri, 12 Dec 2008) | 11 lines
  
  Fix codec capability setup in chan_skinny
  
  Behaviour now is that general codec config flows to default_line and default_device. [devices] stuff amends default_device and similar for [lines]. These are copied to individual device and line as they are created.
  Added confcapability and confprefs for the configured stuff which doesn't change as device and so on are connected. prefs are based on line prefs if they exist, else the device prefs are used (prefs identifies codec order).
  
  (closes issue #13806)
  Reported by: pj
  Patches:
        codecs.diff uploaded by wedhorn (license 30)
  Tested by: pj and me
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2008-12-12 17:16:02 +00:00
Joshua Colp
e93be99823 Blocked revisions 163579 via svnmerge
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  r163579 | file | 2008-12-12 12:55:15 -0400 (Fri, 12 Dec 2008) | 4 lines
  
  Since chan_sip is callback devicestate driven do not pass in actual states, pass in unknown so we get asked. Additionally do not pass in an actual device state value in ast_setstate since the channel may be callback driven.
  (closes issue #13525)
  Reported by: pj
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2008-12-12 16:55:59 +00:00
Russell Bryant
0ac7322a4b Merged revisions 163512 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r163512 | russell | 2008-12-12 08:44:06 -0600 (Fri, 12 Dec 2008) | 13 lines

Merged revisions 163511 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r163511 | russell | 2008-12-12 08:40:31 -0600 (Fri, 12 Dec 2008) | 5 lines

Specify uint32_t for variables storing a CRC32 so that it is actually 32 bits
on 64-bit machines, as well.

(inspired by issue #13879)

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2008-12-12 14:45:44 +00:00
Russell Bryant
d94b6eeeaf Merged revisions 163449 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r163449 | russell | 2008-12-12 07:55:30 -0600 (Fri, 12 Dec 2008) | 34 lines

Merged revisions 163448 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r163448 | russell | 2008-12-12 07:44:08 -0600 (Fri, 12 Dec 2008) | 26 lines

Resolve issues that could cause DTMF to be processed out of order.

These changes come from team/russell/issue_12658

1) Change autoservice to put digits on the head of the channel's frame readq 
   instead of the tail.  If there were frames on the readq that autoservice 
   had not yet read, the previous code would have resulted in out of order 
   processing.  This required a new API call to queue a frame to the head 
   of the queue instead of the tail.

2) Change up the processing of DTMF in ast_read().  Some of the problems 
   were the result of having two sources of pending DTMF frames.  There 
   was the dtmfq and the more generic readq.  Both were used for pending 
   DTMF in various scenarios.  Simplifying things to only use the frame 
   readq avoids some of the problems.

3) Fix a bug where a DTMF END frame could get passed through when it 
   shouldn't have.  If code set END_DTMF_ONLY in the middle of digit emulation,
   and a digit arrived before emulation was complete, digits would get 
   processed out of order.

(closes issue #12658)
Reported by: dimas
Tested by: russell, file
Review: http://reviewboard.digium.com/r/85/

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2008-12-12 14:05:01 +00:00
Tilghman Lesher
40a2446502 Merged revisions 163384 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r163384 | tilghman | 2008-12-11 17:38:56 -0600 (Thu, 11 Dec 2008) | 16 lines
  
  Merged revisions 163383 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r163383 | tilghman | 2008-12-11 17:35:55 -0600 (Thu, 11 Dec 2008) | 9 lines
    
    When a Ctrl-C or Ctrl-D ends a remote console, on certain shells, the terminal
    is messed up.  By intercepting those events with a signal handler in the remote
    console, we can avoid those issues.
    (closes issue #13464)
     Reported by: tzafrir
     Patches: 
           20081110__bug13464.diff.txt uploaded by Corydon76 (license 14)
     Tested by: blitzrage
  ........
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2008-12-11 23:43:53 +00:00
Matthew Nicholson
9cd7139df4 Merged revisions 163317 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r163317 | mnicholson | 2008-12-11 16:49:59 -0600 (Thu, 11 Dec 2008) | 16 lines
  
  Merged revisions 163316 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r163316 | mnicholson | 2008-12-11 16:44:31 -0600 (Thu, 11 Dec 2008) | 9 lines
    
    Clean up the dundi cache every 5 minutes.
    
    (closes issue #13819)
    Reported by: adomjan
    Patches:
          pbx_dundi.c-clearcache.patch uploaded by adomjan (license 487)
          dundi_clearecache3.diff uploaded by mnicholson (license 96)
    Tested by: adomjan
  ........
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2008-12-11 22:52:18 +00:00
Russell Bryant
cb8fbf7b13 Merged revisions 163254 via svnmerge from
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r163254 | russell | 2008-12-11 15:48:08 -0600 (Thu, 11 Dec 2008) | 16 lines

Merged revisions 163253 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r163253 | russell | 2008-12-11 15:46:29 -0600 (Thu, 11 Dec 2008) | 8 lines

Fix some observed slowdowns in dialplan processing.

The change is to remove autoservice usage from dialplan functions that do not
need it because they do not perform operations that potentially block.

(closes issue #13940)
Reported by: tbelder

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2008-12-11 21:49:13 +00:00
Russell Bryant
d878d06d2c Blocked revisions 163241 via svnmerge
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r163241 | russell | 2008-12-11 15:21:31 -0600 (Thu, 11 Dec 2008) | 8 lines

Fix a problem where continuous mode will get inadvertently get turned off if set_rate()
is used while continuous mode was already turned on.

(closes issue #13738)
Reported by: smurfix
Patches:
      res.patch.fixed uploaded by smurfix (license 547)

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2008-12-11 21:22:56 +00:00
Russell Bryant
668ac35844 Merged revisions 163171 via svnmerge from
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r163171 | russell | 2008-12-11 14:07:47 -0600 (Thu, 11 Dec 2008) | 16 lines

Fix the "failed" extension for outgoing calls.  

The conversion to use ast_check_hangup() everywhere instead of checking the softhangup
flag directly introduced this problem.  The issue is that ast_check_hangup() checked
for tech_pvt to be NULL.  Unfortunately, this will be NULL is some valid circumstances,
such as with a dummy channel.

The fix is simple.  Don't check tech_pvt.  It's pointless, because the code path that
sets this to NULL is when the channel hangup callback gets called.  This happens inside
of ast_hangup(), which is the same function responsible for freeing the channel.  Any
code calling ast_check_hangup() better not be calling it after that point, and if so,
we have a bigger problem at hand.

(closes issue #14035)
Reported by: erogoza

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2008-12-11 20:09:10 +00:00
Tilghman Lesher
09a6317035 Merged revisions 163168 via svnmerge from
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  r163168 | tilghman | 2008-12-11 14:02:35 -0600 (Thu, 11 Dec 2008) | 5 lines
  
  Sometimes even Linux needs -lm to link libtonezone, such as when libtonezone
  is compiled statically.
  (closes issue #13887)
   Reported by: tzafrir
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2008-12-11 20:04:15 +00:00
Russell Bryant
2f56ece505 Merged revisions 163094 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r163094 | russell | 2008-12-11 11:06:16 -0600 (Thu, 11 Dec 2008) | 19 lines

Merged revisions 163092 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r163092 | russell | 2008-12-11 10:54:51 -0600 (Thu, 11 Dec 2008) | 11 lines

Fix an issue that made it so you could only have a single caller executing
a custom feature at a time.  This was especially problematic when custom
features ran for any appreciable amount of time.

The fix turned out to be quite simple.  The dynamic features are now stored
in a read/write list instead of a list using a mutex.

(closes issue #13478)
Reported by: neutrino88
Fix suggested by file

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2008-12-11 17:11:50 +00:00
Tilghman Lesher
1ac2518c09 Merged revisions 163089 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r163089 | tilghman | 2008-12-11 10:52:24 -0600 (Thu, 11 Dec 2008) | 13 lines
  
  Merged revisions 163088 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r163088 | tilghman | 2008-12-11 10:51:27 -0600 (Thu, 11 Dec 2008) | 6 lines
    
    Don't wait forever, if there's a specified recording timeout.
    (closes issue #13885)
     Reported by: bamby
     Patches: 
           res_agi.c.patch uploaded by bamby (license 430)
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2008-12-11 16:53:12 +00:00
Mark Michelson
bfbf6f4aac Merged revisions 163085 via svnmerge from
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r163085 | mmichelson | 2008-12-11 10:47:34 -0600 (Thu, 11 Dec 2008) | 12 lines

Merged revisions 163084 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r163084 | mmichelson | 2008-12-11 10:46:22 -0600 (Thu, 11 Dec 2008) | 4 lines

Revert this cast to long. Using time_t here causes build failures on a 
FreeBSD 32-bit build.


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2008-12-11 16:49:18 +00:00
Mark Michelson
5826008ffd Merged revisions 163081 via svnmerge from
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r163081 | mmichelson | 2008-12-11 10:33:16 -0600 (Thu, 11 Dec 2008) | 22 lines

Merged revisions 163080 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r163080 | mmichelson | 2008-12-11 10:24:43 -0600 (Thu, 11 Dec 2008) | 14 lines

Fix a potential crash due to unsafe datastore handling.

This patch also contains a conversion from using long to time_t
for representing times for a queue, as well as some whitespace
fixes.

(closes issue #14060)
Reported by: nivek
Patches:
      datastore_fixup.patch.corrected uploaded by nivek (license 636)
	  with slight modification from me
Tested by: nivek


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2008-12-11 16:40:46 +00:00
Sean Bright
9cfa4c1dfe Blocked revisions 163037 via svnmerge
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  r163037 | seanbright | 2008-12-11 10:40:07 -0500 (Thu, 11 Dec 2008) | 8 lines
  
  Fix some of the grammar issues in doc/tex/qos.tex.
  
  (closes issue #14049)
  Reported by: kshumard
  Patches:
        doc.tex.qos.tex.patch uploaded by kshumard (license 92)
        (Slight modifications by seanbright)
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2008-12-11 15:41:29 +00:00
Joshua Colp
ea9f10f371 Blocked revisions 162997 via svnmerge
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  r162997 | file | 2008-12-11 11:05:49 -0400 (Thu, 11 Dec 2008) | 4 lines
  
  When a device registers to use it is entirely possible that they may be in use, so tell the core that we don't know the devstate and have it ask us for it.
  (closes issue #13525)
  Reported by: pj
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2008-12-11 15:06:49 +00:00
Jeff Peeler
69b008a143 Merged revisions 162927 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r162927 | jpeeler | 2008-12-10 16:53:34 -0600 (Wed, 10 Dec 2008) | 11 lines

Merged revisions 162926 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r162926 | jpeeler | 2008-12-10 16:52:51 -0600 (Wed, 10 Dec 2008) | 3 lines

Oops, inverted logic for a strcasecmp check. Pointed out by mmichelson, thanks!


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2008-12-10 22:54:01 +00:00
Joshua Colp
a75b51ae14 Blocked revisions 162923 via svnmerge
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  r162923 | file | 2008-12-10 18:48:58 -0400 (Wed, 10 Dec 2008) | 4 lines
  
  Fix reloads of aliased CLI commands. Due to changes done to turn it into a single memory allocation we can't just use the existing CLI alias structure. We have to destroy all existing ones and then create new ones.
  (closes issue #14054)
  Reported by: pj
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2008-12-10 22:50:26 +00:00
Jeff Peeler
378383b674 Merged revisions 162891 via svnmerge from
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r162891 | jpeeler | 2008-12-10 16:11:46 -0600 (Wed, 10 Dec 2008) | 13 lines

Merged revisions 162874 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r162874 | jpeeler | 2008-12-10 16:04:18 -0600 (Wed, 10 Dec 2008) | 5 lines

(closes issue #13229)
Reported by: clegall_proformatique

Ensure that moh_generate does not return prematurely before local_ast_moh_stop is called. Also, the sleep in mp3_spawn now only occurs for http locations since it seems to have been added originally only for failing media streams.

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2008-12-10 22:12:40 +00:00
Joshua Colp
b1eb64beec Merged revisions 162805 via svnmerge from
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  r162805 | file | 2008-12-10 15:02:57 -0400 (Wed, 10 Dec 2008) | 13 lines
  
  Merged revisions 162804 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r162804 | file | 2008-12-10 15:01:17 -0400 (Wed, 10 Dec 2008) | 6 lines
    
    Fix subscription based MWI up a bit. We only want to put sip: at the beginning of the URI if it is not already there and revert code to ignore destination check if subscribing for MWI.
    (closes issue #12560)
    Reported by: vsauer
    Patches:
          patch001.diff uploaded by ramonpeek (license 266)
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2008-12-10 19:04:13 +00:00
Joshua Colp
b4d8725ec1 Merged revisions 162739 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r162739 | file | 2008-12-10 13:53:09 -0400 (Wed, 10 Dec 2008) | 13 lines
  
  Merged revisions 162738 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r162738 | file | 2008-12-10 13:50:43 -0400 (Wed, 10 Dec 2008) | 6 lines
    
    When a SIP peer unregisters set the expiry time back to 0 so that the 200 OK contains an expires of 0.
    (closes issue #13599)
    Reported by: hjourdain
    Patches:
          chan_sip.c.diff uploaded by hjourdain (license 583)
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2008-12-10 17:54:06 +00:00
Michiel van Baak
b35132ebeb Blocked revisions 162687 via svnmerge
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  r162687 | mvanbaak | 2008-12-10 18:09:15 +0100 (Wed, 10 Dec 2008) | 8 lines
  
  add tab completion for 'core set debug X filename.c'
  
  (closes issue #13969)
  Reported by: jtodd
  Patches:
        20081205__bug13969.diff.txt uploaded by Corydon76 (license 14)
  Tested by: mvanbaak, eliel
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2008-12-10 17:11:08 +00:00
Mark Michelson
3495f89e85 Blocked revisions 162672 via svnmerge
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r162672 | mmichelson | 2008-12-10 10:46:51 -0600 (Wed, 10 Dec 2008) | 21 lines

Blocked revisions 162670 via svnmerge

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r162670 | mmichelson | 2008-12-10 10:44:37 -0600 (Wed, 10 Dec 2008) | 14 lines

Update to stringfield handling so that side-effects on
parameters are not evaluated multiple times.

An example where this caused a problem was in chan_sip.c, with
the line

  ast_string_field_set(p, fromdomain, ++fromdomain);

This patch was originally uploaded to issue #13783 by
jamessan. While the issue was closed for other reasons, this
patch is valid and fixes a separate problem, and is thus
being committed.


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2008-12-10 16:48:10 +00:00
Mark Michelson
a94a213cc3 Merged revisions 162667 via svnmerge from
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r162667 | mmichelson | 2008-12-10 10:39:10 -0600 (Wed, 10 Dec 2008) | 16 lines

Merged revisions 162659 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r162659 | mmichelson | 2008-12-10 10:10:25 -0600 (Wed, 10 Dec 2008) | 8 lines

Add missing documentation to misdn.txt

(closes issue #14052)
Reported by: festr
Patches:
      misdn.txt.patch uploaded by festr (license 443)


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2008-12-10 16:39:40 +00:00
Mark Michelson
52eea9252e Merged revisions 162664 via svnmerge from
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r162664 | mmichelson | 2008-12-10 10:34:35 -0600 (Wed, 10 Dec 2008) | 19 lines

Merged revisions 162663 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r162663 | mmichelson | 2008-12-10 10:24:56 -0600 (Wed, 10 Dec 2008) | 11 lines

Revert fix for issue 13570. It has caused more problems than
it helped to fix.

(closes issue #13783)
Reported by: navkumar


(closes issue #14025)
Reported by: ffs


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2008-12-10 16:36:38 +00:00
Joshua Colp
05b88f7bb3 Blocked revisions 162660 via svnmerge
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  r162660 | file | 2008-12-10 12:11:06 -0400 (Wed, 10 Dec 2008) | 6 lines
  
  FreeBSD also needs libgen.h
  (closes issue #14051)
  Reported by: ys
  Patches:
        res_http_post.c.diff uploaded by ys (license 281)
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2008-12-10 16:11:38 +00:00
Joshua Colp
db16f8c6bc Merged revisions 162656 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r162656 | file | 2008-12-10 12:06:59 -0400 (Wed, 10 Dec 2008) | 13 lines
  
  Merged revisions 162653 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r162653 | file | 2008-12-10 12:05:29 -0400 (Wed, 10 Dec 2008) | 6 lines
    
    Increment the sequence number on the end packets for RFC2833. After reading the RFC some more and doing some testing I agree with this change.
    (closes issue #12983)
    Reported by: vt
    Patches:
          dtmf_inc_seqnum_on_end_pkts.diff uploaded by vt (license 520)
  ........
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2008-12-10 16:07:51 +00:00
Joshua Colp
62f443578b Merged revisions 162619 via svnmerge from
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  r162619 | file | 2008-12-10 11:22:26 -0400 (Wed, 10 Dec 2008) | 4 lines
  
  When transmitting a register set the socket port to the local one for the transport being used, not the port for the remote server.
  (closes issue #13633)
  Reported by: performer
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2008-12-10 15:26:49 +00:00
Michiel van Baak
5c111c073f Merged revisions 162583 via svnmerge from
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  r162583 | mvanbaak | 2008-12-10 12:34:09 +0100 (Wed, 10 Dec 2008) | 5 lines
  
  Make res_snmp.so compile on OpenBSD.
  
  OpenBSD uses an old version of gcc which throws an error
  if you use a macro that's not #defined
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2008-12-10 11:36:28 +00:00
Joshua Colp
1609a7cf8a Blocked revisions 162542 via svnmerge
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  r162542 | file | 2008-12-09 21:09:06 -0400 (Tue, 09 Dec 2008) | 6 lines
  
  Finish conversion to using ARRAY_LEN and remove it as a janitor project.
  (closes issue #14032)
  Reported by: bkruse
  Patches:
        14032.patch uploaded by bkruse (license 132)
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2008-12-10 01:10:11 +00:00
Mark Michelson
327262b4bb Merged revisions 162488 via svnmerge from
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r162488 | kpfleming | 2008-12-09 17:41:02 -0600 (Tue, 09 Dec 2008) | 1 line

it does help if the compiler attribute syntax is correct
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2008-12-09 23:45:15 +00:00
Tilghman Lesher
a4216d24c2 Merged revisions 162466 via svnmerge from
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  r162466 | tilghman | 2008-12-09 17:10:34 -0600 (Tue, 09 Dec 2008) | 9 lines
  
  Merged revisions 162463 via svnmerge from 
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  ........
    r162463 | tilghman | 2008-12-09 17:08:53 -0600 (Tue, 09 Dec 2008) | 2 lines
    
    Oops, should be "tz", not "zonetag".
  ........
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2008-12-09 23:11:21 +00:00
Russell Bryant
97373c3410 Blocked revisions 162418 via svnmerge
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r162418 | russell | 2008-12-09 16:38:41 -0600 (Tue, 09 Dec 2008) | 7 lines

Add some additional Asterisk project developer documentation.  

After the nightly update of the documentation on asterisk.org, I'll post 
an update to asterisk-dev with a pointer to the changes.  This covers some
release branch and commit policy information.  None of this should be a
surprise, since it's just documenting what we have already been doing.

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2008-12-09 22:38:55 +00:00
Russell Bryant
1f2e9439fd Merged revisions 162414 via svnmerge from
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r162414 | russell | 2008-12-09 16:25:06 -0600 (Tue, 09 Dec 2008) | 16 lines

Merged revisions 162413 via svnmerge from 
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r162413 | russell | 2008-12-09 16:17:39 -0600 (Tue, 09 Dec 2008) | 8 lines

Remove the test_for_thread_safety() function completely.

The test is not valid.  Besides, if we actually suspected that recursive
mutexes were not working, we would get a ton of LOG_ERROR messages when
DEBUG_THREADS is turned on.

(inspired by a discussion on the asterisk-dev list)

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2008-12-09 22:30:52 +00:00
Tilghman Lesher
cd4008a657 Merged revisions 162355 via svnmerge from
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  r162355 | tilghman | 2008-12-09 15:57:09 -0600 (Tue, 09 Dec 2008) | 11 lines
  
  Merged revisions 162348 via svnmerge from 
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  ........
    r162348 | tilghman | 2008-12-09 15:53:25 -0600 (Tue, 09 Dec 2008) | 4 lines
    
    We appear to have documented tz= in the [general] section of voicemail.conf,
    without actually having implemented it.  Oops.
    (Reported by Olivier on the -users list)
  ........
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2008-12-09 21:59:42 +00:00
Joshua Colp
a5202c0285 Merged revisions 162342 via svnmerge from
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  r162342 | file | 2008-12-09 17:16:37 -0400 (Tue, 09 Dec 2008) | 11 lines
  
  Merged revisions 162341 via svnmerge from 
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  ........
    r162341 | file | 2008-12-09 17:14:29 -0400 (Tue, 09 Dec 2008) | 4 lines
    
    Add 'down' as a valid state for directed call pickup. This creeps up when we receive session progress when dialing a device and not ringing.
    (closes issue #14005)
    Reported by: ddl
  ........
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2008-12-09 21:17:25 +00:00
Russell Bryant
f8b80089f1 Merged revisions 162291 via svnmerge from
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r162291 | russell | 2008-12-09 14:59:54 -0600 (Tue, 09 Dec 2008) | 17 lines

Merged revisions 162286 via svnmerge from 
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r162286 | russell | 2008-12-09 14:57:35 -0600 (Tue, 09 Dec 2008) | 9 lines

Fix an issue where callers on an incoming call on an SLA trunk would not hear ringback.

We need to make sure that we don't start writing audio to the trunk channel until we're
actually ready to answer it.  Otherwise, the channel driver will treat it as inband
progress, even though all they are getting is silence.

(closes issue #12471)
Reported by: mthomasslo

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2008-12-09 21:02:01 +00:00
Joshua Colp
1fbbdbf26d Merged revisions 162275 via svnmerge from
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  r162275 | file | 2008-12-09 16:46:11 -0400 (Tue, 09 Dec 2008) | 11 lines
  
  Merged revisions 162273 via svnmerge from 
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  ........
    r162273 | file | 2008-12-09 16:44:32 -0400 (Tue, 09 Dec 2008) | 4 lines
    
    Fix double declaration of 'x' on the PPC platform.
    (closes issue #14038)
    Reported by: ffloimair
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2008-12-09 20:47:11 +00:00
Dwayne M. Hubbard
42b53c5026 Blocked revisions 161115 via svnmerge
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r161115 | dhubbard | 2008-12-04 17:00:30 -0600 (Thu, 04 Dec 2008) | 11 lines

If 'faxdetect=yes' in sip.conf, switch to a 'fax' extension (if it exists) after T38 is negotiated.  
Terry Wilson created the original patch for this functionality, which I slightly modified and added 
the faxdetect=yes|no configuration option.  This patch is only for T38 fax detection and does not 
do anything for G711 over SIP fax detection.  By default, this option is disabled. 

Reviewboard: http://reviewboard.digium.com/r/69/

This functionality is for issue AST-140.



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2008-12-09 20:45:20 +00:00
Steve Murphy
263a912934 Merged revisions 162271 via svnmerge from
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r162271 | murf | 2008-12-09 13:40:31 -0700 (Tue, 09 Dec 2008) | 9 lines

Merged revisions 162264 via svnmerge from 
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r162264 | murf | 2008-12-09 13:20:54 -0700 (Tue, 09 Dec 2008) | 1 line

In discussion with seanbright on #asterisk-dev,  I have added a default rule, and an option to suppress the default rule from being generated in the flex output, for the sake of those OS's where they didn't tweak flex's ECHO macro, and the compiler doesn't like it. The regressions are OK with this.
........

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2008-12-09 20:44:10 +00:00
Mark Michelson
fa979f0828 Merged revisions 162266 via svnmerge from
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r162266 | mmichelson | 2008-12-09 14:30:07 -0600 (Tue, 09 Dec 2008) | 14 lines

Merged revisions 162265 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r162265 | mmichelson | 2008-12-09 14:28:44 -0600 (Tue, 09 Dec 2008) | 6 lines

If we fail to start a thread for the pbx to run in, we need to
be sure to decrease the number of active calls on the system.

This fix may relate to ABE-1713, but it is not certain yet.


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2008-12-09 20:31:21 +00:00
Joshua Colp
3948ee149b Merged revisions 162205 via svnmerge from
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  r162205 | file | 2008-12-09 15:48:35 -0400 (Tue, 09 Dec 2008) | 14 lines
  
  Merged revisions 162204 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r162204 | file | 2008-12-09 15:47:07 -0400 (Tue, 09 Dec 2008) | 7 lines
    
    Make sure that the timestamp for DTMF is not the same as the previous voice frame and do not send audio when transmitting DTMF as this confuses some equipment.
    (closes issue #13209)
    Reported by: ip-rob
    Patches:
          13209.diff uploaded by file (license 11)
    Tested by: ip-rob, bujones
  ........
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2008-12-09 19:50:39 +00:00
Joshua Colp
a38d0e6fa1 Merged revisions 162197 via svnmerge from
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  r162197 | file | 2008-12-09 15:08:39 -0400 (Tue, 09 Dec 2008) | 11 lines
  
  Merged revisions 162188 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r162188 | file | 2008-12-09 15:06:14 -0400 (Tue, 09 Dec 2008) | 4 lines
    
    Take video into account when early bridging RTP.
    (closes issue #13535)
    Reported by: davidw
  ........
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2008-12-09 19:09:40 +00:00
Steve Murphy
7e806f5ddb Merged revisions 162140 via svnmerge from
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r162140 | murf | 2008-12-09 11:35:35 -0700 (Tue, 09 Dec 2008) | 9 lines

Merged revisions 162136 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r162136 | murf | 2008-12-09 11:13:39 -0700 (Tue, 09 Dec 2008) | 1 line

Previous fix used ast_malloc and ast_copy_string and messed up the standalone stuff. Fixed.
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2008-12-09 18:40:49 +00:00
Steve Murphy
a4189e547f Merged revisions 162079 via svnmerge from
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r162079 | murf | 2008-12-09 10:18:03 -0700 (Tue, 09 Dec 2008) | 53 lines

Merged revisions 162013 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r162013 | murf | 2008-12-09 09:31:55 -0700 (Tue, 09 Dec 2008) | 45 lines

(closes issue #14019)
Reported by: ckjohnsonme
Patches:
      14019.diff uploaded by murf (license 17)
Tested by: ckjohnsonme, murf

This crash was the result of a few small errors that
would combine in 64-bit land to result in a crash.

32-bit land might have seen these combine to mysteriously
drop the args to an application call, in certain
circumstances.

Also, in trying to find this bug, I spotted
a situation in the flex input, where, in passing
back a 'word' to the parser, it would allocate
a buffer larger than necessary. I changed the
usage in such situations, so that strdup was
not used, but rather, an ast_malloc, followed
by ast_copy_string.

I removed a field from the pval struct, in
u2, that was never getting used, and set in
one spot in the code. I believe it was an
artifact of a previous fix to make switch
cases work invisibly with extens.

And, for goto's I removed a '!' from
before a strcmp, that has been there
since the initial merging of AEL2, that
might prevent the proper target of a 
goto from being found. This was pretty
harmless on its own, as it would just
louse up a consistency check for users.

Many thanks to ckjohnsonme for providing
a simplified and complete set of information
about the bug, that helped considerably in
finding and fixing the problem.

Now, to get aelparse up and running again
in trunk, and out of its "horribly broken" state,
so I can run the regression suite!


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2008-12-09 17:27:26 +00:00
Eliel C. Sardanons
a10c5cade1 Blocked revisions 161947 via svnmerge
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  r161947 | eliel | 2008-12-09 12:49:30 -0200 (Tue, 09 Dec 2008) | 8 lines
  
  Avoid allocating memory for a thread that don't need it. Also, this memory was not being freed until the
  main thread ends. (That is never).
  
  (closes issue #14040)
  Reported by: eliel
  Patches:
        func_odbc.c.patch uploaded by eliel (license 64)
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2008-12-09 16:54:18 +00:00
Russell Bryant
404c99d4d3 Merged revisions 162016 via svnmerge from
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r162016 | russell | 2008-12-09 10:47:39 -0600 (Tue, 09 Dec 2008) | 13 lines

Merged revisions 162014 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r162014 | russell | 2008-12-09 10:46:53 -0600 (Tue, 09 Dec 2008) | 5 lines

Allow DISA to handle extensions that start with #.

(closes issue #13330)
Reported by: jcovert

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2008-12-09 16:48:57 +00:00
Russell Bryant
6d83552430 Merged revisions 161951 via svnmerge from
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r161951 | russell | 2008-12-09 08:57:39 -0600 (Tue, 09 Dec 2008) | 23 lines

Merged revisions 161948 via svnmerge from 
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r161948 | russell | 2008-12-09 08:52:25 -0600 (Tue, 09 Dec 2008) | 15 lines

Fix a problem with GROUP() settings on a masquerade.

The previous code carried over group settings from the old channel to the new
one.  However, it did nothing with the group settings that were already on the
new channel.  This patch removes all group settings that already existed on the
new channel.

I have a more complicated version of this patch which addresses only the most
blatant problem with this, which is that a channel can end up with multiple
group settings in the same category.  However, I could not think of a use case
for keeping any of the group settings from the old channel, so I went this route
for now.

(closes AST-152)

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2008-12-09 15:02:40 +00:00
Joshua Colp
dd21fc652d Blocked revisions 161869 via svnmerge
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  r161869 | file | 2008-12-08 17:41:50 -0400 (Mon, 08 Dec 2008) | 6 lines
  
  Add alw as a valid file extension for alaw and ulw as a valid file extension for ulaw.
  (closes issue #14001)
  Reported by: henrikw
  Patches:
        alw.diff uploaded by henrikw (license 627)
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2008-12-08 21:42:21 +00:00
Joshua Colp
3c11a93e30 Blocked revisions 161830 via svnmerge
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  r161830 | file | 2008-12-08 16:53:50 -0400 (Mon, 08 Dec 2008) | 2 lines
  
  Update autosupport script with a few changes.
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2008-12-08 20:54:34 +00:00
Tilghman Lesher
8de845fa84 Merged revisions 161790 via svnmerge from
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  r161790 | tilghman | 2008-12-08 12:49:50 -0600 (Mon, 08 Dec 2008) | 6 lines
  
  Allocate enough space initially for the message.
  (closes issue #14027)
   Reported by: junky
   Patches: 
         M14027.diff uploaded by junky (license 177)
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2008-12-08 18:51:30 +00:00
Joshua Colp
fcbd8a7d24 Blocked revisions 161787 via svnmerge
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  r161787 | file | 2008-12-08 14:47:32 -0400 (Mon, 08 Dec 2008) | 6 lines
  
  Fix a regression introduced when the PBX timeouts were converted to milliseconds. collect_digits now gets milliseconds fed to it, not seconds.
  (closes issue #14012)
  Reported by: dveiga
  Patches:
        14012.patch uploaded by bkruse (license 132)
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2008-12-08 18:48:54 +00:00
Joshua Colp
77033f4ad0 Merged revisions 161726 via svnmerge from
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  r161726 | file | 2008-12-08 13:53:32 -0400 (Mon, 08 Dec 2008) | 13 lines
  
  Merged revisions 161725 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r161725 | file | 2008-12-08 13:52:10 -0400 (Mon, 08 Dec 2008) | 6 lines
    
    Make the usereqphone option work again.
    (closes issue #13474)
    Reported by: mmaguire
    Patches:
          20080912_bug13474.diff uploaded by mmaguire (license 571)
  ........
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2008-12-08 17:54:45 +00:00
Eliel C. Sardanons
775cc3c73b Blocked revisions 161637 via svnmerge
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  r161637 | eliel | 2008-12-08 02:23:50 -0200 (Mon, 08 Dec 2008) | 4 lines
  
  - Fix a leak while printing an argument description.
  - Avoid printing the name of an argument in the [Arguments] tag if there is no description
    for that argument.
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2008-12-08 04:25:34 +00:00
Eliel C. Sardanons
f7f38c0048 Blocked revisions 161604 via svnmerge
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  r161604 | eliel | 2008-12-08 01:35:55 -0200 (Mon, 08 Dec 2008) | 10 lines
  
  Add voicemail related applications and functions XML documentation:
      applications:
        - VoiceMail()
        - VoiceMailMain()
        - MailboxExists()
        - VMAuthenticate()
      functions:
        - MAILBOX_EXISTS()
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2008-12-08 03:37:33 +00:00
Eliel C. Sardanons
55ba4370fb Blocked revisions 161571 via svnmerge
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  r161571 | eliel | 2008-12-07 20:43:46 -0200 (Sun, 07 Dec 2008) | 2 lines
  
  Introduce SMS() application XML documentation.
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2008-12-07 22:44:44 +00:00
Eliel C. Sardanons
445293432d Blocked revisions 161536 via svnmerge
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  r161536 | eliel | 2008-12-06 19:18:51 -0200 (Sat, 06 Dec 2008) | 2 lines
  
  Move Speech* applications and functions documentation to XML.
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2008-12-06 21:20:29 +00:00
Mark Michelson
0e2af385ab Merged revisions 161493 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r161493 | mmichelson | 2008-12-05 17:24:38 -0600 (Fri, 05 Dec 2008) | 8 lines

If the autoloop flag is set on a channel, then we need to 
add 1 to the priority when checking if the extension exists. Otherwise,
gosubs will fail.

This was discovered when investigating an asterisk-users mailing list post
made by Gary Hawkins.


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2008-12-05 23:26:45 +00:00
Mark Michelson
aa3e7acc89 Fix a reversed-logic ast_strlen_zero.
This problem was pointed out on the asterisk-users list
by Jim Dickenson



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2008-12-05 22:59:43 +00:00
Sean Bright
d9c4ba0231 Merged revisions 161427 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r161427 | seanbright | 2008-12-05 16:08:43 -0500 (Fri, 05 Dec 2008) | 22 lines
  
  Merged revisions 161426 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ................
    r161426 | seanbright | 2008-12-05 16:02:20 -0500 (Fri, 05 Dec 2008) | 15 lines
    
    Merged revisions 161421 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.2
    
    ........
      r161421 | seanbright | 2008-12-05 15:50:23 -0500 (Fri, 05 Dec 2008) | 8 lines
      
      Fix build errors on FreeBSD (uint -> unsigned int).
      
      (closes issue #14006)
      Reported by: alphaque
      Patches:
            astobj2.h-patch uploaded by alphaque (license 259)
            (Slightly modified by seanbright)
    ........
  ................
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2008-12-05 21:13:44 +00:00
Sean Bright
2e732411d4 Merged revisions 161349-161350 via svnmerge from
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  r161349 | seanbright | 2008-12-05 10:56:15 -0500 (Fri, 05 Dec 2008) | 5 lines
  
  When using IMAP_STORAGE, it's important to convert bare newlines (\n) in
  emailbody and pagerbody to CR-LF so that the IMAP server doesn't spit out an
  error.  This was informally reported on #asterisk-dev a few weeks ago.  Reviewed
  by Mark M. on IRC.
........
  r161350 | seanbright | 2008-12-05 11:04:36 -0500 (Fri, 05 Dec 2008) | 2 lines
  
  Use ast_free() instead of free(), pointed out by eliel on IRC.
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2008-12-05 16:10:56 +00:00
Russell Bryant
39742e18c6 Merged revisions 161288 via svnmerge from
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r161288 | russell | 2008-12-05 08:16:24 -0600 (Fri, 05 Dec 2008) | 10 lines

Merged revisions 161287 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r161287 | russell | 2008-12-05 08:12:14 -0600 (Fri, 05 Dec 2008) | 2 lines

Fix a NULL format string warning found by buildbot.

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2008-12-05 14:17:50 +00:00
Russell Bryant
ab9270440e Merged revisions 161252 via svnmerge from
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r161252 | russell | 2008-12-05 07:46:01 -0600 (Fri, 05 Dec 2008) | 2 lines

Resolve a compiler warning from buildbot about a NULL format string.

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2008-12-05 13:55:58 +00:00
Russell Bryant
29eeedeb5e Compiling is a good thing.
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2008-12-05 13:55:33 +00:00
Eliel C. Sardanons
3a661e91d6 Blocked revisions 161218 via svnmerge
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  r161218 | eliel | 2008-12-05 08:31:25 -0200 (Fri, 05 Dec 2008) | 8 lines
  
  Janitor, use ARRAY_LEN() when possible.
  
  (closes issue #13990)
  Reported by: eliel
  Patches:
        array_len.diff uploaded by eliel (license 64)
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2008-12-05 10:36:45 +00:00
Tilghman Lesher
71a4c78fa8 Merged revisions 161181 via svnmerge from
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  r161181 | tilghman | 2008-12-04 23:41:41 -0600 (Thu, 04 Dec 2008) | 11 lines
  
  The first file should have a blank config filename in the structure, so that
  when a save occurs to a different filename, everything goes to the alternate
  filename, instead of appending to the original.  This is important for the
  AMI command UpdateConfig.
  (closes issue #13301)
   Reported by: trevo
   Patches: 
         20081113__bug13301.diff.txt uploaded by Corydon76 (license 14)
         20081113__bug13301__1.6.0.diff.txt uploaded by Corydon76 (license 14)
   Tested by: Corydon76, blitzrage
........


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2008-12-05 05:43:36 +00:00
Sean Bright
2b4bfa3c77 Merged revisions 161147 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r161147 | seanbright | 2008-12-04 21:47:54 -0500 (Thu, 04 Dec 2008) | 3 lines
  
  Check the return value of fread/fwrite so the compiler doesn't complain.  Only a
  problem when IMAP_STORAGE is enabled.
........


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2008-12-05 02:51:48 +00:00
Eliel C. Sardanons
b1c3c64122 Blocked revisions 161077 via svnmerge
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  r161077 | eliel | 2008-12-04 17:31:48 -0200 (Thu, 04 Dec 2008) | 2 lines
  
  Fix minor coding guidelines introduced with CLI permissions.
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2008-12-04 19:35:46 +00:00
Jeff Peeler
704a528b72 Merged revisions 161014 via svnmerge from
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r161014 | jpeeler | 2008-12-04 12:32:20 -0600 (Thu, 04 Dec 2008) | 17 lines

Merged revisions 161013 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r161013 | jpeeler | 2008-12-04 12:30:41 -0600 (Thu, 04 Dec 2008) | 9 lines

(closes issue #13835)
Reported by: matt_b
Tested by: jpeeler

This mirrors a check that was present in ast_rtp_read to also be in ast_rtp_raw_write to not schedule sending the receiver report if the remote RTCP endpoint address isn't present in the RTCP structure.

Closes AST-142.


........

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2008-12-04 18:36:40 +00:00
Mark Michelson
cf7e1de2d1 Merged revisions 160945 via svnmerge from
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r160945 | mmichelson | 2008-12-04 10:45:06 -0600 (Thu, 04 Dec 2008) | 23 lines

Merged revisions 160943 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r160943 | mmichelson | 2008-12-04 10:44:18 -0600 (Thu, 04 Dec 2008) | 15 lines

Fix a callerid parsing issue. If someone formatted callerid like the
following: "name <number>" (including the quotation marks), then the parts
would be parsed as 

name: "name
number: number

This is because the closing quotation mark was not discovered since the number
and everything after was parsed out of the string earlier. Now, there is a check
to see if the closing quote occurs after the number, so that we can know if we
should strip off the opening quote on the name.

Closes AST-158


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2008-12-04 16:47:19 +00:00
Michiel van Baak
65d469c32d Blocked revisions 160938 via svnmerge
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  r160938 | mvanbaak | 2008-12-04 17:37:13 +0100 (Thu, 04 Dec 2008) | 9 lines
  
  Add debug flag so skinny debug will show information about packets.
  We dont want to scare users with this, so we added a devmode compile flag
  
  (closes issue #13952)
  Reported by: wedhorn
  Patches:
        packetdebug3.diff uploaded by wedhorn (license 30)
  Tested by: mvanbaak, wedhorn
........


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2008-12-04 16:40:27 +00:00
Eliel C. Sardanons
86e0ff40ac Blocked revisions 160896 via svnmerge
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  r160896 | eliel | 2008-12-04 11:45:32 -0200 (Thu, 04 Dec 2008) | 7 lines
  
  Added XML documentation for the following AGI commands:
          - get option
          - get variable
          - hangup
          - noop
........


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2008-12-04 13:50:41 +00:00
Richard Mudgett
a225f81162 Merged revisions 160856 via svnmerge from
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  r160856 | rmudgett | 2008-12-03 19:36:39 -0600 (Wed, 03 Dec 2008) | 1 line
  
  Jcolp pointed out that num will also match number
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2008-12-04 01:38:08 +00:00
Richard Mudgett
3ea1eea638 Merged revisions 160854 via svnmerge from
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  r160854 | rmudgett | 2008-12-03 19:14:22 -0600 (Wed, 03 Dec 2008) | 4 lines
  
  *  Found a couple more places where num/number needed to be done
  so 1.4 upgraders will not have problems.
  *  Added curly braces and minor tweaks.
........


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2008-12-04 01:17:17 +00:00
Tilghman Lesher
0e51fd84c4 Merged revisions 160791 via svnmerge from
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  r160791 | tilghman | 2008-12-03 15:58:21 -0600 (Wed, 03 Dec 2008) | 9 lines
  
  Merged revisions 160770 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r160770 | tilghman | 2008-12-03 15:54:07 -0600 (Wed, 03 Dec 2008) | 2 lines
    
    Some compilers warn on null format strings; some don't (caught by buildbot)
  ........
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2008-12-03 22:01:54 +00:00
Steve Murphy
5acbac37b5 Merged revisions 160760 via svnmerge from
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r160760 | murf | 2008-12-03 14:09:15 -0700 (Wed, 03 Dec 2008) | 23 lines

Merged revisions 160703 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r160703 | murf | 2008-12-03 13:41:42 -0700 (Wed, 03 Dec 2008) | 11 lines

(closes issue #13597)
Reported by: john8675309
Patches:
      patch.13597 uploaded by murf (license 17)
Tested by: murf, john8675309

This patch causes the setcid func to update the CDR
clid after setting the channel field.

I also notice that in trunk, the num/number of 1.4 is
left out; I decided to include the option to use
either in trunk, so as not to have 1.4 upgraders
not to have problems.


........

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2008-12-03 21:23:23 +00:00
Jason Parker
69e24fffbf Merged revisions 160699-160700 via svnmerge from
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  r160699 | qwell | 2008-12-03 14:32:20 -0600 (Wed, 03 Dec 2008) | 7 lines
  
  Fix typo when ListCategories returns none.
  
  (closes issue #13994)
  Reported by: mika
  Patches:
        ListCategoriesActionPatch.diff uploaded by mika (license 624)
........
  r160700 | qwell | 2008-12-03 14:35:36 -0600 (Wed, 03 Dec 2008) | 1 line
  
  Another place this is missing
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2008-12-03 20:36:10 +00:00
Eliel C. Sardanons
e250448a1b Merged revisions 160663 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r160663 | eliel | 2008-12-03 17:25:30 -0200 (Wed, 03 Dec 2008) | 13 lines
  
  - iax2-provision was not freeing iax_templates structure when unloading the chan_iax2.so module.
  - Move the code to start using the LIST macros.
  
  Review: http://reviewboard.digium.com/r/72
  
  (closes issue #13232)
  Reported by: eliel
  Patches:
        iax2-provision.patch.txt uploaded by eliel (license 64)
        (with minor changes pointed by Mark Michelson on review board)
  Tested by: eliel
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2008-12-03 20:02:21 +00:00
Mark Michelson
3d5b168cbc Merged revisions 160626 via svnmerge from
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r160626 | mmichelson | 2008-12-03 12:37:46 -0600 (Wed, 03 Dec 2008) | 16 lines

Add some safety measures when using gosub, especially when using the options
for app_dial and app_queue to run a gosub when the call is answered.

* Check for the existence of the gosub target in gosub_exec. If it is nonexistent,
  then this will cause errors when we attempt to actually run the gosub, including
  a definite memory leak and potential crashes. Return an error in this situation
* Check the return value of pbx_exec in app_dial and app_queue before attempting
  to actually run the gosub routine. If there was an error, we should not attempt
  to run the gosub.
* Change a '|' to a ',' in app_queue.
* Add some extra curly braces where they had been missing previously.

(closes issue #13548)
Reported by: fiddur


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2008-12-03 18:41:28 +00:00
Tilghman Lesher
02ea919584 Merged revisions 160585 via svnmerge from
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................
  r160585 | tilghman | 2008-12-03 11:59:36 -0600 (Wed, 03 Dec 2008) | 11 lines
  
  Blocked revisions 160570 via svnmerge
  
  ........
    r160570 | tilghman | 2008-12-03 11:55:12 -0600 (Wed, 03 Dec 2008) | 5 lines
    
    During bridge code, the channel bridge may return a retry code, if a transfer
    was initiated but not yet completed.  If the bridge is immediately retried,
    then we may send a storm of TXREQ packets, even though the first set is sent
    reliably (retransmitted).  Fixes AST-137.
  ........
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Tilghman Lesher
43386dbc31 Merged revisions 160559 via svnmerge from
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  r160559 | tilghman | 2008-12-03 11:38:59 -0600 (Wed, 03 Dec 2008) | 14 lines
  
  Merged revisions 160558 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r160558 | tilghman | 2008-12-03 11:34:34 -0600 (Wed, 03 Dec 2008) | 7 lines
    
    If an entry is added to the directory during a scan when another entry expires,
    then that new entry will not be processed promptly, but must wait for either a
    future entry to start or a current entry's retry to occur.  If no other entries
    exist in the directory (other than the new entries) when a bunch expire, then
    the new entries must wait until another new entry is added to be processed.
    This was a rather weird race condition, really.  Fixes AST-147.
  ........
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2008-12-03 17:41:07 +00:00
Mark Michelson
f2168e3a67 Merged revisions 160555 via svnmerge from
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r160555 | mmichelson | 2008-12-03 11:07:09 -0600 (Wed, 03 Dec 2008) | 11 lines

When investigating issue #13548, I found that gosub
handling in app_queue was just completely wrong, mostly
because the channel operations being performed were being
done on the incorrect channel.

With this set of changes, a gosub will correctly run on
the answering queue member's channel. There are still crash
issues which occur if there are dialplan syntax errors, so
I cannot yet close the referenced issue.


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Tilghman Lesher
e8a9aba72b Merged revisions 160552 via svnmerge from
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  r160552 | tilghman | 2008-12-03 11:01:03 -0600 (Wed, 03 Dec 2008) | 11 lines
  
  Merged revisions 160551 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r160551 | tilghman | 2008-12-03 10:58:34 -0600 (Wed, 03 Dec 2008) | 4 lines
    
    Don't start scanning the directory until all modules are loaded, because some
    required modules (channels, apps, functions) may not yet be in memory yet.
    Fixes AST-149.
  ........
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Tilghman Lesher
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  r160481 | tilghman | 2008-12-03 08:11:53 -0600 (Wed, 03 Dec 2008) | 14 lines
  
  Merged revisions 160480 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r160480 | tilghman | 2008-12-03 08:09:35 -0600 (Wed, 03 Dec 2008) | 7 lines
    
    Jon Bonilla (Manwe) pointed out on the -dev list:
    "I guess that having only ip-phones in mind is not a good approach. Since it is
    possible to have a sip proxy connected to asterisk we could receive a 407
    (unauthorized) or 483 (too many hops) as response and dialog ending would not be
    a good behavior."
    So modified.
  ........
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2008-12-03 14:13:27 +00:00
Sean Bright
f1c993b17f Fix some 'make menuselect' breakage introduced by recent merges.
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2008-12-03 00:53:56 +00:00
Tilghman Lesher
452f1f9c0d Merged revisions 156388 via svnmerge from
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  r156388 | tilghman | 2008-11-12 15:34:51 -0600 (Wed, 12 Nov 2008) | 12 lines
  
  Merged revisions 156386 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r156386 | tilghman | 2008-11-12 15:18:57 -0600 (Wed, 12 Nov 2008) | 5 lines
    
    When using call limits under 1 second, infinite call lengths are allowed,
    instead.
    (closes issue #13851)
     Reported by: ruddy
  ........
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Tilghman Lesher
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  r156290 | jpeeler | 2008-11-12 13:11:15 -0600 (Wed, 12 Nov 2008) | 11 lines
  
  Merged revisions 156289 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
  r156289 | jpeeler | 2008-11-12 13:10:12 -0600 (Wed, 12 Nov 2008) | 3 lines
  
  For whatever reason, gcc only warned me about the possible use of an uninitialized variable when compiling 1.6.1. 
  
  
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160392 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-02 23:10:07 +00:00
Tilghman Lesher
1a2fee8c28 Merged revisions 156228 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r156228 | jpeeler | 2008-11-12 12:32:46 -0600 (Wed, 12 Nov 2008) | 16 lines
  
  Merged revisions 156178 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
  r156178 | jpeeler | 2008-11-12 11:53:44 -0600 (Wed, 12 Nov 2008) | 8 lines
  
  (closes issue #13173)
  Reported by: pep
  
  This change adds an announce_thread responsible for playing announcements to an existing conference. This allows all announcing to be immediately stopped if necessary but more importantly allows other threads that need to play something to not block. There are multiple benefits to this, but the actual bug is for solving the scenario for a channel to be unusable after hang up for the entire duration of the parting announcement. The parting announcement can be extremely long depending on what the user recorded upon joining the conference.
  
  Reviewed by Russell on Review Board:
  http://reviewboard.digium.com/r/25/
  
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160391 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-02 23:08:17 +00:00
Tilghman Lesher
107dfe787e Merged revisions 152969,153122,154264,154268,154366,155399,155863,156166,156295,156690,156756,158066,158082,158540,158602,159276 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r152969 | tilghman | 2008-10-30 15:35:46 -0500 (Thu, 30 Oct 2008) | 10 lines
  
  Merged revisions 152958 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r152958 | tilghman | 2008-10-30 15:33:28 -0500 (Thu, 30 Oct 2008) | 3 lines
    
    Cannot join detached threads.  See http://www.opengroup.org/onlinepubs/000095399/functions/pthread_join.html
    (Closes issue #13400)
  ........
................
  r153122 | tilghman | 2008-10-31 11:35:21 -0500 (Fri, 31 Oct 2008) | 10 lines
  
  Merged revisions 153114 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r153114 | tilghman | 2008-10-31 11:30:32 -0500 (Fri, 31 Oct 2008) | 3 lines
    
    Turn off qualify on uncached realtime peers.
    (Closes issue #13383)
  ........
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  r154264 | tilghman | 2008-11-04 12:59:48 -0600 (Tue, 04 Nov 2008) | 10 lines
  
  Recorded merge of revisions 154263 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r154263 | tilghman | 2008-11-04 12:58:05 -0600 (Tue, 04 Nov 2008) | 3 lines
    
    Make the monitor thread non-detached, so it can be joined (suggested by Russell
    on -dev list).
  ........
................
  r154268 | rmudgett | 2008-11-04 13:07:26 -0600 (Tue, 04 Nov 2008) | 11 lines
  
  Merged revisions 154266 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r154266 | rmudgett | 2008-11-04 13:01:08 -0600 (Tue, 04 Nov 2008) | 4 lines
    
    JIRA ABE-1703
    mISDN sets the channel to the wrong state when it receives
    the indication AST_CONTROL_RINGING.
  ........
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  r154366 | tilghman | 2008-11-04 14:51:18 -0600 (Tue, 04 Nov 2008) | 16 lines
  
  Merged revisions 154365 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r154365 | tilghman | 2008-11-04 14:49:33 -0600 (Tue, 04 Nov 2008) | 9 lines
    
    On busy systems, it's possible for the values checked within a single line
    of code to change, unless the structure is locked to ensure a consistent
    state.
    (closes issue #13717)
     Reported by: kowalma
     Patches: 
           20081102__bug13717.diff.txt uploaded by Corydon76 (license 14)
     Tested by: kowalma
  ........
................
  r155399 | tilghman | 2008-11-07 16:28:58 -0600 (Fri, 07 Nov 2008) | 14 lines
  
  Merged revisions 155398 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r155398 | tilghman | 2008-11-07 16:27:32 -0600 (Fri, 07 Nov 2008) | 7 lines
    
    Clarify error message.
    (closes issue #13809)
     Reported by: denke
     Patches: 
           20081104__bug13809.diff.txt uploaded by Corydon76 (license 14)
     Tested by: denke
  ........
................
  r155863 | mmichelson | 2008-11-10 15:14:44 -0600 (Mon, 10 Nov 2008) | 22 lines
  
  Merged revisions 155861 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
  r155861 | mmichelson | 2008-11-10 15:07:39 -0600 (Mon, 10 Nov 2008) | 14 lines
  
  Channel drivers assume that when their indicate callback
  is invoked, that the channel on which the callback was called
  is locked. This patch corrects an instance in chan_agent where
  a channel's indicate callback is called directly without first
  locking the channel.
  
  This was leading to some observed locking issues in chan_local,
  but considering that all channel drivers operate under the
  same expectations, the generic fix in chan_agent is the right
  way to go.
  
  AST-126
  
  
  ........
................
  r156166 | russell | 2008-11-12 11:38:20 -0600 (Wed, 12 Nov 2008) | 15 lines
  
  Merged revisions 156164 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
  r156164 | russell | 2008-11-12 11:29:52 -0600 (Wed, 12 Nov 2008) | 7 lines
  
  Move the sanity check that makes sure "always fork" is not set along with the 
  console option to be after the code that reads options from asterisk.conf.  
  This resolves a situation where Asterisk can start taking up 100% when
  misconfigured.
  (Thanks to Bryce Porter (x86 on IRC) for letting me log in to his system to
   figure out what was causing the 100% CPU problem.)
  
  ........
................
  r156295 | tilghman | 2008-11-12 13:28:22 -0600 (Wed, 12 Nov 2008) | 13 lines
  
  Merged revisions 156294 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r156294 | tilghman | 2008-11-12 13:26:45 -0600 (Wed, 12 Nov 2008) | 6 lines
    
    If the SLA thread is not started, then reload causes a memory leak.
    (closes issue #13889)
     Reported by: eliel
     Patches: 
           app_meetme.c.patch uploaded by eliel (license 64)
  ........
................
  r156690 | tilghman | 2008-11-13 15:30:41 -0600 (Thu, 13 Nov 2008) | 14 lines
  
  Merged revisions 156688 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r156688 | tilghman | 2008-11-13 15:24:00 -0600 (Thu, 13 Nov 2008) | 7 lines
    
    Provide more space for all the data which can appear in an originating
    channel name.
    (closes issue #13398)
     Reported by: bamby
     Patches: 
           manager.c.diff uploaded by bamby (license 430)
  ........
................
  r156756 | tilghman | 2008-11-13 18:43:13 -0600 (Thu, 13 Nov 2008) | 13 lines
  
  Merged revisions 156755 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r156755 | tilghman | 2008-11-13 18:41:37 -0600 (Thu, 13 Nov 2008) | 6 lines
    
    ast_waitfordigit() requires that the channel be up, for no good logical
    reason.  This prevents While/EndWhile from working within the "h"
    extension.
    Reported by: jgalarneau (for ABE C.2)
    Fixed by: me
  ........
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  r158066 | mmichelson | 2008-11-20 11:39:06 -0600 (Thu, 20 Nov 2008) | 20 lines
  
  Merged revisions 158053 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
  r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov 2008) | 12 lines
  
  Make sure to set the hangup cause on the calling channel in the case
  that ast_call() fails. For incoming SIP channels, this was causing
  us to send a 603 instead of a 486 when the call-limit was reached on
  the destination channel.
  
  (closes issue #13867)
  Reported by: still_nsk
  Patches:
        13867.diff uploaded by putnopvut (license 60)
  Tested by: blitzrage
  
  
  ........
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  r158082 | mmichelson | 2008-11-20 11:54:31 -0600 (Thu, 20 Nov 2008) | 24 lines
  
  Merged revisions 158071 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
  r158071 | mmichelson | 2008-11-20 11:48:42 -0600 (Thu, 20 Nov 2008) | 16 lines
  
  We don't handle 4XX responses to BYE well. According to
  section 15 of RFC 3261, we should terminate a dialog if we
  receive a 481 or 408 in response to our BYE. Since I am aware
  of at least one phone manufacturer who may sometimes send a 
  404 as well, I am being liberal and saying that any 4XX response
  to a BYE should result in a terminated dialog.
  
  
  (closes issue #12994)
  Reported by: pabelanger
  Patches:
        12994.patch uploaded by putnopvut (license 60)
  
  Closes AST-129
  
  
  ........
................
  r158540 | russell | 2008-11-21 16:12:37 -0600 (Fri, 21 Nov 2008) | 10 lines
  
  Merged revisions 158539 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
  r158539 | russell | 2008-11-21 16:05:55 -0600 (Fri, 21 Nov 2008) | 2 lines
  
  When compiling with DEBUG_THREADS, report the real file/func/line for ao2_lock/ao2_unlock
  
  ........
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  r158602 | tilghman | 2008-11-21 17:14:11 -0600 (Fri, 21 Nov 2008) | 12 lines
  
  Merged revisions 158600 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r158600 | tilghman | 2008-11-21 17:07:46 -0600 (Fri, 21 Nov 2008) | 5 lines
    
    The passed extension may not be the same in the list as the current entry,
    because we strip spaces when copying the extension into the structure.
    Therefore, use the copied item to place the item into the list.
    (found by lmadsen on -dev, fixed by me)
  ........
................
  r159276 | tilghman | 2008-11-25 15:57:59 -0600 (Tue, 25 Nov 2008) | 14 lines
  
  Merged revisions 159269 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r159269 | tilghman | 2008-11-25 15:56:48 -0600 (Tue, 25 Nov 2008) | 7 lines
    
    Don't try to send a response on a NULL pvt.
    (closes issue #13919)
     Reported by: barthpbx
     Patches: 
           chan_iax2.c.patch uploaded by eliel (license 64)
     Tested by: barthpbx
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160389 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-02 22:56:36 +00:00
Tilghman Lesher
b0bad74445 Merged revisions 152216,152287,152369,152467,152569,152605 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r152216 | tilghman | 2008-10-27 16:34:04 -0500 (Mon, 27 Oct 2008) | 13 lines
  
  Merged revisions 152215 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r152215 | tilghman | 2008-10-27 16:32:00 -0500 (Mon, 27 Oct 2008) | 6 lines
    
    Inherit ALL elements of CallerID across a local channel.
    (closes issue #13368)
     Reported by: Peter Schlaile
     Patches: 
           20080826__bug13368.diff.txt uploaded by Corydon76 (license 14)
  ........
................
  r152287 | jpeeler | 2008-10-27 18:31:39 -0500 (Mon, 27 Oct 2008) | 10 lines
  
  Merged revisions 152286 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
  r152286 | jpeeler | 2008-10-27 18:28:49 -0500 (Mon, 27 Oct 2008) | 2 lines
  
  Buffer policy setting for half is not needed.
  
  ........
................
  r152369 | tilghman | 2008-10-28 12:07:39 -0500 (Tue, 28 Oct 2008) | 15 lines
  
  Merged revisions 152368 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r152368 | tilghman | 2008-10-28 12:04:56 -0500 (Tue, 28 Oct 2008) | 8 lines
    
    Reset all DIAL variables back to blank, in case Dial is called multiple times
    per call (which could otherwise lead to inconsistent status reports).
    (closes issue #13216)
     Reported by: ruddy
     Patches: 
           20081014__bug13216.diff.txt uploaded by Corydon76 (license 14)
     Tested by: ruddy
  ........
................
  r152467 | tilghman | 2008-10-28 17:33:40 -0500 (Tue, 28 Oct 2008) | 10 lines
  
  Merged revisions 152463 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r152463 | tilghman | 2008-10-28 17:32:34 -0500 (Tue, 28 Oct 2008) | 3 lines
    
    Quoting in the wrong direction
    (Fixes AST-107)
  ........
................
  r152569 | russell | 2008-10-29 00:34:26 -0500 (Wed, 29 Oct 2008) | 15 lines
  
  Merged revisions 152539 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
  r152539 | russell | 2008-10-29 00:23:51 -0500 (Wed, 29 Oct 2008) | 7 lines
  
  Fix an incorrect usage of sizeof()
  
  (closes issue #13795)
  Reported by: andrew53
  Patches:
  	chan_sip_sizeof.patch uploaded by andrew53 (license 519)
  
  ........
................
  r152605 | murf | 2008-10-29 00:47:13 -0500 (Wed, 29 Oct 2008) | 22 lines
  
  Merged revisions 152538 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
  r152538 | murf | 2008-10-28 23:19:04 -0600 (Tue, 28 Oct 2008) | 14 lines
  
  A little documentation cross-ref between features and
  dial and queue... I wasted some time (stupidly) trying
  to get the one-touch parking stuff working, because it
  didn't occur to me that I had to also have the corresponding
  options in the dial command! Duh! (In all this time, I never
  set this up before!)
  So, to keep some poor fool from suffering the same fate,
  I made the features.conf.sample file mention the corresponding
  opts in dial/queue; and the docs for dial/app specifically
  mention the corresponding decls in the feature.conf file.
  
  I hope this doesn't spoil some vast, eternal plan...
  
  
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160388 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-02 22:19:37 +00:00
Tilghman Lesher
8411899d44 Merged revisions 147518,147689,148000,148112,148268,148917,148988,149062,149131,149201,149205,149208 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r147518 | file | 2008-10-08 09:53:51 -0500 (Wed, 08 Oct 2008) | 9 lines
  
  Merged revisions 147517 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r147517 | file | 2008-10-08 11:51:42 -0300 (Wed, 08 Oct 2008) | 2 lines
    
    If we receive DTMF make sure that the state of the speech structure goes back to being not ready. (issue #LUMENVOX-8)
  ........
................
  r147689 | kpfleming | 2008-10-08 17:26:55 -0500 (Wed, 08 Oct 2008) | 9 lines
  
  Merged revisions 147681 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r147681 | kpfleming | 2008-10-08 17:22:09 -0500 (Wed, 08 Oct 2008) | 3 lines
    
    when parsing a text configuration option, ensure that the buffer on the stack is actually large enough to hold the legal values of that option, and also ensure that sscanf() knows to stop parsing if it would overrun the buffer (without these changes, specifying "buffers=...,immediate" would overflow the buffer on the stack, and could not have worked as expected)
  ........
................
  r148000 | tilghman | 2008-10-09 14:39:34 -0500 (Thu, 09 Oct 2008) | 11 lines
  
  Merged revisions 147997 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r147997 | tilghman | 2008-10-09 14:38:33 -0500 (Thu, 09 Oct 2008) | 4 lines
    
    When blank, callerid name and number should display "unknown caller" in voicemail
    emails.
    (Closes issue #13643)
  ........
................
  r148112 | mmichelson | 2008-10-09 18:15:33 -0500 (Thu, 09 Oct 2008) | 26 lines
  
  Merged revisions 146026 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
  r146026 | murf | 2008-10-03 12:12:54 -0500 (Fri, 03 Oct 2008) | 18 lines
  
  (closes issue #13579)
  Reported by: dwagner
  
  (closes issue #13584)
  Reported by: dwagner
  Tested by: murf, putnopvut
  
  The thought occurred to me that the res= from the extension spawn
  was ending up being returned from the bridge.
  
  "Thou shalt not poison the return value". Made the change
  and it appears to allow blind xfers to work as normal.
  
  If I'm wrong, reopen the bugs. But it looks good to me!
  
  Many thanks to putnopvut for helping me reproduce this!
  
  
  ........
................
  r148268 | tilghman | 2008-10-10 11:31:31 -0500 (Fri, 10 Oct 2008) | 14 lines
  
  Merged revisions 148257 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r148257 | tilghman | 2008-10-10 11:25:31 -0500 (Fri, 10 Oct 2008) | 7 lines
    
    User not notified of temporary greeting, if ODBC storage is in use.
    (closes issue #13659)
     Reported by: moliveras
     Patches: 
           20081009__bug13659.diff.txt uploaded by Corydon76 (license 14)
     Tested by: moliveras
  ........
................
  r148917 | tilghman | 2008-10-14 12:46:48 -0500 (Tue, 14 Oct 2008) | 11 lines
  
  Merged revisions 148916 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r148916 | tilghman | 2008-10-14 12:41:08 -0500 (Tue, 14 Oct 2008) | 4 lines
    
    Ensure that mail headers are 7-bit clean, even when UTF-8 characters are used
    in headers like 'Subject' and 'To'.
    Closes AST-107.
  ........
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  r148988 | tilghman | 2008-10-14 14:03:44 -0500 (Tue, 14 Oct 2008) | 9 lines
  
  Merged revisions 148987 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r148987 | tilghman | 2008-10-14 14:03:08 -0500 (Tue, 14 Oct 2008) | 2 lines
    
    Some compilers warn, some don't.  Fixing.
  ........
................
  r149062 | tilghman | 2008-10-14 15:16:48 -0500 (Tue, 14 Oct 2008) | 13 lines
  
  Merged revisions 149061 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r149061 | tilghman | 2008-10-14 15:09:06 -0500 (Tue, 14 Oct 2008) | 6 lines
    
    Check correct values in the return of ast_waitfor(); also, get rid of a
    possible memory leak.
    (closes issue #13658)
     Reported by: explidous
     Patch by: me
  ........
................
  r149131 | mmichelson | 2008-10-14 16:08:48 -0500 (Tue, 14 Oct 2008) | 15 lines
  
  Merged revisions 149130 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
  r149130 | mmichelson | 2008-10-14 15:49:02 -0500 (Tue, 14 Oct 2008) | 7 lines
  
  Don't allow reserved characters to be used in register
  lines in sip.conf.
  
  (closes issue #13570)
  Reported by: putnopvut
  
  
  ........
................
  r149201 | mmichelson | 2008-10-14 17:41:13 -0500 (Tue, 14 Oct 2008) | 20 lines
  
  Merged revisions 149200 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
  r149200 | mmichelson | 2008-10-14 17:40:42 -0500 (Tue, 14 Oct 2008) | 12 lines
  
  Update the queue with the correct number of calls and
  whether the call was completed within the service level
  when a transfer takes place. This way, we do not "break"
  the leastrecent and fewestcalls strategies by not logging
  a call until after the transferred call has ended.
  
  (closes issue #13395)
  Reported by: Marquis
  Patches:
        app_queue.c.transfer.patch uploaded by Marquis (license 32)
  
  
  ........
................
  r149205 | mmichelson | 2008-10-14 18:04:44 -0500 (Tue, 14 Oct 2008) | 20 lines
  
  Merged revisions 149204 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
  r149204 | mmichelson | 2008-10-14 18:00:01 -0500 (Tue, 14 Oct 2008) | 12 lines
  
  Add a tolerance period for sync-triggered audiohooks
  so that if packetization of audio is close (but not equal)
  we don't end up flushing the audiohooks over small
  inconsistencies in synchronization.
  
  Related to issue #13005, and solves the issue
  for most people who were experiencing the problem.
  However, a small number of people are still experiencing
  the problem on long calls, so I am not closing
  the issue yet
  
  
  ........
................
  r149208 | mmichelson | 2008-10-14 18:15:04 -0500 (Tue, 14 Oct 2008) | 17 lines
  
  Merged revisions 149207 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
  r149207 | mmichelson | 2008-10-14 18:10:26 -0500 (Tue, 14 Oct 2008) | 9 lines
  
  Call register_peer_exten even in the case that the peer's
  IP/port does not change.
  
  (closes issue #13309)
  Reported by: dimas
  Patches:
        v2-13309.patch uploaded by dimas (license 88)
  
  
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160387 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-02 22:16:32 +00:00
Tilghman Lesher
3d50bd62b9 Merged revisions 115313,121770,123272,139624,140205,144257 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r115313 | tilghman | 2008-05-05 15:22:08 -0500 (Mon, 05 May 2008) | 10 lines
  
  Merged revisions 115312 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
  r115312 | tilghman | 2008-05-05 15:17:55 -0500 (Mon, 05 May 2008) | 2 lines
  
  Reverse order, such that user configs override default selections
  
  ........
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  r121770 | crichter | 2008-06-11 06:52:18 -0500 (Wed, 11 Jun 2008) | 9 lines
  
  Merged revisions 121751 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
  r121751 | crichter | 2008-06-11 11:28:04 +0200 (Mi, 11 Jun 2008) | 1 line
  
  fixed issue with previous commit, the find_free_channel test for channels which where inuse was broken.
  ........
................
  r123272 | russell | 2008-06-17 10:52:13 -0500 (Tue, 17 Jun 2008) | 12 lines
  
  Merged revisions 123271 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
  r123271 | russell | 2008-06-17 10:48:31 -0500 (Tue, 17 Jun 2008) | 4 lines
  
  Fix a memory leak in astobj2 that was pointed out by seanbright.  When a container
  got destroyed, the underlying bucket list entry for each object that was in the
  container at that time did not get free'd.
  
  ........
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  r139624 | jpeeler | 2008-08-22 16:57:32 -0500 (Fri, 22 Aug 2008) | 13 lines
  
  Merged revisions 139621 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
  r139621 | jpeeler | 2008-08-22 16:36:13 -0500 (Fri, 22 Aug 2008) | 5 lines
  
  (closes issue #13359)
  Reported by: Laureano
  Patches:
        originate_channel_check.patch uploaded by Laureano (license 265)
  
  ........
................
  r140205 | jpeeler | 2008-08-26 13:48:55 -0500 (Tue, 26 Aug 2008) | 17 lines
  
  Merged revisions 140056 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
  r140056 | jpeeler | 2008-08-26 10:57:02 -0500 (Tue, 26 Aug 2008) | 9 lines
  
  (closes issue #12071)
  Reported by: tzafrir
  Patches:
        dahdi_close.diff uploaded by tzafrir (license 46)
  Tested by: tzafrir, jpeeler
  
  This patch fixes closing open file descriptors in the case of an error.
  
  
  ........
................
  r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) | 9 lines
  
  Merged revisions 144238 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
  r144238 | crichter | 2008-09-24 10:20:52 +0200 (Mi, 24 Sep 2008) | 1 line
  
  improved helptext of misdn_set_opt.
  ........
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2008-12-02 21:46:38 +00:00
Jeff Peeler
07d9e591ac Merged revisions 160333 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r160333 | jpeeler | 2008-12-02 12:04:51 -0600 (Tue, 02 Dec 2008) | 1 line

remove duplicate comment that I accidentally merged
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2008-12-02 18:05:32 +00:00
Jeff Peeler
d3ba6df44e Merged revisions 160319 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r160319 | jpeeler | 2008-12-02 12:00:24 -0600 (Tue, 02 Dec 2008) | 7 lines

(closes issue #13786)
Reported by: tzafrir

Readding DAHDI_CHECK_HOOKSTATE define that was removed in r134260 which fixes not being able to make outgoing calls on some FXO adapters:
http://lists.digium.com/pipermail/asterisk-users/2008-November/thread.html#221553


........


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2008-12-02 18:03:31 +00:00
Tilghman Lesher
89fd15a679 Merged revisions 160308 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r160308 | tilghman | 2008-12-02 11:56:24 -0600 (Tue, 02 Dec 2008) | 17 lines
  
  Merged revisions 160297 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r160297 | tilghman | 2008-12-02 11:42:09 -0600 (Tue, 02 Dec 2008) | 10 lines
    
    When the text does not match exactly (e.g. RTP/SAVP), then the %n conversion
    fails, and the resulting integer is garbage.  Thus, we must initialize the
    integer and check it afterwards for success.
    (closes issue #14000)
     Reported by: folke
     Patches: 
           asterisk-sipbg-sscanf-1.4.22.diff uploaded by folke (license 626)
           asterisk-sipbg-sscanf-1.6.0.1.diff uploaded by folke (license 626)
           asterisk-sipbg-sscanf-trunk-r159896.diff uploaded by folke (license 626)
  ........
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2008-12-02 18:01:47 +00:00
Tilghman Lesher
0a9c41ca89 Merged revisions 160208 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r160208 | tilghman | 2008-12-01 18:37:21 -0600 (Mon, 01 Dec 2008) | 10 lines
  
  Merged revisions 160207 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r160207 | tilghman | 2008-12-01 18:25:16 -0600 (Mon, 01 Dec 2008) | 3 lines
    
    Ensure that Asterisk builds with --enable-dev-mode, even on the latest gcc
    and glibc.
  ........
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2008-12-02 00:57:31 +00:00
Sean Bright
124fb94de6 Merged revisions 160170-160172 via svnmerge from
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................
  r160170 | seanbright | 2008-12-01 18:08:24 -0500 (Mon, 01 Dec 2008) | 1 line
  
  Pay attention to the return value of system(), even if we basically ignore it.
................
  r160171 | seanbright | 2008-12-01 18:18:48 -0500 (Mon, 01 Dec 2008) | 1 line
  
  Silence a build warning. (chan_phone.c:810: warning: value computed is not used)
................
  r160172 | seanbright | 2008-12-01 18:37:49 -0500 (Mon, 01 Dec 2008) | 10 lines
  
  Merged revisions 159976 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r159976 | mvanbaak | 2008-12-01 11:08:36 -0500 (Mon, 01 Dec 2008) | 3 lines
    
    Get rid of the useless format string and argument in the Bogus/ manager channelname.
    Noted by kpfleming and name Bogus/manager suggested by eliel
  ........
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2008-12-01 23:41:16 +00:00
Tilghman Lesher
2ce0cada61 Merged revisions 160097 via svnmerge from
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  r160097 | tilghman | 2008-12-01 15:23:37 -0600 (Mon, 01 Dec 2008) | 2 lines
  
  Use AST_EXT_LIB_SETUP before using AST_EXT_LIB_CHECK or bad things happen.
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2008-12-01 21:45:00 +00:00
Sean Bright
470870f3fc Merged revisions 154919 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r154919 | seanbright | 2008-11-05 17:01:22 -0500 (Wed, 05 Nov 2008) | 2 lines
  
  Fix a problem found while building res_snmp.
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2008-12-01 21:07:48 +00:00
Russell Bryant
d2ac515417 Merged revisions 160004 via svnmerge from
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r160004 | russell | 2008-12-01 11:34:31 -0600 (Mon, 01 Dec 2008) | 14 lines

Merged revisions 160003 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r160003 | russell | 2008-12-01 11:27:30 -0600 (Mon, 01 Dec 2008) | 6 lines

Apply some logic used in iax2_indicate() to iax2_setoption(), as well, since they
both have the potential to send control frames in the middle of call setup.  We
have to wait until we have received a message back from the remote end before
we try to send any more frames.  Otherwise, the remote end will consider it
invalid, and we'll get stuck in an INVAL/VNAK storm.

........

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2008-12-01 17:39:33 +00:00
Michiel van Baak
532d3d5dd2 Merged revisions 159898 via svnmerge from
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  r159898 | mvanbaak | 2008-12-01 15:09:59 +0100 (Mon, 01 Dec 2008) | 11 lines
  
  Merged revisions 159897 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r159897 | mvanbaak | 2008-12-01 15:05:41 +0100 (Mon, 01 Dec 2008) | 4 lines
    
    make manager compile on OpenBSD.
    The last (10th) argument to ast_channel_alloc here should be a pointer
    and NULL is not really a pointer.
  ........
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2008-12-01 16:04:31 +00:00
Russell Bryant
c688256d05 Merged revisions 159911 via svnmerge from
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r159911 | russell | 2008-12-01 08:56:10 -0600 (Mon, 01 Dec 2008) | 10 lines

Merged revisions 159900 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r159900 | russell | 2008-12-01 08:52:56 -0600 (Mon, 01 Dec 2008) | 2 lines

Force a "make clean" to avoid a bizarre build issue ...

........

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2008-12-01 14:56:58 +00:00
Kevin P. Fleming
2eb5c30a3a Merged revisions 159818 via svnmerge from
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........
  r159818 | kpfleming | 2008-11-29 11:57:39 -0600 (Sat, 29 Nov 2008) | 18 lines
  
  incorporates r159808 from branches/1.4:
  ------------------------------------------------------------------------
  r159808 | kpfleming | 2008-11-29 10:58:29 -0600 (Sat, 29 Nov 2008) | 7 lines
  
  update dev-mode compiler flags to match the ones used by default on Ubuntu Intrepid, so all developers will see the same warnings and errors
  
  since this branch already had some printf format attributes, enable checking for them and tag functions that didn't have them
  
  format attributes in a consistent way
  
  
  ------------------------------------------------------------------------
  
  in addition:
  
  move some format attributes from main/utils.c to the header files they belong in, and fix up references to the relevant functions based on new compiler warnings
........


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2008-11-29 18:37:55 +00:00
Mark Michelson
f798f236fb Merged revisions 159554 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r159554 | mmichelson | 2008-11-26 13:57:11 -0600 (Wed, 26 Nov 2008) | 19 lines

Add some necessary hangup commands in the case that forwarding
a call fails

1) Hang up the original destination if the local channel cannot
   be requested.
2) Hang up the local channel (in addition to the original destination)
   if ast_call fails when calling the newly created local channel.

This prevents channels from sticking around forever in the
case of a botched call forward (e.g. to an extension which does not
exist).

(closes issue #13764)
Reported by: davidw
Patches:
      13764_v2.patch uploaded by putnopvut (license 60)
Tested by: putnopvut, davidw


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2008-11-26 19:58:12 +00:00
Kevin P. Fleming
c6bcd9b458 Merged revisions 159534 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r159534 | kpfleming | 2008-11-26 13:08:56 -0600 (Wed, 26 Nov 2008) | 11 lines
  
  Merged revisions 159476 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r159476 | kpfleming | 2008-11-26 12:36:24 -0600 (Wed, 26 Nov 2008) | 7 lines
    
    simplify (and slightly bug-fix) the recent developer-oriented COMPILE_DOUBLE mode
    
    ensure that 'make clean' removes dependency files for .i files that are created in COMPILE_DOUBLE mode
  ........
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2008-11-26 19:18:53 +00:00
Tilghman Lesher
6af6b67bd9 Merged revisions 159475 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r159475 | tilghman | 2008-11-26 12:33:04 -0600 (Wed, 26 Nov 2008) | 7 lines
  
  If the config file does not exist, then the first use crashes Asterisk.
  (closes issue #13848)
   Reported by: klaus3000
   Patches: 
         udptl.c.patch uploaded by eliel (license 64)
   Tested by: blitzrage
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2008-11-26 18:40:28 +00:00
Mark Michelson
bf4326137e Merged revisions 159437 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r159437 | mmichelson | 2008-11-26 08:58:17 -0600 (Wed, 26 Nov 2008) | 10 lines

Don't allow for configuration options to overwrite options
set via channel variables on a reload.

(closes issue #13921)
Reported by: davidw
Patches:
      13921.patch uploaded by putnopvut (license 60)
Tested by: davidw


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2008-11-26 15:01:27 +00:00
Steve Murphy
2ff6ba3c9b Merged revisions 159360 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
r159360 | murf | 2008-11-25 16:03:01 -0700 (Tue, 25 Nov 2008) | 23 lines

Merged revisions 159316 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r159316 | murf | 2008-11-25 15:41:10 -0700 (Tue, 25 Nov 2008) | 15 lines

(closes issue #12694)
Reported by: yraber
Patches:
      12694.2nd.diff uploaded by murf (license 17)
Tested by: murf, laurav

Thanks to file (Joshua Colp) for his IAX fix.

the change to cdr.c allows no-answer to percolate
up into CDR's, and feels like the right place to
locate this fix; if BUSY is done here, no-answer
should be, too.



........

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2008-11-25 23:09:11 +00:00
Mark Michelson
0ecfb8e29e I don't care what anyone says, this change is going into 1.6.0.
Otherwise, the simple act of logging an agent in spams the CLI
with warning messages about failed reads of the alertpipe.



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2008-11-25 22:28:48 +00:00
Tilghman Lesher
1c6382e638 Merged revisions 159247 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r159247 | tilghman | 2008-11-25 15:42:42 -0600 (Tue, 25 Nov 2008) | 21 lines
  
  Merged revisions 159246 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ................
    r159246 | tilghman | 2008-11-25 15:40:28 -0600 (Tue, 25 Nov 2008) | 14 lines
    
    Merged revisions 159245 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.2
    
    ........
      r159245 | tilghman | 2008-11-25 15:37:06 -0600 (Tue, 25 Nov 2008) | 7 lines
      
      Regression fix for last security fix.  Set the iseqno correctly.
      (closes issue #13918)
       Reported by: ffloimair
       Patches: 
             20081119__bug13918.diff.txt uploaded by Corydon76 (license 14)
       Tested by: ffloimair
    ........
  ................
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2008-11-25 21:43:27 +00:00
Terry Wilson
fbcd6b9595 Merged revisions 159093 via svnmerge from
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  r159093 | twilson | 2008-11-25 10:18:53 -0600 (Tue, 25 Nov 2008) | 2 lines
  
  Add missing variable declaration for PPC code
........


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2008-11-25 16:21:25 +00:00
Terry Wilson
8855a46d6e Merged revisions 158992 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r158992 | twilson | 2008-11-24 21:49:30 -0600 (Mon, 24 Nov 2008) | 2 lines
  
  Make chan_usbradio compile under dev mode
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2008-11-25 04:05:31 +00:00
Steve Murphy
3661324455 Merged revisions 158484 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r158484 | murf | 2008-11-21 14:47:16 -0700 (Fri, 21 Nov 2008) | 19 lines

Merged revisions 158483 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r158483 | murf | 2008-11-21 14:19:47 -0700 (Fri, 21 Nov 2008) | 11 lines

(closes issue #13871)
Reported by: mdu113

This one is totally my fault. The code doesn't even
create a bridge CDR if the channel CDR has POST_DISABLED.
I didn't check for that at the end of the bridge.
Fixed with a few small insertions. Tested. Looks
good. No cdr generated, no crash, no unnecc. data
objects created either.


........

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2008-11-21 22:40:28 +00:00
Russell Bryant
c74fff5ebc Merged revisions 158540 via svnmerge from
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r158540 | russell | 2008-11-21 16:12:37 -0600 (Fri, 21 Nov 2008) | 10 lines

Merged revisions 158539 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r158539 | russell | 2008-11-21 16:05:55 -0600 (Fri, 21 Nov 2008) | 2 lines

When compiling with DEBUG_THREADS, report the real file/func/line for ao2_lock/ao2_unlock

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2008-11-21 22:13:42 +00:00
Kevin P. Fleming
15c881c098 Merged revisions 158449 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r158449 | kpfleming | 2008-11-21 14:42:37 -0600 (Fri, 21 Nov 2008) | 3 lines
  
  as suggested by jtodd, document the purposes of the CHANGES and UPGRADE files
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2008-11-21 20:44:23 +00:00
Terry Wilson
ae40f9a6c9 Merged revisions 158374 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r158374 | twilson | 2008-11-21 11:08:16 -0600 (Fri, 21 Nov 2008) | 8 lines
  
  Reloading the config and having no changes still initialized some settings to 0. Initialize settings after doing all of the cfg checks.
  
  (closes issue #13942)
  Reported by: davidw
  Patches: 
        cdr_diff.txt uploaded by otherwiseguy (license 396)
  Tested by: davidw
........


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2008-11-21 17:12:27 +00:00
Mark Michelson
44203a91d2 Merged revisions 158265-158266 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r158265 | mmichelson | 2008-11-20 19:14:20 -0600 (Thu, 20 Nov 2008) | 4 lines

Use some magic constants to get the right size
for this sscanf statement. Thanks Richard!


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r158266 | mmichelson | 2008-11-20 19:22:18 -0600 (Thu, 20 Nov 2008) | 3 lines

Use a more expressive constant for a 64-bit scanned int


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2008-11-21 01:23:18 +00:00
Mark Michelson
28cf66edb8 Merged revisions 158262 via svnmerge from
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r158262 | mmichelson | 2008-11-20 18:59:23 -0600 (Thu, 20 Nov 2008) | 6 lines

Fix the build for 32-bit systems. %lu is only 32-bits
on 32-bit systems, so we need to use %llu instead. Of course
%llu is 128-bits on 64-bit systems, so we have to cast to
unsigned long long. No harm, but it's sure annoying.


........


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2008-11-21 01:00:24 +00:00
Mark Michelson
e2a40b6475 Merged revisions 158230 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r158230 | mmichelson | 2008-11-20 17:12:50 -0600 (Thu, 20 Nov 2008) | 20 lines

Change the remote user agent session version variable
from an int to a uint64_t. This prevents potential comparison
problems from happening if the version string exceeds
INT_MAX. This was an apparent problem for one user who could
not properly place a call on hold since the version in the
SDP of the re-INVITE to place the call on hold greatly 
exceeded INT_MAX.

This also aligns with RFC 2327 better since it recommends
using an NTP timestamp for the version (which is a 
64-bit number).


(closes issue #13531)
Reported by: sgofferj
Patches:
      13531.patch uploaded by putnopvut (license 60)
Tested by: sgofferj


........


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2008-11-20 23:13:33 +00:00
Sean Bright
8bd5236b2f Merged revisions 158188 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r158188 | seanbright | 2008-11-20 14:41:23 -0500 (Thu, 20 Nov 2008) | 10 lines
  
  Fix one case where the application argument was not converted from a pipe to
  a comma.  This was causing problems with switch statements with empty expressions.
  
  (closes issue #13901)
  Reported by: smurfix
  Patches:
        20081118_bug13901.diff uploaded by seanbright (license 71)
  Tested by: seanbright
  Reviewed by: murf
........


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2008-11-20 19:42:31 +00:00
Jason Parker
da39da3e8d Blocked revisions 158061 via svnmerge
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  r158061 | qwell | 2008-11-20 11:37:05 -0600 (Thu, 20 Nov 2008) | 1 line
  
  Whitespace fix
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2008-11-20 17:38:01 +00:00
Kevin P. Fleming
ae3b28aa19 Merged revisions 157974 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r157974 | kpfleming | 2008-11-19 18:08:12 -0600 (Wed, 19 Nov 2008) | 13 lines
  
  Merged revisions 157859 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r157859 | kpfleming | 2008-11-19 15:34:47 -0600 (Wed, 19 Nov 2008) | 7 lines
    
    the gcc optimizer frequently finds broken code (use of uninitalized variables, unreachable code, etc.), which is good. however, developers usually compile with the optimizer turned off, because if they need to debug the resulting code, optimized code makes that process very difficult. this means that we get code changes committed that weren't adequately checked over for these sorts of problems.
    
    with this build system change, if (and only if) --enable-dev-mode was used and DONT_OPTIMIZE is turned on, when a source file is compiled it will actually be preprocessed (into a .i or .ii file), then compiled once with optimization (with the result sent to /dev/null) and again without optimization (but only if the first compile succeeded, of course).
    
    while making these changes, i did some cleanup work in Makefile.rules to move commonly-used combinations of flag variables into their own variables, to make the file easier to read and maintain
  ........
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2008-11-20 00:12:32 +00:00
Kevin P. Fleming
a13f64ef20 Merged revisions 157743 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r157743 | kpfleming | 2008-11-19 07:45:48 -0600 (Wed, 19 Nov 2008) | 1 line
  
  correct small bug introduced during API conversion
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2008-11-19 13:48:47 +00:00
Kevin P. Fleming
ac0c9bbc4d Merged revisions 157706 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r157706 | kpfleming | 2008-11-19 06:42:19 -0600 (Wed, 19 Nov 2008) | 5 lines
  
  make some corrections to the ast_agi_register_multiple(), ast_agi_unregister_multiple() and ast_agi_fdprintf() API calls to be consistent with API guidelines
  
  also, move UPGRADE.txt to UPGRADE-1.6.txt and make the new UPGRADE.txt contain information about upgrading between Asterisk 1.6 releases
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2008-11-19 13:19:49 +00:00
Sean Bright
d974fbe6f7 Merged revisions 157600 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r157600 | seanbright | 2008-11-18 19:27:45 -0500 (Tue, 18 Nov 2008) | 10 lines
  
  Fix a few build problems on Solaris (and check for an md5 utility in
  configure instead of the icky loop I was doing before).
  
  (closes issue #13842)
  Reported by: snuffy
  Patches:
        bug13842_20081106.diff uploaded by snuffy (license 35)
        13842.diff uploaded by seanbright (license 71)
  Tested by: snuffy
........


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2008-11-19 00:33:27 +00:00
Mark Michelson
46ac71b8b4 Merged revisions 157512 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r157512 | mmichelson | 2008-11-18 16:54:08 -0600 (Tue, 18 Nov 2008) | 21 lines

Merged revisions 157503 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r157503 | mmichelson | 2008-11-18 16:47:57 -0600 (Tue, 18 Nov 2008) | 13 lines

Add some missing invite state changes necessary in the sip_write
function. Not setting the invite state correctly on the call was
resulting in the Record application leaving empty files. I also
have updated the doxygen comment next to the declaration of the
INV_EARLY_MEDIA constant to reflect that we also use this state
when we *send* a 18X response to an INVITE.

(closes issue #13878)
Reported by: nahuelgreco
Patches:
      sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco (license 162)
	  Tested by: putnopvut

........

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2008-11-18 22:59:59 +00:00
Mark Michelson
e402952070 Once again, Russell to the rescue. Use the builtin astobj1 lock
of the sip_peer and sip_user instead of adding a new one



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2008-11-18 22:15:31 +00:00
Mark Michelson
35ac1ad736 Merged revisions 157496 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r157496 | mmichelson | 2008-11-18 15:59:24 -0600 (Tue, 18 Nov 2008) | 6 lines

Based on Russell's advice on the asterisk-dev list, I have
changed from using a global lock in update_call_counter to
using the locks within the sip_pvt and sip_peer structures
instead.


........


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2008-11-18 22:11:11 +00:00
Mark Michelson
ddbacb4d33 Merged revisions 157427 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r157427 | mmichelson | 2008-11-18 14:23:58 -0600 (Tue, 18 Nov 2008) | 13 lines

* Add a lock to be used in the update_call_counter function.
* Revert logic to mirror 1.4's in the sense that it will not allow
  the call counter to dip below 0.

These two measures prevent potential races that could cause a SIP peer
to appear to be busy forever.

(closes issue #13668)
Reported by: mjc
Patches:
      hintfix_trunk_rev152649.patch uploaded by wolfelectronic (license 586)


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2008-11-18 20:29:18 +00:00
Mark Michelson
35a7e1deca Merged revisions 157306 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r157306 | mmichelson | 2008-11-18 12:31:08 -0600 (Tue, 18 Nov 2008) | 20 lines

Merged revisions 157305 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r157305 | mmichelson | 2008-11-18 12:25:55 -0600 (Tue, 18 Nov 2008) | 12 lines

Fix a crash in the end_bridge_callback of app_dial and
app_followme which would occur at the end of an attended
transfer. The error occurred because we initially stored
a pointer to an ast_channel which then was hung up due
to a masquerade.

This commit adds a "fixup" callback to the bridge_config
structure to allow for end_bridge_callback_data to be
changed in the case that a new channel pointer is needed
for the end_bridge_callback.


........

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2008-11-18 18:32:18 +00:00
Steve Murphy
a3d874c414 Merged revisions 157302 via svnmerge from
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r157302 | murf | 2008-11-18 11:07:55 -0700 (Tue, 18 Nov 2008) | 18 lines

(closes issue #13420)
Reported by: alex70
Patches:
      13420.13539.patch uploaded by murf (license 17)
Tested by: murf, awk

This fixes two problems: a spurious linefeed insertion
probably left over from pre-precomment times. Only
generated when category had no previous comments.

The other problem: Insertions could get the line-numbering
out of whack and generate negative line numbers, causing 
chunks of line numbers to be emitted, on the scale of the
number of lines up to that point in the file. In such cases,
abort the looping, and all is well.



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2008-11-18 18:10:34 +00:00
Kevin P. Fleming
c1143180c2 Merged revisions 157164 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r157164 | kpfleming | 2008-11-15 20:45:19 +0100 (Sat, 15 Nov 2008) | 13 lines
  
  Merged revisions 157162-157163 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r157162 | kpfleming | 2008-11-15 20:24:24 +0100 (Sat, 15 Nov 2008) | 1 line
    
    dist-clean should remove dependency information files as well
  ........
    r157163 | kpfleming | 2008-11-15 20:31:03 +0100 (Sat, 15 Nov 2008) | 1 line
    
    when an individual directory dist-clean is run, run clean in that directory first, and when running top-level dist-clean, do not run subdirectory clean operations twice
  ........
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2008-11-15 19:47:36 +00:00
Kevin P. Fleming
ef2a405c4a Merged revisions 157105 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r157105 | kpfleming | 2008-11-15 19:00:32 +0100 (Sat, 15 Nov 2008) | 13 lines
  
  major update to doxygen configuration file:
  
  1) update to doxygen 1.5.x style file, as used in trunk
  
  2) tell doxygen where are header files are, so include-file processing can be done
  
  3) make all macros that are used to define variables/functions be expanded, so that doxygen will properly document the resulting variable/function
  
  4) make all macros that are used to provide the contents of a variable (structure) be expanded, so that doxygen will be able to document the resulting fields
  
  5) suppress compiler attributes (__attribute__(xxx)) from being seen by doxygen, so it will properly match up function definition and usage (for an example of th effect of this, look at the doxygen docs for ast_log() from before and afte this commit)
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2008-11-15 18:39:42 +00:00
Tilghman Lesher
ed02eca8fa Merged revisions 156911 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r156911 | tilghman | 2008-11-14 11:02:00 -0600 (Fri, 14 Nov 2008) | 4 lines
  
  Ping is missing the standard double-newline after the event.
  (closes issue #13903)
   Reported by: kebl0155
........


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2008-11-14 17:03:26 +00:00
Mark Michelson
4fbe80cc63 This is the 1.6.0 version of revision 156883 of trunk.
This is different in that it preserves the case-sensitiveness
of processing queues from configuration.

closes issue #13703



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2008-11-14 16:55:53 +00:00
Mark Michelson
bd31630ab0 Merged revisions 156817 via svnmerge from
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r156817 | mmichelson | 2008-11-14 09:20:03 -0600 (Fri, 14 Nov 2008) | 18 lines

Merged revisions 156816 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r156816 | mmichelson | 2008-11-14 09:18:59 -0600 (Fri, 14 Nov 2008) | 10 lines

If the prompt to reenter a voicemail password timed out, it
resulted in the password not being saved, even if the input matched
what you gave when first prompted to enter a new password. This is
because the return value of ast_readstring was checked, but not checked
properly.

This bug was discovered by Jared Smith during an Asterisk training course.
Thanks for reporting it!


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2008-11-14 15:22:06 +00:00
Brandon Kruse
dbc2ed5e21 Update to Coding Guidelines
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2008-11-13 19:26:21 +00:00
Brandon Kruse
f077cc197d Merged revisions 156017 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r156017 | pari | 2008-11-11 17:02:43 -0600 (Tue, 11 Nov 2008) | 5 lines


 Patch by Ryan Brindley -- Make sure that manager refuses any duplicate 'new category' requests in updateconfig

(closes issue #13539)

........


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2008-11-13 19:23:37 +00:00
Steve Murphy
6db3b2ff26 Merged revisions 156299 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r156299 | murf | 2008-11-12 12:47:29 -0700 (Wed, 12 Nov 2008) | 26 lines

Merged revisions 156297 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r156297 | murf | 2008-11-12 12:36:16 -0700 (Wed, 12 Nov 2008) | 18 lines

It turns out that the 0x0XX00 codes being returned for
N, X, and Z are off by one, as per conversation with
jsmith on #asterisk-dev;  he was teaching a class
and disconcerted that this published rule was not
being followed, with patterns _NXX, _[1-8]22 and
_[2-9]22... and NXX was winning, but [1-8] should
have been. 

This change, tested on these 3 patterns now 
picks the proper one.

However, this change may surprise users who
set up dialplans based on previous behavior,
which has been there for what, 2 and half 
years or so now.



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2008-11-12 19:56:34 +00:00
Tilghman Lesher
71cc37f431 Merged revisions 156243 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r156243 | tilghman | 2008-11-12 12:55:18 -0600 (Wed, 12 Nov 2008) | 18 lines
  
  Merged revisions 156229 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r156229 | tilghman | 2008-11-12 12:39:21 -0600 (Wed, 12 Nov 2008) | 11 lines
    
    Revert revision 132506, since it occasionally caused IAX2 HANGUP packets not
    to be sent, and instead, schedule a task to destroy the iax2 pvt structure
    10 seconds later.  This allows the IAX2 HANGUP packet to be queued,
    transmitted, and ACKed before the pvt is destroyed.
    (closes issue #13645)
     Reported by: dzajro
     Patches: 
           20081111__bug13645__3.diff.txt uploaded by Corydon76 (license 14)
     Tested by: vazir
     Reviewed: http://reviewboard.digium.com/r/51/
  ........
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2008-11-12 18:57:06 +00:00
Mark Michelson
221444dd9c Merged revisions 156169 via svnmerge from
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r156169 | mmichelson | 2008-11-12 11:41:56 -0600 (Wed, 12 Nov 2008) | 15 lines

Merged revisions 156167 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r156167 | mmichelson | 2008-11-12 11:38:33 -0600 (Wed, 12 Nov 2008) | 7 lines

When doing some tests, I was having a crash at the end of every call
if an attended transfer occurred during the call. I traced the cause to
the CDR on one of the channels being NULL. murf suggested a check in
the end bridge callback to be sure the CDR is non-NULL before proceeding,
so that's what I'm adding.


........

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2008-11-12 17:47:55 +00:00
Russell Bryant
62eefbec73 Don't blow up if we get NULL when trying to parse out the full name field
(fixed for Jared in the training room)


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2008-11-11 21:28:57 +00:00
Michiel van Baak
1611c5af38 remove prop that shouldn't be here
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2008-11-11 20:04:32 +00:00
Tilghman Lesher
3e62b69b3c Merged revisions 155862 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r155862 | tilghman | 2008-11-10 15:12:28 -0600 (Mon, 10 Nov 2008) | 5 lines
  
  Make documentation of update method match documentation and update update2
  method to match.
  Reported by: atis, via -dev mailing list.
  Fixed by: me
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2008-11-11 19:49:56 +00:00
Tilghman Lesher
57421957b6 Merged revisions 155804 via svnmerge from
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  r155803 | tilghman | 2008-11-10 14:49:59 -0600 (Mon, 10 Nov 2008) | 1 line
  
  I got tired of saying this in every single bugnote referring to this file.
........


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2008-11-10 20:54:15 +00:00
Sean Bright
0591cc3029 Merged revisions 155554 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r155554 | seanbright | 2008-11-08 20:27:00 -0500 (Sat, 08 Nov 2008) | 14 lines

Merged revisions 155553 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r155553 | seanbright | 2008-11-08 20:08:07 -0500 (Sat, 08 Nov 2008) | 6 lines

Use static functions here instead of nested ones.  This requires a small
change to the ast_bridge_config struct as well.  To understand the reason
for this change, see the following post:

    http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html

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2008-11-09 01:34:59 +00:00
Mark Michelson
ac9e7f492a Merged revisions 155467 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r155467 | mmichelson | 2008-11-07 17:41:44 -0600 (Fri, 07 Nov 2008) | 12 lines

Set the invite state to INV_CANCELLED in a place that
makes more sense. Where it was set before, it was impossible
to actually delay sending a CANCEL if we had not yet received
a provisional response to an INVITE.

(closes issue #13626)
Reported by: atis
Patches:
      13626.patch uploaded by putnopvut (license 60)
Tested by: atis


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2008-11-07 23:42:16 +00:00
Mark Michelson
e69fb8b298 Merged revisions 155360 via svnmerge from
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r155360 | mmichelson | 2008-11-07 15:14:49 -0600 (Fri, 07 Nov 2008) | 8 lines

Remove one more instance of the sample configuration
lying about what's possible. The tz cannot be set in a
context like this. It can only be set in the general
section or per-mailbox.

Thanks to sasargen on #asterisk-dev for pointing this out


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2008-11-07 21:15:37 +00:00
Kevin P. Fleming
3577a3c7ca Merged revisions 155121 via svnmerge from
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  r155121 | kpfleming | 2008-11-06 16:49:19 -0600 (Thu, 06 Nov 2008) | 3 lines
  
  don't blindly assume that Darwin and Cygwin need GLOB_ABORTED defined; only define it if it is not already defined
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2008-11-06 22:50:27 +00:00
Mark Michelson
d5e144b150 Merged revisions 155012 via svnmerge from
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r155012 | mmichelson | 2008-11-06 13:46:53 -0600 (Thu, 06 Nov 2008) | 16 lines

Merged revisions 155011 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r155011 | mmichelson | 2008-11-06 13:45:52 -0600 (Thu, 06 Nov 2008) | 8 lines

The documentation listed the ability to set 'maxmsg' per
context. The truth is that you can only set this in the general section
or per mailbox. Thus I am updating the sample config file to be more
accurate.

Thanks to sasargen on IRC for bringing up this issue.


........

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2008-11-06 19:47:45 +00:00
Tilghman Lesher
975b8d582e Recorded merge of revisions 154072 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r154072 | tilghman | 2008-11-03 16:28:12 -0600 (Mon, 03 Nov 2008) | 12 lines
  
  Merged revisions 154066 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r154066 | tilghman | 2008-11-03 16:27:10 -0600 (Mon, 03 Nov 2008) | 5 lines
    
    Attempting to expunge a mailbox when the mailstream is NULL will crash Asterisk.
    (Closes issue #13829)
    Reported by: jaroth
    Patch by: me (modified jaroth's patch)
  ........
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2008-11-03 22:30:13 +00:00
Tilghman Lesher
c8ebf2045a Merged revisions 154060 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r154060 | tilghman | 2008-11-03 15:48:21 -0600 (Mon, 03 Nov 2008) | 3 lines
  
  Remove the potential for a division by zero error.
  (Closes issue #13810)
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2008-11-03 22:04:01 +00:00
Kevin P. Fleming
090cd16d68 record revisions that were manually merged
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@153746 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-03 00:53:10 +00:00
Kevin P. Fleming
ede179faec Merge revision 153709 from trunk
------------------------------------------------------------------------
r153709 | kpfleming | 2008-11-02 17:34:39 -0600 (Sun, 02 Nov 2008) | 3 lines

instead of trying to forcibly load res_agi when app_stack is loaded (even if the administrator didn't want it loaded), use GCC weak symbols to determine whether it was loaded already or not; if it was loaded, then use it.


------------------------------------------------------------------------



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2008-11-03 00:52:05 +00:00
Kevin P. Fleming
fa635ea4b0 port gcc 4.3.x warning fixes from trunk to this branch
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@153743 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-03 00:39:04 +00:00
Terry Wilson
e23be17786 Merged revisions 153181 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r153181 | twilson | 2008-10-31 13:55:33 -0500 (Fri, 31 Oct 2008) | 5 lines
  
  Recent CDR fixes moved execution of the 'h' exten into the bridging code, so variables that were set after ast_bridge_call was called would not show up in the 'h' exten.  Added a callback function to handle setting variables, etc. from w/in the bridging code.  Calls back into a nested function within the function calling ast_bridge_call
  
  (closes issue #13793)
  Reported by: greenfieldtech
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2008-10-31 21:49:23 +00:00
Sean Bright
798db6fe3b Merged revisions 152993 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r152993 | seanbright | 2008-10-30 16:59:17 -0400 (Thu, 30 Oct 2008) | 10 lines

Merged revisions 152992 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r152992 | seanbright | 2008-10-30 16:58:24 -0400 (Thu, 30 Oct 2008) | 2 lines

The -I argument to aclocal needs a space before the include directory name.

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2008-10-30 21:00:13 +00:00
Kevin P. Fleming
3a6940fe60 Merged revisions 152812 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r152812 | kpfleming | 2008-10-30 11:54:29 -0500 (Thu, 30 Oct 2008) | 9 lines
  
  Merged revisions 152811 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r152811 | kpfleming | 2008-10-30 11:53:48 -0500 (Thu, 30 Oct 2008) | 3 lines
    
    instead of comparing the string pointer to 0, let's compare the value that was actually parsed out of the string (found by sparse)
  ........
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2008-10-30 16:54:54 +00:00
Tilghman Lesher
68ccb25f93 Merged revisions 152765 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r152765 | tilghman | 2008-10-29 23:26:34 -0500 (Wed, 29 Oct 2008) | 5 lines
  
  Set up an example stdexten that preserves the original context and extension in
  the CDR.
  (Related to issue #13799)
   Reported by: davidw
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2008-10-30 04:28:06 +00:00
Mark Michelson
84dcf8e6b1 Merged revisions 152646 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r152646 | mmichelson | 2008-10-29 15:53:53 -0500 (Wed, 29 Oct 2008) | 9 lines

If there was no named defined in a voicemail.conf mailbox
entry, then app_directory would crash when attempting to
read that entry from the file. We now check for the NULL
or empty string properly so that there will be no crash.

(closes issue #13804)
Reported by: bluecrow76


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2008-10-29 20:54:49 +00:00
Terry Wilson
969bef1dea Small modification to putnopvut's patch to fix this issue. Thanks for all the help, putnopvut!
(closes issue #12884)
	Reported by: bcnit
	Patches: 
	      12884v4-1.6.0-branch.patch uploaded by otherwiseguy (license 396)
	Tested by: otherwiseguy


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@152644 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-29 20:13:22 +00:00
Tilghman Lesher
01cea8d69d Merged revisions 152442 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r152442 | tilghman | 2008-10-28 16:38:26 -0500 (Tue, 28 Oct 2008) | 7 lines
  
  Only re-add the io port if it was closed, otherwise reload causes a memory
  leak.
  (closes issue #13785)
   Reported by: eliel
   Patches: 
         chan_mgcp.c.patch uploaded by eliel (license 64)
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2008-10-28 21:39:35 +00:00
Tilghman Lesher
8bcea43631 Merged revisions 152134 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r152134 | tilghman | 2008-10-27 11:24:11 -0500 (Mon, 27 Oct 2008) | 4 lines
  
  Oops, only delete the ARG variables once upon release.  The following section
  would have removed them again (removing variables from 2 stack frames, instead
  of just one).
........


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2008-10-27 16:33:16 +00:00
Sean Bright
85b4dc5b11 Merged revisions 152060 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r152060 | seanbright | 2008-10-26 16:25:08 -0400 (Sun, 26 Oct 2008) | 15 lines

Merged revisions 152059 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r152059 | seanbright | 2008-10-26 16:23:36 -0400 (Sun, 26 Oct 2008) | 7 lines

Since passing \0 as the second argument to strchr is valid (and will
match the trailing \0 of a string) we need to check that first, otherwise
we end up with incorrect results.  Fix suggested by reporter.

(closes issue #13787)
Reported by: meitinger

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2008-10-26 20:26:18 +00:00
Russell Bryant
4ebe1cfd46 Fix some memory leaks. These issues are 1.6.0 specific.
- Freeing the peer got accidentally removed from the peer's destructor.  It is
   still needed for astobj, but not for astobj2.
 - Fix some places that called find_user or find_peer, but did not release the
   reference that was returned.

(closes issue #13331)
Reported by: sergee
Patches:
      chan_sip-3leaks-16-r151244.diff uploaded by sergee (license 138)
Tested by: sergee


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@151765 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-23 16:12:01 +00:00
Kevin P. Fleming
ff6a34c1c5 Merged revisions 151242-151243 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r151242 | kpfleming | 2008-10-20 07:59:04 +0300 (Mon, 20 Oct 2008) | 9 lines
  
  Merged revisions 151240 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r151240 | kpfleming | 2008-10-20 07:45:56 +0300 (Mon, 20 Oct 2008) | 3 lines
    
    break up acinclude.m4 into individual files, which will make it easier to maintain, easier to add new macros (less patching) and will ease maintenance of these macros across Asterisk branches
  ........
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  r151243 | kpfleming | 2008-10-20 08:00:56 +0300 (Mon, 20 Oct 2008) | 9 lines
  
  Merged revisions 151241 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r151241 | kpfleming | 2008-10-20 07:57:33 +0300 (Mon, 20 Oct 2008) | 2 lines
    
    rename this macro to properly reflect what it does
  ........
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2008-10-20 05:03:58 +00:00
BJ Weschke
b15c8d915a Merged revisions 150817 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r150817 | bweschke | 2008-10-17 22:18:33 -0400 (Fri, 17 Oct 2008) | 8 lines
  
   Using the GetVar handler in AMI is potentially dangerous (insta-crash [tm]) when you use a dialplan function that requires a channel and then you don't provide one or provide an invalid one in the Channel: parameter. We'll handle this situation exactly the same way it was handled in pbx.c back on r61766.
  We'll create a bogus channel for the function call and destroy it when we're done. If we have trouble allocating the bogus channel then we're not going to try executing the function call at all and run the risk of crashing.
  
  (closes issue #13715)
  reported by: makoto
  patch by: bweschke
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2008-10-18 02:35:47 +00:00
Mark Michelson
6e93187aa1 Instead of merging commit 150307 to 1.6.0, I had
meant to block it in 1.6.1...time to go home :)



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@150313 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-17 00:19:42 +00:00
Mark Michelson
258c4d4932 Merged revisions 150307 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r150307 | mmichelson | 2008-10-16 19:13:35 -0500 (Thu, 16 Oct 2008) | 14 lines

After a long discussion on #asterisk-bugs, it seems kind of
odd that a channel would be named after the port on which it
came in on. For endpoints that always include ":5060" as part
of the From: header, it will mean that you have a ton of
channels with names like "SIP/5060-3ea38a8b."

I am boldly moving forward with this change in trunk, but I'm
not touching other branches with this one since this definitely
would qualify as a behavior change. If there is a problem with
this commit, and I haven't seen the obvious reason why you'd want
to name the channel after the port from which the call originated,
then please feel free to revert this


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2008-10-17 00:14:17 +00:00
Richard Mudgett
757367c8d8 Merged revisions 150125 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r150125 | rmudgett | 2008-10-16 11:04:45 -0500 (Thu, 16 Oct 2008) | 9 lines
  
  Merged revisions 150124 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r150124 | rmudgett | 2008-10-16 10:56:06 -0500 (Thu, 16 Oct 2008) | 1 line
    
    Fix memory leak found by customer
  ........
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2008-10-16 16:10:28 +00:00
BJ Weschke
a301889274 Merged revisions 149756 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r149756 | bweschke | 2008-10-15 16:14:20 -0400 (Wed, 15 Oct 2008) | 10 lines
  
  Merged revisions 149683 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r149683 | bweschke | 2008-10-15 14:28:54 -0400 (Wed, 15 Oct 2008) | 4 lines
    
     An update to the documentation/example of agents.conf.sample with the correct parameter for this feature as defined in chan_agent.c
     (closes issue #13709)
  ........
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2008-10-15 20:18:44 +00:00
Kevin P. Fleming
962a134bcf Merged revisions 149487 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r149487 | kpfleming | 2008-10-15 13:26:36 +0200 (Wed, 15 Oct 2008) | 9 lines
  
  Merged revisions 149452 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r149452 | kpfleming | 2008-10-15 12:30:40 +0200 (Wed, 15 Oct 2008) | 3 lines
    
    fix some problems when parsing SIP messages that have the maximum number of headers or body lines that we support
  ........
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2008-10-15 11:29:25 +00:00
Mark Michelson
4e93c2a778 Merged revisions 148913 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r148913 | mmichelson | 2008-10-14 12:38:06 -0500 (Tue, 14 Oct 2008) | 17 lines

Merged revisions 148912 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r148912 | mmichelson | 2008-10-14 12:33:38 -0500 (Tue, 14 Oct 2008) | 9 lines

Deadlock prevention in chan_local.

(closes issue #13676)
Reported by: tacvbo
Patches:
      13676.patch uploaded by putnopvut (license 60)
Tested by: tacvbo


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2008-10-14 17:39:04 +00:00
Kevin P. Fleming
f9f88ad2d8 Merged revisions 148738 via svnmerge from
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  r148738 | kpfleming | 2008-10-14 12:33:14 +0200 (Tue, 14 Oct 2008) | 9 lines
  
  Merged revisions 148736 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r148736 | kpfleming | 2008-10-14 12:30:54 +0200 (Tue, 14 Oct 2008) | 3 lines
    
    on Ubuntu (at least), recent versions of ld in binutils delete all debugging symbols when -x is supplied; since the reasons why -x is being passed are lost in the mists of time, remove it so debugging will work properly
  ........
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2008-10-14 10:34:14 +00:00
Kevin P. Fleming
158d6c4b64 Merged revisions 148612 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r148612 | kpfleming | 2008-10-14 03:06:45 -0500 (Tue, 14 Oct 2008) | 9 lines
  
  Merged revisions 148611 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r148611 | kpfleming | 2008-10-14 02:54:41 -0500 (Tue, 14 Oct 2008) | 3 lines
    
    it would be nice if this message printing code had actually been tested before it was committed...
  ........
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2008-10-14 08:08:17 +00:00
Steve Murphy
3a39286358 Blocked revisions 148519 via svnmerge
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r148519 | murf | 2008-10-13 11:14:38 -0600 (Mon, 13 Oct 2008) | 15 lines

Hmmm. Nobody (but me) is interested in seeing 
the trie info when they do 'dialplan show ...'
(even with debug set to non-zero); so I set up a 
   'dialplan debug [context]' cli command instead, 
to explicitly show just the trie info.  I even
added an extension_exists() call to make sure the
trie info is built. I moved the explanatory header
to above the extension loop to ensure it only prints
once. And it will do this now, whether debug is set
or not.

I removed the trie printing from the 'dialplan show' 
command entirely. 


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2008-10-13 17:47:51 +00:00
Mark Michelson
2e7e3ee6b3 Merged revisions 148373 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r148373 | mmichelson | 2008-10-10 16:18:10 -0500 (Fri, 10 Oct 2008) | 8 lines

Make sure that the inUse and inRinging fields for 
a sip peer cannot go below zero. This is a regression
from 1.4 and so it will be applied to 1.6.0 as well.

(closes issue #13668)
Reported by: mjc


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2008-10-10 21:18:43 +00:00
Sean Bright
eb87d6dac7 Merged revisions 148200 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r148200 | seanbright | 2008-10-09 20:42:13 -0400 (Thu, 09 Oct 2008) | 12 lines

Don't include logger.h in asterisk.h by default as it is causing problems building
app_voicemail.  Instead, include it where it is needed.  This turned out to be a
relatively minor issue because other headers include logger.h as well.

Need to test -addons before merging this back to 1.6.0.

(closes issue #13605)
Reported by: tomo1657
Patches: 
      13605_seanbright.diff uploaded by seanbright (license 71)
Tested by: mmichelson

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2008-10-10 01:25:31 +00:00
Sean Bright
2b03afb74b Somehow we got conflict markers checked in! Might need a 1.6.0.1 sooner than we'd like.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@148201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-10 01:21:47 +00:00
Mark Michelson
60399ff4ac Merged revisions 148144 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r148144 | mmichelson | 2008-10-09 18:30:47 -0500 (Thu, 09 Oct 2008) | 10 lines

Read the callerid in the correct order and make sure to
read the Urgent flag value from the IMAP headers.

(closes issue #13652)
Reported by: jaroth
Patches:
      imapheaders.patch uploaded by jaroth (license 50)



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2008-10-09 23:31:27 +00:00
Tilghman Lesher
e0f75e891e Merged revisions 148120 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r148120 | tilghman | 2008-10-09 18:25:53 -0500 (Thu, 09 Oct 2008) | 6 lines
  
  Fix example schema
  (closes issue #12860)
   Reported by: flyn
   Patches: 
         res_ldap.conf.patch uploaded by flyn (license 503)
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2008-10-09 23:26:42 +00:00
Michiel van Baak
de553c362d Merged revisions 147899 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r147899 | mvanbaak | 2008-10-09 19:48:53 +0200 (Thu, 09 Oct 2008) | 5 lines
  
  only include this for OpenBSD. At least FreeBSD is borked when including it
  
  (closes issue #13649)
  Reported by: ys
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2008-10-09 17:51:25 +00:00
Tilghman Lesher
2934507592 Merged revisions 147896 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r147896 | tilghman | 2008-10-09 12:46:15 -0500 (Thu, 09 Oct 2008) | 4 lines
  
  Remove "second form" of extensions, as it no longer applies.  Also, cleanup
  the grammar, formatting, and introduce several clarifications to the text.
  (Closes issue #13654)
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2008-10-09 17:47:03 +00:00
Steve Murphy
062d3041fe Merged revisions 147807 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
r147807 | murf | 2008-10-09 08:17:33 -0600 (Thu, 09 Oct 2008) | 15 lines

(closes issue #13557)
Reported by: nickpeirson
Patches:
      pbx.c.patch uploaded by nickpeirson (license 579)
      replace_bzero+bcopy.patch uploaded by nickpeirson (license 579)
Tested by: nickpeirson, murf

1. replaced all refs to bzero and bcopy to memset and memmove instead.
2. added a note to the CODING-GUIDELINES
3. add two macros to asterisk.h to prevent bzero, bcopy from creeping
   back into the source
4. removed bzero from configure, configure.ac, autoconfig.h.in



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2008-10-09 14:56:51 +00:00
Russell Bryant
506f7e187a Remove the sample configuration for configuration sections in chan_dahdi.conf.
This code was not merged into 1.6.0.
Reported by: angler
(closes AST-119)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@147458 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-08 12:16:30 +00:00
Sean Bright
d3edd930a9 Merged revisions 147194 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r147194 | seanbright | 2008-10-07 12:52:02 -0400 (Tue, 07 Oct 2008) | 10 lines

Merged revisions 147193 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r147193 | seanbright | 2008-10-07 12:48:30 -0400 (Tue, 07 Oct 2008) | 2 lines

Make 'imapsecret' an alias to 'imappassword' in voicemail.conf.

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2008-10-07 16:53:40 +00:00
Sean Bright
4d5405c74d Merged revisions 147050 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r147050 | seanbright | 2008-10-07 08:01:36 -0400 (Tue, 07 Oct 2008) | 8 lines

Make sure to compare the correct number of characters when special-casing
our DAHDI operator mode stuff.  Technically, it would work fine, as 'DAH'
is currently unique amongst our channel technologies, but as Jared points
out:

  <@jsmith> Sure... as long as the technology starts whith DAH.... but
            it could be DAHDOO!

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2008-10-07 12:03:07 +00:00
Tilghman Lesher
cdb76a9f1a Merged revisions 146928 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r146928 | tilghman | 2008-10-06 18:21:02 -0500 (Mon, 06 Oct 2008) | 3 lines
  
  Update documentation; AST_THREADSTORAGE() in trunk only takes a single
  argument.
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2008-10-06 23:22:19 +00:00
Michiel van Baak
456b36a4c1 Blocked revisions 146925 via svnmerge
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  r146925 | mvanbaak | 2008-10-07 01:14:33 +0200 (Tue, 07 Oct 2008) | 8 lines
  
  All ODBC parts can now use either unixodbc or iodbc.
  This allows for the ODBC parts to work on OpenBSD as well.
  
  99.99% of the work is done by seanbright (bow, bow) and I actually
  did nothing but test and yell at him that it still didn't work :)
  
  Thanks for helping out !
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2008-10-06 23:17:14 +00:00
Jeff Peeler
69ffdd5f69 Merged revisions 146920 via svnmerge from
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r146920 | jpeeler | 2008-10-06 17:59:58 -0500 (Mon, 06 Oct 2008) | 2 lines

Mvanbaak said this was needed to compile on OpenBSD, so put it in the OpenBSD section.

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2008-10-06 23:03:17 +00:00
Michiel van Baak
49fd015123 Merged revisions 146807 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r146807 | mvanbaak | 2008-10-06 23:18:13 +0200 (Mon, 06 Oct 2008) | 2 lines
  
  make aescrypt.c compile on OpenBSD again
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2008-10-06 21:47:56 +00:00
Tilghman Lesher
45c2df8f99 Blocked revisions 146802 via svnmerge
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  r146802 | tilghman | 2008-10-06 16:09:05 -0500 (Mon, 06 Oct 2008) | 15 lines
  
  Merged revisions 146799 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r146799 | tilghman | 2008-10-06 15:52:04 -0500 (Mon, 06 Oct 2008) | 8 lines
    
    Dialplan functions should not actually return 0, unless they have modified the
    workspace.  To signal an error (and no change to the workspace), -1 should be
    returned instead.
    (closes issue #13340)
     Reported by: kryptolus
     Patches: 
           20080827__bug13340__2.diff.txt uploaded by Corydon76 (license 14)
  ........
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2008-10-06 21:30:42 +00:00
Sean Bright
882bad399e Blocked revisions 146738 via svnmerge
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r146738 | seanbright | 2008-10-06 13:32:02 -0400 (Mon, 06 Oct 2008) | 1 line

Pretty-print a couple configure options
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2008-10-06 17:32:55 +00:00
Tilghman Lesher
eb9fe05fa0 Blocked revisions 146713 via svnmerge
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  r146713 | tilghman | 2008-10-06 11:52:14 -0500 (Mon, 06 Oct 2008) | 16 lines
  
  Merged revisions 146711 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r146711 | tilghman | 2008-10-06 11:51:21 -0500 (Mon, 06 Oct 2008) | 9 lines
    
    Check whether an extension exists in the _call method, rather than the _alloc
    method, because we need to evaluate the callerid (since that data affects
    whether an extension exists).
    (closes issue #13343)
     Reported by: efutch
     Patches: 
           20080915__bug13343.diff.txt uploaded by Corydon76 (license 14)
     Tested by: efutch
  ........
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2008-10-06 16:53:13 +00:00
Mark Michelson
ddc77036bd Blocked revisions 146640 via svnmerge
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r146640 | mmichelson | 2008-10-06 10:29:56 -0500 (Mon, 06 Oct 2008) | 15 lines

This commit introduces a change to how the "joinempty"
and "leavewhenempty" options are configured in queues.conf.

Instead of using vague terms like "yes," "no," "loose," and
"strict," we now accept a comma-separated list of values
to determine when to consider a member available.

Extended details can be found in the queues.conf.sample
file. Note also that the above four referenced values are
still accepted for backwards-compatibility, but are mapped
internally to the new method of representing the option.

AST-105


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2008-10-06 15:36:28 +00:00
Sean Bright
c82ebdb472 Blocked revisions 146597 via svnmerge
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r146597 | seanbright | 2008-10-05 20:36:37 -0400 (Sun, 05 Oct 2008) | 2 lines

Make NOISY_BUILD work for the calls to cp in utils/Makefile

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2008-10-06 00:37:11 +00:00
Sean Bright
a237e8a708 Merged revisions 146555 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r146555 | seanbright | 2008-10-05 20:21:49 -0400 (Sun, 05 Oct 2008) | 1 line

Quote arguments to cp so we can handle spaces in our paths.
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2008-10-06 00:23:03 +00:00
Russell Bryant
23b7ee2026 Merged revisions 146514 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r146514 | russell | 2008-10-05 17:11:30 -0500 (Sun, 05 Oct 2008) | 2 lines

Make this build on my mac.

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2008-10-05 23:35:59 +00:00
Jason Parker
521e821101 Blocked revisions 146449 via svnmerge
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  r146449 | qwell | 2008-10-05 16:21:03 -0500 (Sun, 05 Oct 2008) | 9 lines
  
  Recorded merge of revisions 146448 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r146448 | qwell | 2008-10-05 16:17:44 -0500 (Sun, 05 Oct 2008) | 1 line
    
    Fix silly formatting.
  ........
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2008-10-05 21:23:36 +00:00
Sean Bright
b016fa3371 Blocked revisions 146359 via svnmerge
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r146359 | seanbright | 2008-10-04 12:20:31 -0400 (Sat, 04 Oct 2008) | 10 lines

Fix a bug with the last item in CLI history getting duplicated when
read from the .asterisk_history file (and subsequently being duplicated
when written).  We weren't checking the result of fgets() which meant
that we read the same line twice before feof() actually returned non-
zero.

Also, stop writing out an extra blank line between each item in the
history file, fix a minor off-by-one error, and use symbolic constants
rather than a hardcoded integer.

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2008-10-04 16:21:15 +00:00
Sean Bright
0b5d83c47c Blocked revisions 146312 via svnmerge
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r146312 | seanbright | 2008-10-03 21:54:44 -0400 (Fri, 03 Oct 2008) | 8 lines

Add ability to remotely reboot snom phones.  Also cleaned up and
reorganized sip_notify.conf.sample a bit as well.  Tested snom
reboot on snom 360 and verified snom-check-cfg worked as well.

(closes issue #13601)
Reported by: mjc
Tested by: seanbright

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2008-10-04 01:56:34 +00:00
Sean Bright
db4272571b Blocked revisions 146245 via svnmerge
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r146245 | seanbright | 2008-10-03 18:54:57 -0400 (Fri, 03 Oct 2008) | 13 lines

Blocked revisions 146244 via svnmerge

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r146244 | seanbright | 2008-10-03 18:51:22 -0400 (Fri, 03 Oct 2008) | 6 lines

Change some preprocessor macros to struct definitions so that we get
app_rpt to build with DAHDI.

(closes issue #13576)
Reported by: blitzrage

........

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2008-10-03 22:55:35 +00:00
Sean Bright
28c515cac5 Merged revisions 146198 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r146198 | seanbright | 2008-10-03 18:10:18 -0400 (Fri, 03 Oct 2008) | 7 lines

Resolve a subtle bug where we would never successfully be able to get
the first item in the CLI entry list.  This was preventing '!' from
showing up in either 'help' or in tab completion.

(closes issue #13578)
Reported by: mvanbaak

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2008-10-03 22:12:02 +00:00
Tilghman Lesher
f0b173a9a6 Blocked revisions 146081 via svnmerge
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  r146081 | tilghman | 2008-10-03 13:30:39 -0500 (Fri, 03 Oct 2008) | 2 lines
  
  document meetme schedule changes (related to issue #11040)
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2008-10-03 18:31:41 +00:00
Russell Bryant
ce39baaac3 Merged revisions 145962 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r145962 | russell | 2008-10-02 14:30:45 -0500 (Thu, 02 Oct 2008) | 2 lines

The 'P' command for ExternalIVR was also added in 1.6.0

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2008-10-02 19:30:59 +00:00
Russell Bryant
4ccac83066 Blocked revisions 145959 via svnmerge
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r145959 | russell | 2008-10-02 14:27:37 -0500 (Thu, 02 Oct 2008) | 2 lines

TCP support for ExternalIVR went in to 1.6.1, not 1.6.0

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2008-10-02 19:29:01 +00:00
Michiel van Baak
0f4053a895 Blocked revisions 145915 via svnmerge
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r145915 | mvanbaak | 2008-10-02 20:02:26 +0200 (Thu, 02 Oct 2008) | 6 lines

fix the 'meetme list', 'meetme list concise', 'meetme list $confno' and 'meetme list $confno concise' CLI commands

(closes issue #13586)
Reported by: john8675309
Help and feedback from eliel, thanks!

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2008-10-02 18:03:25 +00:00
Tilghman Lesher
47f29335e9 Blocked revisions 145846 via svnmerge
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  r145846 | tilghman | 2008-10-02 12:16:54 -0500 (Thu, 02 Oct 2008) | 2 lines
  
  Permit the syntax and synopsis fields to be set (for func_odbc).
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2008-10-02 17:17:28 +00:00
Michiel van Baak
56760efcd3 Blocked revisions 145842 via svnmerge
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r145842 | mvanbaak | 2008-10-02 18:42:30 +0200 (Thu, 02 Oct 2008) | 2 lines

make this compile under devmode again

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2008-10-02 16:44:56 +00:00
Sean Bright
f2e9d4db57 Blocked revisions 145771 via svnmerge
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r145771 | seanbright | 2008-10-02 11:28:48 -0400 (Thu, 02 Oct 2008) | 1 line

This is much cleaner, methinks.
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2008-10-02 15:29:31 +00:00
Tilghman Lesher
45a68354bb Blocked revisions 145752 via svnmerge
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  r145752 | tilghman | 2008-10-02 10:17:16 -0500 (Thu, 02 Oct 2008) | 10 lines
  
  Merged revisions 145751 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r145751 | tilghman | 2008-10-02 10:13:21 -0500 (Thu, 02 Oct 2008) | 3 lines
    
    Some sanity checks that may have led to prior crashes, found by codefreeze-lap
    (murf) on IRC.  Also some cleanup of incorrectly-used constants.
  ........
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2008-10-02 15:18:45 +00:00
Russell Bryant
7e6ff1dac8 Merged revisions 145692 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r145692 | seanbright | 2008-10-01 18:48:16 -0500 (Wed, 01 Oct 2008) | 7 lines

Try a test compile using the GMime library.  Some distros install gmime-config
in the base package instead of the -devel package.  Now we print a notice and
disable GMime support instead of bombing during the main compilation.

(closes issue #13583)
Reported by: arkadia

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2008-10-02 00:23:09 +00:00
Russell Bryant
3059d69eef unblock 145692
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@145696 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-02 00:21:18 +00:00
Sean Bright
329ab9cc98 Blocked revisions 145692 via svnmerge
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r145692 | seanbright | 2008-10-01 19:48:16 -0400 (Wed, 01 Oct 2008) | 7 lines

Try a test compile using the GMime library.  Some distros install gmime-config
in the base package instead of the -devel package.  Now we print a notice and
disable GMime support instead of bombing during the main compilation.

(closes issue #13583)
Reported by: arkadia

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2008-10-01 23:53:07 +00:00
Tilghman Lesher
b14ec0380e Blocked revisions 145649 via svnmerge
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  r145649 | tilghman | 2008-10-01 18:02:25 -0500 (Wed, 01 Oct 2008) | 8 lines
  
  Add schedule extensions to app_meetme.  In addition, the reporter found a
  problem within strptime(3), which we are correcting here with ast_strptime().
  (closes issue #11040)
   Reported by: DEA
   Patches: 
         20080910__bug11040.diff.txt uploaded by Corydon76 (license 14)
   Tested by: DEA
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2008-10-01 23:03:11 +00:00
Mark Michelson
4a2c9d25ba Merged revisions 145606 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r145606 | mmichelson | 2008-10-01 17:23:50 -0500 (Wed, 01 Oct 2008) | 11 lines

Okay, this should really do it now. While I did manage
to fix blind transfers with my last commit here, I also
caused an unwanted side-effect. That is, only the first
priority of the 'h' extension would be executed when
a blind transfer occurred instead of all priorities.

Essentially, my last commit corrected the return value
of ast_bridge_call. However, the implementation still
was not 100% correct. Now it is.


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2008-10-01 22:24:16 +00:00
Mark Michelson
2b6112542e Merged revisions 145579 via svnmerge from
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r145579 | mmichelson | 2008-10-01 16:33:11 -0500 (Wed, 01 Oct 2008) | 4 lines

if (!(x) == 0) is the same as
if (x).


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2008-10-01 21:33:55 +00:00
Mark Michelson
64773d186f Merged revisions 145553 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r145553 | mmichelson | 2008-10-01 16:06:26 -0500 (Wed, 01 Oct 2008) | 13 lines

The logic surrounding the return value of ast_spawn_extension
within ast_bridge_call was reversed.

This problem was observed when a blind transfer placed from
the callee channel of a test call failed.

While the problem I am solving here is exactly the same
as what was reported in issue #13584, the difference is
that this fix I am applying is trunk-only. Issue #13584
was reported against the 1.4 branch, and my tests
of 1.4's blind transfers appear to work fine.


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2008-10-01 21:19:46 +00:00
Leif Madsen
107d16c87c Merged revisions 145487 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r145487 | lmadsen | 2008-10-01 13:26:20 -0400 (Wed, 01 Oct 2008) | 12 lines
  
  Merged revisions 145479 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r145479 | lmadsen | 2008-10-01 13:18:30 -0400 (Wed, 01 Oct 2008) | 6 lines
    
    Update the realtime_pgsql.sql script to create the setinterfacevar column.
    
    (closes issue #13549)
    Reported by: fiddur
  ........
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2008-10-01 17:31:12 +00:00
Tilghman Lesher
9f9e461e48 Merged revisions 145428 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r145428 | tilghman | 2008-10-01 10:44:06 -0500 (Wed, 01 Oct 2008) | 7 lines
  
  Initializing buffer prevents a segfault when arguments are incomplete.
  (closes issue #13471)
   Reported by: alecdavis
   Patches: 
         20080916__bug13471.diff.txt uploaded by Corydon76 (license 14)
   Tested by: alecdavis
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Mark Michelson
856f327b9b Blocked revisions 145381 via svnmerge
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r145381 | mmichelson | 2008-10-01 09:44:12 -0500 (Wed, 01 Oct 2008) | 7 lines

Too many times have I mistyped the word
'install' as 'isntall'

Now this typo shall no longer be a problem since
'make isntall' just builds the 'install' target.


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2008-10-01 14:46:40 +00:00
Jeff Peeler
5b1e55af1e Merged revisions 145249 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
r145249 | jpeeler | 2008-09-30 17:21:19 -0500 (Tue, 30 Sep 2008) | 6 lines

(closes issue #13337)
Reported by: pj
Tested by: pj

Set transport to SIP_TRANSPORT_UDP mode if not specified which fixes calls to get_transport returning UNKNOWN.

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2008-09-30 22:25:42 +00:00
Russell Bryant
80d99c22cb Blocked revisions 145226 via svnmerge
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r145226 | russell | 2008-09-30 16:32:53 -0500 (Tue, 30 Sep 2008) | 16 lines

Add support for call pickup on Snom phones.  Asterisk now includes a magic
call-id in the dialog-info event package used with extension state subscriptions
on Snom phones.  Then, when the phone sends an INVITE with Replaces for the
special callid, Asterisk will perform a pickup on the extension that was
subscribed to.

The original code on this issue was submitted by xylome.  However, contributions
have been made by (at least) mgernoth and pkempgen.  The final patch was written
by seanbright, and includes the necessary logic to allow this work in a
technology independent way.

(closes issue #5014)
Reported by: xylome
Patches:
      issue5014-trunk.diff uploaded by seanbright (license 71)

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2008-09-30 21:33:27 +00:00
Tilghman Lesher
e747412f1f Merged revisions 145076 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r145076 | tilghman | 2008-09-28 16:39:07 -0500 (Sun, 28 Sep 2008) | 4 lines
  
  Change several improper "sizeof" to "strlen", as sizeof in that context would
  incorrectly use the size of a pointer, rather than the length of a string.
  (Closes issue #13574)
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2008-09-29 00:13:00 +00:00
Kevin P. Fleming
ae6e72ad45 Blocked revisions 145027 via svnmerge
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  r145027 | kpfleming | 2008-09-28 10:08:12 -0700 (Sun, 28 Sep 2008) | 1 line
  
  rename chandup() and clarify its usage
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2008-09-28 17:09:13 +00:00
Kevin P. Fleming
ccc3bcfb67 Merged revisions 144949-144951 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r144949 | kpfleming | 2008-09-27 10:52:56 -0500 (Sat, 27 Sep 2008) | 17 lines
  
  Merged revisions 144924-144925 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r144924 | kpfleming | 2008-09-27 10:00:48 -0500 (Sat, 27 Sep 2008) | 6 lines
    
    improve header inclusion process in a few small ways:
    
      - it is no longer necessary to forcibly include asterisk/autoconfig.h; every module already includes asterisk.h as its first header (even before system headers), which serves the same purpose
      - astmm.h is now included by asterisk.h when needed, instead of being forced by the Makefile; this means external modules will build properly against installed headers with MALLOC_DEBUG enabled
      - simplify the usage of some of these headers in the AEL-related stuff in the utils directory
  ........
    r144925 | kpfleming | 2008-09-27 10:13:30 -0500 (Sat, 27 Sep 2008) | 2 lines
    
    fix some minor issues with rev 144924
  ........
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  r144950 | kpfleming | 2008-09-27 11:10:33 -0500 (Sat, 27 Sep 2008) | 2 lines
  
  fix bugs caused by r144949 when MALLOC_DEBUG is defined
................
  r144951 | kpfleming | 2008-09-27 11:17:43 -0500 (Sat, 27 Sep 2008) | 1 line
  
  remove incorrect comment
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2008-09-27 16:36:39 +00:00
Michiel van Baak
d9b2f77657 Blocked revisions 144879 via svnmerge
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r144879 | mvanbaak | 2008-09-27 02:49:24 +0200 (Sat, 27 Sep 2008) | 2 lines

fix a couple of CLI commands that did not have a help description.

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2008-09-27 01:01:07 +00:00
Joshua Colp
d14cf1fc79 Merged revisions 144829 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r144829 | file | 2008-09-26 20:12:13 -0300 (Fri, 26 Sep 2008) | 2 lines
  
  Update documentation to include default setting. This is for you jtodd!
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2008-09-26 23:14:29 +00:00
Mark Michelson
6f52d90a92 Blocked revisions 144759 via svnmerge
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r144759 | mmichelson | 2008-09-26 17:15:40 -0500 (Fri, 26 Sep 2008) | 14 lines

Blocked revisions 144758 via svnmerge

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r144758 | mmichelson | 2008-09-26 17:14:59 -0500 (Fri, 26 Sep 2008) | 7 lines

This patch was applied to 1.4 but it completely
does not apply since the "found" pointer is not
passed in to this function. If this is going to
be backported, it needs to be done differently or
a deeper backport needs to be done.


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2008-09-26 22:16:07 +00:00
Steve Murphy
cde0ddac0d Merged revisions 144681 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r144681 | murf | 2008-09-26 12:02:06 -0600 (Fri, 26 Sep 2008) | 14 lines

(closes issue #13564)
Reported by: mnicholson
Patches:
      pbx_lua9.diff uploaded by mnicholson (license 96)

Many thanks to Matt for his upgrade to the lua dialplan
option! 

the Description from the bug:

This patch adds a stack trace to errors encountered while executing lua extensions. The patch also handles out of memory errors reported by lua.



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2008-09-26 18:08:05 +00:00
Steve Murphy
fb1a08e168 Merged revisions 144678 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r144678 | murf | 2008-09-26 11:50:35 -0600 (Fri, 26 Sep 2008) | 20 lines

Merged revisions 144677 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r144677 | murf | 2008-09-26 11:47:13 -0600 (Fri, 26 Sep 2008) | 12 lines

(closes issue #13563)
Reported by: mnicholson
Patches:
      found1.diff uploaded by mnicholson (license 96)

This patch was mainly meant to apply to trunk and 1.6.x,
but I'm applying it to 1.4 also, which should be a perfectly
harmless fix to the vast majority of users who are not using
external switches, but the few who might be affected 
will not have to go to the pain of filing a bug report.


........

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2008-09-26 17:52:04 +00:00
Steve Murphy
d933c94dcb Merged revisions 144635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r144635 | murf | 2008-09-26 10:51:30 -0600 (Fri, 26 Sep 2008) | 1 line

Matt suggests we remove utils/build-extensions-conf.lua, as per bug 12961, it is no longer necessary.
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2008-09-26 16:53:01 +00:00
Steve Murphy
561fd11a1f Merged revisions 144569 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r144569 | murf | 2008-09-25 16:21:28 -0600 (Thu, 25 Sep 2008) | 14 lines

(closes issue #13557)
Reported by: nickpeirson

The user attached a patch, but the license is not yet
recorded. I took the liberty of finding and replacing
ALL index() calls with strchr() calls, and that
involves more than just main/pbx.c;

chan_oss, app_playback, func_cut also had calls
to index(), and I changed them out. 1.4 had no
references to index() at all.



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2008-09-25 22:23:21 +00:00
Steve Murphy
8ad055b83e Merged revisions 144563 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r144563 | murf | 2008-09-25 15:54:11 -0600 (Thu, 25 Sep 2008) | 7 lines

(closes issue #13559)
Reported by: mnicholson
Patches:
      pbx_lua8.diff uploaded by mnicholson (license 96)



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2008-09-25 21:56:30 +00:00
Steve Murphy
660e02982e Merged revisions 144523 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r144523 | murf | 2008-09-25 15:18:12 -0600 (Thu, 25 Sep 2008) | 13 lines

I added a little verbage to hashtab for the hashtab_destroy func.
It was pretty sparsely documented.

This update fleshes out the pbx_lua module, to 
add the switch statements to the extensions in the
extensions.lua file, as well as removing them when
the module is unloaded.

Many thanks to Matt Nicholson for his fine
contribution!



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2008-09-25 21:21:33 +00:00
Steve Murphy
49a9883dc5 Merged revisions 144482 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r144482 | murf | 2008-09-25 11:51:11 -0600 (Thu, 25 Sep 2008) | 14 lines

(closes issue #13558)
Reported by: mnicholson

Considering that the example extensions.lua used nothing but ["12345"] notation,
and that the resulting error message: 

[Sep 24 17:01:16] ERROR[12393]: pbx_lua.c:1204 exec: Error executing lua extension: attempt to call a nil value

is not very informative as to the nature of the problem, I think this bug
fix is a big win!




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2008-09-25 17:55:08 +00:00
Tilghman Lesher
91e2b5bee0 Blocked revisions 144199 via svnmerge
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r144199 | tilghman | 2008-09-24 01:43:05 -0500 (Wed, 24 Sep 2008) | 4 lines

Create a 'hashcompat' option that permits the results of a CURL() able to be
passed directly into the HASH() function.  Requested via the -users list, and
committed at Astricon in the Code Zone.

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2008-09-24 06:44:50 +00:00
Mark Michelson
9c4ebba68a Blocked revisions 144149 via svnmerge
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r144149 | mmichelson | 2008-09-23 18:33:33 -0500 (Tue, 23 Sep 2008) | 3 lines

Fix a conflict in flag values


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2008-09-23 23:34:57 +00:00
Steve Murphy
f1421805ed Merged revisions 144067 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r144067 | murf | 2008-09-23 10:52:32 -0600 (Tue, 23 Sep 2008) | 37 lines

Merged revisions 144066 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r144066 | murf | 2008-09-23 10:41:49 -0600 (Tue, 23 Sep 2008) | 29 lines

(closes issue #13489)
Reported by: DougUDI
Tested by: murf

(closes issue #13490)
Reported by: seanbright
Tested by: murf

(closes issue #13467)
Reported by: edantie
Tested by: murf, edantie, DougUDI


This crash happens because we are unsafely handling old pointers.
The channel whose cdr is being handled, has been hung up and 
destroyed already. I reorganized the code a bit, and tried not
to lose the fork-cdr-chain concepts of the previous code.
I now verify that the 'previous' channel (the channel we
had when the bridge was started), still exists, by looking it up
by name in the channel list. I also do not try to reset the
CDR's of channels involved in bridges. 

Testing shows it solves the crash problem, and should not
negatively impact previous fixes involving CDR's generated
during/after blind transfers. (The reason we need to reset
the CDR's on the "beginning" channels in the first place).



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2008-09-23 16:57:56 +00:00
Mark Michelson
3919045ee6 Merged revisions 144025 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r144025 | mmichelson | 2008-09-23 10:37:00 -0500 (Tue, 23 Sep 2008) | 16 lines

When a promiscuous redirect contained both a user and
host portion in the Contact URI and specifies a 
transport, the parsing done in parse_moved_contact
resulted in a malformed URI.

This commit fixes the parsing so that a proper
Dial string may be formed when the forwarded
call is placed.

(closes issue #13523)
Reported by: mattdarnell
Patches:
      13523v2.patch uploaded by putnopvut (license 60)
Tested by: mattdarnell


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2008-09-23 15:38:06 +00:00
Sean Bright
3758e4813a Merged revisions 143904 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r143904 | seanbright | 2008-09-22 18:50:07 -0400 (Mon, 22 Sep 2008) | 16 lines

Merged revisions 143903 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r143903 | seanbright | 2008-09-22 18:49:00 -0400 (Mon, 22 Sep 2008) | 8 lines

Use the advertised header size in .au files instead of just assuming they
are 24 bytes (the minimum).

(closes issue #13450)
Reported by: jamessan
Patches:
      pcm-header.diff uploaded by jamessan (license 246)

........

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2008-09-22 22:50:55 +00:00
Michiel van Baak
790baaec02 Merged revisions 143843 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r143843 | mvanbaak | 2008-09-21 11:53:01 +0200 (Sun, 21 Sep 2008) | 3 lines
  
  fix privacymanager example so it shows how to use the
  PRIVACYMRGSTATUS variable
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2008-09-21 09:55:28 +00:00
Michiel van Baak
4e8bf25213 Blocked revisions 143840 via svnmerge
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  r143840 | mvanbaak | 2008-09-21 11:31:54 +0200 (Sun, 21 Sep 2008) | 3 lines
  
  document the new context argument for privacymanager
  so people can do pattern matching on the input
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2008-09-21 09:33:12 +00:00
Michiel van Baak
99f6718460 Merged revisions 143837 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r143837 | mvanbaak | 2008-09-21 11:27:08 +0200 (Sun, 21 Sep 2008) | 2 lines
  
  fix privacy documentation. We no longer do priority jumping +101
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2008-09-21 09:28:25 +00:00
Michiel van Baak
18012993f4 Blocked revisions 143799 via svnmerge
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  r143799 | mvanbaak | 2008-09-20 12:39:20 +0200 (Sat, 20 Sep 2008) | 8 lines
  
  make 'module unload chan_skinny.so' actually work.
  
  (closes issue #13524)
  Reported by: wedhorn
  Patches:
        unload.diff uploaded by wedhorn (license 30)
  	  With small tweak by me to prevent a crash
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2008-09-20 10:40:50 +00:00
Sean Bright
3bbb6b6d61 Merged revisions 143737 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r143737 | seanbright | 2008-09-19 20:52:20 -0400 (Fri, 19 Sep 2008) | 17 lines

Merged revisions 143736 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r143736 | seanbright | 2008-09-19 20:50:10 -0400 (Fri, 19 Sep 2008) | 9 lines

Make vmail.cgi work with mailboxes defined in users.conf, too.

(closes issue #13187)
Reported by: netvoice
Patches:
      20080911__bug13187.diff.txt uploaded by Corydon76 (license 14)
      (Slightly modified to take alchamist's comments on mantis into account)
Tested by: msales, alchamist, seanbright

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2008-09-20 00:54:15 +00:00
Sean Bright
dc216d98f0 ast_event_unsubscribe in 1.6.0 returns void, not NULL.
Pointed out by Stefan Gofferje on the asterisk-users mailing list.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@143735 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-20 00:38:51 +00:00
Mark Michelson
f6376d0130 Merged revisions 143609 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r143609 | mmichelson | 2008-09-19 10:43:28 -0500 (Fri, 19 Sep 2008) | 11 lines

We should only unsubscribe to the device state event
subscription if we have previously subscribed. Otherwise
a segfault will occur.

(closes issue #13476)
Reported by: jonnt
Patches:
      13476.patch uploaded by putnopvut (license 60)
Tested by: jonnt


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2008-09-19 15:48:37 +00:00
Steve Murphy
9f74603219 Merged revisions 143559 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r143559 | murf | 2008-09-18 17:41:33 -0600 (Thu, 18 Sep 2008) | 9 lines

Merged revisions 143534 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r143534 | murf | 2008-09-18 16:11:51 -0600 (Thu, 18 Sep 2008) | 1 line

A micro-fix, in sip_park_thread, where d is freed before the func is done using it.
........

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2008-09-18 23:50:25 +00:00
Tilghman Lesher
73e498b461 Merged revisions 143405 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r143405 | tilghman | 2008-09-17 15:57:58 -0500 (Wed, 17 Sep 2008) | 13 lines
  
  Merged revisions 143404 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r143404 | tilghman | 2008-09-17 15:55:47 -0500 (Wed, 17 Sep 2008) | 6 lines
    
    When callerid is blank, we want to use "unknown caller" in those cases, too.
    (closes issue #13486)
     Reported by: tomo1657
     Patches: 
           20080917__bug13486.diff.txt uploaded by Corydon76 (license 14)
  ........
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2008-09-17 20:58:54 +00:00
Mark Michelson
bd9c3357c3 Blocked revisions 143400 via svnmerge
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r143400 | mmichelson | 2008-09-17 15:25:40 -0500 (Wed, 17 Sep 2008) | 16 lines

If attempting to free a NULL pointer when MALLOC_DEBUG
is set, don't bother searching for a region to free, just
immediately exit.

This has the dual benefit of suppressing a warning message
about freeing memory at (nil) and of optimizing the free()
replacement by not having to do any futile searching for
the proper region to free.

(closes issue #13498)
Reported by: pj
Patches:
      13498.patch uploaded by putnopvut (license 60)
Tested by: pj


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2008-09-17 20:26:27 +00:00
Mark Michelson
e9e8dd2efd Blocked revisions 143340 via svnmerge
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r143340 | mmichelson | 2008-09-17 13:26:35 -0500 (Wed, 17 Sep 2008) | 14 lines

Merged revisions 143337 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r143337 | mmichelson | 2008-09-17 13:24:15 -0500 (Wed, 17 Sep 2008) | 6 lines

Allow for "G.729" if offered in an SDP even though
it is not RFC 3551 compliant. Some Cisco switches
will send this in an SDP, and it doesn't hurt to
be able to accept this.


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2008-09-17 18:29:12 +00:00
Tilghman Lesher
e4a1181251 Merged revisions 143141 via svnmerge from
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  r143141 | tilghman | 2008-09-15 16:31:36 -0500 (Mon, 15 Sep 2008) | 13 lines
  
  Merged revisions 143140 via svnmerge from 
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    r143140 | tilghman | 2008-09-15 16:29:32 -0500 (Mon, 15 Sep 2008) | 6 lines
    
    Set the raw formats at the same time as the other formats.
    (closes issue #13240)
     Reported by: jvandal
     Patches: 
           20080813__bug13240.diff.txt uploaded by Corydon76 (license 14)
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2008-09-15 21:32:29 +00:00
Michiel van Baak
4097442bdb Merged revisions 143082 via svnmerge from
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  r143082 | mvanbaak | 2008-09-15 00:16:34 +0200 (Mon, 15 Sep 2008) | 11 lines
  
  plug a couple of memleaks in chan_skinny.
  
  (closes issue #13452)
  Reported by: pj
  Patches:
        memleak5.diff uploaded by wedhorn (license 30)
  Tested by: wedhorn, pj, mvanbaak
  
  (closes issue #13294)
  Reported by: pj
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2008-09-14 22:22:48 +00:00
Sean Bright
6e52004f8b Blocked revisions 143034 via svnmerge
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r143034 | seanbright | 2008-09-13 10:15:08 -0400 (Sat, 13 Sep 2008) | 1 line

Everytime a compile fails, a puppy dies.
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2008-09-13 14:16:10 +00:00
Tilghman Lesher
5302fa36b8 Recorded merge of revisions 143031 via svnmerge from
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  r143031 | tilghman | 2008-09-13 08:54:15 -0500 (Sat, 13 Sep 2008) | 8 lines
  
  Repair IAXVAR implementation so that it works again (regression?)
  (closes issue #13354)
   Reported by: adomjan
   Patches: 
         20080828__bug13354.diff.txt uploaded by Corydon76 (license 14)
         20080829__bug13354__1.6.0.diff.txt uploaded by Corydon76 (license 14)
   Tested by: Corydon76, adomjan
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2008-09-13 13:57:05 +00:00
Tilghman Lesher
1421b14291 Blocked revisions 142992 via svnmerge
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  r142992 | tilghman | 2008-09-12 18:30:03 -0500 (Fri, 12 Sep 2008) | 27 lines
  
  Create a new config file status, CONFIG_STATUS_FILEINVALID for differentiating
  when a file is invalid from when a file is missing.  This is most important when
  we have two configuration files.  Consider the following example:
  
  Old system:
  sip.conf     users.conf     Old result               New result
  ========     ==========     ==========               ==========
  Missing      Missing        SIP doesn't load         SIP doesn't load
  Missing      OK             SIP doesn't load         SIP doesn't load
  Missing      Invalid        SIP doesn't load         SIP doesn't load
  OK           Missing        SIP loads                SIP loads
  OK           OK             SIP loads                SIP loads
  OK           Invalid        SIP loads incompletely   SIP doesn't load
  Invalid      Missing        SIP doesn't load         SIP doesn't load
  Invalid      OK             SIP doesn't load         SIP doesn't load
  Invalid      Invalid        SIP doesn't load         SIP doesn't load
  
  So in the case when users.conf doesn't load because there's a typo that
  disrupts the syntax, we may only partially load users, instead of failing with
  an error, which may cause some calls not to get processed.  Worse yet, the old
  system would do this with no indication that anything was even wrong.
  
  (closes issue #10690)
   Reported by: dtyoo
   Patches: 
         20080716__bug10690.diff.txt uploaded by Corydon76 (license 14)
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2008-09-12 23:32:12 +00:00
Jeff Peeler
9629a4a0a1 Merged revisions 142929 via svnmerge from
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r142929 | jpeeler | 2008-09-12 17:24:13 -0500 (Fri, 12 Sep 2008) | 14 lines

Merged revisions 142927 via svnmerge from 
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r142927 | jpeeler | 2008-09-12 17:22:28 -0500 (Fri, 12 Sep 2008) | 6 lines

(closes issue #12965)
Reported by: rlsutton2

Prevents local channels from playing MOH at each other which was causing ast_generic_bridge to loop much faster.


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2008-09-12 22:25:21 +00:00
Tilghman Lesher
86598fd373 Merged revisions 142866 via svnmerge from
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  r142866 | tilghman | 2008-09-12 15:49:46 -0500 (Fri, 12 Sep 2008) | 18 lines
  
  Merged revisions 142865 via svnmerge from 
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    r142865 | tilghman | 2008-09-12 15:37:18 -0500 (Fri, 12 Sep 2008) | 11 lines
    
    Create rules for disallowing contacts at certain addresses, which may
    improve the security of various installations.  As this does not change
    any default behavior, it is not classified as a direct security fix for
    anything within Asterisk, but may help PBX admins better secure their
    SIP servers.
    (closes issue #11776)
     Reported by: ibc
     Patches: 
           20080829__bug11776.diff.txt uploaded by Corydon76 (license 14)
     Tested by: Corydon76, blitzrage
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Tilghman Lesher
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r142748 | tilghman | 2008-09-12 11:54:44 -0500 (Fri, 12 Sep 2008) | 3 lines

When checking for an encoded character, make sure the string isn't blank, first.
(Closes issue #13470)

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2008-09-12 16:56:49 +00:00
Tilghman Lesher
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r142745 | tilghman | 2008-09-12 11:38:55 -0500 (Fri, 12 Sep 2008) | 12 lines

Merged revisions 142744 via svnmerge from 
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r142744 | tilghman | 2008-09-12 11:38:02 -0500 (Fri, 12 Sep 2008) | 4 lines

Missing merge from 1.2 fixes errant exit on DTMF, only when language is Italian
(cf commit 34242)
(Closes issue #7353)

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Tilghman Lesher
6c55eda169 Merged revisions 142741 via svnmerge from
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r142741 | tilghman | 2008-09-12 11:29:01 -0500 (Fri, 12 Sep 2008) | 12 lines

Merged revisions 142740 via svnmerge from 
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r142740 | tilghman | 2008-09-12 11:27:32 -0500 (Fri, 12 Sep 2008) | 4 lines

Don't return a free'd pointer, when a file cannot be opened.
(closes issue #13462)
 Reported by: wackysalut

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2008-09-12 16:30:07 +00:00
Steve Murphy
3c1c24500b Merged revisions 142676 via svnmerge from
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r142676 | murf | 2008-09-11 22:50:48 -0600 (Thu, 11 Sep 2008) | 40 lines

Merged revisions 142675 via svnmerge from 
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r142675 | murf | 2008-09-11 22:29:34 -0600 (Thu, 11 Sep 2008) | 29 lines

Tested by: sergee, murf, chris-mac, andrew, KNK

This is a "second attempt" to restore the previous "endbeforeh" behavior
in 1.4 and up. In order to capture information concerning all the
legs of transfers in all their infinite combinations, I was forced
to this particular solution by a chain of logical necessities, the
first being that I was not allowed to rewrite the CDR mechanism from 
the ground up!

This change basically leaves the original machinery alone, which allows
IVR and local channel type situations to generate CDR's as normal, but
a channel flag can be set to suppress the normal running of the h exten.
That flag would be set by the code that runs the h exten from the
ast_bridge_call routine, to prevent the h exten from being run twice.
Also, a flag in the ast_bridge_config struct passed into ast_bridge_call
can be used to suppress the running of the h exten in that routine. This
would happen, for instance, if you use the 'g' option in the Dial app.

Running this routine 'early' allows not only the CDR() func to be used
in the h extension for reading CDR variables, but also allows them to
be modified before the CDR is posted to the backends.

While I dearly hope that this patch overcomes all problems, and 
introduces no new problems, reality suggests that surely someone
will have problems. In this case, please re-open 13251 (or 13289),
and we'll see if we can't fix any remaining issues.

** trunk note: some code to suppress the h exten being run 
from app_queue was added; for the 'continue' option available
only in trunk/1.6.x.


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2008-09-12 04:58:38 +00:00
Sean Bright
7d5e83a34f Blocked revisions 142635 via svnmerge
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r142635 | seanbright | 2008-09-11 20:49:31 -0400 (Thu, 11 Sep 2008) | 1 line

Build under dev-mode
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2008-09-12 00:50:42 +00:00
Steve Murphy
d1723e6ae5 Merged revisions 142576 via svnmerge from
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r142576 | murf | 2008-09-11 17:12:53 -0600 (Thu, 11 Sep 2008) | 28 lines

Merged revisions 142575 via svnmerge from 
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r142575 | murf | 2008-09-11 16:55:49 -0600 (Thu, 11 Sep 2008) | 20 lines

(closes issue #13364)
Reported by: mdu113

Well, fundamentally, the problems revealed in 13364 are
because of the ForkCDR call that is done before the dial. 
When the bridge is in place, it's dealing with the first
(and wrong) cdr in the list.

So, I wrote a little func to zip down to the first non-locked
cdr in the chain, and thru-out the ast_bridge_call, these
results are used instead of raw chan->cdr and peer->cdr pointers.
This shouldn't affect anyone who isn't forking cdrs before a
dial, and should correct the cdr's of those that do.

So, this change ends up correcting the dstchannel
and userfield; the disposition was fixed by a previous
patch, it was OK coming into this problem.



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Tilghman Lesher
cdd49ae465 Blocked revisions 142536 via svnmerge
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r142536 | tilghman | 2008-09-11 16:45:07 -0500 (Thu, 11 Sep 2008) | 4 lines

Add usegmtime, as per the recent -users list discussion, and also add my
explanation to the file, since that additional text helps people understand
the concept.

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Steve Murphy
2116207025 Merged revisions 142475 via svnmerge from
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r142475 | murf | 2008-09-10 16:11:27 -0600 (Wed, 10 Sep 2008) | 38 lines

Merged revisions 142474 via svnmerge from 
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r142474 | murf | 2008-09-10 15:58:17 -0600 (Wed, 10 Sep 2008) | 30 lines

(closes issue #12318)
Reported by: krtorio

I made a small change to the code that handles local channel situations.
In that code, I copy the answer time from the peer cdr, to the bridge_cdr,
but I wasn't also copying the disposition from the peer cdr.

So, Now I copy the disposition, and I've tested against 
these cases:

1. phone 1 never answers the phone; no cdr is generated at all.
   this should show up as a manager command failure or something.

2. phone 2 never answers. CDR is generated, says NO ANSWER

3. phone 2 is busy. CDR is generated, says BUSY

4. phone 2 answers: CDR is generated, times are correct; disposition
   is ANSWERED, which is correct. The start time is the time that
   the manager dialed the first phone. The answer time is the time
   the second phone picks up.

I purposely left the cid and src fields blank; since this call really
originates from the manager, there is no 'easy' data to put in these
fields. If you feel strongly that these fields should be filled in,
re-open this bug and I'll dig further.




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2008-09-10 22:14:41 +00:00
Sean Bright
c4f7728d7c Merged revisions 142417 via svnmerge from
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r142417 | seanbright | 2008-09-10 15:09:03 -0400 (Wed, 10 Sep 2008) | 17 lines

Merged revisions 142416 via svnmerge from 
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r142416 | seanbright | 2008-09-10 15:05:46 -0400 (Wed, 10 Sep 2008) | 9 lines

Fix detection of PWLIB and OpenH323 version when spacing in the headers
isn't consistent.

(closes issue #13426)
Reported by: bamby
Patches:
      detect_openh323.diff uploaded by bamby (license 430)
      (Modified by me to use sed instead of tr)

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2008-09-10 19:12:16 +00:00
Tilghman Lesher
a97d655a8d Merged revisions 142359 via svnmerge from
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r142359 | tilghman | 2008-09-10 11:55:31 -0500 (Wed, 10 Sep 2008) | 10 lines

Merged revisions 142358 via svnmerge from 
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r142358 | tilghman | 2008-09-10 11:54:29 -0500 (Wed, 10 Sep 2008) | 2 lines

Publish new extra sounds version.

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2008-09-10 16:57:05 +00:00
Russell Bryant
e6f7951a95 Merged revisions 142355 via svnmerge from
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r142355 | russell | 2008-09-10 11:41:55 -0500 (Wed, 10 Sep 2008) | 15 lines

Merged revisions 142354 via svnmerge from 
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r142354 | russell | 2008-09-10 11:39:53 -0500 (Wed, 10 Sep 2008) | 7 lines

It is a normal situation that a task gets put in the scheduler that should run
as soon as possible.  Accept "0" as an acceptable time to run, and also treat
negative as "run now", and don't print a debug message about it.

(inspired by a message asking about the "request to schedule in the past"
 debug message on the -dev list)

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Philippe Sultan
52826a8645 Merged revisions 142280 via svnmerge from
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r142280 | phsultan | 2008-09-10 00:08:56 +0200 (Wed, 10 Sep 2008) | 6 lines

Disable autoprune by default.
(closes issue #13411)
Reported by: caio1982
Patches:
      res_jabber_autoprune1.diff uploaded by caio1982 (license 22)
Tested by: caio1982
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2008-09-09 22:14:40 +00:00
Mark Michelson
a520ba2e48 Blocked revisions 142219 via svnmerge
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r142219 | mmichelson | 2008-09-09 14:16:30 -0500 (Tue, 09 Sep 2008) | 22 lines

Merged revisions 142218 via svnmerge from 
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r142218 | mmichelson | 2008-09-09 14:15:28 -0500 (Tue, 09 Sep 2008) | 14 lines

Make sure that the branch sent in CANCEL requests
matches the branch of the INVITE it is cancelling.

(closes issue #13381)
Reported by: atca_pres
Patches:
      13381v2.patch uploaded by putnopvut (license 60)
Tested by: atca_pres

(closes issue #13198)
Reported by: rickead2000
Tested by: rickead2000


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Mark Michelson
a4c74eccbd Blocked revisions 142146 via svnmerge
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r142146 | mmichelson | 2008-09-09 12:15:29 -0500 (Tue, 09 Sep 2008) | 21 lines

This is the trunk version of the patch to close
issue 12979. The difference between this and the
1.6.0 and 1.6.1 versions is that this is a much more
invasive change. With this, we completely get rid
of the interfaces list, along with all its helper
functions.

Let me take a moment to say that this change personally
excites me since it may mean huge steps forward regarding
proper lock order in app_queue without having to strew
seemingly unnecessary locks all over the place. It also
results in a huge reduction in lines of code and complexity.
Way to go Brett!

(closes issue #12979)
Reported by: sigxcpu
Patches:
      20080710_issue12979_queue_custom_state_interface_trunk_2.diff uploaded by bbryant (license 36)
Tested by: sigxcpu, putnopvut


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2008-09-09 17:16:33 +00:00
Mark Michelson
fac93e8478 Fix app_queue's device state callback so that it
can correctly parse custom device states (and any
other device which does not contain a '/').

1.6.1 will be getting this patch as well, but trunk
is going to get a much more massive patch by bbryant
which does some very nice overhauling of some
structures in app_queue.

(closes issue #12979)
Reported by: sigxcpu
Patches:
      12979.patch uploaded by putnopvut (license 60)
Tested by: sigxcpu



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2008-09-09 16:26:56 +00:00
Mark Michelson
b12820f102 Blocked revisions 142080 via svnmerge
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r142080 | mmichelson | 2008-09-09 11:20:41 -0500 (Tue, 09 Sep 2008) | 29 lines

Merged revisions 142079 via svnmerge from 
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r142079 | mmichelson | 2008-09-09 11:19:17 -0500 (Tue, 09 Sep 2008) | 21 lines

When determining if codecs used by SIP peers allow
the media to be natively bridged, use the jointcapability
instead of the peercapability.

It seems that the intent of using the peercapability was to
expand the choice of codecs for the call to increase the
chances of being able to native bridge the channels. The 
problem is that if a codec were settled on for the native
bridge and that wasn't a codec that was configured to be used
by Asterisk for that peer, then Asterisk would send a 
REINVITE with no codecs in the SDP which is a bug no matter
how you slice it.


(closes issue #13076)
Reported by: ramonpeek
Patches:
      13076.patch uploaded by putnopvut (license 60)
Tested by: tbelder


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Russell Bryant
f573a4785e Merged revisions 142064 via svnmerge from
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r142064 | russell | 2008-09-09 10:44:10 -0500 (Tue, 09 Sep 2008) | 13 lines

Merged revisions 142063 via svnmerge from 
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r142063 | russell | 2008-09-09 10:40:24 -0500 (Tue, 09 Sep 2008) | 5 lines

Ensure that the stored CDR reference is still valid after the bridge before
poking at it.  Also, keep the channel locked while messing with this CDR.

(fixes crashes reported in issue #13409)

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Bradley Latus
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r142000 | snuffy | 2008-09-09 22:34:32 +1000 (Tue, 09 Sep 2008) | 2 lines

Minor fix to doco

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Mark Michelson
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r141998 | mmichelson | 2008-09-09 07:32:38 -0500 (Tue, 09 Sep 2008) | 7 lines

Use ast_debug for debug messages. I was wondering why debug
messages weren't showing up when I had set the debug level
high for just app_queue.c. It's because we were only checking
the global option_debug variable instead of using the awesome
macro which checks both the global and file-specific value


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Mark Michelson
6a4a774f00 Merged revisions 141995 via svnmerge from
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r141995 | mmichelson | 2008-09-09 05:20:58 -0500 (Tue, 09 Sep 2008) | 8 lines

Fix a memory leak in chan_oss

(closes issue #13311)
Reported by: eliel
Patches:
      chan_oss.c.patch uploaded by eliel (license 64)


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2008-09-09 10:24:32 +00:00
Russell Bryant
449cf2d24a Merged revisions 141949 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r141949 | russell | 2008-09-08 20:47:56 -0500 (Mon, 08 Sep 2008) | 9 lines

Modify ast_answer() to not hold the channel lock while calling ast_safe_sleep()
or when calling ast_waitfor().  These are inappropriate times to hold the channel
lock.  This is what has caused "could not get the channel lock" messages from
chan_sip and has likely caused a negative impact on performance results of SIP
in Asterisk 1.6.  Thanks to file for pointing out this section of code.

(closes issue #13287)
(closes issue #13115)

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2008-09-09 01:49:25 +00:00
Mark Michelson
1be1e4b5e4 Blocked revisions 141906 via svnmerge
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r141906 | mmichelson | 2008-09-08 18:00:06 -0500 (Mon, 08 Sep 2008) | 9 lines

Optimization: The only reason we should check member
status is if the queue has a joinempty or a leavewhenempty
setting which could cause the caller to not join the queue
or exit the queue. Prior to this patch, we could potentially
traverse the entire queue's member list for no reason since
even if the members are currently not available in some way
we're going to let the caller join the queue anyway.


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2008-09-08 23:00:38 +00:00
Mark Michelson
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r141868 | mmichelson | 2008-09-08 17:14:40 -0500 (Mon, 08 Sep 2008) | 4 lines

Um, apparently I didn't actually finish merging before committing.
Bad bad bad


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Mark Michelson
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r141810 | mmichelson | 2008-09-08 16:18:49 -0500 (Mon, 08 Sep 2008) | 22 lines

Merged revisions 141809 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r141809 | mmichelson | 2008-09-08 16:10:10 -0500 (Mon, 08 Sep 2008) | 14 lines

Fix pedantic mode of chan_sip to only check the
remote tag of an endpoint once a dialog has
been confirmed. Up until that point, it is possible
and legal for the far-end to send provisional
responses with a different To: tag each time. With
this patch applied, these provisional messages
will not cause a matching problem.

(closes issue #11536)
Reported by: ibc
Patches:
      11536v2.patch uploaded by putnopvut (license 60)


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2008-09-08 21:19:23 +00:00
Russell Bryant
f86cd6496b Merged revisions 141807 via svnmerge from
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r141807 | russell | 2008-09-08 16:05:01 -0500 (Mon, 08 Sep 2008) | 15 lines

Merged revisions 141806 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r141806 | russell | 2008-09-08 16:02:36 -0500 (Mon, 08 Sep 2008) | 7 lines

When doing an async goto, detect if the channel is already in the middle of a
masquerade.  This can happen when chan_local is trying to optimize itself out.
If this happens, fail the async goto instead of bursting into flames.

(closes issue #13435)
Reported by: geoff2010

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2008-09-08 21:07:12 +00:00
Jason Parker
db347c26e1 Merged revisions 141745 via svnmerge from
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r141745 | qwell | 2008-09-08 15:18:17 -0500 (Mon, 08 Sep 2008) | 16 lines

Merged revisions 141741 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r141741 | qwell | 2008-09-08 15:15:42 -0500 (Mon, 08 Sep 2008) | 8 lines

Remove RPM package targets from Makefile (and all associated parts).
This has never worked in 1.4, and we decided that it makes no sense to be done here.
There are many distros out there that already have "proper" spec files that can be (re)used.

Closes issue #13113
Closes issue #10950
Closes issue #10952

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2008-09-08 20:19:29 +00:00
Sean Bright
6cfd997abd Merged revisions 141682 via svnmerge from
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r141682 | seanbright | 2008-09-08 13:13:04 -0400 (Mon, 08 Sep 2008) | 9 lines

Quote the arguments to grep so that sh on various platforms doesn't choke
on the special characters (like ^).

(closes issue #13417)
Reported by: dougm
Patches:
      13417.make_buildopts_h.patch uploaded by seanbright (license 71)
Tested by: dougm

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2008-09-08 17:14:29 +00:00
Michiel van Baak
aa550bc39a Blocked revisions 141626 via svnmerge
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r141626 | mvanbaak | 2008-09-07 02:04:05 +0200 (Sun, 07 Sep 2008) | 2 lines

make func_curl.c compile under devmode.

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2008-09-07 00:09:12 +00:00
Steve Murphy
0b963bfc68 Merged revisions 141566 via svnmerge from
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r141566 | murf | 2008-09-06 14:19:50 -0600 (Sat, 06 Sep 2008) | 9 lines

Merged revisions 141565 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r141565 | murf | 2008-09-06 14:13:16 -0600 (Sat, 06 Sep 2008) | 1 line

This fix comes from Joshua Colp The Brilliant, who, given the trace, came up with a solution. This will most likely will close 13235 and 13409. I'll wait till Monday to verify, and then close these bugs.
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2008-09-06 20:21:51 +00:00
Tilghman Lesher
f874842ca3 Blocked revisions 141507 via svnmerge
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r141507 | tilghman | 2008-09-06 10:40:15 -0500 (Sat, 06 Sep 2008) | 5 lines

Get rid of the casts that cause warnings on OpenBSD.  The compiler is errantly
detecting warnings when we redefine a structure each time it is used, even
though the structure is identical.
Reported by: mvanbaak, via #asterisk-dev

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Tilghman Lesher
6f43982fee Merged revisions 141504 via svnmerge from
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r141504 | tilghman | 2008-09-06 10:26:45 -0500 (Sat, 06 Sep 2008) | 12 lines

Merged revisions 141503 via svnmerge from 
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r141503 | tilghman | 2008-09-06 10:23:42 -0500 (Sat, 06 Sep 2008) | 4 lines

Reverting behavior change (AGI should not exit non-zero on SUCCESS)
(closes issue #13434)
 Reported by: francesco_r

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2008-09-06 15:28:04 +00:00
Michiel van Baak
daad13b139 Blocked revisions 141464 via svnmerge
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r141464 | mvanbaak | 2008-09-06 14:03:11 +0200 (Sat, 06 Sep 2008) | 23 lines

Some fixes to autocompletion in some commands.

Changes applied by this patch:

- Fix autocompletion in 'sip prune realtime', sip peers where never auto completed. Now we complete this command with:
  'sip prune realtime peer' -> all | like | sip peers
  Also I have modified the syntax in the usage, was wrong...
- Pass ast_cli_args->argv and ast_cli_args->argc while running autocompletion on CLI commands (CLI_GENERATE).
  With this we avoid comparisons on ast_cli_args->line like this:
  strcasestr(a->line, " description")
  strcasestr(a->line, "descriptions ")
  strcasestr(a->line, "realtime peer"), and so on..

  Making the code more confusing (check the spaces in description!).
  The only thing we must be sure is to first check a->pos or a->argc.
														      
- Fix 'iax2 prune realtime' autocompletion, now we autocomplete this command with 'all' & 'iax2 peers', check a look that iax2 peers where all the peers, now only the ones in the cache..

(closes issue #13133)
Reported by: eliel
Patches:
      clichanges.patch uploaded by eliel (license 64)

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Mark Michelson
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r141425 | mmichelson | 2008-09-05 17:03:26 -0500 (Fri, 05 Sep 2008) | 3 lines

Fix func_curl compilation


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Mark Michelson
8628ad06ae Merged revisions 141367 via svnmerge from
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r141367 | mmichelson | 2008-09-05 16:12:09 -0500 (Fri, 05 Sep 2008) | 15 lines

Merged revisions 141366 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r141366 | mmichelson | 2008-09-05 16:10:32 -0500 (Fri, 05 Sep 2008) | 7 lines

Agent's should not try to call a channel's indicate callback
if the channel has been hung up. It will likely crash
otherwise

ABE-1159


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2008-09-05 21:13:06 +00:00
Tilghman Lesher
d216251bf4 Blocked revisions 141328 via svnmerge
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r141328 | tilghman | 2008-09-05 14:12:03 -0500 (Fri, 05 Sep 2008) | 8 lines

Add the CURLOPT dialplan function, which permits setting various options for
use with the CURL dialplan function.
(closes issue #12920)
 Reported by: davevg
 Patches: 
       20080904__bug12920.diff.txt uploaded by Corydon76 (license 14)
 Tested by: Corydon76, davevg

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Mark Michelson
c2ff4c297d Blocked revisions 141271 via svnmerge
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r141271 | mmichelson | 2008-09-05 11:36:48 -0500 (Fri, 05 Sep 2008) | 18 lines

Blocked revisions 141267 via svnmerge

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r141267 | mmichelson | 2008-09-05 11:35:54 -0500 (Fri, 05 Sep 2008) | 11 lines

Since greetings are not stored in IMAP, we should
not be DISPOSE'ing of them the same way we do with
other messages.

(closes issue #13414)
Reported by: mthomasslo
Patches:
      13414v2.patch uploaded by putnopvut (license 60)
Tested by: mthomasslo


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Mark Michelson
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r141218 | mmichelson | 2008-09-05 11:03:11 -0500 (Fri, 05 Sep 2008) | 21 lines

Blocked revisions 141217 via svnmerge

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r141217 | mmichelson | 2008-09-05 11:00:24 -0500 (Fri, 05 Sep 2008) | 14 lines

Commit 140417 had a logic flaw in it which
caused port 5060 to always be used when dialing
a peer if no explicit port was specified. This
broke the behavior of implicitly using the port
from which the peer registered if no port is
specified. This commit fixes the logic flaw.

(closes issue #13424)
Reported by: mdu113
Patches:
      13424.patch uploaded by putnopvut (license 60)
Tested by: mdu113


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Steve Murphy
90d877b2da Merged revisions 141157 via svnmerge from
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r141157 | murf | 2008-09-05 08:18:43 -0600 (Fri, 05 Sep 2008) | 9 lines

Merged revisions 141156 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r141156 | murf | 2008-09-05 08:15:43 -0600 (Fri, 05 Sep 2008) | 1 line

A small change to prevent double-posting of CDR's; thanks to Daniel Ferrer for bringing it to our attention
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Steve Murphy
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r141115 | murf | 2008-09-04 17:31:41 -0600 (Thu, 04 Sep 2008) | 78 lines

Merged revisions 141094 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r141094 | murf | 2008-09-04 17:15:07 -0600 (Thu, 04 Sep 2008) | 70 lines

(closes issue #13357)
Reported by: pj
Tested by: murf

(closes issue #13416)
Reported by: yarns
Tested by: murf

If you find this message overly verbose, relax, it's probably
not meant for you. This message is meant for probably only
two people in the whole world: me, or the poor schnook that
has to maintain this code because I'm either dead or unavailable
at the moment.

This fix solves two reports, both having to do with embedding
a function call in a ${} construct. It was tricky because the
funccall syntax has parenthesis () in it. And up till now,
the 'word' token in the flex stuff didn't allow that, because
it would tend to steal the LP and RP tokens. To be truthful,
the "word" token was the trickiest, most unstable thing in
the whole lexer. I was lucky it made this long without complaints.
I had to choose every character in the pattern with extreme
care, and I knew that someday I'd have to revisit it. Well,
the day has come.

So, my brilliant idea (and I'm being modest), was to use the 
surrounding ${} construct to make a state machine and capture 
everything in it, no matter what it contains. But, I have to now
treat the word token like I did with comments, in that I turn
the whole thing into a state-machine sort of spec, with new
contexts "curlystate", "wordstate", and "brackstate".

Wait a minute, "brackstate"? Yes, well, it didn't take very many
regression tests to point out if I do this for ${} constructs,
I also have to do it with the $[] constructs, too.

I had to create a separate pcbstack2 and pcbstack3 because
these constructs can occur inside macro argument lists, and
when we have two state machines operating on the same structures
we'd get problems otherwise. I guess I could have stopped at
pcbstack2 and had the brackstate stuff share it, but it doesn't
hurt to be safe. So, the pcbpush and pcbpop routines also now
have versions for "2" and "3".

I had to add the {KEYWORD} construct to the initial pattern for
"word", because previously word would match stuff like "default7",
because it was a longer match than the keyword "default". But,
not any more, because the word pattern only matches only one or
two characters now, and it will always lose. So, I made it the
winner again by making an optional match on any of the keywords
before it's normal pattern.

I added another regression test to make sure we don't
lose this in future edits, and had to fix just one regression,
where it no longer reports a 'cascaded' error, which I guess
is a plus.

I've given some thought as to whether to apply these fixes to
1.4 and the 1.6.x releases, vs trunk; I decided to put it in
1.4 because one of the bug reports was against 1.4; and it
is unexpected that AEL cannot handle this situation. It actually
reduced the amount of useless "cascade" error messages that
appeared in the regressions (by one line, ehhem). There is
a possible side-effect in that it does now do more careful
checking of what's in those ${} constructs, as far as matching
parens, and brackets are concerned. Some users may find a an
insidious problem and correct it this way. This should be 
exceedingly rare, I hope.


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2008-09-04 23:47:51 +00:00
Jeff Peeler
4ee5b75c5c Merged revisions 141039 via svnmerge from
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r141039 | jpeeler | 2008-09-04 12:27:56 -0500 (Thu, 04 Sep 2008) | 15 lines

Merged revisions 141028 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r141028 | jpeeler | 2008-09-04 12:00:29 -0500 (Thu, 04 Sep 2008) | 7 lines

(closes issue #11979)
Fixes multiple parking problems:
Crash when executing a park on an extension dialed by AGI due to not returning the proper return code.
Crash when using a builtin feature that was a subset of a enabled dynamic feature.
Crash due to always hanging up the peer despite the fact that the peer was supposed to be parked.


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Mark Michelson
dc4224c694 Merged revisions 140975 via svnmerge from
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r140975 | mmichelson | 2008-09-03 15:16:12 -0500 (Wed, 03 Sep 2008) | 4 lines

Fix some locking order issues in app_queue. This was
brought up by atis on IRC a while ago.


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Michiel van Baak
c58e07d629 Blocked revisions 140938 via svnmerge
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r140938 | mvanbaak | 2008-09-03 20:06:35 +0200 (Wed, 03 Sep 2008) | 12 lines

Added 'skinny show lines verbose'
This will print the subs and their status for every line (if any).

wedhorn did most of the work with his patch which introduced
'skinny show debug' but a discussion on IRC stated that it should be
added to 'skinny show lines'

Input on the output format by Qwell on IRC.

(closes issue #13344)
Reported by: wedhorn

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Mark Michelson
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r140887 | mmichelson | 2008-09-03 09:41:54 -0500 (Wed, 03 Sep 2008) | 3 lines

Fix compilation


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Mark Michelson
b84c5a7ef5 Blocked revisions 140860 via svnmerge
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r140860 | mmichelson | 2008-09-03 09:31:33 -0500 (Wed, 03 Sep 2008) | 17 lines

Merged revisions 140850 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r140850 | mmichelson | 2008-09-03 09:29:15 -0500 (Wed, 03 Sep 2008) | 9 lines

Fix voicemail forwarding when using ODBC storage.

(closes issue #13387)
Reported by: moliveras
Patches:
      13387.patch uploaded by putnopvut (license 60)
Tested by: putnopvut, moliveras


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2008-09-03 14:32:24 +00:00
Steve Murphy
fa34fdcb43 Merged revisions 140749 via svnmerge from
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r140749 | murf | 2008-09-02 17:44:04 -0600 (Tue, 02 Sep 2008) | 11 lines

Merged revisions 140747 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r140747 | murf | 2008-09-02 17:36:56 -0600 (Tue, 02 Sep 2008) | 1 line

I am turning the warnings generated in ast_cdr_free and post_cdr into verbose level 2 messages. Really, they matter little to end users. You either get the CDR's you wanted, or you don't, and it is a bug.

For trunk, I am going one step further. These messages were pretty worthless even for debug, so I'm completely removing them.
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Steve Murphy
a3b41b28b6 Merged revisions 140692 via svnmerge from
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r140692 | murf | 2008-09-02 16:55:12 -0600 (Tue, 02 Sep 2008) | 13 lines

Merged revisions 140690 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r140690 | murf | 2008-09-02 16:40:13 -0600 (Tue, 02 Sep 2008) | 1 line

After reconsidering, with respect to 13409, ast_cdr_detach should be OK, better in fact, than ast_cdr_free, which generates lots of useless warnings that will undoubtably generate complaints.

Hmmm. It doesn't hush the useless warnings, but it does allow control of posting via the detach and post routines, for those possible situations,
where you'd want to post single-channel cdrs.

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Steve Murphy
a561485ccf Merged revisions 140691 via svnmerge from
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r140691 | murf | 2008-09-02 16:50:59 -0600 (Tue, 02 Sep 2008) | 22 lines

Merged revisions 140670 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r140670 | murf | 2008-09-02 16:15:57 -0600 (Tue, 02 Sep 2008) | 14 lines

(closes issue #13409)
Reported by: tomaso
Patches:
      asterisk-1.6.0-rc2-cdrmemleak.patch uploaded by tomaso (license 564)

I basically spent the day, verifying that this patch 
solves the problem, and doesn't hurt in non-problem 
cases. Why valgrind did not plainly reveal this leak
absolutely mystifies and stuns me. 

Many, many thanks to tomaso for finding and providing the fix.



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2008-09-03 14:13:50 +00:00
Sean Bright
ac45770aea Blocked revisions 140821 via svnmerge
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r140821 | seanbright | 2008-09-03 09:48:12 -0400 (Wed, 03 Sep 2008) | 10 lines

Move some duplicated code into a separate function.

Also try to do some wacky stuff in the commit message, like:
a newline \n
a bell \a
a tab \t
a format specification %p

That is all.

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2008-09-03 13:50:06 +00:00
Russell Bryant
f6f5ba6adc Merged revisions 140817 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r140817 | russell | 2008-09-03 08:26:43 -0500 (Wed, 03 Sep 2008) | 12 lines

Merged revisions 140816 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r140816 | russell | 2008-09-03 08:24:35 -0500 (Wed, 03 Sep 2008) | 4 lines

Don't freak out if the poll emulation receives NULL for the pollfds array
(closes issue #13307)
Reported by: jcovert

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2008-09-03 13:27:14 +00:00
Mark Michelson
4eaa38b42b Blocked revisions 140752 via svnmerge
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r140752 | mmichelson | 2008-09-02 18:48:25 -0500 (Tue, 02 Sep 2008) | 14 lines

Merged revisions 140751 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r140751 | mmichelson | 2008-09-02 18:47:49 -0500 (Tue, 02 Sep 2008) | 6 lines

After adding the context checking to app_voicemail
for IMAP storage, I left out a crucial place to 
copy the context to the vm_state structure. This
is the correction.


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2008-09-02 23:50:11 +00:00
Mark Michelson
6edd73586e Blocking some revision. I guess I forgot to
commit this earlier. Oh well.



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2008-09-02 23:49:52 +00:00
Sean Bright
b1d895dba9 Merged revisions 140606 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r140606 | seanbright | 2008-09-02 14:15:54 -0400 (Tue, 02 Sep 2008) | 16 lines

Merged revisions 140605 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r140605 | seanbright | 2008-09-02 14:14:57 -0400 (Tue, 02 Sep 2008) | 8 lines

Make sure to use the correct length of the mohinterpret and mohsuggest
buffers when copying configuration values.

(closes issue #13336)
Reported by: decryptus_proformatique
Patches:
      chan_iax2_mohinterpret_mohsuggest_general_settings.patch uploaded by decryptus (license 555)

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2008-09-02 18:17:42 +00:00
Russell Bryant
08966a95de Merged revisions 140566 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r140566 | russell | 2008-09-02 10:11:53 -0500 (Tue, 02 Sep 2008) | 2 lines

Update instructions for getting libresample

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2008-09-02 15:12:24 +00:00
Russell Bryant
a81f609617 Blocked revisions 140563 via svnmerge
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r140563 | russell | 2008-09-02 10:09:20 -0500 (Tue, 02 Sep 2008) | 3 lines

I'm not sure how these files got to trunk (probably my fault), but they
should not be here

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2008-09-02 15:09:44 +00:00
Sean Bright
8076234eb5 Blocked revisions 140559 via svnmerge
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r140559 | seanbright | 2008-09-02 10:41:41 -0400 (Tue, 02 Sep 2008) | 4 lines

When a call is rejected because of call-limit, the channel driver is behaving
as expected, so we shouldn't report it as an error.  Change to LOG_NOTICE
instead.

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2008-09-02 14:42:26 +00:00
Mark Michelson
8767b7911b Blocked revisions 140433 via svnmerge
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r140433 | mmichelson | 2008-08-29 11:24:37 -0500 (Fri, 29 Aug 2008) | 10 lines

Allow for video files to be opened as well as
audio files.

(closes issue #13372)
Reported by: epicac
Patches:
      13372.patch uploaded by putnopvut (license 60)
Tested by: epicac


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2008-08-29 16:25:17 +00:00
Mark Michelson
f5004d1681 Blocked revisions 140422 via svnmerge
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r140422 | mmichelson | 2008-08-29 11:06:09 -0500 (Fri, 29 Aug 2008) | 20 lines

Merged revisions 140421 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r140421 | mmichelson | 2008-08-29 11:01:07 -0500 (Fri, 29 Aug 2008) | 12 lines

Add context checking when retrieving a vm_state.
This was causing a problem for people who had identically
named mailboxes in separate voicemail contexts.
This commit affects IMAP storage only.

(closes issue #13194)
Reported by: moliveras
Patches:
      13194.patch uploaded by putnopvut (license 60)
Tested by: putnopvut, moliveras


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2008-08-29 16:06:42 +00:00
Mark Michelson
eb27b67366 Blocked revisions 140418 via svnmerge
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r140418 | mmichelson | 2008-08-29 10:32:02 -0500 (Fri, 29 Aug 2008) | 18 lines

Merged revisions 140417 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r140417 | mmichelson | 2008-08-29 10:26:52 -0500 (Fri, 29 Aug 2008) | 10 lines

Fix SIP's parsing so that if a port is specified
in a string to Dial(), it is not ignored.

(closes issue #13355)
Reported by: acunningham
Patches:
      13355v2.patch uploaded by putnopvut (license 60)
Tested by: acunningham


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2008-08-29 15:32:31 +00:00
Tilghman Lesher
eb2623b1d9 Blocked revisions 140355 via svnmerge
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r140355 | tilghman | 2008-08-27 18:23:56 -0500 (Wed, 27 Aug 2008) | 2 lines

Oops

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2008-08-27 23:24:38 +00:00
Mark Michelson
b2f6976dbc Revert commit 140302. Should not be merging
changes like that into a release-candidate branch



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@140304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-27 20:15:29 +00:00
Mark Michelson
f94f437af6 Merged revisions 140301 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r140301 | mmichelson | 2008-08-27 15:11:22 -0500 (Wed, 27 Aug 2008) | 19 lines

Merged revisions 140299 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r140299 | mmichelson | 2008-08-27 14:49:20 -0500 (Wed, 27 Aug 2008) | 11 lines

Fix tag checking in get_sip_pvt_byid_locked when
in pedantic mode. The problem was that the wrong
tags would be compared depending on the direction
of the call.

(closes issue #13353)
Reported by: flefoll
Patches:
      chan_sip.c.br14.139015.patch-refer-pedantic uploaded by flefoll (license 244)


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2008-08-27 20:13:40 +00:00
Tilghman Lesher
5a8fb7cc81 Blocked revisions 140201 via svnmerge
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r140201 | tilghman | 2008-08-26 13:46:07 -0500 (Tue, 26 Aug 2008) | 2 lines

OpenBSD compat fix (reminded by mvanbaak on #asterisk-dev)

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2008-08-26 18:46:54 +00:00
Russell Bryant
0a4475f66a Merged revisions 140169 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r140169 | russell | 2008-08-26 13:11:49 -0500 (Tue, 26 Aug 2008) | 4 lines

Fix building menuselect-tree with PRINT_DIR set.  We _must_ use the --quiet
flag here, or else some arbitrary text will end up in the resulting
menuselect-tree file and things will explode.

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2008-08-26 18:12:08 +00:00
Tilghman Lesher
878517aa7c Blocked revisions 140167 via svnmerge
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r140167 | tilghman | 2008-08-26 13:05:58 -0500 (Tue, 26 Aug 2008) | 5 lines

Standardize the option names for consistency (but continue to work with the
existing names for backwards compatibility).
(closes issue #13370)
 Reported by: jsturtevant

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2008-08-26 18:06:34 +00:00
Russell Bryant
8a03188053 Blocked revisions 140061 via svnmerge
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r140061 | russell | 2008-08-26 11:10:06 -0500 (Tue, 26 Aug 2008) | 14 lines

Merged revisions 140060 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r140060 | russell | 2008-08-26 11:07:58 -0500 (Tue, 26 Aug 2008) | 6 lines

Fix some bogus scheduler usage in chan_sip.  This code used the return value
of a completely unrelated function to determine whether the scheduler should
be run or not.  This would have caused the scheduler to not run in cases where
it should have.  Also, leave a note about another scheduler issue that needs
to be addressed at some point.

........

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2008-08-26 16:10:28 +00:00
Steve Murphy
56ed3a4ce0 Blocked revisions 140057 via svnmerge
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r140057 | murf | 2008-08-26 09:57:49 -0600 (Tue, 26 Aug 2008) | 21 lines

(closes issue #13366)
Reported by: erousseau

This was a reasonable enhancement request, which was
easy to implement. Since it's an enhancement, it 
could only be applied to trunk.

Basically, for accounting where "initiated" seconds
are billed for, if the microseconds field on the end
time is greater than the microseconds field for the
answer time, add one second to the billsec field.

The implementation was requested by erousseau, and
I've implemented it as requested. I've updated the
CHANGES, the cdr.conf.sample, and the .h files
accordingly, to accept and set a flag for the
corresponding new option. cdr.c adds in the extra
second based on the usec fields if the option is
set. Tested, seems to be working fine.


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2008-08-26 15:59:45 +00:00
Russell Bryant
5b2bb04b40 Blocked revisions 140053 via svnmerge
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r140053 | russell | 2008-08-26 10:29:25 -0500 (Tue, 26 Aug 2008) | 23 lines

Merged revisions 140051 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r140051 | russell | 2008-08-26 10:27:23 -0500 (Tue, 26 Aug 2008) | 15 lines

Fix a race condition with the IAX scheduler thread.  A lock and condition are
used here to allow newly scheduled tasks to wake up the scheduler just in case
the new task needs to run sooner than the current wakeup time when the thread
is sleeping.  However, there was a race condition such that a newly scheduled
task would not properly wake up the scheduler or affect the wake up period.
The order of execution would have been:

  1) Scheduler thread determines wake up time of N ms.
  2) Another thread schedules a task and signals the condition, with an
     execution time of < N ms.
  3) Scheduler thread locks and goes to sleep for N ms.

By moving the sleep time determination to inside the critical section, this
possibility is avoided.

........

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2008-08-26 15:29:46 +00:00
Tilghman Lesher
c3845ff6b6 Blocked revisions 139981 via svnmerge
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r139981 | tilghman | 2008-08-25 18:13:32 -0500 (Mon, 25 Aug 2008) | 7 lines

Optional light colored background, for those who use black on white terminals.
(closes issue #13306)
 Reported by: Corydon76
 Patches: 
       20080814__bug13306__3.diff.txt uploaded by Corydon76 (license 14)
 Tested by: Corydon76, pkempgen

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2008-08-25 23:14:08 +00:00
Sean Bright
1b28d6f0c3 Merged revisions 139915 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r139915 | seanbright | 2008-08-25 17:32:10 -0400 (Mon, 25 Aug 2008) | 17 lines

Merged revisions 139909 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r139909 | seanbright | 2008-08-25 17:31:03 -0400 (Mon, 25 Aug 2008) | 9 lines

Some versions of awk (nawk, for example) don't like empty regular expressions
so be slightly more verbose.

(closes issue #13374)
Reported by: dougm
Patches:
      13374.diff uploaded by seanbright (license 71)
Tested by: dougm

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2008-08-25 21:33:21 +00:00
Terry Wilson
0de177bc47 Merged revisions 139870 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r139870 | twilson | 2008-08-25 15:59:58 -0500 (Mon, 25 Aug 2008) | 10 lines

Merged revisions 139869 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r139869 | twilson | 2008-08-25 15:46:10 -0500 (Mon, 25 Aug 2008) | 2 lines

Make SIPADDHEADER() propagate indefinitely

........

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2008-08-25 21:05:57 +00:00
Mark Michelson
7fd12fbedd Blocked revisions 139832 via svnmerge
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r139832 | mmichelson | 2008-08-25 12:24:02 -0500 (Mon, 25 Aug 2008) | 10 lines

Add output of variables to AgentRingNoAnswer manager
event if eventwhencalled is set to "vars" in 
queues.conf. Yay for consistency.

(closes issue #13369)
Reported by: srt
Patches:
      13369_agentringnoanswer_variables.diff uploaded by srt (license 378)


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2008-08-25 17:24:36 +00:00
Tilghman Lesher
0e3107d1ff Blocked revisions 139775 via svnmerge
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r139775 | tilghman | 2008-08-25 11:02:56 -0500 (Mon, 25 Aug 2008) | 7 lines

Realtime capabilities for the Find-Me-Follow-Me application.
(closes issue #13295)
 Reported by: Corydon76
 Patches: 
       20080813__followme_realtime_enabled.diff.txt uploaded by Corydon76 (license 14)
 Tested by: dferrer

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2008-08-25 16:03:32 +00:00
Steve Murphy
ee9f8015b6 Merged revisions 139770 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r139770 | murf | 2008-08-25 09:54:18 -0600 (Mon, 25 Aug 2008) | 17 lines

Merged revisions 139764 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r139764 | murf | 2008-08-25 09:33:14 -0600 (Mon, 25 Aug 2008) | 9 lines

This patch reverts the changes made via 139347, and 139635, as users
are seeing adverse difference. 

I will un-close 13251.

Back to the drawing board/ concept/ beginning/ whatever!



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2008-08-25 16:00:03 +00:00
Mark Michelson
4629d13de9 Blocked revisions 139771 via svnmerge
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r139771 | mmichelson | 2008-08-25 10:55:10 -0500 (Mon, 25 Aug 2008) | 19 lines

Blocked revisions 139769 via svnmerge

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r139769 | mmichelson | 2008-08-25 10:52:59 -0500 (Mon, 25 Aug 2008) | 12 lines

Fix the logic in config_text_file_save so that if an
UpdateConfig manager action is issued and the
file specified in DstFileName does not yet exist,
an error is not returned.

(closes issue #13341)
Reported by: vadim
Patches:
      13341.patch uploaded by putnopvut (license 60)
	  (with small modification from seanbright)


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2008-08-25 15:55:35 +00:00
Tilghman Lesher
ba6a7f8c98 Merged revisions 139707 via svnmerge from
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r139707 | tilghman | 2008-08-24 11:26:48 -0500 (Sun, 24 Aug 2008) | 2 lines

Memory leak

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2008-08-24 16:30:54 +00:00
Tilghman Lesher
5559f3c2fa Blocked revisions 139704 via svnmerge
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r139704 | tilghman | 2008-08-24 11:23:15 -0500 (Sun, 24 Aug 2008) | 2 lines

Eliminate open coding of ast_str

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2008-08-24 16:23:59 +00:00
Steve Murphy
41ed986072 Merged revisions 139662 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r139662 | murf | 2008-08-22 16:32:35 -0600 (Fri, 22 Aug 2008) | 14 lines

Merged revisions 139635 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r139635 | murf | 2008-08-22 16:24:02 -0600 (Fri, 22 Aug 2008) | 6 lines

I found some problems with the code I committed earlier, when
I merged them into trunk, so I'm coming back to clean up.
And, in the process, I found an error in the code I added
to trunk and 1.6.x, that I'll fix using this patch also.


........

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2008-08-22 22:35:14 +00:00
Steve Murphy
ee980c4a33 Merged revisions 139627 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r139627 | murf | 2008-08-22 16:03:13 -0600 (Fri, 22 Aug 2008) | 59 lines

Merged revisions 139347 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r139347 | murf | 2008-08-21 17:03:50 -0600 (Thu, 21 Aug 2008) | 47 lines


(closes issue #13251)
Reported by: sergee
Tested by: murf



THis is a bold move for a static release fix, but I wouldn't have
made it if I didn't feel confident (at least a *bit* confident)
that it wouldn't mess everyone up.

The reasoning goes something like this:

1. We simply cannot do anything with CDR's at the current point
(in pbx.c, after the __ast_pbx_run loop). It's way too late to
have any affect on the CDRs. The CDR is already posted and gone,
and the remnants have been cleared.

2. I was very much afraid that moving the running of the 'h'
extension down into the bridge code (where it would be now
practical to do it), would result in a lot more calls to the
'h' exten, so I implemented it as another exten under another
name, but found, to my pleasant surprise, that there was a 
1:1 correspondence to the running of the 'h' exten in the
pbx_run loop, and the new spot at the end of the bridge.
So, I ifdef'd out the current 'h' loop, and moved it into
the bridge code. The only difference I can see is the stuff
about the AST_PBX_KEEPALIVE, and hopefully, if this 
is still an important decision point, I can replicate it
if there are complaints. To be perfectly honest,
the KEEPALIVE situation is not totally clear to me,
and how it relates to a post-bridge situation is less
clear. I suspect the users will point out everything
in total clarity if this steps on anyone's toes!

3. I temporarily swap the bridge_cdr into the channel
before running the 'h' exten, which makes it possible
for users to edit the cdr before it goes out the door.
And, of course, with the endbeforehexten config var set,
the users can also get at the billsec/duration vals.
After the h exten finishes, the cdr is swapped back
and processing continues as normal.

Please, all who deal with CDR's, please test this version
of Asterisk, and file bug reports as appropriate!


........

I also made a little fix to the app_dial's 'e' option,
that is related to my updates.


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2008-08-22 22:09:31 +00:00
Mark Michelson
503289d053 Blocked revisions 139563 via svnmerge
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r139563 | mmichelson | 2008-08-22 15:20:58 -0500 (Fri, 22 Aug 2008) | 15 lines

The -1 return value from incomplete or improper
headers for the SipNotify manager command was
causing the current manager session to become
disconnected. Change the return value to 0 for
these cases.

Also change a test for a NULL pointer to be
ast_strlen_zero instead.

(closes issue #13351)
Reported by: Laureano
Patches:
      sipnotify_action_fix.patch uploaded by Laureano (license 265)


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2008-08-22 20:21:22 +00:00
Mark Michelson
df90016cf2 Blocked revisions 139558 via svnmerge
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r139558 | mmichelson | 2008-08-22 15:02:35 -0500 (Fri, 22 Aug 2008) | 9 lines

Add missing unique id to ParkedCallGiveUp and ParkedCallTimeOut
manager events

(closes issue #13358)
Reported by: srt
Patches:
      13358_parking_events.diff uploaded by srt (license 378)


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2008-08-22 20:03:31 +00:00
Mark Michelson
c9c22bc30f Merged revisions 139554 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r139554 | mmichelson | 2008-08-22 14:45:41 -0500 (Fri, 22 Aug 2008) | 16 lines

Merged revisions 139553 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r139553 | mmichelson | 2008-08-22 14:45:19 -0500 (Fri, 22 Aug 2008) | 8 lines

Fix compilation when DEBUG_THREAD_LOCALS is selected

(closes issue #13298)
Reported by: snuffy
Patches:
      bug13298_20080822.diff uploaded by snuffy (license 35)


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2008-08-22 19:46:16 +00:00
Mark Michelson
44afac9c8f Blocked revisions 139525 via svnmerge
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r139525 | mmichelson | 2008-08-22 14:24:26 -0500 (Fri, 22 Aug 2008) | 15 lines

Blocked revisions 139521 via svnmerge

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r139521 | mmichelson | 2008-08-22 14:23:02 -0500 (Fri, 22 Aug 2008) | 8 lines

Remove show_frame_stats_deprecated since it is not
used anywhere and causes build errors if building under
dev-mode with TRACE_FRAMES selected in menuselect.

(closes issue #13362)
Reported by: snuffy


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2008-08-22 19:25:10 +00:00
Mark Michelson
ad0ad7429e Merged revisions 139469 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r139469 | mmichelson | 2008-08-22 12:25:12 -0500 (Fri, 22 Aug 2008) | 11 lines

Merged revisions 139466 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r139466 | mmichelson | 2008-08-22 12:24:47 -0500 (Fri, 22 Aug 2008) | 3 lines

Fix the build. Thanks, mvanbaak!


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2008-08-22 17:25:34 +00:00
Mark Michelson
3c9ac8be44 Merged revisions 139457 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r139457 | mmichelson | 2008-08-22 11:58:21 -0500 (Fri, 22 Aug 2008) | 15 lines

Merged revisions 139456 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r139456 | mmichelson | 2008-08-22 11:57:38 -0500 (Fri, 22 Aug 2008) | 7 lines

Prevent a deadlock in chan_iax2 resulting from
incorrect locking order between iax2_pvt and
ast_channel structures.

AST-13


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2008-08-22 16:59:21 +00:00
Jeff Peeler
00f59da391 Merged revisions 139391 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r139391 | jpeeler | 2008-08-21 18:41:50 -0500 (Thu, 21 Aug 2008) | 11 lines

Merged revisions 139387 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r139387 | jpeeler | 2008-08-21 18:39:31 -0500 (Thu, 21 Aug 2008) | 3 lines

Fixes loop that could possibly never exit in the event of a channel never being able to be opened or specify after a restart.
(closes issue #11017)

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2008-08-21 23:46:39 +00:00
Philippe Sultan
9bd17e6a65 Merged revisions 139281 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r139281 | phsultan | 2008-08-21 11:55:31 +0200 (Thu, 21 Aug 2008) | 5 lines

Fix two memory leaks in chan_gtalk, thanks Eliel!
(closes issue #13310)
Reported by: eliel
Patches:
      chan_gtalk.c.patch uploaded by eliel (license 64)
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2008-08-21 10:02:32 +00:00
Russell Bryant
7d1bda1c07 Merged revisions 139215 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r139215 | russell | 2008-08-20 17:16:36 -0500 (Wed, 20 Aug 2008) | 19 lines

Merged revisions 139213 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r139213 | russell | 2008-08-20 17:14:35 -0500 (Wed, 20 Aug 2008) | 11 lines

Fix a crash in the ChanSpy application.  The issue here is that if you call
ChanSpy and specify a spy group, and sit in the application long enough looping
through the channel list, you will eventually run out of stack space and the
application with exit with a seg fault.  The backtrace was always inside of
a harmless snprintf() call, so it was tricky to track down.  However, it turned
out that the call to snprintf() was just the biggest stack consumer in this
code path, so it would always be the first one to hit the boundary.

(closes issue #13338)
Reported by: ruddy

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2008-08-20 22:17:44 +00:00
Jason Parker
3a6e5c4acc Blocked revisions 139210 via svnmerge
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r139210 | qwell | 2008-08-20 17:06:40 -0500 (Wed, 20 Aug 2008) | 7 lines

Fix output of sipshowpeer manager response.

(closes issue #13346)
Reported by: srt
Patches:
      13346_malformed_sip_show_peer_response.diff uploaded by srt (license 378)

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2008-08-20 22:07:27 +00:00
Shaun Ruffell
5a59e03452 Fix bug where the samples were not accurate when in G723 mode, which would
cause the timestamp field of the RTP header to be invalid.



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@139155 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-20 20:12:38 +00:00
Steve Murphy
76be7d91a9 Merged revisions 139083 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r139083 | murf | 2008-08-20 11:25:07 -0600 (Wed, 20 Aug 2008) | 20 lines

Merged revisions 139074 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r139074 | murf | 2008-08-20 11:14:55 -0600 (Wed, 20 Aug 2008) | 12 lines


(closes issue #13263)
Reported by: brainy
Tested by: murf

The specialized reset routine is tromping on the
flags field of the CDR. I made a change to not
reset the DISABLED bit. This should get rid of this
problem.



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2008-08-20 17:30:58 +00:00
Michiel van Baak
e2a327e2cc Blocked revisions 139020 via svnmerge
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r139020 | mvanbaak | 2008-08-20 18:16:53 +0200 (Wed, 20 Aug 2008) | 4 lines

fix unholding phones after hangup on older cisco phones.

Patch by wedhorn.

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2008-08-20 16:18:05 +00:00
Mark Michelson
f084b49eed Merged revisions 139016 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r139016 | mmichelson | 2008-08-20 10:38:47 -0500 (Wed, 20 Aug 2008) | 14 lines

Merged revisions 139015 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r139015 | mmichelson | 2008-08-20 10:37:56 -0500 (Wed, 20 Aug 2008) | 6 lines

sip_read should properly handle a NULL return from sip_rtp_read.

(closes issue #13257)
Reported by: travishein


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2008-08-20 15:39:19 +00:00
Mark Michelson
c6c11dd423 Blocked revisions 138943 via svnmerge
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r138943 | mmichelson | 2008-08-19 18:19:40 -0500 (Tue, 19 Aug 2008) | 19 lines

Merged revisions 138942 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r138942 | mmichelson | 2008-08-19 18:17:17 -0500 (Tue, 19 Aug 2008) | 11 lines

Reset agent_pvt variables back to the values in agents.conf
(from what the corresponding channel variables were set to)
when the agent logs out.

(closes issue #13098)
Reported by: davidw
Patches:
      20080731__issue13098_agent_ackcall_not_reset.diff uploaded by bbryant (license 36)
Tested by: davidw


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2008-08-19 23:20:50 +00:00
Mark Michelson
a63274bf29 Manually add revision 138887 from trunk to the 1.6.0
branch. I had misunderstood the policy for when to merge
to 1.6.0 since it moved to rc status.



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2008-08-19 19:08:51 +00:00
Mark Michelson
502fede002 Blocked revisions 138887 via svnmerge
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r138887 | mmichelson | 2008-08-19 13:52:04 -0500 (Tue, 19 Aug 2008) | 31 lines

Merged revisions 138886 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r138886 | mmichelson | 2008-08-19 13:50:53 -0500 (Tue, 19 Aug 2008) | 23 lines

Add a lock and unlock prior to the destruction of the chanspy_ds
lock to ensure that no other threads still have it locked. While
this should not happen under normal circumstances, it appears that
if the spyer and spyee hang up at nearly the same time, the following
may occur.

1. ast_channel_free is called on the spyee's channel.
2. The chanspy datastore is removed from the spyee's channel in 
   ast_channel_free.
3. In the spyer's thread, the spyer attempts to remove and destroy the datastore
   from the spyee channel, but the datastore has already been removed in step 2, 
   so the spyer continues in the code.
4. The spyee's thread continues and calls the datastore's destroy callback, 
   chanspy_ds_destroy. This involves locking the chanspy_ds.
5. Now the spyer attempts to destroy the chanspy_ds lock. The problem is that in step 4, 
   the spyee has locked this lock, meaning that the spyer is attempting to destroy a lock 
   which is currently locked by another thread.

The backtrace provided in issue #12969 supports the idea that this is possible
(and has even occurred). This commit does not close the issue, but should help
in preventing one type of crash associated with the use of app_chanspy.


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2008-08-19 18:54:19 +00:00
Michiel van Baak
b65c53337c Blocked revisions 138851 via svnmerge
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r138851 | mvanbaak | 2008-08-19 18:56:50 +0200 (Tue, 19 Aug 2008) | 8 lines

chan_skinny now respects callwaiting=no

(closes issue #12691)
Reported by: sbisker
Patches:
      callwaitingv1.diff uploaded by wedhorn (license 30)
Tested by: wedhorn on old skinny phones, mvanbaak on 7960 and 7905 with latest firmware

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2008-08-19 16:58:06 +00:00
Steve Murphy
1ea8431ada Merged revisions 138845 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r138845 | murf | 2008-08-19 10:31:24 -0600 (Tue, 19 Aug 2008) | 1 line

Oops. put a decl in a generated file. My bad, but fixed now.
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2008-08-19 16:38:21 +00:00
Steve Murphy
1f651e547e Merged revisions 138815 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r138815 | murf | 2008-08-19 09:59:12 -0600 (Tue, 19 Aug 2008) | 19 lines

These changes are in regards to bug 13249, where users are being surprised by the changes made
to the Set app in trunk/1.6.x, as they come from the 1.4 world. They are only bitten if
they write their AEL dialplan in the 1.4 world, and then carry it over to a trunk/1.6.x 
installation where a "make samples" was executed, or where they hand-edited the 
asterisk.conf file and added the [compat] category with app_set = 1.6 (or higher).

(this commit does not totally solve 13249, at least not yet)

The change involves issueing a single warning while the AEL file is loading, if:
 1. app_set is present in the config file, and set to 1.6 or higher.
 2. there are double quotes in an assignment statement (eg x = "hi there";)
 3. the warning was not already issued.

The standalone app, aelparse, does not (yet) issue this warning. I'd have to
have it read in the asterisk.conf file, and that's a bit of hassle. I'll add
it if users request it, tho.



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2008-08-19 16:36:22 +00:00
Sean Bright
fe6ed321c8 Merged revisions 138778-138780 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r138778 | seanbright | 2008-08-18 20:08:27 -0400 (Mon, 18 Aug 2008) | 1 line

While we're at it, make this machine parseable too.
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r138779 | seanbright | 2008-08-18 20:09:38 -0400 (Mon, 18 Aug 2008) | 1 line

And remove code we don't need anymore.
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r138780 | seanbright | 2008-08-18 20:10:56 -0400 (Mon, 18 Aug 2008) | 1 line

Let it compile now, too (woops)
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2008-08-19 00:15:28 +00:00
Sean Bright
b688efae65 Merged revisions 138775 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r138775 | seanbright | 2008-08-18 19:42:36 -0400 (Mon, 18 Aug 2008) | 3 lines

Change event header to RegistrationTime to be more consistent (and avoid
breaking existing frameworks).  Pointed out by Laureano on #asterisk-dev.

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2008-08-18 23:43:54 +00:00
Mark Michelson
1633490379 Blocked revisions 138694 via svnmerge
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r138694 | mmichelson | 2008-08-18 15:23:11 -0500 (Mon, 18 Aug 2008) | 10 lines

Change the queue timeout priority logic into less ugly
and confusing code pieces. Clarify the logic within
queues.conf.sample.

(closes issue #12690)
Reported by: atis
Patches:
      queue_timeoutpriority.patch uploaded by atis (license 242)


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2008-08-18 20:23:53 +00:00
Mark Michelson
ac72baa7b5 Merged revisions 138687 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r138687 | mmichelson | 2008-08-18 15:04:10 -0500 (Mon, 18 Aug 2008) | 18 lines

Merged revisions 138685 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r138685 | mmichelson | 2008-08-18 15:01:14 -0500 (Mon, 18 Aug 2008) | 10 lines

Change the inequalities used in app_queue with regards
to timeouts from being strict to non-strict for more
accuracy.

(closes issue #13239)
Reported by: atis
Patches:
      app_queue_timeouts_v2.patch uploaded by atis (license 242)


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2008-08-18 20:05:09 +00:00
Jason Parker
133e628462 Merged revisions 138631 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r138631 | qwell | 2008-08-18 10:54:07 -0500 (Mon, 18 Aug 2008) | 1 line

Remove option that isn't valid here.
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2008-08-18 15:54:40 +00:00
Jeff Peeler
108df903c9 Merged revisions 138518 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r138518 | jpeeler | 2008-08-17 21:13:04 -0500 (Sun, 17 Aug 2008) | 1 line

add missing define for SS7 in dahdi_restart
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2008-08-18 02:14:02 +00:00
Sean Bright
8dcca4a2b0 Merged revisions 138482 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r138482 | seanbright | 2008-08-17 10:12:11 -0400 (Sun, 17 Aug 2008) | 6 lines

Move Uniqueid to the end of the event for those that rely on the position
of the name/value pairs, pointed out by snuffy-home on #asterisk-commits.

For those of you who rely on the position of name/value pairs in manager
events... stop... that is why associative arrays were invented.

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2008-08-17 14:14:46 +00:00
Sean Bright
141673d440 Merged revisions 138479 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r138479 | seanbright | 2008-08-17 09:51:08 -0400 (Sun, 17 Aug 2008) | 7 lines

Add Uniqueid header to ParkedCall manager event.

(closes issue #13323)
Reported by: srt
Patches:
      13323_unique_id_for_parkedcalls_event.diff uploaded by srt (license 378)

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2008-08-17 14:00:54 +00:00
Sean Bright
e509e5b79f Merged revisions 138476 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r138476 | seanbright | 2008-08-17 09:40:36 -0400 (Sun, 17 Aug 2008) | 7 lines

Add missing colons to RTCPReceived and RTCPSent manager events.

(closes issue #13319)
Reported by: srt
Patches:
      13319_rtcp_manager_event_headers.diff uploaded by srt (license 378)

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2008-08-17 13:41:56 +00:00
Sean Bright
2ebec214f2 Merged revisions 138473 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r138473 | seanbright | 2008-08-17 09:31:54 -0400 (Sun, 17 Aug 2008) | 7 lines

Fix the output of the JitterBufStats manager event.

(closes issue #13324)
Reported by: srt
Patches:
      13324_missing_nl_in_jitterbufstats_event_2.diff uploaded by srt (license 378)

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2008-08-17 13:33:21 +00:00
Sean Bright
fe274af25b Merged revisions 138442 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r138442 | seanbright | 2008-08-16 12:40:43 -0400 (Sat, 16 Aug 2008) | 4 lines

Since it's introduction in revision 3497, cdr_tds has *never* read
the port configuration option from cdr_tds.conf.  So go ahead and
remove it from the sample config.

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2008-08-16 16:41:52 +00:00
Tilghman Lesher
8a31f07fe5 Merged revisions 138412 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r138412 | tilghman | 2008-08-16 08:07:08 -0500 (Sat, 16 Aug 2008) | 2 lines

Fix compilation warnings (found with dev-mode)

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2008-08-16 13:07:52 +00:00
Tilghman Lesher
a45a011655 Blocked revisions 138409 via svnmerge
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r138409 | tilghman | 2008-08-16 07:52:06 -0500 (Sat, 16 Aug 2008) | 3 lines

Also make sure hinting won't crash on reload.
(Closes issue #13312)

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2008-08-16 12:53:12 +00:00
Jeff Peeler
f23919f30b Merged revisions 138361 via svnmerge from
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r138361 | jpeeler | 2008-08-15 20:13:26 -0500 (Fri, 15 Aug 2008) | 9 lines

Merged revisions 138360 via svnmerge from 
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r138360 | jpeeler | 2008-08-15 20:12:18 -0500 (Fri, 15 Aug 2008) | 1 line

fixes use count to properly decrement if an active dahdi channel is destroyed allowing module to be unloaded
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2008-08-16 01:14:11 +00:00
Jeff Peeler
47a7ff76f7 Merged revisions 138311 via svnmerge from
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r138311 | jpeeler | 2008-08-15 18:46:09 -0500 (Fri, 15 Aug 2008) | 20 lines

Merged revisions 138119,138151,138238 via svnmerge from 
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r138119 | jpeeler | 2008-08-15 14:21:51 -0500 (Fri, 15 Aug 2008) | 4 lines

Fixes the dahdi restart functionality. Dahdi restart allows one to restart all DAHDI channels, even if they are currently in use. This is different from unloading and then loading the module since unloading requires the use count to be zero. Reloading the module is different in that the signalling is not changed from what it was originally configured. Also, this fixes not closing all the file descriptors for D-channels upon module unload (which would prevent loading the module afterwards).

(closes issue #11017)

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r138151 | jpeeler | 2008-08-15 14:41:29 -0500 (Fri, 15 Aug 2008) | 1 line

declared static mutexes using AST_MUTEX_DEFINE_STATIC macro
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r138238 | jpeeler | 2008-08-15 16:28:26 -0500 (Fri, 15 Aug 2008) | 1 line

initialize condition variable ss_thread_complete using ast_cond_init
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2008-08-16 00:27:27 +00:00
Tilghman Lesher
eec03bcf40 Merged revisions 138260 via svnmerge from
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r138260 | tilghman | 2008-08-15 17:54:57 -0500 (Fri, 15 Aug 2008) | 16 lines

Merged revisions 138258 via svnmerge from 
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r138258 | tilghman | 2008-08-15 17:33:42 -0500 (Fri, 15 Aug 2008) | 8 lines

More fixes for realtime peers.
(closes issue #12921)
 Reported by: Nuitari
 Patches: 
       20080804__bug12921.diff.txt uploaded by Corydon76 (license 14)
       20080815__bug12921.diff.txt uploaded by Corydon76 (license 14)
 Tested by: Corydon76

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2008-08-15 23:03:06 +00:00
Tilghman Lesher
f37a7ab9eb Merged revisions 138206 via svnmerge from
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r138206 | tilghman | 2008-08-15 15:35:24 -0500 (Fri, 15 Aug 2008) | 4 lines

Remove deprecated syntax from sample config file
(closes issue #13314)
 Reported by: kue

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2008-08-15 20:36:35 +00:00
Jeff Peeler
ffff97ded1 rename all zfd instances in chan_dahdi to dfd to match 1.4 (left over from DAHDI transition)
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2008-08-15 20:20:30 +00:00
Jeff Peeler
907dd0a233 Blocked revisions 138155 via svnmerge
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r138155 | jpeeler | 2008-08-15 15:12:19 -0500 (Fri, 15 Aug 2008) | 1 line

rename all zfd instances in chan_dahdi to dfd to match 1.4 (left over from DAHDI transition)
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Tilghman Lesher
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r138148 | tilghman | 2008-08-15 14:36:11 -0500 (Fri, 15 Aug 2008) | 2 lines

Change free to ast_free_ptr, too

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Tilghman Lesher
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r138124 | tilghman | 2008-08-15 14:22:48 -0500 (Fri, 15 Aug 2008) | 4 lines

e->data can be NULL, so use the safe version of ast_strdup()
(closes issue #13312)
 Reported by: pj

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Tilghman Lesher
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r138086 | tilghman | 2008-08-15 13:02:15 -0500 (Fri, 15 Aug 2008) | 2 lines

regseconds is actually stored as the epoch time, not registration length

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2008-08-15 18:03:04 +00:00
Russell Bryant
43b54c74ec Merged revisions 138028 via svnmerge from
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r138028 | russell | 2008-08-15 10:09:46 -0500 (Fri, 15 Aug 2008) | 17 lines

Merged revisions 138027 via svnmerge from 
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r138027 | russell | 2008-08-15 10:07:16 -0500 (Fri, 15 Aug 2008) | 9 lines

Ensure that when a hangup occurs in autoservice, that a hangup frame gets
properly deferred to be read from the channel owner when it gets taken out
of autoservice.

(closes issue #12874)
Reported by: dimas
Patches: 
      v1-12874.patch uploaded by dimas (license 88)

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Tilghman Lesher
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r138024 | tilghman | 2008-08-15 10:03:32 -0500 (Fri, 15 Aug 2008) | 16 lines

Merged revisions 138023 via svnmerge from 
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r138023 | tilghman | 2008-08-15 09:51:12 -0500 (Fri, 15 Aug 2008) | 8 lines

Additional check for more string specifiers than arguments.
(closes issue #13299)
 Reported by: adomjan
 Patches: 
       20080813__bug13299.diff.txt uploaded by Corydon76 (license 14)
       func_strings.c-sprintf.patch uploaded by adomjan (license 487)
 Tested by: adomjan

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2008-08-15 15:04:14 +00:00
Russell Bryant
74c4b84e73 Merged revisions 137987 via svnmerge from
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r137987 | russell | 2008-08-14 17:43:15 -0500 (Thu, 14 Aug 2008) | 2 lines

Fix a bashism that causes an error when trying to build the pdf on ubuntu

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2008-08-14 22:43:36 +00:00
Sean Bright
2bc89083d9 Merged revisions 137933 via svnmerge from
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r137933 | seanbright | 2008-08-14 14:47:28 -0400 (Thu, 14 Aug 2008) | 8 lines

Fix memory leak in cdr_sqlite3_custom.

(closes issue #13304)
Reported by: eliel
Patches:
      sqlite.patch uploaded by eliel (license 64)
      (Slightly modified by me)

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2008-08-14 18:48:39 +00:00
Russell Bryant
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r137901 | russell | 2008-08-14 13:12:16 -0500 (Thu, 14 Aug 2008) | 1 line

Prepare for adding 1.6.2 changes
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Tilghman Lesher
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r137851 | tilghman | 2008-08-14 12:01:00 -0500 (Thu, 14 Aug 2008) | 9 lines

Blocked revisions 137850 via svnmerge

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r137850 | tilghman | 2008-08-14 11:55:20 -0500 (Thu, 14 Aug 2008) | 2 lines

Oops, wrong direction

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Tilghman Lesher
af9f8bf291 Merged revisions 137848 via svnmerge from
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r137848 | tilghman | 2008-08-14 11:52:43 -0500 (Thu, 14 Aug 2008) | 17 lines

Merged revisions 137847 via svnmerge from 
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r137847 | tilghman | 2008-08-14 11:47:30 -0500 (Thu, 14 Aug 2008) | 9 lines

When creating the secondary subchannel name, it is necessary to compare to
the existing channel name without the "Zap/" or "DAHDI/" prefix, since our
test string is also without that prefix.
(closes issue #13027)
 Reported by: dferrer
 Patches: 
       chan_zap-1.4.21.1_fix2.patch uploaded by dferrer (license 525)
       (Slightly modified by me, to compensate for both names)

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2008-08-14 16:54:22 +00:00
Jason Parker
17f520666d Merged revisions 137812 via svnmerge from
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r137812 | qwell | 2008-08-14 10:32:16 -0500 (Thu, 14 Aug 2008) | 8 lines

Make sure we set the socket port, so we don't try to use <ip address>:0.

(closes issue #13255)
Reported by: falves11
Patches:
      13255-socketport.diff uploaded by qwell (license 4)
Tested by: falves11

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2008-08-14 15:37:35 +00:00
Russell Bryant
42d49d655d Merged revisions 137732 via svnmerge from
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r137732 | russell | 2008-08-14 09:15:50 -0500 (Thu, 14 Aug 2008) | 12 lines

Merged revisions 137731 via svnmerge from 
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r137731 | russell | 2008-08-14 09:05:23 -0500 (Thu, 14 Aug 2008) | 4 lines

Comments in this config file were aligned only if your tab size was set to 8.
So, convert tabs to spaces so that things should be aligned regardless of what
tab size you use in your editor.

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2008-08-14 15:20:37 +00:00
Sean Bright
9968e9a09b Merged revisions 137780 via svnmerge from
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r137780 | seanbright | 2008-08-14 11:03:03 -0400 (Thu, 14 Aug 2008) | 8 lines

If we detect that we are no longer connected, try to reconnect a few times
before giving up.  This relies on the timeout settings in the freetds.conf
file and, unfortunately, on a recent version of FreeTDS (0.82 or newer).

I either need to change the current execs to be non-blocking (which I do
not want to do) or we have to force people to run with the latest and
greatest of FreeTDS.  I'm on the fence...

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Kevin P. Fleming
5aa7e82924 Merged revisions 137680 via svnmerge from
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r137680 | kpfleming | 2008-08-13 21:03:47 -0500 (Wed, 13 Aug 2008) | 9 lines

Merged revisions 137679 via svnmerge from 
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r137679 | kpfleming | 2008-08-13 21:03:04 -0500 (Wed, 13 Aug 2008) | 1 line

forgot one module name that changed
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2008-08-14 02:04:05 +00:00
Kevin P. Fleming
666ca122e8 Merged revisions 137640 via svnmerge from
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r137640 | kpfleming | 2008-08-13 18:00:37 -0500 (Wed, 13 Aug 2008) | 1 line

make this script actually work
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2008-08-13 23:00:55 +00:00
Kevin P. Fleming
d0c3c48a54 Merged revisions 137627 via svnmerge from
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r137627 | kpfleming | 2008-08-13 17:33:32 -0500 (Wed, 13 Aug 2008) | 9 lines

Merged revisions 137530 via svnmerge from 
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r137530 | kpfleming | 2008-08-13 15:46:19 -0500 (Wed, 13 Aug 2008) | 1 line

add document describing what users will need to be aware of when upgrading to this version and using DAHDI
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2008-08-13 22:34:25 +00:00
Jason Parker
c6681ec41d Merged revisions 137532 via svnmerge from
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r137532 | qwell | 2008-08-13 16:08:58 -0500 (Wed, 13 Aug 2008) | 8 lines

Correctly end locally ended calls.

(closes issue #12170)
Reported by: pj
Patches:
      20080702__issue12170_clear_pendinginvite.diff uploaded by bbryant (license 36)
Tested by: bbryant, pabelanger

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Jason Parker
84d0f2de87 Merged revisions 137496 via svnmerge from
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r137496 | qwell | 2008-08-13 15:05:50 -0500 (Wed, 13 Aug 2008) | 6 lines

Add FAXMODE variable with what fax transport was used.

(closes issue #13252)
Patches:
      v1-13252.patch uploaded by dimas (license 88)

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Tilghman Lesher
54f95d4596 Blocked revisions 137456 via svnmerge
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r137456 | tilghman | 2008-08-13 12:36:15 -0500 (Wed, 13 Aug 2008) | 6 lines

Convert deprecated routines to the new names.
(closes issue #13297)
 Reported by: snuffy
 Patches: 
       bug13297_20080814.diff uploaded by snuffy (license 35)

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Sean Bright
c4cf3534a5 Merged revisions 137406 via svnmerge from
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r137406 | seanbright | 2008-08-13 10:41:49 -0400 (Wed, 13 Aug 2008) | 9 lines

Merged revisions 137405 via svnmerge from 
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r137405 | seanbright | 2008-08-13 10:33:49 -0400 (Wed, 13 Aug 2008) | 1 line

Update docs to reflect the change to cdr_tds
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Sean Bright
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r137403 | seanbright | 2008-08-13 10:22:47 -0400 (Wed, 13 Aug 2008) | 1 line

Use the ast_vasprintf macro instead of vasprintf directly.
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r137349 | seanbright | 2008-08-12 20:11:13 -0400 (Tue, 12 Aug 2008) | 15 lines

Blocked revisions 137348 via svnmerge

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r137348 | seanbright | 2008-08-12 20:10:35 -0400 (Tue, 12 Aug 2008) | 8 lines

Bring cdr_tds in line with the other CDR backends and have it try to store
CDR(userfield) if it is set.  The new behavior is to check for the userfield
column on module load, and if it exists, we will store CDR(userfield) when
CDRs are written.  A similar patch already went into trunk and 1.6.0.

(closes issue #13290)
Reported by: falves11

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r137301 | russell | 2008-08-12 14:48:38 -0500 (Tue, 12 Aug 2008) | 2 lines

Grammar hax from Qwell

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Russell Bryant
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r137299 | russell | 2008-08-12 14:40:35 -0500 (Tue, 12 Aug 2008) | 3 lines

Note that developer documentation belongs in doxygen, and not integrated with
the user manual stuff in doc/tex/.

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Russell Bryant
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r137239 | russell | 2008-08-11 11:14:29 -0500 (Mon, 11 Aug 2008) | 2 lines

Make PRINT_DIR work as advertised.

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Sean Bright
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r137203 | seanbright | 2008-08-11 10:25:15 -0400 (Mon, 11 Aug 2008) | 7 lines

Log the userfield CDR variable like the other CDR backends, assuming the
column is actually there.  If it's not, we still log everything else as
before.

(closes issue #13281)
Reported by: falves11

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Tilghman Lesher
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r137150 | tilghman | 2008-08-10 19:25:28 -0500 (Sun, 10 Aug 2008) | 13 lines

Merged revisions 137138 via svnmerge from 
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r137138 | tilghman | 2008-08-10 19:20:38 -0500 (Sun, 10 Aug 2008) | 5 lines

Deallocate database connection handle on disconnect, as we allocate another
one on connect.
(closes issue #13271)
 Reported by: dveiga

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Sean Bright
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r137112 | seanbright | 2008-08-10 17:10:04 -0400 (Sun, 10 Aug 2008) | 3 lines

Fix this again so we can compile with shadow warnings enabled and IMAP chosen
in voicemail.

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Sean Bright
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r137110 | seanbright | 2008-08-10 16:57:25 -0400 (Sun, 10 Aug 2008) | 3 lines

That's all, folks.  Not going to update the Makefile until res_jabber is
converted (snuffy, you there? :))

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3ca6fcdb0c Blocked revisions 137089 via svnmerge
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r137089 | seanbright | 2008-08-10 16:23:50 -0400 (Sun, 10 Aug 2008) | 3 lines

Another batch of files from RSW.  The remaining apps and a few more
files from main/

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2008-08-10 20:24:24 +00:00
Sean Bright
06dfd61855 Blocked revisions 137082 via svnmerge
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r137082 | seanbright | 2008-08-10 15:35:50 -0400 (Sun, 10 Aug 2008) | 2 lines

Another big chunk of changes from the RSW branch.  Bunch of stuff from main/

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2008-08-10 19:36:20 +00:00
Sean Bright
c904c89b58 Blocked revisions 137055 via svnmerge
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r137055 | seanbright | 2008-08-10 10:45:25 -0400 (Sun, 10 Aug 2008) | 3 lines

More RSW merges.  Everything from apps/ except for the big offenders
app_voicemail and app_queue.

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2008-08-10 14:47:03 +00:00
Sean Bright
27cb0935bf Blocked revisions 137028 via svnmerge
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r137028 | seanbright | 2008-08-09 20:47:56 -0400 (Sat, 09 Aug 2008) | 1 line

All of the res/ stuff (other than res_jabber) from the RSW branch.
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2008-08-10 00:48:37 +00:00
Tilghman Lesher
80d244e8ec Merged revisions 136947 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r136947 | tilghman | 2008-08-09 10:26:27 -0500 (Sat, 09 Aug 2008) | 18 lines

Merged revisions 136946 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r136946 | tilghman | 2008-08-09 10:25:36 -0500 (Sat, 09 Aug 2008) | 10 lines

Merged revisions 136945 via svnmerge from 
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r136945 | tilghman | 2008-08-09 10:24:36 -0500 (Sat, 09 Aug 2008) | 2 lines

Regression fixes for Solaris

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2008-08-09 15:27:11 +00:00
Sean Bright
b8d6ca0b62 Blocked revisions 136917 via svnmerge
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r136917 | seanbright | 2008-08-09 10:12:34 -0400 (Sat, 09 Aug 2008) | 2 lines

More RSW merges.  This should do it for the channels/ dir.

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Sean Bright
7df6d764db Blocked revisions 136890 via svnmerge
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r136890 | seanbright | 2008-08-09 09:13:48 -0400 (Sat, 09 Aug 2008) | 2 lines

Biggest offender?  chan_dahdi.c!  More RSW merging.

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Sean Bright
d50d40f7fc Blocked revisions 136888 via svnmerge
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r136888 | seanbright | 2008-08-09 08:53:57 -0400 (Sat, 09 Aug 2008) | 4 lines

Merge more changes from the resolve-shadow-warnings branch (henceforth known
as RSW since i am too lazy to keep typing it all out).  This time a few of
the channels.

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2008-08-09 12:54:30 +00:00
Tilghman Lesher
cbf448be6f Merged revisions 136859 via svnmerge from
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r136859 | tilghman | 2008-08-08 20:15:38 -0500 (Fri, 08 Aug 2008) | 4 lines

Update documentation as to the behavior of AGI in 1.6.0 and higher.  Also, add
an OOB message that answers the question of, if AGI no longer shuts down the
connection on hangup, how will FastAGI know when to stop processing the call?

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2008-08-09 01:16:25 +00:00
Sean Bright
2c6408c09f Blocked revisions 136819 via svnmerge
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r136819 | seanbright | 2008-08-08 14:19:33 -0400 (Fri, 08 Aug 2008) | 3 lines

Bring in the configure and makeopts jazz for -Wshadow, but don't add it to the
Makefile yet.

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2008-08-08 18:20:11 +00:00
Mark Michelson
a6df06e84e Merged revisions 136784 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r136784 | mmichelson | 2008-08-08 10:31:31 -0500 (Fri, 08 Aug 2008) | 3 lines

Fix compilation for ODBC voicemail


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2008-08-08 15:33:47 +00:00
Steve Murphy
52d466de03 Merged revisions 136746 via svnmerge from
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r136746 | murf | 2008-08-07 18:48:35 -0600 (Thu, 07 Aug 2008) | 40 lines

Merged revisions 136726 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r136726 | murf | 2008-08-07 18:15:34 -0600 (Thu, 07 Aug 2008) | 32 lines


(closes issue #13236)
Reported by: korihor

Wow, this one was a challenge!

I regrouped and ran a new strategy for
setting the ~~MACRO~~ value; I set it once
per extension, up near the top. It is only
set if there is a switch in the extension.

So, I had to put in a chunk of code to detect
a switch in the pval tree.

I moved the code to insert the set of ~~exten~~
up to the beginning of the gen_prios routine, 
instead of down in the switch code.

I learned that I have to push the detection
of the switches down into the code, so everywhere
I create a new exten in gen_prios, I make sure
to pass onto it the values of the mother_exten
first, and the exten next.

I had to add a couple fields to the exten
struct to accomplish this, in the ael_structs.h
file. The checked field makes it so we don't
repeat the switch search if it's been done.

I also updated the regressions.


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2008-08-08 06:45:43 +00:00
Tilghman Lesher
1222f82205 Merged revisions 136751 via svnmerge from
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r136751 | tilghman | 2008-08-07 21:34:17 -0500 (Thu, 07 Aug 2008) | 2 lines

Removing bad properties

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2008-08-08 02:36:26 +00:00
Mark Michelson
4a5ac6487f This is weird. Either SVN or vim tabbed
a bunch of functions over one level during
a merge.



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2008-08-07 23:42:34 +00:00
Mark Michelson
745f28879f Merged revisions 136722 via svnmerge from
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r136722 | mmichelson | 2008-08-07 18:39:50 -0500 (Thu, 07 Aug 2008) | 3 lines

Remove one last batch of debug messages


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2008-08-07 23:40:47 +00:00
Mark Michelson
8166a7a04d Blocked revisions 136720 via svnmerge
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r136720 | mmichelson | 2008-08-07 18:38:08 -0500 (Thu, 07 Aug 2008) | 6 lines

Fix build for non-IMAP storage and get rid of some
debug messages.
Thanks to eliel for alerting me.
No thanks to buildbot.


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2008-08-07 23:38:30 +00:00
Mark Michelson
5567deefc9 Merged revisions 136715 via svnmerge from
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r136715 | mmichelson | 2008-08-07 17:25:50 -0500 (Thu, 07 Aug 2008) | 18 lines

Merging the imap_consistency_trunk branch to
trunk.

For an explanation of what "imap_consistency" is,
please see svn revision 134223 to the 1.4 branch.

Coincidentally, this also fixes a recent bug report
regarding the inability to save messages to the new
folder when using IMAP storage since they will would
be flagged as "seen" and not be recognized as new
messages.


(closes issue #13234)
Reported by: jaroth



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2008-08-07 23:26:24 +00:00
Michiel van Baak
bb5194554e Blocked revisions 136679 via svnmerge
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r136679 | mvanbaak | 2008-08-07 23:19:39 +0200 (Thu, 07 Aug 2008) | 8 lines

show correct called party id and also store this to the 'placed calls' list once the call is connected.

(closes issue #13180)
Reported by: pj
Patches:
      2008080700_skinny_calledpartyid.diff uploaded by mvanbaak (license 7)
Tested by: mvanbaak, pj

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2008-08-07 21:22:26 +00:00
Shaun Ruffell
732659ef67 Removing code that was commented out.
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2008-08-07 20:41:06 +00:00
Shaun Ruffell
7fbba1d580 Updated codec_dahdi to use the transcoder interface in the DAHDI.
(Issue: DAHDI-42)



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2008-08-07 20:38:38 +00:00
Mark Michelson
eee3091cbf Merged revisions 136660 via svnmerge from
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r136660 | mmichelson | 2008-08-07 15:25:43 -0500 (Thu, 07 Aug 2008) | 4 lines

Bump a LOG_NOTICE message to LOG_DEBUG since it appears
once for every bridged call


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Mark Michelson
f3097eb1af Merged revisions 136635 via svnmerge from
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r136635 | mmichelson | 2008-08-07 14:58:32 -0500 (Thu, 07 Aug 2008) | 5 lines

Don't allow Answer() to accept a negative argument.
Negative argument means an infinite delay and we
don't want that.


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2008-08-07 19:59:00 +00:00
Mark Michelson
23bcd9b4a1 Merged revisions 136633 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r136633 | mmichelson | 2008-08-07 14:54:27 -0500 (Thu, 07 Aug 2008) | 7 lines

Fix a calculation error I had made in the poll. The poll
would reset to 500 ms every time a non-voice frame
was received. The total time we poll should be 500 ms, so
now we save the amount of time left after the poll returned
and use that as our argument for the next call to poll


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2008-08-07 19:54:47 +00:00
Mark Michelson
4ff12eb20c Merged revisions 136631 via svnmerge from
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r136631 | mmichelson | 2008-08-07 14:36:46 -0500 (Thu, 07 Aug 2008) | 13 lines

Scrap the 500 ms delay when Asterisk auto-answers a channel.
Instead, poll the channel until receiving a voice frame. The
cap on this poll is 500 ms.

The optional delay is still allowable in the Answer() application,
but the delay has been moved back to its original position, after
the call to the channel's answer callback. The poll for the voice
frame will not happen if a delay is specified when calling Answer().

(closes issue #12708)
Reported by: kactus


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2008-08-07 19:37:48 +00:00
Richard Mudgett
bca2b8aea0 Merged revisions 136594 via svnmerge from
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r136594 | rmudgett | 2008-08-07 14:01:03 -0500 (Thu, 07 Aug 2008) | 13 lines

Merged revisions 136241 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r136241 | rmudgett | 2008-08-06 16:18:53 -0500 (Wed, 06 Aug 2008) | 5 lines

*  The allowed_bearers setting in misdn.conf misspelled one
of its options: digital_restricted.
*  Fixed some other spelling errors and typos.


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2008-08-07 19:19:49 +00:00
Kevin P. Fleming
eac67f84f4 Merged revisions 136542 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r136542 | kpfleming | 2008-08-07 12:44:20 -0500 (Thu, 07 Aug 2008) | 6 lines

Merged revisions 136541 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

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2008-08-07 17:44:40 +00:00
Kevin P. Fleming
61d378f768 Blocked revisions 136504 via svnmerge
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r136504 | kpfleming | 2008-08-07 12:26:23 -0500 (Thu, 07 Aug 2008) | 2 lines

stop using deprecated API call

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2008-08-07 17:26:46 +00:00
Tilghman Lesher
f67e68de8e Merged revisions 136489 via svnmerge from
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r136489 | tilghman | 2008-08-07 11:55:57 -0500 (Thu, 07 Aug 2008) | 15 lines

Merged revisions 136488 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r136488 | tilghman | 2008-08-07 11:50:47 -0500 (Thu, 07 Aug 2008) | 7 lines

Update persistent state on all exit conditions.
(closes issue #12916)
 Reported by: sgenyuk
 Patches: 
       app_queue.patch.txt uploaded by neutrino88 (license 297)
 Tested by: sgenyuk, aragon

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2008-08-07 16:57:53 +00:00
Kevin P. Fleming
d517702bfb Blocked revisions 136477 via svnmerge
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r136477 | kpfleming | 2008-08-07 11:29:08 -0500 (Thu, 07 Aug 2008) | 2 lines

fix some format strings to actually compile without errors

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2008-08-07 16:30:10 +00:00
Sean Bright
287d7a3c19 Blocked revisions 136408 via svnmerge
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r136408 | seanbright | 2008-08-07 11:16:48 -0400 (Thu, 07 Aug 2008) | 6 lines

More merges from resolve-shadow warnings:

  utils/
  codecs/
  and a change I missed from formats/

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2008-08-07 15:17:49 +00:00
Tilghman Lesher
b2ec2f1e49 Blocked revisions 136406 via svnmerge
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r136406 | tilghman | 2008-08-07 10:10:53 -0500 (Thu, 07 Aug 2008) | 2 lines

Fix runtime symbol error

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2008-08-07 15:12:06 +00:00
Sean Bright
3d6280c057 Blocked revisions 136402 via svnmerge
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r136402 | seanbright | 2008-08-07 10:36:59 -0400 (Thu, 07 Aug 2008) | 1 line

Merge in a few more changes.  This time the include/ directory.
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2008-08-07 14:37:24 +00:00
Tilghman Lesher
244174b9bc Blocked revisions 136349 via svnmerge
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r136349 | tilghman | 2008-08-06 20:32:10 -0500 (Wed, 06 Aug 2008) | 10 lines

Blocked revisions 136348 via svnmerge

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r136348 | tilghman | 2008-08-06 20:31:27 -0500 (Wed, 06 Aug 2008) | 3 lines

Also, parse useincomingcalleridonzaptransfer (and add appropriate deprecation
warnings).

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2008-08-07 01:32:51 +00:00
Tilghman Lesher
ac10ad6d81 Blocked revisions 136305 via svnmerge
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r136305 | tilghman | 2008-08-06 20:18:03 -0500 (Wed, 06 Aug 2008) | 10 lines

Blocked revisions 136304 via svnmerge

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r136304 | tilghman | 2008-08-06 20:17:14 -0500 (Wed, 06 Aug 2008) | 3 lines

For backwards compatibility with previous 1.4 versions which used "zapchan"
in users.conf, ensure that we still support it.

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2008-08-07 01:18:39 +00:00
Sean Bright
bd551b8145 Blocked revisions 136302 via svnmerge
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r136302 | seanbright | 2008-08-06 21:07:40 -0400 (Wed, 06 Aug 2008) | 2 lines

Continue merging in changes from resolve-shadow-warnings.  funcs/ this time.

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2008-08-07 01:08:04 +00:00
Sean Bright
845f98553c Blocked revisions 136300 via svnmerge
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r136300 | seanbright | 2008-08-06 20:52:23 -0400 (Wed, 06 Aug 2008) | 2 lines

More from the resolve-shadow-warnings branch.  This time the cdr/ directory.

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2008-08-07 00:53:12 +00:00
Sean Bright
ad7bedfc61 Blocked revisions 136298 via svnmerge
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r136298 | seanbright | 2008-08-06 20:44:55 -0400 (Wed, 06 Aug 2008) | 5 lines

Start moving in changes from my resolve-shadow-warnings branch.  Going to do
this in pieces so the diffs are a little bit smaller and more reviewable.

pbx/ and formats/ first.

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2008-08-07 00:45:59 +00:00
Dwayne M. Hubbard
38e9396e1f Blocked revisions 136245 via svnmerge
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r136245 | dhubbard | 2008-08-06 16:22:56 -0500 (Wed, 06 Aug 2008) | 1 line

move taskprocessor CLI commands into the core namespace
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2008-08-06 21:28:37 +00:00
Mark Michelson
dbc5b5b357 Merged revisions 136239 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r136239 | mmichelson | 2008-08-06 15:43:58 -0500 (Wed, 06 Aug 2008) | 11 lines

Blocked revisions 136238 via svnmerge

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r136238 | mmichelson | 2008-08-06 15:42:15 -0500 (Wed, 06 Aug 2008) | 4 lines

We only need to unregister the QueueStatus manager
command once on an unload


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2008-08-06 20:45:45 +00:00
Tilghman Lesher
b0cb09c4aa Merged revisions 136191 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r136191 | tilghman | 2008-08-06 15:15:34 -0500 (Wed, 06 Aug 2008) | 12 lines

Merged revisions 136190 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r136190 | tilghman | 2008-08-06 15:14:54 -0500 (Wed, 06 Aug 2008) | 4 lines

-C option takes a filename, not a directory path.
(closes issue #13007)
 Reported by: klaus3000

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2008-08-06 20:16:13 +00:00
Tilghman Lesher
6f7680cccf Blocked revisions 136141 via svnmerge
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r136141 | tilghman | 2008-08-06 12:33:48 -0500 (Wed, 06 Aug 2008) | 6 lines

Janitor ast_str project
(closes issue #13058)
 Reported by: pputman
 Patches: 
       app_meetme_aststr2.patch uploaded by pputman (license 81)

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2008-08-06 17:34:27 +00:00
Tilghman Lesher
af1d2d4538 Merged revisions 136112 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r136112 | tilghman | 2008-08-06 11:58:42 -0500 (Wed, 06 Aug 2008) | 7 lines

Persist DIALGROUP() values in astdb
(closes issue #13138)
 Reported by: Corydon76
 Patches: 
       20080725__bug13138.diff.txt uploaded by Corydon76 (license 14)
 Tested by: pj

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2008-08-06 17:02:06 +00:00
Mark Michelson
b9dfa2fe6c Merged revisions 136063 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r136063 | mmichelson | 2008-08-06 10:59:29 -0500 (Wed, 06 Aug 2008) | 24 lines

Merged revisions 136062 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r136062 | mmichelson | 2008-08-06 10:58:40 -0500 (Wed, 06 Aug 2008) | 16 lines

Since adding the AST_CONTROL_SRCUPDATE frame type,
there are places where ast_rtp_new_source may be called
where the tech_pvt of a channel may not yet have an
rtp structure allocated. This caused a crash in chan_skinny,
which was fixed earlier, but now the same crash has been 
reported against chan_h323 as well. It seems that the best 
solution is to modify ast_rtp_new_source to not attempt to 
set the marker bit if the rtp structure passed in is NULL.

This change to ast_rtp_new_source also allows the removal
of what is now a redundant pointer check from chan_skinny.

(closes issue #13247)
Reported by: pj


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2008-08-06 16:00:07 +00:00
Tilghman Lesher
b4cb58175f Blocked revisions 136034 via svnmerge
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r136034 | tilghman | 2008-08-06 09:51:51 -0500 (Wed, 06 Aug 2008) | 3 lines

Use a dynamic buffer for rendered SQL, instead of hardcoding 2048 bytes.  Also,
switch to using RWLISTs for the linked list of queries.

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2008-08-06 14:53:20 +00:00
Olle Johansson
88c17a771b Merged revisions 136005 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r136005 | oej | 2008-08-06 15:34:08 +0200 (Ons, 06 Aug 2008) | 6 lines

- Formatting
- Changing debug messages from VERBOSE to DEBUG channel
- Adding a few todo's
- Adding a few more "XMPP"'s to compliment Jabber...


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2008-08-06 13:59:32 +00:00
Tilghman Lesher
d645fe1754 Merged revisions 135950 via svnmerge from
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r135950 | tilghman | 2008-08-05 22:55:49 -0500 (Tue, 05 Aug 2008) | 12 lines

Merged revisions 135949 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r135949 | tilghman | 2008-08-05 22:53:36 -0500 (Tue, 05 Aug 2008) | 4 lines

Fix a longstanding bug in channel walking logic, and fix the explanation to
make sense.
(Closes issue #13124)

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2008-08-06 03:56:30 +00:00
Tilghman Lesher
58c8e58fe5 Merged revisions 135938 via svnmerge from
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r135938 | tilghman | 2008-08-05 22:29:42 -0500 (Tue, 05 Aug 2008) | 12 lines

Merged revisions 135915 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r135915 | tilghman | 2008-08-05 22:24:56 -0500 (Tue, 05 Aug 2008) | 4 lines

Since powerof() can return an error condition, it's foolhardy not to detect and
deal with that condition.
(Related to issue #13240)

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2008-08-06 03:30:23 +00:00
Tilghman Lesher
4a854a5e5b Merged revisions 135900 via svnmerge from
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r135900 | tilghman | 2008-08-05 22:04:01 -0500 (Tue, 05 Aug 2008) | 12 lines

Merged revisions 135899 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r135899 | tilghman | 2008-08-05 22:02:59 -0500 (Tue, 05 Aug 2008) | 4 lines

1) Bugfix for debugging code
2) Reduce compiler warnings for another section of debugging code
(Closes issue #13237)

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2008-08-06 03:04:49 +00:00
Mark Michelson
8417f7bfb1 Merged revisions 135851 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r135851 | mmichelson | 2008-08-05 19:30:53 -0500 (Tue, 05 Aug 2008) | 48 lines

Merged revisions 135841,135847,135850 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r135841 | mmichelson | 2008-08-05 19:25:10 -0500 (Tue, 05 Aug 2008) | 27 lines

Merging the issue11259 branch.

The purpose of this branch was to take into account
"burps" which could cause jitterbuffers to misbehave.
One such example is if the L option to Dial() were used
to inject audio into a bridged conversation at regular
intervals. Since the audio here was not passed through
the jitterbuffer, it would cause a gap in the jitterbuffer's
timestamps which would cause a frames to be dropped for a 
brief period.

Now ast_generic_bridge will empty and reset the jitterbuffer
each time it is called. This causes injected audio to be handled
properly.

ast_generic_bridge also will empty and reset the jitterbuffer
if it receives an AST_CONTROL_SRCUPDATE frame since the change
in audio source could negatively affect the jitterbuffer.

All of this was made possible by adding a new public API call
to the abstract_jb called ast_jb_empty_and_reset.

(closes issue #11259)
Reported by: plack
Tested by: putnopvut


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r135847 | mmichelson | 2008-08-05 19:27:54 -0500 (Tue, 05 Aug 2008) | 4 lines

Revert inadvertent changes to app_skel that occurred when
I was testing for a memory leak


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r135850 | mmichelson | 2008-08-05 19:29:54 -0500 (Tue, 05 Aug 2008) | 3 lines

Remove properties that should not be here


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2008-08-06 00:31:28 +00:00
Steve Murphy
638711f573 Merged revisions 135821 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r135821 | murf | 2008-08-05 17:45:32 -0600 (Tue, 05 Aug 2008) | 42 lines

Merged revisions 135799 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r135799 | murf | 2008-08-05 17:13:20 -0600 (Tue, 05 Aug 2008) | 34 lines

(closes issue #12982)
Reported by: bcnit
Tested by: murf

I discovered that also, in the previous bug fixes and changes,
the cdr.conf 'unanswered' option is not being obeyed, so
I fixed this.

And, yes, there are two 'answer' times involved in this
scenario, and I would agree with you, that the first 
answer time is the time that should appear in the CDR.
(the second 'answer' time is the time that the bridge
was begun).

I made the necessary adjustments, recording the first
answer time into the peer cdr, and then using that to
override the bridge cdr's value.

To get the 'unanswered' CDRs to appear, I purposely
output them, using the dial cmd to mark them as
DIALED (with a new flag), and outputting them if
they bear that flag, and you are in the right mode.

I also corrected one small mention of the Zap device
to equally consider the dahdi device.

I heavily tested 10-sec-wait macros in dial, and
without the macro call; I tested hangups while the
macro was running vs. letting the macro complete
and the bridge form. Looks OK. Removed all the
instrumentation and debug.



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2008-08-05 23:52:14 +00:00
Tilghman Lesher
2085534aef Merged revisions 135748 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r135748 | tilghman | 2008-08-05 16:37:35 -0500 (Tue, 05 Aug 2008) | 17 lines

Merged revisions 135747 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r135747 | tilghman | 2008-08-05 16:34:46 -0500 (Tue, 05 Aug 2008) | 9 lines

In a conversion to use ast_strlen_zero, the meaning of the flag IAX_HASCALLERID
was perverted.  This change reverts IAX2 to the original meaning, which was,
that the callerid set on the client should be overridden on the server, even if
that means the resulting callerid is blank.  In other words, if you set
"callerid=" in the IAX config, then the callerid should be overridden to blank,
even if set on the client.  Note that there's a distinction, even on realtime,
between the field not existing (NULL in databases) and the field existing, but
set to blank (override callerid to blank).

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2008-08-05 21:38:29 +00:00
Kevin P. Fleming
e658b7c083 Blocked revisions 135680-135681 via svnmerge
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r135680 | kpfleming | 2008-08-05 11:56:11 -0500 (Tue, 05 Aug 2008) | 2 lines

make datastore creation and destruction a generic API since it is not really channel related, and add the ability to add/find/remove datastores to manager sessions

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r135681 | kpfleming | 2008-08-05 12:05:34 -0500 (Tue, 05 Aug 2008) | 3 lines

datastore inheritance is a channel feature, so move this definition back


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2008-08-05 17:43:07 +00:00
Tilghman Lesher
7d93b5632f Blocked revisions 135648 via svnmerge
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r135648 | tilghman | 2008-08-05 10:30:23 -0500 (Tue, 05 Aug 2008) | 3 lines

Always output a version string, even when we can't figure out what we are.
(Closes issue #13223)

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2008-08-05 15:31:18 +00:00
Sean Bright
7d44270d33 Merged revisions 135598 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r135598 | seanbright | 2008-08-05 09:26:34 -0400 (Tue, 05 Aug 2008) | 9 lines

Merged revisions 135597 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r135597 | seanbright | 2008-08-05 09:25:00 -0400 (Tue, 05 Aug 2008) | 1 line

Use PATH_MAX for filenames
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2008-08-05 13:27:59 +00:00
Russell Bryant
83521808f4 Merged revisions 135537 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r135537 | russell | 2008-08-04 15:15:27 -0500 (Mon, 04 Aug 2008) | 10 lines

Merged revisions 135536 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r135536 | russell | 2008-08-04 15:15:03 -0500 (Mon, 04 Aug 2008) | 2 lines

fix a config sample typo

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2008-08-04 20:15:46 +00:00
Tilghman Lesher
83f0de6aef Merged revisions 135485 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r135485 | tilghman | 2008-08-04 12:12:15 -0500 (Mon, 04 Aug 2008) | 3 lines

Rename Mandrake scripts to Mandriva
(Closes issue #13221)

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2008-08-04 17:12:44 +00:00
Tilghman Lesher
ada9e611dc Merged revisions 135483 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r135483 | tilghman | 2008-08-04 12:08:42 -0500 (Mon, 04 Aug 2008) | 11 lines

Merged revisions 135482 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r135482 | tilghman | 2008-08-04 12:07:52 -0500 (Mon, 04 Aug 2008) | 2 lines

Define ASTSBINDIR for script
(Closes issue #13221)

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2008-08-04 17:09:17 +00:00
Tilghman Lesher
5cf295c4de Merged revisions 135480 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r135480 | tilghman | 2008-08-04 11:58:29 -0500 (Mon, 04 Aug 2008) | 14 lines

Merged revisions 135479 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r135479 | tilghman | 2008-08-04 11:56:19 -0500 (Mon, 04 Aug 2008) | 6 lines

Memory leak on unload
(closes issue #13231)
 Reported by: eliel
 Patches: 
       app_voicemail.leak.patch uploaded by eliel (license 64)

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2008-08-04 17:03:46 +00:00
Tilghman Lesher
7a8515e211 Blocked revisions 135476 via svnmerge
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r135476 | tilghman | 2008-08-04 11:34:04 -0500 (Mon, 04 Aug 2008) | 6 lines

HTTP module memory leaks
(closes issue #13230)
 Reported by: eliel
 Patches: 
       res_http_post_leak.patch uploaded by eliel (license 64)

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2008-08-04 16:35:44 +00:00
Russell Bryant
4f2e6ffa98 Merged revisions 135474 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r135474 | russell | 2008-08-04 11:28:07 -0500 (Mon, 04 Aug 2008) | 10 lines

Merged revisions 135473 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r135473 | russell | 2008-08-04 11:26:17 -0500 (Mon, 04 Aug 2008) | 2 lines

Add a minor clarification to the documentation of mohinterpret and mohsuggest

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2008-08-04 16:28:25 +00:00
Russell Bryant
bf7ca7ec97 Merged revisions 135439 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r135439 | russell | 2008-08-04 10:02:12 -0500 (Mon, 04 Aug 2008) | 4 lines

Be explicit that we don't want a result from this callback.  The callback would
never indicate a match, so nothing would have been returned anyway, but it was
still a poor example of proper usage.

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2008-08-04 15:02:44 +00:00
Sean Bright
6c09b18717 Blocked revisions 135405 via svnmerge
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r135405 | seanbright | 2008-08-03 12:14:14 -0400 (Sun, 03 Aug 2008) | 3 lines

Merge in changes that allow Asterisk to be built against the Hoard
memory allocator.  See doc/hoard.txt for more details.

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2008-08-03 16:15:10 +00:00
Michiel van Baak
ad8494bc11 Blocked revisions 135373 via svnmerge
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r135373 | mvanbaak | 2008-08-03 02:03:53 +0200 (Sun, 03 Aug 2008) | 2 lines

whitespace fixes only.

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2008-08-03 00:06:24 +00:00
Michiel van Baak
c2fb65d6c4 Blocked revisions 135371 via svnmerge
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r135371 | mvanbaak | 2008-08-03 02:00:06 +0200 (Sun, 03 Aug 2008) | 8 lines

Dont coredump on register of non-configured devices

(closes issue #13224)
Reported by: mvanbaak
Patches:
      noncon.diff uploaded by wedhorn (license 30) with whitespace fixes by me
Tested by: wedhorn, mvanbaak

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2008-08-03 00:01:11 +00:00
Michiel van Baak
283d95d035 Blocked revisions 135300 via svnmerge
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r135300 | mvanbaak | 2008-08-02 14:29:23 +0200 (Sat, 02 Aug 2008) | 8 lines

pass device instead of session to transmit_ functions.

(closes issue #10396)
Reported by: wedhorn
Patches:
      transmit3a.diff uploaded by wedhorn (license 30)
Tested by: wedhorn, mvanbaak

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2008-08-02 12:31:33 +00:00
Steve Murphy
88a4540726 Merged revisions 135265 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r135265 | murf | 2008-08-01 22:51:29 -0600 (Fri, 01 Aug 2008) | 31 lines

(closes issue #13202)
Reported by: falves11
Tested by: murf

falves11 ==

The changes I introduce here seem to clear up the problem
for me. However, if they do not for you, please reopen this
bug, and we'll keep digging.

The root of this problem seems to be a subtle memory corruption
introduced when creating an extension with an empty extension
name. While valgrind cannot detect it outside of DEBUG_MALLOC
mode, when compiled with DEBUG_MALLOC, this is certain death.

The code in main/features.c is a puzzle to me. On the initial
module load, the code is attempting to add the parking extension
before the features.conf file has even been opened!

I just wrapped the offending call with an if() that will not
try to add the extension if the extension name is empty. THis
seems to solve the corruption, and let the "memory show allocations"
work as one would expect.

But, really, adding an extension with an empty name is a seriously
bad thing to allow, as it will mess up all the pattern matching 
algorithms, etc. So, I added a statement to the add_extension2 code to return
a -1 if this is attempted.


in 1.6.0, the changes to only main/pbx.c were applicable,
as apparently the code added to main/features by jpeeler
were not included in 1.6.0.

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2008-08-02 05:15:59 +00:00
Terry Wilson
17a509e4d6 Blocked revisions 135235 via svnmerge
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r135235 | twilson | 2008-08-01 16:56:07 -0500 (Fri, 01 Aug 2008) | 2 lines

Fix mime parsing by re-adding support for passing headers to callback functions

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2008-08-01 21:58:06 +00:00
Sean Bright
3f6647d7ad Merged revisions 135197 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r135197 | seanbright | 2008-08-01 15:29:26 -0400 (Fri, 01 Aug 2008) | 6 lines

Remove some code that used to do something but does not anymore, mainly
to get rid of a shadow warning (but this seemed legitimate enough to fix
here instead of in my branch).

Thanks to putnopvut for taking a look as well.

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2008-08-01 19:30:31 +00:00
Russell Bryant
60cc0566b0 Blocked revisions 135158 via svnmerge
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r135158 | russell | 2008-08-01 13:16:24 -0500 (Fri, 01 Aug 2008) | 14 lines

Merge changes from team/bbryant/keyrotation

This set of changes enhances IAX2 encryption support by adding key rotation
to provide enhanced security.  The key used for encryption is rotated right 
after the call gets set up, and then again every few minutes.  This was
discussed at the last AstriDevCon.  For interoperability with older versions
of Asterisk, there is an option that disables key rotation.

(closes issue #13018)
Reported by: bbryant
Patches:
      07072008__iax2_key_rotation.diff uploaded by bbryant (license 36)
Tested by: russell, bbryant

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2008-08-01 18:17:22 +00:00
Tilghman Lesher
45253fa663 Merged revisions 135128 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r135128 | tilghman | 2008-08-01 12:09:50 -0500 (Fri, 01 Aug 2008) | 2 lines

Picky, picky, buildbot

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2008-08-01 17:10:29 +00:00
Tilghman Lesher
4802c70d3a Merged revisions 135126 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r135126 | tilghman | 2008-08-01 11:39:51 -0500 (Fri, 01 Aug 2008) | 9 lines

SIP should use the transport type set in the Moved Temporarily for the next
invite.
(closes issue #11843)
 Reported by: pestermann
 Patches: 
       20080723__issue11843_302_ignores_transport_16branch.diff uploaded by bbryant (license 36)
       20080723__issue11843_302_ignores_transport_trunk.diff uploaded by bbryant (license 36)
 Tested by: pabelanger

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2008-08-01 16:44:32 +00:00
Mark Michelson
52f334eacf Merged revisions 135067-135068 via svnmerge from
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r135067 | mmichelson | 2008-08-01 09:29:48 -0500 (Fri, 01 Aug 2008) | 13 lines

IMAP storage functioned under the assumption that folders
such as "Work" and "Family" would be subfolders of the
INBOX. This is an invalid assumption to make, but it could
be desirable to set up folders in this manner, so a new
option for voicemail.conf, "imapparentfolder" has been
added to allow for this.

(closes issue #13142)
Reported by: jaroth
Patches:
      parentfolder.patch uploaded by jaroth (license 50)


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r135068 | mmichelson | 2008-08-01 09:42:24 -0500 (Fri, 01 Aug 2008) | 3 lines

IMAP-specific items must go in IMAP_STORAGE defines...


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2008-08-01 14:43:19 +00:00
Michiel van Baak
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r135061 | mvanbaak | 2008-08-01 14:17:33 +0200 (Fri, 01 Aug 2008) | 8 lines

Make safe_asterisk work on dash/sh/bash etc.

(closes issue #13111)
Reported by: pabelanger
Patches:
      2008071901_issue13111_safe_asterisk.diff uploaded by mvanbaak (license 7)
Tested by: mvanbaak, pabelanger

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2008-08-01 12:18:18 +00:00
Michiel van Baak
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r135059 | mvanbaak | 2008-08-01 13:47:34 +0200 (Fri, 01 Aug 2008) | 10 lines

Merged revisions 135058 via svnmerge from 
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r135058 | mvanbaak | 2008-08-01 13:43:46 +0200 (Fri, 01 Aug 2008) | 2 lines

make app_ices compile on OpenBSD.

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2008-08-01 11:57:38 +00:00
Michiel van Baak
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r135056 | mvanbaak | 2008-08-01 13:00:13 +0200 (Fri, 01 Aug 2008) | 16 lines

Merged revisions 135055 via svnmerge from 
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r135055 | mvanbaak | 2008-08-01 12:55:27 +0200 (Fri, 01 Aug 2008) | 8 lines

fix some potential deadlocks in chan_skinny

(closes issue #13215)
Reported by: qwell
Patches:
      2008080100_bug13215.diff.txt uploaded by mvanbaak (license 7)
Tested by: mvanbaak

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2008-08-01 11:03:07 +00:00
Kevin P. Fleming
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r135016 | kpfleming | 2008-07-31 17:28:42 -0500 (Thu, 31 Jul 2008) | 11 lines

Merged revisions 134983 via svnmerge from 
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r134983 | kpfleming | 2008-07-31 17:18:11 -0500 (Thu, 31 Jul 2008) | 3 lines

accomodate users who seem to lack a sense of humor :-)


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2008-07-31 22:34:44 +00:00
Tilghman Lesher
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r134980 | tilghman | 2008-07-31 16:55:42 -0500 (Thu, 31 Jul 2008) | 16 lines

Blocked revisions 134976 via svnmerge

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r134976 | tilghman | 2008-07-31 16:53:19 -0500 (Thu, 31 Jul 2008) | 9 lines

Specify codecs in callfiles and manager, to allow video calls to be set up
from callfiles and AMI.
(closes issue #9531)
 Reported by: Geisj
 Patches: 
       20080715__bug9531__1.4.diff.txt uploaded by Corydon76 (license 14)
       20080715__bug9531__1.6.0.diff.txt uploaded by Corydon76 (license 14)
 Tested by: Corydon76

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r134977 | tilghman | 2008-07-31 16:53:59 -0500 (Thu, 31 Jul 2008) | 2 lines

Switch command order, to meet with current specs

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r134925 | tilghman | 2008-07-31 15:10:39 -0500 (Thu, 31 Jul 2008) | 3 lines

Increase column size beyond the minimum required, since PostgreSQL won't let
us modify existing columns.

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2008-07-31 20:11:17 +00:00
Steve Murphy
7a26f22be2 Merged revisions 134922 via svnmerge from
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r134922 | murf | 2008-07-31 13:48:08 -0600 (Thu, 31 Jul 2008) | 63 lines

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r134883 | murf | 2008-07-31 13:23:42 -0600 (Thu, 31 Jul 2008) | 51 lines

(closes issue #11849)
Reported by: greyvoip
Tested by: murf

OK, a few days of debugging, a bunch of instrumentation
in chan_sip, main/channel.c, main/pbx.c, etc. and 5 solid 
notebook pages of notes later, I  have made the small
tweek necc. to get the start time right on the second 
CDR when:

  A Calls B
  B answ.
  A hits Xfer button on sip phone,
  A dials C and hits the OK button,
  A hangs up
  C answers ringing phone
  B and C converse
  B and/or C hangs up

But does not harm the scenario where:

  A Calls B
  B answ.
  B hits xfer button on sip phone,
  B dials C and hits the OK button,
  B hangs up
  C answers ringing phone
  A and C converse
  A and/or C hangs up

The difference in start times on the second CDR is because
of a Masquerade on the B channel when the xfer number is 
sent. It ends up replacing the CDR on the B channel with
a duplicate, which ends up getting tossed out. We keep 
a pointer to the first CDR, and update *that* after the
bridge closes. But, only if the CDR has changed.

I hope this change is specific enough not to muck
up any current CDR-based apps. In my defence, I 
assert that the previous information was wrong,
and this change fixes it, and possibly other
similar scenarios.

I wonder if I should be doing the same thing
for the channel, as I did for the peer, but
I can't think of a scenario this might affect.
I leave it, then, as an exersize for the users,
to find the scenario where the chan's CDR 
changes and loses the proper start time.


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Tilghman Lesher
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r134919 | tilghman | 2008-07-31 14:43:02 -0500 (Thu, 31 Jul 2008) | 6 lines

Two errors:
	1) If a function returns SQLITE_LOCKED, no recovery is possible.
	2) An error message can be allocated, even when no error is signalled.
(closes issue #13109)
 Reported by: gknispel_proformatique

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Russell Bryant
a6439b01e2 Merged revisions 134917 via svnmerge from
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r134917 | russell | 2008-07-31 14:39:50 -0500 (Thu, 31 Jul 2008) | 17 lines

Merged revisions 134915 via svnmerge from 
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r134915 | russell | 2008-07-31 14:37:26 -0500 (Thu, 31 Jul 2008) | 9 lines

Get app_ices working again

(closes issue #12981)
Reported by: dlogan
Patches:
      20080709__app_ices_v2_update_trunk.diff uploaded by bbryant (license 36)
      20080709__app_ices_v2_update_14.diff uploaded by bbryant (license 36)
Tested by: bbryant

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2008-07-31 19:41:11 +00:00
Tilghman Lesher
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r134867 | tilghman | 2008-07-31 14:03:41 -0500 (Thu, 31 Jul 2008) | 4 lines

Optimize frame cache by realloc'ing the smallest frame when the cache is full.
This ensures that we don't just keep a cache of tiny frames, continually doing
an alloc/free for each data frame, thus negating the point of having a cache.

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Russell Bryant
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r134815 | russell | 2008-07-31 11:50:10 -0500 (Thu, 31 Jul 2008) | 15 lines

Merged revisions 134814 via svnmerge from 
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r134814 | russell | 2008-07-31 11:45:31 -0500 (Thu, 31 Jul 2008) | 7 lines

In case we have some processing threads that free more frames than they allocate,
do not let the frame cache grow forever.

(closes issue #13160)
Reported by: tavius
Tested by: tavius, russell

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2008-07-31 16:53:18 +00:00
Russell Bryant
61077f93d3 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@134806 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2008-07-31 16:38:28 +00:00
Mark Michelson
a97827b540 Merged revisions 134759 via svnmerge from
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r134759 | mmichelson | 2008-07-31 11:05:12 -0500 (Thu, 31 Jul 2008) | 24 lines

Merged revisions 134758 via svnmerge from 
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r134758 | mmichelson | 2008-07-31 10:56:18 -0500 (Thu, 31 Jul 2008) | 16 lines

Add more timeout checks into app_queue, specifically
targeting areas where an unknown and potentially
long time has just elapsed. Also added a check
to try_calling() to return early if the timeout
has elapsed instead of potentially setting a negative
timeout for the call (thus making it have *no* timeout
at all).


(closes issue #13186)
Reported by: miquel_cabrespina
Patches:
      13186.diff uploaded by putnopvut (license 60)
Tested by: miquel_cabrespina


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r134705 | tilghman | 2008-07-30 17:40:48 -0500 (Wed, 30 Jul 2008) | 9 lines

Blocked revisions 134704 via svnmerge

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r134704 | tilghman | 2008-07-30 17:39:57 -0500 (Wed, 30 Jul 2008) | 2 lines

Oops, wrong define

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r134703 | tilghman | 2008-07-30 17:38:58 -0500 (Wed, 30 Jul 2008) | 2 lines

Oops, wrong define

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r134650 | tilghman | 2008-07-30 16:40:08 -0500 (Wed, 30 Jul 2008) | 12 lines

Merged revisions 134649 via svnmerge from 
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r134649 | tilghman | 2008-07-30 16:38:50 -0500 (Wed, 30 Jul 2008) | 4 lines

Qwell pointed out, via IRC, that the previous fix only worked when explicitly
set.  When nothing is set, and the option is implied, it breaks, because
configure sets the prefix to 'NONE'.  Fixing.

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Mark Michelson
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r134598 | mmichelson | 2008-07-30 16:05:37 -0500 (Wed, 30 Jul 2008) | 15 lines

Merged revisions 134556 via svnmerge from 
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r134556 | mmichelson | 2008-07-30 15:24:40 -0500 (Wed, 30 Jul 2008) | 7 lines

Fix the parsing of the "reason" parameter in the
Diversion: header.

(closes issue #13195)
Reported by: woodsfsg


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Russell Bryant
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r134596 | russell | 2008-07-30 15:38:35 -0500 (Wed, 30 Jul 2008) | 14 lines

Merged revisions 134595 via svnmerge from 
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r134595 | russell | 2008-07-30 15:37:17 -0500 (Wed, 30 Jul 2008) | 6 lines

Reduce stack consumption by 12.5% of the max stack size to fix a crash when
compiled with LOW_MEMORY.

(closes issue #13154)
Reported by: edantie

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Mark Michelson
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r134556 | mmichelson | 2008-07-30 15:24:40 -0500 (Wed, 30 Jul 2008) | 7 lines

Fix the parsing of the "reason" parameter in the
Diversion: header.

(closes issue #13195)
Reported by: woodsfsg


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r134541 | russell | 2008-07-30 14:55:31 -0500 (Wed, 30 Jul 2008) | 12 lines

Merged revisions 134540 via svnmerge from 
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r134540 | russell | 2008-07-30 14:52:53 -0500 (Wed, 30 Jul 2008) | 4 lines

Fix a memory leak in func_curl.  Every thread that used this function leaked
an allocation the size of a pointer.
(reported by jmls in #asterisk-dev)

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Tilghman Lesher
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r134538 | tilghman | 2008-07-30 14:48:37 -0500 (Wed, 30 Jul 2008) | 12 lines

Merged revisions 134536 via svnmerge from 
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r134536 | tilghman | 2008-07-30 14:47:16 -0500 (Wed, 30 Jul 2008) | 4 lines

Only override sysconfdir and mandir when prefix=/usr
(closes issue #13093)
 Reported by: pabelanger

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r134483 | tilghman | 2008-07-30 14:17:38 -0500 (Wed, 30 Jul 2008) | 4 lines

Let "roundrobin" also reference rrmemory, for the 1.6 release (as described in
UPGRADE-1.4.txt)
(Closes issue #13181)

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r134481 | tilghman | 2008-07-30 14:05:35 -0500 (Wed, 30 Jul 2008) | 13 lines

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r134480 | tilghman | 2008-07-30 14:03:44 -0500 (Wed, 30 Jul 2008) | 5 lines

launch_netscript sometimes returns -1, which fails to set AGISTATUS.  Map
failure to -1, so that AGISTATUS is always set.
(closes issue #13199)
 Reported by: smw1218

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r134476 | mmichelson | 2008-07-30 13:33:12 -0500 (Wed, 30 Jul 2008) | 12 lines

Merged revisions 134475 via svnmerge from 
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r134475 | mmichelson | 2008-07-30 13:31:47 -0500 (Wed, 30 Jul 2008) | 4 lines

Fix a spot where a function could return without bringing
a channel out of autoservice.


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r134442 | tilghman | 2008-07-30 12:32:28 -0500 (Wed, 30 Jul 2008) | 7 lines

Add adaptive capabilities to the sqlite realtime driver
(closes issue #13097)
 Reported by: gknispel_proformatique
 Patches: 
       20080730__bug13097.diff.txt uploaded by Corydon76 (license 14)
 Tested by: Corydon76

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r134443 | tilghman | 2008-07-30 12:36:31 -0500 (Wed, 30 Jul 2008) | 2 lines

Document adaptive capabilities

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r134401 | tilghman | 2008-07-30 11:40:43 -0500 (Wed, 30 Jul 2008) | 7 lines

Move implementation of an attended-transfer-complete sound from one channel
driver into a common place for multiple channel drivers.
(closes issue #13152)
 Reported by: caio1982
 Patches: 
       atxfer_complete_sound3.diff uploaded by caio1982 (license 22)

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r134355 | kpfleming | 2008-07-30 10:32:14 -0500 (Wed, 30 Jul 2008) | 10 lines

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r134352 | kpfleming | 2008-07-30 10:29:17 -0500 (Wed, 30 Jul 2008) | 2 lines

use the proper method for building version.h

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r134353 | tilghman | 2008-07-30 10:30:18 -0500 (Wed, 30 Jul 2008) | 8 lines

Add %u and %g to the ASTERISK_PROMPT settings, for username and group,
respectively.  Also, take the opportunity to clean up the CLI prompt
generation code.
(closes issue #13175)
 Reported by: eliel
 Patches: 
       cliprompt.patch uploaded by eliel (license 64)

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r134312 | tilghman | 2008-07-29 18:00:45 -0500 (Tue, 29 Jul 2008) | 6 lines

Minor changes to reduce packaging changes made by the Fedora maintainer.
(closes issue #12974)
 Reported by: jcollie
 Patches: 
       0001-Don-t-override-duplicate-optimization-flags.patch uploaded by jcollie (license 412)

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2008-07-29 23:01:26 +00:00
Kevin P. Fleming
25fe3c40f1 Merged revisions 134260 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r134260 | kpfleming | 2008-07-29 17:22:13 -0500 (Tue, 29 Jul 2008) | 2 lines

build against the now-typedef-free dahdi/user.h, and remove some #ifdefs for features that will always be present in DAHDI

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2008-07-29 22:29:11 +00:00
Tilghman Lesher
2e8d6fc9b8 Merged revisions 134163 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r134163 | tilghman | 2008-07-28 17:07:12 -0500 (Mon, 28 Jul 2008) | 15 lines

Merged revisions 134161 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r134161 | tilghman | 2008-07-28 16:50:50 -0500 (Mon, 28 Jul 2008) | 7 lines

Detect when sox fails to raise the volume, because sox can't read the file.
(closes issue #12939)
 Reported by: rickbradley
 Patches: 
       20080728__bug12939.diff.txt uploaded by Corydon76 (license 14)
 Tested by: rickbradley

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2008-07-28 22:16:39 +00:00
Mark Michelson
b2eaaae0b5 Merged revisions 134125 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r134125 | mmichelson | 2008-07-28 14:53:56 -0500 (Mon, 28 Jul 2008) | 27 lines

This commit compensates for buggy poll(2)
implementations. Asterisk has, for a long time,
had its own implementation of poll(2) which
just used the input arguments to call select(2).
In 1.4, this internal implementation was used
for Darwin systems. This was removed in Asterisk
trunk at some point, but it seems as though this
was not the right move to make.

On Mac OS X, it appears as though the poll used
to gather CLI input does not respond properly
when connecting via a remote Asterisk console.
Reverting to the use of Asterisk's poll fixed
the issue.

Also, there is now an option for the configure
script, --enable-internal-poll, which will allow
for anyone to use Asterisk's internal poll
implementation in case they suspect that their
system's poll implementation is buggy.

closes issue #11928)
Reported by: adriavidal
Patches:
      1.6.0-configurev2.patch uploaded by putnopvut (license 60)


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2008-07-28 19:55:44 +00:00
Tilghman Lesher
bdbeb72fc4 Blocked revisions 134088 via svnmerge
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r134088 | tilghman | 2008-07-28 11:49:29 -0500 (Mon, 28 Jul 2008) | 7 lines

Change SendImage() to output a more consistent status variable.
(closes issue #13134)
 Reported by: eliel
 Patches: 
       app_image.c.patch uploaded by eliel (license 64)
       UPGRADE.patch uploaded by eliel (license 64)

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2008-07-28 16:50:42 +00:00
Kevin P. Fleming
261a4972f6 Merged revisions 134086 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r134086 | kpfleming | 2008-07-28 11:42:00 -0500 (Mon, 28 Jul 2008) | 3 lines

remove remaining Zaptel references in various places


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2008-07-28 16:49:18 +00:00
Mark Michelson
7eb4b5bf5b Merged revisions 134050 via svnmerge from
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r134050 | mmichelson | 2008-07-28 11:00:19 -0500 (Mon, 28 Jul 2008) | 3 lines

merging the zap_and_dahdi_trunk branch up to trunk


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2008-07-28 16:13:34 +00:00
Russell Bryant
322153ad93 Blocked revisions 134005 via svnmerge
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r134005 | russell | 2008-07-27 16:12:14 -0500 (Sun, 27 Jul 2008) | 4 lines

Add a missing unlock within error handling
(closes issue #13176)
Reported by: pj

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2008-07-27 21:12:43 +00:00
Russell Bryant
86cef15735 Include the licensing page in 1.6.0 as well. Now, this page exists in 1.4, trunk,
and 1.6.0.


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2008-07-26 15:34:42 +00:00
Russell Bryant
a12da2ad99 unblock 133575
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2008-07-26 15:34:02 +00:00
Russell Bryant
42cf9fc54e Merged revisions 133945-133946 via svnmerge from
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r133945 | russell | 2008-07-26 10:15:14 -0500 (Sat, 26 Jul 2008) | 6 lines

ast_device_state() gets called in two different ways.  The first way is when
called from elsewhere in Asterisk to find the current state of a device.  In
that case, we want to use the cached value if it exists.  The other way is when
processing a device state change.  In that case, we do not want to check the
cache because returning the last known state is counter productive.

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r133946 | russell | 2008-07-26 10:16:20 -0500 (Sat, 26 Jul 2008) | 1 line

actually use the cache_cache argument
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2008-07-26 15:17:46 +00:00
Russell Bryant
c34ad0f0a3 Blocked revisions 133943 via svnmerge
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r133943 | russell | 2008-07-26 09:57:50 -0500 (Sat, 26 Jul 2008) | 2 lines

Re-work comment about how device state changes are processed to be a bit more clear

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2008-07-26 14:59:05 +00:00
Russell Bryant
f5b0d8449b Blocked revisions 133941 via svnmerge
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r133941 | russell | 2008-07-26 09:46:13 -0500 (Sat, 26 Jul 2008) | 3 lines

Remove the code that decided when device state changes should be cached or not.
It is no longer needed.

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2008-07-26 14:46:39 +00:00
Tilghman Lesher
c4322a9321 Blocked revisions 133904 via svnmerge
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r133904 | tilghman | 2008-07-25 17:08:06 -0500 (Fri, 25 Jul 2008) | 7 lines

Hebrew syntax for voicemail prompts
(closes issue #13155)
 Reported by: greenfieldtech
 Patches: 
       app_voicemail.c.patch uploaded by greenfieldtech (license 369)
       hebrew.ods uploaded by greenfieldtech (license 369)

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2008-07-25 22:09:43 +00:00
Tilghman Lesher
3763dbdf2f Merged revisions 133902 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r133902 | tilghman | 2008-07-25 16:59:39 -0500 (Fri, 25 Jul 2008) | 6 lines

Update version
(closes issue #13163)
 Reported by: suretec
 Patches: 
       asterisk.ldif uploaded by suretec (license 70)

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2008-07-25 22:00:17 +00:00
Tilghman Lesher
a8eb72453a Blocked revisions 133860 via svnmerge
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r133860 | tilghman | 2008-07-25 16:20:03 -0500 (Fri, 25 Jul 2008) | 2 lines

Deprecate *_device_state_* APIs in favor of *_devstate_* APIs

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2008-07-25 21:20:55 +00:00
Brandon Kruse
2ed19e39b4 Blocking revert of code changes in r133770
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2008-07-25 19:37:03 +00:00
Brandon Kruse
1dc7b64454 Include the http_decode function from trunk
to replace the + with a space.


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2008-07-25 19:34:17 +00:00
Tilghman Lesher
954533dab7 Blocked revisions 133710 via svnmerge
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r133710 | tilghman | 2008-07-25 13:01:58 -0500 (Fri, 25 Jul 2008) | 10 lines

Blocked revisions 133709 via svnmerge

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r133709 | tilghman | 2008-07-25 13:00:33 -0500 (Fri, 25 Jul 2008) | 3 lines

Remove unnecessary mmap flag
(Closes issue #13161)

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2008-07-25 18:02:51 +00:00
Brandon Kruse
3f701cd720 Blocking a fix from trunk for the function
http_decode. 1.6.0 does not have this function.


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2008-07-25 17:33:47 +00:00
Tilghman Lesher
e680b3734f Merged revisions 133665 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r133665 | tilghman | 2008-07-25 12:24:43 -0500 (Fri, 25 Jul 2008) | 16 lines

Merged revisions 133649 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r133649 | tilghman | 2008-07-25 12:19:39 -0500 (Fri, 25 Jul 2008) | 8 lines

Fix some errant device states by making the devicestate API more strict in
terms of the device argument (only without the unique identifier appended).
(closes issue #12771)
 Reported by: davidw
 Patches: 
       20080717__bug12771.diff.txt uploaded by Corydon76 (license 14)
 Tested by: davidw, jvandal, murf

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2008-07-25 17:26:38 +00:00
Russell Bryant
8209b6c64b Merged revisions 133579 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r133579 | russell | 2008-07-25 10:00:49 -0500 (Fri, 25 Jul 2008) | 18 lines

Merged revisions 133578 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r133578 | russell | 2008-07-25 10:00:31 -0500 (Fri, 25 Jul 2008) | 10 lines

Merged revisions 133577 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r133577 | russell | 2008-07-25 10:00:13 -0500 (Fri, 25 Jul 2008) | 2 lines

Fix the IAX2 URI for calling Digium

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2008-07-25 15:01:02 +00:00
Russell Bryant
123e9dfbd5 Blocked revisions 133575 via svnmerge
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r133575 | russell | 2008-07-25 09:57:11 -0500 (Fri, 25 Jul 2008) | 8 lines

Modify the main page of the doxygen documentation to link to a new page dedicated
to Asterisk licensing information.  The licensing page includes the Asterisk license,
as well as a (not yet complete) list of 3rd party libraries that may be used, as well
as what license we receive them under.

Help filling out this list in the format that I have started in doxyref.h would be
much appreciated.  :)

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2008-07-25 14:57:32 +00:00
Mark Michelson
19380639cb Merged revisions 133573 via svnmerge from
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r133573 | mmichelson | 2008-07-25 09:40:52 -0500 (Fri, 25 Jul 2008) | 15 lines

Merged revisions 133572 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r133572 | mmichelson | 2008-07-25 09:40:10 -0500 (Fri, 25 Jul 2008) | 7 lines

We need to make sure to null-terminate the "name"
portion of SIP URI parameters so that there are no
bogus comparisons.

Thanks to bbryant for pointing this out.


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2008-07-25 14:41:40 +00:00
Mark Michelson
78d821c4bd Blocked revisions 133570 via svnmerge
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r133570 | mmichelson | 2008-07-25 08:52:52 -0500 (Fri, 25 Jul 2008) | 5 lines

Add a missing unlock.

Pointed out by Atis Lezdins in #asterisk-dev


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2008-07-25 13:54:19 +00:00
Russell Bryant
6316323815 Merged revisions 133568 via svnmerge from
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r133568 | russell | 2008-07-25 08:01:59 -0500 (Fri, 25 Jul 2008) | 4 lines

Minor coding guidelines tweaks ...
 - Use ast_strlen_zero in one place
 - check for successful string comparison the way most of Asterisk code does it

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2008-07-25 13:25:10 +00:00
Russell Bryant
e80ba0bd83 Blocked revisions 133566 via svnmerge
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r133566 | russell | 2008-07-25 07:42:25 -0500 (Fri, 25 Jul 2008) | 4 lines

When the ast_device_state() function is called to retrieve device state, and
the code checks to see if there is a cached state available, use the aggregate
cached state across all servers, and not just the local state.

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2008-07-25 12:42:55 +00:00
Tilghman Lesher
640d750b3e Merged revisions 133509 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r133509 | tilghman | 2008-07-24 16:27:06 -0500 (Thu, 24 Jul 2008) | 11 lines

Merged revisions 133488 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r133488 | tilghman | 2008-07-24 16:17:55 -0500 (Thu, 24 Jul 2008) | 3 lines

Fix rtautoclear and rtcachefriends
(Closes issue #12707)

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2008-07-24 21:31:29 +00:00
Russell Bryant
b1fb1c8da1 Merged revisions 133486 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r133486 | russell | 2008-07-24 15:40:15 -0500 (Thu, 24 Jul 2008) | 3 lines

I made this change from DEVICE_STATE to DEVICE_STATE_CHANGE, but I had it backwards,
this is the right event to subscribe to ...

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2008-07-24 20:41:36 +00:00
Mark Michelson
38c51fee14 Merged revisions 133448 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r133448 | mmichelson | 2008-07-24 14:53:37 -0500 (Thu, 24 Jul 2008) | 12 lines

Print the correct PID in log messages. Prior to
this commit, only the logger thread's PID would
be printed.

(closes issue #13150)
Reported by: atis
Patches:
      log_pid.diff uploaded by putnopvut (license 60)
Tested by: eliel



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2008-07-24 19:55:12 +00:00
Tilghman Lesher
d944ce506e Merged revisions 133400 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r133400 | tilghman | 2008-07-24 00:21:00 -0500 (Thu, 24 Jul 2008) | 3 lines

Build the logrotate script according to paths
(Closes issue #13147)

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2008-07-24 05:21:56 +00:00
Tilghman Lesher
08aacecb1b Merged revisions 133391 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r133391 | tilghman | 2008-07-23 23:51:42 -0500 (Wed, 23 Jul 2008) | 3 lines

Optionally install logrotate file
(Closes issue #13148)

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2008-07-24 04:52:37 +00:00
Steve Murphy
b2d7015a42 Merged revisions 133299 via svnmerge from
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r133299 | murf | 2008-07-23 16:03:48 -0600 (Wed, 23 Jul 2008) | 27 lines

(closes issue #13144)
Reported by: murf
Tested by: murf
For: J. Geis

The 'data' field in the ast_exten struct was being
'moved' from the current dialplan to the replacement
dialplan. This was not good, as the current dialplan
could have problems in the time between the change
and when the new dialplan is swapped in.

So, I modified the merge_and_delete code to strdup
the 'data' field (the args to the app call), and
then it's freed as normal.

I improved a few messages; I added code to limit
the number of calls to the context_merge_incls_swits_igps_other_registrars()
to one per context. I don't think having it called
multiple times per context was doing anything bad,
but it was inefficient.

I hope this fixes the problems Mr. Geiss was noting in
asterisk-users, see 
http://lists.digium.com/pipermail/asterisk-users/2008-July/215634.html



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Jason Parker
f72194ea8c Merged revisions 133296 via svnmerge from
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r133296 | qwell | 2008-07-23 16:50:20 -0500 (Wed, 23 Jul 2008) | 9 lines

Merged revisions 133295 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r133295 | qwell | 2008-07-23 16:49:03 -0500 (Wed, 23 Jul 2008) | 1 line

inbandrelease is gone - it's now inbanddisconnect
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2008-07-23 21:50:57 +00:00
Brett Bryant
14bb7ff651 Merged revisions 133197 via svnmerge from
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r133197 | bbryant | 2008-07-23 15:33:22 -0500 (Wed, 23 Jul 2008) | 2 lines

Fix issue where tcp in sip is enabled by default, despite what it says in the config sample file. Also fix "sip show settings" for tcp connections.

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2008-07-23 20:39:37 +00:00
Mark Michelson
7d7e554191 Merged revisions 133171 via svnmerge from
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r133171 | mmichelson | 2008-07-23 14:48:03 -0500 (Wed, 23 Jul 2008) | 20 lines

Merged revisions 133169 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r133169 | mmichelson | 2008-07-23 14:39:47 -0500 (Wed, 23 Jul 2008) | 12 lines

As suggested by seanbright, the PSEUDO_CHAN_LEN in 
app_chanspy should be set at load time, not at compile
time, since dahdi_chan_name is determined at load time.

Also changed the next_unique_id_to_use to have the 
static qualifier.

Also added the dahdi_chan_name_len variable so that
strlen(dahdi_chan_name) isn't necessary. Thanks to
seanbright for the suggestion.


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Mark Michelson
492eb19e38 Merged revisions 133106 via svnmerge from
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r133106 | mmichelson | 2008-07-23 14:07:56 -0500 (Wed, 23 Jul 2008) | 13 lines

Merged revisions 133104 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r133104 | mmichelson | 2008-07-23 14:06:16 -0500 (Wed, 23 Jul 2008) | 5 lines

Zap/pseudo is ten characters, but DAHDI/pseudo is
twelve. The strncmp call in next_channel should
account for this.


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2008-07-23 19:08:22 +00:00
Mark Michelson
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r133102 | mmichelson | 2008-07-23 13:58:37 -0500 (Wed, 23 Jul 2008) | 14 lines

Merged revisions 133101 via svnmerge from 
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r133101 | mmichelson | 2008-07-23 13:57:17 -0500 (Wed, 23 Jul 2008) | 6 lines

Update the "last" channel in next_channel in app_chanspy so
that the same pseudo channel isn't constantly returned.

related to issue #13124


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Mark Michelson
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r133041 | mmichelson | 2008-07-23 12:54:03 -0500 (Wed, 23 Jul 2008) | 15 lines

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r133038 | mmichelson | 2008-07-23 12:50:01 -0500 (Wed, 23 Jul 2008) | 7 lines

Small cleanup. Move the declaration of the DAHDI_SPANINFO
variable to the block where it is used. This allows one
less #ifdef HAVE_PRI to clutter things up.

Thanks to Tzafrir for pointing this out on #asterisk-dev


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r132981 | tilghman | 2008-07-23 12:20:43 -0500 (Wed, 23 Jul 2008) | 6 lines

Yet another conversion of '|' to ','
(closes issue #13137)
 Reported by: eliel
 Patches: 
       chan_iax2trunk-IAXPEER.patch uploaded by eliel (license 64)

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r132977 | tilghman | 2008-07-23 12:14:56 -0500 (Wed, 23 Jul 2008) | 6 lines

Add logrotate script for Asterisk
(closes issue #13085)
 Reported by: pabelanger
 Patches: 
       logrotate uploaded by pabelanger (license 224)

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r132975 | mmichelson | 2008-07-23 12:07:08 -0500 (Wed, 23 Jul 2008) | 11 lines

Blocked revisions 132974 via svnmerge

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r132974 | mmichelson | 2008-07-23 12:05:04 -0500 (Wed, 23 Jul 2008) | 4 lines

Fix building of chan_dahdi when HAVE_PRI is not
defined.


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r132883 | crichter | 2008-07-23 07:07:15 -0500 (Wed, 23 Jul 2008) | 9 lines

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r132826 | crichter | 2008-07-23 13:37:50 +0200 (Mi, 23 Jul 2008) | 1 line

another Fix because of r119585, this commit has broken high frequented BRI Ports, there was a possibility that a channel, that was marked as in_use would be reused later, the corresponding port could got stuck then. So it is recommended to upgrade for chan_misdn users.
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r132966 | kpfleming | 2008-07-23 11:38:28 -0500 (Wed, 23 Jul 2008) | 2 lines

use correct function name... please compile with --enable-dev-mode

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r132872 | kpfleming | 2008-07-23 06:52:18 -0500 (Wed, 23 Jul 2008) | 2 lines

minor optimization for stringfields: when a field is being set to a larger value than it currently contains and it happens to be the most recent field allocated from the currentl pool, it is possible to 'grow' it without having to waste the space it is currently using (or potentially even allocate a new pool)

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Olle Johansson
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r132823 | oej | 2008-07-23 10:13:07 +0200 (Ons, 23 Jul 2008) | 8 lines

Well, the content of a channel variable may be longer than the size of a pointer...
Thanks, eliel!

Reported by: eliel
Patches: 
      chan_siptrunk.SIPPEER.patch uploaded by eliel (license 64)
(closes issue #13135)

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Mark Michelson
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r132795 | mmichelson | 2008-07-22 17:17:09 -0500 (Tue, 22 Jul 2008) | 11 lines

Merged revisions 132777 via svnmerge from 
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Allow Spiraled INVITEs to work correctly within Asterisk.

Prior to this change, a spiraled INVITE would cause a 482
Loop Detected to be sent to the caller. With this change,
if a potential loop is detected, the Request-URI is inspected
to see if it has changed from what was originally received. If
pedantic mode is on, then this inspection is fully RFC 3261
compliant. If pedantic mode is not on, then a string comparison
is used to test the equality of the two R-URIs.

This has been tested by using OpenSER to rewrite the R-URI
and send the INVITE back to Asterisk.

(closes issue #7403)
Reported by: stephen_dredge


Modified:
    branches/1.4/channels/chan_sip.c


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r132791 | kpfleming | 2008-07-22 17:14:37 -0500 (Tue, 22 Jul 2008) | 2 lines

correct fix made in r132777... the code *did* compile in dev-mode, as long as libpri was installed and enabled

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Olle Johansson
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r132703 | oej | 2008-07-22 22:46:11 +0200 (Tis, 22 Jul 2008) | 17 lines

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r132645 | oej | 2008-07-22 22:10:26 +0200 (Tis, 22 Jul 2008) | 9 lines

The most common question on the #asterisk iRC channel and on mailing lists
seems to be in regards to an error message when retransmit fails. This
is frequently misunderstood as a failure of Asterisk, not a failure of
the network to reach the other party.

This document tries to assist the Asterisk user in sorting out these
issues by explaining the logic and pointing at some possible 
causes. Hopefully, we will get other questions now :-)

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Tilghman Lesher
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r132778 | tilghman | 2008-07-22 16:53:40 -0500 (Tue, 22 Jul 2008) | 18 lines

Merged revisions 132713 via svnmerge from 
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r132713 | tilghman | 2008-07-22 16:19:39 -0500 (Tue, 22 Jul 2008) | 10 lines

Merged revisions 132711 via svnmerge from 
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r132711 | tilghman | 2008-07-22 16:14:10 -0500 (Tue, 22 Jul 2008) | 2 lines

Fixes for AST-2008-010 and AST-2008-011

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Mark Michelson
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r132777 | mmichelson | 2008-07-22 16:52:24 -0500 (Tue, 22 Jul 2008) | 3 lines

Get chan_dahdi to compile in devmode


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2008-07-22 21:54:21 +00:00
Kevin P. Fleming
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r132721 | kpfleming | 2008-07-22 16:21:56 -0500 (Tue, 22 Jul 2008) | 14 lines

Merged revisions 132712 via svnmerge from 
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r132712 | kpfleming | 2008-07-22 16:17:23 -0500 (Tue, 22 Jul 2008) | 6 lines

ensure that if any alarms exist at channel creation time, they are handled identically to if they occurred later, so that later alarm clearing will work properly and 'make sense'

(closes issue #12160)
Reported by: tzafrir


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Kevin P. Fleming
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r132705 | kpfleming | 2008-07-22 15:54:07 -0500 (Tue, 22 Jul 2008) | 10 lines

Merged revisions 132704 via svnmerge from 
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r132704 | kpfleming | 2008-07-22 15:49:41 -0500 (Tue, 22 Jul 2008) | 2 lines

make AST_C_COMPILE_CHECK able to print a 'pretty' description of what it is doing

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Kevin P. Fleming
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r132643 | kpfleming | 2008-07-22 14:59:10 -0500 (Tue, 22 Jul 2008) | 10 lines

Merged revisions 132641 via svnmerge from 
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r132641 | kpfleming | 2008-07-22 14:49:11 -0500 (Tue, 22 Jul 2008) | 2 lines

use renamed libpri API call for controlling this feature (was improperly named before)

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2008-07-22 20:03:35 +00:00
Kevin P. Fleming
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r132573 | kpfleming | 2008-07-21 17:51:16 -0500 (Mon, 21 Jul 2008) | 10 lines

Merged revisions 132571 via svnmerge from 
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r132571 | kpfleming | 2008-07-21 17:45:16 -0500 (Mon, 21 Jul 2008) | 2 lines

teach chan_dahdi how to find the D-channel on BRI spans, and don't attempt to use channel 24 as a D-channel on spans of unexpected sizes

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2008-07-21 22:52:36 +00:00
Brett Bryant
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r132514 | bbryant | 2008-07-21 16:12:51 -0500 (Mon, 21 Jul 2008) | 8 lines

Update configuration files to add missing options for jingle, gtalk, 
manager.conf, and features.conf.

(closes issue #13128)
Reported by: caio1982
Patches:
      missing_options1.diff uploaded by caio1982 (license 22)

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Tilghman Lesher
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r132511 | tilghman | 2008-07-21 16:00:47 -0500 (Mon, 21 Jul 2008) | 2 lines

(Step 2 of 2)

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Tilghman Lesher
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r132510 | tilghman | 2008-07-21 15:59:03 -0500 (Mon, 21 Jul 2008) | 5 lines

Optionally build integer-based routines for FSK tone decoding (but default
to the more accurate float-based routines).
(Closes issue #11679)
(Step 1 of 2)

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Brett Bryant
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r132508 | bbryant | 2008-07-21 15:54:09 -0500 (Mon, 21 Jul 2008) | 9 lines

Fix a bug where SENDTEXTSTATUS isn't set properly when it isn't 
supported on a channel (yet _another_ useful patch by eliel).

(closes issue #13081)
Reported by: eliel
Patches:
      app_sendtext.c.patch uploaded by eliel (license 64)
Tested by: eliel

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Brett Bryant
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r132468 | bbryant | 2008-07-21 12:42:45 -0500 (Mon, 21 Jul 2008) | 5 lines

Fix bug where ast_parse_arg would inadvertantly enable sip tcp when parsing a tcpbindaddr if it was disabled.

(closes issue #13117)
Reported by: pj

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Brett Bryant
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r132466 | bbryant | 2008-07-21 12:22:02 -0500 (Mon, 21 Jul 2008) | 3 lines

Fix an issue in iax2 where a call that's been rejected still kept an open channel on the side that attempted to make the call (not the side of the 
call that rejected the call). Changes were load tested and also approved by Russell.

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2008-07-21 17:25:09 +00:00
Jeff Peeler
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r132425 | jpeeler | 2008-07-21 10:33:13 -0500 (Mon, 21 Jul 2008) | 2 lines

make buffers config option (chan_dahdi.conf) parsing safer and added logging in case of failure

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2008-07-21 15:34:21 +00:00
Russell Bryant
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r132390 | russell | 2008-07-21 09:47:41 -0500 (Mon, 21 Jul 2008) | 16 lines

Remove libresample from the Asterisk source tree.  It is now available in its
own repository, and must be installed like any other library for Asterisk to
use.  The two modules that require it are codec_resample and app_jack.

To install libresample:

$ svn co http://svn.digium.com/svn/libresample/trunk libresample
$ cd libresample
$ ./configure
$ make
$ sudo make install

This code is currently in our own repository because the build system did not
include the appropriate targets for building a dynamic library or for installing
the library.

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Russell Bryant
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r132388 | russell | 2008-07-21 08:51:05 -0500 (Mon, 21 Jul 2008) | 3 lines

Enable higher quality resampling, as it doesn't have a noticeable performance
impact on my machine ..

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Kevin P. Fleming
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r132312 | kpfleming | 2008-07-19 11:46:33 -0500 (Sat, 19 Jul 2008) | 10 lines

Merged revisions 132311 via svnmerge from 
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r132311 | kpfleming | 2008-07-19 11:45:52 -0500 (Sat, 19 Jul 2008) | 2 lines

grant a license exception to allow distribution of Asterisk binaries that use the UW IMAP Toolkit (which is licensed under a non-GPL-compatible license)

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2008-07-19 16:47:12 +00:00
Michiel van Baak
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r132277 | mvanbaak | 2008-07-19 12:46:12 +0200 (Sat, 19 Jul 2008) | 7 lines

fix a couple of comments in sqlite resource driver.

(closes issue #13110)
Reported by: gknispel_proformatique
Patches:
      res_config_sqlite_comments.patch uploaded by gknispel (license 261)

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Brett Bryant
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r132242 | bbryant | 2008-07-18 17:19:56 -0500 (Fri, 18 Jul 2008) | 4 lines

Fixes problem where manager users loaded from users.conf would be 
removed early (before the routine to load the configuration was 
finished) because a variable wasn't initialized.

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Tilghman Lesher
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r132206 | tilghman | 2008-07-18 15:57:47 -0500 (Fri, 18 Jul 2008) | 5 lines

Russell pointed out that using ast_strdupa() within a loop like this is
probably not a good idea, as we might run out of stack space.  Therefore,
changing this over to use the ast_str infrastructure for buffers is
probably a good idea.

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r132203 | tilghman | 2008-07-18 15:41:02 -0500 (Fri, 18 Jul 2008) | 2 lines

Fix trunk devmode

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Tilghman Lesher
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r132113 | tilghman | 2008-07-18 14:09:39 -0500 (Fri, 18 Jul 2008) | 14 lines

Merged revisions 132112 via svnmerge from 
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r132112 | tilghman | 2008-07-18 14:06:10 -0500 (Fri, 18 Jul 2008) | 6 lines

Fix for Taiwanese number syntax
(closes issue #12319)
 Reported by: CharlesWang
 Patches: 
       saynumber-tw-1.4.18.1.patch uploaded by CharlesWang (license 444)

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2008-07-18 19:10:21 +00:00
Matthew Fredrickson
c1c0583b01 Merged revisions 132108 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r132108 | mattf | 2008-07-18 13:50:00 -0500 (Fri, 18 Jul 2008) | 1 line

Make sure we break the poll so that messages queued will be sent on the SS7 when using CLI commands for blocking and blocking of CICs and linksets.
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2008-07-18 18:53:54 +00:00
Tilghman Lesher
791cf5f2ce Merged revisions 132109 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r132109 | tilghman | 2008-07-18 13:50:37 -0500 (Fri, 18 Jul 2008) | 14 lines

Merged revisions 132107 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r132107 | tilghman | 2008-07-18 13:47:50 -0500 (Fri, 18 Jul 2008) | 6 lines

Textual clarification
(closes issue #13106)
 Reported by: flefoll
 Patches: 
       config.c.br14.120173.patch-unknown-directive uploaded by flefoll (license 244)

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2008-07-18 18:51:20 +00:00
Brett Bryant
86f0d82fc9 Merged revisions 132050 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r132050 | bbryant | 2008-07-18 12:55:41 -0500 (Fri, 18 Jul 2008) | 8 lines

Fix magic Revision keywords in hashtab.c and change cdr_radius.c to use 
the same keyword as the other files (patch by eliel).

(closes issue #13104)
Reported by: eliel
Patches:
      revision.patch uploaded by eliel (license 64)

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2008-07-18 17:56:14 +00:00
Tilghman Lesher
8e886494ec Blocked revisions 132046 via svnmerge
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r132046 | tilghman | 2008-07-18 12:39:18 -0500 (Fri, 18 Jul 2008) | 10 lines

Blocked revisions 132042 via svnmerge

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r132042 | tilghman | 2008-07-18 12:33:42 -0500 (Fri, 18 Jul 2008) | 3 lines

Spinlock within the destroy, to allow a scheduled job to continue, if it's
waiting on the mutex which the destroy thread has.

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2008-07-18 17:40:03 +00:00
Tilghman Lesher
7017604737 Merged revisions 131989 via svnmerge from
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r131989 | tilghman | 2008-07-18 12:10:34 -0500 (Fri, 18 Jul 2008) | 10 lines

Merged revisions 131988 via svnmerge from 
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r131988 | tilghman | 2008-07-18 12:10:01 -0500 (Fri, 18 Jul 2008) | 2 lines

Oops

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2008-07-18 17:11:10 +00:00
Tilghman Lesher
c01f4f499e Merged revisions 131986 via svnmerge from
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r131986 | tilghman | 2008-07-18 11:48:18 -0500 (Fri, 18 Jul 2008) | 10 lines

Merged revisions 131985 via svnmerge from 
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r131985 | tilghman | 2008-07-18 11:46:23 -0500 (Fri, 18 Jul 2008) | 2 lines

Preserve ABI compatibility with last change

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2008-07-18 16:49:44 +00:00
Tilghman Lesher
5abbe30e52 Merged revisions 131982 via svnmerge from
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r131982 | tilghman | 2008-07-18 11:33:56 -0500 (Fri, 18 Jul 2008) | 10 lines

Merged revisions 131970 via svnmerge from 
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r131970 | tilghman | 2008-07-18 11:30:31 -0500 (Fri, 18 Jul 2008) | 2 lines

Make the ast_assert call within ast_sched_del report something useful.

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2008-07-18 16:36:39 +00:00
Kevin P. Fleming
68c5ddc970 Merged revisions 131923 via svnmerge from
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r131923 | kpfleming | 2008-07-18 11:16:12 -0500 (Fri, 18 Jul 2008) | 10 lines

Merged revisions 131921 via svnmerge from 
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r131921 | kpfleming | 2008-07-18 11:15:41 -0500 (Fri, 18 Jul 2008) | 2 lines

remove the dlfcn compatibility stuff, because no platforms that Asterisk currently runs on it use it, and it doesn't build anyway

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2008-07-18 16:16:47 +00:00
Brett Bryant
bd279373b2 Merged revisions 131916 via svnmerge from
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r131916 | bbryant | 2008-07-18 10:38:22 -0500 (Fri, 18 Jul 2008) | 12 lines

Merged revisions 131915 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r131915 | bbryant | 2008-07-18 10:34:42 -0500 (Fri, 18 Jul 2008) | 4 lines

Fix a bug in blind transfers where the BLINDTRANSFER variable isn't always set to the other end of the blind transfer.

(closes issue #12586)

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2008-07-18 15:39:32 +00:00
Jeff Peeler
c27203aedb Merged revisions 131868 via svnmerge from
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r131868 | jpeeler | 2008-07-17 17:40:00 -0500 (Thu, 17 Jul 2008) | 6 lines

Add configuration option to chan_dahdi.conf to allow buffering policy and number of buffers to be configured per channel. Syntax:

buffers=<num of buffers>,<policy>

Where the number of buffers is some non-negative integer and the policy is either "full", "half", or "immediate".

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2008-07-17 22:45:50 +00:00
Mark Michelson
79e4d5873b Merged revisions 131824 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r131824 | mmichelson | 2008-07-17 16:26:41 -0500 (Thu, 17 Jul 2008) | 10 lines

Document that the duration of dtmf may be passed to
the SendDTMF application. Also correct the default
pause between digits.

(closes issue #13102)
Reported by: eliel
Patches:
      app_senddtmf.c.patch uploaded by eliel (license 64)


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2008-07-17 21:27:57 +00:00
Tilghman Lesher
160f124ffc Merged revisions 131791 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r131791 | tilghman | 2008-07-17 15:37:14 -0500 (Thu, 17 Jul 2008) | 15 lines

Merged revisions 131790 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r131790 | tilghman | 2008-07-17 15:35:44 -0500 (Thu, 17 Jul 2008) | 7 lines

Revert part of issue #5620 (revision 6965) as it appears that it was in error.
This should fix talk call progress on analog lines.
(closes issue #12178)
 Reported by: michael-fig
 Patches: 
       20080717__bug12178.diff.txt uploaded by Corydon76 (license 14)

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2008-07-17 20:38:07 +00:00
Tilghman Lesher
8d675a8660 Merged revisions 131753 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r131753 | tilghman | 2008-07-17 13:36:34 -0500 (Thu, 17 Jul 2008) | 6 lines

Fix memory leaks
(closes issue #13099)
 Reported by: gknispel_proformatique
 Patches: 
       res_config_sqlite_leak_on_error.patch uploaded by gknispel (license 261)

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2008-07-17 18:37:22 +00:00
Brett Bryant
384d11190e Merged revisions 131717 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r131717 | bbryant | 2008-07-17 13:14:42 -0500 (Thu, 17 Jul 2008) | 8 lines

Fix a memory leak in register_group_feature when attempting to register 
a feature without specifying a group or feature to register.

(closes issue #13101)
Reported by: eliel
Patches:
      features.c.patch uploaded by eliel (license 64)

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2008-07-17 18:15:10 +00:00
Tilghman Lesher
d98c350ebd Merged revisions 131681 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r131681 | tilghman | 2008-07-17 10:45:25 -0500 (Thu, 17 Jul 2008) | 4 lines

Fix memory leak.
(Closes issue #13096)
Reported by gknispel_proformatique

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2008-07-17 15:46:24 +00:00
Russell Bryant
8388731570 Blocked revisions 131643 via svnmerge
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r131643 | russell | 2008-07-17 09:46:29 -0500 (Thu, 17 Jul 2008) | 5 lines

Instead of attempting to pass through AST_EVENT_DEVICE_STATE, use DEVICE_STATE_CHANGE
instead.  DEVICE_STATE is a state change on one server, and DEVICE_STATE_CHANGE is
the "real" state of that device across all servers sharing state.  This would have
only been a problem with distributed device state.

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2008-07-17 14:47:02 +00:00
Tilghman Lesher
4475eb840e Blocked revisions 131606 via svnmerge
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r131606 | tilghman | 2008-07-17 09:00:27 -0500 (Thu, 17 Jul 2008) | 7 lines

Change several 'core' commands to be 'dialplan' commands (with appropriate
deprecation, of course)
(closes issue #13016)
 Reported by: caio1982
 Patches: 
       dialplan_globals6.diff uploaded by caio1982 (license 22)

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2008-07-17 14:01:09 +00:00
Steve Murphy
3b8753c2c0 The commit from 131570 should not be applied to 1.6.0,
as it is not as necessary, because log_show_lock in trunk
is not available in 1.6.0, and is estimated to be the
only function that might care about the lock_addr values.



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2008-07-16 23:56:00 +00:00
Brett Bryant
da77ba005f Merged revisions 131492 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r131492 | bbryant | 2008-07-16 17:17:36 -0500 (Wed, 16 Jul 2008) | 14 lines

Merged revisions 131491 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r131491 | bbryant | 2008-07-16 17:17:07 -0500 (Wed, 16 Jul 2008) | 6 lines

Fix a bug in iax2 registration that allowed peers to register with 
case-insensitive names (user_cmp_cb and peer_cmp_cb are now both 
case-sensitive).

(closes issue #13091)

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2008-07-16 22:18:12 +00:00
Tilghman Lesher
aedfb1a4e7 Blocked revisions 131482 via svnmerge
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r131482 | tilghman | 2008-07-16 16:50:08 -0500 (Wed, 16 Jul 2008) | 9 lines

Blocked revisions 131480 via svnmerge

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r131480 | tilghman | 2008-07-16 16:46:06 -0500 (Wed, 16 Jul 2008) | 2 lines

Apparently, in certain cases, a callno is already destroyed when iax2_destroy is called.

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2008-07-16 21:55:56 +00:00
Brett Bryant
19700017e1 Merged revisions 131484 via svnmerge from
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r131484 | bbryant | 2008-07-16 16:54:08 -0500 (Wed, 16 Jul 2008) | 4 lines

Fixes sysinfo operator issue also fixed elsewhere in r131445.

(issue #13057)

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2008-07-16 21:54:38 +00:00
Brett Bryant
c252f7f4eb Merged revisions 131445 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r131445 | bbryant | 2008-07-16 16:24:18 -0500 (Wed, 16 Jul 2008) | 9 lines

Fixes an issue with "core show sysinfo" that used the wrong operator to 
calculate the number of bytes from a sysinfo structure.
unsigned long.

(closes issue #13057)
Reported by: eliel
Patches:
      asterisk.c.patch uploaded by eliel (license 64)

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2008-07-16 21:26:13 +00:00
Russell Bryant
1d83b13558 Merged revisions 131422 via svnmerge from
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r131422 | russell | 2008-07-16 15:48:27 -0500 (Wed, 16 Jul 2008) | 15 lines

Merged revisions 131421 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r131421 | russell | 2008-07-16 15:47:53 -0500 (Wed, 16 Jul 2008) | 7 lines

Always ensure that the channel's tech_pvt reference is NULL after calling the
destroy callback.
(closes issue #13060)
Reported by: jpgrayson
Patches:
      chan_iax2_tech_pvt_crash.patch uploaded by jpgrayson (license 492)

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2008-07-16 20:48:49 +00:00
Mark Michelson
8d78d518bf Merged revisions 131375 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r131375 | mmichelson | 2008-07-16 15:24:12 -0500 (Wed, 16 Jul 2008) | 22 lines

Merged revisions 131369 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r131369 | mmichelson | 2008-07-16 15:23:02 -0500 (Wed, 16 Jul 2008) | 14 lines

Move the init_queue call back to where it used to be (changed
Sept 12 last year). It was moved then to prevent a memory leak.
Since then, the same memory leak recurred and was fixed in a 
better way.

Now it has been found that the placement of this init_queue
call can cause problems if a realtime queue has values changed
to an empty string. The problem is that the default value
for that queue parameter would not be set.

(closes issue #13084)
Reported by: elbriga


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2008-07-16 20:24:56 +00:00
Mark Michelson
46f51575e7 Merged revisions 131361 via svnmerge from
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r131361 | mmichelson | 2008-07-16 14:57:02 -0500 (Wed, 16 Jul 2008) | 9 lines

Don't try to dereference the dbfile pointer if
we know that it's NULL.

(closes issue #13092)
Reported by: gknispel_proformatique
Patches:
      trunk_sqlite_check_vars_null.patch uploaded by gknispel (license 261)


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2008-07-16 19:57:29 +00:00
Mark Michelson
fb892b269d Merged revisions 131358 via svnmerge from
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r131358 | mmichelson | 2008-07-16 14:37:42 -0500 (Wed, 16 Jul 2008) | 14 lines

Merged revisions 131357 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r131357 | mmichelson | 2008-07-16 14:37:08 -0500 (Wed, 16 Jul 2008) | 6 lines

Apparently, "thread safety" is important, whatever
that means. :P

(Thanks Russell!)


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2008-07-16 19:39:05 +00:00
Mark Michelson
58a8d7c521 Merged revisions 131300 via svnmerge from
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r131300 | mmichelson | 2008-07-16 13:59:27 -0500 (Wed, 16 Jul 2008) | 21 lines

Merged revisions 131299 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r131299 | mmichelson | 2008-07-16 13:57:34 -0500 (Wed, 16 Jul 2008) | 13 lines

Make absolutely certain that the transfer datastore
is removed from the calling channel once the caller
is finished in the queue. This could have weird con-
sequences when dialing local queue members when multiple
transfers occur on a single call.

Also fixed a memory leak that would occur when an
attended transfer occurred from a queue member.

(closes issue #13047)
Reported by: festr


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2008-07-16 18:59:59 +00:00
Steve Murphy
0559f958e0 Merged revisions 131243 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r131243 | murf | 2008-07-16 11:59:33 -0600 (Wed, 16 Jul 2008) | 27 lines

Merged revisions 131242 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r131242 | murf | 2008-07-16 11:53:43 -0600 (Wed, 16 Jul 2008) | 19 lines

(closes issue #13090)
Reported by: murf

The problem was that, esoteric as it is, because the hangerupper
context immediately preceded the std-priv-extent macro, that 
the checking code accidentally would fall from traversing hangerupper
into the std-priv-exten macro, where it would hit the hangerupper
in the 'includes', and proceed into an infinite recursion.

A small fix to traverse into the statements of the context instead
of the context solves this issue.

I also added some commented out printfs for debug, which were pretty
handy in the face of a dorky gdb.

This was a problem around since the package was first written;
but evidently pretty rare in turning up in the field.


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2008-07-16 18:20:48 +00:00
Luigi Rizzo
accfd377e7 add missing terminator argument for ast_event_subscribe().
Without it the function will randomly walk on the stack
possibly causing a panic


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@131206 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-16 15:04:25 +00:00
Tilghman Lesher
787a601357 Merged revisions 131166 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r131166 | tilghman | 2008-07-15 19:52:48 -0500 (Tue, 15 Jul 2008) | 3 lines

Fix rotate strategy
(Closes issue #13086)

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2008-07-16 00:54:50 +00:00
Steve Murphy
d2d03eb22a Merged revisions 131129 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r131129 | murf | 2008-07-15 17:36:19 -0600 (Tue, 15 Jul 2008) | 21 lines


(closes issue #12960)
Reported by: mnicholson

Spent most of the day on this bug, and the
solution was so simple. Just had to find and
understand the problem.

The problem was, that the routine to copy
the existing switches, includes, and ignorepats
from the old context to the new one, wasn't
getting called when the context is already 
existent. (In other words, if AEL is adding
a new context to the mix, they get copied,
but if pbx_config already defined a context,
then the copy wasn't happening. This made
no sense, so I moved the call to copy the 
includes & etc, no matter the case.



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2008-07-15 23:41:22 +00:00
Russell Bryant
48b083262c Merged revisions 131072 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r131072 | russell | 2008-07-15 13:46:40 -0500 (Tue, 15 Jul 2008) | 5 lines

Fix a couple of places in res_agi where the agi_commands lock would not be
released, causing a deadlock.  (Reported by mvanbaak in #asterisk-dev,
discovered by bbryant's change to the lock tracking code to yell at you
if a thread exits with a lock still held)

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2008-07-15 18:47:12 +00:00
Tilghman Lesher
3570646e54 Merged revisions 131044 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r131044 | tilghman | 2008-07-15 13:25:34 -0500 (Tue, 15 Jul 2008) | 16 lines

Merged revisions 130959 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r130959 | tilghman | 2008-07-15 12:19:13 -0500 (Tue, 15 Jul 2008) | 8 lines

astman_send_error does not need a newline appended -- the API takes care of
that for us.
(closes issue #13068)
 Reported by: gknispel_proformatique
 Patches: 
       asterisk_1_4_astman_send.patch uploaded by gknispel (license 261)
       asterisk_trunk_astman_send.patch uploaded by gknispel (license 261)

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2008-07-15 18:29:31 +00:00
Michiel van Baak
ffe8eef604 Merged revisions 131013 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r131013 | mvanbaak | 2008-07-15 19:49:48 +0200 (Tue, 15 Jul 2008) | 15 lines

Merged revisions 131012 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r131012 | mvanbaak | 2008-07-15 19:47:15 +0200 (Tue, 15 Jul 2008) | 7 lines

remove 4 lines of redundant code.

(closes issue #13080)
Reported by: gknispel_proformatique
Patches:
      trunk_ast_cdr_setapp.patch uploaded by gknispel (license 261)

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2008-07-15 18:00:24 +00:00
Steve Murphy
e45d4f3bf1 Merged revisions 130145 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

Merging this rev from trunk to 1.6.0 was not
simple. Why? Because we've enhanced trunk to
do a [fast] merge-and-delete operation which 
also solved problems with contexts having 
entries from different registrars.
Fast as in the amount of time the contexts
are locked down. That *is* fast, but traversing
the entire dialplan looking for priorities to
delete takes more time overall.
This particular fix involved pulling in those
enhancements from trunk, along with all the
various fixes and refinements made along the
way.

Merging all this from trunk into 1.6 involved:
a. mergetrunk6 in the stuff from 130145;
b. revert all but the prop changes
c. catalog all revisions to pbx.c since 1.6.0 was forked
   (at rev 105596).
d. catalog all revisions to pbx.c in trunk since 1.6.0
   was forked, making special note of all revs that
   were not merged into 1.6.0.
e. study each rev in trunk not applied to 1.6.0, and
   determine if it was involved in the merge_and_delete
   enhancements in trunk. 25 commits were done in 1.6.0,
   all but one (106306) was a merge from trunk.
   Trunk had 22 additional changes, of which 7 were
   involved in the merge_and_delete enhancements:
    106757
    108894
    109169
    116461
    123358
    130145
    130297
f. Go to trunk and collect patches, one by one,
   of the changes made by each rev across the
   entire source tree, using svn diff -c <num> > pfile
g. Apply each patch in order to 1.6.0, and 
   resolve all failures and compilation problems
   before proceding to the next patch.
h. test the stuff.
i. profit!


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r130145 | murf | 2008-07-11 12:24:31 -0600 (Fri, 11 Jul 2008) | 40 lines

(closes issue #13041)
Reported by: eliel
Tested by: murf

(closes issue #12960)
Reported by: mnicholson

In this 'omnibus' fix, I **think** I solved both
the problem in 13041, where unloading pbx_ael.so
caused crashes, or incomplete removal of previous
registrar'ed entries. And I added code to completely
remove all includes, switches, and ignorepats that
had a matching registrar entry, which should
appease 12960.

I also added a lot of seemingly useless brackets
around single statement if's, which helped debug 
so much that I'm leaving them there.

I added a routine to check the correlation between
the extension tree lists and the hashtab 
tables. It can be amazingly helpful when you have
lots of dialplan stuff, and need to narrow
down where a problem is occurring. It's ifdef'd
out by default.

I cleaned up the code around the new CIDmatch code.
It was leaving hanging extens with bad ptrs, getting confused
over which objects to remove, etc. I tightened
up the code and changed the call to remove_exten
in the merge_and_delete code.

I added more conditions to check for empty context
worthy of deletion. It's not empty if there are
any includes, switches, or ignorepats present.

If I've missed anything, please re-open this bug,
and be prepared to supply example dialplan code.


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2008-07-15 13:14:07 +00:00
Tilghman Lesher
d962a11e55 Merged revisions 130890 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r130890 | tilghman | 2008-07-14 18:59:54 -0500 (Mon, 14 Jul 2008) | 16 lines

Merged revisions 130889 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r130889 | tilghman | 2008-07-14 18:59:13 -0500 (Mon, 14 Jul 2008) | 8 lines

Override the callerid in all cases when the callerid is set in the user, not
just when a remote callerid is set.  Also, if not set in the user, allow the
remote CallerID to pass through.
(closes issue #12875)
 Reported by: dimas
 Patches: 
       20080714__bug12875.diff.txt uploaded by Corydon76 (license 14)

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2008-07-15 00:00:34 +00:00
Mark Michelson
9766aed395 Merged revisions 130854 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r130854 | mmichelson | 2008-07-14 17:22:57 -0500 (Mon, 14 Jul 2008) | 9 lines

Fix a memory leak in the case that /dev/null cannot be
opened when running startup commands from cli.conf

(closes issue #13066)
Reported by: eliel
Patches:
      asterisk.c.patch uploaded by eliel (license 64)


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2008-07-14 22:24:01 +00:00
Mark Michelson
033a630385 Merged revisions 130794 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r130794 | mmichelson | 2008-07-14 12:54:11 -0500 (Mon, 14 Jul 2008) | 16 lines

Merged revisions 130792 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r130792 | mmichelson | 2008-07-14 12:50:21 -0500 (Mon, 14 Jul 2008) | 8 lines

Add a check to the CAN_EARLY_BRIDGE macro in app_dial to
be sure there are no audiohooks present on the channels
involved. This fixed a one-way audio situation I had in
my test setup. I couldn't find any open issues that suggested
one-way audio with regards to mixmonitor (or other audiohook)
usage, though.


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2008-07-14 17:55:01 +00:00
Michiel van Baak
4a57bf3e3d Merged revisions 130744 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r130744 | mvanbaak | 2008-07-14 19:21:18 +0200 (Mon, 14 Jul 2008) | 18 lines

Merged revisions 130735 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r130735 | mvanbaak | 2008-07-14 19:10:21 +0200 (Mon, 14 Jul 2008) | 10 lines

notify the user that dnsmgr refresh wont work when dnsmgr is not enabled.
Previously this command would automagically appear and disappear.
This was confusing.

(closes issue #12796)
Reported by: chappell
Patches:
      dnsmgr_refresh_3.diff uploaded by chappell (license 8)
Tested by: russell, chappell, mvanbaak

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2008-07-14 17:22:54 +00:00
Tilghman Lesher
44ab22fe7f Blocked revisions 130697 via svnmerge
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r130697 | tilghman | 2008-07-14 10:44:07 -0500 (Mon, 14 Jul 2008) | 7 lines

Swap "static" and "const", so that "static" appears at the beginning of each
declaration (suppresses a warning).
(closes issue #13070)
 Reported by: gknispel_proformatique
 Patches: 
       asterisk_trunk_const_static.patch uploaded by gknispel (license 261)

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2008-07-14 15:44:49 +00:00
Russell Bryant
0ea127b043 Merged revisions 129987 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r129987 | russell | 2008-07-11 09:22:44 -0500 (Fri, 11 Jul 2008) | 10 lines

Merged revisions 129970 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r129970 | russell | 2008-07-11 09:18:43 -0500 (Fri, 11 Jul 2008) | 2 lines

add a simple ASTOBJ_TRYWRLOCK macro ...

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2008-07-14 10:40:53 +00:00
Russell Bryant
c75d610942 Merged revisions 130635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r130635 | russell | 2008-07-14 05:39:23 -0500 (Mon, 14 Jul 2008) | 10 lines

Merged revisions 130634 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r130634 | russell | 2008-07-14 05:38:14 -0500 (Mon, 14 Jul 2008) | 2 lines

Bump up the debug level for a message.

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2008-07-14 10:40:20 +00:00
Michiel van Baak
170c05fcc4 Merged revisions 130578 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r130578 | mvanbaak | 2008-07-14 01:14:03 +0200 (Mon, 14 Jul 2008) | 15 lines

Make all sed calls Posix sed compatible.
To make sure nobody commits script-modified files we first make a backup
of asterisk.tex, run the script, generate the pdf and / or html,
and put the original asterisk.tex back.
This will guard us for the stuff that happened before that someone committed 
a locally modified asterisk.tex, with changes done by this script.

(closes issue #13062)
Reported by: mvanbaak
Patches:
      sed_without-i-v3.diff uploaded by mvanbaak (license 7)
Tested by: mvanbaak

Feedback from Corydon. Thanks for taking the time to go through this.

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2008-07-13 23:20:11 +00:00
Michiel van Baak
cab0fec957 Blocked revisions 130576 via svnmerge
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r130576 | mvanbaak | 2008-07-14 00:59:35 +0200 (Mon, 14 Jul 2008) | 9 lines

Convert chan_skinny's open-coded linked lists to the list macros

(closes issue #12956)
Reported by: DEA
Patches:
      chan_skinny-linkedlists-v2.txt uploaded by DEA (license 3)
Tested by: DEA, mvanbaak


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2008-07-13 23:01:14 +00:00
Michiel van Baak
b92a2e0e53 Merged revisions 130574 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r130574 | mvanbaak | 2008-07-14 00:50:31 +0200 (Mon, 14 Jul 2008) | 16 lines

Merged revisions 130573 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r130573 | mvanbaak | 2008-07-14 00:48:51 +0200 (Mon, 14 Jul 2008) | 8 lines

fix memory leak when originate from manager cannot create a thread

(closes issue #13069)
Reported by: gknispel_proformatique
Patches:
      asterisk_trunk_action_originate.patch uploaded by gknispel (license 261)
	  Tested by: gknispel_proformatique, mvanbaak

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2008-07-13 22:53:25 +00:00
Tilghman Lesher
21e2e2de49 Merged revisions 130515 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r130515 | tilghman | 2008-07-13 12:58:47 -0500 (Sun, 13 Jul 2008) | 12 lines

Merged revisions 130514 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r130514 | tilghman | 2008-07-13 12:56:10 -0500 (Sun, 13 Jul 2008) | 4 lines

Reverting 2 changesets, as it breaks incoming IAX2 calls
(Related to issue #12963)
Reported by: mvanbaak

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2008-07-13 17:59:34 +00:00
Michiel van Baak
88cbe002af Merged revisions 130479 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r130479 | mvanbaak | 2008-07-13 16:58:40 +0200 (Sun, 13 Jul 2008) | 3 lines

restore ASTERISKVERSION marker to asterisk.tex.
This got lost in commit 97634

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2008-07-13 15:00:09 +00:00
Tilghman Lesher
d7bda637b5 Merged revisions 130444 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r130444 | tilghman | 2008-07-12 21:34:32 -0500 (Sat, 12 Jul 2008) | 2 lines

Unlock list before returning

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2008-07-13 02:35:22 +00:00
Michiel van Baak
bac969582f Blocked revisions 130374 via svnmerge
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r130374 | mvanbaak | 2008-07-12 12:26:55 +0200 (Sat, 12 Jul 2008) | 13 lines

Blocked revisions 130373 via svnmerge

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r130373 | mvanbaak | 2008-07-12 12:25:52 +0200 (Sat, 12 Jul 2008) | 6 lines

in 1.4 the functions still have | as argument seperator.
This commit fixes the use of RAND in the ael random function.

(closes issue #13061)
Reported by: danpwi

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2008-07-12 10:28:39 +00:00
Matthew Fredrickson
6e55f169dd Merged revisions 130293 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r130293 | mattf | 2008-07-11 16:36:26 -0500 (Fri, 11 Jul 2008) | 1 line

Support new TRANSPORT definitions in libss7
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2008-07-11 21:39:48 +00:00
Mark Michelson
ec482339b7 Merged revisions 130237 via svnmerge from
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r130237 | mmichelson | 2008-07-11 15:03:55 -0500 (Fri, 11 Jul 2008) | 11 lines

Merged revisions 130236 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r130236 | mmichelson | 2008-07-11 15:03:23 -0500 (Fri, 11 Jul 2008) | 3 lines

Remove redundant logic


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2008-07-11 20:04:25 +00:00
Tilghman Lesher
251abd9a17 Blocked revisions 130234 via svnmerge
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r130234 | tilghman | 2008-07-11 14:56:34 -0500 (Fri, 11 Jul 2008) | 2 lines

Don't copy on NULL.

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2008-07-11 19:57:27 +00:00
Tilghman Lesher
69c18e0bdb Blocked revisions 130232 via svnmerge
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r130232 | tilghman | 2008-07-11 14:53:38 -0500 (Fri, 11 Jul 2008) | 2 lines

Add some debug code and add a missing release

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2008-07-11 19:54:51 +00:00
Tilghman Lesher
9b5c5f096a Merged revisions 130230 via svnmerge from
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r130230 | tilghman | 2008-07-11 14:40:55 -0500 (Fri, 11 Jul 2008) | 2 lines

Fix trunk breakage

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2008-07-11 19:45:25 +00:00
Mark Michelson
feead8b757 Merged revisions 130174 via svnmerge from
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r130174 | mmichelson | 2008-07-11 14:14:15 -0500 (Fri, 11 Jul 2008) | 15 lines

Merged revisions 130173 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r130173 | mmichelson | 2008-07-11 14:13:29 -0500 (Fri, 11 Jul 2008) | 7 lines

Fix a typo in audiohook_read_frame_both.

While this change has not been proven to fix any
specific issue, it is incorrect and could cause
unforeseen problems.


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2008-07-11 19:14:50 +00:00
Tilghman Lesher
f5e6714b87 Merged revisions 130170 via svnmerge from
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r130170 | tilghman | 2008-07-11 13:52:42 -0500 (Fri, 11 Jul 2008) | 15 lines

Merged revisions 130169 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r130169 | tilghman | 2008-07-11 13:51:56 -0500 (Fri, 11 Jul 2008) | 7 lines

Ensure that a destination callno of 0 will not match for frames that do not
start a dialog (new, lagrq, and ping).
(closes issue #12963)
 Reported by: russellb
 Patches: 
       chan_iax2_dup_new_fix4.patch uploaded by jpgrayson (license 492)

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2008-07-11 18:53:22 +00:00
Sean Bright
2d96db5069 Merged revisions 130167 via svnmerge from
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r130167 | seanbright | 2008-07-11 14:32:26 -0400 (Fri, 11 Jul 2008) | 1 line

Missed one.  Formatting only.
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2008-07-11 18:33:24 +00:00
Brett Bryant
559bbfc457 Merged revisions 130129 via svnmerge from
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r130129 | bbryant | 2008-07-11 13:09:35 -0500 (Fri, 11 Jul 2008) | 8 lines

Janitor patch to change uses of sizeof to ARRAY_LEN

(closes issue #13054)
Reported by: pabelanger
Patches:
      ARRAY_LEN.patch2 uploaded by pabelanger (license 224)
Tested by: seanbright

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2008-07-11 18:14:27 +00:00
Tilghman Lesher
05df0355d9 Merged revisions 130126 via svnmerge from
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r130126 | tilghman | 2008-07-11 12:29:24 -0500 (Fri, 11 Jul 2008) | 17 lines

Merged revisions 130102 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r130102 | tilghman | 2008-07-11 11:50:42 -0500 (Fri, 11 Jul 2008) | 9 lines

Pass the devicestate from an underlying channel up through the Agent channel.
This should make the Agent always report the correct device state, even when
the underlying channel is used for other purposes.
(closes issue #12773)
 Reported by: davidw
 Patches: 
       20080710__bug12773.diff.txt uploaded by Corydon76 (license 14)
 Tested by: davidw

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2008-07-11 17:30:07 +00:00
Kevin P. Fleming
288369308e Merged revisions 130044 via svnmerge from
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r130044 | kpfleming | 2008-07-11 11:18:01 -0500 (Fri, 11 Jul 2008) | 2 lines

clean up a bunch more Zaptel-related references

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2008-07-11 16:18:54 +00:00
Kevin P. Fleming
f95462f418 Merged revisions 130040 via svnmerge from
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r130040 | kpfleming | 2008-07-11 10:57:17 -0500 (Fri, 11 Jul 2008) | 12 lines

Merged revisions 130039 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r130039 | kpfleming | 2008-07-11 10:41:56 -0500 (Fri, 11 Jul 2008) | 4 lines

add support for a configuration parameter for 'inband audio during RELEASE', which is currently mandatory in libpri-1.4.4 but will become configurable in libpri-1.4.5 later today

(related to issue #13042)

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2008-07-11 16:05:02 +00:00
Kevin P. Fleming
5ce44b2532 Merged revisions 129968 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r129968 | kpfleming | 2008-07-11 09:16:15 -0500 (Fri, 11 Jul 2008) | 18 lines

Merged revisions 129966 via svnmerge from 
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r129966 | kpfleming | 2008-07-11 09:03:52 -0500 (Fri, 11 Jul 2008) | 5 lines

fix a flaw found while experimenting with structure alignment and padding; low-fence checking would not work properly on 64-bit platforms, because the compiler was putting 4 bytes of padding between the fence field and the allocation memory block

added a very obvious runtime warning if this condition reoccurs, so the developer who broke it can be chastised into fixing it :-)


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r129967 | kpfleming | 2008-07-11 09:03:52 -0500 (Fri, 11 Jul 2008) | 5 lines

simplify calculation

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2008-07-11 14:16:55 +00:00
Kevin P. Fleming
19e89b7b27 Merged revisions 129916 via svnmerge from
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r129916 | kpfleming | 2008-07-11 07:21:29 -0500 (Fri, 11 Jul 2008) | 10 lines

Merged revisions 129907 via svnmerge from 
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r129907 | kpfleming | 2008-07-11 07:15:42 -0500 (Fri, 11 Jul 2008) | 2 lines

don't attempt to set user/group ownership of extracted sound files (reported on asterisk-users)

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2008-07-11 12:26:43 +00:00
Sean Bright
9b1925c172 Merged revisions 129864 via svnmerge from
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r129864 | seanbright | 2008-07-10 20:55:06 -0400 (Thu, 10 Jul 2008) | 1 line

Fix some usages of snprintf, and clarify a couple variable names.
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2008-07-11 01:01:39 +00:00
Tilghman Lesher
334b1ee95f Merged revisions 129804 via svnmerge from
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r129804 | tilghman | 2008-07-10 17:06:07 -0500 (Thu, 10 Jul 2008) | 16 lines

Merged revisions 129803 via svnmerge from 
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r129803 | tilghman | 2008-07-10 16:57:05 -0500 (Thu, 10 Jul 2008) | 8 lines

Correctly deal with duplicate NEW frames (due to retransmission).  Also, fixup
the destination call number matching to be more strict and reliable.
(closes issue #12963)
 Reported by: jpgrayson
 Patches: 
       chan_iax2_dup_new_fix3.patch uploaded by jpgrayson (license 492)
 Tested by: jpgrayson, Corydon76

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2008-07-10 22:07:12 +00:00
Tilghman Lesher
54d1886d1b Merged revisions 129758 via svnmerge from
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r129758 | tilghman | 2008-07-10 16:23:23 -0500 (Thu, 10 Jul 2008) | 10 lines

Merged revisions 129741 via svnmerge from 
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r129741 | tilghman | 2008-07-10 16:19:48 -0500 (Thu, 10 Jul 2008) | 2 lines

Oops

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2008-07-10 21:24:31 +00:00
Terry Wilson
0f476f87d0 Merged revisions 129738 via svnmerge from
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r129738 | twilson | 2008-07-10 15:56:20 -0500 (Thu, 10 Jul 2008) | 2 lines

Move phoneprov config files to be installed with 'make samples' so changes aren't inadvertently lost on a 'make install'

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2008-07-10 21:05:05 +00:00
Mark Michelson
7ac7889760 Blocked revisions 129734 via svnmerge
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r129734 | mmichelson | 2008-07-10 15:33:13 -0500 (Thu, 10 Jul 2008) | 23 lines

Removed the fn2 field from the vm_state structure.

fn2 was used in three functions. In every case, it was initialized
in the function it was used in. This meant there was no need
to have it in a malloc'd structure just taking up space. Furthermore
two of the functions it was used in were completely unnecessary since
fn2 was set to exactly the same value as the vm_state's fn string.

fn2 was a char array sized at PATH_MAX. On my system, PATH_MAX is 
4096. This equates to a 4K memory savings per vm_state allocated. 
Since there is a vm_state malloc'd for every voicemail user on 
the system, this could potentially add up nicely if there are lots 
of users. In addition, a vm_state is allocated on the stack each 
time a caller calls the VoiceMailMain application, meaning that 
there is a significant stack savings with this patch too.

Of course, a single vm_state struct still takes up approximately
20K on my system (when using IMAP storage. Without IMAP storage,
there would be about another 300 bytes fewer usage), even with 
this removal. Further optimizations are probably possible, 
but most likely not as easy as this one.


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2008-07-10 20:34:12 +00:00
Brett Bryant
f6f918d6a6 Merged revisions 129684 via svnmerge from
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r129684 | bbryant | 2008-07-10 14:13:12 -0500 (Thu, 10 Jul 2008) | 8 lines

Fixes a bug where the interface for a queue member gets reloaded as the state_interface, if a state_interface was set, on reload because the 
state_interface isn't stored in the ast_db.

(closes issue #13043)
Reported by: jvandal
Patches:
      app_queue.patch uploaded by jvandal (license 413)

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2008-07-10 19:14:24 +00:00
Sean Bright
03a55751dc Merged revisions 129642 via svnmerge from
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r129642 | seanbright | 2008-07-10 14:19:17 -0400 (Thu, 10 Jul 2008) | 1 line

A couple more minor text changes
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2008-07-10 18:20:22 +00:00
Sean Bright
663c3a4f2e Merged revisions 129638 via svnmerge from
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r129638 | seanbright | 2008-07-10 14:16:21 -0400 (Thu, 10 Jul 2008) | 1 line

Remove extraneous \n.  Pointed out by eliel on #asterisk-dev.
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2008-07-10 18:18:07 +00:00
Michiel van Baak
2f642bc636 Blocked revisions 129581 via svnmerge
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r129581 | mvanbaak | 2008-07-10 18:21:16 +0200 (Thu, 10 Jul 2008) | 8 lines

Remove deprecated 'show parkedcalls' CLI command

(closes issue #13038)
Reported by: eliel
Patches:
      finish.deprecate.patch uploaded by eliel (license 64)


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2008-07-10 16:24:40 +00:00
Russell Bryant
f6adad7c52 Merged revisions 129569 via svnmerge from
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r129569 | russell | 2008-07-10 11:12:51 -0500 (Thu, 10 Jul 2008) | 11 lines

Merged revisions 129567 via svnmerge from 
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r129567 | russell | 2008-07-10 11:03:59 -0500 (Thu, 10 Jul 2008) | 3 lines

Note that pbx_spool.so is the module used for call files
(inspired by a question in #asterisk)

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2008-07-10 16:13:09 +00:00
Sean Bright
aee532472b Blocked revisions 129506 via svnmerge
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r129506 | seanbright | 2008-07-10 10:09:18 -0400 (Thu, 10 Jul 2008) | 9 lines

Blocked revisions 129505 via svnmerge

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r129505 | seanbright | 2008-07-10 09:57:19 -0400 (Thu, 10 Jul 2008) | 2 lines

Update svn:ignore

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2008-07-10 14:09:52 +00:00
Sean Bright
27ab814e73 Merged revisions 129503 via svnmerge from
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r129503 | seanbright | 2008-07-10 09:54:29 -0400 (Thu, 10 Jul 2008) | 2 lines

Update svn:ignore

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2008-07-10 13:56:10 +00:00
Mark Michelson
f68d786c86 Merged revisions 129437 via svnmerge from
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r129437 | mmichelson | 2008-07-09 14:40:30 -0500 (Wed, 09 Jul 2008) | 21 lines

Merged revisions 129436 via svnmerge from 
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r129436 | mmichelson | 2008-07-09 14:32:20 -0500 (Wed, 09 Jul 2008) | 13 lines

Fix a problem where inbound rfc2833 audio would be sent to the 
core instead of being P2P bridged. When the core regenerated
the rfc2833 packet for the outbound leg, the SSRC would be different
than the RTP audio on the call leg causing DTMF detection issues on
the far end.

(closes issue #12955)
Reported by: tonyredstone
Patches:
      dynamic_rtp.patch uploaded by tsearle (license 373)
Tested by: tonyredstone


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2008-07-09 19:41:31 +00:00
Matthew Fredrickson
d00a013b44 Merged revisions 129399 via svnmerge from
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r129399 | mattf | 2008-07-09 10:57:06 -0500 (Wed, 09 Jul 2008) | 1 line

Add Proceeding() application (#13025)
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2008-07-09 16:01:49 +00:00
Sean Bright
b3ca68f90b Merged revisions 129344 via svnmerge from
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r129344 | seanbright | 2008-07-09 09:44:43 -0400 (Wed, 09 Jul 2008) | 12 lines

Merged revisions 129343 via svnmerge from 
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r129343 | seanbright | 2008-07-09 09:41:21 -0400 (Wed, 09 Jul 2008) | 4 lines

Look for the system installed awk instead of assuming it's at /usr/bin/awk.

Pointed out by jmls via #asterisk-dev.

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2008-07-09 13:46:07 +00:00
Mark Michelson
a44c6c7dea Merged revisions 129270 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r129270 | mmichelson | 2008-07-08 17:56:12 -0500 (Tue, 08 Jul 2008) | 3 lines

Fix compilation error when IMAP storage is enabled


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2008-07-08 22:56:55 +00:00
Mark Michelson
bd9887b558 Blocked revisions 129210 via svnmerge
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r129210 | mmichelson | 2008-07-08 16:32:09 -0500 (Tue, 08 Jul 2008) | 10 lines

Blocked revisions 129208 via svnmerge

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r129208 | mmichelson | 2008-07-08 16:31:51 -0500 (Tue, 08 Jul 2008) | 3 lines

Update documentation to have the correct option name


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2008-07-08 21:32:29 +00:00
Mark Michelson
87a53246fb Blocked revisions 129159 via svnmerge
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r129159 | mmichelson | 2008-07-08 16:10:33 -0500 (Tue, 08 Jul 2008) | 21 lines

Blocked revisions 129158 via svnmerge

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r129158 | mmichelson | 2008-07-08 16:09:10 -0500 (Tue, 08 Jul 2008) | 14 lines

Backport TCP-related timeouts to IMAP voicemail in 1.4
since it should solve bugs people are experiencing. Specifically,
there are times where communication with the IMAP server causes
system calls to block forever. If this should happen when querying
the mailbox so that chan_sip's do_monitor thread can send MWI to
a phone, it means that SIP calls cannot be processed any more.

The timeout options are outlined in doc/imapstorage.txt. Defaults
for the timeouts are sixty seconds.

(closes issue #12987)
Reported by: mthomasslo


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2008-07-08 21:16:31 +00:00
Brett Bryant
1b3cb408f5 Merged revisions 129156 via svnmerge from
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r129156 | bbryant | 2008-07-08 16:00:01 -0500 (Tue, 08 Jul 2008) | 6 lines

Fix a bug in SRV lookups where dnsmgr would discard everything but the 
first SRV result from DNS before processing weights and priorities and 
dns_parse_answer wouldn't report that there were no records in DNS 
unless a failure occured. Also fixed a bug where dnsmgr_refresh would 
report that a entry was being changed when ast_gethostbyname had failed.

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2008-07-08 21:04:56 +00:00
Tilghman Lesher
7d9d663991 Merged revisions 129152 via svnmerge from
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r129152 | tilghman | 2008-07-08 15:30:29 -0500 (Tue, 08 Jul 2008) | 16 lines

Merged revisions 129149 via svnmerge from 
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r129149 | tilghman | 2008-07-08 15:27:47 -0500 (Tue, 08 Jul 2008) | 8 lines

Cause SIP to return a 480 instead of a 404 when a sip peer exists, but is not
registered.
(closes issue #12885)
 Reported by: ibc
 Patches: 
       20080701__bug12885__2.diff.txt uploaded by Corydon76 (license 14)
 Tested by: ibc

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2008-07-08 20:31:43 +00:00
Tilghman Lesher
d42a2e914b Blocked revisions 129114 via svnmerge
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r129114 | tilghman | 2008-07-08 15:17:08 -0500 (Tue, 08 Jul 2008) | 8 lines

Reduce length of time that 'asterisk -rx' waits.
(closes issue #13001)
 Reported by: eliel
 Patches: 
       20080708__bug13001.diff.txt uploaded by Corydon76 (license 14)
       20080708__bug13001.diff.txt.fixed uploaded by eliel (license 64)
 Tested by: Corydon76, eliel

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2008-07-08 20:17:45 +00:00
Tilghman Lesher
297ac91c79 Merged revisions 129048 via svnmerge from
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r129048 | tilghman | 2008-07-08 11:49:01 -0500 (Tue, 08 Jul 2008) | 15 lines

Merged revisions 129047 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r129047 | tilghman | 2008-07-08 11:45:23 -0500 (Tue, 08 Jul 2008) | 7 lines

Timestamp decoding for video mini-frames is bogus, because the timestamp only
includes 15 bits, unlike voice frames, which contain a 16-bit timestamp.
(closes issue #13013)
 Reported by: jpgrayson
 Patches: 
       chan_iax2_unwrap_ts.patch uploaded by jpgrayson (license 492)

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2008-07-08 16:50:29 +00:00
Brett Bryant
b065565bd4 Merged revisions 129045 via svnmerge from
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r129045 | bbryant | 2008-07-08 11:40:28 -0500 (Tue, 08 Jul 2008) | 7 lines

Janitor project to convert sizeof to ARRAY_LEN macro.

(closes issue #13002)
Reported by: caio1982
Patches:
      janitor_arraylen5.diff uploaded by caio1982 (license 22)

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2008-07-08 16:41:31 +00:00
Brett Bryant
bc6553eeec Merged revisions 127621 via svnmerge from
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r127621 | bbryant | 2008-07-02 17:16:29 -0500 (Wed, 02 Jul 2008) | 1 line

Update transport= in sip so that the option is not broken from a recent commit.
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2008-07-08 14:43:30 +00:00
Brett Bryant
096d899b71 Merged revisions 127434 via svnmerge from
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r127434 | bbryant | 2008-07-02 12:27:36 -0500 (Wed, 02 Jul 2008) | 1 line

Fix to sip_parse_host so that it passes the correct information to sip_registry.
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2008-07-08 14:42:03 +00:00
Russell Bryant
05d1cfc350 Merged revisions 129006 via svnmerge from
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r129006 | russell | 2008-07-08 09:17:37 -0500 (Tue, 08 Jul 2008) | 9 lines

Update app_fax for better compatibility with spandsp 0.0.5.  Add a call to
t38_terminal_release, and make sure that the phase E handler gets called
with proper status.

(closes issue #13020)
Reported by: dimas
Patches:
      v1-appfax.patch uploaded by dimas (license 88)

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2008-07-08 14:18:05 +00:00
Olle Johansson
9e194be89a Merged revisions 128951 via svnmerge from
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r128951 | oej | 2008-07-08 12:02:12 +0200 (Tis, 08 Jul 2008) | 19 lines

Merged revisions 128950 via svnmerge from 
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r128950 | oej | 2008-07-08 11:52:21 +0200 (Tis, 08 Jul 2008) | 11 lines

Don't hangup the call if we can't resolve the Contact if there's a proxy
route set for the call.
----
This comment was added a while ago and today it hit me badly. 

/* OEJ: Possible issue that may need a check:
	If we have a proxy route between us and the device,
	should we care about resolving the contact
	or should we just send it?
*/

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2008-07-08 10:06:32 +00:00
Olle Johansson
91dd2f68bf Merged revisions 128927 via svnmerge from
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r128927 | oej | 2008-07-08 11:26:37 +0200 (Tis, 08 Jul 2008) | 15 lines

Merged revisions 128912 via svnmerge from 
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r128912 | oej | 2008-07-08 11:06:08 +0200 (Tis, 08 Jul 2008) | 7 lines

Fix issues where repeated messages where ignored, but retransmitted reliably instead of unreliably.
Reported by: johan
Patches: 
      12746.txt uploaded by oej (license 306)
Tested by: johan
(issue #12746)

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2008-07-08 09:28:56 +00:00
Olle Johansson
16664229b8 Blocking "sip show channelstats" from release candidate (as discussed on asterisk-dev).
Blocked revisions 128197 via svnmerge

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r128197 | oej | 2008-07-05 21:27:42 +0200 (Lör, 05 Jul 2008) | 5 lines

Add new SIP cli command "sip show channelstats" that displays some QoS data (if we have RTCP reports
and not use the p2p rtp bridge). I could not find a way to detect us using the p2p bridge, which
would be nice.


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2008-07-08 09:18:33 +00:00
Tilghman Lesher
2dc658b3b6 Merged revisions 128857 via svnmerge from
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r128857 | tilghman | 2008-07-07 19:02:11 -0500 (Mon, 07 Jul 2008) | 15 lines

Merged revisions 128856 via svnmerge from 
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r128856 | tilghman | 2008-07-07 19:01:30 -0500 (Mon, 07 Jul 2008) | 7 lines

Check for non-NULL before stripping characters.
(closes issue #12954)
 Reported by: bfsworks
 Patches: 
       20080701__bug12954.diff.txt uploaded by Corydon76 (license 14)
 Tested by: deti

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Tilghman Lesher
bf9782a18c Merged revisions 128830 via svnmerge from
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r128830 | tilghman | 2008-07-07 18:25:39 -0500 (Mon, 07 Jul 2008) | 10 lines

Merged revisions 128812 via svnmerge from 
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r128812 | tilghman | 2008-07-07 18:21:52 -0500 (Mon, 07 Jul 2008) | 2 lines

Stop using deprecated method, as requested by Kevin.

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2008-07-07 23:47:24 +00:00
Russell Bryant
0b2f1a8975 Merged revisions 128796 via svnmerge from
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r128796 | russell | 2008-07-07 17:42:30 -0500 (Mon, 07 Jul 2008) | 16 lines

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r128795 | russell | 2008-07-07 17:41:48 -0500 (Mon, 07 Jul 2008) | 8 lines

Fix handling of when a pvt disappears.  Properly return the pvt locked
and don't hold the pvt lock while destroying the ast_channel.

(closes issue #13014)
Reported by: jpgrayson
Patches:
      chan_iax2_ast_iax2_new2.patch uploaded by jpgrayson (license 492)

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2008-07-07 22:44:49 +00:00
Sean Bright
316061549c Merged revisions 128738 via svnmerge from
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r128738 | seanbright | 2008-07-07 16:50:29 -0400 (Mon, 07 Jul 2008) | 17 lines

Merged revisions 128737 via svnmerge from 
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r128737 | seanbright | 2008-07-07 16:47:56 -0400 (Mon, 07 Jul 2008) | 9 lines

Remove spurious trailing whitespace from log messages and fix a spelling error
in a log message.

(closes issue #13017)
Reported by: jpgrayson
Patches:
      chan_iax2_space_after_newline.patch uploaded by jpgrayson (license 492)
      chan_iax2_spelling.patch uploaded by jpgrayson (license 492)

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Mark Michelson
b372ba5e50 Merged revisions 128733 via svnmerge from
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r128733 | mmichelson | 2008-07-07 15:30:46 -0500 (Mon, 07 Jul 2008) | 3 lines

Crap


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2008-07-07 20:31:07 +00:00
Mark Michelson
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r128731 | mmichelson | 2008-07-07 15:28:33 -0500 (Mon, 07 Jul 2008) | 7 lines

If imapfolder=foo were set in voicemail.conf, then when calling VoiceMailMain,
app_voicemail would attempt to play a file called vm-foo instead of playing
vm-INBOX to play the "new" sound file. This commit fixes that issue.

This may fix one of the problems reported in issue #12987


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Mark Michelson
1c094419c4 Blocked revisions 128690 via svnmerge
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r128690 | mmichelson | 2008-07-07 12:34:06 -0500 (Mon, 07 Jul 2008) | 5 lines

Get app_voicemail compiling when IMAP storage is used.

Brought up by reporter on issue #12987


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2008-07-07 17:36:58 +00:00
Mark Michelson
959cf9cdee Merged revisions 128640 via svnmerge from
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r128640 | mmichelson | 2008-07-07 12:09:11 -0500 (Mon, 07 Jul 2008) | 18 lines

Merged revisions 128639 via svnmerge from 
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r128639 | mmichelson | 2008-07-07 12:02:28 -0500 (Mon, 07 Jul 2008) | 10 lines

By using the iaxdynamicthreadcount to identify a thread, it was possible
for thread identifiers to be duplicated. By using a globally-unique monotonically-
increasing integer, this is now avoided.

(closes issue #13009)
Reported by: jpgrayson
Patches:
      chan_iax2_dyn_threadnum.patch uploaded by jpgrayson (license 492)


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Mark Michelson
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r128599 | mmichelson | 2008-07-07 09:35:27 -0500 (Mon, 07 Jul 2008) | 6 lines

Update a few instances of "extensions reload" to "dialplan reload"
in the documentation.

Patch provided by caio1982 (license 22)


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2008-07-07 14:36:30 +00:00
Russell Bryant
8c095156bd Blocked revisions 128564 via svnmerge
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r128564 | russell | 2008-07-07 06:53:52 -0500 (Mon, 07 Jul 2008) | 3 lines

As pointed out on the -dev list, actually use the result of find_peer() so that
a peer reference is not leaked.

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2008-07-07 11:55:08 +00:00
Olle Johansson
175dc59c60 Blocked revisions 128525 via svnmerge
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r128525 | oej | 2008-07-06 22:19:04 +0200 (Sön, 06 Jul 2008) | 7 lines

- Adding alias "udpbindaddr" for the UDP port to comply with "tcpbindaddr" and "tlsbindaddr".
  Note: I don't think we can start properly without UDP port open, that needs to be tested.

- Removing "bindport" from configuration example, not needed to mention this any more

I suggest we deprecate "bindaddr" and "bindport" in trunk (for 1.6.1)

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Olle Johansson
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r128524 | oej | 2008-07-06 22:11:37 +0200 (Sön, 06 Jul 2008) | 5 lines

- Fixing issues with "sip show settings"
- Adding IP address for TCP and/or TLS too if auto-domain is enabled and
  binding to a different IP address
- Fixing documentation in sip.conf.sample

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Olle Johansson
207b9c07e4 Merged revisions 128491 via svnmerge from
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r128491 | oej | 2008-07-06 21:14:06 +0200 (Sön, 06 Jul 2008) | 3 lines

- Remove unused variable "expiry"
- Set global_outboundproxy.force at reload.

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Olle Johansson
c6e18beaae Blocked revisions 128485 via svnmerge
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r128485 | oej | 2008-07-06 20:25:39 +0200 (Sön, 06 Jul 2008) | 2 lines

More doxygen comments.

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Olle Johansson
8353163bdd Blocked revisions 128455 via svnmerge
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r128455 | oej | 2008-07-06 19:58:06 +0200 (Sön, 06 Jul 2008) | 4 lines

- Formatting changes
- Doxygen changes
- Replacing a doxygen description that was copied from another function

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2008-07-06 18:26:01 +00:00
Olle Johansson
ca9b0f89f7 Blocked revisions 128454 via svnmerge
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r128454 | oej | 2008-07-06 19:45:43 +0200 (Sön, 06 Jul 2008) | 2 lines

Adding note about incorrect manager registration...

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2008-07-06 17:58:40 +00:00
Olle Johansson
ce784b9856 The following patch with references to t140red removed, since it only exists
in trunk.

Merged revisions 128417 via svnmerge from 
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r128417 | oej | 2008-07-06 12:13:45 +0200 (Sön, 06 Jul 2008) | 3 lines

Adding documentation on the T.140 support in Asterisk. This is a function that we're
the reference implementation on now. :-)

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Olle Johansson
532284317e Blocked revisions 128381 via svnmerge
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r128381 | oej | 2008-07-06 10:43:08 +0200 (Sön, 06 Jul 2008) | 3 lines

Remove comments that doesn't make sense. The deprecation of type=user will come at a later stage,
as indicated by previous commit message

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Olle Johansson
8e98476573 Blocked revisions 128379 via svnmerge
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r128379 | oej | 2008-07-06 10:32:11 +0200 (Sön, 06 Jul 2008) | 4 lines

Fix severe problem with my previous commit of "kill-the-user". Russell saw a problem with this
code, but not the correct problem. Thanks, anyway! ;-)


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Olle Johansson
d18409c969 Blocked revisions 128378 via svnmerge
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r128378 | oej | 2008-07-06 10:28:58 +0200 (Sön, 06 Jul 2008) | 3 lines

Changing name of global api call to ast_* 
My mistake, pointed out by Russell.

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Olle Johansson
037268e890 Blocking patch that is not needed
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2008-07-06 08:20:09 +00:00
Olle Johansson
db353de824 Merged revisions 128343 via svnmerge from
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r128343 | oej | 2008-07-06 10:10:27 +0200 (Sön, 06 Jul 2008) | 2 lines

Removing the CLI dumpdb command (see asterisk-dev discussion and decision)

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2008-07-06 08:17:34 +00:00
Olle Johansson
a578766ab0 The todo's are not needed in 1.6 release candidate...
Blocked revisions 128294 via svnmerge

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r128294 | oej | 2008-07-06 00:07:37 +0200 (Sön, 06 Jul 2008) | 2 lines

Adding a few reminders

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2008-07-05 22:09:52 +00:00
Olle Johansson
10aaad920e Merged revisions 128290 via svnmerge from
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r128290 | oej | 2008-07-05 23:55:57 +0200 (Lör, 05 Jul 2008) | 5 lines

Adding doxygen comments to missing parts, moving some #define

...trying to get my head around the thoughts behind the TCP/TLS stuff
and figure out what needs to be done to make it useful...

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2008-07-05 22:04:19 +00:00
Olle Johansson
73c1f29476 Merged revisions 128287 via svnmerge from
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r128287 | oej | 2008-07-05 23:37:57 +0200 (Lör, 05 Jul 2008) | 3 lines

Adding TCP and TLS to "sip show settings". 
TLS needs to have one configuration per configured domain at some point.

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2008-07-05 21:56:33 +00:00
Olle Johansson
9b389f95d3 Blocking changes in trunk.
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2008-07-05 21:38:53 +00:00
Michiel van Baak
1504717261 Blocked revisions 128254 via svnmerge
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r128254 | mvanbaak | 2008-07-05 23:20:09 +0200 (Sat, 05 Jul 2008) | 2 lines

fix compiling of chan_sip.c

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Olle Johansson
4d29402dfa Keep the "sip-user" structure in 1.6.0, while testing new funky stuff in trunk.
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2008-07-05 21:02:11 +00:00
Olle Johansson
8f101e3a7c Blocking the AGi changes from 1.6.0. Let's test them for a while in trunk before a release.
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2008-07-05 20:55:50 +00:00
Olle Johansson
d011616e6e Merged revisions 128237 via svnmerge from
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r128237 | oej | 2008-07-05 22:39:54 +0200 (Lör, 05 Jul 2008) | 2 lines

Make TCP disabled by default (it's considered experimental)

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2008-07-05 20:49:40 +00:00
Olle Johansson
cd4363d649 Merged revisions 128236 via svnmerge from
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r128236 | oej | 2008-07-05 22:37:53 +0200 (Lör, 05 Jul 2008) | 2 lines

Reformatting the config sample

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2008-07-05 20:43:50 +00:00
Tilghman Lesher
02cad6c401 Merged revisions 128160 via svnmerge from
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r128160 | tilghman | 2008-07-05 10:17:51 -0500 (Sat, 05 Jul 2008) | 7 lines

LDAP schema updates
(closes issue #12860)
 Reported by: flyn
 Patches: 
       asterisk.ldif uploaded by suretec (license 70)
       asterisk.schema uploaded by suretec (license 70)

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2008-07-05 15:19:08 +00:00
Matthew Fredrickson
5ffa99cdce Merged revisions 128125 via svnmerge from
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r128125 | mattf | 2008-07-04 22:39:07 -0500 (Fri, 04 Jul 2008) | 1 line

It would help if we actually parsed the ss7_explicitacm option in the config file...
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Matthew Fredrickson
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r128122 | mattf | 2008-07-04 22:26:42 -0500 (Fri, 04 Jul 2008) | 1 line

Add option to wait to be able to explicitly send ACM via the Proceeding() application in the dialplan.  Also minor documentation update explaining how to setup multiple signalling links within a linkset
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Tilghman Lesher
6d15b3a980 Blocked revisions 128030 via svnmerge
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r128030 | tilghman | 2008-07-04 11:11:32 -0500 (Fri, 04 Jul 2008) | 9 lines

Blocked revisions 128029 via svnmerge

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r128029 | tilghman | 2008-07-04 11:11:03 -0500 (Fri, 04 Jul 2008) | 2 lines

Move the free down one

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Tilghman Lesher
631361ec04 Merged revisions 128027 via svnmerge from
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r128027 | tilghman | 2008-07-04 11:06:34 -0500 (Fri, 04 Jul 2008) | 16 lines

Merged revisions 127973 via svnmerge from 
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r127973 | tilghman | 2008-07-03 22:30:30 -0500 (Thu, 03 Jul 2008) | 8 lines

Fix the 'dialplan remove extension' logic, so that it a) works with cidmatch,
and b) completes contexts correctly when the extension is ambiguous.
(closes issue #12980)
 Reported by: licedey
 Patches: 
       20080703__bug12980.diff.txt uploaded by Corydon76 (license 14)
 Tested by: Corydon76

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Kevin P. Fleming
a2a2d2d0a0 Merged revisions 127903 via svnmerge from
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r127903 | kpfleming | 2008-07-03 17:23:04 -0500 (Thu, 03 Jul 2008) | 20 lines

Merged revisions 127892,127895 via svnmerge from 
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r127892 | kpfleming | 2008-07-03 17:18:38 -0500 (Thu, 03 Jul 2008) | 6 lines

a couple of small Solaris-related fixes

(closes issue #11885)
Reported by: snuffy, asgaroth


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r127895 | kpfleming | 2008-07-03 17:20:16 -0500 (Thu, 03 Jul 2008) | 3 lines

remove this, it has been moved to the main Makefile


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Mark Michelson
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r127857 | mmichelson | 2008-07-03 15:59:51 -0500 (Thu, 03 Jul 2008) | 3 lines

Make change proposed by andrew53 on bugtracker


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Mark Michelson
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r127856 | mmichelson | 2008-07-03 15:37:21 -0500 (Thu, 03 Jul 2008) | 4 lines

Thanks to a suggestion from seanbright, print a warning if the attachment
of the whisper or barge audiohooks fails.


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Mark Michelson
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r127852 | mmichelson | 2008-07-03 15:23:57 -0500 (Thu, 03 Jul 2008) | 3 lines

Fix build


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Mark Michelson
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r127831 | mmichelson | 2008-07-03 15:19:10 -0500 (Thu, 03 Jul 2008) | 6 lines

Fix a crash when attempting to spy on an unbridged channel.

(closes issue #12986)
Reported by: andrew53


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Steve Murphy
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r127793 | murf | 2008-07-03 11:16:44 -0600 (Thu, 03 Jul 2008) | 38 lines

Merged revisions 127663 via svnmerge from 
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r127663 | murf | 2008-07-02 18:16:25 -0600 (Wed, 02 Jul 2008) | 30 lines

The CDRfix4/5/6 omnibus cdr fixes.

(closes issue #10927)
Reported by: murf
Tested by: murf, deeperror

(closes issue #12907)
Reported by: falves11
Tested by: murf, falves11


(closes issue #11849)
Reported by: greyvoip

As to 11849, I think these changes fix the core problems 
brought up in that bug, but perhaps not the more global
problems created by the limitations of CDR's themselves
not being oriented around transfers.

Reopen if necc, but bug reports are not the best
medium for enhancement discussions. We need to start
a second-generation CDR standardization effort to cover
transfers.

(closes issue #11093)
Reported by: rossbeer
Tested by: greyvoip, murf



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2008-07-03 19:12:14 +00:00
Olle Johansson
5098eedd6d Merged revisions 127791 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r127791 | oej | 2008-07-03 18:48:23 +0200 (Tor, 03 Jul 2008) | 5 lines

Make sure we stop session timers as soon as we start hanging up an active call.

May fix issue 12919.


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2008-07-03 16:50:43 +00:00
Olle Johansson
434c1b619e Merged revisions 127779 via svnmerge from
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r127779 | oej | 2008-07-03 18:25:59 +0200 (Tor, 03 Jul 2008) | 4 lines

Revert some logic for session timers. We do send in-dialog requests that should not have session-timer
require headers, like MESSAGE and REFER. So in the future, only add them on requests and responses
that are related to INVITEs and re-INVITEs.

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2008-07-03 16:33:05 +00:00
Kevin P. Fleming
eeda7c996b Merged revisions 127767 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r127767 | kpfleming | 2008-07-03 11:22:02 -0500 (Thu, 03 Jul 2008) | 2 lines

some minor fixes found while working on issue #12911 (and block the rev from 1.4 since the equivalent is already here)

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2008-07-03 16:24:57 +00:00
Mark Michelson
1eaf84a6e3 Blocked revisions 127720 via svnmerge
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r127720 | mmichelson | 2008-07-03 09:34:25 -0500 (Thu, 03 Jul 2008) | 7 lines

Added a new option, "timeoutpriority" to queues.conf. A detailed
explanation of the change may be found in configs/queues.conf.sample

(closes issue #12690)
Reported by: atis


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2008-07-03 14:34:56 +00:00
Tilghman Lesher
5256257cd8 Blocked revisions 127622 via svnmerge
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r127622 | tilghman | 2008-07-02 17:17:37 -0500 (Wed, 02 Jul 2008) | 2 lines

Oops

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2008-07-02 22:18:12 +00:00
Mark Michelson
cc212c962e Merged revisions 127566 via svnmerge from
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r127566 | mmichelson | 2008-07-02 16:09:18 -0500 (Wed, 02 Jul 2008) | 4 lines

Add a janitor project to use ARRAY_LEN instead of in-line
sizeof() and division.


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2008-07-02 21:10:00 +00:00
Tilghman Lesher
0a06013cd4 Blocked revisions 127564 via svnmerge
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r127564 | tilghman | 2008-07-02 15:52:27 -0500 (Wed, 02 Jul 2008) | 5 lines

Fix some crashlike bugs because flag could be NULL in play_record_review().
(Closes issue #12892)
Reported by: jaroth
Patch originally by jaroth, fixed by me.

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2008-07-02 20:53:11 +00:00
Mark Michelson
2f2fa0b0e9 Merged revisions 127562 via svnmerge from
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r127562 | mmichelson | 2008-07-02 15:49:08 -0500 (Wed, 02 Jul 2008) | 11 lines

Merged revisions 127560 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r127560 | mmichelson | 2008-07-02 15:47:38 -0500 (Wed, 02 Jul 2008) | 3 lines

Fix thread-safety of some of the pbx_builtin_getvar_helper calls


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2008-07-02 20:49:36 +00:00
Mark Michelson
b6f531de7a Blocked revisions 127558 via svnmerge
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r127558 | mmichelson | 2008-07-02 15:43:55 -0500 (Wed, 02 Jul 2008) | 7 lines

The ackcall and endcall options in agents.conf now have supplemental options
acceptdtmf and enddtmf. These allow for the DTMF pressed to be configurable
instead of being hardcoded to '#' and '*'.

(AST-86)


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2008-07-02 20:44:43 +00:00
Tilghman Lesher
37e6a534d2 Blocked revisions 127502 via svnmerge
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r127502 | tilghman | 2008-07-02 14:47:50 -0500 (Wed, 02 Jul 2008) | 21 lines

Blocked revisions 127501 via svnmerge

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r127501 | tilghman | 2008-07-02 14:47:15 -0500 (Wed, 02 Jul 2008) | 14 lines

Merged revisions 127466 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

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r127466 | tilghman | 2008-07-02 13:31:11 -0500 (Wed, 02 Jul 2008) | 6 lines

Solaris fix
(closes issue #12949)
 Reported by: snuffy
 Patches: 
       bug_12949.diff uploaded by snuffy (license 35)

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2008-07-02 19:48:20 +00:00
Tilghman Lesher
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r127466 | tilghman | 2008-07-02 13:31:11 -0500 (Wed, 02 Jul 2008) | 6 lines

Solaris fix
(closes issue #12949)
 Reported by: snuffy
 Patches: 
       bug_12949.diff uploaded by snuffy (license 35)

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2008-07-02 18:31:58 +00:00
Russell Bryant
59b415d1ee Blocked revisions 127401 via svnmerge
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r127401 | russell | 2008-07-02 09:50:45 -0500 (Wed, 02 Jul 2008) | 8 lines

Fix a bunch of places where \arg was used instead of \param.  Using \arg
to document arguments seems logical, and does work, but is not the best
thing to use.

\arg in doxygen is simply for creating non-nested unordered lists.  \param is
the correct tag to use to document function parameters, and will come out
better in the generated documentation.

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2008-07-02 14:51:20 +00:00
Sean Bright
be28cb0858 Merged revisions 127398 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r127398 | seanbright | 2008-07-02 10:30:09 -0400 (Wed, 02 Jul 2008) | 1 line

Fix a bug I noticed while doing the previous merge
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2008-07-02 14:30:38 +00:00
Sean Bright
bbf18851a6 Merged revisions 126226,126513 via svnmerge from
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r126226 | seanbright | 2008-06-28 17:28:16 -0400 (Sat, 28 Jun 2008) | 8 lines

Merge in changes from my cdr-tds-conversion branch.  This changes the internal
implementation from using the volatile libtds, to using the db-lib front end.
The unintended side effect of this is that we support (at least) versions 0.62
through 0.82 of the FreeTDS distribution without any #ifdef ugliness.

(closes issue #12844)
Reported by: jcollie

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r126513 | seanbright | 2008-06-30 07:57:42 -0400 (Mon, 30 Jun 2008) | 4 lines

Cast a few more strings to char *, so that we can compile cleanly against
FreeTDS 0.60.  Update the docs to reflect that we can now compile and run
against all modern releases of FreeTDS (0.60 through 0.82)

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2008-07-02 14:29:18 +00:00
Sean Bright
1231f9e9c8 Unblock some revisions so I can merge the cdr_tds changes from trunk
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@127396 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-02 14:12:23 +00:00
Russell Bryant
46daf77516 Merged revisions 127363 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r127363 | russell | 2008-07-02 07:08:33 -0500 (Wed, 02 Jul 2008) | 13 lines

Add a locking section to the coding guidelines document.

This section covers some locking fundamentals, as well as some information on
locking as it is used in Asterisk.  It describes some of the ways that are used
and could be used to achieve deadlock avoidance.  It also demonstrates the
unfortunate conclusion that with the use of recursive locks, none of the
constructs in use today are failsafe from deadlocks.  Finally, it makes some
recommendations for new code being written.  As proper locking strategies is a
complex subject, this section still has room for expansion and improvement.

This is a result of collaboration between Luigi Rizzo and myself on the 
asterisk-dev mailing list.

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2008-07-02 12:09:08 +00:00
Tilghman Lesher
f53f5f1882 Merged revisions 127297 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r127297 | tilghman | 2008-07-01 21:48:43 -0500 (Tue, 01 Jul 2008) | 12 lines

Change the global timer B to be dependent on the value of the T1 timer, as
recommended in RFC 3261, instead of being hardcoded to 32 seconds.  This is
important for LANs, as it allows autocongestion to occur much more quickly, if
desired by the local PBX administrator.  It also corrects a bug: if the T1
timer was increased beyond 500ms, then timer B would have been set at a much
lower value than recommended.
(closes issue #12544)
 Reported by: kactus
 Patches: 
       20080616__bug12544.diff.txt uploaded by Corydon76 (license 14)
 Tested by: kactus

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2008-07-02 02:49:57 +00:00
Mark Michelson
8d280aa2ae Merged revisions 127245 via svnmerge from
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r127245 | mmichelson | 2008-07-01 18:38:12 -0500 (Tue, 01 Jul 2008) | 13 lines

Merged revisions 127244 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r127244 | mmichelson | 2008-07-01 18:36:40 -0500 (Tue, 01 Jul 2008) | 5 lines

Add error message to failed open(2) calls inside the copy() function of
app_voicemail. This idea came as part of my work in helping to resolve
issue #12764.


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2008-07-01 23:39:42 +00:00
Tilghman Lesher
465dcf9aae Blocked revisions 127169 via svnmerge
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r127169 | tilghman | 2008-07-01 16:21:26 -0500 (Tue, 01 Jul 2008) | 6 lines

Add AMI events for start/stop of MOH
(closes issue #12909)
 Reported by: chris-mac
 Patches: 
       res_musiconhold-event.patch uploaded by chris-mac (license 506)

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2008-07-01 21:22:02 +00:00
Brett Bryant
ecbd67d74e Merged revisions 127154 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r127154 | bbryant | 2008-07-01 16:03:52 -0500 (Tue, 01 Jul 2008) | 2 lines

Add a configuration option so the global outboundproxy can use tcptls without it being defined by each sip user.

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2008-07-01 21:19:59 +00:00
Mark Michelson
d3caa53777 Merged revisions 127157 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r127157 | mmichelson | 2008-07-01 16:16:00 -0500 (Tue, 01 Jul 2008) | 8 lines

Place the delay in __ast_answer prior to the channel-specific answer
callback. This change differs from commit 127113 in that now the 
channel is not set to AST_STATE_UP until after the answer callback.

(closes issue #12924)
Reported by: snyfer


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2008-07-01 21:16:33 +00:00
Mark Michelson
647d2c800a Merging Revision 127113 from trunk
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2008-07-01 21:13:32 +00:00
Jason Parker
0020aa51f0 Merged revisions 127152 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r127152 | qwell | 2008-07-01 15:51:43 -0500 (Tue, 01 Jul 2008) | 7 lines

Fix a typo that caused this asterisk.conf to not get correctly generated.

(closes issue #12966)
Reported by: ibc
Patches:
      12966.patch uploaded by bkruse (license 132)

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2008-07-01 20:52:09 +00:00
Tilghman Lesher
0919561e7c Merged revisions 127143 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r127143 | tilghman | 2008-07-01 15:28:54 -0500 (Tue, 01 Jul 2008) | 10 lines

Merged revisions 127133 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r127133 | tilghman | 2008-07-01 15:25:37 -0500 (Tue, 01 Jul 2008) | 2 lines

Disable the old, slow search for matching callno in chan_iax2 (but allow it to be reenabled for debugging)

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2008-07-01 20:29:53 +00:00
Tilghman Lesher
3af72816ca Blocked revisions 127089 via svnmerge
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r127089 | tilghman | 2008-07-01 14:24:29 -0500 (Tue, 01 Jul 2008) | 9 lines

Blocked revisions 127069 via svnmerge

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r127069 | tilghman | 2008-07-01 14:05:28 -0500 (Tue, 01 Jul 2008) | 2 lines

Oops

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Tilghman Lesher
975815a708 Merged revisions 127074 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r127074 | tilghman | 2008-07-01 14:20:25 -0500 (Tue, 01 Jul 2008) | 16 lines

Merged revisions 127068 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r127068 | tilghman | 2008-07-01 13:52:53 -0500 (Tue, 01 Jul 2008) | 8 lines

Change around how we schedule pings and lagrqs, and fix a reason why the
jobs were not getting properly cancelled.
(closes issue #12903)
 Reported by: stevedavies
 Patches: 
       20080620__bug12903__2.diff.txt uploaded by Corydon76 (license 14)
 Tested by: stevedavies

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2008-07-01 19:23:31 +00:00
Kevin P. Fleming
a52853359d Blocked revisions 127017 via svnmerge
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r127017 | kpfleming | 2008-07-01 12:22:47 -0500 (Tue, 01 Jul 2008) | 2 lines

make the AIS checking a little more generic, and have a more useful configure script command line option for OpenAIS

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2008-07-01 17:23:14 +00:00
Tilghman Lesher
a160a20fb4 Merged revisions 127000 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r127000 | tilghman | 2008-07-01 11:52:29 -0500 (Tue, 01 Jul 2008) | 10 lines

Merged revisions 126999 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r126999 | tilghman | 2008-07-01 11:50:46 -0500 (Tue, 01 Jul 2008) | 2 lines

Suppress annoying warning by finding the remaining cases where the callno is not in the hash.

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2008-07-01 16:53:21 +00:00
Kevin P. Fleming
f5b586edb0 Blocked revisions 126960 via svnmerge
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r126960 | kpfleming | 2008-07-01 11:16:36 -0500 (Tue, 01 Jul 2008) | 2 lines

another minor ast_channel memory size decrease... for nearly all channels, 'dialcontext' is only going to be set once during the channel's lifetime, so make it a string field instead of a char array

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2008-07-01 16:17:38 +00:00
Olle Johansson
221f06f104 Merged revisions 126903 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r126903 | oej | 2008-07-01 17:03:59 +0200 (Tis, 01 Jul 2008) | 15 lines

Merged revisions 126902 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r126902 | oej | 2008-07-01 16:59:31 +0200 (Tis, 01 Jul 2008) | 7 lines

Use domain part of SIP uri in register= configuration as fromdomain.

Reported by: one47
Patches: 
      sip-reg-fromdom2.dpatch uploaded by one47 (license 23)
(closes issue #12474)

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2008-07-01 15:05:19 +00:00
Olle Johansson
927a710cea Merged revisions 126900 via svnmerge from
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r126900 | oej | 2008-07-01 16:32:15 +0200 (Tis, 01 Jul 2008) | 16 lines

Merged revisions 126899 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r126899 | oej | 2008-07-01 16:27:33 +0200 (Tis, 01 Jul 2008) | 8 lines

Handle escaped URI's in call pickups. Patch by oej and IgorG.

Reported by: IgorG
Patches: 
      bug12299-11062-v2.patch uploaded by IgorG (license 20)
Tested by: IgorG, oej
(closes issue #12299)

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2008-07-01 14:34:34 +00:00
Olle Johansson
7204cf07f2 Merged revisions 126845 via svnmerge from
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r126845 | oej | 2008-07-01 14:54:57 +0200 (Tis, 01 Jul 2008) | 14 lines


Merged revisions 126844 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r126844 | oej | 2008-07-01 14:53:01 +0200 (Tis, 01 Jul 2008) | 5 lines

Clear up documentation on "domain=" setting in sip.conf

Reported by: davidw
(closes issue #12413)

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2008-07-01 13:07:47 +00:00
Olle Johansson
b6318501ef Merged revisions 126790 via svnmerge from
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r126790 | oej | 2008-07-01 13:58:17 +0200 (Tis, 01 Jul 2008) | 14 lines

Merged revisions 126789 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r126789 | oej | 2008-07-01 13:51:38 +0200 (Tis, 01 Jul 2008) | 6 lines

Report 200 OK to all in-dialog OPTIONs requests (to confirm that the dialog
exist). Don't bother checking the request URI.

(closes issue #11264)
Reported by: ibc

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2008-07-01 12:02:41 +00:00
Olle Johansson
75ee025252 Merged revisions 126755 via svnmerge from
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r126755 | oej | 2008-07-01 11:51:22 +0200 (Tis, 01 Jul 2008) | 15 lines

Merged revisions 126735 via svnmerge from 
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r126735 | oej | 2008-07-01 09:49:15 +0200 (Tis, 01 Jul 2008) | 7 lines

Fix bad XML for hold notification.
Reported by: gowen72
Patches: 
      hold.patch uploaded by gowen72 (license 432)
(closes issue #12942)


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2008-07-01 09:53:12 +00:00
Jeff Peeler
6bc1dc43ed Merged revisions 126675 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r126675 | jpeeler | 2008-06-30 17:34:08 -0500 (Mon, 30 Jun 2008) | 1 line

rename zapata.conf.sample to chan_dahdi.conf.sample
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2008-06-30 22:34:37 +00:00
Matthew Fredrickson
29c39f4361 Merged revisions 126637 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r126637 | mattf | 2008-06-30 15:25:46 -0500 (Mon, 30 Jun 2008) | 1 line

Add support to see MTP2 down events when the link layer drops in SS7
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2008-06-30 20:32:37 +00:00
Russell Bryant
38999877fb Merged revisions 126574 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r126574 | russell | 2008-06-30 11:07:25 -0500 (Mon, 30 Jun 2008) | 18 lines

Merged revisions 126573 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r126573 | russell | 2008-06-30 11:05:08 -0500 (Mon, 30 Jun 2008) | 10 lines

Fix a typo in the non-DEBUG_THREADS version of the recently added DEADLOCK_AVOIDANCE()
macro.  This caused the lock to not actually be released, and as a result, not
avoid deadlocks at all.  This resolves the issues reported in the last while about
Asterisk locking up all over the place (and most commonly, in chan_iax2).

(closes issue #12927)
(closes issue #12940)
(closes issue #12925)
(potentially closes others ...)

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2008-06-30 16:09:52 +00:00
Olle Johansson
48076d5903 Merged revisions 126517 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r126517 | oej | 2008-06-30 15:03:53 +0200 (MÃ¥n, 30 Jun 2008) | 20 lines

The following patch with some changes for trunk...

Merged revisions 126516 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r126516 | oej | 2008-06-30 14:50:55 +0200 (MÃ¥n, 30 Jun 2008) | 10 lines

Send all responses to an INVITE reliably, so that we retransmit if we don't get an ACK and
also fail if we don't get the very same precious ACK. Based on patch by tsearle, with
my own additions.

(closes issue #12951)

Reported by: tsearle
Patches: 
      busy_retransmit.patch uploaded by tsearle (license 373)

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2008-06-30 13:07:19 +00:00
Sean Bright
9863a997c0 Blocked revisions 126513 via svnmerge
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r126513 | seanbright | 2008-06-30 07:57:42 -0400 (Mon, 30 Jun 2008) | 4 lines

Cast a few more strings to char *, so that we can compile cleanly against
FreeTDS 0.60.  Update the docs to reflect that we can now compile and run
against all modern releases of FreeTDS (0.60 through 0.82)

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2008-06-30 11:58:04 +00:00
Kevin P. Fleming
789d5b96c3 finish converting this module
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2008-06-29 17:02:40 +00:00
Kevin P. Fleming
b8cfb78cc0 Merged revisions 126356 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r126356 | kpfleming | 2008-06-29 09:19:29 -0700 (Sun, 29 Jun 2008) | 9 lines

various minor fixes created while i worked on getting *every* Asterisk module to build on laptop in dev mode:

remove weird pre-setting of LUA paths; they are not necessary; also use the proper path for including the files in pbx_lua.c

make the compiler shut up about some ignored function results in pbx_gtkconsole; this module is badly coded anyway


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2008-06-29 17:02:25 +00:00
Kevin P. Fleming
b148e15762 don't know how this got missed in the DAHDI conversion of this branch
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2008-06-29 17:00:56 +00:00
Sean Bright
1a797f47ea Blocked revisions 126319 via svnmerge
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r126319 | seanbright | 2008-06-29 09:20:01 -0400 (Sun, 29 Jun 2008) | 2 lines

This was bogus, need to find a better way.

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2008-06-29 13:20:31 +00:00
Sean Bright
e3dfce8119 Blocked revisions 126312 via svnmerge
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r126312 | seanbright | 2008-06-29 09:11:55 -0400 (Sun, 29 Jun 2008) | 4 lines

While we're at it, escape all the columns in our TDS queries as well.  Double
quotes seems to be more standard than square brackets (Sybase and SQL Server
both support them).

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2008-06-29 13:13:20 +00:00
Sean Bright
18af01717e Merged revisions 126274 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r126274 | seanbright | 2008-06-29 08:06:46 -0400 (Sun, 29 Jun 2008) | 6 lines

Quote column names when inserting CDRs into postgres to avoid conflicts
with reserved words.

(closes issue #12947)
Reported by: panolex

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2008-06-29 12:07:31 +00:00
Sean Bright
8b7210c2ee Blocked revisions 126226 via svnmerge
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r126226 | seanbright | 2008-06-28 17:28:16 -0400 (Sat, 28 Jun 2008) | 8 lines

Merge in changes from my cdr-tds-conversion branch.  This changes the internal
implementation from using the volatile libtds, to using the db-lib front end.
The unintended side effect of this is that we support (at least) versions 0.62
through 0.82 of the FreeTDS distribution without any #ifdef ugliness.

(closes issue #12844)
Reported by: jcollie

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2008-06-28 21:29:45 +00:00
Kevin P. Fleming
09022d2add Blocked revisions 126187 via svnmerge
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r126187 | kpfleming | 2008-06-28 08:54:04 -0700 (Sat, 28 Jun 2008) | 3 lines

yay for airplane ride optimizations... sort the fields in ast_channel by alignment requirements, saving 36 bytes per instance on a 64-bit platform


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2008-06-28 15:58:11 +00:00
Kevin P. Fleming
9c4c17ae18 update this branch to use the trunk goodness version of menuselect
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@126155 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-28 14:58:07 +00:00
Sean Bright
cf780a9462 Blocked revisions 126115 via svnmerge
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r126115 | seanbright | 2008-06-27 19:29:56 -0400 (Fri, 27 Jun 2008) | 1 line

Pretty up the 'cdr show status' output.
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2008-06-27 23:31:22 +00:00
Tilghman Lesher
b8ee51e4ba Blocked revisions 114174,114298,115258,115518,117524,117812,118059,119077,120063,120372,120672,120732,121993,122315,122434,122616,122664,124744,124909,125894 via svnmerge
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r114174 | qwell | 2008-04-16 12:31:02 -0500 (Wed, 16 Apr 2008) | 14 lines

Blocked revisions 114173 via svnmerge

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r114173 | qwell | 2008-04-16 12:30:09 -0500 (Wed, 16 Apr 2008) | 7 lines

Fix "fallthrough" behavior here, so config options in a previously configured user don't override settings in general.

(closes issue #12458)
Reported by: tzafrir
Patches:
      chanzap_users_sections.diff uploaded by tzafrir (license 46)

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r114298 | tilghman | 2008-04-19 08:53:38 -0500 (Sat, 19 Apr 2008) | 11 lines

Blocked revisions 114297 via svnmerge

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r114297 | tilghman | 2008-04-19 08:49:50 -0500 (Sat, 19 Apr 2008) | 4 lines

MOH usage information needs a terminating newline, or else
"asterisk -rx 'help moh reload'" will hang.  Reported via
-dev list, fixed by me.

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r115258 | bbryant | 2008-05-02 15:26:00 -0500 (Fri, 02 May 2008) | 9 lines

Blocked revisions 115257 via svnmerge

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r115257 | bbryant | 2008-05-02 15:25:42 -0500 (Fri, 02 May 2008) | 2 lines

Add new "pri show version" command to show the libpri version for support reasons.

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r115518 | russell | 2008-05-07 13:17:43 -0500 (Wed, 07 May 2008) | 12 lines

Blocked revisions 115517 via svnmerge

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r115517 | russell | 2008-05-07 13:17:19 -0500 (Wed, 07 May 2008) | 5 lines

Track peer references when stored in the sip_pvt struct as the peer related to
a qualify ping or a subscription.  This fixes some realtime related crashes.
(closes issue #12588)
(closes issue #12555)

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r117524 | tilghman | 2008-05-21 13:45:26 -0500 (Wed, 21 May 2008) | 9 lines

Blocked revisions 117523 via svnmerge

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r117523 | tilghman | 2008-05-21 13:44:53 -0500 (Wed, 21 May 2008) | 2 lines

Revert accidental commit of the last change

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r117812 | tilghman | 2008-05-22 11:50:32 -0500 (Thu, 22 May 2008) | 13 lines

Blocked revisions 117809 via svnmerge

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r117809 | tilghman | 2008-05-22 11:47:03 -0500 (Thu, 22 May 2008) | 6 lines

Take into account the length of delimiters when calculating result string length.
(closes issue #12696)
 Reported by: adomjan
 Patches: 
       func_realtime.c-longdelimiter.patch uploaded by adomjan (license 487)

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r118059 | tilghman | 2008-05-23 08:20:13 -0500 (Fri, 23 May 2008) | 9 lines

Blocked revisions 118055 via svnmerge

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r118055 | tilghman | 2008-05-23 08:18:44 -0500 (Fri, 23 May 2008) | 2 lines

Add format type checking for recently de-inlined function

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r119077 | russell | 2008-05-29 15:49:48 -0500 (Thu, 29 May 2008) | 10 lines

Blocked revisions 119076 via svnmerge

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r119076 | russell | 2008-05-29 15:48:33 -0500 (Thu, 29 May 2008) | 3 lines

Oddly enough, all of the contents of audiohook.h were in there twice.  I have
removed the second copy.

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r120063 | tilghman | 2008-06-03 13:24:14 -0500 (Tue, 03 Jun 2008) | 15 lines

Blocked revisions 120061 via svnmerge

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r120061 | tilghman | 2008-06-03 13:23:32 -0500 (Tue, 03 Jun 2008) | 8 lines

When listing the manager users, managers in users.conf are not shown, even
though they are allowed to connect.
(closes issue #12594)
 Reported by: bkruse
 Patches: 
       12594-managerusers-2.diff uploaded by qwell (license 4)
 Tested by: bkruse

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r120372 | russell | 2008-06-04 11:28:37 -0500 (Wed, 04 Jun 2008) | 11 lines

Blocked revisions 120371 via svnmerge

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r120371 | russell | 2008-06-04 11:26:43 -0500 (Wed, 04 Jun 2008) | 4 lines

Make the "dialplan remove include" CLI command actually work.  Also, tweak
some formatting, and make the success message a little bit more clear.
(closes AST-52)

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r120672 | russell | 2008-06-05 11:39:25 -0500 (Thu, 05 Jun 2008) | 12 lines

Blocked revisions 120671 via svnmerge

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r120671 | russell | 2008-06-05 11:38:52 -0500 (Thu, 05 Jun 2008) | 5 lines

It turns out that searching on the forwarding station isn't very useful for
most people, so pull in the changes that allow searching for SMDI messages
based on other components of the SMDI message.  Also, update the SMDI
documentation.

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r120732 | russell | 2008-06-05 13:01:45 -0500 (Thu, 05 Jun 2008) | 9 lines

Blocked revisions 120731 via svnmerge

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r120731 | russell | 2008-06-05 13:01:25 -0500 (Thu, 05 Jun 2008) | 2 lines

Add the UPGRADE.txt file from Asterisk 1.2, for handy reference.

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r121993 | twilson | 2008-06-11 18:48:38 -0500 (Wed, 11 Jun 2008) | 9 lines

Blocked revisions 121992 via svnmerge

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r121992 | twilson | 2008-06-11 18:47:23 -0500 (Wed, 11 Jun 2008) | 2 lines

Backport fix for 11520--for some reason I didn't do this back in February when I patched for trunk.

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r122315 | jpeeler | 2008-06-12 14:11:23 -0500 (Thu, 12 Jun 2008) | 9 lines

Blocked revisions 122314 via svnmerge

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r122314 | jpeeler | 2008-06-12 14:08:20 -0500 (Thu, 12 Jun 2008) | 2 lines

Adds DAHDI support alongside Zaptel. DAHDI usage favored, but all Zap stuff should continue working. Release announcement to follow.

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r122434 | jpeeler | 2008-06-12 18:09:16 -0500 (Thu, 12 Jun 2008) | 12 lines

Blocked revisions 122208 via svnmerge

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r122208 | jpeeler | 2008-06-12 10:46:08 -0500 (Thu, 12 Jun 2008) | 5 lines

(closes issue #12193)
Reported by: davidw
Patch by: Corydon76, modified by me to work properly with ParkAndAnnounce app


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r122616 | jpeeler | 2008-06-13 12:38:28 -0500 (Fri, 13 Jun 2008) | 13 lines

Blocked revisions 122613 via svnmerge

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r122613 | jpeeler | 2008-06-13 12:36:56 -0500 (Fri, 13 Jun 2008) | 6 lines

(closes issue #12846)
Reported by: Netview
Tested by: jpeeler

Use correct location to search for tonezone.

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r122664 | jpeeler | 2008-06-13 13:58:29 -0500 (Fri, 13 Jun 2008) | 8 lines

Blocked revisions 122663 via svnmerge

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r122663 | jpeeler | 2008-06-13 13:57:24 -0500 (Fri, 13 Jun 2008) | 1 line

fixed dahdi compatability header from assuming either dahdi or zaptel is installed (may not have either)
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r124744 | kpfleming | 2008-06-23 16:24:34 -0500 (Mon, 23 Jun 2008) | 10 lines

Blocked revisions 124743 via svnmerge

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r124743 | kpfleming | 2008-06-23 16:22:08 -0500 (Mon, 23 Jun 2008) | 3 lines

emit a warning if the old IAX2 call searching code finds a call when the new code did not... so that we can get rid of the old code in 2-3 months


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r124909 | tilghman | 2008-06-24 15:55:06 -0500 (Tue, 24 Jun 2008) | 13 lines

Blocked revisions 124908 via svnmerge

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r124908 | tilghman | 2008-06-24 15:52:43 -0500 (Tue, 24 Jun 2008) | 6 lines

Don't access the pvt structure if unable to acquire the lock.
(closes issue #12162)
 Reported by: norman
 Patches: 
       12162-lockfail.diff uploaded by qwell (license 4)

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r125894 | tilghman | 2008-06-27 11:48:05 -0500 (Fri, 27 Jun 2008) | 14 lines

Blocked revisions 125893 via svnmerge

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r125893 | tilghman | 2008-06-27 11:46:05 -0500 (Fri, 27 Jun 2008) | 7 lines

Since HAVE_DAHDI is defined to HAVE_ZAPTEL in dahdi_compat.h, we must first
check for HAVE_ZAPTEL.
(closes issue #12938)
 Reported by: opticron
 Patches: 
       tonezone_compat.diff uploaded by opticron (license 267)

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2008-06-27 22:43:29 +00:00
Tilghman Lesher
d0a5acecf4 Merged revisions 126057 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r126057 | tilghman | 2008-06-27 17:10:34 -0500 (Fri, 27 Jun 2008) | 12 lines

Merged revisions 126056 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r126056 | tilghman | 2008-06-27 17:01:09 -0500 (Fri, 27 Jun 2008) | 4 lines

When we get a 408 Timeout, don't stop trying to re-register.
(closes issue #12863)
 Reported by: ricvil

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2008-06-27 22:12:08 +00:00
Mark Michelson
64174167cb Port revisions 124661 and 123650 from trunk to 1.6.0
Thanks to Atis Lezdins for pointing this out on the asterisk-dev
mailing list



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Tilghman Lesher
ef21aae34b Blocked revisions 126021 via svnmerge
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r126021 | tilghman | 2008-06-27 15:38:59 -0500 (Fri, 27 Jun 2008) | 2 lines

Separate multiple items encoded into a single field with ';'

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2008-06-27 20:40:08 +00:00
Russell Bryant
2d13950a70 Merged revisions 125988 via svnmerge from
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r125988 | russell | 2008-06-27 14:19:08 -0500 (Fri, 27 Jun 2008) | 3 lines

Fix a typo.  Someone on IRC copied this literally and then wondered why it
wasn't working.  :)

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2008-06-27 19:20:27 +00:00
Matthew Fredrickson
968ef39187 Merged revisions 125984 via svnmerge from
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r125984 | mattf | 2008-06-27 14:05:40 -0500 (Fri, 27 Jun 2008) | 1 line

Revert this part of the fix.  We'll fix it in libss7
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2008-06-27 19:06:21 +00:00
Matthew Fredrickson
3fcfde9258 Merged revisions 125982 via svnmerge from
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r125982 | mattf | 2008-06-27 14:00:44 -0500 (Fri, 27 Jun 2008) | 1 line

Obviously somebody didn't compile with libss7 support when doing the DAHDI conversion.
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2008-06-27 19:01:16 +00:00
Matthew Fredrickson
9cb4b02a1a Merged revisions 125980 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r125980 | mattf | 2008-06-27 13:32:17 -0500 (Fri, 27 Jun 2008) | 1 line

Add support for new commands to block/unblock all CICs on a linkset
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2008-06-27 18:33:22 +00:00
Brett Bryant
9da0e24388 Merged revisions 125947 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r125947 | bbryant | 2008-06-27 12:35:41 -0500 (Fri, 27 Jun 2008) | 1 line

Small error in the function that converts peer transports to a string.
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2008-06-27 17:36:14 +00:00
Tilghman Lesher
405af80010 Blocked revisions 125895 via svnmerge
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r125895 | tilghman | 2008-06-27 12:02:56 -0500 (Fri, 27 Jun 2008) | 2 lines

Document DLA_UNLOCK and DLA_LOCK

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2008-06-27 17:03:41 +00:00
Brett Bryant
06c98e7444 Merged revisions 125891 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r125891 | bbryant | 2008-06-27 11:28:06 -0500 (Fri, 27 Jun 2008) | 6 lines

Change the way that the transport option works for sip users. transport will now take multiple arguments, the first one listed will be the one used 
for new dialogs, and the rest listed will be acceptable ways for that peer to contact us. This fixes a minor bug where, because SIP TCP/UDP run on 
the same port, could cause a TCP peer to be saved in the ast_db. There will also be warnings when a transport is changed for an unexpected reason.

(issue #12799)

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2008-06-27 16:29:14 +00:00
Mark Michelson
043e029e48 Blocked revisions 125856 via svnmerge
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r125856 | mmichelson | 2008-06-27 11:18:07 -0500 (Fri, 27 Jun 2008) | 3 lines

Remove debug message


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2008-06-27 16:19:46 +00:00
Mark Michelson
1fbfd723e8 Merged revisions 125855 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r125855 | mmichelson | 2008-06-27 11:16:13 -0500 (Fri, 27 Jun 2008) | 5 lines

Ensure the thread-safety of the monexec variable in app_queue.

Thanks to Russell for pointing out the problem


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2008-06-27 16:19:16 +00:00
Tilghman Lesher
dbdc741920 Merged revisions 125853 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r125853 | tilghman | 2008-06-27 11:00:05 -0500 (Fri, 27 Jun 2008) | 3 lines

Revert half of the fix, as this part may have been unnecessary (related to issue #12914)
Requested here: http://lists.digium.com/pipermail/asterisk-dev/2008-June/033658.html

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2008-06-27 16:01:47 +00:00
Mark Michelson
307dc9d032 Make sure to only include dahdi/user.h if we have
installed DAHDI. 



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@125852 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-27 14:57:28 +00:00
Mark Michelson
489a80e711 I accidentally committed a change to chan_iax2.c in addition to a change
to app_queue.c. Reverting the change to chan_iax2.c, even though it may turn
out that this change is necessary.



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2008-06-27 14:21:01 +00:00
Mark Michelson
aec8cda0c8 Merged revisions 125799 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r125799 | mmichelson | 2008-06-27 09:14:09 -0500 (Fri, 27 Jun 2008) | 3 lines

Remove an unneeded target from the Makefile


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2008-06-27 14:14:46 +00:00
Tilghman Lesher
cdad87e85f Blocked revisions 125796 via svnmerge
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r125796 | tilghman | 2008-06-27 09:08:38 -0500 (Fri, 27 Jun 2008) | 2 lines

Push relatively unused compiler options down the list, keeping the popular options at the top.

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2008-06-27 14:09:15 +00:00
Tilghman Lesher
a12130a093 Merged revisions 125794 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r125794 | tilghman | 2008-06-27 08:54:13 -0500 (Fri, 27 Jun 2008) | 10 lines

Merged revisions 125793 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r125793 | tilghman | 2008-06-27 08:45:03 -0500 (Fri, 27 Jun 2008) | 2 lines

In this debugging function, copy to a buffer instead of using potentially unsafe pointers.

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2008-06-27 14:07:13 +00:00
Tilghman Lesher
d93ad61286 Merged revisions 125741 via svnmerge from
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r125741 | tilghman | 2008-06-27 07:28:38 -0500 (Fri, 27 Jun 2008) | 15 lines

Merged revisions 125740 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r125740 | tilghman | 2008-06-27 07:19:39 -0500 (Fri, 27 Jun 2008) | 7 lines

Add proper deadlock avoidance.
(closes issue #12914)
 Reported by: ozan
 Patches: 
       20080625__bug12914.diff.txt uploaded by Corydon76 (license 14)
 Tested by: ozan

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2008-06-27 12:32:02 +00:00
Philippe Sultan
ca5d70859b Merged revisions 125703 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r125703 | phsultan | 2008-06-27 09:28:17 +0200 (Fri, 27 Jun 2008) | 1 line

Fix a compile time error that occurs if OpenSSL is not installed. Reported by Noel Morais on the users mailing list
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2008-06-27 07:41:54 +00:00
Mark Michelson
a8979176c1 SENTINEL is not defined in 1.6.0
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2008-06-27 01:09:04 +00:00
Mark Michelson
b43587a16e Merged revisions 125666 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r125666 | mmichelson | 2008-06-26 19:22:03 -0500 (Thu, 26 Jun 2008) | 3 lines

Make this compile with dev-mode on


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2008-06-27 00:22:23 +00:00
Mark Michelson
6250e77c59 Merged revisions 125649 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r125649 | mmichelson | 2008-06-26 19:15:54 -0500 (Thu, 26 Jun 2008) | 15 lines

The monitor-join option for queues was deprecated in favor of using
MixMonitor to mix audio. However, it was pointed out to me that because
of this, the command set for the MONITOR_EXEC variable is ignored as well.
This means that people can't do their own custom mixing commands at the end
of recordings in order to make, for instance, stereo recordings of calls.

With this patch, app_queue will set the "joinfiles" variable for the channel's
monitor if MONITOR_EXEC is not zero-length. This means that for normal audio
mixing, MixMonitor is still the preferred choice, but we allow custom
mixing to be done with the two Monitor streams if desired.

(closes issue #12923)
Reported by: snyfer


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2008-06-27 00:17:39 +00:00
Mark Michelson
22bfcffec8 Blocked revisions 125647 via svnmerge
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r125647 | mmichelson | 2008-06-26 18:35:29 -0500 (Thu, 26 Jun 2008) | 16 lines

Improve consistency between app_dial and app_queue with regards
to how language is handled between two channels whose native
language is different. Prior to this patch, app_dial would have
the callee inherit the caller's language, and app_queue would not.

After this patch, app_dial no longer has the language inheritance
capability. This seems to make the most sense since it seems more
natural for a person to hear files played back in his/her native
language instead of the language of the person on the far end of
the call. See the CHANGES file for hints on how to keep the 
previous behavior of app_dial if desired.

(closes issue #12489)
Reported by: bcnit


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2008-06-26 23:36:11 +00:00
Michiel van Baak
3b3e75ff1a Blocked revisions 125596 via svnmerge
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r125596 | mvanbaak | 2008-06-27 01:18:46 +0200 (Fri, 27 Jun 2008) | 5 lines

remove block of commented code to set __ourip
This is now handled in skinny_register and load_config.

part two of chan_skinny cleanup

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2008-06-26 23:19:49 +00:00
Michiel van Baak
3ab5a844b3 Blocked revisions 125593 via svnmerge
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r125593 | mvanbaak | 2008-06-27 01:14:08 +0200 (Fri, 27 Jun 2008) | 4 lines

remove paging device from chan_skinny.
This has never been used, and noone could give us info about what
it is used for.

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2008-06-26 23:15:09 +00:00
Mark Michelson
74e07ac0aa Merged revisions 125591 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r125591 | mmichelson | 2008-06-26 18:06:18 -0500 (Thu, 26 Jun 2008) | 3 lines

Fix a really stupid mistake


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2008-06-26 23:06:47 +00:00
Jason Parker
2de235d2b0 Merged revisions 125589 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r125589 | qwell | 2008-06-26 18:04:18 -0500 (Thu, 26 Jun 2008) | 9 lines

Merged revisions 125587 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r125587 | qwell | 2008-06-26 18:03:15 -0500 (Thu, 26 Jun 2008) | 1 line

Make sure to unlock the lock_info lock (huh?).  Possible deadlock?
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2008-06-26 23:05:04 +00:00
Mark Michelson
9152577c24 Merged revisions 125586 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r125586 | mmichelson | 2008-06-26 18:01:02 -0500 (Thu, 26 Jun 2008) | 19 lines

Merged revisions 125585 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r125585 | mmichelson | 2008-06-26 17:52:39 -0500 (Thu, 26 Jun 2008) | 11 lines

Add the interface of a queue member to the output of the "queue show" command
so that it can easily be associated with a queue member's name. This helps
so that the appropriate queue member can be removed or paused since the 
interface is required, not the member's name.

(closes issue #12783)
Reported by: davevg
Patches:
      app_queue.diff uploaded by davevg (license 209) with small mod from me


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2008-06-26 23:04:12 +00:00
Tilghman Lesher
be0a82a3bb Merged revisions 125583 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r125583 | tilghman | 2008-06-26 17:49:16 -0500 (Thu, 26 Jun 2008) | 2 lines

Don't hang if the command is blank

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2008-06-26 22:50:17 +00:00
Mark Michelson
6d89f9ac86 Merged revisions 125531 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r125531 | mmichelson | 2008-06-26 17:03:54 -0500 (Thu, 26 Jun 2008) | 17 lines

Blocked revisions 125530 via svnmerge

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r125530 | mmichelson | 2008-06-26 17:02:55 -0500 (Thu, 26 Jun 2008) | 10 lines

Backport of attended transfer queue_log patch from trunk.

This patch allows for attended transfers to be logged in the
queue_log the same way that blind transfers have always been.

It was decided by popular opinion on the asterisk-dev mailing
list that this should be backported to 1.4. Thanks to everyone
who gave an opinion.


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2008-06-26 22:06:20 +00:00
Mark Michelson
3f71d4600e Merged revisions 125477 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r125477 | mmichelson | 2008-06-26 15:57:41 -0500 (Thu, 26 Jun 2008) | 19 lines

Merged revisions 125476 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r125476 | mmichelson | 2008-06-26 15:56:01 -0500 (Thu, 26 Jun 2008) | 11 lines

Prior to this patch, the "queue show" command used cached
information for realtime queues instead of giving up-to-date
info. Now realtime is queried for the latest and greatest in
queue info.

(closes issue #12858)
Reported by: bcnit
Patches:
      queue_show.patch uploaded by putnopvut (license 60)


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2008-06-26 20:58:48 +00:00
Tilghman Lesher
4bc9e19d8c Blocked revisions 125438 via svnmerge
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r125438 | tilghman | 2008-06-26 12:40:25 -0500 (Thu, 26 Jun 2008) | 6 lines

Don't play "your message has been saved" twice.
(closes issue #12893)
 Reported by: jaroth
 Patches: 
       duplicate_saved.patch uploaded by jaroth (license 50)

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2008-06-26 17:42:24 +00:00
Olle Johansson
ebb266391f Merged revisions 125385 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r125385 | oej | 2008-06-26 18:54:22 +0200 (Tor, 26 Jun 2008) | 12 lines


Merged revisions 125384 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r125384 | oej | 2008-06-26 18:32:08 +0200 (Tor, 26 Jun 2008) | 3 lines

Add support for peer realm based auth (a few missing lines, the rest is well documented but never worked)


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2008-06-26 17:07:47 +00:00
Tilghman Lesher
a875b5d1bf Blocked revisions 125386 via svnmerge
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r125386 | tilghman | 2008-06-26 12:06:17 -0500 (Thu, 26 Jun 2008) | 6 lines

Convert casts to unions, to fix alignment issues on Solaris
(closes issue #12932)
 Reported by: snuffy
 Patches: 
       bug_12932_20080627.diff uploaded by snuffy (license 35)

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2008-06-26 17:07:05 +00:00
Kevin P. Fleming
e04a7e8ec0 Merged revisions 125333 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r125333 | kpfleming | 2008-06-26 10:50:07 -0500 (Thu, 26 Jun 2008) | 13 lines

Merged revisions 125327 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r125327 | kpfleming | 2008-06-26 10:30:33 -0500 (Thu, 26 Jun 2008) | 5 lines

ensure that (whenever possible) if we generate a log message because an ioctl() call to DAHDI/Zaptel failed, that we include the reason it failed by including the stringified error number

(issue AST-80)


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2008-06-26 15:52:20 +00:00
Kevin P. Fleming
d28ea4a844 Merged revisions 125279 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r125279 | kpfleming | 2008-06-26 07:09:24 -0500 (Thu, 26 Jun 2008) | 2 lines

fix compile failure found by buildbot (go, buildbot!)

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2008-06-26 12:11:19 +00:00
Tilghman Lesher
07126be389 Merged revisions 125277 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r125277 | tilghman | 2008-06-26 06:02:11 -0500 (Thu, 26 Jun 2008) | 15 lines

Merged revisions 125276 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r125276 | tilghman | 2008-06-26 06:01:21 -0500 (Thu, 26 Jun 2008) | 7 lines

Check for rtcp structure before trying to delete schedule.
(closes issue #12872)
 Reported by: destiny6628
 Patches: 
       20080621__bug12872.diff.txt uploaded by Corydon76 (license 14)
 Tested by: destiny6628

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2008-06-26 11:08:34 +00:00
Tilghman Lesher
ea11341760 Merged revisions 125223 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r125223 | tilghman | 2008-06-25 20:25:16 -0500 (Wed, 25 Jun 2008) | 12 lines

Merged revisions 125218 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r125218 | tilghman | 2008-06-25 20:24:26 -0500 (Wed, 25 Jun 2008) | 4 lines

Document ackcall=always.
(closes issue #12852)
 Reported by: davidw

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2008-06-26 01:26:29 +00:00
Tilghman Lesher
8162fdbc32 Merged revisions 125191 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r125191 | tilghman | 2008-06-25 20:11:43 -0500 (Wed, 25 Jun 2008) | 6 lines

Update sample configuration to match what are now the defaults for the prefix.
(closes issue #12838, related to issue #12198)
 Reported by: pabelanger
 Patches: 
       http.conf.diff2 uploaded by pabelanger (license 224)

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2008-06-26 01:13:45 +00:00
Kevin P. Fleming
0b30f14b33 Merged revisions 125138 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r125138 | kpfleming | 2008-06-25 18:05:28 -0500 (Wed, 25 Jun 2008) | 18 lines

Merged revisions 125132 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r125132 | kpfleming | 2008-06-25 17:21:30 -0500 (Wed, 25 Jun 2008) | 10 lines

allow tonezone to live in a different place than DAHDI/Zaptel, since dahdi-tools and dahdi-linux are now separate packages and can be installed in different places

don't include tonezone.h in dahdi_compat.h, because only a couple of modules need it

get app_rpt building again after the DAHDI changes

(closes issue #12911)
Reported by: tzafrir


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2008-06-25 23:20:35 +00:00
Mark Michelson
e6524f22ec Blocked revisions 125135 via svnmerge
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r125135 | mmichelson | 2008-06-25 17:40:41 -0500 (Wed, 25 Jun 2008) | 3 lines

Fix indentation


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2008-06-25 22:40:59 +00:00
Mark Michelson
84505c0c58 Blocked revisions 125133 via svnmerge
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r125133 | mmichelson | 2008-06-25 17:25:20 -0500 (Wed, 25 Jun 2008) | 11 lines

Fix a bug in the rwlock tracking. ast_rwlock_unlock did not take into
account that multiple threads could hold the same rdlock at the same time.
As such, it expected that when a thread released a lock that it must have
been the last to acquire the lock as well. Erroneous error messages would
be sent to the console stating that a thread was attempting to unlock a lock
it did not own.

Now all threads are examined to be sure that the message is only printed 
when it is supposed to be printed.


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2008-06-25 22:25:56 +00:00
Michiel van Baak
6cefd04eb6 Blocked revisions 125096 via svnmerge
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r125096 | mvanbaak | 2008-06-25 21:37:40 +0200 (Wed, 25 Jun 2008) | 10 lines

implement transfer functionality in chan_skinny

(closes issue #9939)
Reported by: wedhorn
Patches:
      transfer_v6.diff uploaded by wedhorn (license 30)
      chan_skinny-transfer-trunk-v10.txt uploaded by DEA (license 3)
      chan_skinny-transfer-trunk-v12.txt uploaded by mvanbaak (license 7)
Tested by: DEA, wedhorn, mvanbaak

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2008-06-25 19:50:08 +00:00
Tilghman Lesher
441da04c0b Merged revisions 124966 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r124966 | tilghman | 2008-06-24 20:08:37 -0500 (Tue, 24 Jun 2008) | 15 lines

Merged revisions 124965 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r124965 | tilghman | 2008-06-24 19:46:24 -0500 (Tue, 24 Jun 2008) | 7 lines

Pvt deadlock causes some channels to get stuck in Reserved status.
(closes issue #12621)
 Reported by: fabianoheringer
 Patches: 
       20080612__bug12621.diff.txt uploaded by Corydon76 (license 14)
 Tested by: fabianoheringer

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2008-06-25 01:13:38 +00:00
Tilghman Lesher
78b60650e4 Merged revisions 124912 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r124912 | tilghman | 2008-06-24 16:18:52 -0500 (Tue, 24 Jun 2008) | 16 lines

Merged revisions 124910 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r124910 | tilghman | 2008-06-24 16:08:52 -0500 (Tue, 24 Jun 2008) | 8 lines

Occasionally control characters find their way into CallerID.  These need to
be stripped prior to placing CallerID in the headers of an email.
(closes issue #12759)
 Reported by: RobH
 Patches: 
       20080602__bug12759__2.diff.txt uploaded by Corydon76 (license 14)
 Tested by: RobH

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2008-06-24 21:39:37 +00:00
Philippe Sultan
a8231e854e Merged revisions 124872 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r124872 | phsultan | 2008-06-24 19:50:22 +0200 (Tue, 24 Jun 2008) | 6 lines

Subscribe to buddy's presence only if we really need to. That is, if
the corresponding roster item has a subscription value set to "none"
or "from".

Make the code more readable.

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2008-06-24 17:52:02 +00:00
Philippe Sultan
4ee5ce0e55 Merged revisions 124870 via svnmerge from
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r124870 | phsultan | 2008-06-24 19:28:39 +0200 (Tue, 24 Jun 2008) | 1 line

Code simplification
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2008-06-24 17:34:53 +00:00
Sean Bright
5cea15f928 Blocked revisions 124835 via svnmerge
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r124835 | seanbright | 2008-06-24 07:02:02 -0400 (Tue, 24 Jun 2008) | 1 line

Update CHANGES and UPGRADE.txt per kpfleming's mail to #asterisk-dev.
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2008-06-24 11:15:38 +00:00
Dwayne M. Hubbard
84b65565d8 blocked revision 124707, taskprocessors are not in 1.6.0
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2008-06-23 15:44:35 +00:00
Steve Murphy
f33fab929e Merged revisions 124541 via svnmerge from
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r124541 | murf | 2008-06-21 20:58:06 -0600 (Sat, 21 Jun 2008) | 17 lines

Merged revisions 124540 via svnmerge from 
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r124540 | murf | 2008-06-21 20:54:52 -0600 (Sat, 21 Jun 2008) | 9 lines

(closes issue #12910)
Reported by: chris-mac

Sorry, my testing did not contain the simple case of forkCDR(v),
I am much embarrassed to admit.  If I had, I would have 
more solidly initialized the opts element for varset.



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2008-06-22 03:18:10 +00:00
Tilghman Lesher
5f20127c71 Merged revisions 124505 via svnmerge from
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r124505 | tilghman | 2008-06-21 07:53:48 -0500 (Sat, 21 Jun 2008) | 4 lines

Reduce warning to debug, otherwise we flood the log when we (legitimately)
can't find a record.
(Closes issue #12908)

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2008-06-21 12:54:42 +00:00
Tilghman Lesher
72cd5e5eac Merged revisions 124451 via svnmerge from
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r124451 | tilghman | 2008-06-20 18:13:21 -0500 (Fri, 20 Jun 2008) | 14 lines

Merged revisions 124450 via svnmerge from 
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r124450 | tilghman | 2008-06-20 18:12:33 -0500 (Fri, 20 Jun 2008) | 6 lines

usleep with a value over 1,000,000 is nonportable.  Changing to use sleep()
instead.  (closes issue #12814)
 Reported by: pputman
 Patches: 
       app_rtp_sleep.patch uploaded by pputman (license 81)

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2008-06-20 23:14:01 +00:00
Tilghman Lesher
cec0e915ee Merged revisions 124396 via svnmerge from
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r124396 | tilghman | 2008-06-20 17:04:37 -0500 (Fri, 20 Jun 2008) | 11 lines

Merged revisions 124395 via svnmerge from 
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r124395 | tilghman | 2008-06-20 17:02:55 -0500 (Fri, 20 Jun 2008) | 3 lines

If the last character in a string to be parsed is the delimiter, then we should
count that final empty string as an additional argument.

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2008-06-20 22:06:01 +00:00
Jeff Gehlbach
748f25bfb6 Merged revisions 124392-124393 via svnmerge from
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r124392 | jeffg | 2008-06-20 17:36:01 -0400 (Fri, 20 Jun 2008) | 9 lines

Merged revisions 124372 via svnmerge from 
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r124372 | jeffg | 2008-06-20 17:14:40 -0400 (Fri, 20 Jun 2008) | 1 line

Fix issues in digium-mib.txt and asterisk-mib.txt to placate smilint - bug 12905
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r124393 | jeffg | 2008-06-20 17:43:18 -0400 (Fri, 20 Jun 2008) | 12 lines

(Missed committing . on previous commit.....)
Merged revisions 124372 via svnmerge from 
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r124372 | jeffg | 2008-06-20 17:14:40 -0400 (Fri, 20 Jun 2008) | 1 line

Fix issues in digium-mib.txt and asterisk-mib.txt to placate smilint - bug 12905
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2008-06-20 21:48:15 +00:00
Tilghman Lesher
fba1394ba0 Merged revisions 124316 via svnmerge from
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r124316 | tilghman | 2008-06-20 15:17:04 -0500 (Fri, 20 Jun 2008) | 16 lines

Merged revisions 124315 via svnmerge from 
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r124315 | tilghman | 2008-06-20 15:16:02 -0500 (Fri, 20 Jun 2008) | 8 lines

When using a Local channel, started by a call file, with a destination of an
AGI script, the AGI script does not always get notified of a hangup if the
underlying channel hangs up early.
(closes issue #11833)
 Reported by: IgorG
 Patches: 
       local_hangup-v1.diff uploaded by IgorG (license 20)

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2008-06-20 20:18:08 +00:00
Mark Michelson
f5ed251dc3 Merged revisions 124278 via svnmerge from
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r124278 | mmichelson | 2008-06-20 11:30:18 -0500 (Fri, 20 Jun 2008) | 6 lines

Change references to doc/channelvariables.txt to
doc/tex/channelvariables.tex.

This issue came up on the asterisk-dev mailing list.


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2008-06-20 16:31:26 +00:00
Mark Michelson
6f0c9715db Merged revisions 124243 via svnmerge from
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r124243 | mmichelson | 2008-06-20 10:20:11 -0500 (Fri, 20 Jun 2008) | 9 lines

Add a missing "ChannelType" header to one of the "PeerStatus" manager
events in chan_sip

(closes issue #12904)
Reported by: eliel
Patches:
      chan_sip.c.patch uploaded by eliel (license 64)


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2008-06-20 15:20:46 +00:00
Tilghman Lesher
553c1e8aab Merged revisions 124183 via svnmerge from
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r124183 | tilghman | 2008-06-19 17:59:41 -0500 (Thu, 19 Jun 2008) | 15 lines

Merged revisions 124182 via svnmerge from 
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r124182 | tilghman | 2008-06-19 17:53:22 -0500 (Thu, 19 Jun 2008) | 7 lines

It's possible for a hangup to be received, even just after the initial cid
spill.
(closes issue #12453)
 Reported by: Alex728
 Patches: 
       20080604__bug12453.diff.txt uploaded by Corydon76 (license 14)

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2008-06-19 23:02:49 +00:00
Mark Michelson
49fffcde19 Blocked revisions 124180 via svnmerge
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r124180 | mmichelson | 2008-06-19 17:34:03 -0500 (Thu, 19 Jun 2008) | 10 lines

Fix attachment behavior when using IMAP storage for voicemails
1. Filenames had an extra "msg" in the attachment name
2. The attachment was being saved twice

(closes issue #12894)
Reported by: jaroth
Patches:
      imap_attach.patch uploaded by jaroth (license 50)


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2008-06-19 22:34:54 +00:00
Michiel van Baak
d026b05b7d Blocked revisions 124127 via svnmerge
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r124127 | mvanbaak | 2008-06-19 22:48:33 +0200 (Thu, 19 Jun 2008) | 10 lines

Older versions of GNU gcc do not allow 'NULL' as sentinel.
They want (char *)NULL as sentinel.
An example is OpenBSD (confirmed on 4.3) that ships with gcc 3.3.4

This commit introduces a contstant SENTINEL which is declared as:
#define SENTINEL ((char *)NULL)

All places I could test compile on my openbsd system are converted.
Update CODING-GUIDELINES to tell about this constant.

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2008-06-19 21:03:54 +00:00
Tilghman Lesher
43353b7374 Blocked revisions 124125 via svnmerge
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r124125 | tilghman | 2008-06-19 15:35:56 -0500 (Thu, 19 Jun 2008) | 2 lines

Oops

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2008-06-19 20:36:38 +00:00
Mark Michelson
c48e31ba8c Merged revisions 124121 via svnmerge from
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r124121 | mmichelson | 2008-06-19 15:30:23 -0500 (Thu, 19 Jun 2008) | 16 lines

Merged revisions 124112 via svnmerge from 
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r124112 | mmichelson | 2008-06-19 15:28:41 -0500 (Thu, 19 Jun 2008) | 8 lines

Fix IMAP forwarding so that messages are sent to the proper mailbox.

(closes issue #12897)
Reported by: jaroth
Patches:
      destination_forward.patch uploaded by jaroth (license 50)


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2008-06-19 20:32:54 +00:00
Tilghman Lesher
ed52cefb14 Blocked revisions 124102 via svnmerge
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r124102 | tilghman | 2008-06-19 15:25:37 -0500 (Thu, 19 Jun 2008) | 3 lines

Make OpenBSD compile again
(reported by mvanbaak via IRC -dev)

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2008-06-19 20:26:31 +00:00
Brett Bryant
d08e081337 Merged revisions 124064 via svnmerge from
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r124064 | bbryant | 2008-06-19 14:48:26 -0500 (Thu, 19 Jun 2008) | 2 lines

Add errors that report any locks held by threads when they are being closed.

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2008-06-19 19:49:54 +00:00
Tilghman Lesher
90afe3757e Blocked revisions 124049 via svnmerge
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r124049 | tilghman | 2008-06-19 14:22:59 -0500 (Thu, 19 Jun 2008) | 6 lines

Allow alternative extensions to be specified for a user.
(closes issue #12830)
 Reported by: jcollie
 Patches: 
       astertisk-trunk-121496-alternate-extensions.patch uploaded by jcollie (license 412)

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2008-06-19 19:23:50 +00:00
Brett Bryant
2652d0d528 Merged revisions 124024 via svnmerge from
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r124024 | bbryant | 2008-06-19 13:57:04 -0500 (Thu, 19 Jun 2008) | 2 lines

Fix bug in sip registration that sets the default port to 5060 for tls.

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Tilghman Lesher
31c415c6f5 Blocked revisions 123988 via svnmerge
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r123988 | tilghman | 2008-06-19 12:55:34 -0500 (Thu, 19 Jun 2008) | 3 lines

Detect if the installed gcc version supports the warn_unused_result attribute.
Reported by mvanbaak via IRC -dev.

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Tilghman Lesher
b8690fbe1f Merged revisions 123952 via svnmerge from
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r123952 | tilghman | 2008-06-19 12:22:27 -0500 (Thu, 19 Jun 2008) | 6 lines

Don't change pointers that need to be later passed back for deallocation.
(closes issue #12572)
 Reported by: flyn
 Patches: 
       20080613__bug12572.diff.txt uploaded by Corydon76 (license 14)

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Tilghman Lesher
b717d74470 Merged revisions 123931 via svnmerge from
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r123931 | tilghman | 2008-06-19 12:02:54 -0500 (Thu, 19 Jun 2008) | 13 lines

Merged revisions 123930 via svnmerge from 
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r123930 | tilghman | 2008-06-19 11:58:19 -0500 (Thu, 19 Jun 2008) | 5 lines

Change informative messages to use the _multiple variant when multiple formats
are possible.
(Closes issue #12848)
Reported by klaus3000

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Tilghman Lesher
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r123913 | tilghman | 2008-06-19 11:26:50 -0500 (Thu, 19 Jun 2008) | 13 lines

Merged revisions 123909 via svnmerge from 
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r123909 | tilghman | 2008-06-19 11:26:03 -0500 (Thu, 19 Jun 2008) | 5 lines

Only process 40 arguments (20 files) at once with xargs, because some older
shells may force xargs to separate on an odd boundary.
(Closes issue #12883)
Reported by Nik Soggia

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Tilghman Lesher
487c0d4465 Merged revisions 123887 via svnmerge from
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r123887 | tilghman | 2008-06-19 11:21:32 -0500 (Thu, 19 Jun 2008) | 12 lines

Merged revisions 123883 via svnmerge from 
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r123883 | tilghman | 2008-06-19 11:20:41 -0500 (Thu, 19 Jun 2008) | 4 lines

Correct description of notifyringing option.
(Closes issue #12890)
Reported by gminet

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Tilghman Lesher
9787f4b97d Merged revisions 123870 via svnmerge from
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r123870 | tilghman | 2008-06-19 11:08:29 -0500 (Thu, 19 Jun 2008) | 14 lines

Merged revisions 123869 via svnmerge from 
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r123869 | tilghman | 2008-06-19 11:07:23 -0500 (Thu, 19 Jun 2008) | 6 lines

The RDTSC instruction was introduced on the Pentium line of microprocessors,
and is not compatible with certain 586 clones, like Cyrix.  Hence, asking for
i386 compatibility was always incorrect. See http://en.wikipedia.org/wiki/RDTSC
(Closes issue #12886)
Reported by tecnoxarxa

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Mark Michelson
94291642d2 Blocked revisions 123867 via svnmerge
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r123867 | mmichelson | 2008-06-19 10:55:10 -0500 (Thu, 19 Jun 2008) | 9 lines

Forwarding non-urgent IMAP messages could inadvertently cause the messages
to be marked urgent. This fixes that issue.

(closes issue #12895)
Reported by: jaroth
Patches:
      urgent_forwarding.patch uploaded by jaroth (license 50)


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Tilghman Lesher
f58e69881d Blocked revisions 123865 via svnmerge
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r123865 | tilghman | 2008-06-19 10:52:17 -0500 (Thu, 19 Jun 2008) | 4 lines

Missing comma
(closes issue #12891)
 Reported by: chris-mac

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Sean Bright
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r123830 | seanbright | 2008-06-19 10:28:56 -0400 (Thu, 19 Jun 2008) | 1 line

Update the queuelog.tex documentation as well.
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Sean Bright
d7d343b433 Blocked revisions 123828 via svnmerge
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r123828 | seanbright | 2008-06-19 10:24:43 -0400 (Thu, 19 Jun 2008) | 7 lines

Include original position in TRANSFER entries written to queue_log.

(closes issue #12888)
Reported by: slavon
Patches:
      app_queue_transfer_patch_trunk.diff uploaded by slavon (license 288)

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Tilghman Lesher
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r123770 | tilghman | 2008-06-18 17:17:17 -0500 (Wed, 18 Jun 2008) | 16 lines

Merged revisions 123769 via svnmerge from 
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r123769 | tilghman | 2008-06-18 17:08:30 -0500 (Wed, 18 Jun 2008) | 8 lines

Add support for saying numbers in Hebrew.
(closes issue #11662)
 Reported by: greenfieldtech
 Patches: 
       say.c.patch-12042008 uploaded by greenfieldtech (license 369)
       Hebrew-Sounds.ods uploaded by greenfieldtech
       (with signficant changes to the spreadsheet by me)

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Tilghman Lesher
04851b1661 Merged revisions 123715 via svnmerge from
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r123715 | tilghman | 2008-06-18 15:23:58 -0500 (Wed, 18 Jun 2008) | 15 lines

Merged revisions 123710 via svnmerge from 
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r123710 | tilghman | 2008-06-18 15:22:42 -0500 (Wed, 18 Jun 2008) | 7 lines

Set the variables top-down, so that if a script sets a variable more than once,
the last one will take precedence.
(closes issue #12673)
 Reported by: phber
 Patches: 
       20080519__bug12673.diff.txt uploaded by Corydon76 (license 14)

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Brett Bryant
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r123692 | bbryant | 2008-06-18 15:07:56 -0500 (Wed, 18 Jun 2008) | 2 lines

Fix a crash in tcp and tls connections related to reference counts.

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Mark Michelson
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r123652 | mmichelson | 2008-06-18 10:08:56 -0500 (Wed, 18 Jun 2008) | 7 lines

A portion of the code which handled the 'c' queue option had been
removed. No telling when it happened. Anyway, it's back in now
and works properly.

(Based on issue reported on mailing list)


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2008-06-18 15:09:19 +00:00
Mark Michelson
8641a52b56 Blocked revisions 123650 via svnmerge
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r123650 | mmichelson | 2008-06-18 09:54:47 -0500 (Wed, 18 Jun 2008) | 4 lines

Silly pointers. This fixes a memory corruption I introduced
with the attended transfer logging.


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2008-06-18 14:55:32 +00:00
Tilghman Lesher
be004ea340 Blocked revisions 123648 via svnmerge
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r123648 | tilghman | 2008-06-18 08:09:02 -0500 (Wed, 18 Jun 2008) | 6 lines

Channel lock janitor -- add locks around retrieval of channel variables
(closes issue #12840)
 Reported by: pputman
 Patches: 
       app_dial_threadsafe3.patch uploaded by pputman (license 81)

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2008-06-18 13:09:49 +00:00
Russell Bryant
2f6a7d610d don't use trunk only API for frame data
(closes issue #12881)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@123647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-18 12:34:22 +00:00
Russell Bryant
95934c81c2 re-add app_fax ... it got accidentally removed
(closes issue #12881)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@123646 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-18 10:14:22 +00:00
Sean Bright
e99222cf45 Blocked revisions 123609 via svnmerge
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r123609 | seanbright | 2008-06-17 20:33:31 -0400 (Tue, 17 Jun 2008) | 1 line

Whitespace only
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2008-06-18 00:33:53 +00:00
Brett Bryant
6891a31c38 Merged revisions 123546 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r123546 | bbryant | 2008-06-17 16:46:57 -0500 (Tue, 17 Jun 2008) | 5 lines

Updates all usages of ast_tcptls_session_instance to be managed by reference counts so that they only get destroyed when all threads are done using 
them, and memory does not get free'd causing strange issues with SIP. 

This code was originally written by russellb in the team/group/issue_11972/ branch.

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2008-06-17 21:57:15 +00:00
Tilghman Lesher
ce6304e162 Blocked revisions 123544 via svnmerge
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r123544 | tilghman | 2008-06-17 16:42:42 -0500 (Tue, 17 Jun 2008) | 7 lines

Add an option, specifying maximum analysis time for talk detection.
(closes issue #12149)
 Reported by: davevg
 Patches: 
       app_talkdetect.c.diff uploaded by davevg (license 209)
       (Plus a few additional cleanups by moi)

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2008-06-17 21:43:27 +00:00
Mark Michelson
aec2526866 Blocked revisions 123541 via svnmerge
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r123541 | mmichelson | 2008-06-17 16:33:40 -0500 (Tue, 17 Jun 2008) | 3 lines

Put quotes around "test"


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2008-06-17 21:34:01 +00:00
Mark Michelson
f552bb9242 Blocked revisions 123526 via svnmerge
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r123526 | mmichelson | 2008-06-17 16:26:31 -0500 (Tue, 17 Jun 2008) | 13 lines

_ys pointed out in #asterisk-bugs that he was experiencing
a memory leak when running the astobj2 test CLI command. After
searching, it appears the leak was in the command handler itself.

Each object was allocated (recount = 1) and then linked into
a container (refounct = 2). Then at the end of the function, 
the container was unreffed, causing all the objects to have
their refcount decremented by one, leaving the refcount for
all objects allocated in that function at 1. I've now added
an extra unref to the mix so that the refcount equals zero
when the container is unreffed.


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2008-06-17 21:28:02 +00:00
Mark Michelson
00d9f25e41 Merged revisions 123486 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r123486 | mmichelson | 2008-06-17 15:28:47 -0500 (Tue, 17 Jun 2008) | 12 lines

Merged revisions 123485 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r123485 | mmichelson | 2008-06-17 15:26:38 -0500 (Tue, 17 Jun 2008) | 4 lines

Make chan_sip build under dev mode with compilers >= GCC 4.2
Thanks to jpeeler for alerting me of this


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2008-06-17 20:30:00 +00:00
Steve Murphy
609bf84b71 block 123448 from trunk; it doesn't apply here.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@123473 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-17 20:23:26 +00:00
Mark Michelson
6c849f921f Blocked revisions 123456 via svnmerge
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r123456 | mmichelson | 2008-06-17 15:19:49 -0500 (Tue, 17 Jun 2008) | 3 lines

Add the same fix from revision 123271 to container_destruct_debug.


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2008-06-17 20:21:08 +00:00
Steve Murphy
0539ac3c7b Blocked 123446 (the cdr_tds.c fix) from 1.6.0, because it's a fix to new code in trunk.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@123447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-17 20:05:42 +00:00
Tilghman Lesher
e41ffcbca3 Merged revisions 123392 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r123392 | tilghman | 2008-06-17 13:57:45 -0500 (Tue, 17 Jun 2008) | 11 lines

Merged revisions 123391 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r123391 | tilghman | 2008-06-17 13:56:53 -0500 (Tue, 17 Jun 2008) | 3 lines

Fix 3 more places where failure to lock the structure could cause the wrong lock to be
unlocked.  (Closes issue #12795)

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2008-06-17 19:01:33 +00:00
Jeff Peeler
83eedba78a Merged revisions 123238 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r123238 | jpeeler | 2008-06-16 18:05:18 -0500 (Mon, 16 Jun 2008) | 1 line

Fix some (more) variables that were forgotten to be renamed, related to 117658
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2008-06-17 18:28:56 +00:00
Jeff Peeler
8e008e0878 Blocked revisions 122243-122244 via svnmerge
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r122243 | jpeeler | 2008-06-12 12:39:38 -0500 (Thu, 12 Jun 2008) | 2 lines

Make sure we hangup any calls we have and NULL out the ss7call value when we get a reset circuit message.  Fixes crash bug. (from mattf r121857, moving from chan_zap to chan_dahdi)

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r122244 | jpeeler | 2008-06-12 12:49:37 -0500 (Thu, 12 Jun 2008) | 2 lines

Fix pseudo channel allocation errors on startup when using SS7. (from mattf r121914, moving from chan_zap to chan_dahdi)

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2008-06-17 18:28:05 +00:00
Tilghman Lesher
83b5751b10 Blocked revisions 123358 via svnmerge
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r123358 | tilghman | 2008-06-17 13:23:01 -0500 (Tue, 17 Jun 2008) | 6 lines

If we don't match registrar when destroying a context, it can cause a crash.
(closes issue #12835)
 Reported by: ys
 Patches: 
       pbx.c.diff uploaded by ys (license 281)

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2008-06-17 18:25:53 +00:00
Mark Michelson
01ec6ca9b1 Merged revisions 123334 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r123334 | mmichelson | 2008-06-17 13:09:54 -0500 (Tue, 17 Jun 2008) | 19 lines

Merged revisions 123333 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r123333 | mmichelson | 2008-06-17 13:09:16 -0500 (Tue, 17 Jun 2008) | 11 lines

Cisco BTS sends SIP responses with a tab between the Cseq number and
SIP request method in the Cseq: header. Asterisk did not handle this
properly, but with this patch, all is well.

(closes issue #12834)
Reported by: tobias_e
Patches:
      12834.patch uploaded by putnopvut (license 60)
Tested by: tobias_e


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2008-06-17 18:10:30 +00:00
Jeff Peeler
f089bbf663 Goodbye Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI. Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@123332 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-17 18:08:09 +00:00
Mark Michelson
8c4f56f424 Merged revisions 123275 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r123275 | mmichelson | 2008-06-17 10:57:43 -0500 (Tue, 17 Jun 2008) | 20 lines

Merged revisions 123274 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r123274 | mmichelson | 2008-06-17 10:56:55 -0500 (Tue, 17 Jun 2008) | 12 lines

davidw pointed out that the holdtime calculation used by
app_queue does not use "boxcar" filtering as the comments
say. The term "boxcar" means that the number of samples used
to calculate stays constant, with new samples replacing the
oldest ones. The queue holdtime calculation uses all holdtime
samples collected since the queue was loaded, so the comment
has been changed to be accurate.

(closes issue #12781)
Reported by: davidw


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2008-06-17 15:58:21 +00:00
Russell Bryant
020dce48e9 Merged revisions 123272 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r123272 | russell | 2008-06-17 10:52:13 -0500 (Tue, 17 Jun 2008) | 12 lines

Merged revisions 123271 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r123271 | russell | 2008-06-17 10:48:31 -0500 (Tue, 17 Jun 2008) | 4 lines

Fix a memory leak in astobj2 that was pointed out by seanbright.  When a container
got destroyed, the underlying bucket list entry for each object that was in the
container at that time did not get free'd.

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2008-06-17 15:52:36 +00:00
Tilghman Lesher
0d3b20487c Blocked revisions 123201 via svnmerge
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r123201 | tilghman | 2008-06-16 16:31:43 -0500 (Mon, 16 Jun 2008) | 2 lines

Oopsie, breakage

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2008-06-16 21:33:14 +00:00
Jeff Peeler
2248db67c7 Fix some variables that were forgotten to be renamed, related to 117658. Couldn't merge from trunk since the chan_dahdi transition has not occurred here yet
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@123178 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-16 21:20:08 +00:00
Chris Tooley
447203ca39 Blocked revisions 123115 via svnmerge
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r123115 | ctooley | 2008-06-16 15:02:06 -0500 (Mon, 16 Jun 2008) | 1 line

Changes response to the ExternalIVR() P command from pipe delimited to comma delimited.  closes issue #12804
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2008-06-16 21:20:07 +00:00
Steve Murphy
3a1b062a16 Merged revisions 123165 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r123165 | murf | 2008-06-16 14:43:46 -0600 (Mon, 16 Jun 2008) | 19 lines

(closes issue #12689)
Reported by: ys

Many thanks to ys for doing the research on this problem.
I didn't think it would be best to unlock the contexts
and then relock them after the remove_extension2() call,
so I added an extra arg to remove_extension2() and set it
appropriately in each call. There were not that many.

I considered forcing the code to lock the contexts before
the call to remove_extension2(), but that would require
a slightly greater degree of changes, especially since
the find_context_locked is local to pbx.c

I did a simple sanity test to make sure the code doesn't
mess things up in general.



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2008-06-16 21:19:23 +00:00
Jeff Peeler
281411716a Blocked revisions 123166 via svnmerge
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r123166 | jpeeler | 2008-06-16 16:15:12 -0500 (Mon, 16 Jun 2008) | 1 line

Fix some variables that were forgotten to be renamed, related to 117658
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2008-06-16 21:18:28 +00:00
Tilghman Lesher
0c73588085 Merged revisions 123114 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r123114 | tilghman | 2008-06-16 14:57:05 -0500 (Mon, 16 Jun 2008) | 10 lines

Merged revisions 123113 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r123113 | tilghman | 2008-06-16 14:50:12 -0500 (Mon, 16 Jun 2008) | 2 lines

Port "hasvoicemail" change from SIP to other channel drivers

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2008-06-16 20:03:18 +00:00
Tilghman Lesher
acd50c01bc Merged revisions 123111 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r123111 | tilghman | 2008-06-16 14:23:51 -0500 (Mon, 16 Jun 2008) | 16 lines

Merged revisions 123110 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r123110 | tilghman | 2008-06-16 14:21:58 -0500 (Mon, 16 Jun 2008) | 8 lines

People expect that if "hasvoicemail" is set in users.conf, even if "mailbox"
isn't set, that SIP will detect a mailbox.
(closes issue #12855)
 Reported by: PLL
 Patches: 
       20080614__bug12855__2.diff.txt uploaded by Corydon76 (license 14)
 Tested by: PLL

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2008-06-16 19:24:44 +00:00
Sean Bright
003cbe46dc Blocked revisions 123076 via svnmerge
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r123076 | seanbright | 2008-06-16 13:33:10 -0400 (Mon, 16 Jun 2008) | 1 line

Last commit for a bit, minor cleanups and move the lock initialization.
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2008-06-16 17:33:52 +00:00
Chris Tooley
a89916022e Fixes and closes bug number 12804
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@123075 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-16 17:29:06 +00:00
Sean Bright
ff61c559ea Blocked revisions 123044 via svnmerge
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r123044 | seanbright | 2008-06-16 13:14:11 -0400 (Mon, 16 Jun 2008) | 1 line

Convert to use stringfields.  Still some more work to do on config load/reload.
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2008-06-16 17:14:47 +00:00
Sean Bright
20a0553d25 Blocked revisions 122834 via svnmerge
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r122834 | seanbright | 2008-06-15 23:33:03 -0400 (Sun, 15 Jun 2008) | 1 line

Resurrected app_fax
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2008-06-16 17:09:14 +00:00
Sean Bright
c3d4dd7fff Blocked revisions 123009,123041 via svnmerge
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r123009 | seanbright | 2008-06-16 11:25:03 -0400 (Mon, 16 Jun 2008) | 1 line

Coding guidelines stuff only.
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r123041 | seanbright | 2008-06-16 12:29:18 -0400 (Mon, 16 Jun 2008) | 1 line

Remove some unused variables
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2008-06-16 17:07:01 +00:00
Russell Bryant
648a3aea44 Blocked revisions 122977 via svnmerge
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r122977 | russell | 2008-06-16 08:31:36 -0500 (Mon, 16 Jun 2008) | 2 lines

Note that only one timing interface should get loaded.

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2008-06-16 13:31:49 +00:00
Russell Bryant
8c0488cb30 Blocked revisions 122926 via svnmerge
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r122926 | russell | 2008-06-16 08:03:40 -0500 (Mon, 16 Jun 2008) | 4 lines

Add a "timing test" CLI command.  It opens a timer and configures it for
50 ticks per second, and then counts to see how many ticks it actually
gets in a second.

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2008-06-16 13:03:58 +00:00
Russell Bryant
5429dcc618 Blocked revisions 122923 via svnmerge
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r122923 | russell | 2008-06-16 07:48:11 -0500 (Mon, 16 Jun 2008) | 5 lines

 - Fix a typo in a timing API call
 - Convert the last part of channel.c over to use the timing API.  This would
   not have made a difference when using the dahdi timing module.  I noticed
   it when trying to use another timing source.  Oops.  :)

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2008-06-16 12:49:00 +00:00
Joshua Colp
caedf75d2c Merged revisions 122920 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r122920 | file | 2008-06-16 09:32:02 -0300 (Mon, 16 Jun 2008) | 14 lines

Merged revisions 122919 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r122919 | file | 2008-06-16 09:31:09 -0300 (Mon, 16 Jun 2008) | 6 lines

Only compare the first 15 characters so that even if the charset is specified we still accept it as SDP.
(closes issue #12803)
Reported by: lanzaandrea
Patches:
      chan_sip.c.diff uploaded by lanzaandrea (license 496)

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2008-06-16 12:32:56 +00:00
Joshua Colp
eb2df0bf9e Merged revisions 122870 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r122870 | file | 2008-06-16 09:09:54 -0300 (Mon, 16 Jun 2008) | 14 lines

Merged revisions 122869 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r122869 | file | 2008-06-16 09:08:28 -0300 (Mon, 16 Jun 2008) | 6 lines

Don't send a BYE on a dialog that is already gone during a REFER.
(closes issue #12865)
Reported by: flefoll
Patches:
      chan_sip.c.br14.121495.patch-ALREADYGONE uploaded by flefoll (license 244)

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2008-06-16 12:10:50 +00:00
Tilghman Lesher
7566296c88 Blocked revisions 122802 via svnmerge
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r122802 | tilghman | 2008-06-15 10:21:16 -0500 (Sun, 15 Jun 2008) | 8 lines

Add some more IAX2-specific information about the channel to the CHANNEL()
function and begin the transition from SIPCHANINFO() to just using CHANNEL().
(closes issue #12856)
 Reported by: mostyn
 Patches: 
       iax_and_sip_channel_info.patch uploaded by mostyn (license 398)
       (with some additional cleanup by me)

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2008-06-15 15:22:39 +00:00
Tilghman Lesher
6e4ad86b78 Blocked revisions 122766 via svnmerge
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r122766 | tilghman | 2008-06-13 17:52:20 -0500 (Fri, 13 Jun 2008) | 2 lines

Document the input for ast_realtime_require_field()

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2008-06-13 22:53:19 +00:00
Tilghman Lesher
ce015bd721 Blocked revisions 122716 via svnmerge
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r122716 | tilghman | 2008-06-13 16:50:28 -0500 (Fri, 13 Jun 2008) | 2 lines

Properly detect the size of char/varchar fields

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2008-06-13 22:01:34 +00:00
Mark Michelson
8940e86621 Merged revisions 122714 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r122714 | mmichelson | 2008-06-13 16:45:21 -0500 (Fri, 13 Jun 2008) | 17 lines

Merged revisions 122713 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r122713 | mmichelson | 2008-06-13 16:44:53 -0500 (Fri, 13 Jun 2008) | 9 lines

Short circuit the loop in autoservice_run if there are no channels to poll.
If we continued, then the result would be calling poll() with a NULL
pollfd array. While this is fine with POSIX's poll(2) system call, those
who use Asterisk's internal poll mechanism (Darwin systems) would have
a failed assertion occur when poll is called.

(related to issue #10342)


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2008-06-13 21:47:53 +00:00
Tilghman Lesher
2a666f21b4 Merged revisions 122557 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r122557 | tilghman | 2008-06-13 09:15:07 -0500 (Fri, 13 Jun 2008) | 7 lines

Convert one more delimiter to use comma.
(closes issue #12850)
 Reported by: bcnit
 Patches: 
       20080613__bug12850.diff.txt uploaded by Corydon76 (license 14)
 Tested by: bcnit

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2008-06-13 14:15:56 +00:00
Russell Bryant
33bb0de90b Blocked revisions 122526 via svnmerge
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r122526 | russell | 2008-06-13 07:53:08 -0500 (Fri, 13 Jun 2008) | 3 lines

Do not allow res_timing_dahdi to be unloaded.  We can re-enable this once we
add automatic use count handling for timing modules.

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2008-06-13 12:53:42 +00:00
Russell Bryant
684ca6bc77 Blocked revisions 122523 via svnmerge
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r122523 | russell | 2008-06-13 07:45:50 -0500 (Fri, 13 Jun 2008) | 11 lines

Merge changes from timing branch

 - Convert chan_iax2 to use the timing API
 - Convert usage of timing in the core to use the timing API instead of
   using DAHDI directly
 - Make a change to the timing API to add the set_rate() function
 - change the timing core to use a rwlock
 - merge a timing implementation, res_timing_dahdi

Basic testing was successful using res_timing_dahdi

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2008-06-13 12:46:16 +00:00
Michiel van Baak
53e887f56b Blocked revisions 122493 via svnmerge
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r122493 | mvanbaak | 2008-06-13 13:20:47 +0200 (Fri, 13 Jun 2008) | 9 lines

Implement call parking in chan_skinny.

(closes issue #11342)
Reported by: DEA
Patches:
      chan_skinny-park.txt uploaded by DEA (license 3)
      chan_skinny-park-v2.diff.txt uploaded by mvanbaak (license 7)
Tested by: DEA, mvanbaak

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2008-06-13 11:21:57 +00:00
Jeff Peeler
c902b29042 Merged revisions 122433 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r122433 | jpeeler | 2008-06-12 18:08:37 -0500 (Thu, 12 Jun 2008) | 4 lines

(closes issue 0012193)
Reported by: davidw
Patch by: Corydon76, modified by me to work properly with ParkAndAnnounce app

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2008-06-13 00:18:02 +00:00
Mark Michelson
541cd0a771 Blocked revisions 122461 via svnmerge
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r122461 | mmichelson | 2008-06-12 18:58:01 -0500 (Thu, 12 Jun 2008) | 5 lines

Fix a segfault by not trying to store a stack address for
long-term use. Instead use the heap. I can't believe this
never happened *once* in my developer branch when I was testing.


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2008-06-12 23:58:26 +00:00
Mark Michelson
b38ba0d12d Blocked revisions 122399 via svnmerge
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r122399 | mmichelson | 2008-06-12 16:23:02 -0500 (Thu, 12 Jun 2008) | 4 lines

Recommitting revision 122228, which was accidentally reverted
as a result of commit 122234.


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2008-06-12 21:23:47 +00:00
Russell Bryant
63dd30cfb2 Blocked revisions 122371 via svnmerge
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r122371 | russell | 2008-06-12 15:38:52 -0500 (Thu, 12 Jun 2008) | 2 lines

Complete the documentation for the timing API.

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2008-06-12 20:39:14 +00:00
Mark Michelson
d8873c2856 Merged revisions 122312 via svnmerge from
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r122312 | mmichelson | 2008-06-12 13:53:17 -0500 (Thu, 12 Jun 2008) | 17 lines

Merged revisions 122311 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r122311 | mmichelson | 2008-06-12 13:50:58 -0500 (Thu, 12 Jun 2008) | 9 lines

Properly play a holdtime message if the announce-holdtime option is
set to "once."

(closes issue #12842)
Reported by: ramonpeek
Patches:
      patch001.diff uploaded by ramonpeek (license 266)


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2008-06-12 18:54:01 +00:00
Russell Bryant
c1bc5f317e Merged revisions 122262 via svnmerge from
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r122262 | russell | 2008-06-12 13:23:54 -0500 (Thu, 12 Jun 2008) | 11 lines

Merged revisions 122259 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r122259 | russell | 2008-06-12 13:22:44 -0500 (Thu, 12 Jun 2008) | 3 lines

Fix some race conditions that cause ast_assert() to report that chan_iax2 tried
to remove an entry that wasn't in the scheduler

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2008-06-12 18:24:33 +00:00
Russell Bryant
f31433df6a Blocked revisions 122241 via svnmerge
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r122241 | russell | 2008-06-12 12:38:27 -0500 (Thu, 12 Jun 2008) | 3 lines

Get default entity ID determination working on Linux again
(closes issue #12839)

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2008-06-12 17:38:46 +00:00
Mark Michelson
192213cb84 Blocked revisions 122228 via svnmerge
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r122228 | mmichelson | 2008-06-12 11:25:09 -0500 (Thu, 12 Jun 2008) | 12 lines

Merging the work done in the queue-log-atxfer branch. The
net result of this work is that attended transfers made
by queue members will now show up in the queue_log as a 
TRANSFER message instead of COMPLETECALLER as it had been.

As far as the details go, I created a datastore which is
attached to the calling channel just prior to when the caller
is bridged with the queue member. If the calling channel
is masqueraded, then during the "fixup" portion, the TRANSFER
will be logged and the datastore will be removed.


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2008-06-12 16:28:14 +00:00
Tilghman Lesher
a435acefbe Merged revisions 122174 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r122174 | tilghman | 2008-06-12 10:26:07 -0500 (Thu, 12 Jun 2008) | 16 lines

Merged revisions 122137 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r122137 | tilghman | 2008-06-12 10:18:39 -0500 (Thu, 12 Jun 2008) | 8 lines

Flipflop the sections for two options, since the section for 'X' (exit context)
may otherwise absorb keypresses meant for 's' (admin/user menu).
(closes issue #12836)
 Reported by: blitzrage
 Patches: 
       20080611__bug12836.diff.txt uploaded by Corydon76 (license 14)
 Tested by: blitzrage

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2008-06-12 15:27:34 +00:00
Tilghman Lesher
24df44d002 Merged revisions 122131 via svnmerge from
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r122131 | tilghman | 2008-06-12 10:14:37 -0500 (Thu, 12 Jun 2008) | 12 lines

Merged revisions 122130 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r122130 | tilghman | 2008-06-12 10:11:30 -0500 (Thu, 12 Jun 2008) | 4 lines

Occasionally, the alertpipe loses its nonblocking status, so detect and correct
that situation before it causes a deadlock.  (Reported and tested by ctooley
via #asterisk-dev)

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2008-06-12 15:15:25 +00:00
Steve Murphy
8230420842 Merged revisions 122128 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r122128 | murf | 2008-06-12 08:56:26 -0600 (Thu, 12 Jun 2008) | 9 lines

Merged revisions 122127 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r122127 | murf | 2008-06-12 08:51:44 -0600 (Thu, 12 Jun 2008) | 1 line

Arkadia tried to warn me, but the code added to ast_cdr_busy, _failed, and _noanswer was redundant. Didn't spot it until I was resolving conflicts in trunk. Ugh. Redundant code removed. It wasn't harmful. Just dumb.
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2008-06-12 15:01:34 +00:00
Steve Murphy
ebdf1fea15 Merged revisions 122091 via svnmerge from
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r122091 | murf | 2008-06-12 08:28:01 -0600 (Thu, 12 Jun 2008) | 45 lines

Merged revisions 122046 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r122046 | murf | 2008-06-12 07:47:34 -0600 (Thu, 12 Jun 2008) | 37 lines

(closes issue #10668)
Reported by: arkadia
Tested by: murf, arkadia

Options added to forkCDR() app and the CDR() func to
remove some roadblocks for CDR applications.

The "show application ForkCDR" output was upgraded
to more fully explain the inner workings of forkCDR.

The A option was added to forkCDR to force the
CDR system to NOT change the disposition on the
original CDR, after the fork. This involves
ast_cdr_answer, _busy, _failed, and so on.

The T option was added to forkCDR to force 
obedience of the cdr LOCKED flag in the
ast_cdr_end, all the disposition changing
funcs (ast_cdr_answer, etc), and in the
ast_cdr_setvar func.

The CHANGES file was updated to explain ALL
the new options added to satisfy this bug report
(and some requests made verbally and via 
email, irc, etc, over the past months/year)

The 's' option was added to the CDR() func,
to force it to skip LOCKED cdr's in the
chain.

Again, the new options should be totally transparent
to existing apps! Current behavior of CDR,
forkCDR, and the rest of the CDR system should
not change one little bit. Until you add the
new options, at least!


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2008-06-12 14:38:46 +00:00
Russell Bryant
bc2cdb7766 Blocked revisions 122047 via svnmerge
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r122047 | russell | 2008-06-12 09:06:38 -0500 (Thu, 12 Jun 2008) | 4 lines

Don't log not being able to set a default EID.  Most people don't care, and
those that do can check their setup using CLI commands.
(closes issue #12839)

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2008-06-12 14:07:10 +00:00
Terry Wilson
53c23298db Blocked revisions 121955 via svnmerge
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r121955 | twilson | 2008-06-11 16:38:40 -0500 (Wed, 11 Jun 2008) | 2 lines

Initialize parkingtime to DEFAULT_PARK_TIME instead of 0

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2008-06-11 21:42:17 +00:00
Matthew Fredrickson
a7e622b086 Merged revisions 121914 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r121914 | mattf | 2008-06-11 13:53:10 -0500 (Wed, 11 Jun 2008) | 1 line

Fix pseudo channel allocation errors on startup when using SS7
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2008-06-11 18:57:15 +00:00
Tilghman Lesher
ec5a235672 Merged revisions 121867 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r121867 | tilghman | 2008-06-11 13:19:24 -0500 (Wed, 11 Jun 2008) | 11 lines

Merged revisions 121861 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r121861 | tilghman | 2008-06-11 13:18:16 -0500 (Wed, 11 Jun 2008) | 3 lines

Make calls to ast_assert() actually test something, so that the error message
printed is not nonsensical (reported by mvanbaak via #asterisk-bugs).

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2008-06-11 18:20:18 +00:00
Matthew Fredrickson
53a3681770 Merged revisions 121857 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r121857 | mattf | 2008-06-11 12:50:17 -0500 (Wed, 11 Jun 2008) | 1 line

Make sure we hangup any calls we have and NULL out the ss7call value when we get a reset circuit message.  Fixes crash bug
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2008-06-11 17:59:43 +00:00
Tilghman Lesher
405f86563f Merged revisions 121855 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r121855 | tilghman | 2008-06-11 12:44:39 -0500 (Wed, 11 Jun 2008) | 3 lines

Expand CDR uniqueid field to 150 chars, to account for maximum systemname.
(Closes issue #12831)

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2008-06-11 17:45:37 +00:00
Jeff Peeler
8f932db851 Merged revisions 121805 via svnmerge from
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r121805 | jpeeler | 2008-06-11 11:11:40 -0500 (Wed, 11 Jun 2008) | 9 lines

Merged revisions 121804 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r121804 | jpeeler | 2008-06-11 11:11:09 -0500 (Wed, 11 Jun 2008) | 1 line

add instructions for logging gdb output via set logging on
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2008-06-11 16:13:00 +00:00
Tilghman Lesher
4ac89d3ca2 Blocked revisions 121683 via svnmerge
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r121683 | tilghman | 2008-06-10 16:14:58 -0500 (Tue, 10 Jun 2008) | 4 lines

Move the table cache routines to res_odbc, so they can be used from other
places (app_voicemail, for example).
(Related to bug #11678)

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2008-06-10 21:15:57 +00:00
Mark Michelson
0f451254a8 Blocked revisions 121649 via svnmerge
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r121649 | mmichelson | 2008-06-10 14:52:21 -0500 (Tue, 10 Jun 2008) | 4 lines

Add an additional sanity check in case an event is passed between 
Asterisk boxes with mismatched ie_maps.


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2008-06-10 19:53:22 +00:00
Donny Kavanagh
cacafd718b Blocked revisions 121599 via svnmerge
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r121599 | juggie | 2008-06-10 15:03:11 -0400 (Tue, 10 Jun 2008) | 2 lines

Revision 117802 changed frame.data to frame.data.ptr however codec_ilbc.c was not updated.  This resolves that oversight.

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2008-06-10 19:06:53 +00:00
Sean Bright
291f2c5b0b Merged revisions 121597 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r121597 | seanbright | 2008-06-10 14:35:37 -0400 (Tue, 10 Jun 2008) | 14 lines

Merged revisions 121596 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r121596 | seanbright | 2008-06-10 14:34:45 -0400 (Tue, 10 Jun 2008) | 6 lines

Fixes a problem with some buggy versions of GNU awk (3.1.3) not liking carriage returns in scripts.

(closes issue #12749)
Reported by: alinux
Tested by: Laureano (on #asterisk-dev), juggie

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2008-06-10 18:36:29 +00:00
Russell Bryant
e769bad031 Blocked revisions 121559 via svnmerge
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r121559 | russell | 2008-06-10 10:12:17 -0500 (Tue, 10 Jun 2008) | 24 lines

Merge another big set of changes from team/russell/events

This commit merges in the rest of the code needed to support distributed device
state.  There are two main parts to this commit.

Core changes:
 - The device state handling in the core has been updated to understand device
   state across a cluster of Asterisk servers.  Every time the state of a device
   changes, it looks at all of the device states on each node, and determines the
   aggregate device state.  That resulting device state is what is provided to
   modules in Asterisk that take actions based on the state of a device.

New module, res_ais:
 - A module has been written to facilitate the communication of events between
   nodes in a cluster of Asterisk servers.  This module uses the SAForum AIS
   (Service Availability Forum Application Interface Specification) CLM and EVT
   services (Cluster Management and Event) to handle this task.  This module
   currently supports sharing Voicemail MWI (Message Waiting Indication) and
   device state events between servers.  It has been tested with openais, though
   other implementations of the spec do exist.

For more information on testing distributed device state, see the following doc:
  - doc/distributed_devstate.txt

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Russell Bryant
da5b26e1c5 Blocked revisions 121555 via svnmerge
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r121555 | russell | 2008-06-10 09:53:40 -0500 (Tue, 10 Jun 2008) | 22 lines

Merge some more changes from team/russell/events

This commit pulls in a batch of improvements and additions to the event API.
Changes include:
 - the ability to dynamically build a subscription.  This is useful if you're
    building a subscription based on something you receive from the network,
    or from options in a configuration file.
 - Add tables of event types and IE types and the corresponding string
    representation for implementing text based protocols that use these
	events, for showing events on the CLI, reading configuration that
	references event information, among other things.
 - Add a table that maps IE types and the corresponding payload type.
 - an API call to get the total size of an event
 - an API call to get all events from the cache that match a subscription
 - a new IE payload type, raw, which I used for transporting the Entity ID in
    my code for handling distributed device state.
 - Code improvements to reduce code duplication
 - Include the Entity ID of the server that originated the event in every event
 - an additional event type, DEVICE_STATE_CHANGE, to help facilitate distributed
    device state.  DEVICE_STATE is a state change on one server, DEVICE_STATE_CHANGE
	is the aggregate device state change across all servers.

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Joshua Colp
2494fa613b git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@121505 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2008-06-10 14:12:51 +00:00
Russell Bryant
e48336cb97 Blocked revisions 121501 via svnmerge
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r121501 | russell | 2008-06-10 09:06:29 -0500 (Tue, 10 Jun 2008) | 9 lines

Merge another change from team/russell/events

This commit breaks out some logic from pbx.c into a simple API.  The hint
processing code had logic for taking the state from multiple devices and
turning that into the state for a single extension.  So, I broke this out
and made an API that lets you take multiple device states and determine
the aggregate device state.  I needed this for some core device state changes
to support distributed device state.

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2008-06-10 14:06:47 +00:00
Joshua Colp
987aadc097 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@121498 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2008-06-10 13:38:40 +00:00
Joshua Colp
3c7c23d269 Merged revisions 121444 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r121444 | file | 2008-06-10 09:54:39 -0300 (Tue, 10 Jun 2008) | 12 lines

Merged revisions 121442 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r121442 | file | 2008-06-10 09:52:06 -0300 (Tue, 10 Jun 2008) | 4 lines

Update BRIDGEPEER variable before we do a generic bridge in case we just broke out of a native bridge and fell through to generic.
(closes issue #12815)
Reported by: ramonpeek

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2008-06-10 12:55:24 +00:00
Russell Bryant
255ef74632 Blocked revisions 121439 via svnmerge
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r121439 | russell | 2008-06-10 07:48:50 -0500 (Tue, 10 Jun 2008) | 13 lines

Merge another change from team/russell/events ...

DUNDi uses a concept called the Entity ID for unique server identifiers.  I have
pulled out the handling of EIDs and made it something available to all of Asterisk.
There is now a global Entity ID that can be used for other purposes as well, such
as code providing distributed device state, which is why I did this.  The global
Entity ID is set automatically, just like it was done in DUNDi, but it can also be
set in asterisk.conf.  DUNDi will now use this global EID unless one is specified
in dundi.conf.

The current EID for the system can be seen in the "core show settings" CLI command.
It is also available in the dialplan via the ENTITYID variable.

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2008-06-10 12:49:09 +00:00
Tilghman Lesher
6e7b2fc321 Blocked revisions 121367 via svnmerge
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r121367 | tilghman | 2008-06-09 17:51:59 -0500 (Mon, 09 Jun 2008) | 2 lines

Expand RQ_INTEGER type out to multiple types, one for each precision

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2008-06-10 03:44:32 +00:00
Russell Bryant
1b8dba640e Merged revisions 121407 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r121407 | russell | 2008-06-09 19:52:46 -0500 (Mon, 09 Jun 2008) | 2 lines

Bump up the debug level of a couple of messages

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2008-06-10 00:53:04 +00:00
Russell Bryant
0588f43631 Blocked revisions 121403 via svnmerge
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r121403 | russell | 2008-06-09 19:43:06 -0500 (Mon, 09 Jun 2008) | 4 lines

Merge a couple of configure script checks in from team/russell/events.  This adds
the checks for the CLM and EVT services from the SAForum AIS.  I'm going to work
on merging in changes from this branch in pieces.

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2008-06-10 00:43:40 +00:00
Terry Wilson
cd38c5937b Blocked revisions 121365 via svnmerge
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r121365 | twilson | 2008-06-09 17:42:08 -0500 (Mon, 09 Jun 2008) | 2 lines

Initialize the lock and destroy lock and cond in the destructor (thanks, mmichelson)

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2008-06-09 22:47:27 +00:00
Tilghman Lesher
aeeaf593b1 Blocked revisions 121334 via svnmerge
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r121334 | tilghman | 2008-06-09 14:33:23 -0500 (Mon, 09 Jun 2008) | 3 lines

Add storage of the useragent in the realtime database.
(Closes AST-38)

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2008-06-09 19:34:04 +00:00
Russell Bryant
3abaf28d74 Merged revisions 121282 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r121282 | russell | 2008-06-09 11:37:08 -0500 (Mon, 09 Jun 2008) | 18 lines

Merged revisions 121280 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r121280 | russell | 2008-06-09 11:35:40 -0500 (Mon, 09 Jun 2008) | 10 lines

Do not attempt to do emulation if an END digit is received and the length is
less than the defined minimum digit length, and the other end only wants END
digits (SIP INFO, for example).

(closes issue #12778)
Reported by: tsearle
Patches:
      12778.rev1.txt uploaded by russell (license 2)
Tested by: tsearle

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2008-06-09 16:37:36 +00:00
Tilghman Lesher
d623090d7e Merged revisions 121279 via svnmerge from
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r121279 | tilghman | 2008-06-09 11:35:06 -0500 (Mon, 09 Jun 2008) | 6 lines

Implement FINDLABEL matching for the new extension matching engine.
(closes issue #12800)
 Reported by: chris-mac
 Patches: 
       20080608__bug12800.diff.txt uploaded by Corydon76 (license 14)

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2008-06-09 16:36:24 +00:00
Mark Michelson
51638f5aa6 Merged revisions 121230 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r121230 | mmichelson | 2008-06-09 10:08:58 -0500 (Mon, 09 Jun 2008) | 27 lines

Merged revisions 121229 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

(Note that this is being merged to trunk/1.6.0 because
it may affect non-callback agents with ackcall set)

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r121229 | mmichelson | 2008-06-09 10:02:37 -0500 (Mon, 09 Jun 2008) | 16 lines

A unique situation of timeouts brought forth a failure situation for
autologoff in chan_agent. If using AgentCallbackLogin-style agents, 
then if the timeout specified by the Dial() to reach the agent's phone
was shorter than the timeout specified in queues.conf, then autologoff
would only work if the caller hung up while the agent's phone was ringing.
This patch allows autologoff to work in this situation when the call in
queue transfers to the next available agent (as it would have if the timeout
in queues.conf were less than the timeout in the Dial()).

(closes issue #12754)
Reported by: Rodrigo
Patches:
      12754.patch uploaded by putnopvut (license 60)
Tested by: Rodrigo


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2008-06-09 15:10:08 +00:00
Michiel van Baak
63d35b9c38 Blocked revisions 121197 via svnmerge
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r121197 | mvanbaak | 2008-06-08 13:40:44 +0200 (Sun, 08 Jun 2008) | 12 lines

add a new argument to PrivacyManager to specify a context
where the entered phone number is checked.

You can now define a set of extensions/exten patterns that describe
valid phone numbers. PrivacyManager will check that context for a match
with the given phone number.
This way you get better control. For example people blindly hitting
10 digits just to get past privacymanager

Example line in extensions.conf:
exten => incoming,n,PrivacyManager(3,10,route-outgoing)

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2008-06-08 11:42:28 +00:00
Jeff Peeler
3e98181722 Merged revisions 121163 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r121163 | jpeeler | 2008-06-07 20:41:59 -0500 (Sat, 07 Jun 2008) | 4 lines

This was accidentally reverted.
Fixes a bug where if a stream monitor thread was not created (caused from failure of opening or starting the stream) pthread_cancel was called with an invalid thread ID.


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2008-06-08 01:43:16 +00:00
Jeff Peeler
5680387bea Merged revisions 121131 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r121131 | jpeeler | 2008-06-07 20:16:25 -0500 (Sat, 07 Jun 2008) | 2 lines

Fixes segfault when using ParkAndAnnounce. Also, loop made more efficient as announce template only needs to be checked until the number of colon separated arguments run out, not the entire pointer storage array. Was done in a similiar fashion in 1.4, but here we're using less variables.

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2008-06-08 01:18:52 +00:00
Russell Bryant
822355271d Merged revisions 121079 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r121079 | russell | 2008-06-07 09:18:44 -0500 (Sat, 07 Jun 2008) | 15 lines

Merged revisions 121078 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r121078 | russell | 2008-06-07 09:10:56 -0500 (Sat, 07 Jun 2008) | 7 lines

Don't run LIST_HEAD_DESTROY on a STATIC list

(closes issue #12807)
Reported by: ys
Patches: 
      chan_agent_local.diff uploaded by ys (license 281)

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2008-06-07 14:19:41 +00:00
Tilghman Lesher
0b8cef4b18 Blocked revisions 121042 via svnmerge
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r121042 | tilghman | 2008-06-06 15:24:11 -0500 (Fri, 06 Jun 2008) | 6 lines

Added a facility for sending arbitrary SIP notify commands from AMI.
(closes issue #12562)
 Reported by: michael-fig
 Patches: 
       20080515__bug12562.diff.txt uploaded by Corydon76 (license 14)

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2008-06-06 20:25:20 +00:00
Tilghman Lesher
9a4ef078a4 Merged revisions 121010 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r121010 | tilghman | 2008-06-06 14:55:08 -0500 (Fri, 06 Jun 2008) | 6 lines

Make extension match characters case-insensitive.
(closes issue #12777)
 Reported by: jsmith
 Patches: 
       lower_case_patterns-trunk-v1.patch uploaded by jsmith (license 15)

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2008-06-06 19:56:08 +00:00
Jeff Peeler
91945123a2 Merged revisions 120960 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r120960 | jpeeler | 2008-06-06 13:30:17 -0500 (Fri, 06 Jun 2008) | 9 lines

Merged revisions 120959 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r120959 | jpeeler | 2008-06-06 13:29:14 -0500 (Fri, 06 Jun 2008) | 1 line

add another LOW_MEMORY define I forgot
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2008-06-06 18:31:31 +00:00
Jeff Peeler
cd601b2b44 Merged revisions 120909 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r120909 | jpeeler | 2008-06-06 13:06:06 -0500 (Fri, 06 Jun 2008) | 9 lines

Merged revisions 120908 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r120908 | jpeeler | 2008-06-06 13:05:15 -0500 (Fri, 06 Jun 2008) | 1 line

only define thread storage variable if necessary for LOW_MEMORY
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2008-06-06 18:13:31 +00:00
Jeff Peeler
53aef9f034 Merged revisions 120906 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r120906 | jpeeler | 2008-06-06 12:50:05 -0500 (Fri, 06 Jun 2008) | 16 lines

Merged revisions 120863,120885 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r120863 | jpeeler | 2008-06-06 10:33:15 -0500 (Fri, 06 Jun 2008) | 3 lines

This fixes a crash when LOW_MEMORY is turned on. Two allocations of the ast_rtp struct that were previously allocated on the stack have been modified to use thread local storage instead.


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r120885 | jpeeler | 2008-06-06 11:39:20 -0500 (Fri, 06 Jun 2008) | 2 lines

Correction to commmit 120863, make sure proper destructor function is called as well define two thread storage local variables.

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2008-06-06 18:02:12 +00:00
Tilghman Lesher
76bce983e2 Merged revisions 120904 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r120904 | tilghman | 2008-06-06 12:34:21 -0500 (Fri, 06 Jun 2008) | 3 lines

For the purpose of making the changed syntax to ExecIf easier to transition,
allow the deprecated syntax (fixed for jmls on -dev).

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2008-06-06 17:35:03 +00:00
Tilghman Lesher
b636496aa1 Blocked revisions 120789 via svnmerge
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r120789 | tilghman | 2008-06-05 14:07:27 -0500 (Thu, 05 Jun 2008) | 3 lines

Merge the adaptive realtime branch, which will make adding new required fields
to realtime less painful in the future.

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2008-06-06 15:39:29 +00:00
Steve Murphy
c24bc84355 Merged revisions 120828 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r120828 | murf | 2008-06-05 15:34:42 -0600 (Thu, 05 Jun 2008) | 1 line

a small fix for a crash that occurs when compiling AEL with global vars
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2008-06-05 21:39:24 +00:00
Philippe Sultan
8c1b4ef44d Merged revisions 120676 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r120676 | phsultan | 2008-06-05 19:02:39 +0200 (Thu, 05 Jun 2008) | 10 lines

Merged revisions 120675 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r120675 | phsultan | 2008-06-05 18:56:15 +0200 (Thu, 05 Jun 2008) | 2 lines

Ignore appended resource when comparing JIDs.

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2008-06-05 17:17:28 +00:00
Brett Bryant
2c8228fdd2 Blocked revisions 120673 via svnmerge
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r120673 | bbryant | 2008-06-05 11:41:36 -0500 (Thu, 05 Jun 2008) | 1 line

Update CHANGES file for the things done in revision 120635.
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2008-06-05 16:42:00 +00:00
Brett Bryant
c745206a72 Blocked revisions 120635 via svnmerge
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r120635 | bbryant | 2008-06-05 11:24:19 -0500 (Thu, 05 Jun 2008) | 14 lines

This patch adds more detailed statistics for RTP channels, and provides an API call to access it, including maximums, minimums, standard deviatinos, 
and normal deviations. Currently this is implemented for chan_sip, but could be added to the func_channel_read callbacks for the CHANNEL function 
for any channel that uses RTP.

(closes issue #10590)
Reported by: gasparz
Patches:
      chan_sip_c.diff uploaded by gasparz (license 219)
      rtp_c.diff uploaded by gasparz (license 219)
      rtp_h.diff uploaded by gasparz (license 219)
      audioqos-trunk.diff uploaded by snuffy (license 35)
      rtpqos-trunk-r119891.diff uploaded by sergee (license 138)
Tested by: jsmith, gasparz, snuffy, marsosa, chappell, sergee

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2008-06-05 16:25:55 +00:00
Tilghman Lesher
ca20011d2c Merged revisions 120602 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r120602 | tilghman | 2008-06-05 10:58:11 -0500 (Thu, 05 Jun 2008) | 4 lines

Conditionally load the AGI command gosub, depending on whether or not res_agi
has been loaded, fix a return value in the loader, and ensure that the help
workhorse header does not print on load.

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2008-06-05 16:01:32 +00:00
Tilghman Lesher
c6dc1218fc Merged revisions 120567 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r120567 | tilghman | 2008-06-05 09:35:47 -0500 (Thu, 05 Jun 2008) | 2 lines

Add info on the [compat] section of asterisk.conf.

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2008-06-05 14:36:32 +00:00
Tilghman Lesher
c5bbfbbd1e Fix frame API for 1.6.0 (Closes issue #12793)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@120566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-05 12:42:44 +00:00
Mark Michelson
a067ee471a Merged revisions 120514 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r120514 | mmichelson | 2008-06-04 17:07:37 -0500 (Wed, 04 Jun 2008) | 14 lines

Merged revisions 120513 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r120513 | mmichelson | 2008-06-04 17:05:33 -0500 (Wed, 04 Jun 2008) | 6 lines

Make sure that the string we set will survive the unref of
the queue member.

Thanks to Russell, who pointed this out.


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2008-06-04 22:08:09 +00:00
Tilghman Lesher
816d355d89 Merged revisions 120477 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r120477 | tilghman | 2008-06-04 15:34:52 -0500 (Wed, 04 Jun 2008) | 2 lines

MSet doesn't necessarily need chan to be set

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2008-06-04 20:35:38 +00:00
Joshua Colp
faddb24782 Merged revisions 120337 via svnmerge from
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r120337 | file | 2008-06-04 12:38:00 -0300 (Wed, 04 Jun 2008) | 2 lines

We like tabs.

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2008-06-04 15:38:56 +00:00
Mark Michelson
d192aa7cca Merged revisions 120286 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r120286 | mmichelson | 2008-06-04 09:12:45 -0500 (Wed, 04 Jun 2008) | 15 lines

Merged revisions 120285 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r120285 | mmichelson | 2008-06-04 09:11:12 -0500 (Wed, 04 Jun 2008) | 7 lines

Tab completion when removing a member should give the member's interface,
not the name, since the interface is what is expected for the command.

(closes issue #12783)
Reported by: davevg


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2008-06-04 14:13:23 +00:00
Joshua Colp
8e26afa2c1 Merged revisions 120283 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r120283 | file | 2008-06-04 10:33:59 -0300 (Wed, 04 Jun 2008) | 14 lines

Merged revisions 120282 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r120282 | file | 2008-06-04 10:31:09 -0300 (Wed, 04 Jun 2008) | 6 lines

Fix a log message and add a message for when the dialplan is done reloading.
(closes issue #12716)
Reported by: chappell
Patches:
      dialplan_reload_2.diff uploaded by chappell (license 8)

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Tilghman Lesher
70ab525211 Blocked revisions 120230 via svnmerge
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r120230 | tilghman | 2008-06-03 18:17:33 -0500 (Tue, 03 Jun 2008) | 7 lines

Add a function, CHANNELS(), which retrieves a list of all active channels.
(closes issue #11330)
 Reported by: rain
 Patches: 
       func_channel-channel_list_function.diff uploaded by rain (license 327)
       (with some additional changes by me, mostly to meet coding guidelines)

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2008-06-03 23:18:11 +00:00
Tilghman Lesher
5183c444bb Merged revisions 120227 via svnmerge from
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r120227 | tilghman | 2008-06-03 17:42:03 -0500 (Tue, 03 Jun 2008) | 16 lines

Merged revisions 120226 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r120226 | tilghman | 2008-06-03 17:41:04 -0500 (Tue, 03 Jun 2008) | 8 lines

Due to incorrect use of the AST_LIST_INSERT_HEAD() macro the loopback switch
cannot perform any translation on the extension number before searching for it
in the target context.
(closes issue #12473)
 Reported by: chappell
 Patches: 
       pbx_loopback.c.diff uploaded by chappell (license 8)

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2008-06-03 22:42:40 +00:00
Jeff Peeler
4954c77f80 Merged revisions 120174 via svnmerge from
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r120174 | jpeeler | 2008-06-03 17:17:07 -0500 (Tue, 03 Jun 2008) | 14 lines

Merged revisions 120173 via svnmerge from 
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r120173 | jpeeler | 2008-06-03 17:15:33 -0500 (Tue, 03 Jun 2008) | 6 lines

(closes issue #11594)
Reported by: yem
Tested by: yem

This change decreases the buffer size allocated on the stack substantially in config_text_file_load when LOW_MEMORY is turned on. This change combined with the fix from revision 117462 (making mkintf not copy the zt_chan_conf structure) was enough to prevent the crash.

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Tilghman Lesher
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r120171 | tilghman | 2008-06-03 17:05:16 -0500 (Tue, 03 Jun 2008) | 5 lines

Move compatibility options into asterisk.conf, default them to on for upgrades,
and off for new installations.  This includes the translation from pipes to commas
for pbx_realtime and the EXEC command for AGI, as well as the change to the Set
application not to support multiple variables at once.

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2008-06-03 22:08:56 +00:00
Russell Bryant
ff438c5343 Merged revisions 120169 via svnmerge from
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r120169 | russell | 2008-06-03 16:35:11 -0500 (Tue, 03 Jun 2008) | 12 lines

Merged revisions 120168 via svnmerge from 
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r120168 | russell | 2008-06-03 16:34:55 -0500 (Tue, 03 Jun 2008) | 4 lines

Fix another place where peer->callno could change at a very bad time, and also
fix a place where a peer was used after the reference was released.
(inspired by rev 120001)

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Mark Michelson
b30c8019b1 Blocked revisions 120166 via svnmerge
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r120166 | mmichelson | 2008-06-03 16:22:52 -0500 (Tue, 03 Jun 2008) | 13 lines

Adding two new queue log events. The ADDMEMBER event is logged when
a dynamic realtime queue member is added to the queue, and the 
REMOVEMEMBER event is logged when a dynamic realtime member is
removed. Since no calling channel is associated with these events
the string "REALTIME" is placed where the channel's unique id is
normally placed.

(closes issue #12774)
Reported by: atis
Patches:
      queue_log_rt_members.patch uploaded by atis (license 242)


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2008-06-03 21:23:33 +00:00
Russell Bryant
0774aac0ef Blocked revisions 120129 via svnmerge
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r120129 | russell | 2008-06-03 14:48:37 -0500 (Tue, 03 Jun 2008) | 2 lines

Use proper return values for a few application modules

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Russell Bryant
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r120066 | russell | 2008-06-03 13:43:01 -0500 (Tue, 03 Jun 2008) | 2 lines

fix build for non debug threads

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2008-06-03 18:44:29 +00:00
Russell Bryant
a321695170 Blocked revisions 120064 via svnmerge
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r120064 | russell | 2008-06-03 13:26:51 -0500 (Tue, 03 Jun 2008) | 10 lines

Add lock tracking for rwlocks.  Previously, lock.h only had the ability to
hold tracking information for mutexes.  Now, the "core show locks" output
will output information about who is holding a rwlock when a thread is
waiting on it.

(closes issue #11279)
Reported by: ys
Patches:
      trunk_lock_utils.v8.diff uploaded by ys (license 281)

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2008-06-03 18:27:26 +00:00
Tilghman Lesher
09d6c17864 Merged revisions 120012 via svnmerge from
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r120012 | tilghman | 2008-06-03 11:19:35 -0500 (Tue, 03 Jun 2008) | 17 lines

Merged revisions 120001 via svnmerge from 
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r120001 | tilghman | 2008-06-03 11:10:53 -0500 (Tue, 03 Jun 2008) | 9 lines

Save the callno when we're poking, because our peer structure could change
during destruction (and thus we unlock the wrong callno, causing a
cascade failure).
(closes issue #12717)
 Reported by: gewfie
 Patches: 
       20080525__bug12717.diff.txt uploaded by Corydon76 (license 14)
 Tested by: gewfie

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2008-06-03 16:24:08 +00:00
Steve Murphy
a83e2b2672 Merged revisions 119998 via svnmerge from
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r119998 | murf | 2008-06-03 09:49:34 -0600 (Tue, 03 Jun 2008) | 16 lines

Merged revisions 119966 via svnmerge from 
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r119966 | murf | 2008-06-03 09:26:56 -0600 (Tue, 03 Jun 2008) | 8 lines

Updated the regressions on AEL. Hadn't updated
this for the changes I made to preserve ${EXTEN}
in switches, which affected several tests because
it adds extra priorities, and at least one needed to be updated
because of the removal of the empty extension warning
message.


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2008-06-03 15:57:26 +00:00
Steve Murphy
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r119930 | murf | 2008-06-03 09:07:20 -0600 (Tue, 03 Jun 2008) | 24 lines

Merged revisions 119929 via svnmerge from 
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r119929 | murf | 2008-06-03 08:49:46 -0600 (Tue, 03 Jun 2008) | 16 lines

as per http://lists.digium.com/pipermail/asterisk-users/2008-June/212934.html,
which is a message from Philipp Kempgen, requesting that the WARNING
that an extension is empty be reduced to a NOTICE or less, as empty
extensions are syntactically possible, and no big deal.

With which I agree, and have removed that WARNING message entirely.
I think it is not necessary to see this message. It didn't 
state that a NoOp() was inserted automatically on your behalf,
and really, as users, who cares? Why freak out dialplan writers
with unnecessary warnings? The details of the machinations a compiler goes
thru to produce working assembly code is of little interest
to most programmers-- we will follow the unix principal of
doing our work silently.



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2008-06-03 15:12:28 +00:00
Joshua Colp
5df551ca7c Merged revisions 119927 via svnmerge from
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r119927 | file | 2008-06-03 11:47:54 -0300 (Tue, 03 Jun 2008) | 10 lines

Merged revisions 119926 via svnmerge from 
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r119926 | file | 2008-06-03 11:46:24 -0300 (Tue, 03 Jun 2008) | 2 lines

Treat ECONNREFUSED as an error that will stop further retransmissions. (issue #AST-58, patch from Switchvox)

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2008-06-03 14:48:53 +00:00
Russell Bryant
f2374dc751 Merged revisions 119892 via svnmerge from
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r119892 | russell | 2008-06-03 08:29:16 -0500 (Tue, 03 Jun 2008) | 9 lines

Do a deep copy of file and function strings to avoid a potential crash when
modules are unloaded.

(closes issue #12780)
Reported by: ys
Patches:
      logger.diff uploaded by ys (license 281)
	  -- modified by me for coding guidelines

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2008-06-03 13:30:39 +00:00
Russell Bryant
3cbe2984ef Merged revisions 119839 via svnmerge from
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r119839 | russell | 2008-06-02 15:08:24 -0500 (Mon, 02 Jun 2008) | 15 lines

Merged revisions 119838 via svnmerge from 
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r119838 | russell | 2008-06-02 15:08:04 -0500 (Mon, 02 Jun 2008) | 7 lines

Revert a change made for issue #12479.  This change caused a regression such that
a dial string such as (IAX2/foo) did not automatically fall back to dialing the 's'
extension anymore.

(closes issue #12770)
Reported by: dagmoller

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2008-06-02 20:08:48 +00:00
Russell Bryant
99982db334 Merged revisions 119801 via svnmerge from
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r119801 | russell | 2008-06-02 11:14:15 -0500 (Mon, 02 Jun 2008) | 4 lines

Add app_fax from asterisk-addons, with some additional changes to resolve compiler
warnings, as well as update to the APIs in spandsp 0.0.5.  Spandsp 0.0.5 is being
distributed under the LGPL, so we can move this module into the main tree.

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2008-06-02 16:14:37 +00:00
Russell Bryant
19721dea99 Merged revisions 119799 via svnmerge from
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r119799 | russell | 2008-06-02 10:57:43 -0500 (Mon, 02 Jun 2008) | 4 lines

After determining that the version of spandsp installed is an acceptable version,
do a build and link test to ensure that the library is usable, and that libtiff
is also available

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2008-06-02 15:58:41 +00:00
Russell Bryant
d206a81d8d Merged revisions 119795 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r119795 | russell | 2008-06-02 10:43:40 -0500 (Mon, 02 Jun 2008) | 2 lines

Add a configure script check for spandsp

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2008-06-02 15:46:07 +00:00
Russell Bryant
6785df630c Merged revisions 119744 via svnmerge from
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r119744 | russell | 2008-06-02 09:41:55 -0500 (Mon, 02 Jun 2008) | 13 lines

Merged revisions 119742 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r119742 | russell | 2008-06-02 09:39:45 -0500 (Mon, 02 Jun 2008) | 5 lines

Improve CLI command blacklist checking for the command manager action.  Previously,
it did not handle case or whitespace properly.  This made it possible for blacklisted
commands to get executed anyway.
(closes issue #12765)

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2008-06-02 14:44:47 +00:00
Philippe Sultan
029ac4cb4d Merged revisions 119741 via svnmerge from
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r119741 | phsultan | 2008-06-02 16:35:24 +0200 (Mon, 02 Jun 2008) | 13 lines

Do not link the guest account with any configured XMPP client (in
jabber.conf). The actual connection is made when a call comes in
Asterisk.

Apply this fix to Jingle too.

Fix the ast_aji_get_client function that was not able to retrieve an
XMPP client from its JID.

(closes issue #12085)
Reported by: junky
Tested by: phsultan

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2008-06-02 14:40:42 +00:00
Russell Bryant
e46eda0e00 Merged revisions 119586,119637 via svnmerge from
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r119586 | crichter | 2008-06-02 03:46:23 -0500 (Mon, 02 Jun 2008) | 9 lines

Merged revisions 119585 via svnmerge from 
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r119585 | crichter | 2008-06-02 10:35:28 +0200 (Mo, 02 Jun 2008) | 1 line

Added counter for unhandled_bmsg Print, this prevents the logs to be flooded to fast and save CPU in this error scenario. Added 'last_used' element to bc structure, when a bchannel changes from used to free this exact time will be marked in last_used. When a new channel is requested the find_free_chan function will check if the new empty channel was used within the last second, if yes it will search for the next channel, if no it will return this channel. This simple mechanism has prooven to prevent race conditions where the NT and TE tried to allocate the exact same channel at the same time (RELEASE cause: 44).
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r119637 | crichter | 2008-06-02 04:35:04 -0500 (Mon, 02 Jun 2008) | 9 lines

Merged revisions 119636 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r119636 | crichter | 2008-06-02 11:29:21 +0200 (Mo, 02 Jun 2008) | 1 line

fixed compile issue when dev-mode is enabled
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2008-06-02 12:32:22 +00:00
Russell Bryant
aeceb3690c Merged revisions 119688 via svnmerge from
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r119688 | russell | 2008-06-02 07:30:42 -0500 (Mon, 02 Jun 2008) | 11 lines

Merged revisions 119687 via svnmerge from 
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r119687 | russell | 2008-06-02 07:30:17 -0500 (Mon, 02 Jun 2008) | 3 lines

Even of the first PING or LAGRQ doesn't get sent because it comes up too soon,
make sure to reschedule so it gets sent later.

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2008-06-02 12:31:05 +00:00
Russell Bryant
8ba94bb258 Merged revisions 119534 via svnmerge from
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r119534 | russell | 2008-06-01 20:08:16 -0500 (Sun, 01 Jun 2008) | 10 lines

Merged revisions 119533 via svnmerge from 
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r119533 | russell | 2008-06-01 20:06:09 -0500 (Sun, 01 Jun 2008) | 2 lines

Change a debug message to an actual debug message

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2008-06-02 01:08:56 +00:00
Russell Bryant
e60f5750ad Merged revisions 119531 via svnmerge from
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r119531 | russell | 2008-06-01 20:04:01 -0500 (Sun, 01 Jun 2008) | 10 lines

Merged revisions 119530 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r119530 | russell | 2008-06-01 20:03:22 -0500 (Sun, 01 Jun 2008) | 2 lines

Fix another typo in documentation

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2008-06-02 01:04:30 +00:00
Michiel van Baak
726f081141 Merged revisions 119479 via svnmerge from
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r119479 | mvanbaak | 2008-06-01 23:06:27 +0200 (Sun, 01 Jun 2008) | 10 lines

Merged revisions 119478 via svnmerge from 
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r119478 | mvanbaak | 2008-06-01 22:47:55 +0200 (Sun, 01 Jun 2008) | 2 lines

small typo fix 'retires' => 'retries'

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2008-06-01 21:59:44 +00:00
Tilghman Lesher
1db927c186 Merged revisions 119419 via svnmerge from
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r119419 | tilghman | 2008-05-30 16:23:14 -0500 (Fri, 30 May 2008) | 14 lines

Merged revisions 119404 via svnmerge from 
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r119404 | tilghman | 2008-05-30 16:17:45 -0500 (Fri, 30 May 2008) | 6 lines

When joinempty=strict, it only failed on join if there were busy members.  If
all members were logged out OR paused, then it (incorrectly) let callers join
the queue.
(closes issue #12451)
 Reported by: davidw

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2008-05-30 21:24:01 +00:00
Joshua Colp
430cfb75d8 Merged revisions 119355 via svnmerge from
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r119355 | file | 2008-05-30 16:47:30 -0300 (Fri, 30 May 2008) | 10 lines

Merged revisions 119354 via svnmerge from 
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r119354 | file | 2008-05-30 16:46:37 -0300 (Fri, 30 May 2008) | 2 lines

Fix a bug I found while testing for another issue.

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2008-05-30 19:48:25 +00:00
Tilghman Lesher
2fde71cb45 Oops, broke 1.6 (thanks MattF)
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2008-05-30 17:13:20 +00:00
Michiel van Baak
c568ee08f3 Merged revisions 119302 via svnmerge from
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r119302 | mvanbaak | 2008-05-30 18:47:24 +0200 (Fri, 30 May 2008) | 22 lines

Merged revisions 119301 via svnmerge from 
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r119301 | mvanbaak | 2008-05-30 18:44:39 +0200 (Fri, 30 May 2008) | 14 lines

dont use a bashism way to check the $VERSION variable.
The rc/init.d scripts, and safe_asterisk work on normal sh now again.
Tested on:
OpenBSD 4.2 (me)
Debian etch (me)
Ubuntu Hardy (me and loloski)
FC9 (loloski)

(closes issue #12687)
Reported by: loloski
Patches:
      20080529-12687-safe_asterisk-fixversion.diff.txt uploaded by mvanbaak (license 7)
	  Tested by: loloski, mvanbaak

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Tilghman Lesher
087a26f492 Merged revisions 119299 via svnmerge from
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r119299 | tilghman | 2008-05-30 11:40:13 -0500 (Fri, 30 May 2008) | 2 lines

Suppress warning about pbx structure already existing

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Tilghman Lesher
caab83a81b Merged revisions 119296 via svnmerge from
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r119296 | tilghman | 2008-05-30 11:10:46 -0500 (Fri, 30 May 2008) | 8 lines

Add native AGI command GOSUB, as invoking Gosub with EXEC does not work
properly.
(closes issue #12760)
 Reported by: Corydon76
 Patches: 
       20080530__bug12760.diff.txt uploaded by Corydon76 (license 14)
 Tested by: tim_ringenbach, Corydon76

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2008-05-30 16:11:31 +00:00
Russell Bryant
ec42d8e877 Merged revisions 119239 via svnmerge from
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r119239 | russell | 2008-05-30 07:59:11 -0500 (Fri, 30 May 2008) | 23 lines

Merged revisions 119238 via svnmerge from 
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r119238 | russell | 2008-05-30 07:55:36 -0500 (Fri, 30 May 2008) | 15 lines

Merged revisions 119237 via svnmerge from 
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r119237 | russell | 2008-05-30 07:49:39 -0500 (Fri, 30 May 2008) | 7 lines

- Instead of only enforcing destination call number checking on an ACK, check
  all full frames except for PING and LAGRQ, which may be sent by older versions
  too quickly to contain the destination call number.
  (As suggested by Tim Panton on the asterisk-dev list)
- Merge changes from team/russell/iax2-frame-race, which prevents PING and LAGRQ
  from being sent before the destination call number is known.

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2008-05-30 13:01:05 +00:00
Russell Bryant
3d1ed5a41e Merged revisions 119157 via svnmerge from
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r119157 | russell | 2008-05-29 17:28:50 -0500 (Thu, 29 May 2008) | 18 lines

Merged revisions 119156 via svnmerge from 
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r119156 | russell | 2008-05-29 17:24:29 -0500 (Thu, 29 May 2008) | 10 lines

Fix a race condition in channel autoservice.  There was still a small window of opportunity
for a DTMF frame, or some other deferred frame type, to come in and get dropped.

(closes issue #12656)
(closes issue #12656)
Reported by: dimas
Patches:
      v3-12656.patch uploaded by dimas (license 88)
	  -- with some modifications by me

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2008-05-29 22:31:44 +00:00
Brett Bryant
8a1ca56cf6 Blocked revisions 119126 via svnmerge
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r119126 | bbryant | 2008-05-29 16:30:37 -0500 (Thu, 29 May 2008) | 7 lines

Adds support for changing logger settingss on remote consoles with a 
new command "logger set level". 

i.e. "logger set level debug off"

(closes issue #10891)

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2008-05-29 21:31:03 +00:00
Tilghman Lesher
5df1e912c8 Merged revisions 119072 via svnmerge from
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r119072 | tilghman | 2008-05-29 15:25:33 -0500 (Thu, 29 May 2008) | 15 lines

Merged revisions 119071 via svnmerge from 
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r119071 | tilghman | 2008-05-29 15:24:11 -0500 (Thu, 29 May 2008) | 7 lines

Call waiting tone occurs too often, because it's getting serviced by both
subchannels.
(closes issue #11354)
 Reported by: cahen
 Patches: 
       20080512__bug11354.diff.txt uploaded by Corydon76 (license 14)

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2008-05-29 20:26:18 +00:00
Michiel van Baak
11145b3feb Blocked revisions 119015 via svnmerge
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r119015 | mvanbaak | 2008-05-29 21:10:31 +0200 (Thu, 29 May 2008) | 10 lines

Make sure the nrfds and nefds are reset to NULL
before we enter manage_parkinglot.
This will get rid of CLI warnings like:
__ast_read: Exception flag set on 'SIP/<NUMBER>-<ID>', but no exception handler

(closes issue #12748)
Reported by: nreinartz
Patches:
      asterisk-multiparking_initialize_filedescr_sets-0.0.1.patch uploaded by nreinartz (license 452)

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2008-05-29 19:13:02 +00:00
Russell Bryant
cbe000d676 Merged revisions 119013 via svnmerge from
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r119013 | russell | 2008-05-29 14:05:33 -0500 (Thu, 29 May 2008) | 12 lines

Merged revisions 119012 via svnmerge from 
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r119012 | russell | 2008-05-29 14:04:52 -0500 (Thu, 29 May 2008) | 4 lines

 - Fix a typo in the argument to Playtones
 - use ast_safe_sleep() instead of calling the wait application
(thanks to tilghman for pointing these out!)

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Russell Bryant
4da50bd4c1 Merged revisions 119010 via svnmerge from
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r119010 | russell | 2008-05-29 13:54:11 -0500 (Thu, 29 May 2008) | 24 lines

Merged revisions 119009 via svnmerge from 
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r119009 | russell | 2008-05-29 13:49:12 -0500 (Thu, 29 May 2008) | 16 lines

Merged revisions 119008 via svnmerge from 
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r119008 | russell | 2008-05-29 13:45:21 -0500 (Thu, 29 May 2008) | 7 lines

Merge changes from team/russell/iax2-another-fix-to-the-fix

As described in the following post to the asterisk-dev mailing list, only
enforce destination call numbers when processing an ACK.

http://lists.digium.com/pipermail/asterisk-dev/2008-May/033217.html

(closes issue #12631)
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2008-05-29 18:57:20 +00:00
Russell Bryant
52b2841fa0 Merged revisions 118962 via svnmerge from
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r118962 | russell | 2008-05-29 12:52:00 -0500 (Thu, 29 May 2008) | 11 lines

Merged revisions 118961 via svnmerge from 
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r118961 | russell | 2008-05-29 12:51:29 -0500 (Thu, 29 May 2008) | 3 lines

 - Mark app_milliwatt dependent on res_indications (thanks to jsmith)
 - fix a typo in a log message (thanks to qwell)

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2008-05-29 17:52:21 +00:00
Russell Bryant
7df9585c72 Merged revisions 118959 via svnmerge from
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r118959 | russell | 2008-05-29 12:46:04 -0500 (Thu, 29 May 2008) | 11 lines

Merged revisions 118956 via svnmerge from 
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r118956 | russell | 2008-05-29 12:38:38 -0500 (Thu, 29 May 2008) | 3 lines

Change milliwatt to use the proper tone by default (1004 Hz) instead of 1000 Hz.
An option is there to use 1000 Hz for anyone that might want it.

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2008-05-29 17:46:57 +00:00
Tilghman Lesher
ef98b16564 Merged revisions 118955,118957 via svnmerge from
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r118955 | tilghman | 2008-05-29 12:35:19 -0500 (Thu, 29 May 2008) | 11 lines

Merged revisions 118953 via svnmerge from 
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r118953 | tilghman | 2008-05-29 12:20:16 -0500 (Thu, 29 May 2008) | 3 lines

Add some debugging code that ensures that when we do deadlock avoidance, we
don't lose the information about how a lock was originally acquired.

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r118957 | tilghman | 2008-05-29 12:39:50 -0500 (Thu, 29 May 2008) | 10 lines

Merged revisions 118954 via svnmerge from 
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r118954 | tilghman | 2008-05-29 12:33:01 -0500 (Thu, 29 May 2008) | 2 lines

Define also when not DEBUG_THREADS

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2008-05-29 17:42:28 +00:00
Sean Bright
08109897c3 Blocked revisions 118911 via svnmerge
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r118911 | seanbright | 2008-05-29 08:12:07 -0400 (Thu, 29 May 2008) | 7 lines

Avoid build warning when execinfo.h isn't available.

(closes issue #12751)
Reported by: ys
Patches:
      check_expr.diff uploaded by ys (license 281)

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2008-05-29 12:16:16 +00:00
Steve Murphy
1051579614 Merged revisions 118880 via svnmerge from
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r118880 | murf | 2008-05-28 19:29:09 -0600 (Wed, 28 May 2008) | 54 lines

Merged revisions 118858 via svnmerge from 
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r118858 | murf | 2008-05-28 18:25:28 -0600 (Wed, 28 May 2008) | 46 lines

(closes issue #10668)
(closes issue #11721)
(closes issue #12726)
Reported by: arkadia
Tested by: murf

These changes:

1. revert the changes made via bug 10668; 
   I should have known that such changes,
   even tho they made sense at the time,
   seemed like an omission, etc, were actually
   integral to the CDR system via forkCDR.
   It makes sense to me now that forkCDR didn't
   natively end any CDR's, but rather depended
   on natively closing them all at hangup time
   via traversing and closing them all, whether
   locked or not. I still don't completely 
   understand the benefits of setvar and answer
   operating on locked cdrs, but I've seen 
   enough to revert those changes also, and
   stop messing up users who depended on that
   behavior. bug 12726 found reverting the changes
   fixed his changes, and after a long review
   and working on forkCDR, I can see why.
2. Apply the suggested enhancements proposed
   in 10668, but in a completely compatible
   way. ForkCDR will behave exactly as before,
   but now has new options that will allow some
   actions to be taken that will slightly 
   modify the outcome and side-effects of
   forkCDR. Based on conversations I've had 
   with various people, these small tweaks
   will allow some users to get the behavior
   they need. For instance, users executing
   forkCDR in an AGI script will find the
   answer time set, and DISPOSITION set, 
   a situation not covered when the routines
  were first written.
3. A small problem in the cdr serializer
   would output answer and end times even
   when they were not set. This is now 
   fixed.



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2008-05-29 04:11:53 +00:00
Michiel van Baak
b3b8fcdad2 Blocked revisions 118824 via svnmerge
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r118824 | mvanbaak | 2008-05-29 00:05:58 +0200 (Thu, 29 May 2008) | 4 lines

formatting changes.
A lot of whitespace issues have been resolved in this commit
Also some doc updates, but that's only 6 lines

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2008-05-28 22:08:34 +00:00
Michiel van Baak
3aeaabd1cc Blocked revisions 118790 via svnmerge
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r118790 | mvanbaak | 2008-05-28 22:00:21 +0200 (Wed, 28 May 2008) | 2 lines

rename DialTone.h to chan_phone.h because chan_phone.c is the only file using it

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2008-05-28 20:02:29 +00:00
Mark Michelson
65d1595160 Blocked revisions 118783 via svnmerge
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r118783 | mmichelson | 2008-05-28 14:56:18 -0500 (Wed, 28 May 2008) | 15 lines

Update to the janitor project for making sure to be thread-safe when
retrieving the value of a channel variable. This covers app_queue.

This commit also incorporates a logical change. Previously, if MixMonitor
is to be used to record the call, all the arguments were parsed first. Then
the MixMonitor app would be located. Now the order of these operations has
been swapped. Now the app is located first so that we only go through the
work of parsing the arguments if the app was found.

(closes issue #12742)
Reported by: snuffy
Patches:
      bug_12742.diff uploaded by snuffy (license 35)


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2008-05-28 19:58:07 +00:00
Michiel van Baak
283ebdb91a Merged revisions 118750 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r118750 | mvanbaak | 2008-05-28 19:58:21 +0200 (Wed, 28 May 2008) | 2 lines

remove unused astobj.h header file from chan_skinny.c

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2008-05-28 18:07:06 +00:00
Joshua Colp
ad57cc89de Merged revisions 118647 via svnmerge from
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r118647 | file | 2008-05-28 11:29:01 -0300 (Wed, 28 May 2008) | 12 lines

Merged revisions 118646 via svnmerge from 
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r118646 | file | 2008-05-28 11:23:34 -0300 (Wed, 28 May 2008) | 4 lines

Add an option to use the source IP address of RTP as the destination IP address of UDPTL when a specific option is enabled. If the remote side is properly configured (ports forwarded) then UDPTL will flow.
(closes issue #10417)
Reported by: cstadlmann

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2008-05-28 14:31:42 +00:00
Philippe Sultan
4d5394383f Merged revisions 118644 via svnmerge from
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r118644 | phsultan | 2008-05-28 16:10:48 +0200 (Wed, 28 May 2008) | 10 lines

Changed to temporary namespaces to match with latest XEPs. As soon as
Jingle is completely standardized, we can set those namespaces to their
final values.

Added two attributes to the jingle_pvt struct to store the content
name attributes. Reported by Robert McQueen on Telepathy's framework
mailing list :
http://lists.freedesktop.org/archives/telepathy/2008-May/001971.html

Keeping working on our Jingle stack!
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2008-05-28 14:13:16 +00:00
Philippe Sultan
93898d77d9 Merged revisions 118614 via svnmerge from
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r118614 | phsultan | 2008-05-28 10:39:10 +0200 (Wed, 28 May 2008) | 1 line

Code simplification
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2008-05-28 08:44:15 +00:00
Brett Bryant
0a8361190d Blocked revisions 118562 via svnmerge
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r118562 | bbryant | 2008-05-27 14:45:41 -0500 (Tue, 27 May 2008) | 4 lines

Remove loop from the detection of a sequence number that acknowledges 
the receiving of a packet that we've kept in memory just incase the 
packet needs to be retransmitted.

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2008-05-27 19:52:43 +00:00
Joshua Colp
5c1d88af55 Merged revisions 118560 via svnmerge from
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r118560 | file | 2008-05-27 16:34:14 -0300 (Tue, 27 May 2008) | 12 lines

Merged revisions 118558 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r118558 | file | 2008-05-27 16:32:38 -0300 (Tue, 27 May 2008) | 4 lines

Fix an issue where codec preferences were not set on dialogs that were not authenticated via a user or peer and allow framing to work without rtpmap in the SDP.
(closes issue #12501)
Reported by: slimey

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2008-05-27 19:35:47 +00:00
Russell Bryant
5794734668 Merged revisions 118556 via svnmerge from
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r118556 | russell | 2008-05-27 14:27:48 -0500 (Tue, 27 May 2008) | 6 lines

Add printf format attribute for vasprintf().
(closes issue #12729)
Reported by: snuffy
Patches:
      bug_12729.diff uploaded by snuffy (license 35)

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2008-05-27 19:28:19 +00:00
Tilghman Lesher
3e82a519b0 Merged revisions 118554 via svnmerge from
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r118554 | tilghman | 2008-05-27 14:21:03 -0500 (Tue, 27 May 2008) | 14 lines

Merged revisions 118551 via svnmerge from 
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r118551 | tilghman | 2008-05-27 14:15:27 -0500 (Tue, 27 May 2008) | 6 lines

When showing an error message for a command, don't shorten the command output,
as it tends to confuse the user (it's fine for suggesting other commands,
however).
 Reported by: seanbright (on #asterisk-dev)
 Fixed by: me

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2008-05-27 19:22:03 +00:00
Mark Michelson
17c015c515 Merged revisions 118514 via svnmerge from
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r118514 | mmichelson | 2008-05-27 14:08:24 -0500 (Tue, 27 May 2008) | 19 lines

Merged revisions 118509 via svnmerge from 
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r118509 | mmichelson | 2008-05-27 14:07:26 -0500 (Tue, 27 May 2008) | 11 lines

Russell noted to me that in the case that separate threads use their
own addressing system, the fix I made for issue 12376 does not guarantee
uniqueness to the datastores' uids. Though I know of no system that works
this way, I am going to change this right now to prevent trying to track
down some future bug that may occur and cause untold hours of debugging
time to track down.

The change involves using a global counter which increases with each new
chanspy_ds which is created. This guarantees uniqueness.


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2008-05-27 19:09:00 +00:00
Tilghman Lesher
c0aa5a8b1c Merged revisions 118466 via svnmerge from
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r118466 | tilghman | 2008-05-27 13:59:06 -0500 (Tue, 27 May 2008) | 16 lines

Merged revisions 118465 via svnmerge from 
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r118465 | tilghman | 2008-05-27 13:58:09 -0500 (Tue, 27 May 2008) | 8 lines

NULL character should terminate only commands back to the core, not log
messages to the console.
(closes issue #12731)
 Reported by: seanbright
 Patches: 
       20080527__bug12731.diff.txt uploaded by Corydon76 (license 14)
 Tested by: seanbright

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2008-05-27 18:59:57 +00:00
Michiel van Baak
2a161cf12d git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@118464 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2008-05-27 17:48:09 +00:00
Michiel van Baak
ef781276a6 small update to the g() option of app_voicemail to
note that gain changes only work on zap channels right now.
issue #12578 shows it's not clear right now.


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2008-05-27 17:25:33 +00:00
Mark Michelson
916e40b9fd Merged revisions 118371 via svnmerge from
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r118371 | mmichelson | 2008-05-27 11:43:36 -0500 (Tue, 27 May 2008) | 22 lines

Merged revisions 118365 via svnmerge from 
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r118365 | mmichelson | 2008-05-27 11:38:38 -0500 (Tue, 27 May 2008) | 14 lines

Add a unique id to the datastore allocated in app_chanspy since
it is possible that multiple spies may be listening to the same
channel.

(closes issue #12376)
Reported by: DougUDI
Patches:
      12376_chanspy_uid.diff uploaded by putnopvut (license 60)
Tested by: destiny6628

(closes issue #12243)
Reported by: atis


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2008-05-27 16:48:50 +00:00
Mark Michelson
3966f2647f Hmm, I apparently forgot to commit the block of revision 118175.
Now I'm doing it.



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Tilghman Lesher
fcf2077659 Merged revisions 118359 via svnmerge from
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r118359 | tilghman | 2008-05-27 10:46:58 -0500 (Tue, 27 May 2008) | 11 lines

Merged revisions 118358 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r118358 | tilghman | 2008-05-27 10:45:37 -0500 (Tue, 27 May 2008) | 3 lines

Add a note that pbx_config.so is needed for Local channels.
(Closes issue #12671)

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2008-05-27 15:47:51 +00:00
Russell Bryant
b365d27f4d Merged revisions 118328 via svnmerge from
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r118328 | russell | 2008-05-27 09:51:13 -0500 (Tue, 27 May 2008) | 2 lines

Add printf attribute to asprintf

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2008-05-27 14:51:43 +00:00
Tilghman Lesher
e3dec82d84 Merged revisions 118302 via svnmerge from
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r118302 | tilghman | 2008-05-27 08:30:10 -0500 (Tue, 27 May 2008) | 6 lines

When binding anonymously, credentials are still needed.
(closes issue #12601)
 Reported by: suretec
 Patches: 
       res_config_ldap.c.patch uploaded by suretec (license 70)

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2008-05-27 13:30:59 +00:00
Tilghman Lesher
138465fbc9 Merged revisions 118300 via svnmerge from
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r118300 | tilghman | 2008-05-27 08:13:17 -0500 (Tue, 27 May 2008) | 4 lines

In compat14 mode, don't translate pipes inside expressions, as they aren't
argument delimiters, but rather 'or' symbols.
(Closes issue #12723)

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2008-05-27 13:14:35 +00:00
Tilghman Lesher
80f7c5ba35 Merged revisions 118252 via svnmerge from
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r118252 | tilghman | 2008-05-25 11:17:05 -0500 (Sun, 25 May 2008) | 20 lines

Merged revisions 118251 via svnmerge from 
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r118251 | tilghman | 2008-05-25 11:02:04 -0500 (Sun, 25 May 2008) | 12 lines

Realtime flag affects construction in multiple ways, so consulting whether
rtcachefriends was set was done too soon (needed to be done inside build_peer,
not just as a flag to build_peer).
Also, fullcontact needed to be reconstructed, because realtime separates the
embedded ';' into multiple fields.
(closes issue #12722)
 Reported by: barthpbx
 Patches: 
       20080525__bug12722.diff.txt uploaded by Corydon76 (license 14)
 Tested by: barthpbx
 (Much of the discussion happened on #asterisk-dev for diagnosing this issue)

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2008-05-25 16:20:32 +00:00
Jeff Peeler
dc2e0b1200 Merged revisions 118178 via svnmerge from
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r118178 | jpeeler | 2008-05-23 20:14:41 -0500 (Fri, 23 May 2008) | 1 line

add document describing API changes from 1.4.0 to 1.6.0
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2008-05-24 01:15:40 +00:00
Jeff Peeler
e3c52c3d8e Blocked revisions 118176 via svnmerge
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r118176 | jpeeler | 2008-05-23 19:37:01 -0500 (Fri, 23 May 2008) | 3 lines

Fixes segfault in parking, patch submitted by bmd.


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2008-05-24 00:53:34 +00:00
Mark Michelson
7791e24f3d Blocked revisions 118173 via svnmerge
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r118173 | mmichelson | 2008-05-23 17:35:50 -0500 (Fri, 23 May 2008) | 15 lines

A new feature thanks to the fine folks at Switchvox!

If a deadlock is detected, then the typical lock information will be
printed along with a backtrace of the stack for the offending threads.
Use of this requires compiling with DETECT_DEADLOCKS and having glibc
installed.

Furthermore, issuing the "core show locks" CLI command will print the
normal lock information as well as a backtraces for each lock. This
requires that DEBUG_THREADS is enabled and that glibc is installed.

All the backtrace features may be disabled by running the configure
script with --without-execinfo as an argument


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2008-05-23 22:36:20 +00:00
Brett Bryant
aec651c6d6 Merged revisions 118161 via svnmerge from
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r118161 | bbryant | 2008-05-23 16:19:42 -0500 (Fri, 23 May 2008) | 3 lines

Add new functionality to http server that requires manager authentication for any path that includes a directory named 'private'. This patch also 
requires manager authentication for any POST's being sent to the server as well to help secure uploads.

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2008-05-23 21:37:07 +00:00
Jeff Peeler
13616f1b68 Merged revisions 118164 via svnmerge from
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r118164 | jpeeler | 2008-05-23 16:26:39 -0500 (Fri, 23 May 2008) | 9 lines

Merged revisions 118163 via svnmerge from 
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r118163 | jpeeler | 2008-05-23 16:21:35 -0500 (Fri, 23 May 2008) | 1 line

Fix a few things I missed to ensure zt_chan_conf structure is not modified in mkintf
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Mark Michelson
bc0da9cd51 Blocked revisions 118159 via svnmerge
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r118159 | mmichelson | 2008-05-23 15:55:02 -0500 (Fri, 23 May 2008) | 4 lines

Get rid of warnings for those silly compilers which warn when freeing
a const pointer


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Mark Michelson
d21a7d4873 Blocked revisions 118157 via svnmerge
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r118157 | mmichelson | 2008-05-23 14:57:40 -0500 (Fri, 23 May 2008) | 10 lines

Use a deep copy on strings that come from ast_events. Otherwise it is
likely that after the event is freed, we no longer refer to valid memory.

(closes issue #12712)
Reported by: tomo1657
Patches:
      12712.patch uploaded by putnopvut (license 60)
Tested by: tomo1657


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2008-05-23 20:00:00 +00:00
Tilghman Lesher
6ece458a1a Merged revisions 118129 via svnmerge from
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r118129 | tilghman | 2008-05-23 13:09:14 -0500 (Fri, 23 May 2008) | 3 lines

Protect the object from changing while the 'odbc show' CLI command is running
(Closes issue #12704)

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2008-05-23 18:15:41 +00:00
Michiel van Baak
4243dbb417 Blocked revisions 118101 via svnmerge
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r118101 | mvanbaak | 2008-05-23 19:12:04 +0200 (Fri, 23 May 2008) | 15 lines

add option 'a' to chanisavail.
If you give chanisavail a list of channels, it will only
return the first available channel.
When this option is set, it will return all the available
channels from the given list.

(closes issue #12248)
Reported by: dagmoller
Patches:
      app_chanisavail-snv.patch-v2.txt uploaded by dagmoller (license 436)
	   - major changes by me because russellb pointed out some buffer overflows
	     and codeguideline issues.
		 Converted it all to the ast_str_* api
Tested by: dagmoller, mvanbaak

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2008-05-23 17:13:31 +00:00
Tilghman Lesher
6232bf8be5 Merged revisions 118053 via svnmerge from
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r118053 | tilghman | 2008-05-23 08:00:10 -0500 (Fri, 23 May 2008) | 11 lines

Merged revisions 118052 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r118052 | tilghman | 2008-05-23 07:59:16 -0500 (Fri, 23 May 2008) | 3 lines

Add information on using the Asterisk console, including tab command line
completion.  (Closes issue #12681)

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2008-05-23 13:00:48 +00:00
Russell Bryant
6d4d094baa Merged revisions 118049 via svnmerge from
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r118049 | russell | 2008-05-23 07:37:31 -0500 (Fri, 23 May 2008) | 17 lines

Merged revisions 118048 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r118048 | russell | 2008-05-23 07:30:53 -0500 (Fri, 23 May 2008) | 9 lines

Don't declare a function that takes variable arguments as inline, because it's
not valid, and on some compilers, will emit a warning.

http://gcc.gnu.org/onlinedocs/gcc/Inline.html#Inline

(closes issue #12289)
Reported by: francesco_r
Patches by Tilghman, final patch by me

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2008-05-23 12:37:58 +00:00
Philippe Sultan
01f7b2b579 Merged revisions 118020 via svnmerge from
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r118020 | phsultan | 2008-05-23 12:33:21 +0200 (Fri, 23 May 2008) | 15 lines

- remove whitespaces between tags in received XML packets before giving
them to the parser ;
- report Gtalk error messages from a buddy to the console.

This patch makes Asterisk "Google Jingle" (chan_gtalk) implementation
work with Empathy. Note that this is only true for audio streams, not
video.

Thank you to PH for his great help!

(closes issue #12647)
Reported by: PH
Patches:
      trunk-12647-1.diff uploaded by phsultan (license 73)
Tested by: phsultan, PH
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2008-05-23 11:02:21 +00:00
Sean Bright
a1b9366aec Blocked revisions 117988 via svnmerge
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r117988 | seanbright | 2008-05-22 17:43:54 -0400 (Thu, 22 May 2008) | 1 line

Split the compile flags out and wire up some dependencies
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2008-05-22 21:44:49 +00:00
Tilghman Lesher
a37fcc7f2f Merged revisions 117986 via svnmerge from
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r117986 | tilghman | 2008-05-22 16:42:50 -0500 (Thu, 22 May 2008) | 2 lines

Add a compatibility option for upgrading realtime extensions

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2008-05-22 21:43:47 +00:00
Tilghman Lesher
85f1e10148 Blocked revisions 117983 via svnmerge
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r117983 | tilghman | 2008-05-22 16:27:00 -0500 (Thu, 22 May 2008) | 2 lines

Fix trunk breakage

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2008-05-22 21:29:08 +00:00
Sean Bright
5a958c150b Blocked revisions 117950 via svnmerge
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r117950 | seanbright | 2008-05-22 16:01:33 -0400 (Thu, 22 May 2008) | 1 line

A couple more places the frame data change was missed.
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2008-05-22 20:02:27 +00:00
Michiel van Baak
5ce17588b7 Blocked revisions 117822,117825,117834,117870 via svnmerge
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r117822 | mvanbaak | 2008-05-22 19:06:00 +0200 (Thu, 22 May 2008) | 2 lines

forgot chan_misdn

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r117825 | mvanbaak | 2008-05-22 19:08:18 +0200 (Thu, 22 May 2008) | 2 lines

oops

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r117834 | mvanbaak | 2008-05-22 19:16:08 +0200 (Thu, 22 May 2008) | 2 lines

chan_console fixes because of ast_frame.data => ast_frame.data.ptr

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r117870 | mvanbaak | 2008-05-22 19:50:40 +0200 (Thu, 22 May 2008) | 2 lines

one more place I forgot

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2008-05-22 19:45:21 +00:00
Tilghman Lesher
04793482a0 Merged revisions 117900 via svnmerge from
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r117900 | tilghman | 2008-05-22 13:54:41 -0500 (Thu, 22 May 2008) | 10 lines

Merged revisions 117899 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r117899 | tilghman | 2008-05-22 13:53:53 -0500 (Thu, 22 May 2008) | 2 lines

Also remove preamble from asynchronous events (reported by jsmith on #asterisk-dev)

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Michiel van Baak
9244f0f433 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@117804 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2008-05-22 16:36:04 +00:00
Mark Michelson
147a00886c Blocked revisions 117794 via svnmerge
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r117794 | mmichelson | 2008-05-22 11:05:18 -0500 (Thu, 22 May 2008) | 3 lines

Committing a fix pointed out by Atis Lezdins on the asterisk-dev list. Thanks!


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2008-05-22 16:06:18 +00:00
Sean Bright
a7d20c7109 Merged revisions 117792 via svnmerge from
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r117792 | seanbright | 2008-05-22 11:49:17 -0400 (Thu, 22 May 2008) | 1 line

Minor text fix.  roster -> resource.
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2008-05-22 15:51:04 +00:00
Russell Bryant
fd15bd07e6 Merged revisions 117756 via svnmerge from
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r117756 | russell | 2008-05-22 08:40:52 -0500 (Thu, 22 May 2008) | 5 lines

Store build-time options as a string in AST_BUILDOPTS in buildopts.h.  Also,
display this information in the "core show settings" CLI command.  This is
useful if you want to verify that you're running a build with DONT_OPTIMIZE,
DEBUG_THREADS, etc.

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2008-05-22 13:41:31 +00:00
Tilghman Lesher
f44eb6a3b8 Blocked revisions 117725 via svnmerge
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r117725 | tilghman | 2008-05-22 00:10:01 -0500 (Thu, 22 May 2008) | 8 lines

Enhance ExternalIVR with new options and commands.
(closes issue #12705)
 Reported by: ctooley
 Patches: 
       new_externalivr_argument_format-v2.diff uploaded by ctooley (license 136)
       new_externalivr_documentation.diff uploaded by ctooley (license 136)
       and a few additional fixes by me

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2008-05-22 05:11:12 +00:00
Mark Michelson
a7a5e26831 Blocked revisions 117693 via svnmerge
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r117693 | mmichelson | 2008-05-21 17:34:27 -0500 (Wed, 21 May 2008) | 5 lines

This change makes it so that logs will report the correct source of verbose messages.
Until this change, all verbose messages in Asterisk's log files reported logger.c
as the source of the message.


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2008-05-21 22:35:17 +00:00
Jeff Peeler
da8a5856be Merged revisions 117658 via svnmerge from
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r117658 | jpeeler | 2008-05-21 16:31:17 -0500 (Wed, 21 May 2008) | 10 lines

Merged revisions 117582 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r117582 | jpeeler | 2008-05-21 15:11:14 -0500 (Wed, 21 May 2008) | 2 lines

Ensure that passed in zt_chan_conf structure is not modified in mkintf.

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Jeff Peeler
ef9c60d095 Merged revisions 117628 via svnmerge from
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r117628 | jpeeler | 2008-05-21 15:44:04 -0500 (Wed, 21 May 2008) | 12 lines

Merged revisions 117462 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r117462 | jpeeler | 2008-05-21 11:58:40 -0500 (Wed, 21 May 2008) | 3 lines

Pass a pointer for the conf parameter to the function mkintf rather than the whole zt_chan_conf structure.
Another commit is following to make sure the zt_chan_conf structure is not modified.


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Mark Michelson
77902e5f19 Blocked revisions 117625 via svnmerge
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r117625 | mmichelson | 2008-05-21 15:27:45 -0500 (Wed, 21 May 2008) | 9 lines

Add a new manager event, AgentRingNoAnswer to
app_queue.

(closes issue #12591)
Reported by: CCHAsteria
Patches:
      app_queue_RNA_event.diff uploaded by CCHAsteria (license 477)


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2008-05-21 20:28:40 +00:00
Joshua Colp
2ded3ac61d Merged revisions 117575 via svnmerge from
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r117575 | file | 2008-05-21 16:39:42 -0300 (Wed, 21 May 2008) | 10 lines

Merged revisions 117574 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r117574 | file | 2008-05-21 16:38:28 -0300 (Wed, 21 May 2008) | 2 lines

Apply the autoframing setting to dialogs that do not get matched against a user or peer.

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Tilghman Lesher
21e9dced65 Merged revisions 117520 via svnmerge from
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r117520 | tilghman | 2008-05-21 13:43:26 -0500 (Wed, 21 May 2008) | 11 lines

Merged revisions 117519 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r117519 | tilghman | 2008-05-21 13:40:14 -0500 (Wed, 21 May 2008) | 3 lines

Strip the preamble from the output also when -rx is not being used
(Related to issue #12702)

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Mark Michelson
bdc7890852 Blocked revisions 117517 via svnmerge
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r117517 | mmichelson | 2008-05-21 13:31:05 -0500 (Wed, 21 May 2008) | 14 lines

Optimize the update_realtime_member_field function by not having
to query the database for the member and instead using a cached
uniqueid. 

Special thanks to atis for creating this and for keeping it up
to date with necessary changes

(closes issue #11896)
Reported by: atis
Patches:
      realtime_uniqueid_v6.patch uploaded by atis (license 242)
Tested by: atis


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2008-05-21 18:31:41 +00:00
Russell Bryant
b469957233 Merged revisions 117515 via svnmerge from
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r117515 | russell | 2008-05-21 13:29:05 -0500 (Wed, 21 May 2008) | 12 lines

Merged revisions 117514 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r117514 | russell | 2008-05-21 13:28:46 -0500 (Wed, 21 May 2008) | 4 lines

Don't filter the magic character in the network verboser.  It gets filtered
once it reaches the client.
(related to issue #12702, pointed out by tilghman)

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2008-05-21 18:29:28 +00:00
Russell Bryant
45bbed1f49 Merged revisions 117508 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r117508 | russell | 2008-05-21 13:20:11 -0500 (Wed, 21 May 2008) | 15 lines

Merged revisions 117507 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r117507 | russell | 2008-05-21 13:19:34 -0500 (Wed, 21 May 2008) | 7 lines

1) Don't print the verbose marker in front of every message from ast_verbose()
   being sent to remote consoles.

2) Fix pbx_gtkconsole to filter out the verbose marker.

(related to issue #12702)

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2008-05-21 18:20:34 +00:00
Russell Bryant
638d9b32f9 Merged revisions 117481 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r117481 | russell | 2008-05-21 13:12:19 -0500 (Wed, 21 May 2008) | 14 lines

Merged revisions 117479 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r117479 | russell | 2008-05-21 13:11:51 -0500 (Wed, 21 May 2008) | 6 lines

Don't display the verbose marker for calls to ast_verbose() that do not include
a VERBOSE_PREFIX in front of the message.
(closes issue #12702)
Reported by: johnlange
Patched by me

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2008-05-21 18:13:08 +00:00
Tilghman Lesher
7dbf39cce8 Blocked revisions 117431 via svnmerge
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r117431 | tilghman | 2008-05-21 08:39:41 -0500 (Wed, 21 May 2008) | 6 lines

On socket-based connections, there is no error FD, so don't try waiting on one.
(closes issue #12697)
 Reported by: ctooley
 Patches: 
       fix_externalivr_waitfor_nandfds-v3.diff uploaded by ctooley (license 136)

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2008-05-21 13:41:39 +00:00
Mark Michelson
9daeb19f80 Merged revisions 117367 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r117367 | mmichelson | 2008-05-20 21:20:31 -0500 (Tue, 20 May 2008) | 19 lines

Be sure that we cache included files for each source file which loads
a configuration file.  As it was, only the first did so. This led to 
a problem if the included file was changed (but not the configuration 
file which includes it) and the second source file attempted to reload 
the configuration. It would not see that the included file had changed.

In this particular example, res_phoneprov and chan_sip both loaded
sip.conf, which included a file call sip.peers.conf. Since res_phoneprov
was the first to load sip.conf, only it cached the fact that sip.conf
included sip.peers.conf. If sip.peers.conf were changed and sip.conf were
not and a sip reload were issued (meaning that chan_sip attempts to 
reload sip.conf only if it and its included files have changed) the changes
made to sip.peers.conf would not be seen and therefore no action would be
taken.

(closes issue #12693)
Reported by: marsosa


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2008-05-21 02:21:38 +00:00
Steve Murphy
2fdbd728b8 Merged revisions 117335 via svnmerge from
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r117335 | murf | 2008-05-20 19:00:28 -0600 (Tue, 20 May 2008) | 10 lines

These changes were made via the comments atis_work made at 4:30am (Mountain Time zone- US)
in #asterisk-dev on 20 May 2008. He noted that a backslash was being inserted before commas
in app call arguments in the extensions.conf.aeldump file that you get from aelparse with the
-w arg. This was being generated from code left over from 1.4, where commas were substituted
with '|', and any remaining commas needed to be escaped.

Many thanks to atis for his comment; please let us know if these changes break anything!



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2008-05-21 01:20:45 +00:00
Tilghman Lesher
de3112d3ee Blocked revisions 117264 via svnmerge
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r117264 | tilghman | 2008-05-20 11:25:16 -0500 (Tue, 20 May 2008) | 3 lines

Increase limit of unshared connections from 1023 to 4.2 billion.
(Related to issue #12677)

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Tilghman Lesher
05e0ef7bfd Blocked revisions 117262 via svnmerge
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r117262 | tilghman | 2008-05-20 11:13:48 -0500 (Tue, 20 May 2008) | 4 lines

Revert part of previous fix, and heavily comment the logic for object
destruction, for future users.
(Closes issue #12677)

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2008-05-20 16:16:13 +00:00
Tilghman Lesher
b762595c7e Blocked revisions 117182 via svnmerge
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r117182 | tilghman | 2008-05-19 15:06:38 -0500 (Mon, 19 May 2008) | 7 lines

Change the default for the pridialplan parameter to the far more common case of
'unknown', and better document the use of each parameter.
(closes issue #12633)
 Reported by: tzafrir
 Patches: 
       pridialplan_unknown_2.diff uploaded by tzafrir (license 46)

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2008-05-19 20:08:00 +00:00
Joshua Colp
5f996f261f Merged revisions 117136 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r117136 | file | 2008-05-19 13:53:33 -0300 (Mon, 19 May 2008) | 14 lines

Merged revisions 117135 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r117135 | file | 2008-05-19 13:50:52 -0300 (Mon, 19 May 2008) | 6 lines

Use the right pthread lock and condition when waiting.
(closes issue #12664)
Reported by: tomo1657
Patches:
      res_smdi.c.patch uploaded by tomo1657 (license 484)

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2008-05-19 16:58:10 +00:00
Joshua Colp
718d8800c0 Blocked revisions 117133 via svnmerge
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r117133 | file | 2008-05-19 13:22:56 -0300 (Mon, 19 May 2008) | 4 lines

Remove a premature mutex destroy (the destruction callback will end up destroying it) and use a callback to purge remaining classes.
(closes issue #12677)
Reported by: falves11

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2008-05-19 16:23:58 +00:00
Tilghman Lesher
aaccfd2de9 Merged revisions 117088 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r117088 | tilghman | 2008-05-19 11:07:09 -0500 (Mon, 19 May 2008) | 10 lines

Merged revisions 117086 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r117086 | tilghman | 2008-05-19 11:05:05 -0500 (Mon, 19 May 2008) | 2 lines

The addition of usleep(2) within ast_assert requires the inclusion of the unistd.h header

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2008-05-19 16:07:53 +00:00
Joshua Colp
69fa2c8a19 Merged revisions 117085 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r117085 | file | 2008-05-19 13:03:33 -0300 (Mon, 19 May 2008) | 4 lines

The logger closes the files it is logging to when reloading so we have to read in the logger configuration even if it has not changed so that the logs get opened again.
(closes issue #12665)
Reported by: DennisD

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2008-05-19 16:05:25 +00:00
Joshua Colp
9256e457f4 Merged revisions 117082 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r117082 | file | 2008-05-19 12:24:44 -0300 (Mon, 19 May 2008) | 14 lines

Merged revisions 117081 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r117081 | file | 2008-05-19 12:22:10 -0300 (Mon, 19 May 2008) | 6 lines

Make chan_h323 work with pwlib 1.12.0
(closes issue #12682)
Reported by: bamby
Patches:
      pwlib_nopipe.diff uploaded by bamby (license 430)

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2008-05-19 15:27:28 +00:00
Russell Bryant
27f6220a41 Merged revisions 116979 via svnmerge from
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r116979 | russell | 2008-05-18 22:44:28 -0500 (Sun, 18 May 2008) | 12 lines

Merged revisions 116978 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r116978 | russell | 2008-05-18 22:44:04 -0500 (Sun, 18 May 2008) | 4 lines

Avoid access of uninitialized memory.  This caused a bunch of crashes for me
while doing load testing of development branch where I'm working on some
performance improvements.

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2008-05-19 03:44:51 +00:00
Tilghman Lesher
0a6450103f Merged revisions 116948 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r116948 | tilghman | 2008-05-18 16:15:58 -0500 (Sun, 18 May 2008) | 4 lines

Add a set of text to the file astcanary uses to communicate back the main
Asterisk process, which explains the purpose for the file being there.  This
should assist people who find the file and wonder why it exists.

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2008-05-18 21:18:06 +00:00
Russell Bryant
d39753ec59 Merged revisions 116919 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r116919 | russell | 2008-05-18 14:58:10 -0500 (Sun, 18 May 2008) | 3 lines

Remove duplicate colon on Reason header
(closes issue #12678)

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2008-05-18 19:59:32 +00:00
Joshua Colp
c1375e6a26 Blocked revisions 116884 via svnmerge
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r116884 | file | 2008-05-17 16:39:35 -0300 (Sat, 17 May 2008) | 6 lines

Improve native transfers when a chain of IAX2 connections are in use.
(closes issue #7567)
Reported by: tjd
Patches:
      bug_7567_update_v2.diff uploaded by snuffy (license 35)

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2008-05-17 19:40:24 +00:00
Joshua Colp
ede214184b Blocked revisions 116848 via svnmerge
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r116848 | file | 2008-05-16 18:34:45 -0300 (Fri, 16 May 2008) | 2 lines

Try to fix attended transfers.

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2008-05-16 21:36:04 +00:00
Joshua Colp
d26ee10058 Merged revisions 116800 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r116800 | file | 2008-05-16 17:30:24 -0300 (Fri, 16 May 2008) | 12 lines

Merged revisions 116799 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r116799 | file | 2008-05-16 17:28:11 -0300 (Fri, 16 May 2008) | 4 lines

Check to make sure an RTP structure exists before calling ast_rtp_new_source on it.
(closes issue #12669)
Reported by: sbisker

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2008-05-16 21:35:33 +00:00
Matthew Fredrickson
7a6e7a7ec1 Merged revisions 116797 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r116797 | mattf | 2008-05-16 15:00:04 -0500 (Fri, 16 May 2008) | 1 line

Try to see if we can make our ringback situation a little better
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2008-05-16 20:03:31 +00:00
Tilghman Lesher
0d4da58df8 Merged revisions 116694 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r116694 | tilghman | 2008-05-15 17:05:47 -0500 (Thu, 15 May 2008) | 4 lines

Add an extra check in ast_strlen_zero, and make ast_assert() not print the
file, line, and function name twice.
(Closes issue #12650)

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2008-05-15 22:07:30 +00:00
Tilghman Lesher
ac19e24dae Merged revisions 116631 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r116631 | tilghman | 2008-05-15 12:58:22 -0500 (Thu, 15 May 2008) | 3 lines

Don't unload config on reload, when config has not changed.
(Closes issue #12652)

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2008-05-15 17:59:17 +00:00
Mark Michelson
25a67cf629 Blocked revisions 116592 via svnmerge
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r116592 | mmichelson | 2008-05-15 10:24:29 -0500 (Thu, 15 May 2008) | 9 lines

Modify externnotify to take the number of urgent voicemails as a final argument instead
of the string "Urgent" 

(closes issue #12660)
Reported by: jaroth
Patches:
      externnotify.patch uploaded by jaroth (license 50)


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2008-05-15 15:24:55 +00:00
Mark Michelson
c41cc86537 Blocked revisions 116590 via svnmerge
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r116590 | mmichelson | 2008-05-15 10:13:11 -0500 (Thu, 15 May 2008) | 8 lines

Prevent crashes from occurring due to a strcmp of a NULL pointer.

(closes issue #12661)
Reported by: jaroth
Patches:
      urgentcompare.patch uploaded by jaroth (license 50)


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2008-05-15 15:13:32 +00:00
Mark Michelson
47e26e5475 Blocked revisions 116522 via svnmerge
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r116522 | mmichelson | 2008-05-14 17:15:12 -0500 (Wed, 14 May 2008) | 8 lines

Adding a new option to Chanspy(). The 'd' option allows for the spy to
press DTMF digits to switch between spying modes. Pressing 4 activates spy mode,
pressing 5 activates whisper mode, and pressing 6 activates barge mode. Use of
this feature overrides the normal operation of DTMF numbers. 

This feature is courtesy of Switchvox.


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2008-05-14 22:16:56 +00:00
Joshua Colp
ecb963aee6 Blocked revisions 116471 via svnmerge
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r116471 | file | 2008-05-14 18:54:03 -0300 (Wed, 14 May 2008) | 2 lines

Fix pedanticness.

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2008-05-14 21:55:17 +00:00
Russell Bryant
8b3150e282 Merged revisions 116469 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r116469 | russell | 2008-05-14 16:40:43 -0500 (Wed, 14 May 2008) | 12 lines

Merged revisions 116463 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r116463 | russell | 2008-05-14 16:32:00 -0500 (Wed, 14 May 2008) | 4 lines

Add ast_assert(), which can be used to handle fatal errors.  It is only compiled
in if dev-mode is enabled, and only aborts if DO_CRASH is defined.
(inspired by issue #12650)

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Tilghman Lesher
6368b8d86e Merged revisions 116467 via svnmerge from
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r116467 | tilghman | 2008-05-14 16:39:06 -0500 (Wed, 14 May 2008) | 15 lines

Merged revisions 116466 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r116466 | tilghman | 2008-05-14 16:38:09 -0500 (Wed, 14 May 2008) | 7 lines

Avoid zombies when the channel exits before the AGI.
(closes issue #12648)
 Reported by: gkloepfer
 Patches: 
       20080514__bug12648.diff.txt uploaded by Corydon76 (license 14)
 Tested by: gkloepfer

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2008-05-14 21:39:47 +00:00
Joshua Colp
209887be27 Blocked revisions 116461 via svnmerge
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r116461 | file | 2008-05-14 18:11:49 -0300 (Wed, 14 May 2008) | 6 lines

Add a missing context unlock.
(closes issue #12649)
Reported by: ys
Patches:
      pbx.c.diff uploaded by ys (license 281)

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2008-05-14 21:12:44 +00:00
Jason Parker
5b30a3d75d Merged revisions 116410 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r116410 | qwell | 2008-05-14 15:43:26 -0500 (Wed, 14 May 2008) | 9 lines

Merged revisions 116409 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r116409 | qwell | 2008-05-14 15:43:08 -0500 (Wed, 14 May 2008) | 1 line

Document exitcontext in app_voicemail sample config
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2008-05-14 20:43:45 +00:00
Jason Parker
191c3196cc Merged revisions 116407 via svnmerge from
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r116407 | qwell | 2008-05-14 15:36:55 -0500 (Wed, 14 May 2008) | 9 lines

Voicemail "* exit" should not require an exitcontext to be specified.
The behavior in 1.4 was that it would use the current context if an exitcontext existed.

(closes issue #12605)
Reported by: kenjreno
Patches:
      12605-starexit.diff uploaded by qwell (license 4)
Tested by: file

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2008-05-14 20:37:22 +00:00
Joshua Colp
91d87c10a5 Merged revisions 116353 via svnmerge from
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r116353 | file | 2008-05-14 15:54:16 -0300 (Wed, 14 May 2008) | 12 lines

Merged revisions 116352 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r116352 | file | 2008-05-14 15:53:39 -0300 (Wed, 14 May 2008) | 4 lines

Add linux-gnueabi in.
(closes issue #12529)
Reported by: tzafrir

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2008-05-14 18:54:58 +00:00
Joshua Colp
a90f428994 Merged revisions 116350 via svnmerge from
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r116350 | file | 2008-05-14 15:25:54 -0300 (Wed, 14 May 2008) | 4 lines

Make the ldap version setting work without having both version and protocol set.
(closes issue #12613)
Reported by: suretec

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2008-05-14 18:27:06 +00:00
Tilghman Lesher
c348875fd0 Merged revisions 116298 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r116298 | tilghman | 2008-05-14 11:53:23 -0500 (Wed, 14 May 2008) | 15 lines

Merged revisions 116296 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r116296 | tilghman | 2008-05-14 11:46:48 -0500 (Wed, 14 May 2008) | 2 lines

Detect another way for a connection to have gone away.
(closes issue #12618)
 Reported by: ctooley
 Patches: 
       1.4-externalivr-test_fd.diff uploaded by ctooley (license 136)
       trunk-externalivr-test_fd.diff uploaded by ctooley (license 136)

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2008-05-14 17:01:53 +00:00
Olle Johansson
6d13b0bf81 Merged revisions 116234 via svnmerge from
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r116234 | oej | 2008-05-14 15:05:15 +0200 (Ons, 14 Maj 2008) | 11 lines

Merged revisions 116230 via svnmerge from 
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r116230 | oej | 2008-05-14 14:51:06 +0200 (Ons, 14 Maj 2008) | 3 lines

Accept text messages even with
Content-Type: text/plain;charset=Södermanländska

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2008-05-14 13:13:18 +00:00
Mark Michelson
923ab249ed Blocked revisions 116138 via svnmerge
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r116138 | mmichelson | 2008-05-13 19:20:05 -0500 (Tue, 13 May 2008) | 6 lines

Undo inadvertent changes to chan_skinny caused by the merging of urgent messaging
support.

Thanks to Damien Wedhorn for pointing out the problem.


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Mark Michelson
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r116089 | mmichelson | 2008-05-13 18:54:01 -0500 (Tue, 13 May 2008) | 20 lines

Merged revisions 116088 via svnmerge from 
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r116088 | mmichelson | 2008-05-13 18:47:49 -0500 (Tue, 13 May 2008) | 12 lines

A change to the way channel locks are handled when DEBUG_CHANNEL_LOCKS is defined.

After debugging a deadlock, it was noticed that when DEBUG_CHANNEL_LOCKS
is enabled in menuselect, the actual origin of channel locks is obscured
by the fact that all channel locks appear to happen in the function
ast_channel_lock(). This code change redefines ast_channel_lock to be a
macro which maps to __ast_channel_lock(), which then relays the proper
file name, line number, and function name information to the core lock
functions so that this information will be displayed in the case that
there is some sort of locking error or core show locks is issued.


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2008-05-13 23:58:34 +00:00
Russell Bryant
20925d6a81 Merged revisions 116039 via svnmerge from
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r116039 | russell | 2008-05-13 16:18:55 -0500 (Tue, 13 May 2008) | 32 lines

Merged revisions 116038 via svnmerge from 
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r116038 | russell | 2008-05-13 16:17:23 -0500 (Tue, 13 May 2008) | 24 lines

Fix a deadlock involving channel autoservice and chan_local that was debugged
and fixed by mmichelson and me.

We observed a system that had a bunch of threads stuck in ast_autoservice_stop().
The reason these threads were waiting around is because this function waits to
ensure that the channel list in the autoservice thread gets rebuilt before the
stop() function returns.  However, the autoservice thread was also locked, so
the autoservice channel list was never getting rebuilt.

The autoservice thread was stuck waiting for the channel lock on a local channel.
However, the local channel was locked by a thread that was stuck in the autoservice
stop function.

It turned out that the issue came down to the local_queue_frame() function in
chan_local.  This function assumed that one of the channels passed in as an
argument was locked when called.  However, that was not always the case.  There
were multiple cases in which this channel was not locked when the function was
called.  We fixed up chan_local to indicate to this function whether this channel
was locked or not.  The previous assumption had caused local_queue_frame() to
improperly return with the channel locked, where it would then never get unlocked.

(closes issue #12584)
(related to issue #12603)

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Russell Bryant
61348297c7 Merged revisions 116001 via svnmerge from
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r116001 | russell | 2008-05-13 16:07:59 -0500 (Tue, 13 May 2008) | 13 lines

Merged revisions 115990 via svnmerge from 
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r115990 | russell | 2008-05-13 16:05:57 -0500 (Tue, 13 May 2008) | 5 lines

Fix an issue that I noticed in autoservice while mmichelson and I were debugging
a different problem.  I noticed that it was theoretically possible for two threads
to attempt to start the autoservice thread at the same time.  This change makes the
process of starting the autoservice thread, thread-safe.

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2008-05-13 21:12:10 +00:00
Joshua Colp
e422336191 Merged revisions 115945 via svnmerge from
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r115945 | file | 2008-05-13 17:29:27 -0300 (Tue, 13 May 2008) | 12 lines

Merged revisions 115944 via svnmerge from 
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r115944 | file | 2008-05-13 17:28:23 -0300 (Tue, 13 May 2008) | 4 lines

Use the right flag to open the audio in non-blocking.
(closes issue #12616)
Reported by: nicklewisdigiumuser

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2008-05-13 20:30:21 +00:00
Matthew Fredrickson
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r115941 | mattf | 2008-05-13 15:18:04 -0500 (Tue, 13 May 2008) | 1 line

Need to clear calling_party_cat variable after we retrieve it
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Matthew Fredrickson
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r115939 | mattf | 2008-05-13 15:11:20 -0500 (Tue, 13 May 2008) | 1 line

Add support for receiving calling party category
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Tilghman Lesher
22a7310578 Merged revisions 115886 via svnmerge from
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r115886 | tilghman | 2008-05-13 13:38:11 -0500 (Tue, 13 May 2008) | 11 lines

Merged revisions 115884 via svnmerge from 
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r115884 | tilghman | 2008-05-13 13:36:13 -0500 (Tue, 13 May 2008) | 3 lines

If the socket dies (read returns 0=EOF), return immediately.
(Closes issue #12637)

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Russell Bryant
ceee0ce6d4 Blocked revisions 115850 via svnmerge
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r115850 | russell | 2008-05-13 12:42:17 -0500 (Tue, 13 May 2008) | 2 lines

Re-introduce proper error handling that was removed in recent commits.

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2008-05-13 17:48:06 +00:00
Russell Bryant
ba49e91504 Merged revisions 115847 via svnmerge from
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r115847 | russell | 2008-05-13 12:14:22 -0500 (Tue, 13 May 2008) | 2 lines

Initialize the start time in smdi_msg_wait.  Somehow this code got lost in trunk.

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Tilghman Lesher
773afb20d2 Blocked revisions 115813 via svnmerge
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r115813 | tilghman | 2008-05-12 15:34:38 -0500 (Mon, 12 May 2008) | 8 lines

Add a script which installs every package needed for a Debian install of
Asterisk, and includes possible support (to be contributed) for various other
distributions.
(closes issue #10523)
 Reported by: tzafrir
 Patches: 
       install_prereq_2 uploaded by tzafrir (license 46)

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2008-05-12 20:36:13 +00:00
Mark Michelson
bc90252b89 Merged revisions 115737 via svnmerge from
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r115737 | mmichelson | 2008-05-12 12:55:08 -0500 (Mon, 12 May 2008) | 15 lines

Merged revisions 115735 via svnmerge from 
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r115735 | mmichelson | 2008-05-12 12:51:14 -0500 (Mon, 12 May 2008) | 7 lines

If a thread holds no locks, do not print any information on the thread when issuing
a core show locks command. This will help to de-clutter output somewhat.

Russell said it would be fine to place this improvement in the 1.4 branch, so that's
why it's going here too.


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2008-05-12 17:57:03 +00:00
Jason Parker
fc07d2e1b7 Merged revisions 115705 via svnmerge from
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r115705 | qwell | 2008-05-12 11:35:50 -0500 (Mon, 12 May 2008) | 1 line

Correctly document state interface for AddQueueMember.  Discovered while looking at issue #12626.
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Brett Bryant
c9d6041a8b Merged revisions 115669 via svnmerge from
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r115669 | bbryant | 2008-05-12 10:17:32 -0500 (Mon, 12 May 2008) | 3 lines

A small change to fix iax2 native bridging.


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2008-05-12 15:18:10 +00:00
Matthew Fredrickson
77705a0808 Merged revisions 115600 via svnmerge from
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r115600 | mattf | 2008-05-10 22:23:05 -0500 (Sat, 10 May 2008) | 1 line

Add Zap MTP2 support to chan_zap
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2008-05-11 03:27:05 +00:00
Matthew Fredrickson
1e42ac8e05 Merged revisions 115598 via svnmerge from
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r115598 | mattf | 2008-05-10 21:19:21 -0500 (Sat, 10 May 2008) | 1 line

Open up audio channel when we get ACM on SS7 event
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2008-05-11 02:24:39 +00:00
Tilghman Lesher
cbdbbf7819 Merged revisions 115596 via svnmerge from
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r115596 | tilghman | 2008-05-10 09:19:41 -0500 (Sat, 10 May 2008) | 2 lines

Ensure that "calldate" is acceptable for a column name.

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2008-05-10 14:22:11 +00:00
Mark Michelson
5deec3d5dd Blocked revisions 115591 via svnmerge
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r115591 | mmichelson | 2008-05-09 17:36:50 -0500 (Fri, 09 May 2008) | 3 lines

Remove a debug line


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2008-05-09 22:37:23 +00:00
Mark Michelson
b1c9a7449a Blocked revisions 115588 via svnmerge
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r115588 | mmichelson | 2008-05-09 16:22:42 -0500 (Fri, 09 May 2008) | 19 lines

Adding support for "urgent" voicemail messages. Messages which are
marked "urgent" are considered to be higher priority than other messages
and so they will be played before any other messages in a user's mailbox.

There are two ways to leave an urgent message. 
1. send the 'U' option to VoiceMail().
2. Set review=yes in voicemail.conf. This will give instructions for 
   a caller to mark a message as urgent after the message has been recorded.

I have tested that this works correctly with file and ODBC storage, and James
Rothenberger (who wrote initial support for this feature) has tested its use
with IMAP storage.

(closes issue #11817)
Reported by: jaroth
	Based on branch http://svn.digium.com/svn/asterisk/team/jrothenberger/asterisk-urgent
Tested by: putnopvut, jaroth


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2008-05-09 21:24:02 +00:00
Brett Bryant
2fbf6f13bc Blocked revisions 115586 via svnmerge
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r115586 | bbryant | 2008-05-09 15:05:50 -0500 (Fri, 09 May 2008) | 2 lines

Update CHANGES file for previous commit of ENUM and TXCIDNAME changes.

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2008-05-09 20:06:30 +00:00
Brett Bryant
c2620074e8 Blocked revisions 115584 via svnmerge
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r115584 | bbryant | 2008-05-09 14:54:45 -0500 (Fri, 09 May 2008) | 15 lines

The following patch adds new options and alters the default behavior of the ENUM* functions. The TXCIDNAME lookup function has also gotten a 
new paramater. The new options for ENUM* functions include 'u', 's', 'i', and 'd' which return the full uri, trigger isn specific rewriting, look 
for branches into an infrastructure enum tree, or do a direct dns lookup of a number respectively. The new paramater for TXCIDNAME adds a 
zone-suffix argument for looking up caller id's in DNS that aren't e164.arpa.

This patch is based on the original code from otmar, modified by snuffy, and tested by jtodd, me, and others.

(closes issue #8089)
Reported by: otmar
Patches:
      20080508_bug8089-1.diff 
	- original code by otmar (license 480), 
	- revised by snuffy (license 35)
Tested by: oej, otmar, jtodd, Corydon76, snuffy, alexnikolov, bbryant

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2008-05-09 19:55:35 +00:00
Tilghman Lesher
f9ed11a788 Blocked revisions 115582 via svnmerge
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r115582 | tilghman | 2008-05-09 12:28:06 -0500 (Fri, 09 May 2008) | 7 lines

Allow a password change to be validated by an external script.
(closes issue #12090)
 Reported by: jaroth
 Patches: 
       vm-check-newpassword.diff.txt uploaded by mvanbaak (license 7)
       20080509__bug12090.diff.txt uploaded by Corydon76 (license 14)

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2008-05-09 17:29:25 +00:00
Joshua Colp
1d5d0976cc Merged revisions 115580 via svnmerge from
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r115580 | file | 2008-05-09 13:36:58 -0300 (Fri, 09 May 2008) | 10 lines

Merged revisions 115579 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r115579 | file | 2008-05-09 13:34:08 -0300 (Fri, 09 May 2008) | 2 lines

Improve res_ninit and res_ndestroy autoconf logic on the Darwin platform.

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2008-05-09 16:38:38 +00:00
Russell Bryant
3ec3e555c0 Merged revisions 115569 via svnmerge from
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r115569 | russell | 2008-05-08 14:20:35 -0500 (Thu, 08 May 2008) | 10 lines

Merged revisions 115568 via svnmerge from 
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r115568 | russell | 2008-05-08 14:19:50 -0500 (Thu, 08 May 2008) | 2 lines

Remove debug output.

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2008-05-08 19:21:08 +00:00
Russell Bryant
24d6ade06e Merged revisions 115566 via svnmerge from
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r115566 | russell | 2008-05-08 14:17:04 -0500 (Thu, 08 May 2008) | 41 lines

Merged revisions 115565 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r115565 | russell | 2008-05-08 14:15:25 -0500 (Thu, 08 May 2008) | 33 lines

Merged revisions 115564 via svnmerge from 
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r115564 | russell | 2008-05-08 14:14:04 -0500 (Thu, 08 May 2008) | 25 lines

Fix a race condition that bbryant just found while doing some IAX2 testing.
He was running Asterisk trunk running IAX2 calls through a few Asterisk boxes,
however, the audio was extremely choppy.  We looked at a packet trace and saw
a storm of INVAL and VNAK frames being sent from one box to another.

It turned out that what had happened was that one box tried to send a CONTROL
frame before the 3 way handshake had completed.  So, that frame did not include
the destination call number, because it didn't have it yet.  Part of our recent
work for security issues included an additional check to ensure that frames that
are supposed to include the destination call number have the correct one.  This
caused the frame to be rejected with an INVAL.  The frame would get retransmitted
for forever, rejected every time ...

This race condition exists in all versions that got the security changes,
in theory.  However, it is really only likely that this would cause a problem in
Asterisk trunk.  There was a control frame being sent (SRCUPDATE) at the _very_
beginning of the call, which does not exist in 1.2 or 1.4.  However, I am fixing
all versions that could potentially be affected by the introduced race condition.

These changes are what bbryant and I came up with to fix the issue.  Instead of
simply dropping control frames that get sent before the handshake is complete,
the code attempts to wait a little while, since in most cases, the handshake
will complete very quickly.  If it doesn't complete after yielding for a little
while, then the frame gets dropped.

........

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Russell Bryant
f42c651268 Merged revisions 115562 via svnmerge from
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r115562 | russell | 2008-05-08 11:14:08 -0500 (Thu, 08 May 2008) | 11 lines

Merged revisions 115561 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r115561 | russell | 2008-05-08 11:11:33 -0500 (Thu, 08 May 2008) | 3 lines

Don't give up on attempting an outbound registration if we receive a 408 Timeout.
(closes issue #12323)

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Russell Bryant
fd3eea1ba8 Merged revisions 115558 via svnmerge from
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r115558 | russell | 2008-05-08 10:38:27 -0500 (Thu, 08 May 2008) | 11 lines

Merged revisions 115557 via svnmerge from 
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r115557 | russell | 2008-05-08 10:37:49 -0500 (Thu, 08 May 2008) | 3 lines

remove postgres_cdr.sql, as the CDR schema is in realtime_pgsql.sql, as well
(closes issue #9676)

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2008-05-08 15:38:55 +00:00
Russell Bryant
eadd1e8063 Merged revisions 115555 via svnmerge from
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r115555 | russell | 2008-05-08 10:32:48 -0500 (Thu, 08 May 2008) | 11 lines

Merged revisions 115554 via svnmerge from 
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r115554 | russell | 2008-05-08 10:32:08 -0500 (Thu, 08 May 2008) | 3 lines

Don't exit the script if Asterisk is not running.
(closes issue #12611)

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Russell Bryant
ddc2373fe1 Merged revisions 115552 via svnmerge from
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r115552 | russell | 2008-05-08 10:26:49 -0500 (Thu, 08 May 2008) | 12 lines

Merged revisions 115551 via svnmerge from 
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r115551 | russell | 2008-05-08 10:24:54 -0500 (Thu, 08 May 2008) | 4 lines

Don't use a channel before checking for channel allocation failure.
(closes issue #12609)
Reported by: edantie

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Matthew Fredrickson
886d1aef09 Merged revisions 115548 via svnmerge from
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r115548 | mattf | 2008-05-08 10:04:45 -0500 (Thu, 08 May 2008) | 1 line

Remove unused code as well as demote an error message to a debug message
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2008-05-08 15:08:26 +00:00
Russell Bryant
4909bf569a Merged revisions 115546 via svnmerge from
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r115546 | russell | 2008-05-08 09:41:12 -0500 (Thu, 08 May 2008) | 12 lines

Merged revisions 115545 via svnmerge from 
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r115545 | russell | 2008-05-08 09:40:53 -0500 (Thu, 08 May 2008) | 4 lines

Use the same method for executing Asterisk as the rest of the script.
(closes issue #12611)
Reported by: b_plessis

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2008-05-08 14:41:34 +00:00
Russell Bryant
9a7f4fc302 Blocked revisions 115537 via svnmerge
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r115537 | russell | 2008-05-07 16:11:33 -0500 (Wed, 07 May 2008) | 10 lines

Fix up a problem that was introduced into the scheduler when it was converted
to use doubly linked lists.  The schedule() function had an optimization that
had it try to guess which direction would be better for the traversal to insert
the task into the scheduler queue.  However, if the code chose the path where
it traversed the queue in reverse, and the result was that the task should be
at the head of the queue, then the code would actually put it at the tail,
instead.

(Problem found by bbryant, debugged and fixed by bbryant and me)

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2008-05-07 21:12:13 +00:00
Tilghman Lesher
072d10f23b Blocked revisions 115535 via svnmerge
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r115535 | tilghman | 2008-05-07 15:22:09 -0500 (Wed, 07 May 2008) | 2 lines

Advance to next sounds release

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2008-05-07 20:23:11 +00:00
Tilghman Lesher
6ace54528a Blocked revisions 115525 via svnmerge
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r115525 | tilghman | 2008-05-07 13:40:21 -0500 (Wed, 07 May 2008) | 2 lines

Don't free the object on destroy, as astobj2 takes care of that for you

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2008-05-07 18:41:34 +00:00
Russell Bryant
f4eeb3481b Merged revisions 115523 via svnmerge from
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r115523 | russell | 2008-05-07 13:33:50 -0500 (Wed, 07 May 2008) | 6 lines

Only save a password if a username exists.

(closes issue #12600)
Reported By: suretec
Patch by me

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2008-05-07 18:35:52 +00:00
Russell Bryant
41a4500980 Merged revisions 115521 via svnmerge from
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r115521 | russell | 2008-05-07 13:30:12 -0500 (Wed, 07 May 2008) | 7 lines

Use the default that the log output claims will be used for the basedn

(closes issue #12599)
Reported by: suretec
Patches:
      12599.patch uploaded by juggie (license 24)

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2008-05-07 18:30:29 +00:00
Russell Bryant
ae3dda883f Merged revisions 115519 via svnmerge from
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r115519 | russell | 2008-05-07 13:24:51 -0500 (Wed, 07 May 2008) | 2 lines

Let chan_h323 build in dev mode

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2008-05-07 18:25:08 +00:00
Russell Bryant
fefdbda7a6 Blocked revisions 115515 via svnmerge
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r115515 | russell | 2008-05-07 12:38:36 -0500 (Wed, 07 May 2008) | 2 lines

re-add dlinkedlists.h to trunk, oops!

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2008-05-07 17:38:58 +00:00
Russell Bryant
09d71ed55f Merged revisions 115513 via svnmerge from
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r115513 | russell | 2008-05-07 12:28:19 -0500 (Wed, 07 May 2008) | 19 lines

Merged revisions 115512 via svnmerge from 
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r115512 | russell | 2008-05-07 11:24:09 -0500 (Wed, 07 May 2008) | 11 lines

Merged revisions 115511 via svnmerge from 
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r115511 | russell | 2008-05-07 11:22:49 -0500 (Wed, 07 May 2008) | 3 lines

Remove remnants of dlinkedlists.  I didn't actually use them in the final version
of my IAX2 improvements.

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2008-05-07 17:36:44 +00:00
Tilghman Lesher
fa1a079f43 Merged revisions 115509 via svnmerge from
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r115509 | tilghman | 2008-05-07 08:49:15 -0500 (Wed, 07 May 2008) | 2 lines

Update typos in description fields
(closes issue #12598)
 Reported by: suretec
 Patches: 
       asterisk_schema_changes.patch uploaded by suretec (license 70)

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2008-05-07 13:49:50 +00:00
Joshua Colp
f663ee9346 Blocked revisions 115507 via svnmerge
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r115507 | file | 2008-05-07 10:41:25 -0300 (Wed, 07 May 2008) | 4 lines

Remove redundant header getting.
(closes issue #12597)
Reported by: hooi

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2008-05-07 13:42:20 +00:00
Mark Michelson
7a9f945fc7 Blocked revisions 115473 via svnmerge
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r115473 | mmichelson | 2008-05-06 15:15:02 -0500 (Tue, 06 May 2008) | 3 lines

Get refcounter to build with LOW_MEMORY defined


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2008-05-06 20:15:46 +00:00
Jason Parker
31891c5ad8 Merged revisions 115423 via svnmerge from
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r115423 | qwell | 2008-05-06 14:55:45 -0500 (Tue, 06 May 2008) | 23 lines

Merged revisions 115422 via svnmerge from 
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r115422 | qwell | 2008-05-06 14:55:29 -0500 (Tue, 06 May 2008) | 15 lines

Merged revisions 115421 via svnmerge from 
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r115421 | qwell | 2008-05-06 14:54:57 -0500 (Tue, 06 May 2008) | 7 lines

read requires an argument on some non-bash shells

(closes issue #12593)
Reported by: bkruse
Patches:
      getilbc.sh_12593_v1.diff uploaded by bkruse (license 132)

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2008-05-06 19:56:00 +00:00
Jason Parker
4b4910e35b Merged revisions 115419 via svnmerge from
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r115419 | qwell | 2008-05-06 14:38:44 -0500 (Tue, 06 May 2008) | 15 lines

Merged revisions 115418 via svnmerge from 
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r115418 | qwell | 2008-05-06 14:34:58 -0500 (Tue, 06 May 2008) | 7 lines

Switch to using ast_random() rather than just rand().
This does not fix the bug reported, but I believe it is correct.

(from issue #12446)
Patches:
      bug_12446.diff uploaded by snuffy (license 35)

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2008-05-06 19:39:11 +00:00
Tilghman Lesher
12c3dd0067 Merged revisions 115416 via svnmerge from
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r115416 | tilghman | 2008-05-06 14:32:29 -0500 (Tue, 06 May 2008) | 10 lines

Merged revisions 115415 via svnmerge from 
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r115415 | tilghman | 2008-05-06 14:31:39 -0500 (Tue, 06 May 2008) | 2 lines

Don't print the terminating NUL.  (Closes issue #12589)

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2008-05-06 19:33:14 +00:00
Russell Bryant
257786c0bc Blocked revisions 115344 via svnmerge
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r115344 | russell | 2008-05-06 10:14:55 -0500 (Tue, 06 May 2008) | 2 lines

Change some NOTICE log messages to debug.

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2008-05-06 15:15:52 +00:00
Joshua Colp
33e5f9e766 Merged revisions 115342 via svnmerge from
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r115342 | file | 2008-05-06 10:55:44 -0300 (Tue, 06 May 2008) | 10 lines

Merged revisions 115341 via svnmerge from 
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r115341 | file | 2008-05-06 10:54:15 -0300 (Tue, 06 May 2008) | 2 lines

Add in missing argument.

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Tilghman Lesher
a12adeaa22 Merged revisions 115334 via svnmerge from
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r115334 | tilghman | 2008-05-05 18:00:31 -0500 (Mon, 05 May 2008) | 15 lines

Merged revisions 115333 via svnmerge from 
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r115333 | tilghman | 2008-05-05 17:50:31 -0500 (Mon, 05 May 2008) | 7 lines

Separate verbose output from CLI output, by using a preamble.
(closes issue #12402)
 Reported by: Corydon76
 Patches: 
       20080410__no_verbose_in_rx_output.diff.txt uploaded by Corydon76 (license 14)
       20080501__no_verbose_in_rx_output__1.4.diff.txt uploaded by Corydon76 (license 14)

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2008-05-05 23:01:22 +00:00
Joshua Colp
0ed86b307e Merged revisions 115328 via svnmerge from
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r115328 | file | 2008-05-05 19:13:57 -0300 (Mon, 05 May 2008) | 10 lines

Merged revisions 115327 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r115327 | file | 2008-05-05 19:10:05 -0300 (Mon, 05 May 2008) | 2 lines

Make sure that either the main speex library contains preprocess functions or that speexdsp does. If both fail then speex stuff can not be built.

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2008-05-05 22:17:05 +00:00
Mark Michelson
cd04c3207d Merged revisions 115329 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r115329 | mmichelson | 2008-05-05 17:14:06 -0500 (Mon, 05 May 2008) | 15 lines

#execing the same file multiple times led to warning messages saying that the same file was
being #included twice. This was due to the fact that #exec created a temporary file which
was then #included. The name of the temporary file was the name of the #exec'd file, with
the Unix timestamp and thread ID concatenated. The issue was that if multiple #exec statements
of the same file were reached in the same second, then the result was that the temporary files
would have duplicate names. To resolve this, the temporary file now has microsecond resolution
for the timestamp portion.

(closes issue #12574)
Reported by: jmls
Patches:
      12574.patch uploaded by putnopvut (license 60)
Tested by: jmls, putnopvut


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2008-05-05 22:14:42 +00:00
Russell Bryant
57a35ae1b3 Blocked revisions 115324 via svnmerge
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r115324 | russell | 2008-05-05 17:01:56 -0500 (Mon, 05 May 2008) | 2 lines

Simplify code by using a taskprocessor for dispatching events in the Asterisk core.

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2008-05-05 22:02:27 +00:00
Mark Michelson
cb62c9cba8 Merged revisions 115321 via svnmerge from
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r115321 | mmichelson | 2008-05-05 16:43:21 -0500 (Mon, 05 May 2008) | 21 lines

Merged revisions 115320 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r115320 | mmichelson | 2008-05-05 16:41:34 -0500 (Mon, 05 May 2008) | 13 lines

Don't consider a caller "handled" until the caller is bridged with
a queue member. There was too much of an opportunity for the member
to hang up (either during a delay, announcement, or overly long
agi) between the time that he answered the phone and the time when
he actually was bridged with the caller. The consequence of this
was that if the member hung up in that interval, then proper
abandonment details would not be noted in the queue log if the caller
were to hang up at any point after the member hangup.

(closes issue #12561)
Reported by: ablackthorn


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2008-05-05 21:44:04 +00:00
Russell Bryant
b46f8226e1 Merged revisions 115315 via svnmerge from
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r115315 | russell | 2008-05-05 15:28:17 -0500 (Mon, 05 May 2008) | 2 lines

Remove my rant, since I have now replaced the rant with code.

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2008-05-05 20:28:37 +00:00
Tilghman Lesher
c8753fb014 Merged revisions 115309 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r115309 | tilghman | 2008-05-05 14:57:28 -0500 (Mon, 05 May 2008) | 10 lines

Merged revisions 115308 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r115308 | tilghman | 2008-05-05 14:55:55 -0500 (Mon, 05 May 2008) | 2 lines

Err, the documentation on the return value of ast_odbc_backslash_is_escape is exactly backwards.

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2008-05-05 19:58:25 +00:00
Russell Bryant
9a981dfa35 Merged revisions 115305 via svnmerge from
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r115305 | russell | 2008-05-05 14:50:24 -0500 (Mon, 05 May 2008) | 13 lines

Merged revisions 115304 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r115304 | russell | 2008-05-05 14:49:25 -0500 (Mon, 05 May 2008) | 5 lines

Avoid putting opaque="" in Digest authentication.  This patch came from switchvox.
It fixes authentication with Primus in Canada, and has been in use for a very long
time without causing problems with any other providers.
(closes issue AST-36)

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2008-05-05 19:50:43 +00:00
Tilghman Lesher
dfb7b247c0 Merged revisions 115302 via svnmerge from
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r115302 | tilghman | 2008-05-05 14:42:36 -0500 (Mon, 05 May 2008) | 2 lines

Note change for ExecIf syntax (caught by jmls on IRC)

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2008-05-05 19:43:35 +00:00
Joshua Colp
6e9f6a36e5 Blocked revisions 115290 via svnmerge
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r115290 | file | 2008-05-05 10:52:44 -0300 (Mon, 05 May 2008) | 6 lines

Document the 'B' option of app_chanspy.
(closes issue #12582)
Reported by: IgorG
Patches:
      app_chanspy_B_option.diff uploaded by IgorG (license 20)

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2008-05-05 13:53:26 +00:00
Kevin P. Fleming
b00df35ba7 Merged revisions 115288 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r115288 | kpfleming | 2008-05-05 05:55:09 -0500 (Mon, 05 May 2008) | 2 lines

clarify wording

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2008-05-05 10:55:40 +00:00
Tilghman Lesher
87dfc2f6f8 Merged revisions 115286 via svnmerge from
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r115286 | tilghman | 2008-05-04 22:25:35 -0500 (Sun, 04 May 2008) | 15 lines

Merged revisions 115285 via svnmerge from 
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r115285 | tilghman | 2008-05-04 22:22:25 -0500 (Sun, 04 May 2008) | 7 lines

When starting Asterisk, bug out if Asterisk is already running.
(closes issue #12525)
 Reported by: explidous
 Patches: 
       20080428__bug12525.diff.txt uploaded by Corydon76 (license 14)
 Tested by: mvanbaak

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2008-05-05 03:26:35 +00:00
Joshua Colp
b4ca837667 Merged revisions 115283 via svnmerge from
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r115283 | file | 2008-05-03 23:11:01 -0300 (Sat, 03 May 2008) | 10 lines

Merged revisions 115282 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r115282 | file | 2008-05-03 23:09:44 -0300 (Sat, 03 May 2008) | 2 lines

Expand the test function for GCC attributes so that more complex attributes are properly recognized.

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2008-05-04 02:12:12 +00:00
Joshua Colp
7ccc4fe461 Merged revisions 115280 via svnmerge from
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r115280 | file | 2008-05-03 22:52:00 -0300 (Sat, 03 May 2008) | 10 lines

Merged revisions 115279 via svnmerge from 
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r115279 | file | 2008-05-03 22:50:59 -0300 (Sat, 03 May 2008) | 2 lines

For my next trick I will make these work with what our autoconf header file gives us.

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2008-05-04 01:53:02 +00:00
Joshua Colp
a33ad4ebf0 Merged revisions 115277 via svnmerge from
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r115277 | file | 2008-05-03 22:45:21 -0300 (Sat, 03 May 2008) | 10 lines

Merged revisions 115276 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r115276 | file | 2008-05-03 22:43:26 -0300 (Sat, 03 May 2008) | 2 lines

Treat warnings as errors when checking if a GCC attribute exists. We have to do this as GCC will just ignore the attribute and pop up a warning, it won't actually fail to compile.

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2008-05-04 01:47:11 +00:00
Dwayne M. Hubbard
a03eef2580 block voicemail mwi notification subscriptions taskprocessor
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2008-05-03 04:25:24 +00:00
Dwayne M. Hubbard
14251c61ed block pbx taskprocessor
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2008-05-03 04:19:40 +00:00
Dwayne M. Hubbard
84997860ed block app_queue taskprocessor
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2008-05-03 04:04:18 +00:00
Dwayne M. Hubbard
643898e91b blocked taskprocessors
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2008-05-03 03:51:55 +00:00
Mark Michelson
e85fbb0aad Blocked revisions 115199 via svnmerge
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r115199 | mmichelson | 2008-05-02 09:51:59 -0500 (Fri, 02 May 2008) | 3 lines

Make res/snmp/agent.c build


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2008-05-02 14:55:24 +00:00
Mark Michelson
78673e20fc Merged revisions 115197 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r115197 | mmichelson | 2008-05-02 09:28:55 -0500 (Fri, 02 May 2008) | 14 lines

Merged revisions 115196 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r115196 | mmichelson | 2008-05-02 09:28:19 -0500 (Fri, 02 May 2008) | 6 lines

Clarify a comment that was, well, just wrong. It turns out that
ignoring the way that macros expand. Instead, I have clarified in the
comment why the macro will work even if the scheduler id for the
task to be deleted changes during the execution of the macro.


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Tilghman Lesher
940eb5779e Blocked revisions 115159 via svnmerge
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r115159 | tilghman | 2008-05-01 21:56:39 -0500 (Thu, 01 May 2008) | 2 lines

Okay, maybe FreeBSD will like this better.

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Tilghman Lesher
e6ea155b9a Blocked revisions 115157 via svnmerge
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r115157 | tilghman | 2008-05-01 21:33:04 -0500 (Thu, 01 May 2008) | 2 lines

Add attributes to various API calls, to help track down bugs (and remove a deprecated function)

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2008-05-02 02:34:53 +00:00
Tilghman Lesher
23d923acc3 Merged revisions 115104 via svnmerge from
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r115104 | tilghman | 2008-05-01 18:21:13 -0500 (Thu, 01 May 2008) | 10 lines

Merged revisions 115102 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r115102 | tilghman | 2008-05-01 18:20:25 -0500 (Thu, 01 May 2008) | 2 lines

Change the comment of deprecated to an actual compiler deprecation

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2008-05-01 23:22:02 +00:00
Brett Bryant
5d1b5862a6 Blocked revisions 115078 via svnmerge
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r115078 | bbryant | 2008-05-01 18:09:08 -0500 (Thu, 01 May 2008) | 2 lines

Add two new console commands "pri show version" and "ss7 show version" that will show the version of each library respectively.

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Tilghman Lesher
560342760a Blocked revisions 115076 via svnmerge
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r115076 | tilghman | 2008-05-01 18:06:23 -0500 (Thu, 01 May 2008) | 7 lines

Modify TIMEOUT() to be accurate down to the millisecond.
(closes issue #10540)
 Reported by: spendergrass
 Patches: 
       20080417__bug10540.diff.txt uploaded by Corydon76 (license 14)
 Tested by: blitzrage

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Russell Bryant
d518bd9091 Blocked revisions 115021 via svnmerge
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r115021 | russell | 2008-05-01 14:05:36 -0500 (Thu, 01 May 2008) | 9 lines

Merge changes from team/russell/smdi-msg-searching

This commit adds some new features to the SMDI_MSG_RETRIEVE() dialplan function.
Previously, this function only allowed searching by the forwarding station.
I have added some options to allow you to also search for messages in the queue
by the message desk terminal ID, as well as the message desk number.

This originally came up as a suggestion on the asterisk-dev mailing list.

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2008-05-01 19:05:56 +00:00
Tilghman Lesher
335111dd89 Merged revisions 115018 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r115018 | tilghman | 2008-05-01 14:00:18 -0500 (Thu, 01 May 2008) | 14 lines

Merged revisions 115017 via svnmerge from 
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r115017 | tilghman | 2008-05-01 13:59:08 -0500 (Thu, 01 May 2008) | 6 lines

'#' is another reserved character for URIs that also needs to be escaped.
(closes issue #10543)
 Reported by: blitzrage
 Patches: 
       20080418__bug10543.diff.txt uploaded by Corydon76 (license 14)

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Brett Bryant
25ea5b5dea Blocked revisions 114977 via svnmerge
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r114977 | bbryant | 2008-05-01 13:28:38 -0500 (Thu, 01 May 2008) | 4 lines

Add "read" capability to new libspeex functions in func_speex.c.

func_speex.c is based on contributions from Switchvox.

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2008-05-01 18:29:02 +00:00
Russell Bryant
7c57c0816f Merged revisions 114931 via svnmerge from
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r114931 | russell | 2008-05-01 12:28:25 -0500 (Thu, 01 May 2008) | 4 lines

Clarify the deprecation notice about Macro() to note that it will not be removed
for the sake of backwards compatibility, since it is a non-trivial task to convert
existing large dialplans that depend on Macro() to use GoSub(), instead.

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Brett Bryant
c64f4ac817 Blocked revisions 114926 via svnmerge
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r114926 | bbryant | 2008-05-01 11:57:19 -0500 (Thu, 01 May 2008) | 5 lines

Add two new dialplan functions from libspeex for applying audio gain control 
and denoising to a channel, AGC() and DENOISE(). Also included, is a change 
to the audiohook API to add a new function (ast_audiohook_remove) that can 
remove an audiohook from a channel before it is detached.

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2008-05-01 16:58:30 +00:00
Jason Parker
76e257a8f9 Merged revisions 114922 via svnmerge from
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r114922 | qwell | 2008-05-01 11:49:24 -0500 (Thu, 01 May 2008) | 10 lines

Allow dringXrange to properly default to 10, as was done in 1.4.

dringXrange is a new feature that was added, and it attempted to default, but only when the option was specified.

(closes issue #12536)
Reported by: bjm
Patches:
      12536-dringXrange.diff uploaded by qwell (license 4)
Tested by: bjm

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Joshua Colp
87b670f1ff Blocked revisions 114912 via svnmerge
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r114912 | file | 2008-04-30 17:51:17 -0300 (Wed, 30 Apr 2008) | 2 lines

Add support for specifying the registration expiry on a per registration basis in the register line. This comes from a Switchvox patch. (issue AST-24)

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2008-04-30 20:51:45 +00:00
Russell Bryant
9f1ccdbea3 Add the dlinkedlists implementation from trunk
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2008-04-30 20:20:46 +00:00
Mark Michelson
fe72ca1329 Make 1.6.0 compile
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2008-04-30 20:17:20 +00:00
Mark Michelson
a3e1649a9a Blocked revisions 114906 via svnmerge
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r114906 | mmichelson | 2008-04-30 14:30:41 -0500 (Wed, 30 Apr 2008) | 12 lines

Adding new configuration options to app_queue. This adds two new values
to announce-position, "limit" and "more," as well as a new option, 
announce-position-limit. For more information on the use of these options,
see CHANGES or configs/queues.conf.sample.

(closes issue #10991)
Reported by: slavon
Patches:
      app_q.diff uploaded by slavon (license 288)
Tested by: slavon, putnopvut


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2008-04-30 19:31:19 +00:00
Tilghman Lesher
b7444c2679 Blocked revisions 114904 via svnmerge
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r114904 | tilghman | 2008-04-30 14:21:04 -0500 (Wed, 30 Apr 2008) | 8 lines

Lock around variables retrieved, and copy the values, if they stay persistent,
since another thread could remove them.
(closes issue #12541)
 Reported by: snuffy
 Patches: 
       bug_12156_apps.diff uploaded by snuffy (license 35)
       Several additional changes by me

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2008-04-30 19:21:41 +00:00
Olle Johansson
71c72192ea Merged revisions 114899 via svnmerge from
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r114899 | oej | 2008-04-30 18:55:49 +0200 (Ons, 30 Apr 2008) | 15 lines

Merged revisions 114890 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114890 | oej | 2008-04-30 18:23:17 +0200 (Ons, 30 Apr 2008) | 7 lines

Don't crash on bad SIP replys.
Fix created in Huntsville together with Mark M (putnopvut)

(closes issue #12363)
Reported by: jvandal
Tested by: putnopvut, oej

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2008-04-30 17:06:18 +00:00
Russell Bryant
567cb9a041 Merged revisions 114892 via svnmerge from
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r114892 | russell | 2008-04-30 11:34:24 -0500 (Wed, 30 Apr 2008) | 36 lines

Merged revisions 114891 via svnmerge from 
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r114891 | russell | 2008-04-30 11:30:01 -0500 (Wed, 30 Apr 2008) | 28 lines

Merge changes from team/russell/iax2_find_callno and iax2_find_callno_1.4

These changes address a critical performance issue introduced in the latest
release.  The fix for the latest security issue included a change that made
Asterisk randomly choose call numbers to make them more difficult to guess by
attackers.  However, due to some inefficient (this is by far, an understatement)
code, when Asterisk chose high call numbers, chan_iax2 became unusable after
just a small number of calls.  On a small embedded platform, it would not be
able to handle a single call.  On my Intel Core 2 Duo @ 2.33 GHz, I couldn't
run more than about 16 IAX2 channels.  Ouch.

These changes address some performance issues of the find_callno() function
that have bothered me for a very long time.  On every incoming media frame,
it iterated through every possible call number trying to find a matching
active call.  This involved a mutex lock and unlock for each call number
checked.  So, if the random call number chosen was 20000, then every media
frame would cause 20000 locks and unlocks.  Previously, this problem was
not as obvious since Asterisk always chose the lowest call number it could.

A second container for IAX2 pvt structs has been added.  It is an astobj2
hash table.  When we know the remote side's call number, the pvt goes into
the hash table with a hash value of the remote side's call number.  Then,
lookups for incoming media frames are a very fast hash lookup instead of an
absolutely insane array traversal.

In a quick test, I was able to get more than 3600% more IAX2 channels
on my machine with these changes.

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2008-04-30 16:41:33 +00:00
Jeff Peeler
d4f34710bf Merged revisions 114888 via svnmerge from
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r114888 | jpeeler | 2008-04-30 11:14:43 -0500 (Wed, 30 Apr 2008) | 3 lines

Fixes a bug where if a stream monitor thread was not created (caused from failure of opening or starting the stream) pthread_cancel was called with an invalid thread ID.


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Kevin P. Fleming
ced1cda76c Merged revisions 114884 via svnmerge from
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r114884 | kpfleming | 2008-04-30 09:49:51 -0500 (Wed, 30 Apr 2008) | 10 lines

Merged revisions 114880 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114880 | kpfleming | 2008-04-30 09:46:57 -0500 (Wed, 30 Apr 2008) | 2 lines

use the ARRAY_LEN macro for indexing through the iaxs/iaxsl arrays so that the size of the arrays can be adjusted in one place, and change the size of the arrays from 32768 calls to 2048 calls when LOW_MEMORY is defined

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Kevin P. Fleming
483c30870a Merged revisions 114876 via svnmerge from
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r114876 | kpfleming | 2008-04-30 07:15:43 -0500 (Wed, 30 Apr 2008) | 10 lines

Merged revisions 114875 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114875 | kpfleming | 2008-04-30 07:14:07 -0500 (Wed, 30 Apr 2008) | 2 lines

pay attention to *all* header files for dependency tracking, not just the local ones (inspired by r578 of asterisk-addons by tilghman)

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Jeff Peeler
d868e4643c Merged revisions 114866 via svnmerge from
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r114866 | jpeeler | 2008-04-29 17:54:14 -0500 (Tue, 29 Apr 2008) | 2 lines

Fixes a problem where all the templates were marked as dead no matter what. The templates should only be marked as dead if a configuration file has been successfully loaded and has changes. Bug found while making API documentation for 1.6.0.

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Mark Michelson
a4bfe4880e Blocked revisions 114857 via svnmerge
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r114857 | mmichelson | 2008-04-29 16:07:36 -0500 (Tue, 29 Apr 2008) | 15 lines

Patching app_chanspy to jibe better with what is documented. This allows for 
a colon-delimited list of spygroups to be specified when calling the ChanSpy application
with the 'g' option. Prior to this, you could only specify a single group when using the
'g' option.

I also have upped the maximum number of spygroups to 128 and added a #define so that this
can be easily increased or decreased later.

(closes issue #12497)
Reported by: jsmith
Patches:
      app_chanspy_multiple_groups_v2.patch uploaded by jsmith (license 15)
Tested by: atis, jvandal


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2008-04-29 21:09:07 +00:00
Mark Michelson
ec4819f368 Merged revisions 114849 via svnmerge from
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r114849 | mmichelson | 2008-04-29 14:42:04 -0500 (Tue, 29 Apr 2008) | 22 lines

Merged revisions 114848 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114848 | mmichelson | 2008-04-29 14:40:06 -0500 (Tue, 29 Apr 2008) | 14 lines

Use the MACRO_CONTEXT and MACRO_EXTEN channel variables instead of the channel's macrocontext
and macroexten fields. This is needed because if macros are daisy-chained, the incorrect 
context and extension are placed on the new channel. I also added locking to the channel prior
to accessing these variables as noted in trunk's janitor project file.


(closes issue #12549)
Reported by: darren1713
Patches:
      app_queue.c.macroextenpatch uploaded by darren1713 (license 116)
	       (with modifications from me)
Tested by: putnopvut


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2008-04-29 19:42:43 +00:00
Kevin P. Fleming
d5b5113743 bug is not present in this branch
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Mark Michelson
75bb9a755a Blocked revisions 114841 via svnmerge
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r114841 | mmichelson | 2008-04-29 13:48:26 -0500 (Tue, 29 Apr 2008) | 4 lines

Make app_directory dependent on app_voicemail. This is because the function
which says the person's name is handled inside app_voicemail now.


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Mark Michelson
b47f8fc35d Blocked revisions 114834 via svnmerge
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r114834 | mmichelson | 2008-04-29 12:56:13 -0500 (Tue, 29 Apr 2008) | 8 lines

Since there is now a globally available function for saying someone's name, a LOT of
functions in app_directory can be removed since the ODBC-specific lookups are accomplished
within app_voicemail. This change greatly reduces the amount of lines in app_directory that
were solely for the purpose of looking up a name when ODBC_STORAGE is specified for voicemail.

This commit also makes the name-saying interruptable via DTMF.


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Mark Michelson
a78efc7fcd Blocked revisions 114832 via svnmerge
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r114832 | mmichelson | 2008-04-29 12:31:26 -0500 (Tue, 29 Apr 2008) | 3 lines

Fix a crash happening in app_directory. This crash would occur if a users.conf existed.


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Jason Parker
fff476985f Merged revisions 114830 via svnmerge from
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r114830 | qwell | 2008-04-29 12:10:55 -0500 (Tue, 29 Apr 2008) | 9 lines

Merged revisions 114829 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114829 | qwell | 2008-04-29 12:08:55 -0500 (Tue, 29 Apr 2008) | 1 line

Change warning message to debug, since there are cases where 0 results is perfectly fine.
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Kevin P. Fleming
3df627a0fd Merged revisions 114824 via svnmerge from
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r114824 | kpfleming | 2008-04-29 07:54:31 -0500 (Tue, 29 Apr 2008) | 18 lines

Merged revisions 114823 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114823 | kpfleming | 2008-04-29 07:53:12 -0500 (Tue, 29 Apr 2008) | 10 lines

Merged revisions 114822 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r114822 | kpfleming | 2008-04-29 07:52:32 -0500 (Tue, 29 Apr 2008) | 2 lines

stop script from appending source code if run multiple times

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Mark Michelson
88645b3b3a Blocked revisions 114813 via svnmerge
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r114813 | mmichelson | 2008-04-28 17:38:07 -0500 (Mon, 28 Apr 2008) | 10 lines

Adding a new option 'n' to app_chanspy. This option allows for the name of the spied-on
party to be spoken instead of the channel name or number.

This was accomplished by adding a new function pointer to point to a function in app_voicemail
which retrieves the name file and plays it. This makes for an easy way that applications may play
a user's name should it be necessary. app_directory, in particular, can be simplified greatly by
this change.

This change comes as a suggestion from Switchvox, which already has this feature. AST-23

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Matthew Fredrickson
d27eb35d56 Merged revisions 114776 via svnmerge from
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r114776 | mattf | 2008-04-28 12:00:38 -0500 (Mon, 28 Apr 2008) | 1 line

Fix deadlock issue in chan_zap with libss7 due to channel variables being set with the channel pvt lock being held. #12512
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2008-04-28 17:04:23 +00:00
Tilghman Lesher
392c610211 Blocked revisions 114773 via svnmerge
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r114773 | tilghman | 2008-04-28 11:37:45 -0500 (Mon, 28 Apr 2008) | 8 lines

Add incomplete matching to PBX code and app_dial
(closes issue #12351)
 Reported by: Corydon76
 Patches: 
       20080402__pbx_incomplete__3.diff.txt uploaded by Corydon76 (license 14)
       pbx_incomplete_with_timeout.diff uploaded by fabled (license 448)
 Tested by: Corydon76, fabled

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2008-04-28 16:39:34 +00:00
Joshua Colp
39767a7948 Merged revisions 114713 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r114713 | file | 2008-04-28 10:42:13 -0300 (Mon, 28 Apr 2008) | 2 lines

Update autoconf logic with latest API change for libss7.

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2008-04-28 13:44:24 +00:00
Tilghman Lesher
4ab60760e3 Merged revisions 114709 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r114709 | tilghman | 2008-04-27 23:53:20 -0500 (Sun, 27 Apr 2008) | 13 lines

Merged revisions 114708 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114708 | tilghman | 2008-04-27 23:47:39 -0500 (Sun, 27 Apr 2008) | 5 lines

When modules are embedded, they take on a different name, without the ".so"
extension.  Specifically check for this name, when we're checking if a module
is loaded.
(Closes issue #12534)

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2008-04-28 04:54:13 +00:00
Tilghman Lesher
32ec7da129 Blocked revisions 114706 via svnmerge
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r114706 | tilghman | 2008-04-27 23:30:02 -0500 (Sun, 27 Apr 2008) | 2 lines

Fix breakage caused by #12028.  (Closes issue #12535)

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2008-04-28 04:32:44 +00:00
Michiel van Baak
ba2d863314 Merged revisions 114700 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

Merged to 1.6 because it fixes a crash.

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r114700 | mvanbaak | 2008-04-27 17:17:18 +0200 (Sun, 27 Apr 2008) | 8 lines

Make MWI in chan_skinny event based modeled after chan_zap and chan_mgcp.

(closes issue #12214)
Reported by: DEA
Patches:
      chan_skinny-vm-events-v3.txt uploaded by DEA (license 3)
	  Tested by: DEA and me

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2008-04-27 15:20:45 +00:00
Sean Bright
ae5149f4fe Merged revisions 114696 via svnmerge from
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r114696 | seanbright | 2008-04-26 21:28:32 -0400 (Sat, 26 Apr 2008) | 13 lines

Merged revisions 114695 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114695 | seanbright | 2008-04-26 21:26:15 -0400 (Sat, 26 Apr 2008) | 5 lines

When we don't explicitly pass a path to the --with-tds configure option, we
may end up finding tds.h in /usr/local/include instead of /usr/include.  If
this happens, the grep that looks for the version (from tdsver.h) will fail
and we'll have some problems during the build.

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2008-04-27 01:30:44 +00:00
Tilghman Lesher
cec3fba722 Blocked revisions 114692 via svnmerge
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r114692 | tilghman | 2008-04-26 10:08:51 -0500 (Sat, 26 Apr 2008) | 2 lines

Unleak reference

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2008-04-26 15:09:31 +00:00
Tilghman Lesher
d3a72a96af Merged revisions 114690 via svnmerge from
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r114690 | tilghman | 2008-04-26 08:17:19 -0500 (Sat, 26 Apr 2008) | 14 lines

Merged revisions 114689 via svnmerge from 
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r114689 | tilghman | 2008-04-26 08:15:21 -0500 (Sat, 26 Apr 2008) | 6 lines

Clicking forward without selecting a message leaves an errant .lock file.
(closes issue #12528)
 Reported by: pukepail
 Patches: 
       patch.diff uploaded by pukepail (license 431)

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2008-04-26 13:18:20 +00:00
Tilghman Lesher
238df44b01 Blocked revisions 114683 via svnmerge
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r114683 | tilghman | 2008-04-25 21:48:56 -0500 (Fri, 25 Apr 2008) | 8 lines

Add 'sip qualify peer <peer>' command (with AMI SIPqualifypeer)
(closes issue #12524)
 Reported by: ctooley
 Patches: 
       sip_qualify_peer.diff.2 uploaded by ctooley (license 136)
       some modifications for trunk by Corydon76
 Tested by: Corydon76

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2008-04-26 02:49:54 +00:00
Mark Michelson
0ef5d9d15b Blocked revisions 114678 via svnmerge
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r114678 | mmichelson | 2008-04-25 17:24:32 -0500 (Fri, 25 Apr 2008) | 11 lines

Adding a new option, 'B' to app_chanspy. This option allows the spy to
barge on the call. It is like the existing whisper option, except that
it allows the spy to talk to both sides of the conversation on which
he is spying.

This feature has existed in Switchvox, and this merges the functionality
into Asterisk.

(AST-32)


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2008-04-25 22:25:48 +00:00
Russell Bryant
49ba484a53 Merged revisions 114676 via svnmerge from
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r114676 | russell | 2008-04-25 17:04:46 -0500 (Fri, 25 Apr 2008) | 7 lines

Lock the channel around datastore access

(closes issue #12527)
Reported by: mnicholson
Patches:
      pbx_lua4.diff uploaded by mnicholson (license 96)

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2008-04-25 22:05:26 +00:00
Russell Bryant
35c3c7770f Merged revisions 114674 via svnmerge from
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r114674 | russell | 2008-04-25 17:00:35 -0500 (Fri, 25 Apr 2008) | 11 lines

Merged revisions 114673 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114673 | russell | 2008-04-25 16:54:40 -0500 (Fri, 25 Apr 2008) | 3 lines

Use consistent logic for checking to see if a call number has been chosen yet.
Also, remove some redundant logic I recently added in a fix.

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2008-04-25 22:03:25 +00:00
Russell Bryant
583b8b1cb8 Blocked revisions 114650 via svnmerge
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r114650 | tilghman | 2008-04-25 10:58:36 -0500 (Fri, 25 Apr 2008) | 13 lines

Blocked revisions 114649 via svnmerge

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r114649 | tilghman | 2008-04-25 10:53:52 -0500 (Fri, 25 Apr 2008) | 6 lines

Reference documentation files that actually exist.
(closes issue #12516)
 Reported by: linuxmaniac
 Patches: 
       diff_rev114611.patch uploaded by linuxmaniac (license 472)

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2008-04-25 21:46:36 +00:00
Russell Bryant
c9fdb8cd09 Blocked revisions 114655 via svnmerge
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r114655 | russell | 2008-04-25 13:18:27 -0500 (Fri, 25 Apr 2008) | 10 lines

Merge code from team/russell/parking_updates

Add some additional features to the core park_call_full() function, and expose
them as options to the Park() application.  The functionality being added is the
ability to specify a custom return extension/context/priority, a custom timeout,
and a couple of options.  The options are to play ringing instead of MOH to the
parked caller, and to randomize parking spot selection.

(code inspired by the patch in AST-17, code from switchvox)

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Tilghman Lesher
09c3e0f618 Blocked revisions 114667 via svnmerge
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r114667 | tilghman | 2008-04-25 15:20:10 -0500 (Fri, 25 Apr 2008) | 2 lines

Whitespace changes only

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2008-04-25 20:20:54 +00:00
Tilghman Lesher
27ef21d404 Blocked revisions 114665 via svnmerge
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r114665 | tilghman | 2008-04-25 14:39:26 -0500 (Fri, 25 Apr 2008) | 2 lines

Oops, this isn't necessarily AGI that is forking anymore

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2008-04-25 19:40:20 +00:00
Mark Michelson
070497048a Merged revisions 114663 via svnmerge from
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r114663 | mmichelson | 2008-04-25 14:33:27 -0500 (Fri, 25 Apr 2008) | 12 lines

Merged revisions 114662 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114662 | mmichelson | 2008-04-25 14:32:02 -0500 (Fri, 25 Apr 2008) | 4 lines

Move the unlock of the spyee channel to outside the start_spying() function so that
the channel is not unlocked twice when using whisper mode.


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2008-04-25 19:34:11 +00:00
Jason Parker
bfdca9e0f2 Blocked revisions 114660 via svnmerge
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r114660 | qwell | 2008-04-25 13:32:22 -0500 (Fri, 25 Apr 2008) | 12 lines

Merge app_pickupchan with app_directed_pickup, for AST-27.

Initially, this was to be a new feature, with a patch from Switchvox,
 but after discussions, it was noted that this feature already existed in trunk.

The resulting discussions ended in a comment that was along the lines of
 "the patch provided here is a lot smaller than what is already in trunk,
 because it doesn't create a new application and duplicate existing code"

It was decided that these two applications could be easily merged to reduce
 code duplication.  SO, that's what this does.

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2008-04-25 19:09:08 +00:00
Mark Michelson
4a927adba5 Blocked revisions 114656 via svnmerge
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r114656 | mmichelson | 2008-04-25 13:18:30 -0500 (Fri, 25 Apr 2008) | 13 lines

This patch allows for forwarding a message with a "comment" attachment
if using IMAP storage for voicemail. The comment will be recorded and attached
as a second attachment in addition to the original message. This will be invoked
if you choose to prepend a message the way you would with file or ODBC storage


(closes issue #12028)
Reported by: jaroth
Patches:
      forward_with_comment_v2.patch uploaded by jaroth (license 50)
Tested by: jaroth


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2008-04-25 18:20:01 +00:00
Russell Bryant
326c20838b Blocked revisions 114655 via svnmerge
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r114655 | russell | 2008-04-25 13:18:27 -0500 (Fri, 25 Apr 2008) | 10 lines

Merge code from team/russell/parking_updates

Add some additional features to the core park_call_full() function, and expose
them as options to the Park() application.  The functionality being added is the
ability to specify a custom return extension/context/priority, a custom timeout,
and a couple of options.  The options are to play ringing instead of MOH to the
parked caller, and to randomize parking spot selection.

(code inspired by the patch in AST-17, code from switchvox)

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2008-04-25 18:19:24 +00:00
Mark Michelson
c2f8b21868 Merged revisions 114651 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r114651 | mmichelson | 2008-04-25 11:25:17 -0500 (Fri, 25 Apr 2008) | 4 lines

Fix a memory leak and protect against potential dereferences of a NULL
pointer.


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2008-04-25 16:26:08 +00:00
Sean Bright
f35b986bfd Blocked revisions 114644 via svnmerge
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r114644 | seanbright | 2008-04-25 09:56:05 -0400 (Fri, 25 Apr 2008) | 1 line

Speaking of building...
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2008-04-25 13:56:57 +00:00
Michiel van Baak
72bad89c93 Blocked revisions 114637 via svnmerge
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r114637 | mvanbaak | 2008-04-25 00:16:48 +0200 (Fri, 25 Apr 2008) | 8 lines

Pass the hangup cause all the way to the calling app/channel.

(closes issue #11328)
Reported by: rain
Patches:
      20071207__pass_cause_in_hangup_control_frame.diff.txt uploaded by Corydon76 (license 14)
brought up-to-date to trunk by me

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2008-04-24 22:18:58 +00:00
Joshua Colp
7b66e9fcb5 Merged revisions 114635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r114635 | file | 2008-04-24 19:11:46 -0300 (Thu, 24 Apr 2008) | 4 lines

Hey look, it builds.
(closes issue #12519)
Reported by: falves11

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2008-04-24 22:14:03 +00:00
Mark Michelson
6f5321c1bf Merged revisions 114633 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r114633 | mmichelson | 2008-04-24 16:35:39 -0500 (Thu, 24 Apr 2008) | 19 lines

Merged revisions 114632 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114632 | mmichelson | 2008-04-24 16:35:08 -0500 (Thu, 24 Apr 2008) | 11 lines

Re-invite RTP during a masquerade so that, for instance, an AMI
redirect of two channels which are natively bridged will preserve audio
on both channels. This prevents a problem with Asterisk not re-inviting
due to one of the channels having being a zombie.

(closes issue #12513)
Reported by: mneuhauser
Patches:
      asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by mneuhauser (license 425)


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2008-04-24 21:36:05 +00:00
Mark Michelson
9ebf88ab3d Merged revisions 114629 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r114629 | mmichelson | 2008-04-24 15:43:52 -0500 (Thu, 24 Apr 2008) | 16 lines

Merged revisions 114628 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114628 | mmichelson | 2008-04-24 15:43:03 -0500 (Thu, 24 Apr 2008) | 8 lines

Output of channel variables when eventwhencalled=vars was set
was being truncated two characters. This patch corrects the
problem.

(closes issue #12493)
Reported by: davidw


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2008-04-24 20:44:37 +00:00
Mark Michelson
0b3951288d Merged revisions 114625 via svnmerge from
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r114625 | mmichelson | 2008-04-24 15:06:06 -0500 (Thu, 24 Apr 2008) | 18 lines

Merged revisions 114624 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114624 | mmichelson | 2008-04-24 15:04:24 -0500 (Thu, 24 Apr 2008) | 10 lines

Resolve a deadlock in chan_local by releasing the channel lock
temporarily.

(closes issue #11712)
Reported by: callguy
Patches:
      11712.patch uploaded by putnopvut (license 60)
Tested by: acunningham


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2008-04-24 20:06:38 +00:00
Tilghman Lesher
a48962b43d Merged revisions 114622 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r114622 | tilghman | 2008-04-24 14:54:57 -0500 (Thu, 24 Apr 2008) | 12 lines

Merged revisions 114621 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114621 | tilghman | 2008-04-24 14:53:36 -0500 (Thu, 24 Apr 2008) | 4 lines

Ensure that when we set the accountcode, it actually shows up in the CDR.
(Fix for AMI Originate)
(Closes issue #12007)

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2008-04-24 19:55:55 +00:00
Tilghman Lesher
b4a9866bd5 Merged revisions 114617 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r114617 | tilghman | 2008-04-24 14:24:31 -0500 (Thu, 24 Apr 2008) | 6 lines

Fix DST calculation, and fix bug in calculation of whether conf has started yet or not
(Closes issue #12292)
 Reported by: DEA
 Patches: 
       app_meetme-rt-dst-sched-fix.txt uploaded by DEA (license 3)

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2008-04-24 19:37:45 +00:00
Jason Parker
9668fba027 Merged revisions 114612 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r114612 | qwell | 2008-04-24 11:47:01 -0500 (Thu, 24 Apr 2008) | 17 lines

Merged revisions 51989 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

(closes issue #12496)
Reported by: daniele
Patches:
      misdn-moh-1.6.0-beta7.1.patch uploaded by daniele (license 471)
Tested by: daniele

Technically, I didn't use the patch above except to find out what revision to merge - but it's the same thing as this revision.

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r51989 | crichter | 2007-01-24 06:57:22 -0600 (Wed, 24 Jan 2007) | 1 line

added fix from #8899
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2008-04-24 16:48:05 +00:00
Russell Bryant
7f3b41e6b3 Merged revisions 114609 via svnmerge from
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r114609 | russell | 2008-04-24 10:56:55 -0500 (Thu, 24 Apr 2008) | 12 lines

Merged revisions 114608 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114608 | russell | 2008-04-24 10:55:21 -0500 (Thu, 24 Apr 2008) | 4 lines

Fix a silly mistake in a change I made yesterday that caused chan_iax2 to blow
up very quickly.
(issue #12515)

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2008-04-24 15:57:24 +00:00
Olle Johansson
d6cf0c7823 Merged revisions 114606 via svnmerge from
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r114606 | oej | 2008-04-24 16:59:05 +0200 (Tor, 24 Apr 2008) | 11 lines

Merged revisions 114603 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114603 | oej | 2008-04-24 16:55:18 +0200 (Tor, 24 Apr 2008) | 3 lines

Only have one max-forwards header in outbound REFERs.
Discovered in the Asterisk SIP Masterclass in Orlando. Thanks Joe!

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2008-04-24 15:00:50 +00:00
Russell Bryant
d8234a6f34 Merged revisions 114604 via svnmerge from
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r114604 | russell | 2008-04-24 09:55:21 -0500 (Thu, 24 Apr 2008) | 3 lines

Change a verbose message to debug.
(closes issue #12514)

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Russell Bryant
25eca96f81 Merged revisions 114601 via svnmerge from
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r114601 | russell | 2008-04-23 17:53:20 -0500 (Wed, 23 Apr 2008) | 14 lines

Merged revisions 114600 via svnmerge from 
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r114600 | russell | 2008-04-23 17:18:12 -0500 (Wed, 23 Apr 2008) | 6 lines

Improve some broken cookie parsing code.  Previously, manager login over HTTP
would only work if the mansession_id cookie was first.  Now, the code builds
a list of all of the cookies in the Cookie header.  This fixes a problem
observed by users of the Asterisk GUI.
(closes AST-20)

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2008-04-23 22:54:41 +00:00
Russell Bryant
e5c15bef2a Merged revisions 114598 via svnmerge from
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r114598 | russell | 2008-04-23 15:53:05 -0500 (Wed, 23 Apr 2008) | 18 lines

Merged revisions 114597 via svnmerge from 
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r114597 | russell | 2008-04-23 15:49:18 -0500 (Wed, 23 Apr 2008) | 10 lines

Fix an issue that caused getting the correct next channel to not always work.
Also, remove setting the amount of time to wait for a digit from 5 seconds back
down to 1/10 of a second.  I believe this was so the beep didn't get played over
and over really fast, but a while back I put in another fix for that issue.

(closes issue #12498)
Reported by: jsmith
Patches:
      app_chanspy_channel_walk.trunk.patch uploaded by jsmith (license 15)

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2008-04-23 20:53:27 +00:00
Jason Parker
9b9c0983eb Merged revisions 114595 via svnmerge from
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r114595 | qwell | 2008-04-23 13:33:28 -0500 (Wed, 23 Apr 2008) | 16 lines

Merged revisions 114594 via svnmerge from 
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r114594 | qwell | 2008-04-23 13:28:44 -0500 (Wed, 23 Apr 2008) | 8 lines

Fix reload/unload for res_musiconhold module.

(closes issue #11575)
Reported by: sunder
Patches:
      M11575_14_rev3.diff uploaded by junky (license 177)
      bug11575_trunk.diff.txt uploaded by jamesgolovich (license 176)

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2008-04-23 18:34:50 +00:00
Russell Bryant
94dddcf2a8 Merged revisions 114592 via svnmerge from
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r114592 | russell | 2008-04-23 13:01:00 -0500 (Wed, 23 Apr 2008) | 13 lines

Merged revisions 114591 via svnmerge from 
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r114591 | russell | 2008-04-23 12:55:31 -0500 (Wed, 23 Apr 2008) | 5 lines

Store the manager session ID explicitly as 4 byte ID instead of a ulong.  The
mansession_id cookie is coded to be limited to 8 characters of hex, and this
could break logins from 64-bit machines in some cases.
(inspired by AST-20)

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2008-04-23 18:01:42 +00:00
Russell Bryant
1b3ee9e4f0 Merged revisions 114588 via svnmerge from
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r114588 | russell | 2008-04-23 12:18:29 -0500 (Wed, 23 Apr 2008) | 10 lines

Merged revisions 114587 via svnmerge from 
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r114587 | russell | 2008-04-23 12:16:32 -0500 (Wed, 23 Apr 2008) | 2 lines

Fix find_callno_locked() to actually return the callno locked in some more cases.

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2008-04-23 17:18:59 +00:00
Olle Johansson
3435ef6663 Merged revisions 114585 via svnmerge from
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r114585 | oej | 2008-04-23 18:53:34 +0200 (Ons, 23 Apr 2008) | 10 lines

Merged revisions 114584 via svnmerge from 
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r114584 | oej | 2008-04-23 18:51:41 +0200 (Ons, 23 Apr 2008) | 2 lines

Add 502 support for both directions, not only one...  (see r114571)

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2008-04-23 16:57:28 +00:00
Joshua Colp
bfd90e4f11 Merged revisions 114580 via svnmerge from
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r114580 | file | 2008-04-23 11:55:03 -0300 (Wed, 23 Apr 2008) | 12 lines

Merged revisions 114579 via svnmerge from 
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r114579 | file | 2008-04-23 11:54:11 -0300 (Wed, 23 Apr 2008) | 4 lines

Instead of stopping dialplan execution when SayNumber attempts to say a large number that it can not print out a message informing the user and continue on.
(closes issue #12502)
Reported by: bcnit

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Mark Michelson
2513d6d403 Blocked revisions 114577 via svnmerge
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r114577 | mmichelson | 2008-04-22 19:58:49 -0500 (Tue, 22 Apr 2008) | 23 lines

Round 2 of IMAP_STORAGE app_voicemail.c fixes:

This fixes a bug that was thought to be fixed already.

app_voicemail, if using IMAP_STORAGE, has a problem because
the IMAP header files include syslog.h, which define LOG_WARNING
and LOG_DEBUG to be different than what Asterisk uses for those
same macros. This was "fixed" in the past by including all the 
IMAP header files prior to including asterisk.h. This fix worked...
unless you were to try to compile with MALLOC_DEBUG. MALLOC_DEBUG
prepends the inclusion of astmm.h to every file, which means that no
matter what order the includes are in in app_voicemail, the unexpected
values for LOG_WARNING and LOG_DEBUG will be in place.

The action taken for this fix was to define AST_LOG_* macros in addition
to the LOG_* macros already defined. These new macros are used in app_voicemail.c,
logger.h, and astobj.h right now, and their use will be encouraged in the future.

In consideration of those who have written third-party modules which use 
the LOG_* macros, these will NOT be removed from the source, however future use
of these macros is discouraged.


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Mark Michelson
2ee5b15459 Merged revisions 114575 via svnmerge from
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r114575 | mmichelson | 2008-04-22 19:40:30 -0500 (Tue, 22 Apr 2008) | 10 lines

Round 1 of IMAP_STORAGE-related app_voicemail changes

This makes IMAP_STORAGE include the proper headers if you
have specified the "system" option for --with-imap when running
the configure script and your IMAP-related headers exist in 
/usr/include/c-client.

This change is due to a hasty merge of a 1.4 change I made.


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2008-04-23 00:41:12 +00:00
Tilghman Lesher
6d39bdb8cf Merged revisions 114572 via svnmerge from
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r114572 | tilghman | 2008-04-22 18:58:19 -0500 (Tue, 22 Apr 2008) | 10 lines

Merged revisions 114571 via svnmerge from 
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r114571 | tilghman | 2008-04-22 18:51:44 -0500 (Tue, 22 Apr 2008) | 2 lines

Treat a 502 just like a 503, when it comes to processing a response code

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2008-04-22 23:59:11 +00:00
Russell Bryant
f3db442032 Merged revisions 114559 via svnmerge from
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r114559 | russell | 2008-04-22 17:17:31 -0500 (Tue, 22 Apr 2008) | 13 lines

Merged revisions 114558 via svnmerge from 
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r114558 | russell | 2008-04-22 17:15:36 -0500 (Tue, 22 Apr 2008) | 5 lines

When we receive a full frame that is supposed to contain our call number,
ensure that it has the correct one.
(closes issue #10078)
(AST-2008-006)

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2008-04-22 22:18:59 +00:00
Steve Murphy
7d7db965ad Merged revisions 114553 via svnmerge from
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r114553 | murf | 2008-04-22 15:57:57 -0600 (Tue, 22 Apr 2008) | 14 lines


(closes issue #12469)
Reported by: triccyx

I had a bit a problem reproducing this in my setup (trying not to disturb my other stuff)
but finally, I got it. The problem appears to be that the extension is being added in
replace mode, which kinda assumes that the pattern trie has been formed, when in fact,
in this case, it was not. The checks being done are not nec. when the tree is not yet
formed, as changes like this will be summarized when the trie is formed in the future.

I tested the fix, and the crash no longer happens. Feel free to open the bug again if
this fix doesn't cure the problem.


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Russell Bryant
a04a1268e3 Blocked revisions 114551 via svnmerge
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r114551 | russell | 2008-04-22 16:15:41 -0500 (Tue, 22 Apr 2008) | 11 lines

Blocked revisions 114550 via svnmerge

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r114550 | russell | 2008-04-22 16:14:55 -0500 (Tue, 22 Apr 2008) | 4 lines

I thought I was going to be able to leave 1.4 alone, but that was not the case.
I ran into some problems with G.722 in 1.4, so I have merged in all of the fixes
in this area that I have made in trunk/1.6.0, and things are happy again.

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2008-04-22 21:16:15 +00:00
Russell Bryant
2a69116229 Merged revisions 114548 via svnmerge from
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r114548 | russell | 2008-04-22 15:25:56 -0500 (Tue, 22 Apr 2008) | 2 lines

re-add a fix that got lost with a recent change

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Russell Bryant
ff9c78771f Blocked revisions 114546 via svnmerge
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r114546 | russell | 2008-04-22 14:45:12 -0500 (Tue, 22 Apr 2008) | 9 lines

Blocked revisions 114545 via svnmerge

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r114545 | russell | 2008-04-22 14:45:00 -0500 (Tue, 22 Apr 2008) | 2 lines

Trivial change to read the number of samples from a frame before calling ast_write()

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2008-04-22 19:45:25 +00:00
Russell Bryant
e1f8040a80 Blocked revisions 114543 via svnmerge
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r114543 | russell | 2008-04-22 13:30:47 -0500 (Tue, 22 Apr 2008) | 10 lines

Blocked revisions 114542 via svnmerge

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r114542 | russell | 2008-04-22 13:29:56 -0500 (Tue, 22 Apr 2008) | 3 lines

After a parked call times out, allow the call back to the parker to time out.
(closes issue #10890)

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2008-04-22 18:31:00 +00:00
Jason Parker
8ec76fdc91 Merged revisions 114540 via svnmerge from
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r114540 | qwell | 2008-04-22 13:14:09 -0500 (Tue, 22 Apr 2008) | 8 lines

Allow setqueuevar=yes (et al) to work, after changes to pbx_builtin_setvar()

(closes issue #12490)
Reported by: bcnit
Patches:
      12490-queuevars-3.diff uploaded by qwell (license 4)
Tested by: qwell

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2008-04-22 18:14:44 +00:00
Russell Bryant
f283f17f4d Merged revisions 114538 via svnmerge from
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r114538 | russell | 2008-04-22 13:04:39 -0500 (Tue, 22 Apr 2008) | 17 lines

Merged revisions 114537 via svnmerge from 
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r114537 | russell | 2008-04-22 13:03:33 -0500 (Tue, 22 Apr 2008) | 9 lines

If the dial string passed to the call channel callback does not indicate an
extension, then consider the extension on the channel before falling back
to the default.

(closes issue #12479)
Reported by: darren1713
Patches:
      exten_dial_fix_chan_iax2.c.patch uploaded by darren1713 (license 116)

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Russell Bryant
7a61cf5371 Blocked revisions 114533 via svnmerge
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r114533 | russell | 2008-04-22 11:47:00 -0500 (Tue, 22 Apr 2008) | 4 lines

Add a c() option for the Jack() application and JACK_HOOK() funciton for supplying
a custom client name.  Using the channel name is still the default.  This was done
at the request of Jared Smith.

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Joshua Colp
0a17789ab3 Blocked revisions 114529 via svnmerge
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r114529 | file | 2008-04-22 12:54:06 -0300 (Tue, 22 Apr 2008) | 6 lines

Add support for authenticating on a NOTIFY request. This is useful for phones that require it when sending them a special packet to get them to do something (such as reload their configuration).
(closes issue #9896)
Reported by: IgorG
Patches:
      sipnotify-113980-v14.patch uploaded by IgorG (license 20)

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2008-04-22 15:54:37 +00:00
Russell Bryant
4b8c7ee1c9 Merged revisions 114527 via svnmerge from
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r114527 | russell | 2008-04-22 10:46:01 -0500 (Tue, 22 Apr 2008) | 8 lines

Correct action_ping() and action_events() with regards to Manager 1.1
documentation.  Also, fix a bug in xml_translate().

(closes issue #11649)
Reported by: ys
Patches:
      trunk_manager.c.diff uploaded by ys (license 281)

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Russell Bryant
defe682c1d Blocked revisions 114523 via svnmerge
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r114523 | russell | 2008-04-22 10:20:53 -0500 (Tue, 22 Apr 2008) | 20 lines

Blocked revisions 114522 via svnmerge

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r114522 | russell | 2008-04-22 10:20:37 -0500 (Tue, 22 Apr 2008) | 13 lines

Merge changes from team/russell/issue_9520

These changes make sure that the reference count for sip_peer objects properly
reflects the fact that the peer is sitting in the scheduler for a scheduled
callback for qualifying peers or for expiring registrations.  Without this, it
was possible for these callbacks to happen at the same time that the peer was
being destroyed.  This was especially likely to happen with realtime peers, and
for people making use of the realtime prune CLI command.

(closes issue #9520)
Reported by: kryptolus
Committed patch by me

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Steve Murphy
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r114520 | murf | 2008-04-22 08:38:46 -0600 (Tue, 22 Apr 2008) | 15 lines


Hopefully, this will resolve the issues that russellb had with this log_show_lock().
I gathered the code that filled the string, and put it in a different func which
I cryptically call "append_lock_information()".
Now, both log_show_lock(), and handle_show_locks() both call this code to do
the work. Tested, seems to work fine. 
Also, log_show_lock was modified to use the ast_str stuff, along with checking
for successful ast_str creation, and freeing the ast_str obj when finished.
A break was inserted to terminate the search for the lock; we should never
see it twice.

An example usage in chan_sip.c was created as a comment, for instructional
purposes.


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2008-04-22 14:50:55 +00:00
Matthew Fredrickson
a49ac36ed8 Merged revisions 114389 via svnmerge from
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r114389 | mattf | 2008-04-21 13:44:35 -0500 (Mon, 21 Apr 2008) | 1 line

Add support for generic name transmission (#12484) on SS7 in chan_zap
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Matthew Fredrickson
8ae8851835 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@114421 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2008-04-21 20:22:08 +00:00
Matthew Fredrickson
e2505e605f git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@114420 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2008-04-21 20:17:35 +00:00
Matthew Fredrickson
8b97c5fd92 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@114390 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2008-04-21 18:56:04 +00:00
Jeff Peeler
37b0abc1b7 Merged revisions 114327 via svnmerge from
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r114327 | jpeeler | 2008-04-21 10:34:37 -0500 (Mon, 21 Apr 2008) | 2 lines

This removes an invalid warning message for an incorrectly entered pin, but more importantly removes an inapplicable check. If the first argument passed to app_authenticate does not contain a '/', the argument should be treated as the sole fixed "password" to match against and that is all. (Previous behavior was attempting to open a file based on the pin.)

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Russell Bryant
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r114325 | russell | 2008-04-21 10:01:04 -0500 (Mon, 21 Apr 2008) | 2 lines

Add a simple janitor project

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2008-04-21 15:01:23 +00:00
Joshua Colp
877fc5a8c2 Merged revisions 114323 via svnmerge from
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r114323 | file | 2008-04-21 11:40:33 -0300 (Mon, 21 Apr 2008) | 12 lines

Merged revisions 114322 via svnmerge from 
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r114322 | file | 2008-04-21 11:39:32 -0300 (Mon, 21 Apr 2008) | 4 lines

Only drop audio if we receive it without a progress indication. We allow other frames through such as DTMF because they may be needed to complete the call.
(closes issue #12440)
Reported by: aragon

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2008-04-21 14:42:03 +00:00
Joshua Colp
6299cfdad7 Merged revisions 114320 via svnmerge from
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r114320 | file | 2008-04-21 11:34:06 -0300 (Mon, 21 Apr 2008) | 6 lines

Only print out the error message if ldap_modify_ext_s actually returns an error, and not success.
(closes issue #12438)
Reported by: gservat
Patches:
      res_config_ldap.c-patch-code uploaded by gservat (license 466)

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2008-04-21 14:35:53 +00:00
Sean Bright
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r114314 | seanbright | 2008-04-20 10:52:07 -0400 (Sun, 20 Apr 2008) | 1 line

Minor logging cleanups
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Matthew Fredrickson
7cfd909969 SS7:Added - Generic Name / Access Transport / Redirecting Number handling. #12425
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2008-04-19 17:00:15 +00:00
Russell Bryant
27cf0cc766 Merged revisions 114285 via svnmerge from
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r114285 | russell | 2008-04-18 16:51:05 -0500 (Fri, 18 Apr 2008) | 10 lines

Merged revisions 114284 via svnmerge from 
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r114284 | russell | 2008-04-18 16:48:06 -0500 (Fri, 18 Apr 2008) | 2 lines

Don't destroy a manager session if poll() returns an error of EAGAIN.

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Russell Bryant
1cd288d8b8 Merged revisions 114279 via svnmerge from
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r114279 | russell | 2008-04-18 15:01:47 -0500 (Fri, 18 Apr 2008) | 10 lines

Merged revisions 114278 via svnmerge from 
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r114278 | russell | 2008-04-18 15:01:09 -0500 (Fri, 18 Apr 2008) | 2 lines

ensure directories are created before we try to install stuff into them

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Russell Bryant
bb6f2f32c7 Merged revisions 114276 via svnmerge from
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r114276 | russell | 2008-04-18 14:59:17 -0500 (Fri, 18 Apr 2008) | 10 lines

Merged revisions 114275 via svnmerge from 
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r114275 | russell | 2008-04-18 14:58:55 -0500 (Fri, 18 Apr 2008) | 2 lines

SUBDIRS_INSTALL is already listed as a subtarget for bininstall

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Joshua Colp
feda76658e Merged revisions 114271 via svnmerge from
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r114271 | file | 2008-04-18 16:35:33 -0300 (Fri, 18 Apr 2008) | 4 lines

Make sure ADSI is marked as unavailable on Unistim channels so voicemail does not try to do some ADSI jazz.
(closes issue #12460)
Reported by: PerryB

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Joshua Colp
07e6364afa Blocked revisions 114261 via svnmerge
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r114261 | file | 2008-04-18 15:15:11 -0300 (Fri, 18 Apr 2008) | 6 lines

Add MEETME_INFO dialplan function that allows querying various properties of a Meetme conference.
(closes issue #11691)
Reported by: junky
Patches:
      meetme_info.patch uploaded by jpeeler (license 325)

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Mark Michelson
ef448d123d Merged revisions 114259 via svnmerge from
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r114259 | mmichelson | 2008-04-18 13:03:06 -0500 (Fri, 18 Apr 2008) | 14 lines

Merged revisions 114257 via svnmerge from 
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r114257 | mmichelson | 2008-04-18 12:44:29 -0500 (Fri, 18 Apr 2008) | 6 lines

Clearing up error messages so they make a bit more sense. Also removing a redundant error
message.

Issue AST-15


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Joshua Colp
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r114254 | file | 2008-04-18 13:11:27 -0300 (Fri, 18 Apr 2008) | 4 lines

If the parsing of the config file fails make sure we unlock ldap_lock.
(closes issue #12477)
Reported by: IgorG

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Russell Bryant
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r114249 | russell | 2008-04-18 10:25:50 -0500 (Fri, 18 Apr 2008) | 12 lines

Blocked revisions 114248 via svnmerge

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r114248 | russell | 2008-04-18 10:24:09 -0500 (Fri, 18 Apr 2008) | 5 lines

Ensure that we don't ast_strdupa(NULL)
(closes issue #12476)
Reported by: davidw
Patch by me

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Sean Bright
0f70968ff5 Merged revisions 114246 via svnmerge from
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r114246 | seanbright | 2008-04-18 09:38:07 -0400 (Fri, 18 Apr 2008) | 9 lines

Merged revisions 114245 via svnmerge from 
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r114245 | seanbright | 2008-04-18 09:33:32 -0400 (Fri, 18 Apr 2008) | 1 line

Only complete the SIP channel name once for 'sip show channel <channel>'
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Tilghman Lesher
69c922d088 Merged revisions 114243 via svnmerge from
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r114243 | tilghman | 2008-04-18 01:53:47 -0500 (Fri, 18 Apr 2008) | 11 lines

Merged revisions 114242 via svnmerge from 
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r114242 | tilghman | 2008-04-18 01:49:16 -0500 (Fri, 18 Apr 2008) | 3 lines

For consistency sake, ensure that the values that ${CALLINGPRES} returns are valid as an
input to SetCallingPres.  (Closes issue #12472)

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Russell Bryant
2192aa0cfc Merged revisions 114151 via svnmerge from
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r114151 | oej | 2008-04-15 15:39:29 -0500 (Tue, 15 Apr 2008) | 10 lines

Merged revisions 114148 via svnmerge from 
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r114148 | oej | 2008-04-15 22:26:05 +0200 (Tis, 15 Apr 2008) | 2 lines

Handle subscribe queues in all situations... Thanks to festr_ on irc for telling me about this bug.

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Russell Bryant
bb661f904b Merged revisions 114150 via svnmerge from
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r114150 | oej | 2008-04-15 15:31:08 -0500 (Tue, 15 Apr 2008) | 2 lines

Adding chanvar to SIPPEER from 1.4 branch

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Russell Bryant
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r114194 | seanbright | 2008-04-17 07:25:23 -0500 (Thu, 17 Apr 2008) | 1 line

Update the CHANGES file with yesterday's ChanSpy change.  Sorry Kevin, just saw your e-mail.
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Russell Bryant
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r114229 | jpeeler | 2008-04-17 16:09:37 -0500 (Thu, 17 Apr 2008) | 1 line

added info describing DNS manager
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r114181 | tilghman | 2008-04-16 15:00:27 -0500 (Wed, 16 Apr 2008) | 10 lines

Blocked revisions 114180 via svnmerge

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r114180 | tilghman | 2008-04-16 14:59:37 -0500 (Wed, 16 Apr 2008) | 3 lines

Backport revisions for latest vpb drivers to 1.4
(Closes issue #12457)

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Russell Bryant
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r114077 | dbailey | 2008-04-11 17:09:04 -0500 (Fri, 11 Apr 2008) | 2 lines

Change the number of line keys per registration from 2 to 1 

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Russell Bryant
59082b1829 Merged revisions 114233 via svnmerge from
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r114233 | russell | 2008-04-17 17:24:00 -0500 (Thu, 17 Apr 2008) | 14 lines

Merged revisions 114230 via svnmerge from 
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r114230 | russell | 2008-04-17 17:15:43 -0500 (Thu, 17 Apr 2008) | 6 lines

Remove redundant safety net.  The check for the autoservice channel list state
accomplishes the same goal in a better way.

(issue #12470)
Reported By: atis

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Russell Bryant
4ee2ba0704 Blocked revisions 114231 via svnmerge
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r114231 | russell | 2008-04-17 17:21:32 -0500 (Thu, 17 Apr 2008) | 3 lines

Make this file compile.  The variable str is never set anywhere.  Furthermore,
it duplicates a lot of code.  I will leave it to murf to clean up.

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Mark Michelson
392b19e844 Merged revisions 114227 via svnmerge from
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r114227 | mmichelson | 2008-04-17 16:04:40 -0500 (Thu, 17 Apr 2008) | 17 lines

Merged revisions 114226 via svnmerge from 
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r114226 | mmichelson | 2008-04-17 16:03:29 -0500 (Thu, 17 Apr 2008) | 9 lines

Declaration of the peer channel in this scope was making it so the peer variable defined
in the outer scope was never set properly, therefore making iterating through the channel
list always restart from the beginning. This bug would have affected anyone who called
chanspy without specifying a first argument.

(closes issue #12461)
Reported by: stever28


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Steve Murphy
f067a409ec Blocked revisions 114201 via svnmerge
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r114201 | murf | 2008-04-17 08:45:16 -0600 (Thu, 17 Apr 2008) | 1 line

Thanks to snuff for finding these omissions
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Steve Murphy
535549d60b Blocked revisions 114190 via svnmerge
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r114190 | murf | 2008-04-16 17:53:27 -0600 (Wed, 16 Apr 2008) | 1 line

This is the scariest commit I've done in a long time. This is the astobj2-ification of chan_sip. I've tested a number of scenarios like crazy. It used to have 4x the call setup/teardown performance of trunk, but now it's roughly at parity. I will attempt to find the bottlenecks and get it back to the 4x mark. The changes made were somewhat invasive, but the value to the community of these upgrades outweighs waiting further for more testing. Every change being made to chan_sip was lousing this code up when we tried to merge. Peers, Users, Dialogs, are all now astobj2 objects, indexed via hashtables. Refcounting is used to track objects and free them at the bitter end of their lives. Please file issues on bugs.digium.com, and PLEASE, please, please be patient. One natural advantage to all the hash-table work is that loading large sip.conf files full of thousands of peers now goes much faster. One more please: PLEASE help thrash this code and test it.
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Steve Murphy
9fa2d84191 Blocked revisions 114187 via svnmerge
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r114187 | murf | 2008-04-16 14:54:41 -0600 (Wed, 16 Apr 2008) | 1 line

A small enhancement-- I added the routine log_show_lock to utils.c, which if the mentioned lock has been acquired, this routine will log to the console the normal info about that lock you'd see from the CLI when you do a 'core show locks'. It's solely for debug-- if the lock is NOT acquired, there is no output. I use it to show 'unexpected' locks, to see where/why a lock is pre-locked. This command is to be called from points of interest, like just before a trylock, and helps to spot fleeting, highly temporal locks that normally are not locked...
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Steve Murphy
25c534180d Blocked revisions 114183 via svnmerge
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r114183 | murf | 2008-04-16 14:28:08 -0600 (Wed, 16 Apr 2008) | 1 line

Introducing a small optimization to event_unsubscribe; events now use a Doubly-Linked list for events, gives fast deletions, for the sake of channel driver mwi events. From team/murf/bug11210.
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Steve Murphy
3e3e4b6e0c Blocked revisions 114182 via svnmerge
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r114182 | murf | 2008-04-16 14:09:39 -0600 (Wed, 16 Apr 2008) | 1 line

Introducing a small upgrade to the ast_sched_xxx  facility, to keep it from eating up lots of cpu cycles. See CHANGES. From the team/murf/bug11210 branch.
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Steve Murphy
d29597b342 Blocked revisions 114175 via svnmerge
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r114175 | murf | 2008-04-16 11:45:28 -0600 (Wed, 16 Apr 2008) | 1 line

Introducing various astobj2 enhancements, chief being a refcount tracing feature, and various documentation updates in astobj2.h, and the addition of standalone utility, refcounter, that will filter the trace output for unbalanced, unfreed objects. This comes from the team/murf/bug11210 branch.
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Steve Murphy
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r114172 | murf | 2008-04-16 11:14:18 -0600 (Wed, 16 Apr 2008) | 1 line

Introducing doubly linked lists to trunk from branch team/murf/bug11210.
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Mark Michelson
a85a24b719 Blocked revisions 114212 via svnmerge
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r114212 | mmichelson | 2008-04-17 11:51:09 -0500 (Thu, 17 Apr 2008) | 11 lines

Blocked revisions 114211 via svnmerge

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r114211 | mmichelson | 2008-04-17 11:50:46 -0500 (Thu, 17 Apr 2008) | 4 lines

Add prototype for ast_dsp_frame_freed. I'm not sure how this was
compiling before...


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Mark Michelson
1dd0e2df7c Merged revisions 114208 via svnmerge from
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r114208 | mmichelson | 2008-04-17 11:40:12 -0500 (Thu, 17 Apr 2008) | 20 lines

Merged revisions 114207 via svnmerge from 
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r114207 | mmichelson | 2008-04-17 11:28:03 -0500 (Thu, 17 Apr 2008) | 12 lines

It was possible for a reference to a frame which was part of a freed DSP to still be
referenced, leading to memory corruption and eventual crashes. This code change ensures
that the dsp is freed when we are finished with the frame. This change is very similar
to a change Russell made with translators back a month or so ago.

(closes issue #11999)
Reported by: destiny6628
Patches:
      11999.patch uploaded by putnopvut (license 60)
Tested by: destiny6628, victoryure


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Russell Bryant
a52d360944 Merged revisions 114205 via svnmerge from
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r114205 | russell | 2008-04-17 11:25:29 -0500 (Thu, 17 Apr 2008) | 11 lines

Merged revisions 114204 via svnmerge from 
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r114204 | russell | 2008-04-17 11:23:45 -0500 (Thu, 17 Apr 2008) | 3 lines

Fix the bininstall target to install from subdirs, as well.
(closes issue AST-8, patch from bmd at switchvox)

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Tilghman Lesher
8479b5c666 Merged revisions 114202 via svnmerge from
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r114202 | tilghman | 2008-04-17 10:12:52 -0500 (Thu, 17 Apr 2008) | 2 lines

fileio.h does not exist; io.h does, though.

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Philippe Sultan
c35209317d Merged revisions 114199 via svnmerge from
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r114199 | phsultan | 2008-04-17 15:46:17 +0200 (Thu, 17 Apr 2008) | 10 lines

Merged revisions 114198 via svnmerge from 
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r114198 | phsultan | 2008-04-17 15:42:23 +0200 (Thu, 17 Apr 2008) | 2 lines

Use keepalives effectively in order diagnose bug #12432.

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Tilghman Lesher
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r114196 | tilghman | 2008-04-17 07:59:04 -0500 (Thu, 17 Apr 2008) | 16 lines

Merged revisions 114195 via svnmerge from 
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r114195 | tilghman | 2008-04-17 07:56:38 -0500 (Thu, 17 Apr 2008) | 8 lines

Add special case for when the agi cannot be executed, to comply with the documentation that
we return failure in that case.
(closes issue #12462)
 Reported by: fmueller
 Patches: 
       20080416__bug12462.diff.txt uploaded by Corydon76 (license 14)
 Tested by: fmueller

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Sean Bright
ddef13ea2a Merged revisions 114192 via svnmerge from
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r114192 | seanbright | 2008-04-17 06:55:05 -0400 (Thu, 17 Apr 2008) | 9 lines

Merged revisions 114191 via svnmerge from 
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r114191 | seanbright | 2008-04-17 06:51:20 -0400 (Thu, 17 Apr 2008) | 1 line

Make sure we have enough room for the recording's filename.
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Tilghman Lesher
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r114188 | tilghman | 2008-04-16 17:57:54 -0500 (Wed, 16 Apr 2008) | 2 lines

Standardized routines for forking processes (keeps all the specialized code in one place).

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Kevin P. Fleming
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r114185 | kpfleming | 2008-04-16 15:47:30 -0500 (Wed, 16 Apr 2008) | 14 lines

Merged revisions 114184 via svnmerge from 
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r114184 | kpfleming | 2008-04-16 15:46:38 -0500 (Wed, 16 Apr 2008) | 6 lines

use the ZT_SET_DIALPARAMS ioctl properly by initializing the structure to all zeroes in case it contains fields that we don't write values into (which it does as of Zaptel 1.4.10)

(closes issue #12456)
Reported by: fnordian


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Joshua Colp
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r114168 | file | 2008-04-16 11:11:18 -0300 (Wed, 16 Apr 2008) | 13 lines

Blocked revisions 114167 via svnmerge

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r114167 | file | 2008-04-16 11:10:46 -0300 (Wed, 16 Apr 2008) | 6 lines

Include the proper headers for using mkdir on FreeBSD.
(closes issue #12430)
Reported by: ys
Patches:
      app_meetme.c.diff uploaded by ys (license 281)

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2008-04-16 14:12:03 +00:00
Sean Bright
be7215325f Blocked revisions 114165 via svnmerge
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r114165 | seanbright | 2008-04-16 08:23:13 -0400 (Wed, 16 Apr 2008) | 1 line

Add the ability to disable channel technology name playback when speaking the current channel name
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2008-04-16 12:24:03 +00:00
Tilghman Lesher
14900e903f Merged revisions 114152 via svnmerge from
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r114152 | tilghman | 2008-04-15 15:51:08 -0500 (Tue, 15 Apr 2008) | 2 lines

Oops, buffer wasn't long enough for query

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2008-04-15 20:53:43 +00:00
Steve Murphy
af0e618767 Merged revisions 114146 via svnmerge from
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r114146 | murf | 2008-04-15 13:59:50 -0600 (Tue, 15 Apr 2008) | 8 lines

These changes: 

   a. fix a self-found problem with SPAWN-ing an extension,
      where matches were not being found
   b. correct some wording in a comment
   c. Add some debug for future debugging.


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2008-04-15 20:09:26 +00:00
Sean Bright
a0b3026199 Blocked revisions 114143 via svnmerge
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r114143 | seanbright | 2008-04-15 13:54:38 -0400 (Tue, 15 Apr 2008) | 2 lines

I'm not sure why, but "this" bothers me.  Ba dum dum.

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2008-04-15 18:03:36 +00:00
Jason Parker
8a9eb5810c Merged revisions 114141 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r114141 | qwell | 2008-04-15 12:21:58 -0500 (Tue, 15 Apr 2008) | 8 lines

Shorten the mac address pattern, since some phones use different identifiers (such as the i2050 softphone).

(closes issue #12398)
Reported by: c_hans
Patches:
      chan_unistim_svn.diff uploaded by c (license 460)
Tested by: c_hans

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2008-04-15 17:22:23 +00:00
Jason Parker
19d96ce22f Merged revisions 114139 via svnmerge from
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r114139 | qwell | 2008-04-15 12:17:37 -0500 (Tue, 15 Apr 2008) | 15 lines

Merged revisions 114138 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114138 | qwell | 2008-04-15 12:17:18 -0500 (Tue, 15 Apr 2008) | 7 lines

Update Digium autosupport script, for more useful information.

(closes issue #12452)
Reported by: angler
Patches:
      autosupport.diff uploaded by angler (license 106)

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2008-04-15 17:17:53 +00:00
Jason Parker
d98b6009ce Merged revisions 114134 via svnmerge from
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r114134 | qwell | 2008-04-15 11:18:38 -0500 (Tue, 15 Apr 2008) | 16 lines

Merged revisions 114133 via svnmerge from 
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r114133 | qwell | 2008-04-15 11:18:08 -0500 (Tue, 15 Apr 2008) | 8 lines

Allow autofill to work in the general section of queues.conf.
Additionally, don't try to (re)set options when they have empty values in realtime (all unset columns would have an empty value).

(closes issue #12445)
Reported by: atis
Patches:
      12445-autofill.diff uploaded by qwell (license 4)

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2008-04-15 16:19:06 +00:00
Jason Parker
bd2a864426 Blocked revisions 114131 via svnmerge
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r114131 | qwell | 2008-04-15 10:20:47 -0500 (Tue, 15 Apr 2008) | 7 lines

Convert several DEBUG logs into ast_debug.

(closes issue #12444)
Reported by: IgorG
Patches:
      channel_c_debug.diff uploaded by IgorG (license 20)

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2008-04-15 15:21:18 +00:00
Terry Wilson
9dbd614f0e Blocked revisions 114124 via svnmerge
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r114124 | twilson | 2008-04-14 14:12:27 -0500 (Mon, 14 Apr 2008) | 2 lines

Don't unref user twice on failure.  Also, when adding sorted list of users, it is best to check the entry already in the list for a "next" entry instead of the newly created entry...

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2008-04-14 20:00:33 +00:00
Terry Wilson
df9a6370a3 Blocked revisions 114127 via svnmerge
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r114127 | twilson | 2008-04-14 14:58:52 -0500 (Mon, 14 Apr 2008) | 2 lines

Need a new buffer for each loop

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2008-04-14 19:59:36 +00:00
Jason Parker
1aa20cb6c0 Merged revisions 114121 via svnmerge from
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r114121 | qwell | 2008-04-14 13:34:17 -0500 (Mon, 14 Apr 2008) | 15 lines

Merged revisions 114120 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114120 | qwell | 2008-04-14 13:31:57 -0500 (Mon, 14 Apr 2008) | 7 lines

The call_token on the pvt can occasionally be NULL, causing a crash.

If it is NULL, we can skip this channel, since it can't the one we're looking for.

(closes issue #9299)
Reported by: vazir

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2008-04-14 18:34:37 +00:00
Mark Michelson
128e5426c4 Merged revisions 114118 via svnmerge from
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r114118 | mmichelson | 2008-04-14 12:42:20 -0500 (Mon, 14 Apr 2008) | 19 lines

Merged revisions 114117 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114117 | mmichelson | 2008-04-14 12:41:03 -0500 (Mon, 14 Apr 2008) | 11 lines

Increase the retry count when attempting to show channels. This apparently
cleared an issue someone was seeing when attempting to show channels when
the load was high.

(closes issue #11667)
Reported by: falves11
Patches:
      11677.txt uploaded by russell (license 2)
Tested by: falves11


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2008-04-14 17:42:55 +00:00
Tilghman Lesher
6c96df8368 Merged revisions 114115 via svnmerge from
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r114115 | tilghman | 2008-04-14 11:32:59 -0500 (Mon, 14 Apr 2008) | 2 lines

Make tab-completion work for all cases

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2008-04-14 16:33:59 +00:00
Mark Michelson
96e74f94f0 Merged revisions 114113 via svnmerge from
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r114113 | mmichelson | 2008-04-14 11:25:09 -0500 (Mon, 14 Apr 2008) | 17 lines

Merged revisions 114112 via svnmerge from 
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r114112 | mmichelson | 2008-04-14 11:24:22 -0500 (Mon, 14 Apr 2008) | 9 lines

If the datastore has been moved to another channel due to a masquerade, then
freeing the datastore here causes an eventual double free when the new channel
hangs up. We should only free the datastore if we were able to successfully remove
it from the channel we are referencing (i.e. the datastore was not moved).

(closes issue #12359)
Reported by: pguido


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2008-04-14 16:25:57 +00:00
Joshua Colp
5b77e4588d Blocked revisions 114109 via svnmerge
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r114109 | file | 2008-04-14 12:36:02 -0300 (Mon, 14 Apr 2008) | 2 lines

During hangup it is possible for p->chan or p->owner to be NULL, so just return what the channel is bridged to instead of what they are *really* bridged to. Thanks Matt Nicholson!

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2008-04-14 15:37:10 +00:00
Mark Michelson
c243c96214 Merged revisions 114107 via svnmerge from
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r114107 | mmichelson | 2008-04-14 10:01:36 -0500 (Mon, 14 Apr 2008) | 13 lines

Merged revisions 114106 via svnmerge from 
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r114106 | mmichelson | 2008-04-14 09:58:02 -0500 (Mon, 14 Apr 2008) | 5 lines

Save a local copy of the generate callback prior to unlocking the channel in
case the generate callback goes NULL on us after the channel is unlocked. Thanks
to Russell for pointing this need out to me.


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2008-04-14 15:02:43 +00:00
Joshua Colp
e2f8a90b78 Merged revisions 114104 via svnmerge from
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r114104 | file | 2008-04-14 11:53:33 -0300 (Mon, 14 Apr 2008) | 12 lines

Merged revisions 114103 via svnmerge from 
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r114103 | file | 2008-04-14 11:52:46 -0300 (Mon, 14 Apr 2008) | 4 lines

It is possible for the remote side to say they want T38 but not give any capabilities.
(closes issue #12414)
Reported by: MVF

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Joshua Colp
8d1e1eb09e Merged revisions 114101 via svnmerge from
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r114101 | file | 2008-04-14 10:53:33 -0300 (Mon, 14 Apr 2008) | 12 lines

Merged revisions 114100 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114100 | file | 2008-04-14 10:52:49 -0300 (Mon, 14 Apr 2008) | 4 lines

Don't change the SSRC when a new source comes into play, this might happen quite often and depending on the remote side... they might not like this.
(closes issue #12353)
Reported by: dimas

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2008-04-14 13:54:25 +00:00
Tilghman Lesher
570cafe27e Merged revisions 114098 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r114098 | tilghman | 2008-04-13 21:55:41 -0500 (Sun, 13 Apr 2008) | 3 lines

Add tab command-line completion
(Closes issue #12428)

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2008-04-14 02:59:50 +00:00
Tilghman Lesher
175b5c09dc Merged revisions 114096 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r114096 | tilghman | 2008-04-13 09:35:43 -0500 (Sun, 13 Apr 2008) | 3 lines

Use ast_mkdir instead of mkdir
(Closes issue #12430)

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2008-04-13 14:38:35 +00:00
Matthew Fredrickson
35bf7be972 Make sure linkset is locked exiting ss7_start_call
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@114095 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-12 16:22:44 +00:00
Matthew Fredrickson
808e06b1a9 Make sure we start incoming calls on SS7 with echo cancellation enabled. Also make sure when completing a COT we call ss7_start_call with the proper locks held. Lastly, make sure if we fail to get a channel from zt_new that we don't assume it's there.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@114094 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-12 16:22:00 +00:00
Tilghman Lesher
903e7f1afd Merged revisions 114090 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r114090 | tilghman | 2008-04-11 18:26:56 -0500 (Fri, 11 Apr 2008) | 3 lines

If any field is not null, but has no default, then it must be set or the insert will fail.
(Closes issue #12285)

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2008-04-11 23:27:39 +00:00
Tilghman Lesher
7ef743b835 Merged revisions 114088 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r114088 | tilghman | 2008-04-11 18:21:54 -0500 (Fri, 11 Apr 2008) | 3 lines

Make the sample config match the contributed LDAP schema
(Closes issue #12421)

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2008-04-11 23:23:10 +00:00
Terry Wilson
8b58c0dd5a Merged revisions 114084 via svnmerge from
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r114084 | twilson | 2008-04-11 17:48:52 -0500 (Fri, 11 Apr 2008) | 15 lines

Merged revisions 114083 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114083 | twilson | 2008-04-11 17:32:51 -0500 (Fri, 11 Apr 2008) | 7 lines

Several places in the code called find_callno() (which releases the lock on the pvt structure) and then immediately locked the call and did things with it. Unfortunately, the call can disappear between the find_callno and the lock, causing Bad Stuff(tm) to happen.

Added find_callno_locked() function to return the callno withtout unlocking for instances that it is needed.

(issue #12400)
Reported by: ztel

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Tilghman Lesher
50071e070c Merged revisions 114085 via svnmerge from
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r114085 | tilghman | 2008-04-11 18:12:16 -0500 (Fri, 11 Apr 2008) | 7 lines

Use the correct function for free'ing objects, and maybe we won't crash.
(closes issue #12163)
 Reported by: gservat
 Patches: 
       20080411__bug12163.diff.txt uploaded by Corydon76 (license 14)
 Tested by: gservat

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2008-04-11 23:13:07 +00:00
Terry Wilson
87a4e63f6d Blocked revisions 114080 via svnmerge
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r114080 | twilson | 2008-04-11 17:23:34 -0500 (Fri, 11 Apr 2008) | 2 lines

Make sure that ${LINE} is set even if linenumber is not set in users.conf

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2008-04-11 22:24:41 +00:00
Jason Parker
7db4d7d006 Merged revisions 114073 via svnmerge from
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r114073 | qwell | 2008-04-11 16:39:44 -0500 (Fri, 11 Apr 2008) | 13 lines

Blocked revisions 114072 via svnmerge

Already fixed here.

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r114072 | qwell | 2008-04-11 16:35:16 -0500 (Fri, 11 Apr 2008) | 4 lines

It's possible that a channel can have an async goto on the successful execution of an application as well.

Closes issue #12172.

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Terry Wilson
c3747a1ace Blocked revisions 114067 via svnmerge
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r114067 | twilson | 2008-04-11 16:04:46 -0500 (Fri, 11 Apr 2008) | 3 lines

Fix the fact that global_variables 1) weren't being updated on reload (thanks for the report, Doug), and 2) weren't actually being appended to the list of profile variables because build_profile was called before the list was populated. Also needed to free the contents returned by load_file().


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2008-04-11 21:08:02 +00:00
Mark Michelson
a3a59c0f89 Merged revisions 114064 via svnmerge from
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r114064 | mmichelson | 2008-04-11 10:49:35 -0500 (Fri, 11 Apr 2008) | 19 lines

Merged revisions 114063 via svnmerge from 
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r114063 | mmichelson | 2008-04-11 10:44:28 -0500 (Fri, 11 Apr 2008) | 11 lines

Fix a race condition that may happen between a sip hangup
and a "core show channel" command. This patch adds locking
to prevent the resulting crash.

(closes issue #12155)
Reported by: tsearle
Patches:
      show_channels_crash2.patch uploaded by tsearle (license 373)
Tested by: tsearle


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Tilghman Lesher
f26629cd1e Merged revisions 114061 via svnmerge from
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r114061 | tilghman | 2008-04-11 09:54:22 -0500 (Fri, 11 Apr 2008) | 6 lines

Errors are all greater than 0
(closes issue #12422)
 Reported by: nito
 Patches: 
       res_config_ldap_result_check_patch.diff uploaded by nito (license 340)

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2008-04-11 14:56:12 +00:00
Mark Michelson
fcbd180cec Merged revisions 114052 via svnmerge from
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r114052 | mmichelson | 2008-04-10 17:02:32 -0500 (Thu, 10 Apr 2008) | 11 lines

Merged revisions 114051 via svnmerge from 
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r114051 | mmichelson | 2008-04-10 15:59:49 -0500 (Thu, 10 Apr 2008) | 3 lines

Fix 1.4 build when LOW_MEMORY is enabled.


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2008-04-10 22:23:14 +00:00
Joshua Colp
74408cd537 Blocked revisions 114049 via svnmerge
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r114049 | file | 2008-04-10 17:28:40 -0300 (Thu, 10 Apr 2008) | 2 lines

A 'b' option has been added which causes chan_local to return the actual channel that is behind it when queried. This is useful for transfer scenarios as the actual channel will be transferred, not the Local channel. If you have been using Local channels as queue members and having issues when the agent did a blind transfer this option may solve the issue.

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2008-04-10 20:30:07 +00:00
Mark Michelson
3690b75234 Merged revisions 114046 via svnmerge from
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r114046 | mmichelson | 2008-04-10 14:58:36 -0500 (Thu, 10 Apr 2008) | 14 lines

Merged revisions 114045 via svnmerge from 
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r114045 | mmichelson | 2008-04-10 14:55:33 -0500 (Thu, 10 Apr 2008) | 6 lines

Be sure that we're not about to set bridgepvt NULL prior to dereferencing it.

(closes issue #11775)
Reported by: fujin


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Tilghman Lesher
f997490e97 Merged revisions 114042 via svnmerge from
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r114042 | tilghman | 2008-04-10 14:04:29 -0500 (Thu, 10 Apr 2008) | 7 lines

The hydra grows yet another head...
(closes issue #12401)
 Reported by: davevg
 Patches: 
       astcli.diff2 uploaded by davevg (license 209)
 Tested by: davevg, Corydon76

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2008-04-10 19:09:37 +00:00
Jason Parker
63336eb9f6 Merged revisions 114036 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r114036 | qwell | 2008-04-10 12:27:16 -0500 (Thu, 10 Apr 2008) | 18 lines

Merged revisions 114035 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114035 | qwell | 2008-04-10 12:26:10 -0500 (Thu, 10 Apr 2008) | 10 lines

Only try to prefix language if we are not using an absolute path (suffix it otherwise).

en/var/lib/asterisk/sounds/blah.gsm is a very silly path.

(closes issue #12379)
Reported by: kuj
Patches:
      12379-absolutepath.diff uploaded by qwell (license 4)
Tested by: kuj, qwell

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2008-04-10 17:27:35 +00:00
Joshua Colp
bcdf05dc3d Blocked revisions 114033 via svnmerge
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r114033 | file | 2008-04-10 12:59:13 -0300 (Thu, 10 Apr 2008) | 13 lines

Blocked revisions 114032 via svnmerge

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r114032 | file | 2008-04-10 12:58:39 -0300 (Thu, 10 Apr 2008) | 6 lines

Forgot the 1.4 branch for russian language fix.
(closes issue #12404)
Reported by: IgorG
Patches:
      voicemail_ru_hardcoded-v1.patch uploaded by IgorG (license 20)

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2008-04-10 16:00:25 +00:00
Joshua Colp
4f40fd35ab Merged revisions 114030 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r114030 | file | 2008-04-10 12:10:47 -0300 (Thu, 10 Apr 2008) | 14 lines

Merged revisions 114029 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114029 | file | 2008-04-10 12:09:04 -0300 (Thu, 10 Apr 2008) | 6 lines

Create the directory where name recordings will go if it does not exist.
(closes issue #12311)
Reported by: rkeene
Patches:
      12311-mkdir.diff uploaded by qwell (license 4)

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2008-04-10 15:11:34 +00:00
Joshua Colp
39d2b684ca Merged revisions 114027 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r114027 | file | 2008-04-10 11:53:19 -0300 (Thu, 10 Apr 2008) | 6 lines

Don't hardcode ru into the digits filename so that languageprefix can work.
(closes issue #12404)
Reported by: IgorG
Patches:
      voicemail_ru_hardcoded-v1.patch uploaded by IgorG (license 20)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@114028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-10 14:54:01 +00:00
Joshua Colp
f850d7d9c0 Merged revisions 114024 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r114024 | file | 2008-04-10 10:45:45 -0300 (Thu, 10 Apr 2008) | 4 lines

Fix spelling of existent in a few places.
(closes issue #12409)
Reported by: candlerb

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2008-04-10 13:46:33 +00:00
Joshua Colp
8c044be7be Merged revisions 114022 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r114022 | file | 2008-04-10 10:28:30 -0300 (Thu, 10 Apr 2008) | 14 lines

Merged revisions 114021 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114021 | file | 2008-04-10 10:27:11 -0300 (Thu, 10 Apr 2008) | 6 lines

Don't add custom URI options if they don't exist OR they are empty.
(closes issue #12407)
Reported by: homesick
Patches:
      uri_options-1.4.diff uploaded by homesick (license 91)

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2008-04-10 13:30:32 +00:00
Mark Michelson
3dd87d6d77 Merged revisions 113980 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r113980 | mmichelson | 2008-04-09 17:32:32 -0500 (Wed, 09 Apr 2008) | 8 lines

Fix a crash that happened due to accessing free'd memory

(closes issue #12396)
Reported by: tcalosi
Patches:
      12396.patch uploaded by putnopvut (license 60)
	  Tested by: tcalosi

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2008-04-09 22:34:04 +00:00
Mark Michelson
2389278c4a Merged revisions 113928 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r113928 | mmichelson | 2008-04-09 15:56:14 -0500 (Wed, 09 Apr 2008) | 16 lines

Merged revisions 113927 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r113927 | mmichelson | 2008-04-09 15:54:31 -0500 (Wed, 09 Apr 2008) | 8 lines

We need to set the persistant_route [sic] parameter for the sip_pvt
during the initial INVITE, no matter if we're building the route set from
an INVITE request or response.

(closes issue #12391)
Reported by: benjaminbohlmann
Tested by: benjaminbohlmann

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2008-04-09 20:56:51 +00:00
Tilghman Lesher
81136ad1fe Merged revisions 113875 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r113875 | tilghman | 2008-04-09 14:00:40 -0500 (Wed, 09 Apr 2008) | 12 lines

Merged revisions 113874 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r113874 | tilghman | 2008-04-09 13:57:33 -0500 (Wed, 09 Apr 2008) | 4 lines

If the [csv] section does not exist in cdr.conf, then an unload/load sequence
is needed to correct the problem.  Track whether the load succeeded with a
variable, so we can fix this with a simple reload event, instead.

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2008-04-09 19:02:27 +00:00
Joshua Colp
5699f7ce1e Blocked revisions 113840 via svnmerge
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r113840 | file | 2008-04-09 15:05:40 -0300 (Wed, 09 Apr 2008) | 4 lines

Enable enough RTP bridging to allow P2P to work.
(closes issue #11901)
Reported by: pj

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2008-04-09 18:06:46 +00:00
Jason Parker
2c7bcd89b6 Merged revisions 113838 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r113838 | qwell | 2008-04-09 12:56:07 -0500 (Wed, 09 Apr 2008) | 2 lines

Fix a small file handle "leak" pointed out by jjshoe on #asterisk.

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2008-04-09 17:56:28 +00:00
Mark Michelson
dc6dcc47c3 Merged revisions 113836 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r113836 | mmichelson | 2008-04-09 12:48:33 -0500 (Wed, 09 Apr 2008) | 14 lines

There was a subtle logical difference between 1.4 and trunk with regards to how timeouts
were handled. In 1.4, if the absolute timeout were reached on a call, no matter what
the return value of ast_spawn_extension was, the pbx would attempt to go to the 'T'
extension or hangup otherwise. The rearrangement of this function in trunk made this check
only happen in the case that ast_spawn_extension returned 0. If ast_spawn_extension returned
1, then the fact that the timeout expired resulted in a no-op, and would cause an infinite
loop to occur in __ast_pbx_run. This change fixes this problem. Now timeouts will
behave as they did in 1.4

(closes issue #11550)
Reported by: pj
Tested by: putnopvut


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2008-04-09 17:50:00 +00:00
Jason Parker
876f7bcc0a Blocked revisions 113834 via svnmerge
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r113834 | qwell | 2008-04-09 12:41:09 -0500 (Wed, 09 Apr 2008) | 7 lines

Move all messages wrapped in skinnydebug from debug to verbose.

(closes issue #12224)
Reported by: DEA
Patches:
      chan_skinny-debug-log.txt uploaded by DEA (license 3)

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2008-04-09 17:41:26 +00:00
Joshua Colp
950e46f357 Merged revisions 113785 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r113785 | file | 2008-04-09 13:52:04 -0300 (Wed, 09 Apr 2008) | 12 lines

Merged revisions 113784 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r113784 | file | 2008-04-09 13:50:45 -0300 (Wed, 09 Apr 2008) | 4 lines

If we receive an AUTHREQ from the remote server and we are unable to reply (for example they have a secret configured, but we do not) then queue a hangup frame on the Asterisk channel. This will cause the channel to hangup and a HANGUP to be sent via IAX2 to the remote side which is the proper thing to do in this scenario.
(closes issue #12385)
Reported by: viraptor

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2008-04-09 16:53:52 +00:00
Tilghman Lesher
f18fbcd8c9 Blocked revisions 113752 via svnmerge
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r113752 | tilghman | 2008-04-09 11:23:30 -0500 (Wed, 09 Apr 2008) | 2 lines

Mark recent additions from #11954 and #12254

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2008-04-09 16:24:47 +00:00
Tilghman Lesher
51bdc38e79 Blocked revisions 113731 via svnmerge
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r113731 | tilghman | 2008-04-09 11:16:44 -0500 (Wed, 09 Apr 2008) | 6 lines

Permit message wrap-around during message retrieval.
(closes issue #12254)
 Reported by: andrew
 Patches: 
       bug-12253.diff uploaded by snuffy (license 35)

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2008-04-09 16:18:13 +00:00
Mark Michelson
5f1b7ecd09 Merged revisions 113682 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r113682 | mmichelson | 2008-04-09 09:41:58 -0500 (Wed, 09 Apr 2008) | 17 lines

Merged revisions 113681 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r113681 | mmichelson | 2008-04-09 09:40:05 -0500 (Wed, 09 Apr 2008) | 9 lines

If Asterisk receives a 488 on an INVITE (not a reinvite), then
we should not send a BYE.

(closes issue #12392)
Reported by: fnordian
Patches:
      chan_sip.patch uploaded by fnordian (license 110) with small modification from me


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2008-04-09 14:42:48 +00:00
Tilghman Lesher
00c18989b7 Blocked revisions 113649 via svnmerge
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r113649 | tilghman | 2008-04-09 08:55:28 -0500 (Wed, 09 Apr 2008) | 6 lines

Permit callee to continue in the dialplan, after caller has hung up.
(closes issue #11954)
 Reported by: johan
 Patches: 
       app_dial_rev104031.patch uploaded by johan (license 334)

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2008-04-09 13:56:32 +00:00
Tilghman Lesher
13366c64b8 Merged revisions 113647 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r113647 | tilghman | 2008-04-09 08:23:44 -0500 (Wed, 09 Apr 2008) | 6 lines

Additional enhancements
(closes issue #12390)
 Reported by: tzafrir
 Patches: 
       astcli_fixes.diff uploaded by tzafrir (license 46)

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2008-04-09 13:24:50 +00:00
Terry Wilson
de98ac9a6c Merged revisions 113597 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r113597 | twilson | 2008-04-08 20:36:58 -0500 (Tue, 08 Apr 2008) | 10 lines

Merged revisions 113596 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r113596 | twilson | 2008-04-08 20:34:25 -0500 (Tue, 08 Apr 2008) | 2 lines

Initialize fr->cacheable to make valgrind happy

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2008-04-09 01:40:50 +00:00
Tilghman Lesher
7857509e29 Merged revisions 113559 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r113559 | tilghman | 2008-04-08 16:33:11 -0500 (Tue, 08 Apr 2008) | 6 lines

Add commandline tool for doing CLI commands through AMI (instead of using asterisk -rx)
(closes issue #12389)
 Reported by: davevg
 Patches: 
       astcli uploaded by davevg (license 209)

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2008-04-08 21:34:03 +00:00
Mark Michelson
c88831e436 Blocked revisions 113508 via svnmerge
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r113508 | mmichelson | 2008-04-08 14:09:16 -0500 (Tue, 08 Apr 2008) | 15 lines

Blocked revisions 113507 via svnmerge

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r113507 | mmichelson | 2008-04-08 14:07:38 -0500 (Tue, 08 Apr 2008) | 8 lines

Fix potential buffer overflow that could happen if more than 100 announce files
were specified when calling ParkAndAnnounce. This overflow is not exploitable remotely
and so there is no need for a security advisory.

(closes issue #12386)
Reported by: davidw


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2008-04-08 19:10:06 +00:00
Jason Parker
5e7a6ce8a3 Merged revisions 113505 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r113505 | qwell | 2008-04-08 13:49:21 -0500 (Tue, 08 Apr 2008) | 9 lines

Merged revisions 113504 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r113504 | qwell | 2008-04-08 13:48:55 -0500 (Tue, 08 Apr 2008) | 1 line

Add a little more that is required for previously added devices.
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2008-04-08 18:49:36 +00:00
Jason Parker
da72762cb6 Merged revisions 113455 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r113455 | qwell | 2008-04-08 13:08:35 -0500 (Tue, 08 Apr 2008) | 12 lines

Merged revisions 113454 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r113454 | qwell | 2008-04-08 13:07:49 -0500 (Tue, 08 Apr 2008) | 4 lines

Add support for several new(ish) devices - most notably, 7942/7945, 7962/7965, 7975.

Thanks to Greg Oliver for providing me the required information.

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2008-04-08 18:09:09 +00:00
Jason Parker
78f25bb0b8 Blocked revisions 113452 via svnmerge
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r113452 | qwell | 2008-04-08 12:32:42 -0500 (Tue, 08 Apr 2008) | 9 lines

Move AST_FEATURE_FLAG_* and FEATURE_RETURN_* to features.h so that they can be used by modules.

(closes issue #12384)
Reported by: fnordian
Patches:
      features.patch uploaded by fnordian (license 110)

(patch modified by me, to give FEATURE_RETURN_* an AST_ prefix)

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2008-04-08 17:32:59 +00:00
Jason Parker
2b454613e7 Merged revisions 113403 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r113403 | qwell | 2008-04-08 12:00:55 -0500 (Tue, 08 Apr 2008) | 9 lines

Merged revisions 113402 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r113402 | qwell | 2008-04-08 11:56:52 -0500 (Tue, 08 Apr 2008) | 1 line

Work around some silliness caused by sys/capability.h - this should fix compile errors a number of users have been experiencing.
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2008-04-08 17:01:41 +00:00
Tilghman Lesher
362adae65d Merged revisions 113400 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r113400 | tilghman | 2008-04-08 11:54:21 -0500 (Tue, 08 Apr 2008) | 14 lines

Merged revisions 113399 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r113399 | tilghman | 2008-04-08 11:51:28 -0500 (Tue, 08 Apr 2008) | 6 lines

Add security note on astgenkey's manpage.
(closes issue #12373)
 Reported by: lmamane
 Patches: 
       20080406__bug12373.diff.txt uploaded by Corydon76 (license 14)

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2008-04-08 16:56:43 +00:00
Tilghman Lesher
b22b4bb40c Merged revisions 113349 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r113349 | tilghman | 2008-04-08 10:48:58 -0500 (Tue, 08 Apr 2008) | 15 lines

Merged revisions 113348 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r113348 | tilghman | 2008-04-08 10:39:16 -0500 (Tue, 08 Apr 2008) | 7 lines

Move check for still-bridged channels out a little further, to avoid possible
deadlocks.  (Closes issue #12252)
Reported by: callguy
 Patches: 
       20080319__bug12252.diff.txt uploaded by Corydon76 (license 14)
 Tested by: callguy

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2008-04-08 15:50:52 +00:00
Joshua Colp
55af227d7c Blocked revisions 112874 via svnmerge
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r112874 | murf | 2008-04-04 22:33:13 -0300 (Fri, 04 Apr 2008) | 15 lines

Found a little problem with the sip request handling that could lead to a quick crash of asterisk, and a road to a DOS attack if left unfixed.

Attaching to a running asterisk with "telnet hostname 5060", I would input "something", then hit return three times, and asterisk crashes.

I traced it to handle_request_do(), which zeroes out the data (an ast_str ptr) if the string is too short. 
Instead of freeing the struct and nulling the pointer, it now just resets it, because this 
ast_str is expected by the calling routine to still be there after handle_request_do() returns.

This appears to fix the crash. I assume that it was introduced with ast_str's being adopted.  It's a subtle and easy-to-miss sort of problem.

I also found all the places where the req.data is freed, and made sure the ptr is Nulled out as well; 
no good leaving bad ptrs laying around-- I didn't need to do this, but it seemed a good thing to do...



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2008-04-08 15:10:05 +00:00
Joshua Colp
6b383a1bb5 Merged revisions 113297 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r113297 | file | 2008-04-08 12:05:35 -0300 (Tue, 08 Apr 2008) | 12 lines

Merged revisions 113296 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r113296 | file | 2008-04-08 12:03:43 -0300 (Tue, 08 Apr 2008) | 4 lines

If audio suddenly gets fed into one side of a channel after a lapse of frames flush the other factory so that old audio does not remain in the factory causing the sync code to not execute.
(closes issue #12296)
Reported by: jvandal

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2008-04-08 15:06:28 +00:00
Tilghman Lesher
7dca269ec0 Merged revisions 113245 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r113245 | tilghman | 2008-04-07 17:16:46 -0500 (Mon, 07 Apr 2008) | 2 lines

Additional note

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2008-04-07 22:17:53 +00:00
Jason Parker
143c8d59bc Merged revisions 113243 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r113243 | qwell | 2008-04-07 16:49:27 -0500 (Mon, 07 Apr 2008) | 1 line

Document 'originate' permission in manager sample config.
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2008-04-07 21:49:50 +00:00
Jeff Peeler
b9935ade93 Merged revisions 113241 via svnmerge from
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r113241 | jpeeler | 2008-04-07 16:35:48 -0500 (Mon, 07 Apr 2008) | 23 lines

Merged revisions 113013 via svnmerge from 
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r113013 | jpeeler | 2008-04-07 10:18:10 -0500 (Mon, 07 Apr 2008) | 15 lines

Merged revisions 113012 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008) | 7 lines

(closes issue #12362)
(closes issue #12372)
Reported by: vinsik
Tested by: tecnoxarxa

This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one.

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2008-04-07 21:36:47 +00:00
Mark Michelson
9fb92870ba Blocked revisions 113207 via svnmerge
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r113207 | mmichelson | 2008-04-07 15:22:06 -0500 (Mon, 07 Apr 2008) | 7 lines

This is a "fix" for something that's been bugging the crap out of me for a while.
The variable name "flag" to distinguish between whether a message is being forwarded or
is new is not a helpful name. The newly added doxygen documentation to app_voicemail is
tremendously helpful, but I still just...hate this variable name. I think is_new_message
is more indicative of what its purpose is.


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2008-04-07 20:26:34 +00:00
Jason Parker
bb9fa0affd Merged revisions 113119 via svnmerge from
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r113119 | qwell | 2008-04-07 13:02:51 -0500 (Mon, 07 Apr 2008) | 16 lines

Merged revisions 113118 via svnmerge from 
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r113118 | qwell | 2008-04-07 13:00:09 -0500 (Mon, 07 Apr 2008) | 8 lines

Allow playback with noanswer (and add earlyrtp option).

(closes issue #9077)
Reported by: pj
Patches:
      earlyrtp.diff uploaded by wedhorn (license 30)
Tested by: pj, qwell, DEA, wedhorn

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2008-04-07 19:10:45 +00:00
Tilghman Lesher
f0cf3b0fbe Merged revisions 113172 via svnmerge from
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r113172 | tilghman | 2008-04-07 14:06:46 -0500 (Mon, 07 Apr 2008) | 11 lines

Merged revisions 113117 via svnmerge from 
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r113117 | tilghman | 2008-04-07 12:51:49 -0500 (Mon, 07 Apr 2008) | 3 lines

Force ast_mktime() to check for DST, since strptime(3) does not.
(Closes issue #12374)

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2008-04-07 19:08:45 +00:00
Terry Wilson
e2df0aabd5 Blocked revisions 113170 via svnmerge
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r113170 | twilson | 2008-04-07 13:57:21 -0500 (Mon, 07 Apr 2008) | 2 lines

atoi(NULL) is bad

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2008-04-07 18:58:30 +00:00
Mark Michelson
a0d427bd18 Merged revisions 113066 via svnmerge from
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r113066 | mmichelson | 2008-04-07 11:12:30 -0500 (Mon, 07 Apr 2008) | 21 lines

Merged revisions 113065 via svnmerge from 
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r113065 | mmichelson | 2008-04-07 11:08:45 -0500 (Mon, 07 Apr 2008) | 13 lines

This fix prevents a deadlock that was experienced in chan_local. There was
deadlock prevention in place in chan_local, but it would not work in a specific
case because the channel was recursively locked. By unlocking the channel prior
to calling the generator's generate callback in ast_read_generator_actions(), we
prevent the recursive locking, and therefore the deadlock.

(closes issue #12307)
Reported by: callguy
Patches:
      12307.patch uploaded by putnopvut (license 60)
Tested by: callguy


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2008-04-07 16:13:23 +00:00
Jeff Peeler
d3b67ac906 Merged revisions 113013 via svnmerge from
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r113013 | jpeeler | 2008-04-07 10:18:10 -0500 (Mon, 07 Apr 2008) | 15 lines

Merged revisions 113012 via svnmerge from 
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r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008) | 7 lines

(closes issue #12362)
(closes issue #12372)
Reported by: vinsik
Tested by: tecnoxarxa

This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one.

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2008-04-07 15:28:23 +00:00
Joshua Colp
b2ad5c6b24 Blocked revisions 113009 via svnmerge
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r113009 | file | 2008-04-07 11:54:42 -0300 (Mon, 07 Apr 2008) | 2 lines

Put my slinfactory changes back in.

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2008-04-07 14:55:57 +00:00
Tilghman Lesher
613dcc788b Merged revisions 112972 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r112972 | tilghman | 2008-04-05 08:24:12 -0500 (Sat, 05 Apr 2008) | 6 lines

AsyncAGI should not close the manager session on error.
(closes issue #12370)
 Reported by: srt
 Patches: 
       asterisk-12370.diff uploaded by srt (license 378)

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2008-04-05 13:30:54 +00:00
Tilghman Lesher
7e5764f561 Erm, it's early... that should have been merged
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@112974 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-05 13:27:46 +00:00
Tilghman Lesher
f3e2d7c912 Blocked revisions 112972 via svnmerge
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r112972 | tilghman | 2008-04-05 08:24:12 -0500 (Sat, 05 Apr 2008) | 6 lines

AsyncAGI should not close the manager session on error.
(closes issue #12370)
 Reported by: srt
 Patches: 
       asterisk-12370.diff uploaded by srt (license 378)

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2008-04-05 13:26:01 +00:00
Terry Wilson
677ce67c4c Blocked revisions 112939 via svnmerge
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r112939 | twilson | 2008-04-05 02:58:42 -0500 (Sat, 05 Apr 2008) | 2 lines

Clean up some memory leak/ref counting issues

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2008-04-05 07:59:39 +00:00
Terry Wilson
ace4ba10e6 Blocked revisions 112906 via svnmerge
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r112906 | twilson | 2008-04-04 23:59:25 -0500 (Fri, 04 Apr 2008) | 2 lines

Multi-line support for phoneprov

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2008-04-05 05:01:11 +00:00
Philippe Sultan
8205252c82 Merged revisions 112821 via svnmerge from
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r112821 | phsultan | 2008-04-04 21:28:49 +0200 (Fri, 04 Apr 2008) | 9 lines

Merged revisions 112820 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r112820 | phsultan | 2008-04-04 21:26:15 +0200 (Fri, 04 Apr 2008) | 1 line

Free newly allocated channel before returning
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2008-04-04 19:30:35 +00:00
Philippe Sultan
acf6ac5a7f Merged revisions 112785 via svnmerge from
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r112785 | phsultan | 2008-04-04 19:32:46 +0200 (Fri, 04 Apr 2008) | 15 lines

Merged revisions 112766 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r112766 | phsultan | 2008-04-04 19:16:59 +0200 (Fri, 04 Apr 2008) | 7 lines

Prevent call connections when codecs don't match.

(closes issue #10604)
Reported by: keepitcool
Patches:
      branch-1.4-10604-2.diff uploaded by phsultan (license 73)
Tested by: phsultan
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2008-04-04 17:43:01 +00:00
Joshua Colp
c9c67f0685 Blocked revisions 112708,112710 via svnmerge
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r112708 | dhubbard | 2008-04-03 21:32:05 -0300 (Thu, 03 Apr 2008) | 1 line

blocked for trunk....woot
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r112710 | file | 2008-04-03 21:49:01 -0300 (Thu, 03 Apr 2008) | 9 lines

Blocked revisions 112709 via svnmerge

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r112709 | file | 2008-04-03 21:48:01 -0300 (Thu, 03 Apr 2008) | 2 lines

One thing at a time... let's get 1.4 building.

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2008-04-04 01:10:12 +00:00
Dwayne M. Hubbard
63a98b72eb Merged revisions 112653,112656,112714 via svnmerge from
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r112653 | dhubbard | 2008-04-03 17:13:11 -0500 (Thu, 03 Apr 2008) | 1 line

add a Zaptel timer check to verify the timer is responding when Zaptel support is compiled into Asterisk and Zaptel drivers are loaded.  This will help people not waste their valuable time debugging side effects.
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r112656 | dhubbard | 2008-04-03 17:19:43 -0500 (Thu, 03 Apr 2008) | 1 line

satisfy buildbot
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r112714 | dhubbard | 2008-04-03 19:57:33 -0500 (Thu, 03 Apr 2008) | 1 line

sleep long enough for the zaptel timer error message to display before exit
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2008-04-04 01:08:31 +00:00
Joshua Colp
81d7cf5925 Merged revisions 112712 via svnmerge from
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r112712 | file | 2008-04-03 21:53:19 -0300 (Thu, 03 Apr 2008) | 10 lines

Merged revisions 112711 via svnmerge from 
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r112711 | file | 2008-04-03 21:52:36 -0300 (Thu, 03 Apr 2008) | 2 lines

Pass in the path to Zaptel for systems that install Zaptel headers in a separate location.

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2008-04-04 00:54:22 +00:00
Mark Michelson
472c4a22dd Merged revisions 112600 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r112600 | mmichelson | 2008-04-03 09:35:47 -0500 (Thu, 03 Apr 2008) | 17 lines

Merged revisions 112599 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r112599 | mmichelson | 2008-04-03 09:32:20 -0500 (Thu, 03 Apr 2008) | 9 lines

Fix the testing of the "res" variable so that it is more logically correct and 
makes the correct warning and debug messages print.

(closes issue #12361)
Reported by: one47
Patches:
      chan_zap_deferred_digit.patch uploaded by one47 (license 23)


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2008-04-03 14:42:58 +00:00
Tilghman Lesher
30d749bde2 Blocked revisions 112321 via svnmerge
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r112321 | jpeeler | 2008-04-01 17:07:30 -0500 (Tue, 01 Apr 2008) | 2 lines

Existing DNS manager lookups extended to check for SRV records.

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2008-04-03 07:51:17 +00:00
Tilghman Lesher
e6c0a30bca Blocked revisions 112564 via svnmerge
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r112564 | tilghman | 2008-04-03 02:49:05 -0500 (Thu, 03 Apr 2008) | 7 lines

Use a 32k file buffer on recordings, which increases the efficiency of file recording.
(closes issue #11962)
 Reported by: garlew
 Patches: 
       recording.patch uploaded by garlew (license 376)
       bug-11962.diff uploaded by snuffy (license 35)

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2008-04-03 07:49:51 +00:00
Terry Wilson
2f899cfcb0 Blocked revisions 112426 via svnmerge
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r112426 | twilson | 2008-04-02 10:25:48 -0500 (Wed, 02 Apr 2008) | 2 lines

Re-add HTTP post support by moving to res_http_post.c

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2008-04-02 19:53:34 +00:00
Tilghman Lesher
bdf9950e55 Blocked revisions 112520 via svnmerge
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r112520 | tilghman | 2008-04-02 14:34:52 -0500 (Wed, 02 Apr 2008) | 6 lines

Make MISDN generate channel rename events when the name changes.
(closes issue #11142)
 Reported by: julianjm
 Patches: 
       chan_misdn_tmpchan_trunk_v1.diff uploaded by julianjm (license 99)

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2008-04-02 19:36:05 +00:00
Mark Michelson
88eacb2e27 Merged revisions 112469 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r112469 | mmichelson | 2008-04-02 12:36:49 -0500 (Wed, 02 Apr 2008) | 21 lines

Merged revisions 112468 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r112468 | mmichelson | 2008-04-02 12:36:04 -0500 (Wed, 02 Apr 2008) | 13 lines

Fix a race condition in the manager. It is possible that a new manager event
could be appended during a brief time when the manager is not waiting for input.
If an event comes during this period, we need to set an indicator that there is an
event pending so that the manager doesn't attempt to wait forever for an event that
already happened.

(closes issue #12354)
Reported by: bamby
Patches:
      manager_race_condition.diff uploaded by bamby (license 430)
	  (comments added by me)


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2008-04-02 17:37:52 +00:00
Joshua Colp
b928d453af Merged revisions 112431 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r112431 | file | 2008-04-02 12:26:51 -0300 (Wed, 02 Apr 2008) | 7 lines

Since the SIP request structure gets reused multiple times with TCP handling we have to clear the debug state or else we will keep spitting out debug even after it has been turned off.
(closes issue #12169)
Reported by: pj
Patches:
      12169-debugoff-2.diff uploaded by qwell (license 4)
Tested by: pj

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2008-04-02 15:27:56 +00:00
Mark Michelson
bb75d40666 Merged revisions 112394 via svnmerge from
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r112394 | mmichelson | 2008-04-02 09:32:43 -0500 (Wed, 02 Apr 2008) | 14 lines

Merged revisions 112393 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r112393 | mmichelson | 2008-04-02 09:32:00 -0500 (Wed, 02 Apr 2008) | 6 lines

Ensure that there is no timeout if none is specified.

(closes issue #12349)
Reported by: johnlange


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2008-04-02 14:33:33 +00:00
Jeff Peeler
c8163da651 Blocked revisions 112360 via svnmerge
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r112360 | jpeeler | 2008-04-01 17:55:28 -0500 (Tue, 01 Apr 2008) | 2 lines

Added dnsmgr status output for sip show registry.

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2008-04-01 22:57:20 +00:00
Steve Murphy
f4492e33c7 Merged revisions 112357 via svnmerge from
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r112357 | murf | 2008-04-01 16:45:10 -0600 (Tue, 01 Apr 2008) | 1 line

Bumped across another test set for the new exten pattern matcher, which revealed a problem with the CANMATCH/MATCHMORE modes. Direct matches were getting in the way. Fixed.
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2008-04-01 22:48:06 +00:00
Russell Bryant
429dee1033 Blocked revisions 112351 via svnmerge
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r112351 | russell | 2008-04-01 17:25:45 -0500 (Tue, 01 Apr 2008) | 2 lines

Fix a typo that prevented configuration of non-dynamic peers.

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2008-04-01 22:26:08 +00:00
Steve Murphy
52c5c0e2b2 Merged revisions 112289 via svnmerge from
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r112289 | murf | 2008-04-01 14:02:19 -0600 (Tue, 01 Apr 2008) | 21 lines

(closes issue #12298)
Reported by: falves11
Patches:
      12298.patch1 uploaded by murf (license 17)
Tested by: murf

I have hopes that the changes made over the last few days will
finalize and solidify this code. While there are bound to be 
small tweaks still needed, I feel that the job (at last) is
somewhat completed. Finally, I had a chance to comprehend how
the scoring of extension patterns was done in the previous
version, and I've come very close to using the exact same
criteria in the new pattern matching code. The left-right
sorting is now replicated in the trie structure itself, such
that the first match found will the 'best' match. Compared
the results against 1.4 for several extensions. Replicated
falves11's setup and it works. Used some devious patterns
provided by jsmith, supplemented with a few of my own.
Looks good.


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2008-04-01 20:20:34 +00:00
Tilghman Lesher
d2c02a474f Blocked revisions 112124,112234 via svnmerge
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r112124 | russell | 2008-04-01 11:35:04 -0500 (Tue, 01 Apr 2008) | 2 lines

Now that zaptel trunk has been removed, add the PSTN deprecation notice to chan_zap, as well.

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r112234 | tilghman | 2008-04-01 13:23:40 -0500 (Tue, 01 Apr 2008) | 2 lines

Fix last commit

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2008-04-01 18:46:11 +00:00
Jason Parker
c5d0fc87e7 Blocked revisions 112241 via svnmerge
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r112241 | qwell | 2008-04-01 13:24:56 -0500 (Tue, 01 Apr 2008) | 7 lines

More voicemail doxygen additions/cleanup.

(issue #12343)
Reported by: travishein
Patches:
      app_voicemail_code_documentation.patch uploaded by travishein (license 385)

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2008-04-01 18:28:30 +00:00
Jason Parker
163802fbd5 Blocked revisions 112252 via svnmerge
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r112252 | qwell | 2008-04-01 13:27:08 -0500 (Tue, 01 Apr 2008) | 7 lines

Minor formatting cleanup.

(closes issue #12343)
Reported by: travishein
Patches:
      app_voicemail_code_convention.patch uploaded by travishein (license 385)

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2008-04-01 18:28:02 +00:00
Joshua Colp
8566c2eb9e Merged revisions 112210 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r112210 | file | 2008-04-01 15:06:13 -0300 (Tue, 01 Apr 2008) | 12 lines

Merged revisions 112209 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r112209 | file | 2008-04-01 15:02:43 -0300 (Tue, 01 Apr 2008) | 4 lines

Disable Packet2Packet bridging when we need to feed DTMF frames into the core. Some implementations do not like how we switch between things.
(closes issue #12212)
Reported by: bamby

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2008-04-01 18:09:01 +00:00
Jeff Peeler
0ff33fbcd8 Blocked revisions 112207 via svnmerge
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r112207 | jpeeler | 2008-04-01 12:53:08 -0500 (Tue, 01 Apr 2008) | 3 lines

This adds DNS SRV record support to DNS manager.  If there is a SRV record for a given domain, the hostname and port listed in the SRV record will be used. If no SRV record exists or a SRV lookup is not attempted, the DNS lookup on the specified domain will be performed as normal. Chan_sip has been modified to take advantage of the new SRV support.


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2008-04-01 17:54:48 +00:00
Joshua Colp
2dfca4e149 Merged revisions 112205 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r112205 | file | 2008-04-01 14:48:52 -0300 (Tue, 01 Apr 2008) | 12 lines

Merged revisions 112204 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r112204 | file | 2008-04-01 14:43:46 -0300 (Tue, 01 Apr 2008) | 4 lines

Do not pass audio until the remote side has indicated they are providing early media, or if the channel has been answered.
(closes issue #11823)
Reported by: SDamm

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2008-04-01 17:52:44 +00:00
Joshua Colp
597fc244b3 Blocked revisions 112155 via svnmerge
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r112155 | file | 2008-04-01 14:24:45 -0300 (Tue, 01 Apr 2008) | 6 lines

Demote a log message down to a warning.
(closes issue #12345)
Reported by: caio1982
Patches:
      limit_msg.diff uploaded by caio1982 (license 22)

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2008-04-01 17:27:42 +00:00
Mark Michelson
bfb1ccad29 Merged revisions 112148 via svnmerge from
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r112148 | mmichelson | 2008-04-01 12:23:19 -0500 (Tue, 01 Apr 2008) | 18 lines

Merged revisions 112138 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r112138 | mmichelson | 2008-04-01 12:21:21 -0500 (Tue, 01 Apr 2008) | 10 lines

Initialize the __res_state structure used for dns purposes
to all 0's prior to using it. This is due to valgrind's complaints
on issue #12284 as well as an excerpt found in "Description" portion
of the online man page found here:

http://www.iti.cs.tu-bs.de/cgi-bin/UNIXhelp/man-cgi?res_nquery+3RESOLV

(pertains to issue #12284 but does not necessarily close it)


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2008-04-01 17:25:05 +00:00
Joshua Colp
346c1b5ae6 Merged revisions 112126 via svnmerge from
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r112126 | file | 2008-04-01 13:50:37 -0300 (Tue, 01 Apr 2008) | 13 lines

Merged revisions 112125 via svnmerge from 
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r112125 | file | 2008-04-01 13:45:14 -0300 (Tue, 01 Apr 2008) | 5 lines

Ensure that we do not exceed the hold's maximum size with a single frame.
(closes issue #12047)
Reported by: fabianoheringer
Tested by: fabianoheringer

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2008-04-01 16:57:16 +00:00
Jason Parker
4358d42970 Blocked revisions 112071 via svnmerge
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r112071 | qwell | 2008-03-31 17:16:34 -0500 (Mon, 31 Mar 2008) | 7 lines

I missed a place when this define was changed.

(closes issue #12334)
Reported by: ovi
Patches:
      12334-asterisk.patch uploaded by dimas (license 88)

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2008-03-31 22:17:53 +00:00
Jason Parker
c0e61fdcaf Merged revisions 112069 via svnmerge from
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r112069 | qwell | 2008-03-31 16:48:30 -0500 (Mon, 31 Mar 2008) | 13 lines

Merged revisions 112068 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r112068 | qwell | 2008-03-31 16:48:05 -0500 (Mon, 31 Mar 2008) | 5 lines

Fix a silly infinite loop when choosing an invalid option.

(closes issue #12315)
Reported by: jmls

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Terry Wilson
ae356a6d1f Blocked revisions 112035 via svnmerge
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r112035 | twilson | 2008-03-31 16:01:59 -0500 (Mon, 31 Mar 2008) | 2 lines

Yeah, simplify that logic a bit...

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Terry Wilson
8991cce7b4 Merged revisions 112033 via svnmerge from
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r112033 | twilson | 2008-03-31 15:45:05 -0500 (Mon, 31 Mar 2008) | 2 lines

Handle blank prefix= in http.conf

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2008-03-31 20:55:40 +00:00
Russell Bryant
d265edff9f Merged revisions 111998 via svnmerge from
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r111998 | russell | 2008-03-31 12:14:58 -0500 (Mon, 31 Mar 2008) | 7 lines

Ensure configure gets run on a clean checkout.

(closes issue #12197)
Reported by: juggie
Patches:
      12197.diff uploaded by juggie (license 24)

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Russell Bryant
e72038d2c9 Blocked revisions 111996 via svnmerge
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r111996 | russell | 2008-03-31 11:37:13 -0500 (Mon, 31 Mar 2008) | 2 lines

This fixes a high fence violation that MALLOC_DEBUG reported to me.

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Joshua Colp
e6e67bde76 Merged revisions 111961 via svnmerge from
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r111961 | file | 2008-03-31 11:20:39 -0300 (Mon, 31 Mar 2008) | 4 lines

Initialize all these here tmp pointers at declaration. They confused some compilers a wee bit.
(closes issue #12333)
Reported by: ovi

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Russell Bryant
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r111908 | russell | 2008-03-28 17:45:43 -0500 (Fri, 28 Mar 2008) | 3 lines

Note a minor race condition that I noticed while reviewing Jeff's changes
to this code.

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r111909 | russell | 2008-03-28 17:50:46 -0500 (Fri, 28 Mar 2008) | 3 lines

Make some notes about common usage of pbx_builtin_getvar_helper() that is not
thread-safe.

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2008-03-28 22:52:02 +00:00
Jason Parker
557219e0dc Merged revisions 111857 via svnmerge from
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r111857 | qwell | 2008-03-28 16:46:02 -0500 (Fri, 28 Mar 2008) | 20 lines

Merged revisions 111856 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r111856 | qwell | 2008-03-28 16:45:35 -0500 (Fri, 28 Mar 2008) | 12 lines

Allow gsm to compile correctly on x86 with gcc4 optimizations.

(closes issue #11243)
Reported by: whiskerp
Patches:
      11243-maybe-asm.diff uploaded by qwell (license 4)
Tested by: Seggy (IRC)

Note: While I did write this patch, I would not have found this if fossil
 had not reported and fixed issue #12253.  A huge thanks to him for helping
 to (indirectly) find the problem here.

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Mark Michelson
b185f4f733 Blocked revisions 111811 via svnmerge
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r111811 | mmichelson | 2008-03-28 15:03:16 -0500 (Fri, 28 Mar 2008) | 11 lines

This time the fix is proper for issue 12284. I have tested it thoroughly and found
that valgrind no longer complains and that calls do complete correctly.

The fix is along the same lines as before: Make sure the final null terminator gets copied
into the new sip_request's data pointer. Without it, parse_request will read and potentially
write past the end of the string, causing potential crashes.

(closes issue #12284...for real this time!)
reported by falves11


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2008-03-28 20:03:48 +00:00
Mark Michelson
339c3eda73 Blocked revisions 111777 via svnmerge
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r111777 | mmichelson | 2008-03-28 14:14:51 -0500 (Fri, 28 Mar 2008) | 6 lines

Temporary revert of 111662. It's causing lots of trouble and appears to not be
the proper solution to the problem reported anyway.

(related to issue #12884)


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Jason Parker
14b73f54b3 Blocked revisions 111773-111774 via svnmerge
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r111773 | qwell | 2008-03-28 14:06:54 -0500 (Fri, 28 Mar 2008) | 7 lines

Add a bit of doxygen documentation for app_voicemail.

(issue #12326)
Reported by: travishein
Patches:
      app_voicemail_code_documentation.patch uploaded by travishein (license 385)

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r111774 | qwell | 2008-03-28 14:08:35 -0500 (Fri, 28 Mar 2008) | 8 lines

Replace magic number size from msgArray array with a define.

(same patch as before, I just split this part out)
(close issue #12326)
Reported by: travishein
Patches:
      app_voicemail_code_documentation.patch uploaded by travishein (license 385)

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2008-03-28 19:11:14 +00:00
Jason Parker
1b23f850e9 Merged revisions 111721 via svnmerge from
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r111721 | qwell | 2008-03-28 12:57:12 -0500 (Fri, 28 Mar 2008) | 9 lines

Merged revisions 111720 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r111720 | qwell | 2008-03-28 12:55:05 -0500 (Fri, 28 Mar 2008) | 1 line

Remove unimplemented softkeys.  Prompted by issue #12325.
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2008-03-28 17:57:31 +00:00
Mark Michelson
2caae1b4c0 Blocked revisions 111662 via svnmerge
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r111662 | mmichelson | 2008-03-28 11:36:59 -0500 (Fri, 28 Mar 2008) | 9 lines

The copy_request function did not take into account the necessary null terminator
for the string to be copied into. This resulted in parse_request reading invalid
memory beyond the end of the string, and in some cases led to crashes. Thanks
to falves11 for providing the valgrind output which led to the closure of this issue.

(closes issue #12284)
Reported by: falves11


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Jason Parker
89596690ee Merged revisions 111659 via svnmerge from
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r111659 | qwell | 2008-03-28 11:20:59 -0500 (Fri, 28 Mar 2008) | 16 lines

Merged revisions 111658 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r111658 | qwell | 2008-03-28 11:19:56 -0500 (Fri, 28 Mar 2008) | 8 lines

The file size of WAV49 does not need to be an even number.

(closes issue #12128)
Reported by: mdu113
Patches:
      12128-noevenlength.diff uploaded by qwell (license 4)
Tested by: qwell, mdu113

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2008-03-28 16:21:31 +00:00
Tilghman Lesher
4ffd366972 Blocked revisions 110930 via svnmerge
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r110930 | juggie | 2008-03-26 12:28:49 -0500 (Wed, 26 Mar 2008) | 1 line

revert something dumb, because i was running svn diff in a subfolder not the root of trunk, before doing my commit and did not see it
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2008-03-28 14:43:11 +00:00
Tilghman Lesher
3d35d8702d Merged revisions 111606 via svnmerge from
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r111606 | tilghman | 2008-03-28 09:37:28 -0500 (Fri, 28 Mar 2008) | 11 lines

Merged revisions 111605 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r111605 | tilghman | 2008-03-28 09:35:45 -0500 (Fri, 28 Mar 2008) | 3 lines

Update debugging text, since Valgrind eliminated the --log-file-exactly option.
(Closes issue #12320)

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2008-03-28 14:38:18 +00:00
Joshua Colp
0ee953a249 Merged revisions 111565 via svnmerge from
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r111565 | file | 2008-03-27 21:55:47 -0300 (Thu, 27 Mar 2008) | 2 lines

Forgetting to unregister a manager action is bad, mmmk?

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2008-03-28 00:56:44 +00:00
Mark Michelson
ec8ef5229c Merged revisions 111533 via svnmerge from
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r111533 | mmichelson | 2008-03-27 19:12:52 -0500 (Thu, 27 Mar 2008) | 10 lines

Fix a crash that would happen when attempting to unload the app_queue module.

The problem was that when the refcount on the queue hit 0, the destructor was
called, and inside the destructor, another function was called which would increase
the refcount back to 1 again and then decrease it again back to 0 for every member
in the queue. This meant that the destructor was being recursively called, leading
to a double free of the queue. This is now fixed by making sure to unlink the
queue from the queues container prior to the final unref of the queue.


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Terry Wilson
a850a9a13e Blocked revisions 111213 via svnmerge
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r111213 | twilson | 2008-03-26 16:23:29 -0500 (Wed, 26 Mar 2008) | 2 lines

Stupid strcasecmp function :-)

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Terry Wilson
c7c4e7a56a Blocked revisions 111500 via svnmerge
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r111500 | twilson | 2008-03-27 17:10:25 -0500 (Thu, 27 Mar 2008) | 2 lines

Fix another little http problem.  In making it match coding guidelines, a comparison was dropped

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3a151beb5f Merged revisions 111497 via svnmerge from
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r111497 | murf | 2008-03-27 15:25:55 -0600 (Thu, 27 Mar 2008) | 1 line

comment cleanup and iron out a really dumb mistake in handling the '.'-wildcard in the new exten pattern matcher.
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Tilghman Lesher
8955b42372 Merged revisions 111443 via svnmerge from
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r111443 | tilghman | 2008-03-27 14:26:45 -0500 (Thu, 27 Mar 2008) | 14 lines

Merged revisions 111442 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r111442 | tilghman | 2008-03-27 14:23:12 -0500 (Thu, 27 Mar 2008) | 6 lines

For FreeBSD, at least, the ifa_addr element could be NULL.
(closes issue #12300)
 Reported by: festr
 Patches: 
       acl.c.patch uploaded by festr (license 443)

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Steve Murphy
b218cc2c99 Merged revisions 111410 via svnmerge from
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r111410 | murf | 2008-03-27 07:29:41 -0600 (Thu, 27 Mar 2008) | 17 lines

Merged revisions 111391 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r111391 | murf | 2008-03-27 07:03:28 -0600 (Thu, 27 Mar 2008) | 9 lines

These small documentation updates made in response to a query in
asterisk-users, where a user was using Playback, but needed the
features of Background, and had no idea that Background existed,
or that it might provide the features he needed. I thought the
best way to avert these kinds of queries was to provide "See Also"
references in all three of "Background", "Playback", "WaitExten".
Perhaps a project to do this with all related apps is in order.


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Steve Murphy
c51c46ba7f Merged revisions 111360 via svnmerge from
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r111360 | murf | 2008-03-26 22:47:12 -0600 (Wed, 26 Mar 2008) | 23 lines

Merged revisions 111341 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r111341 | murf | 2008-03-26 21:21:05 -0600 (Wed, 26 Mar 2008) | 15 lines


(closes issue #12302)
Reported by: pj
Tested by: murf

These changes will set a channel variable ~~EXTEN~~ just before generating code
for a switch, with the value of ${EXTEN}. The exten is marked as having a switch, 
and ever after that, till the end of the exten, we substitute any ${EXTEN} 
with ${~~EXTEN~~} instead in application arguments; (and the ${EXTEN: also). 
The reason for this, is that because switches are coded using 
separate extensions to provide pattern matching, and
jumping to/from these switch extensions messes up the ${EXTEN} value, 
which blows the minds of users.


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Jason Parker
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r111295 | qwell | 2008-03-26 19:27:35 -0500 (Wed, 26 Mar 2008) | 1 line

But we can change the API here.
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r111285 | qwell | 2008-03-26 19:25:56 -0500 (Wed, 26 Mar 2008) | 9 lines

Merged revisions 111280 via svnmerge from 
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r111280 | qwell | 2008-03-26 19:25:13 -0500 (Wed, 26 Mar 2008) | 1 line

Put this flag back so we don't change the API.
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r111246 | qwell | 2008-03-26 18:27:33 -0500 (Wed, 26 Mar 2008) | 17 lines

Merged revisions 111245 via svnmerge from 
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r111245 | qwell | 2008-03-26 18:26:33 -0500 (Wed, 26 Mar 2008) | 9 lines

Remove excessive smoother optimization that was causing audio glitches (small "pops")
 after (about 200ms later) an "incorrectly" sized frame was received.

While it would be very nice to keep this as optimized as possible, it makes no sense
 for the smoother to be dropping random bits of audio like this.  Isn't that the
 whole point of a smoother?

Closes issue #12093.

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Tilghman Lesher
203f3ba522 Blocked revisions 111185 via svnmerge
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r111185 | tilghman | 2008-03-26 15:34:05 -0500 (Wed, 26 Mar 2008) | 2 lines

Oops, missed one

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Tilghman Lesher
c6595f5201 Blocked revisions 111132 via svnmerge
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r111132 | tilghman | 2008-03-26 14:58:09 -0500 (Wed, 26 Mar 2008) | 2 lines

Simplify new macro, simplify configfile logic, now that list is sorted

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Joshua Colp
c61c83eedb Merged revisions 111130 via svnmerge from
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r111130 | file | 2008-03-26 16:56:40 -0300 (Wed, 26 Mar 2008) | 14 lines

Merged revisions 111129 via svnmerge from 
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r111129 | file | 2008-03-26 16:55:08 -0300 (Wed, 26 Mar 2008) | 6 lines

Update autosupport script.
(closes issue #12310)
Reported by: angler
Patches:
      autosupport.diff uploaded by angler (license 106)

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2008-03-26 19:57:54 +00:00
Kevin P. Fleming
477bf823b0 Merged revisions 111127 via svnmerge from
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r111127 | kpfleming | 2008-03-26 14:52:27 -0500 (Wed, 26 Mar 2008) | 18 lines

Merged revisions 111126 via svnmerge from 
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r111126 | kpfleming | 2008-03-26 14:51:24 -0500 (Wed, 26 Mar 2008) | 10 lines

Merged revisions 111125 via svnmerge from 
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r111125 | kpfleming | 2008-03-26 14:49:30 -0500 (Wed, 26 Mar 2008) | 2 lines

update UPGRADE notes to document usage of the script

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2008-03-26 19:53:12 +00:00
Mark Michelson
b0951fcb92 Merged revisions 111123 via svnmerge from
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r111123 | mmichelson | 2008-03-26 14:39:23 -0500 (Wed, 26 Mar 2008) | 12 lines

Merged revisions 111121 via svnmerge from 
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r111121 | mmichelson | 2008-03-26 14:37:36 -0500 (Wed, 26 Mar 2008) | 4 lines

This code change is made just for clarification. It does exactly
the same thing as before. It just doesn't look as wrong.


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Joshua Colp
ff7deea0b5 Blocked revisions 111083 via svnmerge
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r111083 | file | 2008-03-26 16:29:26 -0300 (Wed, 26 Mar 2008) | 4 lines

Add expiry value to the sip show subscriptions CLI command.
(closes issue #12025)
Reported by: agx

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2008-03-26 19:30:50 +00:00
Mark Michelson
2b6b4258cb Merged revisions 111067 via svnmerge from
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r111067 | mmichelson | 2008-03-26 14:26:23 -0500 (Wed, 26 Mar 2008) | 17 lines

Merged revisions 111049 via svnmerge from 
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r111049 | mmichelson | 2008-03-26 14:22:16 -0500 (Wed, 26 Mar 2008) | 9 lines

Add a lock to the vm_state structure and use the lock around mail_open calls
to prevent concurrent access of the same mailstream. This, along with trunk's
ability to configure TCP timeouts for IMAP storage will help to prevent
crashes and hangs when using voicemail with IMAP storage.

(closes issue #10487)
Reported by: ewilhelmsen


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Tilghman Lesher
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r111036 | tilghman | 2008-03-26 14:19:31 -0500 (Wed, 26 Mar 2008) | 2 lines

Add a linkedlist macro that maintains a sorted list

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Jason Parker
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r111028 | qwell | 2008-03-26 14:16:31 -0500 (Wed, 26 Mar 2008) | 4 lines

Only try to detect silence when we actually need to, instead of...always.

If this is wrong, I'd love to hear why.

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Jason Parker
5f0c300242 Blocked revisions 111022 via svnmerge
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r111022 | qwell | 2008-03-26 14:05:51 -0500 (Wed, 26 Mar 2008) | 23 lines

Large cleanup of DSP code

Per comments from dimas:
1. The code now generates DTMF_BEGIN frames in addition to DTMF_END ones.

2. "quelching" rewritten - now each detector (MF/DTMF/generic tone) may mark fragment of a frame for suppression (squelching, muting) with a call to mute_fragment. Actual muting happens only once at the very end of ast_dsp_process where all marked fragments are zeroed. This way every detector sees original data in the frame without any piece of a frame being zeroed by a detector which was run before.

3. DTMF detector tries to "mute" one block before and one block after the block where actual tone was detected. Muting of previois block is something new for this patch. Obviously this operation is not always possible - if current frame does not contain data for previous block - it is too late. But at least we make our best.
Muting of next block was already done by the old code but it only affects part of the next block which is in the frame being processed. New code keeps this information in state structures so it will mute proper number of samples in the next frame(s) too.

4. Removed ast_dsp_digitdetect and ast_dsp_getdigits APIs because these are not used.

5. DSP API extended a bit - ast_dsp_was_muted() function added which returns true if DSP code was muting any fragment in the last frame. chan_zap uses this function to decide it needs to turn on confmute on the channel.
This is to replace AST_FRAME_DTMF 'm'/'u' (mute/unmute) functionality.


(closes issue #11968)
Reported by: dimas
Patches:
      v2-11968-dsp.patch uploaded by dimas (license 88)
      v4-11968-zap.patch uploaded by dimas (license 88)
Tested by: dimas, qwell

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Kevin P. Fleming
7d84d643e0 Merged revisions 111025 via svnmerge from
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r111025 | kpfleming | 2008-03-26 14:08:00 -0500 (Wed, 26 Mar 2008) | 18 lines

Merged revisions 111024 via svnmerge from 
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r111024 | kpfleming | 2008-03-26 14:06:56 -0500 (Wed, 26 Mar 2008) | 10 lines

Merged revisions 111019 via svnmerge from 
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r111019 | kpfleming | 2008-03-26 13:58:37 -0500 (Wed, 26 Mar 2008) | 2 lines

add a script to make getting the iLBC source code simple for end users

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Joshua Colp
6506b05636 Merged revisions 111021 via svnmerge from
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r111021 | file | 2008-03-26 16:05:42 -0300 (Wed, 26 Mar 2008) | 12 lines

Merged revisions 111020 via svnmerge from 
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r111020 | file | 2008-03-26 16:04:35 -0300 (Wed, 26 Mar 2008) | 4 lines

If we are requested to authenticate a reinvite make sure that it contains T38 SDP if need be.
(closes issue #11995)
Reported by: fall

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Joshua Colp
838e70cdfa Merged revisions 111017 via svnmerge from
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r111017 | file | 2008-03-26 15:42:52 -0300 (Wed, 26 Mar 2008) | 12 lines

Merged revisions 110628 via svnmerge from 
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r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 lines

Add an option (transmit_silence) which transmits silence during both Record() and DTMF generation. The reason this is an option is that in order to transmit silence we have to setup a translation path. This may not be needed/wanted in all cases.
(closes issue #10058)
Reported by: tracinet

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2008-03-26 18:44:32 +00:00
Tilghman Lesher
c9b2181168 Blocked revisions 111012-111013 via svnmerge
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r111012 | tilghman | 2008-03-26 13:39:06 -0500 (Wed, 26 Mar 2008) | 3 lines

Add the "config reload <conffile>" command, which allows you to tell Asterisk
to reload any file that references a given configuration file.

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r111013 | tilghman | 2008-03-26 13:41:27 -0500 (Wed, 26 Mar 2008) | 2 lines

Oops, fix this, too

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2008-03-26 18:42:31 +00:00
Kevin P. Fleming
28154344e7 Merged revisions 110963 via svnmerge from
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r110963 | kpfleming | 2008-03-26 12:44:09 -0500 (Wed, 26 Mar 2008) | 10 lines

Merged revisions 110962 via svnmerge from 
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r110962 | kpfleming | 2008-03-26 12:43:02 -0500 (Wed, 26 Mar 2008) | 2 lines

add note that the user will need to enable codec_ilbc to get it to build

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2008-03-26 17:44:35 +00:00
Donny Kavanagh
ea128d010a Merged revisions 110911 via svnmerge from
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r110911 | juggie | 2008-03-26 13:24:54 -0400 (Wed, 26 Mar 2008) | 8 lines

update documentation to reflect the changes in the way configure detects net-snmp.

(closes issue #12067)
Reported by: juggie
Patches:
      12067_snmp_doc.patch uploaded by juggie (license 24)
Tested by: juggie

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2008-03-26 17:35:25 +00:00
Kevin P. Fleming
085ef90375 Merged revisions 110881 via svnmerge from
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r110881 | kpfleming | 2008-03-26 10:10:28 -0700 (Wed, 26 Mar 2008) | 18 lines

Merged revisions 110880 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110880 | kpfleming | 2008-03-26 09:42:35 -0700 (Wed, 26 Mar 2008) | 10 lines

Merged revisions 110869 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar 2008) | 2 lines

due to licensing restrictions, we cannot distribute the source code for iLBC encoding and decoding... so remove it, and add instructions on how the user can obtain it themselves

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2008-03-26 17:15:06 +00:00
Joshua Colp
fbe5770d1d Merged revisions 110726 via svnmerge from
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r110726 | jpeeler | 2008-03-25 17:02:57 -0300 (Tue, 25 Mar 2008) | 2 lines

This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one.

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2008-03-26 15:33:21 +00:00
Joshua Colp
f0320a42b9 Blocked revisions 110639 via svnmerge
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r110639 | mmichelson | 2008-03-25 12:44:01 -0300 (Tue, 25 Mar 2008) | 3 lines

Oops here too. I need to stop coding for a while...


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2008-03-26 15:31:28 +00:00
Mark Michelson
4cd89a7e8f Merged revisions 110831 via svnmerge from
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r110831 | mmichelson | 2008-03-25 19:02:31 -0500 (Tue, 25 Mar 2008) | 6 lines

This ensures that the manager interface is not enabled by default. Prior to this
change, it was possible to start Asterisk with the manager interface enabled, then
either comment out the enabled option or make manager.conf unopenable and the manager
interface would still be enabled.


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2008-03-26 00:03:12 +00:00
Jason Parker
ac941589fe Merged revisions 110780 via svnmerge from
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r110780 | qwell | 2008-03-25 17:51:55 -0500 (Tue, 25 Mar 2008) | 14 lines

Merged revisions 110779 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110779 | qwell | 2008-03-25 17:51:17 -0500 (Tue, 25 Mar 2008) | 6 lines

Make file access in cdr_custom similar to cdr_csv.

Fixes issue #12268.

Patch borrowed from r82344

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2008-03-25 22:52:24 +00:00
Jeff Peeler
e9f4445378 This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one.
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2008-03-25 22:11:03 +00:00
Tilghman Lesher
362eda6282 Merged revisions 110691 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r110691 | tilghman | 2008-03-25 12:46:34 -0500 (Tue, 25 Mar 2008) | 6 lines

Update sample configurations to make virtual hosting more obvious.
(closes issue #11969)
 Reported by: pprindeville
 Patches: 
       acme-virtualpbx.1.6.patch uploaded by pprindeville (license 347)

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Tilghman Lesher
46d5e9c5c9 Merged revisions 110689 via svnmerge from
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r110689 | tilghman | 2008-03-25 12:40:28 -0500 (Tue, 25 Mar 2008) | 6 lines

Update the sample configuration, to use Macro less (since it's now deprecated).
(closes issue #12293)
 Reported by: pprindeville
 Patches: 
       bugid-0012293.1.6.patch uploaded by pprindeville (license 347)

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2008-03-25 17:41:20 +00:00
Mark Michelson
913e19eb60 Oops.
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2008-03-25 15:43:24 +00:00
Mark Michelson
6f5612a7b0 Merged revisions 110636 via svnmerge from
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r110636 | mmichelson | 2008-03-25 10:41:33 -0500 (Tue, 25 Mar 2008) | 15 lines

Merged revisions 110635 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110635 | mmichelson | 2008-03-25 10:40:33 -0500 (Tue, 25 Mar 2008) | 7 lines

When reverting a commit, I accidentally left in this bit which was an experiment
to see what would happen. It passed the compile test, and I didn't notice I had
left this change in too.

So this is a revert of a revert...sort of.


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Joshua Colp
ffde763e29 Blocked revisions 110625 via svnmerge
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r110625 | oej | 2008-03-25 07:54:07 -0300 (Tue, 25 Mar 2008) | 6 lines

Use the "Server" header when responding to SIP requests.
(closes issue #12278)
Reported by: rjain
Patches: 
      chan_sip.c.diff uploaded by rjain (license 226)

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Joshua Colp
deb2cfeb5f Blocked revisions 110444 via svnmerge
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r110444 | tilghman | 2008-03-20 22:44:38 -0300 (Thu, 20 Mar 2008) | 2 lines

Add note of the added Directory options, from commit 110237 (closes issue #7151)

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Joshua Colp
9abb24214c Blocked revisions 110631 via svnmerge
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r110631 | file | 2008-03-25 12:18:41 -0300 (Tue, 25 Mar 2008) | 4 lines

Add a special dialplan variable to chan_sip which will cause an audio file to be played upon completion of an attended transfer.
(closes issue #9239)
Reported by: sunder

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Joshua Colp
be093c0508 Merged revisions 110629 via svnmerge from
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r110629 | file | 2008-03-25 11:39:45 -0300 (Tue, 25 Mar 2008) | 12 lines

Merged revisions 110628 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 lines

Add an option (transmit_silence) which transmits silence during both Record() and DTMF generation. The reason this is an option is that in order to transmit silence we have to setup a translation path. This may not be needed/wanted in all cases.
(closes issue #10058)
Reported by: tracinet

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Mark Michelson
4b6fe33940 Blocked revisions 110621 via svnmerge
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r110621 | mmichelson | 2008-03-24 15:14:07 -0500 (Mon, 24 Mar 2008) | 11 lines

Remove the "Event: registration" header from Asterisk-generated
SIP REGISTER requests. rjain points out that RFC 3265 specifies
that the Event: header is not a valid header for REGISTER requests
and that the "registration" value is not defined at IANA.

(closes issue #12279)
Reported by: rjain
Patches:
      chan_sip.c.diff uploaded by rjain (license 226)


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2008-03-24 20:14:28 +00:00
Mark Michelson
1a433add7a Merged revisions 110619 via svnmerge from
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r110619 | mmichelson | 2008-03-24 14:19:37 -0500 (Mon, 24 Mar 2008) | 23 lines

Merged revisions 110618 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110618 | mmichelson | 2008-03-24 14:17:41 -0500 (Mon, 24 Mar 2008) | 15 lines

This is a revert for revision 108288. The reason is that that revision
was not for an actual bug fix per se, and so it really should not have been in 1.4 in
the first place. Plus, people who compile with DO_CRASH are more likely
to encounter a crash due to this change. While I think the usage of DO_CRASH
in ast_sched_del is a bit absurd, this sort of change is beyond the scope of 1.4
and should be done instead in a developer branch based on trunk 
so that all scheduler functions are fixed at once.

I also am reverting the change to trunk and 1.6 since they also suffer from
the DO_CRASH potential.

(closes issue #12272)
Reported by: qq12345


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2008-03-24 19:20:30 +00:00
Russell Bryant
1ed7dc48f3 Merged revisions 110615 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r110615 | russell | 2008-03-24 12:36:04 -0500 (Mon, 24 Mar 2008) | 10 lines

Merged revisions 110614 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110614 | russell | 2008-03-24 12:34:56 -0500 (Mon, 24 Mar 2008) | 2 lines

Turn a NOTICE into a DEBUG message.

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2008-03-24 17:36:23 +00:00
Joshua Colp
00bf34e216 Merged revisions 110610 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r110610 | file | 2008-03-24 12:28:25 -0300 (Mon, 24 Mar 2008) | 6 lines

Only print out the set_address_from_contact host verbose message if debugging is enabled on the dialog.
(closes issue #12280)
Reported by: rjain
Patches:
      chan_sip.c.diff uploaded by rjain (license 226)

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2008-03-24 15:29:38 +00:00
Jason Parker
5519fd72b9 Merged revisions 110578 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r110578 | qwell | 2008-03-21 16:52:06 -0500 (Fri, 21 Mar 2008) | 1 line

Update to 1.4.11 core sounds.
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2008-03-21 21:52:47 +00:00
Joshua Colp
dbd1692779 Blocked revisions 110542 via svnmerge
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r110542 | file | 2008-03-21 14:58:59 -0300 (Fri, 21 Mar 2008) | 2 lines

Merge over ast_audiohook_volume branch. This adds API calls for use by developers to adjust the volume on a channel.

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2008-03-21 18:00:06 +00:00
Russell Bryant
6aa9c04ee9 Merged revisions 110499 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r110499 | russell | 2008-03-21 10:24:43 -0500 (Fri, 21 Mar 2008) | 3 lines

Note that the TCP and TLS support is currently considered experimental and
is subject to change while we work out the remaining issues.

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2008-03-21 15:25:06 +00:00
Jason Parker
57d3a27f1b Merged revisions 110475 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r110475 | qwell | 2008-03-21 09:36:17 -0500 (Fri, 21 Mar 2008) | 15 lines

Merged revisions 110474 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110474 | qwell | 2008-03-21 09:32:52 -0500 (Fri, 21 Mar 2008) | 7 lines

Don't attempt to do optimizations of gsm on mips platforms either.

(closes issue #12270)
Reported by: zandbelt
Patches:
      026-gsm-mips.patch uploaded by zandbelt (license 33)

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2008-03-21 14:36:43 +00:00
Russell Bryant
4cc349589c Merged revisions 110396 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r110396 | russell | 2008-03-20 18:14:13 -0500 (Thu, 20 Mar 2008) | 17 lines

Merged revisions 110395 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110395 | russell | 2008-03-20 18:13:56 -0500 (Thu, 20 Mar 2008) | 9 lines

Shorten the ast_waitfor() timeout from 500 ms to 50 ms in the autoservice thread.
This really should not make a difference except in very rare cases.  That case would
be that all of the channels in autoservice are not generating any frames.  In that
case, this change reduces the potential amount of time that a thread waits in
ast_autoservice_stop() for the autoservice thread to wrap back around to the beginning
of its loop.

(closes issue #12266, reported by dimas)

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2008-03-20 23:14:26 +00:00
Russell Bryant
2adbae37a8 Merged revisions 110339 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r110339 | russell | 2008-03-20 17:02:20 -0500 (Thu, 20 Mar 2008) | 3 lines

Use the correct buffer for g722tolin16_sample.  This shouldn't have caused any
problems, but Qwell noticed the typo here.

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2008-03-20 22:02:46 +00:00
Russell Bryant
573e35bba1 Merged revisions 110337 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r110337 | russell | 2008-03-20 16:55:50 -0500 (Thu, 20 Mar 2008) | 22 lines

Merged revisions 110336 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110336 | russell | 2008-03-20 16:54:58 -0500 (Thu, 20 Mar 2008) | 14 lines

Merged revisions 110335 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r110335 | russell | 2008-03-20 16:53:27 -0500 (Thu, 20 Mar 2008) | 6 lines

Fix some very broken code that was introduced in 1.2.26 as a part of the security
fix.  The dnsmgr is not appropriate here.  The dnsmgr takes a pointer to an address
structure that a background thread continuously updates.  However, in these cases,
a stack variable was passed.  That means that the dnsmgr thread would be continuously
writing to bogus memory.

........

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2008-03-20 21:56:22 +00:00
Russell Bryant
5f3c8c12e6 Merged revisions 110303 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r110303 | russell | 2008-03-20 15:08:26 -0500 (Thu, 20 Mar 2008) | 8 lines

Fix a bug when using zaptel timing for playing back files that have a sample rate
other than 8 kHz.  The issue here is that format modules give a "whennext" sample
value, which is used to calculate when to set a timer for to retrieve the next
frame.  However, the zaptel timer operates on 8 kHz samples, so this must be taken
into account.

(another part of issue #12164, reported by milazzo and jsmith, patch by me)

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2008-03-20 20:10:27 +00:00
Mark Michelson
f56d9c2483 Merged revisions 110272 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r110272 | mmichelson | 2008-03-20 13:01:36 -0500 (Thu, 20 Mar 2008) | 3 lines

Add missing unlock


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2008-03-20 18:02:42 +00:00
Russell Bryant
951a25d277 Blocked revisions 110270 via svnmerge
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r110270 | russell | 2008-03-20 12:45:29 -0500 (Thu, 20 Mar 2008) | 2 lines

Remove astobj.h from some places where it wasn't needed

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2008-03-20 17:45:57 +00:00
Russell Bryant
30f7c61f66 Merged revisions 110268 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r110268 | russell | 2008-03-20 12:41:22 -0500 (Thu, 20 Mar 2008) | 27 lines

Add some fixes that I made in regards to wideband codec handling to get
G.722 music on hold working for me.

(issue #12164, reported by milazzo and jsmith, patches by me)

res/res_musiconhold.c:
 - I moved a single line so that the sample queue update happened before
   ast_write().  The reason that this was a bug is that the G.722 frame
   originally says it has 320 samples in it (which is correct).  However,
   when the frame is written to a channel that uses RTP, main/rtp.c modifies
   the frame to cut the number of samples in half before it sends it on
   the wire.  This is to account for the stupid incorrect G.722 spec that
   makes it so we have to lie about the number of samples with RTP.  I should
   probably go and re-work the RTP code so it doesn't modify the frame so
   that a bug like this won't happen in the future.  However, this change to
   MOH is harmless.

main/channel.c:
 - I made two fixes in regards to generator timing.  Generators use samples
   for timing.  However, this code assumed 8 kHz samples.  In one case, it was
   a hard coded 160 samples, that is now written as the sample rate / 50.  The
   other place was dealing with timing a generator based on frames coming from
   the other direction.  However, that would have only worked if the sample
   rates for the formats in both directions were the same.  The code now takes
   into account that the sample rates may differ, and scales the generator
   samples accordingly.

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2008-03-20 17:41:45 +00:00
Tilghman Lesher
386038d8d0 Blocked revisions 110237 via svnmerge
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r110237 | tilghman | 2008-03-20 00:06:12 -0500 (Thu, 20 Mar 2008) | 2 lines

Upgrade the sounds version, and add several directory enhancements

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2008-03-20 05:45:34 +00:00
Tilghman Lesher
1bd1dddfb6 Blocked revisions 110211 via svnmerge
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r110211 | tilghman | 2008-03-19 22:14:59 -0500 (Wed, 19 Mar 2008) | 2 lines

Fix recent trunk breakage

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2008-03-20 03:19:49 +00:00
Russell Bryant
631b48efba Merged revisions 110164 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r110164 | russell | 2008-03-19 17:58:33 -0500 (Wed, 19 Mar 2008) | 13 lines

Merged revisions 110163 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110163 | russell | 2008-03-19 17:57:59 -0500 (Wed, 19 Mar 2008) | 5 lines

Fix a bug where when calls on the trunk side hang up while on hold, the state
is not properly reflected.

(closes issue #11990, reported by anakaoka, patched by me)

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2008-03-19 23:00:23 +00:00
Jason Parker
e740375bb7 Blocked revisions 110161 via svnmerge
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r110161 | qwell | 2008-03-19 17:25:34 -0500 (Wed, 19 Mar 2008) | 5 lines

Rename DSP_FEATURE_DTMF_DETECT, because we are *NOT* only detecting DTMF digits.
This was very misleading.

Early cleanup for issue #11968

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2008-03-19 22:27:44 +00:00
Jason Parker
780688bc19 Blocked revisions 110132 via svnmerge
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r110132 | qwell | 2008-03-19 16:56:15 -0500 (Wed, 19 Mar 2008) | 1 line

Rename very poorly named function to reflect what it actually does.  This was causing quite a bit of confusion for me...
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2008-03-19 21:57:58 +00:00
Jeff Peeler
851dde73ac marking rev 110087 from trunk as not applying
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@110088 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-19 21:06:46 +00:00
Mark Michelson
a2a08286fc Merged revisions 110084 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r110084 | mmichelson | 2008-03-19 15:34:13 -0500 (Wed, 19 Mar 2008) | 12 lines

Merged revisions 110083 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110083 | mmichelson | 2008-03-19 15:33:03 -0500 (Wed, 19 Mar 2008) | 4 lines

Add a missing unlock in the case that memory allocation fails in app_chanspy.
Thanks to Russell for confirming that this was an issue.


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2008-03-19 20:37:18 +00:00
Joshua Colp
38a75f1b82 Merged revisions 110036 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r110036 | file | 2008-03-19 16:13:39 -0300 (Wed, 19 Mar 2008) | 12 lines

Merged revisions 110035 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110035 | file | 2008-03-19 16:11:33 -0300 (Wed, 19 Mar 2008) | 4 lines

Add sanity checking for position resuming. We *have* to make sure that the position does not exceed the total number of files present, and we have to make sure that the position's filename is the same as previous. These values can change if a music class is reloaded and give unpredictable behavior.
(closes issue #11663)
Reported by: junky

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2008-03-19 19:14:56 +00:00
Russell Bryant
02471f21dd Blocked revisions 110023 via svnmerge
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r110023 | russell | 2008-03-19 13:57:16 -0500 (Wed, 19 Mar 2008) | 2 lines

remove svnmerge-blocked property that is not supposed to be here

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2008-03-19 19:00:29 +00:00
Russell Bryant
06f978bd58 Merged revisions 109974 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r109974 | qwell | 2008-03-19 12:15:14 -0500 (Wed, 19 Mar 2008) | 13 lines

Merged revisions 109973 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r109973 | qwell | 2008-03-19 12:12:52 -0500 (Wed, 19 Mar 2008) | 5 lines

People report bugs about Asterisk crashing with DO_CRASH enabled was getting a little silly...

Now we only show certain cflags when you run configure with --enable-dev-mode
(corresponding menuselect change to follow)

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2008-03-19 19:00:11 +00:00
Russell Bryant
17710ede58 Blocked revisions 109926 via svnmerge
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r109926 | kpfleming | 2008-03-19 11:21:36 -0500 (Wed, 19 Mar 2008) | 2 lines

ensure that res_phoneprov's HTTP handler tells the dispatcher what method it can handle

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2008-03-19 18:59:48 +00:00
Russell Bryant
28beb7c71f Blocked revisions 109912 via svnmerge
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r109912 | kpfleming | 2008-03-19 11:18:29 -0500 (Wed, 19 Mar 2008) | 2 lines

actually implement HTTP request dispatching based on both URI and method; reduce duplication of data when generating responses using ast_http_error()

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2008-03-19 18:59:32 +00:00
Russell Bryant
a4ef06e955 Blocked revisions 109910 via svnmerge
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r109910 | russell | 2008-03-19 10:45:49 -0500 (Wed, 19 Mar 2008) | 2 lines

Fix some more breakage that I introduced when changing extension state callbacks to the list macros.

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2008-03-19 18:59:17 +00:00
Russell Bryant
c0d64c79c3 Blocked revisions 109909 via svnmerge
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r109909 | kpfleming | 2008-03-19 10:41:54 -0500 (Wed, 19 Mar 2008) | 2 lines

clean up code to conform to coding guidelines

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2008-03-19 18:59:00 +00:00
Russell Bryant
2db2038706 Blocked revisions 109907 via svnmerge
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r109907 | russell | 2008-03-19 10:22:13 -0500 (Wed, 19 Mar 2008) | 3 lines

Remove an unneeded variable.  This compiled, but I missed the uninitialized warning
because I always compile without optimizations turned on.  Sorry!

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2008-03-19 18:58:43 +00:00
Russell Bryant
0183961a5a Blocked revisions 109883 via svnmerge
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r109883 | russell | 2008-03-18 23:32:13 -0500 (Tue, 18 Mar 2008) | 2 lines

Convert handling of extension state callbacks to the list macros.

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2008-03-19 18:58:28 +00:00
Russell Bryant
7caac77e86 Blocked revisions 109841-109842 via svnmerge
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r109841 | russell | 2008-03-18 23:09:55 -0500 (Tue, 18 Mar 2008) | 2 lines

Minor change to use Asterisk macros

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r109842 | russell | 2008-03-18 23:14:12 -0500 (Tue, 18 Mar 2008) | 2 lines

Minor coding style changes, including adding handling for memory allocation failure

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2008-03-19 18:58:08 +00:00
Joshua Colp
aaf6b67e5c Merged revisions 110020 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r110020 | file | 2008-03-19 15:25:33 -0300 (Wed, 19 Mar 2008) | 14 lines

Merged revisions 110019 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110019 | file | 2008-03-19 15:20:28 -0300 (Wed, 19 Mar 2008) | 6 lines

Make sure that the mark bit does not incorrectly cause video frame timestamps to be calculated as if they are audio frames.
(closes issue #11429)
Reported by: sperreault
Patches:
      11429-frametype.diff uploaded by qwell (license 4)

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2008-03-19 18:26:24 +00:00
Joshua Colp
0a31c19fd7 Blocked revisions 109970 via svnmerge
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r109970 | file | 2008-03-19 13:54:12 -0300 (Wed, 19 Mar 2008) | 7 lines

Add the ability to use a pattern match for a hint.
(closes issue #7767)
Reported by: Corydon76
Patches:
      20070314__simple_hint_lookup.diff.txt uploaded by Corydon76
      pbx-trunk-98436.diff uploaded by plack (license 365)

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2008-03-19 16:55:00 +00:00
Steve Murphy
301b8b156a Merged revisions 109942 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r109942 | murf | 2008-03-19 10:24:51 -0600 (Wed, 19 Mar 2008) | 80 lines

Merged revisions 109908 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r109908 | murf | 2008-03-19 09:41:13 -0600 (Wed, 19 Mar 2008) | 72 lines

(closes issue #11442)
Reported by: tzafrir
Patches:
      11442.patch uploaded by murf (license 17)
Tested by: murf

I didn't give tzafrir very much time to test this, but if he does 
still have remaining issues, he is welcome to 
re-open this bug, and we'll do what is called for.

I reproduced the problem, and tested the fix, so I hope I
am not jumping by just going ahead and committing the fix.

The problem was with what file_save does with templates; 
firstly, it tended to print out multiple options:

[my_category](!)(templateref)

instead of 

[my_category](!,templateref)

which is fixed by this patch.


Nextly, the code to suppress output of duplicate declarations
that would occur because the reader copies inherited declarations
down the hierarchy, was not working. Thus:


 [master-template](!)
 mastervar = bar


 [template](!,master-template)
 tvar = value


 [cat](template)
 catvar = val


would be rewritten as:

 ;!
 ;! Automatically generated configuration file
 ;! Filename: experiment.conf (/etc/asterisk/experiment.conf)
 ;! Generator: Manager
 ;! Creation Date: Tue Mar 18 23:17:46 2008
 ;!
 
 [master-template](!)
 mastervar = bar

 
 [template](!,master-template)
 mastervar = bar
 tvar = value

 
 [cat](template)
 mastervar = bar
 tvar = value
 catvar = val

This has been fixed. Since the config reader 'explodes' inherited
vars into the category, users may, in certain circumstances, see
output different from what they originally entered, but it should
be both correct and equivalent.



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2008-03-19 16:46:12 +00:00
Russell Bryant
cf0f0d5971 Merged revisions 109839 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r109839 | russell | 2008-03-18 23:06:31 -0500 (Tue, 18 Mar 2008) | 10 lines

Merged revisions 109838 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r109838 | russell | 2008-03-18 23:06:05 -0500 (Tue, 18 Mar 2008) | 2 lines

Tweak spacing in a recent change because I'm very picky.

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2008-03-19 04:06:56 +00:00
Russell Bryant
8da200be71 Merged revisions 109764 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r109764 | russell | 2008-03-18 17:36:02 -0500 (Tue, 18 Mar 2008) | 11 lines

Merged revisions 109763 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r109763 | russell | 2008-03-18 17:34:42 -0500 (Tue, 18 Mar 2008) | 3 lines

Fix one place where the chanspy datastore isn't removed from a channel.
(issue #12243, reported by atis, patch by me)

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2008-03-19 04:04:13 +00:00
Russell Bryant
5b528ad229 Blocked revisions 109762 via svnmerge
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r109762 | kpfleming | 2008-03-18 17:32:26 -0500 (Tue, 18 Mar 2008) | 2 lines

start the process of changing HTTP request dispatching to do it based on *both* URI and method, so that POST support can move into a module; move http.c's private function prototypes into _private.h

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2008-03-19 04:03:37 +00:00
Russell Bryant
c7b86f005b Blocked revisions 109680 via svnmerge
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r109680 | tilghman | 2008-03-18 14:53:02 -0500 (Tue, 18 Mar 2008) | 2 lines

Comment debug

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2008-03-19 04:02:48 +00:00
Russell Bryant
84cc3e6449 Blocked revisions 109833 via svnmerge
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r109833 | russell | 2008-03-18 22:51:06 -0500 (Tue, 18 Mar 2008) | 3 lines

Set req->data to NULL after free'ing to ensure that it never gets accidentally
double free'd.  (reported by dhubbard directly to me)

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2008-03-19 04:02:22 +00:00
Mark Michelson
4783a79cce Blocked revisions 109802 via svnmerge
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r109802 | mmichelson | 2008-03-18 18:32:58 -0500 (Tue, 18 Mar 2008) | 5 lines

Fix a typo which caused a double free in chan_zap. This was discovered
by Juggie while attempting to load chan_zap. Apparently this would happen
if an error were encountered while trying to process zapata.conf.


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2008-03-18 23:35:53 +00:00
Tilghman Lesher
276e3411c6 Merged revisions 109775 via svnmerge from
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r109775 | tilghman | 2008-03-18 18:22:25 -0500 (Tue, 18 Mar 2008) | 3 lines

Change back to using ldap_initialize() and let the user specify a URL directly,
instead of trying to piece it together, badly.

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2008-03-18 23:23:37 +00:00
Mark Michelson
3431f1483a Merged revisions 109714 via svnmerge from
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r109714 | mmichelson | 2008-03-18 15:59:02 -0500 (Tue, 18 Mar 2008) | 20 lines

Merged revisions 109713 via svnmerge from 
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r109713 | mmichelson | 2008-03-18 15:52:15 -0500 (Tue, 18 Mar 2008) | 12 lines

This patch makes it so that all queue member status changes are handled through device state
code. This removes several problems people were seeing where their queue members would get into
an "unknown" state. Huge props go to atis on this one since he was the one who found the code
section that was causing the problem and proposed the solution. I just wrote what he suggested :)

(closes issue #12127)
Reported by: atis
Patches:
      12127v3.patch uploaded by putnopvut (license 60)
Tested by: atis, jvandal


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2008-03-18 21:03:52 +00:00
Tilghman Lesher
39b162b00c Merged revisions 109683 via svnmerge from
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r109683 | tilghman | 2008-03-18 15:13:40 -0500 (Tue, 18 Mar 2008) | 4 lines

Set protocol version, port number correctly.
(closes issue #12211, closes issue #12209)
 Reported by: sylvain

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2008-03-18 20:14:44 +00:00
Mark Michelson
0aa3619019 Blocked revisions 109681 via svnmerge
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r109681 | mmichelson | 2008-03-18 15:02:26 -0500 (Tue, 18 Mar 2008) | 7 lines

Since a sip request's data field is now a stringfield, we not only have to check
if the string is zero-length, but also if the data field is non-null.

(closes issue #12250)
Reported by: caio1982


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2008-03-18 20:03:04 +00:00
Jason Parker
4f2ec37cbd Merged revisions 109651 via svnmerge from
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r109651 | qwell | 2008-03-18 14:24:15 -0500 (Tue, 18 Mar 2008) | 15 lines

Merged revisions 109648 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r109648 | qwell | 2008-03-18 14:23:44 -0500 (Tue, 18 Mar 2008) | 7 lines

Allow codecs that use log2comp (g726) to compile correctly on x86 with gcc4 optimizations.

(closes issue #12253)
Reported by: fossil
Patches:
      log2comp.patch uploaded by fossil (license 140)

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2008-03-18 19:24:45 +00:00
Mark Michelson
81019b3690 Blocked revisions 109621 via svnmerge
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r109621 | mmichelson | 2008-03-18 13:58:42 -0500 (Tue, 18 Mar 2008) | 9 lines

Add option 'randomperiodicannounce' to queues.conf. Setting this will
allow the list of periodic announcments specified to be played in a random
order instead of being played sequentially.

(closes issue #6681)
Reported by: alt_phil
Tested by: putnopvut


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2008-03-18 19:00:30 +00:00
Mark Michelson
7783cbdafa Merged revisions 109576 via svnmerge from
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r109576 | mmichelson | 2008-03-18 12:59:18 -0500 (Tue, 18 Mar 2008) | 14 lines

Merged revisions 109575 via svnmerge from 
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r109575 | mmichelson | 2008-03-18 12:58:11 -0500 (Tue, 18 Mar 2008) | 6 lines

Make sure an agent doesn't try to send dtmf to a NULL channel

closes issue #12242
Reported by Yourname


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2008-03-18 17:59:49 +00:00
Mark Michelson
2bac84986e Merged revisions 109545 via svnmerge from
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r109545 | mmichelson | 2008-03-18 12:00:53 -0500 (Tue, 18 Mar 2008) | 3 lines

Add format attribute to printf-style functions in astmm.h


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2008-03-18 17:01:36 +00:00
Kevin P. Fleming
26ea819943 Merged revisions 109475 via svnmerge from
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r109475 | kpfleming | 2008-03-18 11:23:05 -0500 (Tue, 18 Mar 2008) | 2 lines

fix up various warnings found via the addition of format string checking... some of these were really, really bad code

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2008-03-18 16:26:59 +00:00
Russell Bryant
4af367c3a9 Merged revisions 109447 via svnmerge from
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r109447 | twilson | 2008-03-18 10:43:34 -0500 (Tue, 18 Mar 2008) | 3 lines

Go through and fix a bunch of places where character strings were being interpreted as format strings. Most of these changes are solely to make compiling with -Wsecurity and -Wformat=2 happy, and were not
actual problems, per se.  I also added format attributes to any printf wrapper functions I found that didn't have them.  -Wsecurity and -Wmissing-format-attribute added to --enable-dev-mode.

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2008-03-18 15:58:50 +00:00
Russell Bryant
1df02c7c33 Blocked revisions 109394 via svnmerge
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r109394 | qwell | 2008-03-18 10:10:45 -0500 (Tue, 18 Mar 2008) | 1 line

Block this here.  Already committed.
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2008-03-18 15:55:33 +00:00
Russell Bryant
6192c7a750 Blocked revisions 109316 via svnmerge
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r109316 | oej | 2008-03-18 02:23:45 -0500 (Tue, 18 Mar 2008) | 7 lines

Add manager peerstatus events when peer can't authenticate.

(closes issue #11959)
Reported by: mostyn
Patches: 
      peerstatus3.patch uploaded by mostyn (license 398)

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2008-03-18 15:55:12 +00:00
Russell Bryant
dd7e5d925d Blocked revisions 109172 via svnmerge
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r109172 | mvanbaak | 2008-03-17 12:58:34 -0500 (Mon, 17 Mar 2008) | 3 lines

block rev 109171 that is already here


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2008-03-18 15:54:36 +00:00
Russell Bryant
e871d50c63 Blocked revisions 109169 via svnmerge
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r109169 | murf | 2008-03-17 12:47:36 -0500 (Mon, 17 Mar 2008) | 20 lines

(closes issue #12238)
Reported by: mvanbaak
Tested by: murf, mvanbaak

Due to a bug that occurred when merge_contexts_and_delete scanned the "old" or existing contexts, and found a context
that doesn't exist in the new set, yet owned by a different registrar. The context is created in the new set, with the
old registrar, and and all the priorities and extens that have a different registrar are copied into it. But, not the
includes, ignorepats, and switches. I added code to do this immediately after the context is created.

This still leaves a logical hole in the code. If you define a context in two places, (eg. in extensions.conf and also 
in extensions.ael), and they both have includes, but different in composition, no new context will be generated, and
therefore the 'old' includes, switches, and ignorepats will not be copied. I'd have added code to simply add any non-duplicates
into the 'new' context that had a different registrar, but there is one big complication: includes, and switches are definitely
order dependent. (ignorepats I'm not sure about). And we'll have to develop some sort of policy about how we 
merge order dependent lists, especially if the intersection of the two sets is empty. (in other words, they do not have any
elements in common). Do the new go first, or the old? I've elected to punt this issue until a user complains. Hopefully,
this is pretty rare thing.



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2008-03-18 15:53:20 +00:00
Russell Bryant
f4d6e28823 Merged revisions 109111 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r109111 | qwell | 2008-03-17 11:37:31 -0500 (Mon, 17 Mar 2008) | 10 lines

Add sample events for aastra phones.

aastra-check-cfg is the same as the other check-cfg entries,
 and aastra-xml is to load a pre-configured xml script.

(closes issue #12229)
Reported by: gowen72
Patches:
      aastra.patch uploaded by gowen72 (license 432)

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2008-03-18 15:52:41 +00:00
Kevin P. Fleming
a076cb58fa Merged revisions 109451 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r109451 | kpfleming | 2008-03-18 10:50:29 -0500 (Tue, 18 Mar 2008) | 2 lines

ensure that dependencies on AST_C_DEFINE_CHECK symbols work properly

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2008-03-18 15:50:53 +00:00
Russell Bryant
b56c938c91 Blocked revisions 109058 via svnmerge
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r109058 | qwell | 2008-03-17 10:16:41 -0500 (Mon, 17 Mar 2008) | 11 lines

Blocked revisions 109057 via svnmerge

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r109057 | qwell | 2008-03-17 10:15:56 -0500 (Mon, 17 Mar 2008) | 4 lines

Backport revision 106439 from trunk.  I didn't realize this was broken in 1.4 as well.

Closes issue #12222.

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2008-03-18 15:50:35 +00:00
Russell Bryant
3986cfc196 Blocked revisions 109054 via svnmerge
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r109054 | file | 2008-03-17 09:37:40 -0500 (Mon, 17 Mar 2008) | 4 lines

Make sure that the temporary sip_request structure is empty so that copy_request doesn't think it already has an ast_str.
(closes issue #12231)
Reported by: IgorG

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2008-03-18 15:49:56 +00:00
Russell Bryant
af39c6e896 Merged revisions 108962 via svnmerge from
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r108962 | mvanbaak | 2008-03-16 16:50:58 -0500 (Sun, 16 Mar 2008) | 15 lines

Merged revisions 108961 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r108961 | mvanbaak | 2008-03-16 22:47:10 +0100 (Sun, 16 Mar 2008) | 7 lines

add missing break to case AST_CONTROL_SRCUPDATE

(closes issue #12228)
Reported by: andrew
Patches:
      SRC.patch uploaded by andrew (license 240)

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2008-03-18 15:49:23 +00:00
Russell Bryant
0dd2065118 Blocked revisions 108857 via svnmerge
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r108857 | russell | 2008-03-14 16:20:36 -0500 (Fri, 14 Mar 2008) | 2 lines

Doxygenify dnsmgr.h

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2008-03-18 15:48:56 +00:00
Joshua Colp
8e7ad98e63 Merged revisions 109396 via svnmerge from
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r109396 | file | 2008-03-18 12:13:07 -0300 (Tue, 18 Mar 2008) | 3 lines

Make sure values are interpreted as character strings and not format strings.
(AST-2008-004)

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2008-03-18 15:16:14 +00:00
Steve Murphy
6c23fca72a Merged revisions 109357 via svnmerge from
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r109357 | murf | 2008-03-18 08:09:50 -0600 (Tue, 18 Mar 2008) | 25 lines

Merged revisions 109309 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r109309 | murf | 2008-03-18 00:37:15 -0600 (Tue, 18 Mar 2008) | 17 lines

(closes issue #11903)
Reported by: atis

Many thanks to atis for spotting this problem and reporting it.
The fix was to straighten out how items are placed on and removed
from the file stack. Regressions as well as the provided test case
helped to straighten out all code paths. valgrind was used to make
sure all memory allocated was freed.

Sorry for not solving this earlier. I got distracted.

Added the ntest23 regression test, which is mainly a copy of ntest22, 
but with a few juicy errors thrown in, to replicate the kind of 
error that atis spotted.



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2008-03-18 15:14:22 +00:00
Jason Parker
5996a61f4b Merged revisions 109389 via svnmerge from
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r109389 | qwell | 2008-03-18 10:07:04 -0500 (Tue, 18 Mar 2008) | 3 lines

Do not return with a successful authentication if the From header ends up empty.
(AST-2008-003)

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2008-03-18 15:11:25 +00:00
Joshua Colp
627d3f04d4 Merged revisions 109390 via svnmerge from
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r109390 | file | 2008-03-18 12:08:09 -0300 (Tue, 18 Mar 2008) | 11 lines

Merged revisions 109386 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r109386 | file | 2008-03-18 11:58:39 -0300 (Tue, 18 Mar 2008) | 3 lines

Put a maximum limit on the number of payloads accepted, and also make sure a given payload does not exceed our maximum value.
(AST-2008-002)

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2008-03-18 15:09:39 +00:00
Sean Bright
cc4c5f1a6d Merged revisions 109282 via svnmerge from
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r109282 | seanbright | 2008-03-17 20:28:39 -0400 (Mon, 17 Mar 2008) | 1 line

Fix a typo
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2008-03-18 00:40:32 +00:00
Terry Wilson
9a4405a7e5 Merged revisions 109229 via svnmerge from
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r109229 | twilson | 2008-03-17 17:10:06 -0500 (Mon, 17 Mar 2008) | 5 lines

Replace minimime with superior GMime library so that the entire contents of an http post are not read into memory.
This does introduce a dependency on the GMime library for handling HTTP POSTs, but it is available in most distros.

If the library is present, then the compile flag for ENABLE_UPLOADS is enabled by default in menuselect.

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2008-03-17 22:24:46 +00:00
Mark Michelson
cbdd59040c Merged revisions 109227 via svnmerge from
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r109227 | mmichelson | 2008-03-17 17:06:44 -0500 (Mon, 17 Mar 2008) | 20 lines

Merged revisions 109226 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r109226 | mmichelson | 2008-03-17 17:05:49 -0500 (Mon, 17 Mar 2008) | 12 lines

Fix a logic flaw in the code that stores lock info which is displayed
via the "core show locks" command. The idea behind this section of code was
to remove the previous lock from the list if it was a trylock that had failed.
Unfortunately, instead of checking the status of the previous lock, we were referencing
the index immediately following the previous lock in the lock_info->locks array. 
The result of this problem, under the right circumstances, was that the lock which 
we currently in the process of attempting to acquire could "overwrite" the previous lock 
which was acquired. While this does not in any way affect typical operation, it *could*
lead to misleading "core show locks" output.



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2008-03-17 22:07:29 +00:00
Michiel van Baak
f58d1ed4c3 Merged revisions 109168 via svnmerge from
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r109168 | mvanbaak | 2008-03-17 18:43:46 +0100 (Mon, 17 Mar 2008) | 11 lines

Update the directory of placed calls on skinny phones
when dialing a channel that does not provide progress (analog ZAP lines)

The phone does handle the double update on calls to channels that do
provide progress and wont insert duplicate items

(closes issue #12239)
Reported by: DEA
Patches:
      chan_skinny-call-log.txt uploaded by DEA (license 3)

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2008-03-17 18:11:19 +00:00
Kevin P. Fleming
cc5d6ba88d Merged revisions 109166 via svnmerge from
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r109166 | kpfleming | 2008-03-17 12:31:46 -0500 (Mon, 17 Mar 2008) | 3 lines

don't define Zaptel features as libraries, they aren't, and we don't want '--with-zaptel-<foo>' configure options for them
also some minor cleanups

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2008-03-17 17:42:20 +00:00
Joshua Colp
32ec8b7b15 Blocked revisions 109113 via svnmerge
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r109113 | file | 2008-03-17 13:47:02 -0300 (Mon, 17 Mar 2008) | 2 lines

Remove something that is never ever used.

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Joshua Colp
2d567f7434 Merged revisions 109108 via svnmerge from
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r109108 | file | 2008-03-17 13:26:36 -0300 (Mon, 17 Mar 2008) | 12 lines

Merged revisions 109107 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r109107 | file | 2008-03-17 13:24:29 -0300 (Mon, 17 Mar 2008) | 4 lines

200 OKs in response to a reinvite need to be sent reliably. If the remote side does not receive one the dialog will be torn down.
(closes issue #12208)
Reported by: atrash

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Mark Michelson
e59eb16daa Merged revisions 109024 via svnmerge from
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r109024 | mmichelson | 2008-03-17 09:21:14 -0500 (Mon, 17 Mar 2008) | 14 lines

Merged revisions 109012 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r109012 | mmichelson | 2008-03-17 09:18:26 -0500 (Mon, 17 Mar 2008) | 6 lines

Make sure that we release the lock on the spyee channel if the spyee or spy has hung up

(closes issue #12232)
Reported by: atis


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2008-03-17 14:21:48 +00:00
Russell Bryant
6b057dcbe9 Blocked revisions 108929 via svnmerge
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r108929 | russell | 2008-03-16 12:55:53 -0500 (Sun, 16 Mar 2008) | 2 lines

Remove an unnecessary thread attribute instance

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2008-03-16 17:56:11 +00:00
Russell Bryant
4b5da21ef2 Merged revisions 108927 via svnmerge from
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r108927 | russell | 2008-03-16 12:53:46 -0500 (Sun, 16 Mar 2008) | 7 lines

Fix polling for mailbox changes in mailboxes that are not in the default vm context.

(closes issue #12223)
Reported by: DEA
Patches: 
      vm-polled-imap.txt uploaded by DEA (license 3)

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2008-03-16 17:54:13 +00:00
Russell Bryant
7e74bd7557 Blocked revisions 108894 via svnmerge
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r108894 | russell | 2008-03-15 11:21:04 -0500 (Sat, 15 Mar 2008) | 8 lines

Remove a double write lock of the contexts lock in ast_wrlock_contexts().
How did this ever work?

(closes issue #12219)
Reported by: ys
Patches: 
      pbx.c.diff uploaded by ys (license 281)

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2008-03-15 16:21:50 +00:00
Russell Bryant
a36eb6597e Merged revisions 108799 via svnmerge from
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r108799 | russell | 2008-03-14 15:14:06 -0500 (Fri, 14 Mar 2008) | 8 lines

Make sure configure is run before menuselect on a clean checkout

(closes issue #12197)
Reported by: juggie
Patches:
      12197.diff uploaded by juggie (license 24)


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2008-03-14 20:14:46 +00:00
Russell Bryant
3120fe50ba Merged revisions 108797 via svnmerge from
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r108797 | russell | 2008-03-14 15:09:37 -0500 (Fri, 14 Mar 2008) | 13 lines

Merged revisions 108796 via svnmerge from 
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r108796 | russell | 2008-03-14 15:09:22 -0500 (Fri, 14 Mar 2008) | 5 lines

Fix a channel name issue.  chan_oss registers the "Console" channel type,
but it created channels with an "OSS" prefix.

(closes issue #12194, reported by davidw, patched by me)

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2008-03-14 20:09:53 +00:00
Russell Bryant
a5cc1f0552 Blocked revisions 108404,108439,108523,108639 via svnmerge
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r108404 | jpeeler | 2008-03-13 13:59:04 -0500 (Thu, 13 Mar 2008) | 6 lines

(closes issue #11827)
Reported by: ctooley
Patches:
      eivr_tcp_generic.patch uploaded by jpeeler (license 325)
This change adds the ability to communicate over a TCP socket instead of forking a child process.

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r108439 | russell | 2008-03-13 14:54:44 -0500 (Thu, 13 Mar 2008) | 8 lines

Merge changes from team/jamesgolovich/chan_sip-ast_str

This set of changes removes the hard coded maximum packet size of 4kB from chan_sip.
It now starts by allocating 1kB, and growing the buffer as needed to accommodate large
packets.

(closes issue #8556, reported by mikma, patch by jamesgolovich)

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r108523 | jpeeler | 2008-03-13 15:38:56 -0500 (Thu, 13 Mar 2008) | 1 line

set variable to NULL to prevent uninitialized warning
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r108639 | jpeeler | 2008-03-13 18:12:59 -0500 (Thu, 13 Mar 2008) | 1 line

documenting changes as a result of adding TCP functionality to ExternalIVR
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Russell Bryant
c1aa9978a0 Merged revisions 108793 via svnmerge from
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r108793 | russell | 2008-03-14 15:04:56 -0500 (Fri, 14 Mar 2008) | 12 lines

Merged revisions 108792 via svnmerge from 
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r108792 | russell | 2008-03-14 15:04:35 -0500 (Fri, 14 Mar 2008) | 4 lines

Update the SuSE init script to start networking before asterisk, as well.

(closes issue #12200, reported by and change suggested by reinerotto)

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Russell Bryant
220d0a17cc Merged revisions 108740 via svnmerge from
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r108740 | russell | 2008-03-14 12:05:11 -0500 (Fri, 14 Mar 2008) | 5 lines

Do a link test in AST_EXT_TOOL_CHECK() to ensure we have all the required libs
reported by the tool.

(closes issue #12067, reported by Juggie, patched by me)

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Mark Michelson
2c71d8fa63 Merged revisions 108738 via svnmerge from
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r108738 | mmichelson | 2008-03-14 11:52:51 -0500 (Fri, 14 Mar 2008) | 41 lines

Merged revisions 108737 via svnmerge from 
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r108737 | mmichelson | 2008-03-14 11:44:08 -0500 (Fri, 14 Mar 2008) | 33 lines

Fix a race condition in the SIP packet scheduler which could cause a crash.

chan_sip uses the scheduler API in order to schedule retransmission of reliable
packets (such as INVITES). If a retransmission of a packet is occurring, then the
packet is removed from the scheduler and retrans_pkt is called. Meanwhile, if
a response is received from the packet as previously transmitted, then when we 
ACK the response, we will remove the packet from the scheduler and free the packet.

The problem is that both the ACK function and retrans_pkt attempt to acquire the
same lock at the beginning of the function call. This means that if the ACK function
acquires the lock first, then it will free the packet which retrans_pkt is about to
read from and write to. The result is a crash.

The solution:

1. If the ACK function fails to remove the packet from the scheduler and the retransmit
   id of the packet is not -1 (meaning that we have not reached the maximum number of 
   retransmissions) then release the lock and yield so that retrans_pkt may acquire the
   lock and operate.

2. Make absolutely certain that the ACK function does not recursively lock the lock in
   question. If it does, then releasing the lock will do no good, since retrans_pkt will
   still be unable to acquire the lock.

(closes issue #12098)
Reported by: wegbert
(closes issue #12089)
Reported by: PTorres
Patches:
      12098-putnopvutv3.patch uploaded by putnopvut (license 60)
Tested by: jvandal


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2008-03-14 16:54:36 +00:00
Jason Parker
c6a0820063 Merged revisions 108683 via svnmerge from
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r108683 | qwell | 2008-03-14 09:32:55 -0500 (Fri, 14 Mar 2008) | 12 lines

Merged revisions 108682 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r108682 | qwell | 2008-03-14 09:29:05 -0500 (Fri, 14 Mar 2008) | 4 lines

Fix a potential segfault if chan (or chan->music_state) is NULL.

Closes issue #12210, credit to edantie for pointing this out.

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2008-03-14 14:33:33 +00:00
Mark Michelson
180bdba9e5 Merged revisions 108586 via svnmerge from
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r108586 | mmichelson | 2008-03-13 16:47:55 -0500 (Thu, 13 Mar 2008) | 3 lines

Make this compile


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2008-03-13 21:48:38 +00:00
Russell Bryant
e2d7bed99a Merged revisions 108584 via svnmerge from
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r108584 | russell | 2008-03-13 16:40:43 -0500 (Thu, 13 Mar 2008) | 19 lines

Merged revisions 108583 via svnmerge from 
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r108583 | russell | 2008-03-13 16:38:16 -0500 (Thu, 13 Mar 2008) | 11 lines

Fix another issue that was causing crashes in chanspy.  This introduces a new
datastore callback, called chan_fixup().  The concept is exactly like the
fixup callback that is used in the channel technology interface.  This callback
gets called when the owning channel changes due to a masquerade.  Before this
was introduced, if a masquerade happened on a channel being spyed on, the
channel pointer in the datastore became invalid.

(closes issue #12187)
(reported by, and lots of testing from atis)
(props to file for the help with ideas)

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2008-03-13 21:41:14 +00:00
Mark Michelson
27803e8d3e Merged revisions 108529 via svnmerge from
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r108529 | mmichelson | 2008-03-13 15:59:00 -0500 (Thu, 13 Mar 2008) | 11 lines

Fixing a potential buffer overflow in the manager command ModuleCheck.

Though this overflow is exploitable remotely, we are NOT issuing a security
advisory for this since in order to exploit the overflow, the attacker would
have to establish an authenticated manager session AND have the system privilege.
By gaining this privilege, the attacker already has more powerful weapons at his
disposal than overflowing a buffer with a malformed manager header, so the vulnerability
in this case really lies with the authentication method that allowed the attacker to 
gain the system privilege in the first place.


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Russell Bryant
bcdaa03501 Merged revisions 108531 via svnmerge from
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r108531 | russell | 2008-03-13 16:06:52 -0500 (Thu, 13 Mar 2008) | 18 lines

Merged revisions 108530 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r108530 | russell | 2008-03-13 16:06:33 -0500 (Thu, 13 Mar 2008) | 10 lines

Make a tweak that gets the LEDs on polycom phones to blink when an extension that
has been subscribed to goes on hold.  Otherwise, they just stay on like it does
when an extension is in use.

(closes issue #11263)
Reported by: russell
Patches:
      notify_hold.rev1.txt uploaded by russell (license 2)
Tested by: russell

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2008-03-13 21:07:14 +00:00
Russell Bryant
a75ec6f0ba Merged revisions 108508 via svnmerge from
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r108508 | russell | 2008-03-13 15:35:28 -0500 (Thu, 13 Mar 2008) | 2 lines

Fix a place where configuration values could cause an overflow of a buffer.

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Russell Bryant
42a70f149d Merged revisions 108472 via svnmerge from
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r108472 | russell | 2008-03-13 15:26:59 -0500 (Thu, 13 Mar 2008) | 12 lines

Merged revisions 108469 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r108469 | russell | 2008-03-13 15:26:28 -0500 (Thu, 13 Mar 2008) | 4 lines

Fix a couple uses of sprintf.  The second one could actually cause an overflow
of a stack buffer.  It's not a security issue though, it only depends on your
configuration.

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2008-03-13 20:27:24 +00:00
Russell Bryant
53f827fd60 Blocked revisions 107721,107787,107790,107928,108034 via svnmerge
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r107721 | tilghman | 2008-03-11 15:58:42 -0500 (Tue, 11 Mar 2008) | 2 lines

Add contributed script for separation of database access from Asterisk

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r107787 | russell | 2008-03-11 17:21:19 -0500 (Tue, 11 Mar 2008) | 6 lines

Add a trivial new dialplan function, AST_CONFIG(), which allows you to access
a variable from an Asterisk configuration file in the dialplan, or anywhere
else where dialplan functions can be used.

(Inspired by a discussion with Tilghman and Pari)

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r107790 | russell | 2008-03-11 17:54:25 -0500 (Tue, 11 Mar 2008) | 2 lines

remove documentation of an argument that i did not implement

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r107928 | tilghman | 2008-03-11 21:53:21 -0500 (Tue, 11 Mar 2008) | 2 lines

Cache config files, when possible, for speed

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r108034 | russell | 2008-03-12 12:06:37 -0500 (Wed, 12 Mar 2008) | 4 lines

 - Add Tilghman to the copyright info ... he wrote the hard part :)
 - Remove some magic in unload_module that isn't needed.  Module use counts
   already ensure that the function isn't going to be in use at this point.

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Russell Bryant
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r107719 | russell | 2008-03-11 15:54:32 -0500 (Tue, 11 Mar 2008) | 8 lines

This patch adds support for extended help prompts in voicemail.  These prompts
are in the 1.4.9 sounds release.

(closes issue #11705)
Reported by: jaroth
Patches:
      helpprompts.patch uploaded by jaroth (license 50)

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Russell Bryant
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r107551 | tilghman | 2008-03-11 11:28:03 -0500 (Tue, 11 Mar 2008) | 2 lines

Whitespace changes only

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Russell Bryant
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r107475 | kpfleming | 2008-03-11 10:19:05 -0500 (Tue, 11 Mar 2008) | 2 lines

block fix that is already here

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Russell Bryant
68ebba346c Merged revisions 107465 via svnmerge from
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r107465 | file | 2008-03-11 10:05:17 -0500 (Tue, 11 Mar 2008) | 4 lines

Clarify comment about masquerading and playback of the parking slot.
(closes issue #12180)
Reported by: davidw

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Russell Bryant
82922e116c Merged revisions 107157 via svnmerge from
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r107157 | file | 2008-03-10 15:00:21 -0500 (Mon, 10 Mar 2008) | 4 lines

If we receive a 488 on a T38 request reinvite back to audio. As well reinvite across a bridge back to audio if one side doesn't negotiate to T38.
(closes issue #8677)
Reported by: alex-911

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Russell Bryant
871cb71a98 Merged revisions 106892 via svnmerge from
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r106892 | mattf | 2008-03-07 16:36:49 -0600 (Fri, 07 Mar 2008) | 1 line

Make sure we don't start a call when we have already done so in response to a COT message
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Russell Bryant
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r106843 | qwell | 2008-03-07 16:15:20 -0600 (Fri, 07 Mar 2008) | 13 lines

Merged revisions 106842 via svnmerge from 
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r106842 | qwell | 2008-03-07 16:14:45 -0600 (Fri, 07 Mar 2008) | 5 lines

Fix hardcoded grep in editline, were GNU grep is required.

(closes issue #12124)
Reported by: dmartin

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Russell Bryant
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r106757 | murf | 2008-03-07 12:57:57 -0600 (Fri, 07 Mar 2008) | 126 lines

(closes issue #6002)
Reported by: rizzo
Tested by: murf

Proposal of the changes to be made, and then an announcement of how they were accomplished:

http://lists.digium.com/pipermail/asterisk-dev/2008-February/032065.html

and:

http://lists.digium.com/pipermail/asterisk-dev/2008-March/032124.html

Here is a recap, file by file, of what I have done:

pbx/pbx_config.c
pbx/pbx_ael.c

All funcs that were passed a ptr to the context list, now will ALSO be passed a hashtab ptr to the same set.
Why? because (for the time being), the dialplan is stored in both, to facilitate a quick, low-cost move to
hash-tables to speed up dialplan processing. If it was deemed necessary to pass the context LIST, well, it
is just as necessary to have the TABLE available. This is because the list/table in question might not be
the global one, but temporary ones we would use to stage the dialplan on, and then swap into the global
position when things are ready.

We now have one external function for apps to use, "ast_context_find_or_create()" instead of the pre-existing
"find" and "create", as all existing usages used both in tandem anyway.

pbx_config, and pbx_ael, will stage the reloaded dialplan into local lists and tables, and 
then call merge_contexts_and_delete, which will merge (now) existing contexts and 
priorities from other registrars into this local set by copying them. Then, merge_contexts_and_delete will
lock down the contexts, swap the lists and tables, and unlock (real quick), and then 
destroy the old dialplan.



chan_sip.c
chan_iax.c
chan_skinny.c

All the channel drivers that would add regcontexts now use the ast_context_find_or_create now.

chan_sip also includes a small fix to get rid of warnings about removing priorities that never got entered.


apps/app_meetme.c
apps/app_dial.c
apps/app_queue.c

All the apps that added a context/exten/priority were also modified to use ast_context_find_or_create instead.


include/asterisk/pbx.h

ast_context_create() is removed. Find_or_create_ is the new method.
ast_context_find_or_create()  interface gets the hashtab added.
ast_merge_contexts_and_delete() gets the local hashtab arg added.
ast_wrlock_contexts_version() is added so you can detect if someone else got a writelock between your readlocking and writelocking.
ast_hashtab_compare_contexts was made public for use in pbx_config/pbx_ael
ast_hashtab_hash_contexts was in like fashion make public.


include/asterisk/pval.h

ast_compile_ael2() interface changed to include the local hashtab table ptr.


main/features.c

For the sake of the parking context, we use ast_context_find_or_create().



main/pbx.c

I changed all the "tree" names to "table" instead. That's because the original
implementation was based on binary trees. (had a free library). Then I moved
to hashtabs. Now, the names move forward too.

refcount field added to contexts, so you can keep track of how many modules
wanted this context to exist.

Some log messages that are warnings were inflated from LOG_NOTICE to LOG_WARNING.

Added some calls to ast_verb(3,...) for debug messages

Lots of little mods to ast_context_remove_extension2, which is now excersized in ways
it was not previously; one definite bug fixed.

find_or_create was upgraded to handle both local lists/tables as well as the globals.

context_merge() was added to do the per-context merging of the old/present contexts/extens/prios into the new/proposed local list/tables

ast_merge_contexts_and_delete() was heavily modified.

ast_add_extension2() was also upgraded to handle changes. 

the context_destroy() code was re-engineered to handle the new way of doing things,
by exten/prio instead of by context.



res/ael/pval.c
res/ael/ael.tab.c
res/ael/ael.tab.h
res/ael/ael.y
res/ael/ael_lex.c
res/ael/ael.flex
utils/ael_main.c
utils/extconf.c
utils/conf2ael.c
utils/Makefile

Had to change the interface to ast_compile_ael2(), to include the hashtab ptr.
This ended up involving several external apps.  The main gotcha was I had to 
include lock.h and hashtab.h in several places.


As a side note, I tested this stuff pretty thoroughly, I replicated the problems
originally reported by Luigi, and made triply sure that reloads worked, and everything
worked thru "stop gracefully". I found a and fixed a few bugs as I was merging into
trunk, that did not appear in my tests of bug6002.

How's this for verbose commit messages?



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2008-03-12 22:51:38 +00:00
Russell Bryant
50de1f9043 Blocked revisions 108226 via svnmerge
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r108226 | file | 2008-03-12 16:06:50 -0500 (Wed, 12 Mar 2008) | 2 lines

Doxygenify slinfactory a bit.

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2008-03-12 22:51:03 +00:00
Russell Bryant
cc9f77eef9 Blocked revisions 108286 via svnmerge
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r108286 | kpfleming | 2008-03-12 16:37:40 -0500 (Wed, 12 Mar 2008) | 13 lines

add support for named sections in zapata.conf, and fix a few bugs in config file parsing

(closes issue #9503)
Reported by: tzafrir
Patches:
      fix_cleanups uploaded by tzafrir (license 46)
      zapata_sections uploaded by tzafrir (license 46)
      skipchannel_options uploaded by tzafrir (license 46)
      conf_sample uploaded by tzafrir (license 46)

patches updated by me to better conform to coding guidelines and fix some problems


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2008-03-12 22:50:45 +00:00
Russell Bryant
967d9d2c40 Merged revisions 108295 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r108295 | russell | 2008-03-12 17:13:18 -0500 (Wed, 12 Mar 2008) | 3 lines

Rename ast_tcptls_server_instance to session_instance, since this pertains to
server and client usage.

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2008-03-12 22:50:14 +00:00
Russell Bryant
64a69c1348 Merged revisions 108346 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r108346 | russell | 2008-03-12 17:49:26 -0500 (Wed, 12 Mar 2008) | 4 lines

Make the default prefix empty, like it was in Asterisk 1.4.

(closes issue #12198, reported by bkruse, patched by me)

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2008-03-12 22:49:47 +00:00
Mark Michelson
78e9f784f1 Merged revisions 108293 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r108293 | mmichelson | 2008-03-12 17:09:52 -0500 (Wed, 12 Mar 2008) | 3 lines

Let's get this to compile


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Mark Michelson
e3af4b3632 Merged revisions 108289 via svnmerge from
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r108289 | mmichelson | 2008-03-12 16:57:41 -0500 (Wed, 12 Mar 2008) | 22 lines

Merged revisions 108288 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r108288 | mmichelson | 2008-03-12 16:53:46 -0500 (Wed, 12 Mar 2008) | 14 lines

Change AST_SCHED_DEL use to ast_sched_del for autocongestion in chan_sip.

The scheduler callback will always return 0. This means that this id 
is never rescheduled, so it makes no sense to loop trying to delete
the id from the scheduler queue. If we fail to remove the item from the
queue once, it will fail every single time.

(Yes I realize that in this case, the macro would exit early because the
id is set to -1 in the callback, but it still makes no sense to use
that macro in favor of calling ast_sched_del once and being done with it)

This is the first of potentially several such fixes.


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2008-03-12 21:58:29 +00:00
Mark Michelson
2b948919be Merged revisions 108238 via svnmerge from
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r108238 | mmichelson | 2008-03-12 16:19:30 -0500 (Wed, 12 Mar 2008) | 20 lines

Merged revisions 108227 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r108227 | mmichelson | 2008-03-12 16:16:28 -0500 (Wed, 12 Mar 2008) | 12 lines

Added a large comment before the AST_SCHED_DEL macro to explain its purpose as well as when
it is appropriate and when it is not appropriate to use it.

I also removed the part of the debug message that mentions that this is probably a bug because
there are some perfectly legitimate places where ast_sched_del may fail to delete an entry (e.g.
when the scheduler callback manually reschedules with a new id instead of returning non-zero to
tell the scheduler to reschedule with the same idea). I also raised the debug level of the debug
message in AST_SCHED_DEL since it seems like it could come up quite frequently since the macro
is probably being used in several places where it shouldn't be. Also removed the redundant line,
file, and function information since that is provided by ast_log.


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2008-03-12 21:21:02 +00:00
Kevin P. Fleming
3592c06d29 Merged revisions 108191 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r108191 | kpfleming | 2008-03-12 15:27:01 -0500 (Wed, 12 Mar 2008) | 14 lines

Merged revisions 108086 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r108086 | kpfleming | 2008-03-12 14:16:07 -0500 (Wed, 12 Mar 2008) | 6 lines

if we receive an INVITE with a Content-Length that is not a valid number, or is zero, then don't process the rest of the message body looking for an SDP

closes issue #11475
Reported by: andrebarbosa


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2008-03-12 20:29:50 +00:00
Russell Bryant
45de934e14 Merged revisions 108137 via svnmerge from
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r108137 | russell | 2008-03-12 14:59:05 -0500 (Wed, 12 Mar 2008) | 48 lines

Merged revisions 108135 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r108135 | russell | 2008-03-12 14:57:42 -0500 (Wed, 12 Mar 2008) | 40 lines

(closes issue #12187, reported by atis, fixed by me after some brainstorming
 on the issue with mmichelson)

- Update copyright info on app_chanspy.

- Fix a race condition that caused app_chanspy to crash.  The issue was that
  the chanspy datastore magic that was used to ensure that spyee channels did
  not disappear out from under the code did not completely solve the problem.
  It was actually possible for chanspy to acquire a channel reference out of
  its datastore to a channel that was in the middle of being destroyed.  That
  was because datastore destruction in ast_channel_free() was done near the
  end.  So, this left the code in app_chanspy accessing a channel that was
  partially, or completely invalid because it was in the process of being free'd
  by another thread.  The following sort of shows the code path where the race 
  occurred:

  =============================================================================
  Thread 1 (PBX thread for spyee chan)  ||   Thread 2 (chanspy)
  --------------------------------------||-------------------------------------
  ast_channel_free()                    ||
    - remove channel from channel list  ||
    - lock/unlock the channel to ensure ||
      that no references retrieved from ||
      the channel list exist.           ||
  --------------------------------------||-------------------------------------
                                        || channel_spy()
    - destroy some channel data         ||  - Lock chanspy datastore
                                        ||  - Retrieve reference to channel
                                        ||  - lock channel
                                        ||  - Unlock chanspy datastore
  --------------------------------------||-------------------------------------
     - destroy channel datastores       ||
        - call chanspy datastore d'tor  ||  
          which NULL's out the ds'      ||  - Operate on the channel ...
          reference to the channel      ||     
                                        ||
    - free the channel                  || 
                                        ||
                                        ||  - unlock the channel
  --------------------------------------||-------------------------------------
  =============================================================================

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2008-03-12 19:59:24 +00:00
Joshua Colp
de10acaa07 Merged revisions 108084 via svnmerge from
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r108084 | file | 2008-03-12 15:29:33 -0300 (Wed, 12 Mar 2008) | 12 lines

Merged revisions 108083 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r108083 | file | 2008-03-12 15:26:37 -0300 (Wed, 12 Mar 2008) | 4 lines

Add a trigger mode that triggers on both read and write. The actual function that returns the combined audio frame though will wait until both sides have fed in audio, or until one side stops (such as the case when you call Wait).
(closes issue #11945)
Reported by: xheliox

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2008-03-12 18:31:07 +00:00
Russell Bryant
3e717dce36 Merged revisions 108032 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r108032 | russell | 2008-03-12 12:02:57 -0500 (Wed, 12 Mar 2008) | 12 lines

Merged revisions 108031 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r108031 | russell | 2008-03-12 11:59:07 -0500 (Wed, 12 Mar 2008) | 4 lines

Destroy the channel lock after the channel datastores.

(inspired by issue #12187)

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2008-03-12 17:03:18 +00:00
Tilghman Lesher
a92dc508ec Merged revisions 107998 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r107998 | tilghman | 2008-03-12 02:43:03 -0500 (Wed, 12 Mar 2008) | 7 lines

Deadlock fixes
(closes issue #12143)
 Reported by: kactus
 Patches: 
       20080312__bug12143__2.diff.txt uploaded by Corydon76 (license 14)
 Tested by: kactus

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2008-03-12 07:44:28 +00:00
Tilghman Lesher
02519b9df1 Merged revisions 107960 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r107960 | tilghman | 2008-03-12 00:46:39 -0500 (Wed, 12 Mar 2008) | 4 lines

Revert several changes from revision 102525, as the changes were not
compatible, and, in fact, introduced regressions.
(Closes issue #12190)

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2008-03-12 05:47:37 +00:00
Tilghman Lesher
4206a8f4f6 Merged revisions 107878 via svnmerge from
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r107878 | tilghman | 2008-03-11 20:54:00 -0500 (Tue, 11 Mar 2008) | 10 lines

Merged revisions 107877 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r107877 | tilghman | 2008-03-11 20:52:40 -0500 (Tue, 11 Mar 2008) | 2 lines

Document all of the possible realtime fields

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2008-03-12 01:55:41 +00:00
Jason Parker
007b33e9c6 Merged revisions 107827 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r107827 | qwell | 2008-03-11 18:38:00 -0500 (Tue, 11 Mar 2008) | 15 lines

Merged revisions 107826 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r107826 | qwell | 2008-03-11 18:37:05 -0500 (Tue, 11 Mar 2008) | 7 lines

Update documentation for pgsql ODBC voicemail.

(closes issue #12186)
Reported by: jsmith
Patches:
      vm_pgsql_doc_update.patch uploaded by jsmith (license 15)

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2008-03-11 23:38:31 +00:00
Tilghman Lesher
0d5ad6e21c Merged revisions 107791 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r107791 | tilghman | 2008-03-11 17:55:16 -0500 (Tue, 11 Mar 2008) | 5 lines

An offhand comment from Russell made me realize that the configuration file
caching would not work properly for users.conf and any other file read from
more than one place.  I needed to add the filename which requested the config
file to get it to work properly.

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2008-03-11 22:59:52 +00:00
Tilghman Lesher
082e13ee49 Blocked revisions 107722 via svnmerge
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r107722 | tilghman | 2008-03-11 16:10:45 -0500 (Tue, 11 Mar 2008) | 7 lines

Convert prepare_and_execute to direct_execute for speed
(closes issue #11935)
 Reported by: falves11
 Patches: 
       20080208__bug11935.diff.txt uploaded by Corydon76 (license 14)
 Tested by: falves11, Corydon76

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2008-03-11 21:11:31 +00:00
Jason Parker
2b1b649a97 Merged revisions 107718 via svnmerge from
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r107718 | qwell | 2008-03-11 15:53:48 -0500 (Tue, 11 Mar 2008) | 13 lines

Merged revisions 107714 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r107714 | qwell | 2008-03-11 15:49:56 -0500 (Tue, 11 Mar 2008) | 5 lines

Copy voicemail dependency logic for res_adsi to chan_gtalk and chan_jingle (for jabber).

(closes issue #12014)
Reported by: junky

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2008-03-11 20:54:38 +00:00
Kevin P. Fleming
2ada3e4bba Merged revisions 107715 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r107715 | kpfleming | 2008-03-11 15:50:57 -0500 (Tue, 11 Mar 2008) | 10 lines

Merged revisions 107713 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r107713 | kpfleming | 2008-03-11 15:48:58 -0500 (Tue, 11 Mar 2008) | 2 lines

get chan_vpb to build properly in dev mode

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2008-03-11 20:51:18 +00:00
Joshua Colp
e86f6f817a Merged revisions 107710 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r107710 | file | 2008-03-11 17:36:14 -0300 (Tue, 11 Mar 2008) | 6 lines

Dial a device even if it's state is unknown.
(closes issue #12184)
Reported by: bluecrow76
Patches:
      asterisk-svn-app_page.c.devicestate_unknown.diff uploaded by bluecrow76 (license 270)

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2008-03-11 20:37:09 +00:00
Joshua Colp
a241dd0b5d Merged revisions 107659 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r107659 | file | 2008-03-11 16:23:28 -0300 (Tue, 11 Mar 2008) | 12 lines

Merged revisions 107646 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r107646 | file | 2008-03-11 16:20:01 -0300 (Tue, 11 Mar 2008) | 4 lines

Make sure the visible indication is on the right channel so when the masquerade happens the proper indication is enacted.
(closes issue #11707)
Reported by: iam

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2008-03-11 19:25:00 +00:00
Joshua Colp
aad2069958 Merged revisions 107638 via svnmerge from
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r107638 | file | 2008-03-11 15:48:59 -0300 (Tue, 11 Mar 2008) | 12 lines

Merged revisions 107637 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r107637 | file | 2008-03-11 15:47:33 -0300 (Tue, 11 Mar 2008) | 4 lines

Add an additional check for setting conference parameter when using the marked user options. It was possible for it to return to a no listen/no talk state if a masquerade happened.
(closes issue #12136)
Reported by: aragon

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2008-03-11 18:50:15 +00:00
Joshua Colp
f0649d1105 Blocked revisions 107583 via svnmerge
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r107583 | file | 2008-03-11 14:34:46 -0300 (Tue, 11 Mar 2008) | 11 lines

Blocked revisions 107582 via svnmerge

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r107582 | file | 2008-03-11 14:32:17 -0300 (Tue, 11 Mar 2008) | 4 lines

Fix a minor spelling error.
(closes issue #12183)
Reported by: darrylc

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Mark Michelson
0996e4b518 Blocked revisions 107530 via svnmerge
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r107530 | mmichelson | 2008-03-11 10:59:32 -0500 (Tue, 11 Mar 2008) | 7 lines

Remove some redundant logic from wait_for_answer. This also let's us get rid of one of 
those XXX comments from the code.

The redundancy occurs because the 'single' flag implies that the 'r' and 'm' flags are
not set, so there's no need to explicitly check them again.


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2008-03-11 16:04:05 +00:00
Kevin P. Fleming
2932ac2f59 Merged revisions 107525 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r107525 | kpfleming | 2008-03-11 10:39:37 -0500 (Tue, 11 Mar 2008) | 2 lines

fix another potential bug found by gcc 4.3

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2008-03-11 15:39:53 +00:00
Kevin P. Fleming
19ca2892c0 Merged revisions 107466 via svnmerge from
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r107466 | kpfleming | 2008-03-11 10:13:38 -0500 (Tue, 11 Mar 2008) | 10 lines

Merged revisions 107464 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r107464 | kpfleming | 2008-03-11 09:53:03 -0500 (Tue, 11 Mar 2008) | 2 lines

fix various other problems found by gcc 4.3

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Kevin P. Fleming
07e93e4598 Merged revisions 107462 via svnmerge from
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r107462 | kpfleming | 2008-03-11 09:37:03 -0500 (Tue, 11 Mar 2008) | 10 lines

Merged revisions 107461 via svnmerge from 
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r107461 | kpfleming | 2008-03-11 09:33:45 -0500 (Tue, 11 Mar 2008) | 2 lines

stop checking for mktime() in the configure script... we don't use it, and the test is buggy under gcc 4.3

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2008-03-11 14:37:23 +00:00
Kevin P. Fleming
ca8d368a80 Merged revisions 107409 via svnmerge from
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r107409 | kpfleming | 2008-03-11 09:09:49 -0500 (Tue, 11 Mar 2008) | 13 lines

Merged revisions 107408 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r107408 | kpfleming | 2008-03-11 09:07:59 -0500 (Tue, 11 Mar 2008) | 5 lines

check for compiler support for -fno-strict-overflow before using it (tested with Debian's gcc 4.3, 4.1 and 3.4)

(closes issue #12179)
Reported by: Netview

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2008-03-11 14:10:08 +00:00
Kevin P. Fleming
ba3a3a37ef Merged revisions 107406 via svnmerge from
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r107406 | kpfleming | 2008-03-11 08:58:37 -0500 (Tue, 11 Mar 2008) | 10 lines

Merged revisions 107405 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r107405 | kpfleming | 2008-03-11 08:57:08 -0500 (Tue, 11 Mar 2008) | 2 lines

fix small bug in IMAP toolkit testing

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2008-03-11 13:59:27 +00:00
Kevin P. Fleming
b2fabfdb4c Merged revisions 107373 via svnmerge from
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r107373 | kpfleming | 2008-03-11 06:36:51 -0500 (Tue, 11 Mar 2008) | 19 lines

Merged revisions 107352 via svnmerge from 
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r107352 | kpfleming | 2008-03-11 06:04:29 -0500 (Tue, 11 Mar 2008) | 11 lines

fix up various compiler warnings found with gcc-4.3:

- the output of flex includes a static function called 'input' that is not used, so for the moment we'll stop having the compiler tell us about unused variables in the flex source files (a better fix would be to improve our flex post-processing to remove the unused function)

- main/stdtime/localtime.c makes assumptions about signed integer overflow, and gcc-4.3's improved optimizer tries to take advantage of handling potential overflow conditions at compile time; for now, suppress these optimizations until we can fiure out if the code needs improvement

- main/udptl.c has some references to uninitialized variables; in one case there was no bug, but in the other it was certainly possibly for unexpected behavior to occur

- main/editline/readline.c had an unused variable


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2008-03-11 11:39:32 +00:00
Terry Wilson
c9411643ea Merged revisions 107292 via svnmerge from
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r107292 | twilson | 2008-03-10 20:09:46 -0500 (Mon, 10 Mar 2008) | 10 lines

Merged revisions 107290 via svnmerge from 
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r107290 | twilson | 2008-03-10 19:59:18 -0500 (Mon, 10 Mar 2008) | 2 lines

If we fail to alloc a channel, we should re-lock the pvt structure before returning.

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2008-03-11 01:27:23 +00:00
Steve Murphy
e261ad9b4c Merged revisions 107019 via svnmerge from
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r107019 | murf | 2008-03-10 08:55:21 -0600 (Mon, 10 Mar 2008) | 1 line

way back in July, in r.75706, a fix was made ot the strftime usages, which was good, but in this case, the check for a nil time was accidentally removed, and now it is restored, to keep timevals like '1969-12-31 17:00:00' from showing up in the cdrs. No idea what databases will do with this. No bugs filed as yet, but it felt like a bug.
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2008-03-10 23:46:02 +00:00
Tilghman Lesher
109b374629 Blocked revisions 107231 via svnmerge from
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r107231 | tilghman | 2008-03-10 16:48:20 -0500 (Mon, 10 Mar 2008) | 6 lines

(closes issue #6019)
 Reported by: ssokol
 Patches: 
       20080304__bug6019.diff.txt uploaded by Corydon76 (license 14)
 Tested by: putnopvut

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2008-03-10 22:02:33 +00:00
Jason Parker
3d33906019 Merged revisions 107177 via svnmerge from
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r107177 | qwell | 2008-03-10 15:28:33 -0500 (Mon, 10 Mar 2008) | 13 lines

Merged revisions 107173 via svnmerge from 
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r107173 | qwell | 2008-03-10 15:27:08 -0500 (Mon, 10 Mar 2008) | 5 lines

Make sure to reenable echo can after a "failed" (canceled, etc) three-way call.

(closes issue #11335)
Reported by: rebuild

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2008-03-10 20:29:21 +00:00
Russell Bryant
95289aa843 Merged revisions 107162 via svnmerge from
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r107162 | russell | 2008-03-10 15:17:37 -0500 (Mon, 10 Mar 2008) | 16 lines

Merged revisions 107161 via svnmerge from 
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r107161 | russell | 2008-03-10 15:17:11 -0500 (Mon, 10 Mar 2008) | 8 lines

Fix another bug specifically related to asynchronous call origination.  Once the
PBX is started on the channel using ast_pbx_start(), then the ownership of the
channel has been passed on to another thread.  We can no longer access it in this
code.  If the channel gets hung up very quickly, it is possible that we could
access a channel that has been free'd.

(inspired by BE-386)

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2008-03-10 20:18:13 +00:00
Russell Bryant
bf658f9798 Merged revisions 107159 via svnmerge from
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r107159 | russell | 2008-03-10 15:05:12 -0500 (Mon, 10 Mar 2008) | 17 lines

Merged revisions 107158 via svnmerge from 
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r107158 | russell | 2008-03-10 15:04:27 -0500 (Mon, 10 Mar 2008) | 9 lines

Fix some bugs related to originating calls.  If the code failed to start a PBX
on the channel (such as if you set a call limit based on the system's load
average), then there were cases where a channel that has already been free'd
using ast_hangup() got accessed.  This caused weird memory corruption and
crashes to occur.

(fixes issue BE-386)
(much debugging credit goes to twilson, final patch written by me)

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Russell Bryant
9d24a5691f Merged revisions 107103 via svnmerge from
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r107103 | russell | 2008-03-10 12:13:34 -0500 (Mon, 10 Mar 2008) | 10 lines

Merged revisions 107102 via svnmerge from 
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r107102 | russell | 2008-03-10 12:13:17 -0500 (Mon, 10 Mar 2008) | 2 lines

Resolve a compiler warning.

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2008-03-10 17:13:54 +00:00
Russell Bryant
8d28aa58f6 Merged revisions 107100 via svnmerge from
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r107100 | russell | 2008-03-10 11:59:13 -0500 (Mon, 10 Mar 2008) | 11 lines

Merged revisions 107099 via svnmerge from 
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r107099 | russell | 2008-03-10 11:58:57 -0500 (Mon, 10 Mar 2008) | 3 lines

Fix a race condition where the generator can go away
(closes issue #12175, reported by edantie, patched by me)

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2008-03-10 16:59:33 +00:00
Mark Michelson
1036d0869d Merged revisions 107068 via svnmerge from
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r107068 | mmichelson | 2008-03-10 10:45:13 -0500 (Mon, 10 Mar 2008) | 10 lines

app_queue has now been doxygenified thanks to snuffy! The ony thing I changed
was the way that locks are referenced, since the old 1.2 names were still used
in the comments.

(closes issue #11997)
Reported by: snuffy
Patches:
      bug_11997_queue_doxy.diff uploaded by snuffy (license 35)


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2008-03-10 15:46:39 +00:00
Joshua Colp
1b56c4ce2d Merged revisions 107017 via svnmerge from
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r107017 | file | 2008-03-10 11:36:16 -0300 (Mon, 10 Mar 2008) | 15 lines

Merged revisions 107016 via svnmerge from 
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r107016 | file | 2008-03-10 11:33:02 -0300 (Mon, 10 Mar 2008) | 7 lines

Move where unanswered CDRs are dropped to the CDR core, not everything uses app_dial.
(closes issue #11516)
Reported by: ys
Patches:
      branch_1.4_cdr.diff uploaded by ys (license 281)
Tested by: anest, jcapp, dartvader

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2008-03-10 14:38:01 +00:00
Matthew Fredrickson
3be8b59e7c Make sure we don't start a call on a channel that has already started a call
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2008-03-08 17:54:59 +00:00
Kevin P. Fleming
0eea9ee544 Merged revisions 106946 via svnmerge from
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r106946 | kpfleming | 2008-03-08 10:03:48 -0600 (Sat, 08 Mar 2008) | 10 lines

Merged revisions 106945 via svnmerge from 
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r106945 | kpfleming | 2008-03-08 09:59:42 -0600 (Sat, 08 Mar 2008) | 2 lines

don't generate D-Channel "up" and "down" messages unless the channel state is actually changing; also, generate the "up" message when an implicit "up" occurs due to reception of a normal event when we thought the channel was "down"

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2008-03-08 16:14:57 +00:00
Russell Bryant
265c460d19 Merged revisions 106896 via svnmerge from
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r106896 | russell | 2008-03-07 16:52:46 -0600 (Fri, 07 Mar 2008) | 10 lines

Merged revisions 106895 via svnmerge from 
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r106895 | russell | 2008-03-07 16:51:23 -0600 (Fri, 07 Mar 2008) | 2 lines

Only start the SLA thread if SLA has actually been configured.

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2008-03-07 22:53:06 +00:00
Joshua Colp
254ea518ef Merged revisions 106789 via svnmerge from
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r106789 | file | 2008-03-07 15:33:09 -0400 (Fri, 07 Mar 2008) | 12 lines

Merged revisions 106788 via svnmerge from 
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r106788 | file | 2008-03-07 15:32:00 -0400 (Fri, 07 Mar 2008) | 4 lines

Ignore source update control frame.
(closes issue #12168)
Reported by: plack

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Russell Bryant
b48e9a1f08 Merged revisions 106707 via svnmerge from
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r106707 | russell | 2008-03-07 11:17:30 -0600 (Fri, 07 Mar 2008) | 16 lines

Merged revisions 106704 via svnmerge from 
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r106704 | russell | 2008-03-07 11:16:58 -0600 (Fri, 07 Mar 2008) | 8 lines

Change a warning message to a debug message.  This is happening quite frequently,
and it is not worth spamming users with these messages unless we are pretty confident
that it should never happen.  As it stands today, it _will_ and _does_ happen and
until that gets cleaned up a reasonable amount on the development side, let's not
spam the logs of everyone else.

(closes issue #12154)

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Russell Bryant
9a6ac737ae Merged revisions 106684 via svnmerge from
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r106684 | russell | 2008-03-07 10:31:48 -0600 (Fri, 07 Mar 2008) | 2 lines

fix example usage

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Tilghman Lesher
061a42ab6f Merged revisions 106654 via svnmerge from
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r106654 | tilghman | 2008-03-07 10:26:07 -0600 (Fri, 07 Mar 2008) | 11 lines

Merged revisions 106635 via svnmerge from 
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r106635 | tilghman | 2008-03-07 10:22:11 -0600 (Fri, 07 Mar 2008) | 3 lines

Warn the user when a temporary greeting exists
(Closes issue #11409)

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Tilghman Lesher
d334a88e36 Merged revisions 106607 via svnmerge from
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r106607 | tilghman | 2008-03-07 09:22:34 -0600 (Fri, 07 Mar 2008) | 11 lines

Merged revisions 106606 via svnmerge from 
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r106606 | tilghman | 2008-03-07 09:20:52 -0600 (Fri, 07 Mar 2008) | 3 lines

Properly initialize rtp->schedid
(Closes issue #12154)

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Tilghman Lesher
fd382738e3 Merged revisions 106553 via svnmerge from
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r106553 | tilghman | 2008-03-07 00:54:47 -0600 (Fri, 07 Mar 2008) | 14 lines

Merged revisions 106552 via svnmerge from 
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r106552 | tilghman | 2008-03-07 00:36:33 -0600 (Fri, 07 Mar 2008) | 6 lines

Safely use the strncat() function.
(closes issue #11958)
 Reported by: norman
 Patches: 
       20080209__bug11958.diff.txt uploaded by Corydon76 (license 14)

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2008-03-07 06:57:44 +00:00
Russell Bryant
591f6c5012 Merged revisions 106518 via svnmerge from
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r106518 | russell | 2008-03-06 19:19:02 -0600 (Thu, 06 Mar 2008) | 1 line

minor text changes
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Russell Bryant
ef1ad01921 Merged revisions 106507 via svnmerge from
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r106507 | russell | 2008-03-06 19:15:36 -0600 (Thu, 06 Mar 2008) | 2 lines

Add updated SMDI documentation that I had only sitting in my email ... oops

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Russell Bryant
ccf90e8186 Merged revisions 106501 via svnmerge from
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r106501 | russell | 2008-03-06 18:24:58 -0600 (Thu, 06 Mar 2008) | 28 lines

Merge changes from team/russell/g722-sillyness ...

Fix a number of other places where the number of samples in a G722 frame was
not properly handled because of various reasons.

main/rtp.c:
 - When a G722 frame is read from the smoother, the number of samples in the
   frame must be divided by 2 before being sent out over the network.  Even
   though G722 is 16 kHz, an error in some previous spec has made it so that
   we have to list the number of samples such as if it was 8 kHz.

main/file.c:
 - When scheduling the next time to expect a frame, take into account that the
   format of the file we're reading from may not be 8 kHz.

codecs/codec_g722.c:
 - When converting from G722 to slinear, g722_decode() expects its samples
   parameter to be in the silly (real samples / 2) format.  Make it so.
 - When converting from slinear to G722, properly set the number of samples in
   the frame to be the number of bytes of output * 2.

formats/format_pcm.c:
 - This format module handles G722, among a number of other formats.  However,
   the read() and seek() functions did not account for the fact that G722 has
   2 samples per byte.

(closes issue #12130, reported by rickross, patched by me)

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Mark Michelson
29e469fabc Merged revisions 106438 via svnmerge from
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r106438 | mmichelson | 2008-03-06 16:11:26 -0600 (Thu, 06 Mar 2008) | 16 lines

Merged revisions 106437 via svnmerge from 
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r106437 | mmichelson | 2008-03-06 16:10:07 -0600 (Thu, 06 Mar 2008) | 8 lines

Quell an annoying message that is likely to print every single time that 
ast_pbx_outgoing_app is called. The reason is that __ast_request_and_dial
allocates the cdr for the channel, so it should be expected that the channel
will have a cdr on it.

Thanks to joetester on IRC for pointing this out


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2008-03-06 22:16:55 +00:00
Jason Parker
3b548ed386 Merged revisions 106439 via svnmerge from
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r106439 | qwell | 2008-03-06 16:11:30 -0600 (Thu, 06 Mar 2008) | 8 lines

Fix file playback in many cases.

(closes issue #12115)
Reported by: pj
Patches:
      v2-fileexists.patch uploaded by dimas (license 88) (with modifications by me)
Tested by: dimas, qwell, russell

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2008-03-06 22:15:03 +00:00
Donny Kavanagh
69b6111450 Merged revisions 106399 via svnmerge from
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r106399 | juggie | 2008-03-06 14:31:50 -0500 (Thu, 06 Mar 2008) | 9 lines

trivial fix for an agi error when attempting to use EAGI on a dead/hungup channel, we now print an error that makes sense 
given our removal of deadagi as an actual application.

(closes issue #12161)
Reported by: explidous
Patches:
      res_agi_12161.patch uploaded by juggie (license 24)
Tested by: juggie

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2008-03-06 20:39:49 +00:00
Tilghman Lesher
8abd853f7f Merged revisions 106346 via svnmerge from
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r106346 | tilghman | 2008-03-05 23:21:39 -0600 (Wed, 05 Mar 2008) | 7 lines

Missing braces, fix parsing
(closes issue #12112)
 Reported by: cyrenity
 Patches: 
       res_config_ldap.patch-03-03-2008 uploaded by cyrenity (license 416)
 Tested by: cyrenity, Corydon76

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Tilghman Lesher
dff3671f09 Merged revisions 106329 via svnmerge from
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r106329 | tilghman | 2008-03-05 22:45:16 -0600 (Wed, 05 Mar 2008) | 10 lines

Merged revisions 106328 via svnmerge from 
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r106328 | tilghman | 2008-03-05 22:40:06 -0600 (Wed, 05 Mar 2008) | 2 lines

Upgrade to the next release of sounds

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Russell Bryant
3969f432af Blocked revisions 106250 via svnmerge
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r106250 | qwell | 2008-03-05 17:21:52 -0600 (Wed, 05 Mar 2008) | 1 line

Add a cmenuselect/cmenuconfig, to force curses.
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Russell Bryant
b4088e0fef Blocked revisions 106240 via svnmerge
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r106240 | russell | 2008-03-05 16:50:00 -0600 (Wed, 05 Mar 2008) | 5 lines

Add the nmenuselect makefile targets.  This is a newt menuselect interface which
is much sexier than the curses one that was written by seanbright.  :)

(issue #12139)

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Russell Bryant
90046983f7 Merged revisions 106239 via svnmerge from
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r106239 | file | 2008-03-05 16:43:22 -0600 (Wed, 05 Mar 2008) | 12 lines

Merged revisions 106235 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r106235 | file | 2008-03-05 18:32:10 -0400 (Wed, 05 Mar 2008) | 4 lines

Add a control frame to indicate the source of media has changed. Depending on the underlying technology it may need to change some things.
(closes issue #12148)
Reported by: jcomellas

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2008-03-06 00:22:20 +00:00
Russell Bryant
16456f61f6 Merged revisions 106238 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r106238 | russell | 2008-03-05 16:40:58 -0600 (Wed, 05 Mar 2008) | 11 lines

Merged revisions 106237 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r106237 | russell | 2008-03-05 16:37:09 -0600 (Wed, 05 Mar 2008) | 3 lines

Fix a potential deadlock and a few different potential crashes.
(closes issue #12145, reported by thiagarcia, patched by me)

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2008-03-06 00:21:24 +00:00
Russell Bryant
3f996ba1b3 Blocked revisions 106236 via svnmerge
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r106236 | mmichelson | 2008-03-05 16:33:05 -0600 (Wed, 05 Mar 2008) | 11 lines

Adding the Atxfer manager command. With this, you may initiate
an attended transfer over AMI

(closes issue #10585)
Reported by: ornati
Patches:
      atxfer-trunk-r90428.diff uploaded by ornati (license 210)
	  (with modifications from me)
Tested by: putnopvut


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2008-03-06 00:20:55 +00:00
Russell Bryant
a3a7672808 Merged revisions 106186 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r106186 | mvanbaak | 2008-03-05 15:19:06 -0600 (Wed, 05 Mar 2008) | 7 lines

document var_metric usage to prevent bugreports that are actually configuration issues

(closes issue #12151)
Reported by: caio1982
Patches:
      DB_metric3.diff uploaded by caio1982 (license 22)

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2008-03-06 00:20:35 +00:00
Russell Bryant
0fe91ba8d9 Blocked revisions 106179 via svnmerge
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r106179 | mvanbaak | 2008-03-05 15:15:57 -0600 (Wed, 05 Mar 2008) | 12 lines

Blocked revisions 106178 via svnmerge

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r106178 | mvanbaak | 2008-03-05 22:12:36 +0100 (Wed, 05 Mar 2008) | 5 lines

document var_metric so no bugreports will come in when it's actually a configuration issue.
(issue #12151)
Reported and patched by: caio1982
1.4 patch by me

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2008-03-06 00:20:00 +00:00
Russell Bryant
717e0179cf Blocked revisions 106110 via svnmerge
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r106110 | file | 2008-03-05 10:39:22 -0600 (Wed, 05 Mar 2008) | 2 lines

Fix code up to what it was meant to be.

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2008-03-06 00:18:05 +00:00
Russell Bryant
24d1a5334f Blocked revisions 106072 via svnmerge
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r106072 | tilghman | 2008-03-05 10:23:44 -0600 (Wed, 05 Mar 2008) | 7 lines

Create a centralized configuration option for silencethreshold
(closes issue #11236)
 Reported by: philipps
 Patches: 
       20080218__bug11236.diff.txt uploaded by Corydon76 (license 14)
 Tested by: philipps

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2008-03-06 00:17:33 +00:00
Russell Bryant
3e8a42c9c3 Blocked revisions 105984 via svnmerge
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r105984 | russell | 2008-03-04 22:34:29 -0600 (Tue, 04 Mar 2008) | 3 lines

 - simplify a few statements with ARRAY_LEN()
 - constify the stregy int to string mappings array

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2008-03-06 00:16:57 +00:00
Russell Bryant
78393c6698 Merged revisions 105933 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r105933 | russell | 2008-03-04 19:54:16 -0600 (Tue, 04 Mar 2008) | 13 lines

Merged revisions 105932 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r105932 | russell | 2008-03-04 19:52:18 -0600 (Tue, 04 Mar 2008) | 5 lines

Fix a bug that I just noticed in the RTP code.  The calculation for setting the
len field in an ast_frame of audio was wrong when G.722 is in use.  The len field
represents the number of ms of audio that the frame contains.  It would have
set the value to be twice what it should be.

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2008-03-06 00:16:30 +00:00
Russell Bryant
51f5f4470a Merged revisions 105899 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r105899 | russell | 2008-03-04 18:45:39 -0600 (Tue, 04 Mar 2008) | 3 lines

Fix the SHARED() read callback to properly unlock the channel.  This function
could not have worked, as it left the channel locked in all cases.

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2008-03-06 00:15:54 +00:00
Russell Bryant
b66de7ac7b Merged revisions 105864 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r105864 | mmichelson | 2008-03-04 17:24:56 -0600 (Tue, 04 Mar 2008) | 5 lines

There are several places in manager.c where BUFSIZ is used for a buffer
which will contain nowhere near that amount of data. This makes these buffers
more reasonably sized.


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2008-03-06 00:15:30 +00:00
Russell Bryant
e09e677a81 Merged revisions 105841 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r105841 | tilghman | 2008-03-04 17:10:45 -0600 (Tue, 04 Mar 2008) | 2 lines

Fix minor misuses of snprintf

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2008-03-06 00:15:02 +00:00
Russell Bryant
9c09333526 Merged revisions 105840 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r105840 | tilghman | 2008-03-04 17:04:29 -0600 (Tue, 04 Mar 2008) | 2 lines

Whitespace changes only

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2008-03-06 00:14:32 +00:00
Russell Bryant
908eddfe58 Merged revisions 105804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r105804 | russell | 2008-03-04 16:28:03 -0600 (Tue, 04 Mar 2008) | 2 lines

add a destroy API call for a server instance

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2008-03-06 00:13:54 +00:00
Russell Bryant
36f5261290 Merged revisions 105785 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r105785 | russell | 2008-03-04 16:23:21 -0600 (Tue, 04 Mar 2008) | 2 lines

More public API name changes to use an appropriate ast_ prefix

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2008-03-06 00:13:10 +00:00
Russell Bryant
78a863b9a2 Merged revisions 105773 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r105773 | russell | 2008-03-04 16:15:18 -0600 (Tue, 04 Mar 2008) | 2 lines

Rename public object server_instance to ast_tcptls_server_instance

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2008-03-06 00:11:09 +00:00
Russell Bryant
c6f570d433 Merged revisions 105734 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r105734 | russell | 2008-03-04 14:36:16 -0600 (Tue, 04 Mar 2008) | 6 lines

Fix some bugs in the SIP tcp helper thread.
 - fix a spot where a lock wouldn't get unlocked in an error condition
 - call ast_mutex_destroy() on the lock before freeing its memory

(related to issue #11972)

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2008-03-06 00:10:33 +00:00
Russell Bryant
28f18301fd Merged revisions 105733 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r105733 | twilson | 2008-03-04 14:32:55 -0600 (Tue, 04 Mar 2008) | 2 lines

Set username to default to the category name if it isn't overridden by a usernmae= setting in users.conf

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2008-03-06 00:10:08 +00:00
Russell Bryant
83942c273a Merged revisions 105677 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r105677 | file | 2008-03-04 12:11:38 -0600 (Tue, 04 Mar 2008) | 10 lines

Merged revisions 105676 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r105676 | file | 2008-03-04 14:10:34 -0400 (Tue, 04 Mar 2008) | 2 lines

In addition to setting the marker bit let's change our ssrc so they know for sure it is a different source.

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2008-03-06 00:08:11 +00:00
Russell Bryant
b11caafd7e Merged revisions 105675 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r105675 | file | 2008-03-04 12:08:42 -0600 (Tue, 04 Mar 2008) | 16 lines

Merged revisions 105674 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r105674 | file | 2008-03-04 14:05:28 -0400 (Tue, 04 Mar 2008) | 8 lines

When a new source of audio comes in (such as music on hold) make sure the marker bit gets set.
(closes issue #10355)
Reported by: wdecarne
Patches:
      10355.diff uploaded by file (license 11)
(closes issue #11491)
Reported by: kanderson

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2008-03-06 00:07:09 +00:00
Tilghman Lesher
12b513fe48 Merged revisions 106139 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r106139 | tilghman | 2008-03-05 11:40:42 -0600 (Wed, 05 Mar 2008) | 3 lines

Should check these values for non-NULL before scanning.
(Closes issue #12147)

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2008-03-05 17:42:36 +00:00
Kevin P. Fleming
e6e2862eb8 Merged revisions 106040 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r106040 | kpfleming | 2008-03-05 09:40:40 -0600 (Wed, 05 Mar 2008) | 15 lines

Merged revisions 106038 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r106038 | kpfleming | 2008-03-05 09:32:35 -0600 (Wed, 05 Mar 2008) | 7 lines

when a PRI call must be moved to a different B channel at the request of the other endpoint, ensure that any DSP active on the original channel is moved to the new one

(closes issue #11917)
Reported by: mavetju
Tested by: mavetju


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2008-03-05 15:43:11 +00:00
Tilghman Lesher
b7201beb50 Merged revisions 106036 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r106036 | tilghman | 2008-03-05 09:23:32 -0600 (Wed, 05 Mar 2008) | 15 lines

Merged revisions 106015 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r106015 | tilghman | 2008-03-05 09:17:16 -0600 (Wed, 05 Mar 2008) | 7 lines

Correctly initialize retransid in SIP, and ensure that the warning when failing to delete a schedule entry can actually hit the log.
(closes issue #12140)
 Reported by: slavon
 Patches: 
       sch2.patch uploaded by slavon (license 288)
(Patch slightly modified by me)

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2008-03-05 15:31:37 +00:00
Russell Bryant
764077c366 change the versions in the heading
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@105600 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-04 17:00:01 +00:00
Russell Bryant
107c67a6f4 Blocked revisions 105597 via svnmerge
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r105597 | russell | 2008-03-04 10:55:17 -0600 (Tue, 04 Mar 2008) | 2 lines

Update CHANGES heading

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2008-03-04 16:57:30 +00:00
Russell Bryant
4d49cdf23e set up trunk-merged property
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@105598 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-04 16:56:37 +00:00
Russell Bryant
39244fb614 Create 1.6.0 branch (email to asterisk-dev coming soon to explain)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@105596 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-04 16:53:42 +00:00
639 changed files with 114424 additions and 71548 deletions

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.lastclean Normal file
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.version Normal file
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@@ -0,0 +1 @@
1.6.0.13-rc1

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@@ -1,6 +1,36 @@
======================================================================
===
=== This file documents the new and/or enhanced functionality added in
=== the Asterisk versions listed below. This file does NOT include
=== changes in behavior that would not be backwards compatible with
=== previous versions; for that information see the UPGRADE.txt file
=== and the other UPGRADE files for older releases.
===
======================================================================
-----------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.6.0.10 to Asterisk 1.6.0.11 -------------
-----------------------------------------------------------------------------------
SIP Changes
-----------
* Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
(either globally or for a specific peer), chan_sip will treat any SDP data
it receives as new data and update the media stream accordingly. By
default, Asterisk will only modify the media stream if the SDP session
version received is different from the current SDP session version. This
option is required to interoperate with devices that have non-standard SDP
session version implementations (observed with Microsoft OCS). This option
is disabled by default. In addition, this behavior is automatic when the SDP received
is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
since the call will fail if Asterisk does not process the incoming SDP, Asterisk
will accept the SDP even if the SDP version number is not properly incremented,
but will generate a warning in the log indicating that the SIP peer that sent
the SDP should have the 'ignoresdpversion' option set.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
------------------------------------------------------------------------------
--- Functionality changes since Asterisk 1.4-beta was branched ----------------
-------------------------------------------------------------------------------
AMI - The manager (TCP/TLS/HTTP)
--------------------------------
@@ -82,6 +112,10 @@ Dialplan functions
ID for the call (not the Asterisk call ID or unique ID), provided that the
channel driver supports this. For SIP, you get the SIP call-ID for the
bridged channel which you can store in the CDR with a custom field.
* Added the function AUDIOHOOK_INHERIT. This actually is already in Asterisk
1.4, but since it was added late in the release cycle, I felt it was a good
idea to list it here as well. See the CLI output for "core show function
AUDIOHOOK_INHERIT" for more details
CLI Changes
-----------
@@ -149,8 +183,9 @@ SIP changes
* Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
were not properly torn down due to network or endpoint failures during an established
SIP session.
* Added TCP and TLS support for SIP. See doc/siptls.txt and configs/sip.conf.sample for
more information on how it is used.
* Added experimental TCP and TLS support for SIP. See doc/siptls.txt and
configs/sip.conf.sample for more information on how it is used.
* Added t38pt_usertpsource option. See sip.conf.sample for details.
IAX2 changes
------------
@@ -169,7 +204,7 @@ Skinny changes
* Proper codec support in chan_skinny.
* Added settings for IP and Ethernet QoS requests
MGCP changes
------------
* Added separate settings for media QoS in mgcp.conf
@@ -200,28 +235,28 @@ Local channel changes
feature is enabled by using the 'j' option in the Dial string to the Local
channel in conjunction with the existing 'n' option for local channels.
Zaptel channel driver (chan_zap) Changes
DAHDI channel driver (chan_dahdi) Changes
----------------------------------------
* SS7 support in chan_zap (via libss7 library)
* SS7 support (via libss7 library)
* In India, some carriers transmit CID via dtmf. Some code has been added
that will handle some situations. The cidstart=polarity_IN choice has been added for
those carriers that transmit CID via dtmf after a polarity change.
* CID matching information is now shown when doing 'dialplan show'.
* Added zap show version CLI command to chan_zap.
* Added setvar support to zapata.conf channel entries.
* Added dahdi show version CLI command.
* Added setvar support to chan_dahdi.conf channel entries.
* Added two new options: mwimonitor and mwimonitornotify. These options allow
you to enable MWI monitoring on FXO lines. When the MWI state changes,
the script specified in the mwimonitornotify option is executed. An internal
event indicating the new state of the mailbox is also generated, so that
the normal MWI facilities in Asterisk work as usual.
* Added signalling type 'auto', which attempts to use the same signalling type
for a channel as configured in Zaptel. This is primarily designed for analog
for a channel as configured in DAHDI. This is primarily designed for analog
ports, but will also work for digital ports that are configured for FXS or FXO
signalling types. This mode is also the default now, so if your zapata.conf
signalling types. This mode is also the default now, so if your chan_dahdi.conf
does not specify signalling for a channel (which is unlikely as the sample
configuration file has always recommended specifying it for every channel) then
the 'auto' mode will be used for that channel if possible.
* Added a 'zap set dnd' command to allow CLI control of the Do-Not-Disturb
* Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
state for a channel; also ensured that the DNDState Manager event is
emitted no matter how the DND state is set or cleared.
@@ -363,7 +398,7 @@ Other Dialplan Application Changes
WaitForRing() now takes floating pt timeout arg.
SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
* Added 's' option to Page application.
* Added 'E' and 'V' commands to ExternalIVR.
* Added 'E', 'V', and 'P' commands to ExternalIVR.
* Added 'o' and 'X' options to Chanspy.
* Added a new dialplan application, Bridge, which allows you to bridge the
calling channel to any other active channel on the system.
@@ -381,6 +416,12 @@ Other Dialplan Application Changes
* The ChannelRedirect application no longer exits the dialplan if the given channel
does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
or NOCHANNEL if the given channel was not found.
* ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
to obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
Music On Hold Changes
---------------------
@@ -422,6 +463,13 @@ AEL Changes
Call Features (res_features) Changes
------------------------------------
* Added the parkedcalltransfers option to features.conf
* Added parkedcallparking option to control one touch parking w/ parking
pickup
* Added parkedcallhangup option to control disconnect feature w/ parking
pickup
* Added parkedcallrecording option to control one-touch record w/ parking
pickup
* Added BRIDGE_FEATURES variable to set available features for a channel
* The built-in method for doing attended transfers has been updated to
include some new options that allow you to have the transferee sent
back to the person that did the transfer if the transfer is not successful.
@@ -446,6 +494,9 @@ AGI Changes
-----------
* Added SPEECH commands for speech recognition. A complete listing can be found
using agi show.
* If app_stack is loaded, GOSUB is a native AGI command that may be used to
invoke subroutines in the dialplan. Note that calling EXEC with Gosub
does not behave as expected; the native command needs to be used, instead.
Logger changes
--------------
@@ -522,6 +573,9 @@ Miscellaneous New Modules
Miscellaneous
-------------
* res_jabber: autoprune has been disabled by default, to avoid misconfiguration
that would end up being interpreted as a bug once Asterisk started removing
the contacts from a user list.
* Ability to use libcap to set high ToS bits when non-root
on Linux. If configure is unable to find libcap then you
can use --with-cap to specify the path.
@@ -555,4 +609,10 @@ Miscellaneous
* Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
turned on, via the CHANNEL(trace) dialplan function. Could be useful for
dialplan debugging.
* iLBC source code no longer included (see UPGRADE.txt for details)
* A new option for the configure script, --enable-internal-poll, has been added
for use with systems which may have a buggy implementation of the poll system
call. If you notice odd behavior such as the CLI being unresponsive on remote
consoles, you may want to try using this option. This option is enabled by default
on Darwin systems since it is known that the Darwin poll() implementation has
odd issues.

54594
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View File

@@ -31,8 +31,8 @@ there is no requirement that you provide the same exception in your
GPL'd products (although if you've written a module for Asterisk we
would strongly encourage you to make the same exception that we do).
Specific permission is also granted to link Asterisk with OpenSSL and
OpenH323 and distribute the resulting binary files.
Specific permission is also granted to link Asterisk with OpenSSL, OpenH323
and/or the UW IMAP Toolkit and distribute the resulting binary files.
In addition, Asterisk implements two management/control protocols: the
Asterisk Manager Interface (AMI) and the Asterisk Gateway Interface
@@ -60,7 +60,7 @@ contact us:
+1.877.344.4861 (via telephone in the USA)
+1.256.428.6000 (via telephone outside the USA)
+1.256.864.0464 (via FAX inside or outside the USA)
IAX2/misery.digium.com/6000 (via IAX2)
IAX2/pbx.digium.com (via IAX2)
licensing@digium.com (via email)
Digium, Inc.

232
Makefile
View File

@@ -13,29 +13,23 @@
# All Makefiles use the following variables:
#
# ASTCFLAGS - compiler options
# ASTLDFLAGS - linker flags (not libraries)
# ASTCFLAGS - compiler options provided by the user (if any)
# _ASTCFLAGS - compiler options provided by the build system
# ASTLDFLAGS - linker flags (not libraries) provided by the user (if any)
# _ASTLDFLAGS - linker flags (not libraries) provided by the build system
# LIBS - additional libraries, at top-level for all links,
# on a single object just for that object
# SOLINK - linker flags used only for creating shared objects (.so files),
# used for all .so links
#
# Initial values for ASTCFLAGS and ASTLDFLAGS can be specified in the
# Values for ASTCFLAGS and ASTLDFLAGS can be specified in the
# environment when running make, as follows:
#
# $ ASTCFLAGS="-Werror" make ...
#
# note that this is different from
# or as a variable value on the make command line itself:
#
# $ make ASTCFLAGS="-Werror" ...
#
# If you need to pass compiler/linker flags as 'make' variables, please use
#
# $ make COPTS="..." LDOPTS="..." ...
#
#
# You can add the path of local module subdirs from the command line with
# make LOCAL_MOD_SUBDIRS= ....
export ASTTOPDIR # Top level dir, used in subdirs' Makefiles
export ASTERISKVERSION
@@ -92,6 +86,7 @@ export DOWNLOAD
export AWK
export GREP
export ID
export MD5
# even though we could use '-include makeopts' here, use a wildcard
# lookup anyway, so that make won't try to build makeopts if it doesn't
@@ -170,7 +165,7 @@ HTTP_DOCSDIR=/var/www/html
HTTP_CGIDIR=/var/www/cgi-bin
# Uncomment this to use the older DSP routines
#ASTCFLAGS+=-DOLD_DSP_ROUTINES
#_ASTCFLAGS+=-DOLD_DSP_ROUTINES
# If the file .asterisk.makeopts is present in your home directory, you can
# include all of your favorite menuselect options so that every time you download
@@ -220,65 +215,69 @@ ifeq ($(OSARCH),linux-gnu)
endif
endif
ifeq ($(findstring -save-temps,$(ASTCFLAGS)),)
ASTCFLAGS+=-pipe
ifeq ($(findstring -save-temps,$(_ASTCFLAGS) $(ASTCFLAGS)),)
ifeq ($(findstring -pipe,$(_ASTCFLAGS) $(ASTCFLAGS)),)
_ASTCFLAGS+=-pipe
endif
endif
ASTCFLAGS+=-Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations $(DEBUG)
ifeq ($(findstring -Wall,$(_ASTCFLAGS) $(ASTCFLAGS)),)
_ASTCFLAGS+=-Wall
endif
ASTCFLAGS+=-include $(ASTTOPDIR)/include/asterisk/autoconfig.h
_ASTCFLAGS+=-Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations $(DEBUG)
ifeq ($(AST_DEVMODE),yes)
ASTCFLAGS+=-Werror -Wunused -Wundef $(AST_DECLARATION_AFTER_STATEMENT)
_ASTCFLAGS+=-Werror
_ASTCFLAGS+=-Wunused
_ASTCFLAGS+=$(AST_DECLARATION_AFTER_STATEMENT)
_ASTCFLAGS+=$(AST_FORTIFY_SOURCE)
_ASTCFLAGS+=-Wundef
_ASTCFLAGS+=-Wmissing-format-attribute
_ASTCFLAGS+=-Wformat=2
endif
ifneq ($(findstring BSD,$(OSARCH)),)
ASTCFLAGS+=-I/usr/local/include
ASTLDFLAGS+=-L/usr/local/lib
_ASTCFLAGS+=-I/usr/local/include
endif
ifneq ($(PROC),ultrasparc)
ASTCFLAGS+=$(shell if $(CC) -march=$(PROC) -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then echo "-march=$(PROC)"; fi)
_ASTCFLAGS+=$(shell if $(CC) -march=$(PROC) -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then echo "-march=$(PROC)"; fi)
endif
ifeq ($(PROC),ppc)
ASTCFLAGS+=-fsigned-char
_ASTCFLAGS+=-fsigned-char
endif
ifeq ($(OSARCH),FreeBSD)
# -V is understood by BSD Make, not by GNU make.
BSDVERSION=$(shell make -V OSVERSION -f /usr/share/mk/bsd.port.subdir.mk)
ASTCFLAGS+=$(shell if test $(BSDVERSION) -lt 500016 ; then echo "-D_THREAD_SAFE"; fi)
_ASTCFLAGS+=$(shell if test $(BSDVERSION) -lt 500016 ; then echo "-D_THREAD_SAFE"; fi)
endif
ifeq ($(OSARCH),NetBSD)
ASTCFLAGS+=-pthread -I/usr/pkg/include
_ASTCFLAGS+=-pthread -I/usr/pkg/include
endif
ifeq ($(OSARCH),OpenBSD)
ASTCFLAGS+=-pthread
_ASTCFLAGS+=-pthread
endif
ifeq ($(OSARCH),SunOS)
ASTCFLAGS+=-Wcast-align -DSOLARIS -I../include/solaris-compat -I/opt/ssl/include -I/usr/local/ssl/include -D_XPG4_2
_ASTCFLAGS+=-Wcast-align -DSOLARIS -I../include/solaris-compat -I/opt/ssl/include -I/usr/local/ssl/include -D_XPG4_2
endif
ASTERISKVERSION:=$(shell GREP=$(GREP) AWK=$(AWK) build_tools/make_version .)
ifneq ($(wildcard .version),)
ASTERISKVERSIONNUM:=$(shell $(AWK) -F. '{printf "%01d%02d%02d", $$1, $$2, $$3}' .version)
RPMVERSION:=$(shell sed 's/[-\/:]/_/g' .version)
else
RPMVERSION=unknown
endif
ifneq ($(wildcard .svn),)
ASTERISKVERSIONNUM:=999999
endif
# XXX MALLOC_DEBUG is probably unused, Makefile.moddir_rules adds the
# value directly to ASTCFLAGS
ASTCFLAGS+=$(MALLOC_DEBUG)$(OPTIONS)
_ASTCFLAGS+=$(OPTIONS)
MOD_SUBDIRS:=channels pbx apps codecs formats cdr funcs tests main res $(LOCAL_MOD_SUBDIRS)
OTHER_SUBDIRS:=utils agi
@@ -293,18 +292,18 @@ MOD_SUBDIRS_EMBED_LIBS:=$(MOD_SUBDIRS:%=%-embed-libs)
MOD_SUBDIRS_MENUSELECT_TREE:=$(MOD_SUBDIRS:%=%-menuselect-tree)
ifneq ($(findstring darwin,$(OSARCH)),)
ASTCFLAGS+=-D__Darwin__
_ASTCFLAGS+=-D__Darwin__
SOLINK=-dynamic -bundle -undefined suppress -force_flat_namespace
else
# These are used for all but Darwin
SOLINK=-shared -Xlinker -x
SOLINK=-shared
ifneq ($(findstring BSD,$(OSARCH)),)
LDFLAGS+=-L/usr/local/lib
_ASTLDFLAGS+=-L/usr/local/lib
endif
endif
ifeq ($(OSARCH),SunOS)
SOLINK=-shared -fpic -L/usr/local/ssl/lib
SOLINK=-shared -fpic -L/usr/local/ssl/lib -lrt
endif
# comment to print directories during submakes
@@ -312,7 +311,7 @@ endif
SILENTMAKE:=$(MAKE) --quiet --no-print-directory
ifneq ($(PRINT_DIR)$(NOISY_BUILD),)
SUBMAKE:=$(MAKE) --quiet
SUBMAKE:=$(MAKE)
else
SUBMAKE:=$(MAKE) --quiet --no-print-directory
endif
@@ -350,28 +349,28 @@ makeopts: configure
@echo "****"
@exit 1
menuselect.makeopts: menuselect/menuselect menuselect-tree
menuselect/menuselect --check-deps $(GLOBAL_MAKEOPTS) $(USER_MAKEOPTS) menuselect.makeopts
menuselect.makeopts: menuselect/menuselect menuselect-tree makeopts
menuselect/menuselect --check-deps menuselect.makeopts $(GLOBAL_MAKEOPTS) $(USER_MAKEOPTS)
$(MOD_SUBDIRS_EMBED_LDSCRIPT):
@echo "EMBED_LDSCRIPTS+="`$(SILENTMAKE) -C $(@:-embed-ldscript=) SUBDIR=$(@:-embed-ldscript=) __embed_ldscript` >> makeopts.embed_rules
+@echo "EMBED_LDSCRIPTS+="`$(SILENTMAKE) -C $(@:-embed-ldscript=) SUBDIR=$(@:-embed-ldscript=) __embed_ldscript` >> makeopts.embed_rules
$(MOD_SUBDIRS_EMBED_LDFLAGS):
@echo "EMBED_LDFLAGS+="`$(SILENTMAKE) -C $(@:-embed-ldflags=) SUBDIR=$(@:-embed-ldflags=) __embed_ldflags` >> makeopts.embed_rules
+@echo "EMBED_LDFLAGS+="`$(SILENTMAKE) -C $(@:-embed-ldflags=) SUBDIR=$(@:-embed-ldflags=) __embed_ldflags` >> makeopts.embed_rules
$(MOD_SUBDIRS_EMBED_LIBS):
@echo "EMBED_LIBS+="`$(SILENTMAKE) -C $(@:-embed-libs=) SUBDIR=$(@:-embed-libs=) __embed_libs` >> makeopts.embed_rules
+@echo "EMBED_LIBS+="`$(SILENTMAKE) -C $(@:-embed-libs=) SUBDIR=$(@:-embed-libs=) __embed_libs` >> makeopts.embed_rules
$(MOD_SUBDIRS_MENUSELECT_TREE):
@$(SUBMAKE) -C $(@:-menuselect-tree=) SUBDIR=$(@:-menuselect-tree=) moduleinfo
@$(SUBMAKE) -C $(@:-menuselect-tree=) SUBDIR=$(@:-menuselect-tree=) makeopts
+@$(SUBMAKE) -C $(@:-menuselect-tree=) SUBDIR=$(@:-menuselect-tree=) moduleinfo
+@$(SUBMAKE) -C $(@:-menuselect-tree=) SUBDIR=$(@:-menuselect-tree=) makeopts
makeopts.embed_rules: menuselect.makeopts
@echo "Generating embedded module rules ..."
@rm -f $@
@$(MAKE) $(PRINT_DIR) $(MOD_SUBDIRS_EMBED_LDSCRIPT)
@$(MAKE) $(PRINT_DIR) $(MOD_SUBDIRS_EMBED_LDFLAGS)
@$(MAKE) $(PRINT_DIR) $(MOD_SUBDIRS_EMBED_LIBS)
+@$(SUBMAKE) $(MOD_SUBDIRS_EMBED_LDSCRIPT)
+@$(SUBMAKE) $(MOD_SUBDIRS_EMBED_LDFLAGS)
+@$(SUBMAKE) $(MOD_SUBDIRS_EMBED_LIBS)
$(SUBDIRS): main/version.c include/asterisk/version.h include/asterisk/build.h include/asterisk/buildopts.h defaults.h makeopts.embed_rules
@@ -394,22 +393,22 @@ res: main
endif
$(MOD_SUBDIRS):
@ASTCFLAGS="$(MOD_SUBDIR_CFLAGS) $(ASTCFLAGS)" ASTLDFLAGS="$(ASTLDFLAGS)" $(MAKE) $(PRINT_DIR) --no-builtin-rules -C $@ SUBDIR=$@ all
@_ASTCFLAGS="$(MOD_SUBDIR_CFLAGS) $(_ASTCFLAGS)" ASTCFLAGS="$(ASTCFLAGS)" _ASTLDFLAGS="$(_ASTLDFLAGS)" ASTLDFLAGS="$(ASTLDFLAGS)" $(SUBMAKE) --no-builtin-rules -C $@ SUBDIR=$@ all
$(OTHER_SUBDIRS):
@ASTCFLAGS="$(OTHER_SUBDIR_CFLAGS) $(ASTCFLAGS)" ASTLDFLAGS="$(ASTLDFLAGS)" $(MAKE) $(PRINT_DIR) --no-builtin-rules -C $@ SUBDIR=$@ all
@_ASTCFLAGS="$(OTHER_SUBDIR_CFLAGS) $(_ASTCFLAGS)" ASTCFLAGS="$(ASTCFLAGS)" _ASTLDFLAGS="$(_ASTLDFLAGS)" ASTLDFLAGS="$(ASTLDFLAGS)" $(SUBMAKE) --no-builtin-rules -C $@ SUBDIR=$@ all
defaults.h: makeopts
@build_tools/make_defaults_h > $@.tmp
@cmp -s $@.tmp $@ || mv $@.tmp $@
@rm -f $@.tmp
main/version.c:
main/version.c: FORCE
@build_tools/make_version_c > $@.tmp
@cmp -s $@.tmp $@ || mv $@.tmp $@
@rm -f $@.tmp
include/asterisk/version.h:
include/asterisk/version.h: FORCE
@build_tools/make_version_h > $@.tmp
@cmp -s $@.tmp $@ || mv $@.tmp $@
@rm -f $@.tmp
@@ -425,12 +424,14 @@ include/asterisk/build.h:
@rm -f $@.tmp
$(SUBDIRS_CLEAN):
@$(MAKE) $(PRINT_DIR) -C $(@:-clean=) clean
+@$(SUBMAKE) -C $(@:-clean=) clean
$(SUBDIRS_DIST_CLEAN):
@$(MAKE) $(PRINT_DIR) -C $(@:-dist-clean=) dist-clean
+@$(SUBMAKE) -C $(@:-dist-clean=) dist-clean
clean: $(SUBDIRS_CLEAN)
clean: $(SUBDIRS_CLEAN) _clean
_clean:
rm -f defaults.h
rm -f include/asterisk/build.h
rm -f main/version.c
@@ -440,12 +441,12 @@ clean: $(SUBDIRS_CLEAN)
dist-clean: distclean
distclean: $(SUBDIRS_DIST_CLEAN) clean
distclean: $(SUBDIRS_DIST_CLEAN) _clean
@$(MAKE) -C menuselect dist-clean
@$(MAKE) -C sounds dist-clean
rm -f menuselect.makeopts makeopts menuselect-tree menuselect.makedeps
rm -f makeopts.embed_rules
rm -f config.log config.status
rm -f config.log config.status config.cache
rm -rf autom4te.cache
rm -f include/asterisk/autoconfig.h
rm -f include/asterisk/buildopts.h
@@ -453,14 +454,10 @@ distclean: $(SUBDIRS_DIST_CLEAN) clean
rm -f build_tools/menuselect-deps
datafiles: _all
if [ x`$(ID) -un` = xroot ]; then CFLAGS="$(ASTCFLAGS)" sh build_tools/mkpkgconfig $(DESTDIR)/usr/lib/pkgconfig; fi
if [ x`$(ID) -un` = xroot ]; then CFLAGS="$(_ASTCFLAGS) $(ASTCFLAGS)" bash build_tools/mkpkgconfig $(DESTDIR)/usr/lib/pkgconfig; fi
# Should static HTTP be installed during make samples or even with its own target ala
# webvoicemail? There are portions here that *could* be customized but might also be
# improved a lot. I'll put it here for now.
mkdir -p $(DESTDIR)$(ASTDATADIR)/phoneprov
for x in phoneprov/*; do \
$(INSTALL) -m 644 $$x $(DESTDIR)$(ASTDATADIR)/phoneprov ; \
done
mkdir -p $(DESTDIR)$(ASTDATADIR)/static-http
for x in static-http/*; do \
$(INSTALL) -m 644 $$x $(DESTDIR)$(ASTDATADIR)/static-http ; \
@@ -498,7 +495,7 @@ update:
NEWHEADERS=$(notdir $(wildcard include/asterisk/*.h))
OLDHEADERS=$(filter-out $(NEWHEADERS),$(notdir $(wildcard $(DESTDIR)$(ASTHEADERDIR)/*.h)))
bininstall: _all
installdirs:
mkdir -p $(DESTDIR)$(MODULES_DIR)
mkdir -p $(DESTDIR)$(ASTSBINDIR)
mkdir -p $(DESTDIR)$(ASTETCDIR)
@@ -510,6 +507,8 @@ bininstall: _all
mkdir -p $(DESTDIR)$(ASTSPOOLDIR)/tmp
mkdir -p $(DESTDIR)$(ASTSPOOLDIR)/meetme
mkdir -p $(DESTDIR)$(ASTSPOOLDIR)/monitor
bininstall: _all installdirs $(SUBDIRS_INSTALL)
$(INSTALL) -m 755 main/asterisk $(DESTDIR)$(ASTSBINDIR)/
$(LN) -sf asterisk $(DESTDIR)$(ASTSBINDIR)/rasterisk
$(INSTALL) -m 755 contrib/scripts/astgenkey $(DESTDIR)$(ASTSBINDIR)/
@@ -541,7 +540,7 @@ bininstall: _all
fi
$(SUBDIRS_INSTALL):
@DESTDIR="$(DESTDIR)" ASTSBINDIR="$(ASTSBINDIR)" $(MAKE) --quiet $(PRINT_DIR) -C $(@:-install=) install
@DESTDIR="$(DESTDIR)" ASTSBINDIR="$(ASTSBINDIR)" $(SUBMAKE) -C $(@:-install=) install
NEWMODS:=$(foreach d,$(MOD_SUBDIRS),$(notdir $(wildcard $(d)/*.so)))
OLDMODS=$(filter-out $(NEWMODS),$(notdir $(wildcard $(DESTDIR)$(MODULES_DIR)/*.so)))
@@ -571,7 +570,7 @@ ifneq ($(findstring ~,$(DESTDIR)),)
@exit 1
endif
install: badshell datafiles bininstall $(SUBDIRS_INSTALL)
install: badshell datafiles bininstall
@if [ -x /usr/sbin/asterisk-post-install ]; then \
/usr/sbin/asterisk-post-install $(DESTDIR) . ; \
fi
@@ -674,6 +673,7 @@ samples: adsi
echo ";cache_record_files = yes ; Cache recorded sound files to another directory during recording" ; \
echo ";record_cache_dir = /tmp ; Specify cache directory (used in cnjunction with cache_record_files)" ; \
echo ";transmit_silence_during_record = yes ; Transmit SLINEAR silence while a channel is being recorded" ; \
echo ";transmit_silence = yes ; Transmit SLINEAR silence while a channel is being recorded or DTMF is being generated" ; \
echo ";transcode_via_sln = yes ; Build transcode paths via SLINEAR, instead of directly" ; \
echo ";runuser = asterisk ; The user to run as" ; \
echo ";rungroup = asterisk ; The group to run as" ; \
@@ -684,12 +684,35 @@ samples: adsi
echo ";astctlowner = root" ; \
echo ";astctlgroup = apache" ; \
echo ";astctl = asterisk.ctl" ; \
echo "" ; \
echo "[compat]" ; \
echo "pbx_realtime=1.6" ; \
echo "res_agi=1.6" ; \
echo "app_set=1.6" ; \
) > $(DESTDIR)$(ASTCONFPATH) ; \
else \
echo "Skipping asterisk.conf creation"; \
fi
mkdir -p $(DESTDIR)$(ASTSPOOLDIR)/voicemail/default/1234/INBOX
build_tools/make_sample_voicemail $(DESTDIR)/$(ASTDATADIR) $(DESTDIR)/$(ASTSPOOLDIR)
@mkdir -p $(DESTDIR)$(ASTDATADIR)/phoneprov
@for x in phoneprov/*; do \
dst="$(DESTDIR)$(ASTDATADIR)/$$x" ; \
if [ -f $${dst} ]; then \
if [ "$(OVERWRITE)" = "y" ]; then \
if cmp -s $${dst} $$x ; then \
echo "Config file $$x is unchanged"; \
continue; \
fi ; \
mv -f $${dst} $${dst}.old ; \
else \
echo "Skipping config file $$x"; \
continue; \
fi ;\
fi ; \
echo "Installing file $$x"; \
$(INSTALL) -m 644 $$x $${dst} ;\
done
webvmail:
@[ -d $(DESTDIR)$(HTTP_DOCSDIR)/ ] || ( printf "http docs directory not found.\nUpdate assignment of variable HTTP_DOCSDIR in Makefile!\n" && exit 1 )
@@ -717,24 +740,18 @@ webvmail:
@echo " + +"
@echo " +-------------------------------------------+"
spec:
sed "s/^Version:.*/Version: $(RPMVERSION)/g" redhat/asterisk.spec > asterisk.spec ; \
rpm: __rpm
__rpm: main/version.c include/asterisk/version.h include/asterisk/buildopts.h spec
rm -rf /tmp/asterisk ; \
mkdir -p /tmp/asterisk/redhat/RPMS/i386 ; \
$(MAKE) DESTDIR=/tmp/asterisk install ; \
$(MAKE) DESTDIR=/tmp/asterisk samples ; \
mkdir -p /tmp/asterisk/etc/rc.d/init.d ; \
cp -f contrib/init.d/rc.redhat.asterisk /tmp/asterisk/etc/rc.d/init.d/asterisk ; \
rpmbuild --rcfile /usr/lib/rpm/rpmrc:redhat/rpmrc -bb asterisk.spec
progdocs:
(cat contrib/asterisk-ng-doxygen; echo "HAVE_DOT=$(HAVEDOT)"; \
echo "PROJECT_NUMBER=$(ASTERISKVERSION)") | doxygen -
install-logrotate:
if [ ! -d $(ASTETCDIR)/../logrotate.d ]; then \
mkdir $(ASTETCDIR)/../logrotate.d ; \
fi
sed 's#__LOGDIR__#$(ASTLOGDIR)#g' < contrib/scripts/asterisk.logrotate | sed 's#__SBINDIR__#$(ASTSBINDIR)#g' > contrib/scripts/asterisk.logrotate.tmp
install -m 0644 contrib/scripts/asterisk.logrotate.tmp $(ASTETCDIR)/../logrotate.d/asterisk
rm -f contrib/scripts/asterisk.logrotate.tmp
config:
@if [ "${OSARCH}" = "linux-gnu" ]; then \
if [ -f /etc/redhat-release -o -f /etc/fedora-release ]; then \
@@ -747,13 +764,13 @@ config:
$(INSTALL) -m 755 contrib/init.d/rc.gentoo.asterisk $(DESTDIR)/etc/init.d/asterisk; \
if [ -z "$(DESTDIR)" ]; then /sbin/rc-update add asterisk default; fi; \
elif [ -f /etc/mandrake-release -o -f /etc/mandriva-release ]; then \
$(INSTALL) -m 755 contrib/init.d/rc.mandrake.asterisk $(DESTDIR)/etc/rc.d/init.d/asterisk; \
$(INSTALL) -m 755 contrib/init.d/rc.mandriva.asterisk $(DESTDIR)/etc/rc.d/init.d/asterisk; \
if [ -z "$(DESTDIR)" ]; then /sbin/chkconfig --add asterisk; fi; \
elif [ -f /etc/SuSE-release -o -f /etc/novell-release ]; then \
$(INSTALL) -m 755 contrib/init.d/rc.suse.asterisk $(DESTDIR)/etc/init.d/asterisk; \
if [ -z "$(DESTDIR)" ]; then /sbin/chkconfig --add asterisk; fi; \
elif [ -f /etc/slackware-version ]; then \
echo "Slackware is not currently supported, although an init script does exist for it." \
echo "Slackware is not currently supported, although an init script does exist for it."; \
else \
echo "We could not install init scripts for your distribution."; \
fi \
@@ -772,7 +789,7 @@ cleantest:
@cmp -s .cleancount .lastclean || $(MAKE) clean
$(SUBDIRS_UNINSTALL):
@$(MAKE) $(PRINT_DIR) -C $(@:-uninstall=) uninstall
@$(SUBMAKE) -C $(@:-uninstall=) uninstall
_uninstall: $(SUBDIRS_UNINSTALL)
rm -f $(DESTDIR)$(MODULES_DIR)/*
@@ -811,34 +828,63 @@ uninstall-all: _uninstall
menuconfig: menuselect
cmenuconfig: cmenuselect
gmenuconfig: gmenuselect
menuselect: menuselect/menuselect menuselect-tree
-@menuselect/menuselect $(GLOBAL_MAKEOPTS) $(USER_MAKEOPTS) menuselect.makeopts && (echo "menuselect changes saved!"; rm -f channels/h323/Makefile.ast main/asterisk) || echo "menuselect changes NOT saved!"
nmenuconfig: nmenuselect
menuselect: menuselect/cmenuselect menuselect/nmenuselect menuselect/gmenuselect
@if [ -x menuselect/nmenuselect ]; then \
$(MAKE) nmenuselect; \
elif [ -x menuselect/cmenuselect ]; then \
$(MAKE) cmenuselect; \
elif [ -x menuselect/gmenuselect ]; then \
$(MAKE) gmenuselect; \
else \
echo "No menuselect user interface found. Install ncurses,"; \
echo "newt or GTK libraries to build one and re-rerun"; \
echo "'make menuselect'."; \
fi
cmenuselect: menuselect/cmenuselect menuselect-tree
-@menuselect/cmenuselect menuselect.makeopts $(GLOBAL_MAKEOPTS) $(USER_MAKEOPTS) && (echo "menuselect changes saved!"; rm -f channels/h323/Makefile.ast main/asterisk) || echo "menuselect changes NOT saved!"
gmenuselect: menuselect/gmenuselect menuselect-tree
-@menuselect/gmenuselect $(GLOBAL_MAKEOPTS) $(USER_MAKEOPTS) menuselect.makeopts && (echo "menuselect changes saved!"; rm -f channels/h323/Makefile.ast main/asterisk) || echo "menuselect changes NOT saved!"
-@menuselect/gmenuselect menuselect.makeopts $(GLOBAL_MAKEOPTS) $(USER_MAKEOPTS) && (echo "menuselect changes saved!"; rm -f channels/h323/Makefile.ast main/asterisk) || echo "menuselect changes NOT saved!"
nmenuselect: menuselect/nmenuselect menuselect-tree
-@menuselect/nmenuselect menuselect.makeopts $(GLOBAL_MAKEOPTS) $(USER_MAKEOPTS) && (echo "menuselect changes saved!"; rm -f channels/h323/Makefile.ast main/asterisk) || echo "menuselect changes NOT saved!"
# options for make in menuselect/
MAKE_MENUSELECT=CC="$(HOST_CC)" CXX="$(CXX)" LD="" AR="" RANLIB="" CFLAGS="" $(MAKE) -C menuselect CONFIGURE_SILENT="--silent"
menuselect/menuselect: menuselect/makeopts
$(MAKE_MENUSELECT)
+$(MAKE_MENUSELECT) menuselect
menuselect/cmenuselect: menuselect/makeopts
+$(MAKE_MENUSELECT) cmenuselect
menuselect/gmenuselect: menuselect/makeopts
$(MAKE_MENUSELECT) gmenuselect
+$(MAKE_MENUSELECT) gmenuselect
menuselect/makeopts:
$(MAKE_MENUSELECT) makeopts
menuselect/nmenuselect: menuselect/makeopts
+$(MAKE_MENUSELECT) nmenuselect
menuselect-tree: $(foreach dir,$(filter-out main,$(MOD_SUBDIRS)),$(wildcard $(dir)/*.c) $(wildcard $(dir)/*.cc)) build_tools/cflags.xml sounds/sounds.xml build_tools/embed_modules.xml configure
menuselect/makeopts: makeopts
+$(MAKE_MENUSELECT) makeopts
menuselect-tree: $(foreach dir,$(filter-out main,$(MOD_SUBDIRS)),$(wildcard $(dir)/*.c) $(wildcard $(dir)/*.cc)) build_tools/cflags.xml build_tools/cflags-devmode.xml sounds/sounds.xml build_tools/embed_modules.xml configure
@echo "Generating input for menuselect ..."
@echo "<?xml version=\"1.0\"?>" > $@
@echo >> $@
@echo "<menu name=\"Asterisk Module and Build Option Selection\">" >> $@
@for dir in $(sort $(filter-out main,$(MOD_SUBDIRS))); do $(SUBMAKE) -C $${dir} SUBDIR=$${dir} moduleinfo >> $@; done
@for dir in $(sort $(filter-out main,$(MOD_SUBDIRS))); do $(SUBMAKE) -C $${dir} SUBDIR=$${dir} makeopts >> $@; done
+@for dir in $(sort $(filter-out main,$(MOD_SUBDIRS))); do $(SILENTMAKE) -C $${dir} SUBDIR=$${dir} moduleinfo >> $@; done
+@for dir in $(sort $(filter-out main,$(MOD_SUBDIRS))); do $(SILENTMAKE) -C $${dir} SUBDIR=$${dir} makeopts >> $@; done
@cat build_tools/cflags.xml >> $@
@if [ "${AST_DEVMODE}" = "yes" ]; then \
cat build_tools/cflags-devmode.xml >> $@; \
fi
@cat build_tools/embed_modules.xml >> $@
@cat sounds/sounds.xml >> $@
@echo "</menu>" >> $@
@@ -847,4 +893,6 @@ pdf: asterisk.pdf
asterisk.pdf:
$(MAKE) -C doc/tex asterisk.pdf
.PHONY: menuselect main sounds clean dist-clean distclean all prereqs cleantest uninstall _uninstall uninstall-all pdf dont-optimize $(SUBDIRS_INSTALL) $(SUBDIRS_DIST_CLEAN) $(SUBDIRS_CLEAN) $(SUBDIRS_UNINSTALL) $(SUBDIRS) $(MOD_SUBDIRS_EMBED_LDSCRIPT) $(MOD_SUBDIRS_EMBED_LDFLAGS) $(MOD_SUBDIRS_EMBED_LIBS) badshell main/version.c include/asterisk/version.h
.PHONY: menuselect menuselect.makeopts main sounds clean dist-clean distclean all prereqs cleantest uninstall _uninstall uninstall-all pdf dont-optimize $(SUBDIRS_INSTALL) $(SUBDIRS_DIST_CLEAN) $(SUBDIRS_CLEAN) $(SUBDIRS_UNINSTALL) $(SUBDIRS) $(MOD_SUBDIRS_EMBED_LDSCRIPT) $(MOD_SUBDIRS_EMBED_LDFLAGS) $(MOD_SUBDIRS_EMBED_LIBS) badshell main/version.c include/asterisk/version.h installdirs
FORCE:

View File

@@ -18,14 +18,8 @@
# Also note that we can only set one variable per rule, so we have to
# repeat the left hand side to set multiple variables.
ifneq ($(findstring MALLOC_DEBUG,$(MENUSELECT_CFLAGS)),)
ifeq ($(findstring astmm.h,$(ASTCFLAGS)),)
ASTCFLAGS+=-include $(ASTTOPDIR)/include/asterisk/astmm.h
endif
endif
ifeq ($(findstring LOADABLE_MODULES,$(MENUSELECT_CFLAGS)),)
ASTCFLAGS+=${GC_CFLAGS}
_ASTCFLAGS+=${GC_CFLAGS}
endif
ifneq ($(findstring STATIC_BUILD,$(MENUSELECT_CFLAGS)),)
@@ -64,18 +58,18 @@ endif
# (they are stored in file 'makeopts')
$(addsuffix .oo,$(CC_MODS)) $(addsuffix .o,$(C_MODS)): \
ASTCFLAGS+= -DAST_MODULE=\"$*\" $(MENUSELECT_OPTS_$*:%=-D%) $(foreach dep,$(MENUSELECT_DEPENDS_$*),$(value $(dep)_INCLUDE))
_ASTCFLAGS+= -DAST_MODULE=\"$*\" $(MENUSELECT_OPTS_$*:%=-D%) $(foreach dep,$(MENUSELECT_DEPENDS_$*),$(value $(dep)_INCLUDE))
ifeq ($(findstring $(OSARCH), mingw32 cygwin ),)
# don't define -fPIC on mingw32 and cygwin, it is the default
$(LOADABLE_MODS:%=%.so): ASTCFLAGS+=-fPIC
$(LOADABLE_MODS:%=%.so): _ASTCFLAGS+=-fPIC
endif
# For loadable modules, pass _LIB and _LDFLAGS from menuselect.
$(LOADABLE_MODS:%=%.so): LIBS+=$(foreach dep,$(MENUSELECT_DEPENDS_$*),$(value $(dep)_LIB))
$(LOADABLE_MODS:%=%.so): ASTLDFLAGS+=$(foreach dep,$(MENUSELECT_DEPENDS_$*),$(value $(dep)_LDFLAGS))
$(LOADABLE_MODS:%=%.so): _ASTLDFLAGS+=$(foreach dep,$(MENUSELECT_DEPENDS_$*),$(value $(dep)_LDFLAGS))
$(EMBEDDED_MODS:%=%.o): ASTCFLAGS+=-DEMBEDDED_MODULE=$*
$(EMBEDDED_MODS:%=%.o): _ASTCFLAGS+=-DEMBEDDED_MODULE=$*
$(addsuffix .so,$(filter $(LOADABLE_MODS),$(C_MODS))): %.so: %.o
$(addsuffix .so,$(filter $(LOADABLE_MODS),$(CC_MODS))): %.so: %.oo
@@ -116,8 +110,8 @@ modules.link:
@for file in $(patsubst %,$(SUBDIR)/%,$(filter-out %.eo,$^)); do echo "INPUT (../$${file})" >> $@; done
clean::
rm -f *.so *.o *.oo *.eo
rm -f .*.o.d .*.oo.d
rm -f *.so *.o *.oo *.eo *.i *.ii
rm -f .*.d
rm -f *.s *.i
rm -f modules.link

View File

@@ -3,7 +3,7 @@
#
# Makefile rules
#
# Copyright (C) 2006, Digium, Inc.
# Copyright (C) 2006-2008, Digium, Inc.
#
# Kevin P. Fleming <kpfleming@digium.com>
#
@@ -11,7 +11,6 @@
# the GNU General Public License
#
# Rules for various build phases.
# Each command is preceded by a short comment on what to do.
# Prefixing one or the other with @\# or @ or nothing makes the desired
# behaviour. ECHO_PREFIX prefixes the comment, CMD_PREFIX prefixes the command.
@@ -20,64 +19,94 @@
.PHONY: dist-clean
# If 'make' decides to create intermediate files to satisfy a build requirement
# (like producing a .i from a .c), we want to keep them, so tell make to keep
# all intermediate files
.SECONDARY:
# extra cflags to build dependencies. Recursively expanded.
MAKE_DEPS= -MMD -MT $@ -MF .$(subst /,_,$@).d -MP
MAKE_DEPS=-MD -MT $@ -MF .$(subst /,_,$@).d -MP
ifeq ($(NOISY_BUILD),)
ECHO_PREFIX=@
CMD_PREFIX=@
else
ECHO_PREFIX=@\#
ECHO_PREFIX=@\#
CMD_PREFIX=
endif
ifeq ($(findstring DONT_OPTIMIZE,$(MENUSELECT_CFLAGS)),)
# More GSM codec optimization
# Uncomment to enable MMXTM optimizations for x86 architecture CPU's
# which support MMX instructions. This should be newer pentiums,
# ppro's, etc, as well as the AMD K6 and K7.
#K6OPT=-DK6OPT
OPTIMIZE?=-O6
OPTIMIZE?=-O6
ASTCFLAGS+=$(OPTIMIZE)
ifeq ($(findstring DONT_OPTIMIZE,$(MENUSELECT_CFLAGS)),)
_ASTCFLAGS+=$(OPTIMIZE)
endif
# build rules for various targets
%.o: %.c
$(ECHO_PREFIX) echo " [CC] $< -> $@"
$(CMD_PREFIX) $(CC) -o $@ -c $< $(PTHREAD_CFLAGS) $(ASTCFLAGS) $(MAKE_DEPS)
# shortcuts for common combinations of flags; these must be recursively expanded so that
# per-target settings will be applied
CC_CFLAGS=$(PTHREAD_CFLAGS) $(_ASTCFLAGS) $(ASTCFLAGS)
CXX_CFLAGS=$(PTHREAD_CFLAGS) $(filter-out -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations $(AST_DECLARATION_AFTER_STATEMENT),$(_ASTCFLAGS) $(ASTCFLAGS))
CC_LDFLAGS_SO=$(PTHREAD_CFLAGS) $(_ASTLDFLAGS) $(SOLINK) $(ASTLDFLAGS)
CXX_LDFLAGS_SO=$(PTHREAD_CFLAGS) $(_ASTLDFLAGS) $(SOLINK) $(ASTLDFLAGS)
CC_LIBS=$(PTHREAD_LIBS) $(LIBS)
CXX_LIBS=$(PTHREAD_LIBS) $(LIBS)
%.o: %.i
$(ECHO_PREFIX) echo " [CCi] $< -> $@"
$(CMD_PREFIX) $(CC) -o $@ -c $< $(PTHREAD_CFLAGS) $(ASTCFLAGS) $(MAKE_DEPS)
%.i: %.c
$(ECHO_PREFIX) echo " [CPP] $< -> $@"
$(CMD_PREFIX) $(CC) -o $@ -E $< $(PTHREAD_CFLAGS) $(ASTCFLAGS) $(MAKE_DEPS)
# determine whether to double-compile so that the optimizer can report code path problems
# this is only done when developer mode and DONT_OPTIMIZE are both enabled
# in that case, we run the preprocessor to produce a .i or .ii file from the source
# code, then compile once with optimizer enabled (and the output to /dev/null),
# and if that doesn't fail then compile again with optimizer disabled
ifeq ($(findstring DONT_OPTIMIZE,$(MENUSELECT_CFLAGS))$(AST_DEVMODE),DONT_OPTIMIZEyes)
COMPILE_DOUBLE=yes
endif
%.o: %.s
$(ECHO_PREFIX) echo " [AS] $< -> $@"
$(CMD_PREFIX) $(CC) -o $@ -c $< $(PTHREAD_CFLAGS) $(ASTCFLAGS) $(MAKE_DEPS)
ifeq ($(COMPILE_DOUBLE),yes)
$(CMD_PREFIX) $(CC) -o /dev/null -c $< $(OPTIMIZE) $(CC_CFLAGS)
endif
$(CMD_PREFIX) $(CC) -o $@ -c $< $(CC_CFLAGS)
%.o: %.i
$(ECHO_PREFIX) echo " [CCi] $< -> $@"
ifeq ($(COMPILE_DOUBLE),yes)
$(CMD_PREFIX) $(CC) -o /dev/null -c $< $(OPTIMIZE) $(CC_CFLAGS)
endif
$(CMD_PREFIX) $(CC) -o $@ -c $< $(CC_CFLAGS)
ifneq ($(COMPILE_DOUBLE),yes)
%.o: %.c
$(ECHO_PREFIX) echo " [CC] $< -> $@"
$(CMD_PREFIX) $(CC) -o $@ -c $< $(MAKE_DEPS) $(CC_CFLAGS)
endif
%.i: %.c
$(ECHO_PREFIX) echo " [CPP] $< -> $@"
$(CMD_PREFIX) $(CC) -o $@ -E $< $(MAKE_DEPS) $(CC_CFLAGS)
%.oo: %.ii
$(ECHO_PREFIX) echo " [CXXi] $< -> $@"
ifeq ($(COMPILE_DOUBLE),yes)
$(CMD_PREFIX) $(CXX) -o /dev/null -c $< $(OPTIMIZE) $(CXX_CFLAGS)
endif
$(CMD_PREFIX) $(CXX) -o $@ -c $< $(CXX_CFLAGS)
ifneq ($(COMPILE_DOUBLE),yes)
%.oo: %.cc
$(ECHO_PREFIX) echo " [CXX] $< -> $@"
$(CMD_PREFIX) $(CXX) -o $@ -c $< $(PTHREAD_CFLAGS) $(filter-out -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations,$(ASTCFLAGS)) $(MAKE_DEPS)
$(CMD_PREFIX) $(CXX) -o $@ -c $< $(MAKE_DEPS) $(CXX_CFLAGS)
endif
%.c: %.y
$(ECHO_PREFIX) echo " [BISON] $< -> $@"
$(CMD_PREFIX) bison -o $@ -d --name-prefix=ast_yy $<
%.c: %.fl
$(ECHO_PREFIX) echo " [FLEX] $< -> $@"
$(CMD_PREFIX) flex -o $@ --full $<
%.ii: %.cc
$(ECHO_PREFIX) echo " [CPP] $< -> $@"
$(CMD_PREFIX) $(CXX) -o $@ -E $< $(MAKE_DEPS) $(CXX_CFLAGS)
%.so: %.o
$(ECHO_PREFIX) echo " [LD] $^ -> $@"
$(CMD_PREFIX) $(CC) $(STATIC_BUILD) -o $@ $(PTHREAD_CFLAGS) $(ASTLDFLAGS) $(SOLINK) $^ $(PTHREAD_LIBS) $(LIBS)
$(CMD_PREFIX) $(CC) $(STATIC_BUILD) -o $@ $(CC_LDFLAGS_SO) $^ $(CC_LIBS)
%.so: %.oo
$(ECHO_PREFIX) echo " [LDXX] $^ -> $@"
$(CMD_PREFIX) $(CXX) $(STATIC_BUILD) -o $@ $(PTHREAD_CFLAGS) $(ASTLDFLAGS) $(SOLINK) $^ $(PTHREAD_LIBS) $(LIBS)
$(CMD_PREFIX) $(CXX) $(STATIC_BUILD) -o $@ $(CXX_LDFLAGS_SO) $^ $(CXX_LIBS)
%.eo: %.o
$(ECHO_PREFIX) echo " [EMBED] $< -> $@"
@@ -93,6 +122,6 @@ endif
%: %.o
$(ECHO_PREFIX) echo " [LD] $^ -> $@"
$(CMD_PREFIX) $(CXX) $(STATIC_BUILD) -o $@ $(PTHREAD_CFLAGS) $(ASTLDFLAGS) $^ $(PTHREAD_LIBS) $(LIBS)
$(CMD_PREFIX) $(CXX) $(STATIC_BUILD) -o $@ $(PTHREAD_CFLAGS) $(_ASTLDFLAGS) $^ $(CXX_LIBS) $(ASTLDFLAGS)
dist-clean::
dist-clean:: clean

2
README
View File

@@ -175,7 +175,7 @@ they're used only to help make the configuration file easier to
understand, and do not affect how it is actually parsed.
Entries of the form 'variable=value' set the value of some parameter in
asterisk. For example, in zapata.conf, one might specify:
asterisk. For example, in dahdi.conf, one might specify:
switchtype=national

View File

@@ -1,10 +1,17 @@
=========================================================
===
=== Information for upgrading from Asterisk 1.0 to 1.2
===
=== This file documents all the changes that MUST be taken
=== into account when upgrading between the Asterisk
=== versions listed below. These changes may require that
=== you modify your configuration files, dialplan or (in
=== some cases) source code if you have your own Asterisk
=== modules or patches. These files also includes advance
=== notice of any functionality that has been marked as
=== 'deprecated' and may be removed in a future release,
=== along with the suggested replacement functionality.
===
=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
=== UPGRADE.txt -- Upgrade info for 1.4 to 1.6
=========================================================
Compiling:

View File

@@ -1,10 +1,19 @@
=========================================================
===
=== Information for upgrading from Asterisk 1.2 to 1.4
===
=== These files document all the changes that MUST be taken
=== into account when upgrading between the Asterisk
=== versions listed below. These changes may require that
=== you modify your configuration files, dialplan or (in
=== some cases) source code if you have your own Asterisk
=== modules or patches. These files also includes advance
=== notice of any functionality that has been marked as
=== 'deprecated' and may be removed in a future release,
=== along with the suggested replacement functionality.
===
=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
=== UPGRADE.txt -- Upgrade info for 1.4 to 1.6
===
=========================================================
Build Process (configure script):

265
UPGRADE-1.6.txt Normal file
View File

@@ -0,0 +1,265 @@
=========================================================
===
=== Information for upgrading from Asterisk 1.4 to 1.6
===
=== These files document all the changes that MUST be taken
=== into account when upgrading between the Asterisk
=== versions listed below. These changes may require that
=== you modify your configuration files, dialplan or (in
=== some cases) source code if you have your own Asterisk
=== modules or patches. These files also includes advance
=== notice of any functionality that has been marked as
=== 'deprecated' and may be removed in a future release,
=== along with the suggested replacement functionality.
===
=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
===
=========================================================
AEL:
* Macros are now implemented underneath with the Gosub() application.
Heaven Help You if you wrote code depending on any aspect of this!
Previous to 1.6, macros were implemented with the Macro() app, which
provided a nice feature of auto-returning. The compiler will do its
best to insert a Return() app call at the end of your macro if you did
not include it, but really, you should make sure that all execution
paths within your macros end in "return;".
* The conf2ael program is 'introduced' in this release; it is in a rather
crude state, but deemed useful for making a first pass at converting
extensions.conf code into AEL. More intelligence will come with time.
Core:
* The 'languageprefix' option in asterisk.conf is now deprecated, and
the default sound file layout for non-English sounds is the 'new
style' layout introduced in Asterisk 1.4 (and used by the automatic
sound file installer in the Makefile).
* The ast_expr2 stuff has been modified to handle floating-point numbers.
Numbers of the format D.D are now acceptable input for the expr parser,
Where D is a string of base-10 digits. All math is now done in "long double",
if it is available on your compiler/architecture. This was half-way between
a bug-fix (because the MATH func returns fp by default), and an enhancement.
Also, for those counting on, or needing, integer operations, a series of
'functions' were also added to the expr language, to allow several styles
of rounding/truncation, along with a set of common floating point operations,
like sin, cos, tan, log, pow, etc. The ability to call external functions
like CDR(), etc. was also added, without having to use the ${...} notation.
* The delimiter passed to applications has been changed to the comma (','), as
that is what people are used to using within extensions.conf. If you are
using realtime extensions, you will need to translate your existing dialplan
to use this separator. To use a literal comma, you need merely to escape it
with a backslash ('\'). Another possible side effect is that you may need to
remove the obscene level of backslashing that was necessary for the dialplan
to work correctly in 1.4 and previous versions. This should make writing
dialplans less painful in the future, albeit with the pain of a one-time
conversion. If you would like to avoid this conversion immediately, set
pbx_realtime=1.4 in the [compat] section of asterisk.conf. After
transitioning, set pbx_realtime=1.6 in the same section.
* For the same purpose as above, you may set res_agi=1.4 in the [compat]
section of asterisk.conf to continue to use the '|' delimiter in the EXEC
arguments of AGI applications. After converting to use the ',' delimiter,
change this option to res_agi=1.6.
* As a side effect of the application delimiter change, many places that used
to need quotes in order to get the proper meaning are no longer required.
You now only need to quote strings in configuration files if you literally
want quotation marks within a string.
* The logger.conf option 'rotatetimestamp' has been deprecated in favor of
'rotatestrategy'. This new option supports a 'rotate' strategy that more
closely mimics the system logger in terms of file rotation.
* The concise versions of various CLI commands are now deprecated. We recommend
using the manager interface (AMI) for application integration with Asterisk.
Voicemail:
* The voicemail configuration values 'maxmessage' and 'minmessage' have
been changed to 'maxsecs' and 'minsecs' to clarify their purpose and
to make them more distinguishable from 'maxmsgs', which sets folder
size. The old variables will continue to work in this version, albeit
with a deprecation warning.
* If you use any interface for modifying voicemail aside from the built in
dialplan applications, then the option "pollmailboxes" *must* be set in
voicemail.conf for message waiting indication (MWI) to work properly. This
is because Voicemail notification is now event based instead of polling
based. The channel drivers are no longer responsible for constantly manually
checking mailboxes for changes so that they can send MWI information to users.
Examples of situations that would require this option are web interfaces to
voicemail or an email client in the case of using IMAP storage.
Applications:
* ChanIsAvail() now has a 't' option, which allows the specified device
to be queried for state without consulting the channel drivers. This
performs mostly a 'ChanExists' sort of function.
* ChannelRedirect() will not terminate the channel that fails to do a
channelredirect as it has done previously. Instead CHANNELREDIRECT_STATUS
will reflect if the attempt was successful of not.
* SetCallerPres() has been replaced with the CALLERPRES() dialplan function
and is now deprecated.
* DISA()'s fifth argument is now an options argument. If you have previously
used 'NOANSWER' in this argument, you'll need to convert that to the new
option 'n'.
* Macro() is now deprecated. If you need subroutines, you should use the
Gosub()/Return() applications. To replace MacroExclusive(), we have
introduced dialplan functions LOCK(), TRYLOCK(), and UNLOCK(). You may use
these functions in any location where you desire to ensure that only one
channel is executing that path at any one time. The Macro() applications
are deprecated for performance reasons. However, since Macro() has been
around for a long time and so many dialplans depend heavily on it, for the
sake of backwards compatibility it will not be removed . It is also worth
noting that using both Macro() and GoSub() at the same time is _heavily_
discouraged.
* Read() now sets a READSTATUS variable on exit. It does NOT automatically
return -1 (and hangup) anymore on error. If you want to hangup on error,
you need to do so explicitly in your dialplan.
* Privacy() no longer uses privacy.conf, so any options must be specified
directly in the application arguments.
* MusicOnHold application now has duration parameter which allows specifying
timeout in seconds.
* WaitMusicOnHold application is now deprecated in favor of extended MusicOnHold.
* SetMusicOnHold is now deprecated. You should use Set(CHANNEL(musicclass)=...)
instead.
* The arguments in ExecIf changed a bit, to be more like other applications.
The syntax is now ExecIf(<cond>?appiftrue(args):appiffalse(args)).
* The behavior of the Set application now depends upon a compatibility option,
set in asterisk.conf. To use the old 1.4 behavior, which allowed Set to take
multiple key/value pairs, set app_set=1.4 in [compat] in asterisk.conf. To
use the new behavior, which permits variables to be set with embedded commas,
set app_set=1.6 in [compat] in asterisk.conf. Note that you can have both
behaviors at the same time, if you switch to using MSet if you want the old
behavior.
Dialplan Functions:
* QUEUE_MEMBER_COUNT() has been deprecated in favor of the QUEUE_MEMBER() function. For
more information, issue a "show function QUEUE_MEMBER" from the CLI.
CDR:
* The cdr_sqlite module has been marked as deprecated in favor of
cdr_sqlite3_custom. It will potentially be removed from the tree
after Asterisk 1.6 is released.
* The cdr_odbc module now uses res_odbc to manage its connections. The
username and password parameters in cdr_odbc.conf, therefore, are no
longer used. The dsn parameter now points to an entry in res_odbc.conf.
* The uniqueid field in the core Asterisk structure has been changed from a
maximum 31 character field to a 149 character field, to account for all
possible values the systemname prefix could be. In the past, if the
systemname was too long, the uniqueid would have been truncated.
* The cdr_tds module now supports all versions of FreeTDS that contain
the db-lib frontend. It will also now log the userfield variable if
the target database table contains a column for it.
Formats:
* format_wav: The GAIN preprocessor definition and source code that used it
is removed. This change was made in response to user complaints of
choppiness or the clipping of loud signal peaks. To increase the volume
of voicemail messages, use the 'volgain' option in voicemail.conf
Channel Drivers:
* SIP: a small upgrade to support the "Record" button on the SNOM360,
which sends a sip INFO message with a "Record: on" or "Record: off"
header. If Asterisk is set up (via features.conf) to accept "One Touch Monitor"
requests (by default, via '*1'), then the user-configured dialpad sequence
is generated, and recording can be started and stopped via this button. The
file names and formats are all controlled via the normal mechanisms. If the
user has not configured the automon feature, the normal "415 Unsupported media type"
is returned, and nothing is done.
* SIP: The "call-limit" option is marked as deprecated. It still works in this version of
Asterisk, but will be removed in the following version. Please use the groupcount functions
in the dialplan to enforce call limits. The "limitonpeer" configuration option is
now renamed to "counteronpeer".
* SIP: The "username" option is now renamed to "defaultuser" to match "defaultip".
These are used only before registration to call a peer with the uri
sip:defaultuser@defaultip
The "username" setting still work, but is deprecated and will not work in
the next version of Asterisk.
* chan_local.c: the comma delimiter inside the channel name has been changed to a
semicolon, in order to make the Local channel driver compatible with the comma
delimiter change in applications.
* H323: The "tos" setting has changed name to "tos_audio" and "cos" to "cos_audio"
to be compatible with settings in sip.conf. The "tos" and "cos" configuration
is deprecated and will stop working in the next release of Asterisk.
* Console: A new console channel driver, chan_console, has been added to Asterisk.
This new module can not be loaded at the same time as chan_alsa or chan_oss. The
default modules.conf only loads one of them (chan_oss by default). So, unless you
have modified your modules.conf to not use the autoload option, then you will need
to modify modules.conf to add another "noload" line to ensure that only one of
these three modules gets loaded.
* DAHDI: The chan_zap module that supported PSTN interfaces using
Zaptel has been renamed to chan_dahdi, and only supports the DAHDI
telephony driver package for PSTN interfaces. See the
Zaptel-to-DAHDI.txt file for more details on this transition.
* DAHDI: The "msdstrip" option has been deprecated, as it provides no value over
the method of stripping digits in the dialplan using variable substring syntax.
Configuration:
* pbx_dundi.c: tos parameter changed to use new values. Old values like lowdelay,
lowcost and other is not acceptable now. Look into qos.tex for description of
this parameter.
* queues.conf: the queue-lessthan sound file option is no longer available, and the
queue-round-seconds option no longer takes '1' as a valid parameter.
Manager:
* Manager has been upgraded to version 1.1 with a lot of changes.
Please check doc/manager_1_1.txt for information
* The IAXpeers command output has been changed to more closely resemble the
output of the SIPpeers command.
* cdr_manager now reports at the "cdr" level, not at "call" You may need to
change your manager.conf to add the level to existing AMI users, if they
want to see the CDR events generated.
* The Originate command now requires the Originate write permission. For
Originate with the Application parameter, you need the additional System
privilege if you want to do anything that calls out to a subshell.
iLBC Codec:
* Previously, the Asterisk source code distribution included the iLBC
encoder/decoder source code, from Global IP Solutions
(http://www.gipscorp.com). This code is not licensed for
distribution, and thus has been removed from the Asterisk source
code distribution. If you wish to use codec_ilbc to support iLBC
channels in Asterisk, you can run the contrib/scripts/get_ilbc_source.sh
script to download the source and put it in the proper place in
the Asterisk build tree. Once that is done you can follow your normal
steps of building Asterisk. You will need to run 'menuselect' and enable
the iLBC codec in the 'Codec Translators' category.

View File

@@ -1,186 +1,60 @@
=========================================================
=== Information for upgrading from Asterisk 1.4 to 1.6
===========================================================
===
=== Information for upgrading between Asterisk 1.6 versions
===
=== These files document all the changes that MUST be taken
=== into account when upgrading between the Asterisk
=== versions listed below. These changes may require that
=== you modify your configuration files, dialplan or (in
=== some cases) source code if you have your own Asterisk
=== modules or patches. These files also includes advance
=== notice of any functionality that has been marked as
=== 'deprecated' and may be removed in a future release,
=== along with the suggested replacement functionality.
===
=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
=== UPGRADE.txt -- Upgrade info for 1.4 to 1.6
=========================================================
=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
===
===========================================================
AEL:
From 1.6.0.10 to 1.6.0.11:
* Macros are now implemented underneath with the Gosub() application.
Heaven Help You if you wrote code depending on any aspect of this!
Previous to 1.6, macros were implemented with the Macro() app, which
provided a nice feature of auto-returning. The compiler will do its
best to insert a Return() app call at the end of your macro if you did
not include it, but really, you should make sure that all execution
paths within your macros end in "return;".
* Beginning with this release, Asterisk's internal methods of
negotiating T.38 (FAX over IP) sessions changed in
non-backwards-compatible ways. Any applications that previously used
AST_CONTROL_T38 control frames will have to be upgraded to use
AST_CONTROL_T38_PARAMETERS control frames instead; app_fax.c is a good
example of how to generate and respond to these frames. These changes
were made to solve significant T.38 interoperability problems between
Asterisk and various SIP/T.38 endpoints identified by many users of
Asterisk.
* The conf2ael program is 'introduced' in this release; it is in a rather
crude state, but deemed useful for making a first pass at converting
extensions.conf code into AEL. More intelligence will come with time.
From 1.6.0.9 to 1.6.0.10:
Core:
* Support for Taiwanese was incorrectly supported with the "tw" language code.
In reality, the "tw" language code is reserved for the Twi language, native
to Ghana. If you were previously using the "tw" language code, you should
switch to using either "zh" (for Mandarin Chinese) or "zh_TW" for Taiwan
specific localizations. Additionally, "mx" should be changed to "es_MX",
Georgian was incorrectly specified as "ge" but should be "ka", and Czech is
"cs", not "cz".
* The 'languageprefix' option in asterisk.conf is now deprecated, and
the default sound file layout for non-English sounds is the 'new
style' layout introduced in Asterisk 1.4 (and used by the automatic
sound file installer in the Makefile).
From 1.6.0.1 to 1.6.0.2:
* The ast_expr2 stuff has been modified to handle floating-point numbers.
Numbers of the format D.D are now acceptable input for the expr parser,
Where D is a string of base-10 digits. All math is now done in "long double",
if it is available on your compiler/architecture. This was half-way between
a bug-fix (because the MATH func returns fp by default), and an enhancement.
Also, for those counting on, or needing, integer operations, a series of
'functions' were also added to the expr language, to allow several styles
of rounding/truncation, along with a set of common floating point operations,
like sin, cos, tan, log, pow, etc. The ability to call external functions
like CDR(), etc. was also added, without having to use the ${...} notation.
* The delimiter passed to applications has been changed to the comma (','), as
that is what people are used to using within extensions.conf. If you are
using realtime extensions, you will need to translate your existing dialplan
to use this separator. To use a literal comma, you need merely to escape it
with a backslash ('\'). Another possible side effect is that you may need to
remove the obscene level of backslashing that was necessary for the dialplan
to work correctly in 1.4 and previous versions. This should make writing
dialplans less painful in the future, albeit with the pain of a one-time
conversion.
* The ast_agi_register_multiple() and ast_agi_unregister_multiple()
API calls were added in 1.6.0, so that modules that provide multiple
AGI commands could register/unregister them all with a single
step. However, these API calls were not implemented properly, and did
not allow the caller to know whether registration or unregistration
succeeded or failed. They have been redefined to now return success
or failure, but this means any code using these functions will need
be recompiled after upgrading to a version of Asterisk containing
these changes. In addition, the source code using these functions
should be reviewed to ensure it can properly react to failure
of registration or unregistration of its API commands.
* The logger.conf option 'rotatetimestamp' has been deprecated in favor of
'rotatestrategy'. This new option supports a 'rotate' strategy that more
closely mimics the system logger in terms of file rotation.
* The concise versions of various CLI commands are now deprecated. We recommend
using the manager interface (AMI) for application integration with Asterisk.
Voicemail:
* The voicemail configuration values 'maxmessage' and 'minmessage' have
been changed to 'maxsecs' and 'minsecs' to clarify their purpose and
to make them more distinguishable from 'maxmsgs', which sets folder
size. The old variables will continue to work in this version, albeit
with a deprecation warning.
* If you use any interface for modifying voicemail aside from the built in
dialplan applications, then the option "pollmailboxes" *must* be set in
voicemail.conf for message waiting indication (MWI) to work properly. This
is because Voicemail notification is now event based instead of polling
based. The channel drivers are no longer responsible for constantly manually
checking mailboxes for changes so that they can send MWI information to users.
Examples of situations that would require this option are web interfaces to
voicemail or an email client in the case of using IMAP storage.
Applications:
* ChanIsAvail() now has a 't' option, which allows the specified device
to be queried for state without consulting the channel drivers. This
performs mostly a 'ChanExists' sort of function.
* ChannelRedirect() will not terminate the channel that fails to do a
channelredirect as it has done previously. Instead CHANNELREDIRECT_STATUS
will reflect if the attempt was successful of not.
* SetCallerPres() has been replaced with the CALLERPRES() dialplan function
and is now deprecated.
* DISA()'s fifth argument is now an options argument. If you have previously
used 'NOANSWER' in this argument, you'll need to convert that to the new
option 'n'.
* Macro() is now deprecated. If you need subroutines, you should use the
Gosub()/Return() applications. To replace MacroExclusive(), we have
introduced dialplan functions LOCK(), TRYLOCK(), and UNLOCK(). You may use
these functions in any location where you desire to ensure that only one
channel is executing that path at any one time.
* Read() now sets a READSTATUS variable on exit. It does NOT automatically
return -1 (and hangup) anymore on error. If you want to hangup on error,
you need to do so explicitly in your dialplan.
* Privacy() no longer uses privacy.conf, so any options must be specified
directly in the application arguments.
* MusicOnHold application now has duration parameter which allows specifying
timeout in seconds.
* WaitMusicOnHold application is now deprecated in favor of extended MusicOnHold.
* SetMusicOnHold is now deprecated. You should use Set(CHANNEL(musicclass)=...)
instead.
Dialplan Functions:
* QUEUE_MEMBER_COUNT() has been deprecated in favor of the QUEUE_MEMBER() function. For
more information, issue a "show function QUEUE_MEMBER" from the CLI.
CDR:
* The cdr_sqlite module has been marked as deprecated in favor of
cdr_sqlite3_custom. It will potentially be removed from the tree
after Asterisk 1.6 is released.
* The cdr_odbc module now uses res_odbc to manage its connections. The
username and password parameters in cdr_odbc.conf, therefore, are no
longer used. The dsn parameter now points to an entry in res_odbc.conf.
Formats:
* format_wav: The GAIN preprocessor definition and source code that used it
is removed. This change was made in response to user complaints of
choppiness or the clipping of loud signal peaks. To increase the volume
of voicemail messages, use the 'volgain' option in voicemail.conf
Channel Drivers:
* SIP: a small upgrade to support the "Record" button on the SNOM360,
which sends a sip INFO message with a "Record: on" or "Record: off"
header. If Asterisk is set up (via features.conf) to accept "One Touch Monitor"
requests (by default, via '*1'), then the user-configured dialpad sequence
is generated, and recording can be started and stopped via this button. The
file names and formats are all controlled via the normal mechanisms. If the
user has not configured the automon feature, the normal "415 Unsupported media type"
is returned, and nothing is done.
* SIP: The "call-limit" option is marked as deprecated. It still works in this version of
Asterisk, but will be removed in the following version. Please use the groupcount functions
in the dialplan to enforce call limits. The "limitonpeer" configuration option is
now renamed to "counteronpeer".
* SIP: The "username" option is now renamed to "defaultuser" to match "defaultip".
These are used only before registration to call a peer with the uri
sip:defaultuser@defaultip
The "username" setting still work, but is deprecated and will not work in
the next version of Asterisk.
* chan_local.c: the comma delimiter inside the channel name has been changed to a
semicolon, in order to make the Local channel driver compatible with the comma
delimiter change in applications.
* H323: The "tos" setting has changed name to "tos_audio" and "cos" to "cos_audio"
to be compatible with settings in sip.conf. The "tos" and "cos" configuration
is deprecated and will stop working in the next release of Asterisk.
* Console: A new console channel driver, chan_console, has been added to Asterisk.
This new module can not be loaded at the same time as chan_alsa or chan_oss. The
default modules.conf only loads one of them (chan_oss by default). So, unless you
have modified your modules.conf to not use the autoload option, then you will need
to modify modules.conf to add another "noload" line to ensure that only one of
these three modules gets loaded.
* Zap: The "msdstrip" option has been deprecated, as it provides no value over
the method of stripping digits in the dialplan using variable substring syntax.
Configuration:
* pbx_dundi.c: tos parameter changed to use new values. Old values like lowdelay,
lowcost and other is not acceptable now. Look into qos.tex for description of
this parameter.
* queues.conf: the queue-lessthan sound file option is no longer available, and the
queue-round-seconds option no longer takes '1' as a valid parameter.
Manager:
* Manager has been upgraded to version 1.1 with a lot of changes.
Please check doc/manager_1_1.txt for information
* The IAXpeers command output has been changed to more closely resemble the
output of the SIPpeers command.
* cdr_manager now reports at the "cdr" level, not at "call" You may need to
change your manager.conf to add the level to existing AMI users, if they
want to see the CDR events generated.
* The Originate command now requires the Originate write permission. For
Originate with the Application parameter, you need the additional System
privilege if you want to do anything that calls out to a subshell.
* The ast_agi_fdprintf() API call has been renamed to ast_agi_send()
to better match what it really does, and the argument order has been
changed to be consistent with other API calls that perform similar
operations.

53
Zaptel-to-DAHDI.txt Normal file
View File

@@ -0,0 +1,53 @@
=========================================================
=== Information for upgrading from Zaptel to DAHDI ===
=========================================================
As announced in early 2008, Digium is renaming the Zaptel telephony
interface project to DAHDI (Digium Asterisk Hardware Device Interface)
to accommodate the desires of the owner of the Zaptel trademark for
telephony purposes.
This version of Asterisk can only be built using DAHDI, and as a
result there are number of changes that will be visible to previous
users of Asterisk with Zaptel.
First, the modules that directly use services from DAHDI have been
renamed; the new names are:
chan_zap.so -> chan_dahdi.so
app_zapbarge.so -> app_dahdibarge.so
app_zapras.so -> app_dahdiras.so
app_zapscan.so -> app_dahdiscan.so
codec_zap.so -> codec_dahdi.so
Second, the behavior of many modules has changed due to the switch to
DAHDI; the changes are listed below.
chan_dahdi.so:
Incoming and outgoing channels managed by this module will be
'DAHDI' channels instead of 'Zap' channels.
All CLI commands that began with 'zap' have been renamed to 'dahdi'
commands.
All Asterisk Manager Interface (AMI) actions that began with 'Zap'
have changed to 'DAHDI' prefixes.
The ZapSendKeypadFacility dialplan application has been renamed to
DAHDISendKeypadFacility.
The configuration for the channel driver will be read from
/etc/asterisk/chan_dahdi.conf instead of /etc/asterisk/zapata.conf.
app_dahdibarge.so:
The ZapBarge application has been renamed to DAHDIBarge.
app_dahdiras.so:
The ZapRAS application has been renamed to DAHDIRAS.
app_dahdiscan.so:
The ZapScan application has been renamed to DAHDIScan.

File diff suppressed because it is too large Load Diff

View File

@@ -25,6 +25,8 @@ endif
include $(ASTTOPDIR)/Makefile.rules
_ASTCFLAGS+=-DSTANDALONE
all: $(AGIS)
strcompat.c: ../main/strcompat.c
@@ -43,8 +45,7 @@ uninstall:
clean:
rm -f *.so *.o look eagi-test eagi-sphinx-test
rm -f .*.o.d .*.oo.d
rm -f *.s *.i
rm -f .*.d *.s *.i
rm -f strcompat.c
ifneq ($(wildcard .*.d),)

View File

@@ -79,7 +79,9 @@ static int read_environment(void)
char *val;
/* Read environment */
for(;;) {
fgets(buf, sizeof(buf), stdin);
if (!fgets(buf, sizeof(buf), stdin)) {
return -1;
}
if (feof(stdin))
return -1;
buf[strlen(buf) - 1] = '\0';
@@ -130,7 +132,9 @@ static char *wait_result(void)
return NULL;
}
if (FD_ISSET(STDIN_FILENO, &fds)) {
fgets(astresp, sizeof(astresp), stdin);
if (!fgets(astresp, sizeof(astresp), stdin)) {
return NULL;
}
if (feof(stdin)) {
fprintf(stderr, "Got hungup on apparently\n");
return NULL;
@@ -141,9 +145,10 @@ static char *wait_result(void)
}
if (FD_ISSET(AUDIO_FILENO, &fds)) {
res = read(AUDIO_FILENO, audiobuf, sizeof(audiobuf));
if (res > 0) {
if (sphinx_sock > -1)
write(sphinx_sock, audiobuf, res);
if ((res > 0) && (sphinx_sock > -1)) {
if (write(sphinx_sock, audiobuf, res) < 0) {
fprintf(stderr, "write() failed: %s\n", strerror(errno));
}
}
}
if ((sphinx_sock > -1) && FD_ISSET(sphinx_sock, &fds)) {

View File

@@ -24,7 +24,9 @@ static int read_environment(void)
char *val;
/* Read environment */
for(;;) {
fgets(buf, sizeof(buf), stdin);
if (!fgets(buf, sizeof(buf), stdin)) {
return -1;
}
if (feof(stdin))
return -1;
buf[strlen(buf) - 1] = '\0';
@@ -68,7 +70,9 @@ static char *wait_result(void)
return NULL;
}
if (FD_ISSET(STDIN_FILENO, &fds)) {
fgets(astresp, sizeof(astresp), stdin);
if (!fgets(astresp, sizeof(astresp), stdin)) {
return NULL;
}
if (feof(stdin)) {
fprintf(stderr, "Got hungup on apparently\n");
return NULL;

View File

@@ -22,12 +22,6 @@ ifneq ($(findstring ODBC_STORAGE,$(MENUSELECT_OPTS_app_voicemail)),)
endif
ifneq ($(findstring IMAP_STORAGE,$(MENUSELECT_OPTS_app_voicemail)),)
MENUSELECT_DEPENDS_app_voicemail+=$(MENUSELECT_DEPENDS_IMAP_STORAGE)
MENUSELECT_DEPENDS_app_directory+=$(MENUSELECT_DEPENDS_IMAP_STORAGE)
endif
ifeq (SunOS,$(shell uname))
MENUSELECT_DEPENDS_app_chanspy+=RT
RT_LIB=-lrt
endif
all: _all

View File

@@ -515,7 +515,7 @@ static int geteventbyname(char *name)
{
int x;
for (x = 0; x < sizeof(events) / sizeof(events[0]); x++) {
for (x = 0; x < ARRAY_LEN(events); x++) {
if (!strcasecmp(events[x].name, name))
return events[x].id;
}
@@ -527,7 +527,7 @@ static int getjustifybyname(char *name)
{
int x;
for (x = 0; x <sizeof(justify) / sizeof(justify[0]); x++) {
for (x = 0; x < ARRAY_LEN(justify); x++) {
if (!strcasecmp(justify[x].name, name))
return justify[x].id;
}
@@ -935,7 +935,7 @@ static int process_returncode(struct adsi_soft_key *key, char *code, char *args,
int x, res;
char *unused;
for (x = 0; x < sizeof(kcmds) / sizeof(kcmds[0]); x++) {
for (x = 0; x < ARRAY_LEN(kcmds); x++) {
if ((kcmds[x].id > -1) && !strcasecmp(kcmds[x].name, code)) {
if (kcmds[x].add_args) {
res = kcmds[x].add_args(key->retstr + key->retstrlen,
@@ -964,7 +964,7 @@ static int process_opcode(struct adsi_subscript *sub, char *code, char *args, st
int x, res, max = sub->id ? MAX_SUB_LEN : MAX_MAIN_LEN;
char *unused;
for (x = 0; x < sizeof(opcmds) / sizeof(opcmds[0]); x++) {
for (x = 0; x < ARRAY_LEN(opcmds); x++) {
if ((opcmds[x].id > -1) && !strcasecmp(opcmds[x].name, code)) {
if (opcmds[x].add_args) {
res = opcmds[x].add_args(sub->data + sub->datalen,
@@ -1369,7 +1369,9 @@ static struct adsi_script *compile_script(char *script)
/* Create "main" as first subroutine */
getsubbyname(scr, "main", NULL, 0);
while (!feof(f)) {
fgets(buf, sizeof(buf), f);
if (!fgets(buf, sizeof(buf), f)) {
continue;
}
if (!feof(f)) {
lineno++;
/* Trim off trailing return */

View File

@@ -126,9 +126,7 @@ static int auth_exec(struct ast_channel *chan, void *data)
/* Compare against a fixed password */
if (!strcmp(passwd, arglist.password))
break;
}
if (ast_test_flag(&flags,OPT_DATABASE)) {
} else if (ast_test_flag(&flags,OPT_DATABASE)) {
char tmp[256];
/* Compare against a database key */
if (!ast_db_get(arglist.password + 1, passwd, tmp, sizeof(tmp))) {
@@ -153,7 +151,9 @@ static int auth_exec(struct ast_channel *chan, void *data)
if (feof(f))
break;
fgets(buf, sizeof(buf), f);
if (!fgets(buf, sizeof(buf), f)) {
continue;
}
if (ast_strlen_zero(buf))
continue;

View File

@@ -71,7 +71,7 @@ static int chanavail_exec(struct ast_channel *chan, void *data)
);
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "ChanIsAvail requires an argument (Zap/1&Zap/2)\n");
ast_log(LOG_WARNING, "ChanIsAvail requires an argument (DAHDI/1&DAHDI/2)\n");
return -1;
}

View File

@@ -75,6 +75,9 @@ static int asyncgoto_exec(struct ast_channel *chan, void *data)
return 0;
}
if (chan2->pbx) {
ast_set_flag(chan2, AST_FLAG_BRIDGE_HANGUP_DONT); /* don't let the after-bridge code run the h-exten */
}
res = ast_async_parseable_goto(chan2, args.label);
pbx_builtin_setvar_helper(chan, "CHANNELREDIRECT_STATUS", "SUCCESS");
ast_channel_unlock(chan2);

View File

@@ -2,7 +2,7 @@
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2005 Anthony Minessale II (anthmct@yahoo.com)
* Copyright (C) 2005 - 2006, Digium, Inc.
* Copyright (C) 2005 - 2008, Digium, Inc.
*
* A license has been granted to Digium (via disclaimer) for the use of
* this code.
@@ -23,6 +23,8 @@
* \brief ChanSpy: Listen in on any channel.
*
* \author Anthony Minessale II <anthmct@yahoo.com>
* \author Joshua Colp <jcolp@digium.com>
* \author Russell Bryant <russell@digium.com>
*
* \ingroup applications
*/
@@ -32,6 +34,7 @@
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include <ctype.h>
#include <errno.h>
#include "asterisk/paths.h" /* use ast_config_AST_MONITOR_DIR */
#include "asterisk/file.h"
@@ -45,6 +48,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/translate.h"
#include "asterisk/module.h"
#include "asterisk/lock.h"
#include "asterisk/options.h"
#define AST_NAME_STRLEN 256
@@ -164,6 +168,7 @@ AST_APP_OPTIONS(spy_opts, {
AST_APP_OPTION('X', OPTION_EXIT),
});
static int next_unique_id_to_use = 0;
struct chanspy_translation_helper {
/* spy data */
@@ -187,7 +192,7 @@ static void spy_release(struct ast_channel *chan, void *data)
static int spy_generate(struct ast_channel *chan, void *data, int len, int samples)
{
struct chanspy_translation_helper *csth = data;
struct ast_frame *f = NULL;
struct ast_frame *f, *cur;
ast_audiohook_lock(&csth->spy_audiohook);
if (csth->spy_audiohook.status != AST_AUDIOHOOK_STATUS_RUNNING) {
@@ -203,13 +208,18 @@ static int spy_generate(struct ast_channel *chan, void *data, int len, int sampl
if (!f)
return 0;
if (ast_write(chan, f)) {
ast_frfree(f);
return -1;
}
for (cur = f; cur; cur = AST_LIST_NEXT(cur, frame_list)) {
if (ast_write(chan, cur)) {
ast_frfree(f);
return -1;
}
if (csth->fd)
write(csth->fd, f->data, f->datalen);
if (csth->fd) {
if (write(csth->fd, cur->data, cur->datalen) < 0) {
ast_log(LOG_WARNING, "write() failed: %s\n", strerror(errno));
}
}
}
ast_frfree(f);
@@ -229,19 +239,18 @@ static int start_spying(struct ast_channel *chan, const char *spychan_name, stru
ast_log(LOG_NOTICE, "Attaching %s to %s\n", spychan_name, chan->name);
ast_set_flag(audiohook, AST_AUDIOHOOK_TRIGGER_SYNC | AST_AUDIOHOOK_SMALL_QUEUE);
res = ast_audiohook_attach(chan, audiohook);
if (!res && ast_test_flag(chan, AST_FLAG_NBRIDGE) && (peer = ast_bridged_channel(chan))) {
ast_channel_unlock(chan);
if (!res && ast_test_flag(chan, AST_FLAG_NBRIDGE) && (peer = ast_bridged_channel(chan))) {
ast_softhangup(peer, AST_SOFTHANGUP_UNBRIDGE);
} else
ast_channel_unlock(chan);
}
return res;
}
struct chanspy_ds {
struct ast_channel *chan;
char unique_id[20];
ast_mutex_t lock;
};
@@ -273,8 +282,10 @@ static int channel_spy(struct ast_channel *chan, struct chanspy_ds *spyee_chansp
/* We now hold the channel lock on spyee */
if (ast_check_hangup(chan) || ast_check_hangup(spyee))
if (ast_check_hangup(chan) || ast_check_hangup(spyee)) {
ast_channel_unlock(spyee);
return 0;
}
name = ast_strdupa(spyee->name);
ast_verb(2, "Spying on channel %s\n", name);
@@ -283,16 +294,18 @@ static int channel_spy(struct ast_channel *chan, struct chanspy_ds *spyee_chansp
ast_audiohook_init(&csth.spy_audiohook, AST_AUDIOHOOK_TYPE_SPY, "ChanSpy");
if (start_spying(spyee, spyer_name, &csth.spy_audiohook)) { /* Unlocks spyee */
if (start_spying(spyee, spyer_name, &csth.spy_audiohook)) {
ast_audiohook_destroy(&csth.spy_audiohook);
ast_channel_unlock(spyee);
return 0;
}
if (ast_test_flag(flags, OPTION_WHISPER)) {
ast_audiohook_init(&csth.whisper_audiohook, AST_AUDIOHOOK_TYPE_WHISPER, "ChanSpy");
start_spying(spyee, spyer_name, &csth.whisper_audiohook); /* Unlocks spyee */
start_spying(spyee, spyer_name, &csth.whisper_audiohook);
}
ast_channel_unlock(spyee);
spyee = NULL;
csth.volfactor = *volfactor;
@@ -425,9 +438,19 @@ static void chanspy_ds_destroy(void *data)
ast_mutex_unlock(&chanspy_ds->lock);
}
static void chanspy_ds_chan_fixup(void *data, struct ast_channel *old_chan, struct ast_channel *new_chan)
{
struct chanspy_ds *chanspy_ds = data;
ast_mutex_lock(&chanspy_ds->lock);
chanspy_ds->chan = new_chan;
ast_mutex_unlock(&chanspy_ds->lock);
}
static const struct ast_datastore_info chanspy_ds_info = {
.type = "chanspy",
.destroy = chanspy_ds_destroy,
.chan_fixup = chanspy_ds_chan_fixup,
};
static struct chanspy_ds *chanspy_ds_free(struct chanspy_ds *chanspy_ds)
@@ -443,7 +466,7 @@ static struct chanspy_ds *chanspy_ds_free(struct chanspy_ds *chanspy_ds)
chan = chanspy_ds->chan;
ast_channel_lock(chan);
if ((datastore = ast_channel_datastore_find(chan, &chanspy_ds_info, NULL))) {
if ((datastore = ast_channel_datastore_find(chan, &chanspy_ds_info, chanspy_ds->unique_id))) {
ast_channel_datastore_remove(chan, datastore);
/* chanspy_ds->chan is NULL after this call */
chanspy_ds_destroy(datastore->data);
@@ -464,16 +487,15 @@ static struct chanspy_ds *setup_chanspy_ds(struct ast_channel *chan, struct chan
ast_mutex_lock(&chanspy_ds->lock);
chanspy_ds->chan = chan;
if (!(datastore = ast_channel_datastore_alloc(&chanspy_ds_info, NULL))) {
if (!(datastore = ast_channel_datastore_alloc(&chanspy_ds_info, chanspy_ds->unique_id))) {
ast_mutex_unlock(&chanspy_ds->lock);
chanspy_ds = chanspy_ds_free(chanspy_ds);
ast_channel_unlock(chan);
return NULL;
}
chanspy_ds->chan = chan;
datastore->data = chanspy_ds;
ast_channel_datastore_add(chan, datastore);
return chanspy_ds;
@@ -483,30 +505,32 @@ static struct chanspy_ds *next_channel(struct ast_channel *chan,
const struct ast_channel *last, const char *spec,
const char *exten, const char *context, struct chanspy_ds *chanspy_ds)
{
struct ast_channel *this;
struct ast_channel *next;
const size_t pseudo_len = strlen("DAHDI/pseudo");
redo:
if (!ast_strlen_zero(spec))
this = ast_walk_channel_by_name_prefix_locked(last, spec, strlen(spec));
next = ast_walk_channel_by_name_prefix_locked(last, spec, strlen(spec));
else if (!ast_strlen_zero(exten))
this = ast_walk_channel_by_exten_locked(last, exten, context);
next = ast_walk_channel_by_exten_locked(last, exten, context);
else
this = ast_channel_walk_locked(last);
next = ast_channel_walk_locked(last);
if (!this)
if (!next)
return NULL;
if (!strncmp(this->name, "Zap/pseudo", 10)) {
ast_channel_unlock(this);
if (!strncmp(next->name, "DAHDI/pseudo", pseudo_len)) {
last = next;
ast_channel_unlock(next);
goto redo;
} else if (this == chan) {
last = this;
ast_channel_unlock(this);
} else if (next == chan) {
last = next;
ast_channel_unlock(next);
goto redo;
}
return setup_chanspy_ds(this, chanspy_ds);
return setup_chanspy_ds(next, chanspy_ds);
}
static int common_exec(struct ast_channel *chan, const struct ast_flags *flags,
@@ -522,7 +546,7 @@ static int common_exec(struct ast_channel *chan, const struct ast_flags *flags,
char *ptr;
int num;
int num_spyed_upon = 1;
struct chanspy_ds chanspy_ds;
struct chanspy_ds chanspy_ds = { 0, };
if (ast_test_flag(flags, OPTION_EXIT)) {
const char *c;
@@ -536,6 +560,8 @@ static int common_exec(struct ast_channel *chan, const struct ast_flags *flags,
ast_mutex_init(&chanspy_ds.lock);
snprintf(chanspy_ds.unique_id, sizeof(chanspy_ds.unique_id), "%d", ast_atomic_fetchadd_int(&next_unique_id_to_use, +1));
if (chan->_state != AST_STATE_UP)
ast_answer(chan);
@@ -590,19 +616,9 @@ static int common_exec(struct ast_channel *chan, const struct ast_flags *flags,
chanspy_ds_free(peer_chanspy_ds), prev = peer,
peer_chanspy_ds = next_chanspy_ds ? next_chanspy_ds :
next_channel(chan, prev, spec, exten, context, &chanspy_ds), next_chanspy_ds = NULL) {
const char *group;
int igrp = !mygroup;
char *groups[25];
int num_groups = 0;
char *dup_group;
int x;
char *s;
char *buffer;
char *end;
char *ext;
char *form_enforced;
int ienf = !myenforced;
struct ast_channel *peer;
char *s;
peer = peer_chanspy_ds->chan;
@@ -631,10 +647,15 @@ static int common_exec(struct ast_channel *chan, const struct ast_flags *flags,
}
if (mygroup) {
int num_groups = 0;
char dup_group[512];
char *groups[25];
const char *group;
int x;
if ((group = pbx_builtin_getvar_helper(peer, "SPYGROUP"))) {
dup_group = ast_strdupa(group);
ast_copy_string(dup_group, group, sizeof(dup_group));
num_groups = ast_app_separate_args(dup_group, ':', groups,
sizeof(groups) / sizeof(groups[0]));
ARRAY_LEN(groups));
}
for (x = 0; x < num_groups; x++) {
@@ -651,64 +672,57 @@ static int common_exec(struct ast_channel *chan, const struct ast_flags *flags,
}
if (myenforced) {
char ext[AST_CHANNEL_NAME + 3];
char buffer[512];
char *end;
/* We don't need to allocate more space than just the
length of (peer->name) for ext as we will cut the
channel name's ending before copying into ext */
snprintf(buffer, sizeof(buffer) - 1, ":%s:", myenforced);
ext = alloca(strlen(peer->name));
form_enforced = alloca(strlen(myenforced) + 3);
strcpy(form_enforced, ":");
strcat(form_enforced, myenforced);
strcat(form_enforced, ":");
buffer = ast_strdupa(peer->name);
if ((end = strchr(buffer, '-'))) {
ast_copy_string(ext + 1, peer->name, sizeof(ext) - 1);
if ((end = strchr(ext, '-'))) {
*end++ = ':';
*end = '\0';
}
strcpy(ext, ":");
strcat(ext, buffer);
ext[0] = ':';
if (strcasestr(form_enforced, ext))
if (strcasestr(buffer, ext)) {
ienf = 1;
}
}
if (!ienf)
if (!ienf) {
continue;
}
strcpy(peer_name, "spy-");
strncat(peer_name, peer->name, AST_NAME_STRLEN);
strncat(peer_name, peer->name, AST_NAME_STRLEN - 4 - 1);
ptr = strchr(peer_name, '/');
*ptr++ = '\0';
for (s = peer_name; s < ptr; s++)
*s = tolower(*s);
/* We have to unlock the peer channel here to avoid a deadlock.
* So, when we need it again, we have to lock the datastore and get
* the pointer from there to see if the channel is still valid. */
* So, when we need to dereference it again, we have to lock the
* datastore and get the pointer from there to see if the channel
* is still valid. */
ast_channel_unlock(peer);
peer = NULL;
if (!ast_test_flag(flags, OPTION_QUIET)) {
if (ast_fileexists(peer_name, NULL, NULL) != -1) {
res = ast_streamfile(chan, peer_name, chan->language);
if (!res)
res = ast_waitstream(chan, "");
if (res)
if (res) {
chanspy_ds_free(peer_chanspy_ds);
break;
}
} else
res = ast_say_character_str(chan, peer_name, "", chan->language);
if ((num = atoi(ptr)))
ast_say_digits(chan, atoi(ptr), "", chan->language);
}
waitms = 5000;
res = channel_spy(chan, peer_chanspy_ds, &volfactor, fd, flags, exitcontext);
num_spyed_upon++;
@@ -717,6 +731,7 @@ static int common_exec(struct ast_channel *chan, const struct ast_flags *flags,
goto exit;
} else if (res == -2) {
res = 0;
chanspy_ds_free(peer_chanspy_ds);
goto exit;
} else if (res > 1 && spec) {
struct ast_channel *next;
@@ -753,6 +768,8 @@ exit:
ast_channel_setoption(chan, AST_OPTION_TXGAIN, &zero_volume, sizeof(zero_volume), 0);
ast_mutex_lock(&chanspy_ds.lock);
ast_mutex_unlock(&chanspy_ds.lock);
ast_mutex_destroy(&chanspy_ds.lock);
return res;
@@ -814,7 +831,7 @@ static int chanspy_exec(struct ast_channel *chan, void *data)
}
if (recbase) {
char filename[512];
char filename[PATH_MAX];
snprintf(filename, sizeof(filename), "%s/%s.%d.raw", ast_config_AST_MONITOR_DIR, recbase, (int) time(NULL));
if ((fd = open(filename, O_CREAT | O_WRONLY | O_TRUNC, AST_FILE_MODE)) <= 0) {
@@ -893,7 +910,7 @@ static int extenspy_exec(struct ast_channel *chan, void *data)
}
if (recbase) {
char filename[512];
char filename[PATH_MAX];
snprintf(filename, sizeof(filename), "%s/%s.%d.raw", ast_config_AST_MONITOR_DIR, recbase, (int) time(NULL));
if ((fd = open(filename, O_CREAT | O_WRONLY | O_TRUNC, AST_FILE_MODE)) <= 0) {

View File

@@ -21,7 +21,7 @@
/*! \file
*
* \brief Zap Barge support
* \brief DAHDI Barge support
*
* \author Mark Spencer <markster@digium.com>
*
@@ -32,14 +32,15 @@
*/
/*** MODULEINFO
<depend>zaptel</depend>
<depend>dahdi</depend>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/zapata.h"
#include <sys/ioctl.h>
#include <dahdi/user.h>
#include "asterisk/lock.h"
#include "asterisk/file.h"
@@ -52,12 +53,12 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/say.h"
#include "asterisk/utils.h"
static char *app = "ZapBarge";
static char *app = "DAHDIBarge";
static char *synopsis = "Barge in (monitor) Zap channel";
static char *synopsis = "Barge in (monitor) DAHDI channel";
static char *descrip =
" ZapBarge([channel]): Barges in on a specified zap\n"
" DAHDIBarge([channel]): Barges in on a specified DAHDI\n"
"channel or prompts if one is not specified. Returns\n"
"-1 when caller user hangs up and is independent of the\n"
"state of the channel being monitored.";
@@ -86,7 +87,7 @@ static int careful_write(int fd, unsigned char *data, int len)
static int conf_run(struct ast_channel *chan, int confno, int confflags)
{
int fd;
struct zt_confinfo ztc;
struct dahdi_confinfo dahdic;
struct ast_frame *f;
struct ast_channel *c;
struct ast_frame fr;
@@ -95,11 +96,11 @@ static int conf_run(struct ast_channel *chan, int confno, int confflags)
int nfds;
int res;
int flags;
int retryzap;
int retrydahdi;
int origfd;
int ret = -1;
ZT_BUFFERINFO bi;
struct dahdi_bufferinfo bi;
char __buf[CONF_SIZE + AST_FRIENDLY_OFFSET];
char *buf = __buf + AST_FRIENDLY_OFFSET;
@@ -115,11 +116,11 @@ static int conf_run(struct ast_channel *chan, int confno, int confflags)
goto outrun;
}
ast_indicate(chan, -1);
retryzap = strcasecmp(chan->tech->type, "Zap");
zapretry:
retrydahdi = strcasecmp(chan->tech->type, "DAHDI");
dahdiretry:
origfd = chan->fds[0];
if (retryzap) {
fd = open("/dev/zap/pseudo", O_RDWR);
if (retrydahdi) {
fd = open("/dev/dahdi/pseudo", O_RDWR);
if (fd < 0) {
ast_log(LOG_WARNING, "Unable to open pseudo channel: %s\n", strerror(errno));
goto outrun;
@@ -139,10 +140,10 @@ zapretry:
/* Setup buffering information */
memset(&bi, 0, sizeof(bi));
bi.bufsize = CONF_SIZE;
bi.txbufpolicy = ZT_POLICY_IMMEDIATE;
bi.rxbufpolicy = ZT_POLICY_IMMEDIATE;
bi.txbufpolicy = DAHDI_POLICY_IMMEDIATE;
bi.rxbufpolicy = DAHDI_POLICY_IMMEDIATE;
bi.numbufs = 4;
if (ioctl(fd, ZT_SET_BUFINFO, &bi)) {
if (ioctl(fd, DAHDI_SET_BUFINFO, &bi)) {
ast_log(LOG_WARNING, "Unable to set buffering information: %s\n", strerror(errno));
close(fd);
goto outrun;
@@ -153,34 +154,34 @@ zapretry:
fd = chan->fds[0];
nfds = 0;
}
memset(&ztc, 0, sizeof(ztc));
memset(&dahdic, 0, sizeof(dahdic));
/* Check to see if we're in a conference... */
ztc.chan = 0;
if (ioctl(fd, ZT_GETCONF, &ztc)) {
dahdic.chan = 0;
if (ioctl(fd, DAHDI_GETCONF, &dahdic)) {
ast_log(LOG_WARNING, "Error getting conference\n");
close(fd);
goto outrun;
}
if (ztc.confmode) {
if (dahdic.confmode) {
/* Whoa, already in a conference... Retry... */
if (!retryzap) {
ast_debug(1, "Zap channel is in a conference already, retrying with pseudo\n");
retryzap = 1;
goto zapretry;
if (!retrydahdi) {
ast_debug(1, "DAHDI channel is in a conference already, retrying with pseudo\n");
retrydahdi = 1;
goto dahdiretry;
}
}
memset(&ztc, 0, sizeof(ztc));
memset(&dahdic, 0, sizeof(dahdic));
/* Add us to the conference */
ztc.chan = 0;
ztc.confno = confno;
ztc.confmode = ZT_CONF_MONITORBOTH;
dahdic.chan = 0;
dahdic.confno = confno;
dahdic.confmode = DAHDI_CONF_MONITORBOTH;
if (ioctl(fd, ZT_SETCONF, &ztc)) {
if (ioctl(fd, DAHDI_SETCONF, &dahdic)) {
ast_log(LOG_WARNING, "Error setting conference\n");
close(fd);
goto outrun;
}
ast_debug(1, "Placed channel %s in ZAP channel %d monitor\n", chan->name, confno);
ast_debug(1, "Placed channel %s in DAHDI channel %d monitor\n", chan->name, confno);
for(;;) {
outfd = -1;
@@ -188,13 +189,13 @@ zapretry:
c = ast_waitfor_nandfds(&chan, 1, &fd, nfds, NULL, &outfd, &ms);
if (c) {
if (c->fds[0] != origfd) {
if (retryzap) {
if (retrydahdi) {
/* Kill old pseudo */
close(fd);
}
ast_debug(1, "Ooh, something swapped out under us, starting over\n");
retryzap = 0;
goto zapretry;
retrydahdi = 0;
goto dahdiretry;
}
f = ast_read(c);
if (!f)
@@ -236,10 +237,10 @@ zapretry:
else {
/* Take out of conference */
/* Add us to the conference */
ztc.chan = 0;
ztc.confno = 0;
ztc.confmode = 0;
if (ioctl(fd, ZT_SETCONF, &ztc)) {
dahdic.chan = 0;
dahdic.confno = 0;
dahdic.confmode = 0;
if (ioctl(fd, DAHDI_SETCONF, &dahdic)) {
ast_log(LOG_WARNING, "Error setting conference\n");
}
}
@@ -258,9 +259,9 @@ static int conf_exec(struct ast_channel *chan, void *data)
char confstr[80] = "";
if (!ast_strlen_zero(data)) {
if ((sscanf(data, "Zap/%d", &confno) != 1) &&
if ((sscanf(data, "DAHDI/%d", &confno) != 1) &&
(sscanf(data, "%d", &confno) != 1)) {
ast_log(LOG_WARNING, "ZapBarge Argument (if specified) must be a channel number, not '%s'\n", (char *)data);
ast_log(LOG_WARNING, "DAHDIBarge Argument (if specified) must be a channel number, not '%s'\n", (char *)data);
return 0;
}
}
@@ -296,4 +297,4 @@ static int load_module(void)
return ((ast_register_application(app, conf_exec, synopsis, descrip)) ? AST_MODULE_LOAD_FAILURE : AST_MODULE_LOAD_SUCCESS);
}
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Barge in on Zap channel application");
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Barge in on DAHDI channel application");

View File

@@ -26,7 +26,8 @@
*/
/*** MODULEINFO
<depend>zaptel</depend>
<depend>dahdi</depend>
<depend>working_fork</depend>
***/
#include "asterisk.h"
@@ -41,9 +42,12 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include <signal.h>
#endif /* __linux__ */
#include <fcntl.h>
#include <dahdi/user.h>
#include "asterisk/zapata.h"
#include <fcntl.h>
#ifdef HAVE_CAP
#include <sys/capability.h>
#endif /* HAVE_CAP */
#include "asterisk/lock.h"
#include "asterisk/file.h"
@@ -51,15 +55,15 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/pbx.h"
#include "asterisk/module.h"
static char *app = "ZapRAS";
static char *app = "DAHDIRAS";
static char *synopsis = "Executes Zaptel ISDN RAS application";
static char *synopsis = "Executes DAHDI ISDN RAS application";
static char *descrip =
" ZapRAS(args): Executes a RAS server using pppd on the given channel.\n"
"The channel must be a clear channel (i.e. PRI source) and a Zaptel\n"
" DAHDIRAS(args): Executes a RAS server using pppd on the given channel.\n"
"The channel must be a clear channel (i.e. PRI source) and a DAHDI\n"
"channel to be able to use this function (No modem emulation is included).\n"
"Your pppd must be patched to be zaptel aware. Arguments should be\n"
"Your pppd must be patched to be dahdi aware. Arguments should be\n"
"separated by , characters.\n";
@@ -76,6 +80,9 @@ static pid_t spawn_ras(struct ast_channel *chan, char *args)
int argc = 0;
char *stringp=NULL;
sigset_t fullset, oldset;
#ifdef HAVE_CAP
cap_t cap;
#endif
sigfillset(&fullset);
pthread_sigmask(SIG_BLOCK, &fullset, &oldset);
@@ -87,6 +94,16 @@ static pid_t spawn_ras(struct ast_channel *chan, char *args)
return pid;
}
#ifdef HAVE_CAP
cap = cap_from_text("cap_net_admin-eip");
if (cap_set_proc(cap)) {
/* Careful with order! Logging cannot happen after we close FDs */
ast_log(LOG_WARNING, "Unable to remove capabilities.\n");
}
cap_free(cap);
#endif
/* Restore original signal handlers */
for (x=0;x<NSIG;x++)
signal(x, SIG_DFL);
@@ -108,7 +125,7 @@ static pid_t spawn_ras(struct ast_channel *chan, char *args)
memset(argv, 0, sizeof(argv));
/* First argument is executable, followed by standard
arguments for zaptel PPP */
arguments for dahdi PPP */
argv[argc++] = PPP_EXEC;
argv[argc++] = "nodetach";
@@ -121,7 +138,7 @@ static pid_t spawn_ras(struct ast_channel *chan, char *args)
}
argv[argc++] = "plugin";
argv[argc++] = "zaptel.so";
argv[argc++] = "dahdi.so";
argv[argc++] = "stdin";
/* Finally launch PPP */
@@ -136,10 +153,10 @@ static void run_ras(struct ast_channel *chan, char *args)
int status;
int res;
int signalled = 0;
struct zt_bufferinfo savebi;
struct dahdi_bufferinfo savebi;
int x;
res = ioctl(chan->fds[0], ZT_GET_BUFINFO, &savebi);
res = ioctl(chan->fds[0], DAHDI_GET_BUFINFO, &savebi);
if(res) {
ast_log(LOG_WARNING, "Unable to check buffer policy on channel %s\n", chan->name);
return;
@@ -165,19 +182,20 @@ static void run_ras(struct ast_channel *chan, char *args)
if (res < 0) {
ast_log(LOG_WARNING, "wait4 returned %d: %s\n", res, strerror(errno));
}
if (WIFEXITED(status))
if (WIFEXITED(status)) {
ast_verb(3, "RAS on %s terminated with status %d\n", chan->name, WEXITSTATUS(status));
else if (WIFSIGNALED(status))
ast_verb(3, "RAS on %s terminated with signal %d\n", chan->name, WTERMSIG(status));
else
ast_verbose(VERBOSE_PREFIX_3 "RAS on %s terminated weirdly.\n", chan->name);
} else if (WIFSIGNALED(status)) {
ast_verb(3, "RAS on %s terminated with signal %d\n",
chan->name, WTERMSIG(status));
} else {
ast_verb(3, "RAS on %s terminated weirdly.\n", chan->name);
}
/* Throw back into audio mode */
x = 1;
ioctl(chan->fds[0], ZT_AUDIOMODE, &x);
ioctl(chan->fds[0], DAHDI_AUDIOMODE, &x);
/* Restore saved values */
res = ioctl(chan->fds[0], ZT_SET_BUFINFO, &savebi);
res = ioctl(chan->fds[0], DAHDI_SET_BUFINFO, &savebi);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set buffer policy on channel %s\n", chan->name);
}
@@ -186,11 +204,11 @@ static void run_ras(struct ast_channel *chan, char *args)
}
}
static int zapras_exec(struct ast_channel *chan, void *data)
static int dahdiras_exec(struct ast_channel *chan, void *data)
{
int res=-1;
char *args;
ZT_PARAMS ztp;
struct dahdi_params dahdip;
if (!data)
data = "";
@@ -200,16 +218,16 @@ static int zapras_exec(struct ast_channel *chan, void *data)
/* Answer the channel if it's not up */
if (chan->_state != AST_STATE_UP)
ast_answer(chan);
if (strcasecmp(chan->tech->type, "Zap")) {
/* If it's not a zap channel, we're done. Wait a couple of
if (strcasecmp(chan->tech->type, "DAHDI")) {
/* If it's not a DAHDI channel, we're done. Wait a couple of
seconds and then hangup... */
ast_verb(2, "Channel %s is not a Zap channel\n", chan->name);
ast_verb(2, "Channel %s is not a DAHDI channel\n", chan->name);
sleep(2);
} else {
memset(&ztp, 0, sizeof(ztp));
if (ioctl(chan->fds[0], ZT_GET_PARAMS, &ztp)) {
ast_log(LOG_WARNING, "Unable to get zaptel parameters\n");
} else if (ztp.sigtype != ZT_SIG_CLEAR) {
memset(&dahdip, 0, sizeof(dahdip));
if (ioctl(chan->fds[0], DAHDI_GET_PARAMS, &dahdip)) {
ast_log(LOG_WARNING, "Unable to get DAHDI parameters\n");
} else if (dahdip.sigtype != DAHDI_SIG_CLEAR) {
ast_verb(2, "Channel %s is not a clear channel\n", chan->name);
} else {
/* Everything should be okay. Run PPP. */
@@ -228,8 +246,8 @@ static int unload_module(void)
static int load_module(void)
{
return ((ast_register_application(app, zapras_exec, synopsis, descrip)) ? AST_MODULE_LOAD_FAILURE : AST_MODULE_LOAD_SUCCESS);
return ((ast_register_application(app, dahdiras_exec, synopsis, descrip)) ? AST_MODULE_LOAD_FAILURE : AST_MODULE_LOAD_SUCCESS);
}
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Zaptel ISDN Remote Access Server");
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "DAHDI ISDN Remote Access Server");

View File

@@ -5,7 +5,7 @@
*
* Mark Spencer <markster@digium.com>
*
* Modified from app_zapbarge by David Troy <dave@toad.net>
* Modified from app_dahdibarge by David Troy <dave@toad.net>
*
* Special thanks to comphealth.com for sponsoring this
* GPL application.
@@ -23,7 +23,7 @@
/*! \file
*
* \brief Zap Scanner
* \brief DAHDI Scanner
*
* \author Mark Spencer <markster@digium.com>
*
@@ -31,14 +31,15 @@
*/
/*** MODULEINFO
<depend>zaptel</depend>
<depend>dahdi</depend>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/zapata.h"
#include <sys/ioctl.h>
#include <dahdi/user.h>
#include "asterisk/lock.h"
#include "asterisk/file.h"
@@ -51,22 +52,22 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/cli.h"
#include "asterisk/say.h"
static char *app = "ZapScan";
static char *app = "DAHDIScan";
static char *synopsis = "Scan Zap channels to monitor calls";
static char *synopsis = "Scan DAHDI channels to monitor calls";
static char *descrip =
" ZapScan([group]) allows a call center manager to monitor Zap channels in\n"
" DAHDIScan([group]) allows a call center manager to monitor DAHDI channels in\n"
"a convenient way. Use '#' to select the next channel and use '*' to exit\n"
"Limit scanning to a channel GROUP by setting the option group argument.\n";
#define CONF_SIZE 160
static struct ast_channel *get_zap_channel_locked(int num) {
static struct ast_channel *get_dahdi_channel_locked(int num) {
char name[80];
snprintf(name,sizeof(name),"Zap/%d-1",num);
snprintf(name, sizeof(name), "DAHDI/%d-1", num);
return ast_get_channel_by_name_locked(name);
}
@@ -91,7 +92,7 @@ static int careful_write(int fd, unsigned char *data, int len)
static int conf_run(struct ast_channel *chan, int confno, int confflags)
{
int fd;
struct zt_confinfo ztc;
struct dahdi_confinfo dahdic;
struct ast_frame *f;
struct ast_channel *c;
struct ast_frame fr;
@@ -100,13 +101,13 @@ static int conf_run(struct ast_channel *chan, int confno, int confflags)
int nfds;
int res;
int flags;
int retryzap;
int retrydahdi;
int origfd;
int ret = -1;
char input[4];
int ic=0;
ZT_BUFFERINFO bi;
struct dahdi_bufferinfo bi;
char __buf[CONF_SIZE + AST_FRIENDLY_OFFSET];
char *buf = __buf + AST_FRIENDLY_OFFSET;
@@ -122,11 +123,11 @@ static int conf_run(struct ast_channel *chan, int confno, int confflags)
goto outrun;
}
ast_indicate(chan, -1);
retryzap = strcasecmp(chan->tech->type, "Zap");
zapretry:
retrydahdi = strcasecmp(chan->tech->type, "DAHDI");
dahdiretry:
origfd = chan->fds[0];
if (retryzap) {
fd = open("/dev/zap/pseudo", O_RDWR);
if (retrydahdi) {
fd = open("/dev/dahdi/pseudo", O_RDWR);
if (fd < 0) {
ast_log(LOG_WARNING, "Unable to open pseudo channel: %s\n", strerror(errno));
goto outrun;
@@ -136,7 +137,7 @@ static int conf_run(struct ast_channel *chan, int confno, int confflags)
if (flags < 0) {
ast_log(LOG_WARNING, "Unable to get flags: %s\n", strerror(errno));
close(fd);
goto outrun;
goto outrun;
}
if (fcntl(fd, F_SETFL, flags | O_NONBLOCK)) {
ast_log(LOG_WARNING, "Unable to set flags: %s\n", strerror(errno));
@@ -146,48 +147,48 @@ static int conf_run(struct ast_channel *chan, int confno, int confflags)
/* Setup buffering information */
memset(&bi, 0, sizeof(bi));
bi.bufsize = CONF_SIZE;
bi.txbufpolicy = ZT_POLICY_IMMEDIATE;
bi.rxbufpolicy = ZT_POLICY_IMMEDIATE;
bi.txbufpolicy = DAHDI_POLICY_IMMEDIATE;
bi.rxbufpolicy = DAHDI_POLICY_IMMEDIATE;
bi.numbufs = 4;
if (ioctl(fd, ZT_SET_BUFINFO, &bi)) {
if (ioctl(fd, DAHDI_SET_BUFINFO, &bi)) {
ast_log(LOG_WARNING, "Unable to set buffering information: %s\n", strerror(errno));
close(fd);
goto outrun;
}
nfds = 1;
nfds = 1;
} else {
/* XXX Make sure we're not running on a pseudo channel XXX */
fd = chan->fds[0];
nfds = 0;
}
memset(&ztc, 0, sizeof(ztc));
memset(&dahdic, 0, sizeof(dahdic));
/* Check to see if we're in a conference... */
ztc.chan = 0;
if (ioctl(fd, ZT_GETCONF, &ztc)) {
dahdic.chan = 0;
if (ioctl(fd, DAHDI_GETCONF, &dahdic)) {
ast_log(LOG_WARNING, "Error getting conference\n");
close(fd);
goto outrun;
}
if (ztc.confmode) {
if (dahdic.confmode) {
/* Whoa, already in a conference... Retry... */
if (!retryzap) {
ast_debug(1, "Zap channel is in a conference already, retrying with pseudo\n");
retryzap = 1;
goto zapretry;
if (!retrydahdi) {
ast_debug(1, "DAHDI channel is in a conference already, retrying with pseudo\n");
retrydahdi = 1;
goto dahdiretry;
}
}
memset(&ztc, 0, sizeof(ztc));
memset(&dahdic, 0, sizeof(dahdic));
/* Add us to the conference */
ztc.chan = 0;
ztc.confno = confno;
ztc.confmode = ZT_CONF_MONITORBOTH;
dahdic.chan = 0;
dahdic.confno = confno;
dahdic.confmode = DAHDI_CONF_MONITORBOTH;
if (ioctl(fd, ZT_SETCONF, &ztc)) {
if (ioctl(fd, DAHDI_SETCONF, &dahdic)) {
ast_log(LOG_WARNING, "Error setting conference\n");
close(fd);
goto outrun;
}
ast_debug(1, "Placed channel %s in ZAP channel %d monitor\n", chan->name, confno);
ast_debug(1, "Placed channel %s in DAHDI channel %d monitor\n", chan->name, confno);
for(;;) {
outfd = -1;
@@ -195,13 +196,13 @@ static int conf_run(struct ast_channel *chan, int confno, int confflags)
c = ast_waitfor_nandfds(&chan, 1, &fd, nfds, NULL, &outfd, &ms);
if (c) {
if (c->fds[0] != origfd) {
if (retryzap) {
if (retrydahdi) {
/* Kill old pseudo */
close(fd);
}
ast_debug(1, "Ooh, something swapped out under us, starting over\n");
retryzap = 0;
goto zapretry;
retrydahdi = 0;
goto dahdiretry;
}
f = ast_read(c);
if (!f)
@@ -223,7 +224,7 @@ static int conf_run(struct ast_channel *chan, int confno, int confflags)
input[ic++] = '\0';
ic=0;
ret = atoi(input);
ast_verb(3, "Zapscan: change channel to %d\n",ret);
ast_verb(3, "DAHDIscan: change channel to %d\n",ret);
break;
}
}
@@ -263,10 +264,10 @@ static int conf_run(struct ast_channel *chan, int confno, int confflags)
else {
/* Take out of conference */
/* Add us to the conference */
ztc.chan = 0;
ztc.confno = 0;
ztc.confmode = 0;
if (ioctl(fd, ZT_SETCONF, &ztc)) {
dahdic.chan = 0;
dahdic.confno = 0;
dahdic.confmode = 0;
if (ioctl(fd, DAHDI_SETCONF, &dahdic)) {
ast_log(LOG_WARNING, "Error setting conference\n");
}
}
@@ -310,7 +311,7 @@ static int conf_exec(struct ast_channel *chan, void *data)
ast_frfree(f);
ichan = NULL;
if(input) {
ichan = get_zap_channel_locked(input);
ichan = get_dahdi_channel_locked(input);
input = 0;
}
@@ -329,8 +330,8 @@ static int conf_exec(struct ast_channel *chan, void *data)
continue;
}
}
if (tempchan && (!strcmp(tempchan->tech->type, "Zap")) && (tempchan != chan) ) {
ast_verb(3, "Zap channel %s is in-use, monitoring...\n", tempchan->name);
if (tempchan && (!strcmp(tempchan->tech->type, "DAHDI")) && (tempchan != chan) ) {
ast_verb(3, "DAHDI channel %s is in-use, monitoring...\n", tempchan->name);
ast_copy_string(confstr, tempchan->name, sizeof(confstr));
ast_channel_unlock(tempchan);
if ((tmp = strchr(confstr,'-'))) {
@@ -359,5 +360,5 @@ static int load_module(void)
return ((ast_register_application(app, conf_exec, synopsis, descrip)) ? AST_MODULE_LOAD_FAILURE : AST_MODULE_LOAD_SUCCESS);
}
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Scan Zap channels application");
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Scan DAHDI channels application");

View File

@@ -122,14 +122,18 @@ static char *descrip =
" Optionally, an extension, or extension and context may be specified. \n"
" Otherwise, the current extension is used. You cannot use any additional\n"
" action post answer options in conjunction with this option.\n"
" h - Allow the called party to hang up by sending the '*' DTMF digit.\n"
" H - Allow the calling party to hang up by hitting the '*' DTMF digit.\n"
" h - Allow the called party to hang up by sending the '*' DTMF digit, or\n"
" whatever sequence was defined in the featuremap section for\n"
" 'disconnect' in features.conf\n"
" H - Allow the calling party to hang up by hitting the '*' DTMF digit, or\n"
" whatever sequence was defined in the featuremap section for\n"
" 'disconnect' in features.conf\n"
" i - Asterisk will ignore any forwarding requests it may receive on this\n"
" dial attempt.\n"
" k - Allow the called party to enable parking of the call by sending\n"
" the DTMF sequence defined for call parking in features.conf.\n"
" the DTMF sequence defined for call parking in the featuremap section of features.conf.\n"
" K - Allow the calling party to enable parking of the call by sending\n"
" the DTMF sequence defined for call parking in features.conf.\n"
" the DTMF sequence defined for call parking in the featuremap section of features.conf.\n"
" L(x[:y][:z]) - Limit the call to 'x' ms. Play a warning when 'y' ms are\n"
" left. Repeat the warning every 'z' ms. The following special\n"
" variables can be used with this option:\n"
@@ -160,6 +164,8 @@ static char *descrip =
" You cannot use any additional action post answer options in conjunction\n"
" with this option. Also, pbx services are not run on the peer (called) channel,\n"
" so you will not be able to set timeouts via the TIMEOUT() function in this macro.\n"
" Be aware of the limitations that macros have, specifically with regards to use of\n"
" the WaitExten application. For more information, see the documentation for Macro()\n"
" n - This option is a modifier for the screen/privacy mode. It specifies\n"
" that no introductions are to be saved in the priv-callerintros\n"
" directory.\n"
@@ -168,8 +174,8 @@ static char *descrip =
" o - Specify that the CallerID that was present on the *calling* channel\n"
" be set as the CallerID on the *called* channel. This was the\n"
" behavior of Asterisk 1.0 and earlier.\n"
" O([x]) - \"Operator Services\" mode (Zaptel channel to Zaptel channel\n"
" only, if specified on non-Zaptel interface, it will be ignored).\n"
" O([x]) - \"Operator Services\" mode (DAHDI channel to DAHDI channel\n"
" only, if specified on non-DAHDI interface, it will be ignored).\n"
" When the destination answers (presumably an operator services\n"
" station), the originator no longer has control of their line.\n"
" They may hang up, but the switch will not release their line\n"
@@ -188,9 +194,11 @@ static char *descrip =
" S(x) - Hang up the call after 'x' seconds *after* the called party has\n"
" answered the call.\n"
" t - Allow the called party to transfer the calling party by sending the\n"
" DTMF sequence defined in features.conf.\n"
" DTMF sequence defined in the blindxfer setting in the featuremap section\n"
" of features.conf.\n"
" T - Allow the calling party to transfer the called party by sending the\n"
" DTMF sequence defined in features.conf.\n"
" DTMF sequence defined in the blindxfer setting in the featuremap section\n"
" of features.conf.\n"
" U(x[^arg]) - Execute via Gosub the routine 'x' for the *called* channel before connecting\n"
" to the calling channel. Arguments can be specified to the Gosub\n"
" using '^' as a delimeter. The Gosub routine can set the variable\n"
@@ -207,13 +215,17 @@ static char *descrip =
" with this option. Also, pbx services are not run on the peer (called) channel,\n"
" so you will not be able to set timeouts via the TIMEOUT() function in this routine.\n"
" w - Allow the called party to enable recording of the call by sending\n"
" the DTMF sequence defined for one-touch recording in features.conf.\n"
" the DTMF sequence defined in the automon setting in the featuremap section\n"
" of features.conf.\n"
" W - Allow the calling party to enable recording of the call by sending\n"
" the DTMF sequence defined for one-touch recording in features.conf.\n"
" the DTMF sequence defined in the automon setting in the featuremap section\n"
" of features.conf.\n"
" x - Allow the called party to enable recording of the call by sending\n"
" the DTMF sequence defined for one-touch automixmonitor in features.conf\n"
" the DTMF sequence defined in the automixmon setting in the featuremap section\n"
" of features.conf.\n"
" X - Allow the calling party to enable recording of the call by sending\n"
" the DTMF sequence defined for one-touch automixmonitor in features.conf\n";
" the DTMF sequence defined in the automixmon setting in the featuremap section\n"
" of features.conf.\n";
/* RetryDial App by Anthony Minessale II <anthmct@yahoo.com> Jan/2005 */
static char *rapp = "RetryDial";
@@ -222,7 +234,7 @@ static char *rdescrip =
" RetryDial(announce,sleep,retries,dialargs): This application will attempt to\n"
"place a call using the normal Dial application. If no channel can be reached,\n"
"the 'announce' file will be played. Then, it will wait 'sleep' number of\n"
"seconds before retying the call. After 'retires' number of attempts, the\n"
"seconds before retrying the call. After 'retries' number of attempts, the\n"
"calling channel will continue at the next priority in the dialplan. If the\n"
"'retries' setting is set to 0, this application will retry endlessly.\n"
" While waiting to retry a call, a 1 digit extension may be dialed. If that\n"
@@ -299,8 +311,6 @@ AST_APP_OPTIONS(dial_exec_options, BEGIN_OPTIONS
AST_APP_OPTION('i', OPT_IGNORE_FORWARDING),
AST_APP_OPTION('k', OPT_CALLEE_PARK),
AST_APP_OPTION('K', OPT_CALLER_PARK),
AST_APP_OPTION('k', OPT_CALLEE_PARK),
AST_APP_OPTION('K', OPT_CALLER_PARK),
AST_APP_OPTION_ARG('L', OPT_DURATION_LIMIT, OPT_ARG_DURATION_LIMIT),
AST_APP_OPTION_ARG('m', OPT_MUSICBACK, OPT_ARG_MUSICBACK),
AST_APP_OPTION_ARG('M', OPT_CALLEE_MACRO, OPT_ARG_CALLEE_MACRO),
@@ -321,9 +331,10 @@ AST_APP_OPTIONS(dial_exec_options, BEGIN_OPTIONS
AST_APP_OPTION('X', OPT_CALLER_MIXMONITOR),
END_OPTIONS );
#define CAN_EARLY_BRIDGE(flags) (!ast_test_flag64(flags, OPT_CALLEE_HANGUP | \
#define CAN_EARLY_BRIDGE(flags,chan,peer) (!ast_test_flag64(flags, OPT_CALLEE_HANGUP | \
OPT_CALLER_HANGUP | OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER | \
OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR | OPT_CALLEE_PARK | OPT_CALLER_PARK))
OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR | OPT_CALLEE_PARK | OPT_CALLER_PARK) && \
!chan->audiohooks && !peer->audiohooks)
/*
* The list of active channels
@@ -334,6 +345,7 @@ struct chanlist {
uint64_t flags;
};
static int detect_disconnect(struct ast_channel *chan, char code, struct ast_str *featurecode);
static void hanguptree(struct chanlist *outgoing, struct ast_channel *exception, int answered_elsewhere)
{
@@ -381,12 +393,18 @@ static void handle_cause(int cause, struct cause_args *num)
num->congestion++;
break;
case AST_CAUSE_NO_ROUTE_DESTINATION:
case AST_CAUSE_UNREGISTERED:
if (cdr)
ast_cdr_failed(cdr);
num->nochan++;
break;
case AST_CAUSE_NO_ANSWER:
if (cdr) {
ast_cdr_noanswer(cdr);
}
break;
case AST_CAUSE_NORMAL_CLEARING:
break;
@@ -480,6 +498,9 @@ static void do_forward(struct chanlist *o,
tech = tmpchan;
} else {
const char *forward_context = pbx_builtin_getvar_helper(c, "FORWARD_CONTEXT");
if (ast_strlen_zero(forward_context)) {
forward_context = NULL;
}
snprintf(tmpchan, sizeof(tmpchan), "%s@%s", c->call_forward, forward_context ? forward_context : c->context);
stuff = tmpchan;
tech = "Local";
@@ -505,6 +526,7 @@ static void do_forward(struct chanlist *o,
if (!c) {
ast_clear_flag64(o, DIAL_STILLGOING);
handle_cause(cause, num);
ast_hangup(original);
} else {
char *new_cid_num, *new_cid_name;
struct ast_channel *src;
@@ -532,6 +554,7 @@ static void do_forward(struct chanlist *o,
ast_log(LOG_NOTICE, "Failed to dial on local channel for call forward to '%s'\n", tmpchan);
ast_clear_flag64(o, DIAL_STILLGOING);
ast_hangup(original);
ast_hangup(c);
c = o->chan = NULL;
num->nochan++;
} else {
@@ -544,6 +567,9 @@ static void do_forward(struct chanlist *o,
/* Hangup the original channel now, in case we needed it */
ast_hangup(original);
}
if (single) {
ast_indicate(in, -1);
}
}
}
@@ -570,7 +596,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
#ifdef HAVE_EPOLL
struct chanlist *epollo;
#endif
struct ast_str *featurecode = ast_str_alloca(FEATURE_MAX_LEN + 1);
if (single) {
/* Turn off hold music, etc */
ast_deactivate_generator(in);
@@ -661,6 +687,10 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
if (!peer) {
ast_verb(3, "%s answered %s\n", c->name, in->name);
peer = c;
if (peer->cdr) {
peer->cdr->answer = ast_tvnow();
peer->cdr->disposition = AST_CDR_ANSWERED;
}
ast_copy_flags64(peerflags, o,
OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
@@ -670,7 +700,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
DIAL_NOFORWARDHTML);
ast_copy_string(c->dialcontext, "", sizeof(c->dialcontext));
ast_copy_string(c->exten, "", sizeof(c->exten));
if (CAN_EARLY_BRIDGE(peerflags))
if (CAN_EARLY_BRIDGE(peerflags, in, peer))
/* Setup early bridge if appropriate */
ast_channel_early_bridge(in, peer);
}
@@ -697,7 +727,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
case AST_CONTROL_RINGING:
ast_verb(3, "%s is ringing\n", c->name);
/* Setup early media if appropriate */
if (single && CAN_EARLY_BRIDGE(peerflags))
if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
ast_channel_early_bridge(in, c);
if (!(pa->sentringing) && !ast_test_flag64(outgoing, OPT_MUSICBACK)) {
ast_indicate(in, AST_CONTROL_RINGING);
@@ -707,7 +737,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
case AST_CONTROL_PROGRESS:
ast_verb(3, "%s is making progress passing it to %s\n", c->name, in->name);
/* Setup early media if appropriate */
if (single && CAN_EARLY_BRIDGE(peerflags))
if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
ast_channel_early_bridge(in, c);
if (!ast_test_flag64(outgoing, OPT_RINGBACK))
ast_indicate(in, AST_CONTROL_PROGRESS);
@@ -716,9 +746,13 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
ast_verb(3, "%s requested a video update, passing it to %s\n", c->name, in->name);
ast_indicate(in, AST_CONTROL_VIDUPDATE);
break;
case AST_CONTROL_SRCUPDATE:
ast_verb(3, "%s requested a source update, passing it to %s\n", c->name, in->name);
ast_indicate(in, AST_CONTROL_SRCUPDATE);
break;
case AST_CONTROL_PROCEEDING:
ast_verb(3, "%s is proceeding passing it to %s\n", c->name, in->name);
if (single && CAN_EARLY_BRIDGE(peerflags))
if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
ast_channel_early_bridge(in, c);
if (!ast_test_flag64(outgoing, OPT_RINGBACK))
ast_indicate(in, AST_CONTROL_PROCEEDING);
@@ -797,8 +831,8 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
}
if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP) &&
(f->subclass == '*')) { /* hmm it it not guaranteed to be '*' anymore. */
ast_verb(3, "User hit %c to disconnect call.\n", f->subclass);
detect_disconnect(in, f->subclass, featurecode)) {
ast_verb(3, "User requested call disconnect.\n");
*to = 0;
strcpy(pa->status, "CANCEL");
ast_cdr_noanswer(in->cdr);
@@ -819,7 +853,8 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
if (single && (f->frametype == AST_FRAME_CONTROL) &&
((f->subclass == AST_CONTROL_HOLD) ||
(f->subclass == AST_CONTROL_UNHOLD) ||
(f->subclass == AST_CONTROL_VIDUPDATE))) {
(f->subclass == AST_CONTROL_VIDUPDATE) ||
(f->subclass == AST_CONTROL_SRCUPDATE))) {
ast_verb(3, "%s requested special control %d, passing it to %s\n", in->name, f->subclass, outgoing->chan->name);
ast_indicate_data(outgoing->chan, f->subclass, f->data, f->datalen);
}
@@ -830,23 +865,6 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
if (!*to || ast_check_hangup(in))
ast_cdr_noanswer(in->cdr);
}
if (peer && !ast_cdr_log_unanswered()) {
/* suppress the CDR's that didn't win */
struct chanlist *o;
for (o = outgoing; o; o = o->next) {
struct ast_channel *c = o->chan;
if (c && c != peer && c->cdr)
ast_set_flag(c->cdr, AST_CDR_FLAG_POST_DISABLED);
}
} else if (!peer && !ast_cdr_log_unanswered()) {
/* suppress the CDR's that didn't win */
struct chanlist *o;
for (o = outgoing; o; o = o->next) {
struct ast_channel *c = o->chan;
if (c && c->cdr)
ast_set_flag(c->cdr, AST_CDR_FLAG_POST_DISABLED);
}
}
#ifdef HAVE_EPOLL
for (epollo = outgoing; epollo; epollo = epollo->next) {
@@ -858,6 +876,26 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
return peer;
}
static int detect_disconnect(struct ast_channel *chan, char code, struct ast_str *featurecode)
{
struct ast_flags features = { AST_FEATURE_DISCONNECT }; /* only concerned with disconnect feature */
struct ast_call_feature feature = { 0, };
int res;
ast_str_append(&featurecode, 1, "%c", code);
res = ast_feature_detect(chan, &features, ast_str_buffer(featurecode), &feature);
if (res != AST_FEATURE_RETURN_STOREDIGITS) {
ast_str_reset(featurecode);
}
if (feature.feature_mask & AST_FEATURE_DISCONNECT) {
return 1;
}
return 0;
}
static void replace_macro_delimiter(char *s)
{
for (; *s; s++)
@@ -951,7 +989,12 @@ static int do_timelimit(struct ast_channel *chan, struct ast_bridge_config *conf
*calldurationlimit = 0;
/* more efficient to do it like S(x) does since no advanced opts */
if (!config->play_warning && !config->start_sound && !config->end_sound && config->timelimit) {
*calldurationlimit = config->timelimit / 1000;
if (config->timelimit > 1000) {
*calldurationlimit = config->timelimit / 1000;
} else if (config->timelimit > 0) {
/* Not enough granularity to make it less, but we can't use the special value 0 */
*calldurationlimit = 1;
}
ast_verb(3, "Setting call duration limit to %d seconds.\n",
*calldurationlimit);
config->timelimit = play_to_caller = play_to_callee =
@@ -1208,17 +1251,33 @@ static int setup_privacy_args(struct privacy_args *pa,
return 1; /* success */
}
static void set_dial_features(struct ast_flags64 *opts, struct ast_dial_features *features)
static void end_bridge_callback(void *data)
{
struct ast_flags64 perm_opts = {.flags = 0};
char buf[80];
time_t end;
struct ast_channel *chan = data;
ast_copy_flags64(&perm_opts, opts,
OPT_CALLER_TRANSFER | OPT_CALLER_PARK | OPT_CALLER_MONITOR | OPT_CALLER_MIXMONITOR | OPT_CALLER_HANGUP |
OPT_CALLEE_TRANSFER | OPT_CALLEE_PARK | OPT_CALLEE_MONITOR | OPT_CALLEE_MIXMONITOR | OPT_CALLEE_HANGUP);
if (!chan->cdr) {
return;
}
memset(features->options, 0, sizeof(features->options));
time(&end);
ast_app_options2str64(dial_exec_options, &perm_opts, features->options, sizeof(features->options));
ast_channel_lock(chan);
if (chan->cdr->answer.tv_sec) {
snprintf(buf, sizeof(buf), "%ld", end - chan->cdr->answer.tv_sec);
pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", buf);
}
if (chan->cdr->start.tv_sec) {
snprintf(buf, sizeof(buf), "%ld", end - chan->cdr->start.tv_sec);
pbx_builtin_setvar_helper(chan, "DIALEDTIME", buf);
}
ast_channel_unlock(chan);
}
static void end_bridge_callback_data_fixup(struct ast_bridge_config *bconfig, struct ast_channel *originator, struct ast_channel *terminator) {
bconfig->end_bridge_callback_data = originator;
}
static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags64 *peerflags, int *continue_exec)
@@ -1244,7 +1303,6 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
int sentringing = 0, moh = 0;
const char *outbound_group = NULL;
int result = 0;
time_t start_time;
char *parse;
int opermode = 0;
AST_DECLARE_APP_ARGS(args,
@@ -1256,11 +1314,15 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
struct ast_flags64 opts = { 0, };
char *opt_args[OPT_ARG_ARRAY_SIZE];
struct ast_datastore *datastore = NULL;
struct ast_datastore *ds_caller_features = NULL;
struct ast_datastore *ds_callee_features = NULL;
struct ast_dial_features *caller_features;
int fulldial = 0, num_dialed = 0;
/* Reset all DIAL variables back to blank, to prevent confusion (in case we don't reset all of them). */
pbx_builtin_setvar_helper(chan, "DIALSTATUS", "");
pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", "");
pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", "");
pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", "");
pbx_builtin_setvar_helper(chan, "DIALEDTIME", "");
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "Dial requires an argument (technology/number)\n");
pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
@@ -1320,6 +1382,10 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
res = -1; /* reset default */
}
if (ast_test_flag64(&opts, OPT_DTMF_EXIT) || ast_test_flag64(&opts, OPT_CALLER_HANGUP)) {
__ast_answer(chan, 0, 0);
}
if (continue_exec)
*continue_exec = 0;
@@ -1333,28 +1399,6 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
ast_copy_flags64(peerflags, &opts, OPT_DTMF_EXIT | OPT_GO_ON | OPT_ORIGINAL_CLID | OPT_CALLER_HANGUP | OPT_IGNORE_FORWARDING);
/* Create datastore for channel dial features for caller */
if (!(ds_caller_features = ast_channel_datastore_alloc(&dial_features_info, NULL))) {
ast_log(LOG_WARNING, "Unable to create channel datastore for dial features. Aborting!\n");
goto out;
}
if (!(caller_features = ast_malloc(sizeof(*caller_features)))) {
ast_log(LOG_WARNING, "Unable to allocate memory for feature flags. Aborting!\n");
goto out;
}
ast_copy_flags(&(caller_features->features_callee), &(config.features_caller), AST_FLAGS_ALL);
caller_features->is_caller = 1;
set_dial_features(&opts, caller_features);
ds_caller_features->inheritance = DATASTORE_INHERIT_FOREVER;
ds_caller_features->data = caller_features;
ast_channel_lock(chan);
ast_channel_datastore_add(chan, ds_caller_features);
ast_channel_unlock(chan);
/* loop through the list of dial destinations */
rest = args.peers;
while ((cur = strsep(&rest, "&")) ) {
@@ -1366,7 +1410,6 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
char *tech = strsep(&number, "/");
/* find if we already dialed this interface */
struct ast_dialed_interface *di;
struct ast_dial_features *callee_features;
AST_LIST_HEAD(, ast_dialed_interface) *dialed_interfaces;
num_dialed++;
if (!number) {
@@ -1468,6 +1511,7 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
/* Inherit specially named variables from parent channel */
ast_channel_inherit_variables(chan, tc);
ast_channel_datastore_inherit(chan, tc);
tc->appl = "AppDial";
tc->data = "(Outgoing Line)";
@@ -1506,30 +1550,6 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
else
ast_copy_string(tc->exten, chan->exten, sizeof(tc->exten));
/* Save callee features */
if (!(ds_callee_features = ast_channel_datastore_alloc(&dial_features_info, NULL))) {
ast_log(LOG_WARNING, "Unable to create channel datastore for dial features. Aborting!\n");
ast_free(tmp);
goto out;
}
if (!(callee_features = ast_malloc(sizeof(*callee_features)))) {
ast_log(LOG_WARNING, "Unable to allocate memory for feature flags. Aborting!\n");
ast_free(tmp);
goto out;
}
ast_copy_flags(&(callee_features->features_callee), &(config.features_callee), AST_FLAGS_ALL);
callee_features->is_caller = 0;
set_dial_features(&opts, callee_features);
ds_callee_features->inheritance = DATASTORE_INHERIT_FOREVER;
ds_callee_features->data = callee_features;
ast_channel_lock(chan);
ast_channel_datastore_add(tc, ds_callee_features);
ast_channel_unlock(chan);
res = ast_call(tc, numsubst, 0); /* Place the call, but don't wait on the answer */
/* Save the info in cdr's that we called them */
@@ -1541,6 +1561,9 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
/* Again, keep going even if there's an error */
ast_debug(1, "ast call on peer returned %d\n", res);
ast_verb(3, "Couldn't call %s\n", numsubst);
if (tc->hangupcause) {
chan->hangupcause = tc->hangupcause;
}
ast_hangup(tc);
tc = NULL;
ast_free(tmp);
@@ -1599,11 +1622,16 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
}
}
time(&start_time);
peer = wait_for_answer(chan, outgoing, &to, peerflags, &pa, &num, &result);
ast_channel_datastore_remove(chan, datastore);
ast_channel_datastore_free(datastore);
/* The ast_channel_datastore_remove() function could fail here if the
* datastore was moved to another channel during a masquerade. If this is
* the case, don't free the datastore here because later, when the channel
* to which the datastore was moved hangs up, it will attempt to free this
* datastore again, causing a crash
*/
if (!ast_channel_datastore_remove(chan, datastore))
ast_channel_datastore_free(datastore);
if (!peer) {
if (result) {
res = result;
@@ -1615,10 +1643,9 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
/* almost done, although the 'else' block is 400 lines */
} else {
const char *number;
time_t end_time, answer_time = time(NULL);
char toast[80]; /* buffer to set variables */
strcpy(pa.status, "ANSWER");
pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
/* Ah ha! Someone answered within the desired timeframe. Of course after this
we will always return with -1 so that it is hung up properly after the
conversation. */
@@ -1693,6 +1720,10 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
theapp = pbx_findapp("Macro");
if (theapp && !res) { /* XXX why check res here ? */
/* Set peer->exten and peer->context so that MACRO_EXTEN and MACRO_CONTEXT get set */
ast_copy_string(peer->context, chan->context, sizeof(peer->context));
ast_copy_string(peer->exten, chan->exten, sizeof(peer->exten));
replace_macro_delimiter(opt_args[OPT_ARG_CALLEE_MACRO]);
res = pbx_exec(peer, theapp, opt_args[OPT_ARG_CALLEE_MACRO]);
ast_debug(1, "Macro exited with status %d\n", res);
@@ -1703,7 +1734,6 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
}
if (ast_autoservice_stop(chan) < 0) {
ast_log(LOG_ERROR, "Could not stop autoservice on calling channel\n");
res = -1;
}
@@ -1761,23 +1791,35 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
ast_copy_string(peer->exten, "s", sizeof(peer->exten));
peer->priority = 0;
gosub_argstart = strchr(opt_args[OPT_ARG_CALLEE_GOSUB], '|');
gosub_argstart = strchr(opt_args[OPT_ARG_CALLEE_GOSUB], ',');
if (gosub_argstart) {
*gosub_argstart = 0;
asprintf(&gosub_args, "%s|s|1(%s)", opt_args[OPT_ARG_CALLEE_GOSUB], gosub_argstart + 1);
*gosub_argstart = '|';
if (asprintf(&gosub_args, "%s,s,1(%s)", opt_args[OPT_ARG_CALLEE_GOSUB], gosub_argstart + 1) < 0) {
ast_log(LOG_WARNING, "asprintf() failed: %s\n", strerror(errno));
gosub_args = NULL;
}
*gosub_argstart = ',';
} else {
asprintf(&gosub_args, "%s|s|1", opt_args[OPT_ARG_CALLEE_GOSUB]);
if (asprintf(&gosub_args, "%s,s,1", opt_args[OPT_ARG_CALLEE_GOSUB]) < 0) {
ast_log(LOG_WARNING, "asprintf() failed: %s\n", strerror(errno));
gosub_args = NULL;
}
}
if (gosub_args) {
res = pbx_exec(peer, theapp, gosub_args);
ast_pbx_run(peer);
if (!res) {
struct ast_pbx_args args;
/* A struct initializer fails to compile for this case ... */
memset(&args, 0, sizeof(args));
args.no_hangup_chan = 1;
ast_pbx_run_args(peer, &args);
}
ast_free(gosub_args);
if (option_debug)
ast_log(LOG_DEBUG, "Gosub exited with status %d\n", res);
} else
ast_debug(1, "Gosub exited with status %d\n", res);
} else {
ast_log(LOG_ERROR, "Could not Allocate string for Gosub arguments -- Gosub Call Aborted!\n");
}
res = 0;
} else {
@@ -1836,10 +1878,9 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
res = ast_dtmf_stream(chan, peer, dtmfcalling, 250, 0);
}
}
if (res) { /* some error */
res = -1;
end_time = time(NULL);
} else {
if (ast_test_flag64(peerflags, OPT_CALLEE_TRANSFER))
ast_set_flag(&(config.features_callee), AST_FEATURE_REDIRECT);
@@ -1861,7 +1902,13 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMIXMON);
if (ast_test_flag64(peerflags, OPT_CALLER_MIXMONITOR))
ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMIXMON);
if (ast_test_flag64(peerflags, OPT_GO_ON))
ast_set_flag(&(config.features_caller), AST_FEATURE_NO_H_EXTEN);
config.end_bridge_callback = end_bridge_callback;
config.end_bridge_callback_data = chan;
config.end_bridge_callback_data_fixup = end_bridge_callback_data_fixup;
if (moh) {
moh = 0;
ast_moh_stop(chan);
@@ -1879,9 +1926,9 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
res = -1;
goto done;
}
if (opermode && !strncmp(chan->name, "Zap", 3) && !strncmp(peer->name, "Zap", 3)) {
/* what's this special handling for Zap <-> Zap ?
* A: Zap to Zap calls are natively bridged at the kernel driver
if (opermode && !strncmp(chan->tech->type, "DAHDI", 5) && !strncmp(peer->name, "DAHDI", 5)) {
/* what's this special handling for dahdi <-> dahdi ?
* A: dahdi to dahdi calls are natively bridged at the kernel driver
* level, so we need to ensure that this mode gets propagated
* all the way down. */
struct oprmode oprmode;
@@ -1892,17 +1939,6 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
ast_channel_setoption(chan, AST_OPTION_OPRMODE, &oprmode, sizeof(oprmode), 0);
}
res = ast_bridge_call(chan, peer, &config);
end_time = time(NULL);
snprintf(toast, sizeof(toast), "%ld", (long)(end_time - answer_time));
pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", toast);
}
snprintf(toast, sizeof(toast), "%ld", (long)(end_time - start_time));
pbx_builtin_setvar_helper(chan, "DIALEDTIME", toast);
if (ast_test_flag64(&opts, OPT_PEER_H)) {
ast_log(LOG_NOTICE, "PEER context: %s; PEER exten: %s; PEER priority: %d\n",
peer->context, peer->exten, peer->priority);
}
strcpy(peer->context, chan->context);
@@ -1915,7 +1951,7 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
autoloopflag = ast_test_flag(peer, AST_FLAG_IN_AUTOLOOP); /* save value to restore at the end */
ast_set_flag(peer, AST_FLAG_IN_AUTOLOOP);
while ((res = ast_spawn_extension(peer, peer->context, peer->exten, peer->priority, peer->cid.cid_num, &found, 1)))
while ((res = ast_spawn_extension(peer, peer->context, peer->exten, peer->priority, peer->cid.cid_num, &found, 1)) == 0)
peer->priority++;
if (found && res) {
@@ -1925,11 +1961,9 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
}
ast_set2_flag(peer, autoloopflag, AST_FLAG_IN_AUTOLOOP); /* set it back the way it was */
}
if (res != AST_PBX_NO_HANGUP_PEER) {
if (!ast_check_hangup(chan))
chan->hangupcause = peer->hangupcause;
ast_hangup(peer);
}
if (!ast_check_hangup(chan))
chan->hangupcause = peer->hangupcause;
ast_hangup(peer);
}
out:
if (moh) {
@@ -1944,13 +1978,12 @@ out:
pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
senddialendevent(chan, pa.status);
ast_debug(1, "Exiting with DIALSTATUS=%s.\n", pa.status);
if ((ast_test_flag64(peerflags, OPT_GO_ON)) && !ast_check_hangup(chan) && (res != AST_PBX_KEEPALIVE)) {
if (calldurationlimit)
chan->whentohangup = 0;
res = 0;
}
if (ast_test_flag64(peerflags, OPT_GO_ON) && !ast_check_hangup(chan)) {
if (calldurationlimit)
chan->whentohangup = 0;
res = 0;
}
done:
return res;
}
@@ -1985,10 +2018,12 @@ static int retrydial_exec(struct ast_channel *chan, void *data)
parse = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, parse);
if ((sleep = atoi(args.sleep)))
if (!ast_strlen_zero(args.sleep) && (sleep = atoi(args.sleep)))
sleep *= 1000;
loops = atoi(args.retries);
if (!ast_strlen_zero(args.retries)) {
loops = atoi(args.retries);
}
if (!args.dialdata) {
ast_log(LOG_ERROR, "%s requires a 4th argument (dialdata)\n", rapp);
@@ -2076,7 +2111,7 @@ static int unload_module(void)
res |= ast_unregister_application(rapp);
if ((con = ast_context_find("app_dial_gosub_virtual_context"))) {
ast_context_remove_extension2(con, "s", 1, NULL);
ast_context_remove_extension2(con, "s", 1, NULL, 0);
ast_context_destroy(con, "app_dial"); /* leave nothing behind */
}
@@ -2088,13 +2123,11 @@ static int load_module(void)
int res;
struct ast_context *con;
con = ast_context_find("app_dial_gosub_virtual_context");
if (!con)
con = ast_context_create(NULL, "app_dial_gosub_virtual_context", "app_dial");
con = ast_context_find_or_create(NULL, NULL, "app_dial_gosub_virtual_context", "app_dial");
if (!con)
ast_log(LOG_ERROR, "Dial virtual context 'app_dial_gosub_virtual_context' does not exist and unable to create\n");
else
ast_add_extension2(con, 1, "s", 1, NULL, NULL, "KeepAlive", ast_strdup(""), ast_free_ptr, "app_dial");
ast_add_extension2(con, 1, "s", 1, NULL, NULL, "NoOp", ast_strdup(""), ast_free_ptr, "app_dial");
res = ast_register_application(app, dial_exec, synopsis, descrip);
res |= ast_register_application(rapp, retrydial_exec, rsynopsis, rdescrip);

View File

@@ -116,7 +116,9 @@ static int dictate_exec(struct ast_channel *chan, void *data)
return -1;
}
ast_answer(chan);
if (chan->_state != AST_STATE_UP) {
ast_answer(chan);
}
ast_safe_sleep(chan, 200);
for (res = 0; !res;) {
if (ast_strlen_zero(filename)) {

View File

@@ -79,7 +79,7 @@ static int pickup_do(struct ast_channel *chan, struct ast_channel *target)
/* Helper function that determines whether a channel is capable of being picked up */
static int can_pickup(struct ast_channel *chan)
{
if (!chan->pbx && (chan->_state == AST_STATE_RINGING || chan->_state == AST_STATE_RING))
if (!chan->pbx && (chan->_state == AST_STATE_RINGING || chan->_state == AST_STATE_RING || chan->_state == AST_STATE_DOWN))
return 1;
else
return 0;

View File

@@ -222,6 +222,10 @@ static int compare(const char *text, const char *template)
{
char digit;
if (ast_strlen_zero(text)) {
return -1;
}
while (*template) {
digit = toupper(*text++);
switch (digit) {
@@ -492,6 +496,9 @@ static struct ast_config *realtime_directory(char *context)
if (!cat) {
ast_log(LOG_WARNING, "Out of memory\n");
ast_config_destroy(cfg);
if (rtdata) {
ast_config_destroy(rtdata);
}
return NULL;
}
ast_category_append(cfg, cat);
@@ -578,7 +585,7 @@ static int search_directory(const char *context, struct ast_config *vmcfg, struc
strsep(&bufptr, ",");
pos = strsep(&bufptr, ",");
res = check_match(&item, pos, v->name, ext, use_first_name);
res = check_match(&item, S_OR(pos, ""), v->name, ext, use_first_name);
if (!res)
continue;
else if (res < 0)

View File

@@ -102,7 +102,7 @@ AST_APP_OPTIONS(app_opts, {
static void play_dialtone(struct ast_channel *chan, char *mailbox)
{
const struct ind_tone_zone_sound *ts = NULL;
const struct tone_zone_sound *ts = NULL;
if(ast_app_has_voicemail(mailbox, NULL))
ts = ast_get_indication_tone(chan->zone, "dialrecall");
else
@@ -279,8 +279,14 @@ static int disa_exec(struct ast_channel *chan, void *data)
continue;
}
} else {
if (j == '#') { /* end of extension */
break;
if (j == '#') { /* end of extension .. maybe */
if (i == 0 &&
(ast_matchmore_extension(chan, args.context, "#", 1, chan->cid.cid_num) ||
ast_exists_extension(chan, args.context, "#", 1, chan->cid.cid_num)) ) {
/* Let the # be the part of, or the entire extension */
} else {
break;
}
}
}

View File

@@ -141,8 +141,8 @@ static int dumpchan_exec(struct ast_channel *chan, void *data)
pbx_builtin_serialize_variables(chan, &vars);
serialize_showchan(chan, info, sizeof(info));
if (level > 0)
ast_verb(level, "\nDumping Info For Channel: %s:\n%s\nInfo:\n%s\nVariables:\n%s%s\n", chan->name, line, info, vars->str, line);
if (option_verbose >= level)
ast_verbose("\nDumping Info For Channel: %s:\n%s\nInfo:\n%s\nVariables:\n%s%s\n", chan->name, line, info, vars->str, line);
return 0;
}

View File

@@ -148,7 +148,7 @@ static int tryexec_exec(struct ast_channel *chan, void *data)
static int execif_exec(struct ast_channel *chan, void *data)
{
int res = 0;
char *truedata = NULL, *falsedata = NULL, *end;
char *truedata = NULL, *falsedata = NULL, *end, *firstcomma, *firstquestion;
struct ast_app *app = NULL;
AST_DECLARE_APP_ARGS(expr,
AST_APP_ARG(expr);
@@ -160,24 +160,49 @@ static int execif_exec(struct ast_channel *chan, void *data)
);
char *parse = ast_strdupa(data);
AST_NONSTANDARD_APP_ARGS(expr, parse, '?');
if (ast_strlen_zero(expr.remainder)) {
ast_log(LOG_ERROR, "Usage: ExecIf(<expr>?<appiftrue>(<args>)[:<appiffalse>(<args)])\n");
return -1;
}
firstcomma = strchr(parse, ',');
firstquestion = strchr(parse, '?');
AST_NONSTANDARD_APP_ARGS(apps, expr.remainder, ':');
if ((firstcomma != NULL && firstquestion != NULL && firstcomma < firstquestion) || (firstquestion == NULL)) {
/* Deprecated syntax */
AST_DECLARE_APP_ARGS(depr,
AST_APP_ARG(expr);
AST_APP_ARG(appname);
AST_APP_ARG(appargs);
);
AST_STANDARD_APP_ARGS(depr, parse);
if (apps.t && (truedata = strchr(apps.t, '('))) {
*truedata++ = '\0';
if ((end = strrchr(truedata, ')')))
*end = '\0';
}
ast_log(LOG_WARNING, "Deprecated syntax found. Please upgrade to using ExecIf(<expr>?%s(%s))\n", depr.appname, depr.appargs);
if (apps.f && (falsedata = strchr(apps.f, '('))) {
*falsedata++ = '\0';
if ((end = strrchr(falsedata, ')')))
*end = '\0';
/* Make the two syntaxes look the same */
expr.expr = depr.expr;
apps.t = depr.appname;
apps.f = NULL;
truedata = depr.appargs;
} else {
/* Preferred syntax */
AST_NONSTANDARD_APP_ARGS(expr, parse, '?');
if (ast_strlen_zero(expr.remainder)) {
ast_log(LOG_ERROR, "Usage: ExecIf(<expr>?<appiftrue>(<args>)[:<appiffalse>(<args)])\n");
return -1;
}
AST_NONSTANDARD_APP_ARGS(apps, expr.remainder, ':');
if (apps.t && (truedata = strchr(apps.t, '('))) {
*truedata++ = '\0';
if ((end = strrchr(truedata, ')'))) {
*end = '\0';
}
}
if (apps.f && (falsedata = strchr(apps.f, '('))) {
*falsedata++ = '\0';
if ((end = strrchr(falsedata, ')'))) {
*end = '\0';
}
}
}
if (pbx_checkcondition(expr.expr)) {

View File

@@ -31,11 +31,18 @@
* \ingroup applications
*/
/*** MODULEINFO
<depend>working_fork</depend>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include <signal.h>
#ifdef HAVE_CAP
#include <sys/capability.h>
#endif /* HAVE_CAP */
#include "asterisk/lock.h"
#include "asterisk/file.h"
@@ -258,28 +265,20 @@ static void ast_eivr_getvariable(struct ast_channel *chan, char *data, char *out
static void ast_eivr_setvariable(struct ast_channel *chan, char *data)
{
char buf[1024];
char *value;
char *inbuf, *variable;
char *inbuf = ast_strdupa(data), *variable;
int j;
for (j = 1, inbuf = data; ; j++, inbuf = NULL) {
variable = strsep(&inbuf, ",");
ast_chan_log(LOG_DEBUG, chan, "Setting up a variable: %s\n", variable);
if(variable) {
/* variable contains "varname=value" */
ast_copy_string(buf, variable, sizeof(buf));
value = strchr(buf, '=');
if(!value)
value="";
else
*value++ = '\0';
pbx_builtin_setvar_helper(chan, buf, value);
for (variable = strsep(&inbuf, ","); variable; variable = strsep(&inbuf, ",")) {
ast_debug(1, "Setting up a variable: %s\n", variable);
/* variable contains "varname=value" */
value = strchr(variable, '=');
if (!value) {
value = "";
} else {
*value++ = '\0';
}
else
break;
pbx_builtin_setvar_helper(chan, variable, value);
}
};
@@ -304,7 +303,7 @@ static int app_exec(struct ast_channel *chan, void *data)
int res = -1;
int gen_active = 0;
int pid;
char *buf, *pipe_delim_argbuf, *pdargbuf_ptr;
char *buf, *comma_delim_argbuf;
struct ivr_localuser foo = {
.playlist = AST_LIST_HEAD_INIT_VALUE,
.finishlist = AST_LIST_HEAD_INIT_VALUE,
@@ -330,9 +329,7 @@ static int app_exec(struct ast_channel *chan, void *data)
AST_STANDARD_APP_ARGS(args, buf);
/* copy args and replace commas with pipes */
pipe_delim_argbuf = ast_strdupa(data);
while((pdargbuf_ptr = strchr(pipe_delim_argbuf, ',')) != NULL)
pdargbuf_ptr[0] = '|';
comma_delim_argbuf = ast_strdupa(data);
if (pipe(child_stdin)) {
ast_chan_log(LOG_WARNING, chan, "Could not create pipe for child input: %s\n", strerror(errno));
@@ -364,6 +361,15 @@ static int app_exec(struct ast_channel *chan, void *data)
if (!pid) {
/* child process */
int i;
#ifdef HAVE_CAP
cap_t cap = cap_from_text("cap_net_admin-eip");
if (cap_set_proc(cap)) {
/* Careful with order! Logging cannot happen after we close FDs */
ast_log(LOG_WARNING, "Unable to remove capabilities.\n");
}
cap_free(cap);
#endif
signal(SIGPIPE, SIG_DFL);
pthread_sigmask(SIG_UNBLOCK, &fullset, NULL);
@@ -388,7 +394,7 @@ static int app_exec(struct ast_channel *chan, void *data)
child_stdout[1] = 0;
close(child_stderr[1]);
child_stderr[1] = 0;
res = eivr_comm(chan, u, child_stdin[1], child_stdout[0], child_stderr[0], pipe_delim_argbuf);
res = eivr_comm(chan, u, child_stdin[1], child_stdout[0], child_stderr[0], comma_delim_argbuf);
exit:
if (gen_active)
@@ -432,6 +438,7 @@ static int eivr_comm(struct ast_channel *chan, struct ivr_localuser *u,
struct ast_channel *rchan;
char *command;
int res = -1;
int test_available_fd = -1;
FILE *eivr_commands = NULL;
FILE *eivr_errors = NULL;
@@ -451,6 +458,8 @@ static int eivr_comm(struct ast_channel *chan, struct ivr_localuser *u,
goto exit;
}
}
test_available_fd = open("/dev/null", O_RDONLY);
setvbuf(eivr_events, NULL, _IONBF, 0);
setvbuf(eivr_commands, NULL, _IONBF, 0);
@@ -477,7 +486,7 @@ static int eivr_comm(struct ast_channel *chan, struct ivr_localuser *u,
errno = 0;
exception = 0;
rchan = ast_waitfor_nandfds(&chan, 1, waitfds, 2, &exception, &ready_fd, &ms);
rchan = ast_waitfor_nandfds(&chan, 1, waitfds, (eivr_errors_fd == 0) ? 1 : 2, &exception, &ready_fd, &ms);
if (!AST_LIST_EMPTY(&u->finishlist)) {
AST_LIST_LOCK(&u->finishlist);
@@ -522,7 +531,7 @@ static int eivr_comm(struct ast_channel *chan, struct ivr_localuser *u,
} else if (ready_fd == eivr_commands_fd) {
char input[1024];
if (exception || feof(eivr_commands)) {
if (exception || (dup2(eivr_commands_fd, test_available_fd) == -1) || feof(eivr_commands)) {
ast_chan_log(LOG_WARNING, chan, "Child process went away\n");
res = -1;
break;
@@ -633,7 +642,11 @@ static int eivr_comm(struct ast_channel *chan, struct ivr_localuser *u,
exit:
if (eivr_events)
if (test_available_fd > -1) {
close(test_available_fd);
}
if (eivr_events)
fclose(eivr_events);
if (eivr_commands)

831
apps/app_fax.c Normal file
View File

@@ -0,0 +1,831 @@
/*
* Asterisk -- A telephony toolkit for Linux.
*
* Simple fax applications
*
* 2007-2008, Dmitry Andrianov <asterisk@dima.spb.ru>
*
* Code based on original implementation by Steve Underwood <steveu@coppice.org>
*
* This program is free software, distributed under the terms of
* the GNU General Public License
*
*/
/*** MODULEINFO
<depend>spandsp</depend>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include <string.h>
#include <stdlib.h>
#include <stdio.h>
#include <inttypes.h>
#include <pthread.h>
#include <errno.h>
#include <tiffio.h>
#include <spandsp.h>
#ifdef HAVE_SPANDSP_EXPOSE_H
#include <spandsp/expose.h>
#endif
#include <spandsp/version.h>
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/app.h"
#include "asterisk/dsp.h"
#include "asterisk/module.h"
#include "asterisk/manager.h"
static char *app_sndfax_name = "SendFAX";
static char *app_sndfax_synopsis = "Send a FAX";
static char *app_sndfax_desc =
" SendFAX(filename[|options]):\n"
"Send a given TIFF file to the channel as a FAX.\n"
"The option string may contain zero or more of the following characters:\n"
" 'a' - makes the application behave as an answering machine\n"
" The default behaviour is to behave as a calling machine.\n"
"\n"
"This application uses following variables:\n"
" LOCALSTATIONID to identify itself to the remote end.\n"
" LOCALHEADERINFO to generate a header line on each page.\n"
"\n"
"This application sets the following channel variables upon completion:\n"
" FAXSTATUS - status of operation:\n"
" SUCCESS | FAILED\n"
" FAXERROR - Error when FAILED\n"
" FAXMODE - Mode used:\n"
" audio | T38\n"
" REMOTESTATIONID - CSID of the remote side.\n"
" FAXPAGES - number of pages sent.\n"
" FAXBITRATE - transmition rate.\n"
" FAXRESOLUTION - resolution.\n"
"\n"
"Returns -1 in case of user hang up or any channel error.\n"
"Returns 0 on success.\n";
static char *app_rcvfax_name = "ReceiveFAX";
static char *app_rcvfax_synopsis = "Receive a FAX";
static char *app_rcvfax_desc =
" ReceiveFAX(filename[|options]):\n"
"Receives a fax from the channel into the given filename overwriting\n"
"the file if it already exists. File created will have TIFF format.\n"
"The option string may contain zero or more of the following characters:\n"
" 'c' -- makes the application behave as a calling machine\n"
" The default behaviour is to behave as an answering machine.\n"
"\n"
"This application uses following variables:\n"
" LOCALSTATIONID to identify itself to the remote end.\n"
" LOCALHEADERINFO to generate a header line on each page.\n"
"\n"
"This application sets the following channel variables upon completion:\n"
" FAXSTATUS - status of operation:\n"
" SUCCESS | FAILED\n"
" FAXERROR - Error when FAILED\n"
" FAXMODE - Mode used:\n"
" audio | T38\n"
" REMOTESTATIONID - CSID of the remote side.\n"
" FAXPAGES - number of pages sent.\n"
" FAXBITRATE - transmition rate.\n"
" FAXRESOLUTION - resolution.\n"
"\n"
"Returns -1 in case of user hang up or any channel error.\n"
"Returns 0 on success.\n";
#define MAX_SAMPLES 240
/* Watchdog. I have seen situations when remote fax disconnects (because of poor line
quality) while SpanDSP continues staying in T30_STATE_IV_CTC state forever.
To avoid this, we terminate when we see that T30 state does not change for 5 minutes.
We also terminate application when more than 30 minutes passed regardless of
state changes. This is just a precaution measure - no fax should take that long */
#define WATCHDOG_TOTAL_TIMEOUT 30 * 60
#define WATCHDOG_STATE_TIMEOUT 5 * 60
typedef struct {
struct ast_channel *chan;
enum ast_t38_state t38state; /* T38 state of the channel */
int direction; /* Fax direction: 0 - receiving, 1 - sending */
int caller_mode;
char *file_name;
struct ast_control_t38_parameters t38parameters;
volatile int finished;
} fax_session;
static void span_message(int level, const char *msg)
{
if (level == SPAN_LOG_ERROR) {
ast_log(LOG_ERROR, "%s", msg);
} else if (level == SPAN_LOG_WARNING) {
ast_log(LOG_WARNING, "%s", msg);
} else {
ast_log(LOG_DEBUG, "%s", msg);
}
}
static int t38_tx_packet_handler(t38_core_state_t *s, void *user_data, const uint8_t *buf, int len, int count)
{
struct ast_channel *chan = (struct ast_channel *) user_data;
struct ast_frame outf = {
.frametype = AST_FRAME_MODEM,
.subclass = AST_MODEM_T38,
.src = __FUNCTION__,
};
/* TODO: Asterisk does not provide means of resending the same packet multiple
times so count is ignored at the moment */
AST_FRAME_SET_BUFFER(&outf, buf, 0, len);
if (ast_write(chan, &outf) < 0) {
ast_log(LOG_WARNING, "Unable to write frame to channel; %s\n", strerror(errno));
return -1;
}
return 0;
}
static void phase_e_handler(t30_state_t *f, void *user_data, int result)
{
const char *local_ident;
const char *far_ident;
char buf[20];
fax_session *s = (fax_session *) user_data;
t30_stats_t stat;
int pages_transferred;
ast_debug(1, "Fax phase E handler. result=%d\n", result);
t30_get_transfer_statistics(f, &stat);
s = (fax_session *) user_data;
if (result != T30_ERR_OK) {
s->finished = -1;
/* FAXSTATUS is already set to FAILED */
pbx_builtin_setvar_helper(s->chan, "FAXERROR", t30_completion_code_to_str(result));
ast_log(LOG_WARNING, "Error transmitting fax. result=%d: %s.\n", result, t30_completion_code_to_str(result));
return;
}
s->finished = 1;
local_ident = t30_get_tx_ident(f);
far_ident = t30_get_rx_ident(f);
pbx_builtin_setvar_helper(s->chan, "FAXSTATUS", "SUCCESS");
pbx_builtin_setvar_helper(s->chan, "FAXERROR", NULL);
pbx_builtin_setvar_helper(s->chan, "REMOTESTATIONID", far_ident);
#if SPANDSP_RELEASE_DATE >= 20090220
pages_transferred = (s->direction) ? stat.pages_tx : stat.pages_rx;
#else
pages_transferred = stat.pages_transferred;
#endif
snprintf(buf, sizeof(buf), "%d", pages_transferred);
pbx_builtin_setvar_helper(s->chan, "FAXPAGES", buf);
snprintf(buf, sizeof(buf), "%d", stat.y_resolution);
pbx_builtin_setvar_helper(s->chan, "FAXRESOLUTION", buf);
snprintf(buf, sizeof(buf), "%d", stat.bit_rate);
pbx_builtin_setvar_helper(s->chan, "FAXBITRATE", buf);
ast_debug(1, "Fax transmitted successfully.\n");
ast_debug(1, " Remote station ID: %s\n", far_ident);
ast_debug(1, " Pages transferred: %d\n", pages_transferred);
ast_debug(1, " Image resolution: %d x %d\n", stat.x_resolution, stat.y_resolution);
ast_debug(1, " Transfer Rate: %d\n", stat.bit_rate);
manager_event(EVENT_FLAG_CALL,
s->direction ? "FaxSent" : "FaxReceived",
"Channel: %s\r\n"
"Exten: %s\r\n"
"CallerID: %s\r\n"
"RemoteStationID: %s\r\n"
"LocalStationID: %s\r\n"
"PagesTransferred: %d\r\n"
"Resolution: %d\r\n"
"TransferRate: %d\r\n"
"FileName: %s\r\n",
s->chan->name,
s->chan->exten,
S_OR(s->chan->cid.cid_num, ""),
far_ident,
local_ident,
pages_transferred,
stat.y_resolution,
stat.bit_rate,
s->file_name);
}
/* === Helper functions to configure fax === */
/* Setup SPAN logging according to Asterisk debug level */
static int set_logging(logging_state_t *state)
{
int level = SPAN_LOG_WARNING + option_debug;
span_log_set_message_handler(state, span_message);
span_log_set_level(state, SPAN_LOG_SHOW_SEVERITY | SPAN_LOG_SHOW_PROTOCOL | level);
return 0;
}
static void set_local_info(t30_state_t *state, fax_session *s)
{
const char *x;
x = pbx_builtin_getvar_helper(s->chan, "LOCALSTATIONID");
if (!ast_strlen_zero(x))
t30_set_tx_ident(state, x);
x = pbx_builtin_getvar_helper(s->chan, "LOCALHEADERINFO");
if (!ast_strlen_zero(x))
t30_set_tx_page_header_info(state, x);
}
static void set_file(t30_state_t *state, fax_session *s)
{
if (s->direction)
t30_set_tx_file(state, s->file_name, -1, -1);
else
t30_set_rx_file(state, s->file_name, -1);
}
static void set_ecm(t30_state_t *state, int ecm)
{
t30_set_ecm_capability(state, ecm);
t30_set_supported_compressions(state, T30_SUPPORT_T4_1D_COMPRESSION | T30_SUPPORT_T4_2D_COMPRESSION | T30_SUPPORT_T6_COMPRESSION);
}
/* === Generator === */
/* This function is only needed to return passed params so
generator_activate will save it to channel's generatordata */
static void *fax_generator_alloc(struct ast_channel *chan, void *params)
{
return params;
}
static int fax_generator_generate(struct ast_channel *chan, void *data, int len, int samples)
{
fax_state_t *fax = (fax_state_t*) data;
uint8_t buffer[AST_FRIENDLY_OFFSET + MAX_SAMPLES * sizeof(uint16_t)];
int16_t *buf = (int16_t *) (buffer + AST_FRIENDLY_OFFSET);
struct ast_frame outf = {
.frametype = AST_FRAME_VOICE,
.subclass = AST_FORMAT_SLINEAR,
.src = __FUNCTION__,
};
if (samples > MAX_SAMPLES) {
ast_log(LOG_WARNING, "Only generating %d samples, where %d requested\n", MAX_SAMPLES, samples);
samples = MAX_SAMPLES;
}
if ((len = fax_tx(fax, buf, samples)) > 0) {
outf.samples = len;
AST_FRAME_SET_BUFFER(&outf, buffer, AST_FRIENDLY_OFFSET, len * sizeof(int16_t));
if (ast_write(chan, &outf) < 0) {
ast_log(LOG_WARNING, "Failed to write frame to '%s': %s\n", chan->name, strerror(errno));
return -1;
}
}
return 0;
}
struct ast_generator generator = {
alloc: fax_generator_alloc,
generate: fax_generator_generate,
};
/* === Transmission === */
static int transmit_audio(fax_session *s)
{
int res = -1;
int original_read_fmt = AST_FORMAT_SLINEAR;
int original_write_fmt = AST_FORMAT_SLINEAR;
fax_state_t fax;
t30_state_t *t30state;
struct ast_frame *inf = NULL;
int last_state = 0;
struct timeval now, start, state_change;
enum ast_t38_state t38_state;
struct ast_control_t38_parameters t38_parameters = { .version = 0,
.max_ifp = 800,
.rate = AST_T38_RATE_9600,
.rate_management = AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF,
.fill_bit_removal = 1,
.transcoding_mmr = 1,
.transcoding_jbig = 1,
};
/* if in receive mode, try to use T.38 */
if (!s->direction) {
/* check if we are already in T.38 mode (unlikely), or if we can request
* a switch... if so, request it now and wait for the result, rather
* than starting an audio FAX session that will have to be cancelled
*/
if ((t38_state = ast_channel_get_t38_state(s->chan)) == T38_STATE_NEGOTIATED) {
return 1;
} else if ((t38_state != T38_STATE_UNAVAILABLE) &&
(t38_parameters.request_response = AST_T38_REQUEST_NEGOTIATE,
(ast_indicate_data(s->chan, AST_CONTROL_T38_PARAMETERS, &t38_parameters, sizeof(t38_parameters)) == 0))) {
/* wait up to five seconds for negotiation to complete */
unsigned int timeout = 5000;
int ms;
ast_debug(1, "Negotiating T.38 for receive on %s\n", s->chan->name);
while (timeout > 0) {
ms = ast_waitfor(s->chan, 1000);
if (ms < 0) {
ast_log(LOG_WARNING, "something bad happened while channel '%s' was polling.\n", s->chan->name);
return -1;
}
if (!ms) {
/* nothing happened */
if (timeout > 0) {
timeout -= 1000;
continue;
} else {
ast_log(LOG_WARNING, "channel '%s' timed-out during the T.38 negotiation.\n", s->chan->name);
break;
}
}
if (!(inf = ast_read(s->chan))) {
return -1;
}
if ((inf->frametype == AST_FRAME_CONTROL) &&
(inf->subclass == AST_CONTROL_T38_PARAMETERS) &&
(inf->datalen == sizeof(t38_parameters))) {
struct ast_control_t38_parameters *parameters = inf->data;
switch (parameters->request_response) {
case AST_T38_NEGOTIATED:
ast_debug(1, "Negotiated T.38 for receive on %s\n", s->chan->name);
res = 1;
break;
case AST_T38_REFUSED:
ast_log(LOG_WARNING, "channel '%s' refused to negotiate T.38\n", s->chan->name);
break;
default:
ast_log(LOG_ERROR, "channel '%s' failed to negotiate T.38\n", s->chan->name);
break;
}
ast_frfree(inf);
if (res == 1) {
return 1;
} else {
break;
}
}
ast_frfree(inf);
}
}
}
#if SPANDSP_RELEASE_DATE >= 20080725
/* for spandsp shaphots 0.0.6 and higher */
t30state = &fax.t30;
#else
/* for spandsp release 0.0.5 */
t30state = &fax.t30_state;
#endif
original_read_fmt = s->chan->readformat;
if (original_read_fmt != AST_FORMAT_SLINEAR) {
res = ast_set_read_format(s->chan, AST_FORMAT_SLINEAR);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set to linear read mode, giving up\n");
goto done;
}
}
original_write_fmt = s->chan->writeformat;
if (original_write_fmt != AST_FORMAT_SLINEAR) {
res = ast_set_write_format(s->chan, AST_FORMAT_SLINEAR);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set to linear write mode, giving up\n");
goto done;
}
}
/* Initialize T30 terminal */
fax_init(&fax, s->caller_mode);
/* Setup logging */
set_logging(&fax.logging);
set_logging(&t30state->logging);
/* Configure terminal */
set_local_info(t30state, s);
set_file(t30state, s);
set_ecm(t30state, TRUE);
fax_set_transmit_on_idle(&fax, TRUE);
t30_set_phase_e_handler(t30state, phase_e_handler, s);
start = state_change = ast_tvnow();
ast_activate_generator(s->chan, &generator, &fax);
while (!s->finished) {
res = ast_waitfor(s->chan, 20);
if (res < 0)
break;
else if (res > 0)
res = 0;
inf = ast_read(s->chan);
if (inf == NULL) {
ast_debug(1, "Channel hangup\n");
res = -1;
break;
}
ast_debug(10, "frame %d/%d, len=%d\n", inf->frametype, inf->subclass, inf->datalen);
/* Check the frame type. Format also must be checked because there is a chance
that a frame in old format was already queued before we set chanel format
to slinear so it will still be received by ast_read */
if (inf->frametype == AST_FRAME_VOICE && inf->subclass == AST_FORMAT_SLINEAR) {
if (fax_rx(&fax, inf->data, inf->samples) < 0) {
/* I know fax_rx never returns errors. The check here is for good style only */
ast_log(LOG_WARNING, "fax_rx returned error\n");
res = -1;
break;
}
/* Watchdog */
if (last_state != t30state->state) {
state_change = ast_tvnow();
last_state = t30state->state;
}
} else if ((inf->frametype == AST_FRAME_CONTROL) &&
(inf->subclass == AST_CONTROL_T38_PARAMETERS)) {
struct ast_control_t38_parameters *parameters = inf->data;
if (parameters->request_response == AST_T38_NEGOTIATED) {
/* T38 switchover completed */
s->t38parameters = *parameters;
ast_debug(1, "T38 negotiated, finishing audio loop\n");
res = 1;
break;
} else if (parameters->request_response == AST_T38_REQUEST_NEGOTIATE) {
t38_parameters.request_response = AST_T38_NEGOTIATED;
ast_debug(1, "T38 request received, accepting\n");
/* Complete T38 switchover */
ast_indicate_data(s->chan, AST_CONTROL_T38_PARAMETERS, &t38_parameters, sizeof(t38_parameters));
/* Do not break audio loop, wait until channel driver finally acks switchover
* with AST_T38_NEGOTIATED
*/
}
}
ast_frfree(inf);
inf = NULL;
/* Watchdog */
now = ast_tvnow();
if (ast_tvdiff_sec(now, start) > WATCHDOG_TOTAL_TIMEOUT || ast_tvdiff_sec(now, state_change) > WATCHDOG_STATE_TIMEOUT) {
ast_log(LOG_WARNING, "It looks like we hung. Aborting.\n");
res = -1;
break;
}
}
ast_debug(1, "Loop finished, res=%d\n", res);
if (inf)
ast_frfree(inf);
ast_deactivate_generator(s->chan);
/* If we are switching to T38, remove phase E handler. Otherwise it will be executed
by t30_terminate, display diagnostics and set status variables although no transmittion
has taken place yet. */
if (res > 0) {
t30_set_phase_e_handler(t30state, NULL, NULL);
}
t30_terminate(t30state);
fax_release(&fax);
done:
if (original_write_fmt != AST_FORMAT_SLINEAR) {
if (ast_set_write_format(s->chan, original_write_fmt) < 0)
ast_log(LOG_WARNING, "Unable to restore write format on '%s'\n", s->chan->name);
}
if (original_read_fmt != AST_FORMAT_SLINEAR) {
if (ast_set_read_format(s->chan, original_read_fmt) < 0)
ast_log(LOG_WARNING, "Unable to restore read format on '%s'\n", s->chan->name);
}
return res;
}
static int transmit_t38(fax_session *s)
{
int res = 0;
t38_terminal_state_t t38;
struct ast_frame *inf = NULL;
int last_state = 0;
struct timeval now, start, state_change, last_frame;
t30_state_t *t30state;
t38_core_state_t *t38state;
#if SPANDSP_RELEASE_DATE >= 20080725
/* for spandsp shaphots 0.0.6 and higher */
t30state = &t38.t30;
t38state = &t38.t38_fe.t38;
#else
/* for spandsp releases 0.0.5 */
t30state = &t38.t30_state;
t38state = &t38.t38;
#endif
/* Initialize terminal */
memset(&t38, 0, sizeof(t38));
if (t38_terminal_init(&t38, s->caller_mode, t38_tx_packet_handler, s->chan) == NULL) {
ast_log(LOG_WARNING, "Unable to start T.38 termination.\n");
return -1;
}
t38_set_max_datagram_size(t38state, s->t38parameters.max_ifp);
if (s->t38parameters.fill_bit_removal) {
t38_set_fill_bit_removal(t38state, TRUE);
}
if (s->t38parameters.transcoding_mmr) {
t38_set_mmr_transcoding(t38state, TRUE);
}
if (s->t38parameters.transcoding_jbig) {
t38_set_jbig_transcoding(t38state, TRUE);
}
/* Setup logging */
set_logging(&t38.logging);
set_logging(&t30state->logging);
set_logging(&t38state->logging);
/* Configure terminal */
set_local_info(t30state, s);
set_file(t30state, s);
set_ecm(t30state, TRUE);
t30_set_phase_e_handler(t30state, phase_e_handler, s);
now = start = state_change = ast_tvnow();
while (!s->finished) {
res = ast_waitfor(s->chan, 20);
if (res < 0)
break;
else if (res > 0)
res = 0;
last_frame = now;
now = ast_tvnow();
t38_terminal_send_timeout(&t38, ast_tvdiff_us(now, last_frame) / (1000000 / 8000));
inf = ast_read(s->chan);
if (inf == NULL) {
ast_debug(1, "Channel hangup\n");
res = -1;
break;
}
ast_debug(10, "frame %d/%d, len=%d\n", inf->frametype, inf->subclass, inf->datalen);
if (inf->frametype == AST_FRAME_MODEM && inf->subclass == AST_MODEM_T38) {
t38_core_rx_ifp_packet(t38state, inf->data, inf->datalen, inf->seqno);
/* Watchdog */
if (last_state != t30state->state) {
state_change = ast_tvnow();
last_state = t30state->state;
}
} else if (inf->frametype == AST_FRAME_CONTROL && inf->subclass == AST_CONTROL_T38_PARAMETERS) {
struct ast_control_t38_parameters *parameters = inf->data;
if (parameters->request_response == AST_T38_TERMINATED) {
ast_debug(1, "T38 down, finishing\n");
break;
}
}
ast_frfree(inf);
inf = NULL;
/* Watchdog */
if (ast_tvdiff_sec(now, start) > WATCHDOG_TOTAL_TIMEOUT || ast_tvdiff_sec(now, state_change) > WATCHDOG_STATE_TIMEOUT) {
ast_log(LOG_WARNING, "It looks like we hung. Aborting.\n");
res = -1;
break;
}
}
ast_debug(1, "Loop finished, res=%d\n", res);
if (inf)
ast_frfree(inf);
t30_terminate(t30state);
t38_terminal_release(&t38);
return res;
}
static int transmit(fax_session *s)
{
int res = 0;
/* Clear all channel variables which to be set by the application.
Pre-set status to error so in case of any problems we can just leave */
pbx_builtin_setvar_helper(s->chan, "FAXSTATUS", "FAILED");
pbx_builtin_setvar_helper(s->chan, "FAXERROR", "Channel problems");
pbx_builtin_setvar_helper(s->chan, "FAXMODE", NULL);
pbx_builtin_setvar_helper(s->chan, "REMOTESTATIONID", NULL);
pbx_builtin_setvar_helper(s->chan, "FAXPAGES", NULL);
pbx_builtin_setvar_helper(s->chan, "FAXRESOLUTION", NULL);
pbx_builtin_setvar_helper(s->chan, "FAXBITRATE", NULL);
if (s->chan->_state != AST_STATE_UP) {
/* Shouldn't need this, but checking to see if channel is already answered
* Theoretically asterisk should already have answered before running the app */
res = ast_answer(s->chan);
if (res) {
ast_log(LOG_WARNING, "Could not answer channel '%s'\n", s->chan->name);
return res;
}
}
s->t38state = ast_channel_get_t38_state(s->chan);
if (s->t38state != T38_STATE_NEGOTIATED) {
/* T38 is not negotiated on the channel yet. First start regular transmission. If it switches to T38, follow */
pbx_builtin_setvar_helper(s->chan, "FAXMODE", "audio");
res = transmit_audio(s);
if (res > 0) {
/* transmit_audio reports switchover to T38. Update t38state */
s->t38state = ast_channel_get_t38_state(s->chan);
if (s->t38state != T38_STATE_NEGOTIATED) {
ast_log(LOG_ERROR, "Audio loop reports T38 switchover but t38state != T38_STATE_NEGOTIATED\n");
}
}
}
if (s->t38state == T38_STATE_NEGOTIATED) {
pbx_builtin_setvar_helper(s->chan, "FAXMODE", "T38");
res = transmit_t38(s);
}
if (res) {
ast_log(LOG_WARNING, "Transmission error\n");
res = -1;
} else if (s->finished < 0) {
ast_log(LOG_WARNING, "Transmission failed\n");
} else if (s->finished > 0) {
ast_debug(1, "Transmission finished Ok\n");
}
return res;
}
/* === Application functions === */
static int sndfax_exec(struct ast_channel *chan, void *data)
{
int res = 0;
char *parse;
fax_session session = { 0, };
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(file_name);
AST_APP_ARG(options);
);
if (chan == NULL) {
ast_log(LOG_ERROR, "Fax channel is NULL. Giving up.\n");
return -1;
}
/* The next few lines of code parse out the filename and header from the input string */
if (ast_strlen_zero(data)) {
/* No data implies no filename or anything is present */
ast_log(LOG_ERROR, "SendFAX requires an argument (filename)\n");
return -1;
}
parse = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, parse);
session.caller_mode = TRUE;
if (args.options) {
if (strchr(args.options, 'a'))
session.caller_mode = FALSE;
}
/* Done parsing */
session.direction = 1;
session.file_name = args.file_name;
session.chan = chan;
session.finished = 0;
res = transmit(&session);
return res;
}
static int rcvfax_exec(struct ast_channel *chan, void *data)
{
int res = 0;
char *parse;
fax_session session;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(file_name);
AST_APP_ARG(options);
);
if (chan == NULL) {
ast_log(LOG_ERROR, "Fax channel is NULL. Giving up.\n");
return -1;
}
/* The next few lines of code parse out the filename and header from the input string */
if (ast_strlen_zero(data)) {
/* No data implies no filename or anything is present */
ast_log(LOG_ERROR, "ReceiveFAX requires an argument (filename)\n");
return -1;
}
parse = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, parse);
session.caller_mode = FALSE;
if (args.options) {
if (strchr(args.options, 'c'))
session.caller_mode = TRUE;
}
/* Done parsing */
session.direction = 0;
session.file_name = args.file_name;
session.chan = chan;
session.finished = 0;
res = transmit(&session);
return res;
}
static int unload_module(void)
{
int res;
res = ast_unregister_application(app_sndfax_name);
res |= ast_unregister_application(app_rcvfax_name);
return res;
}
static int load_module(void)
{
int res ;
res = ast_register_application(app_sndfax_name, sndfax_exec, app_sndfax_synopsis, app_sndfax_desc);
res |= ast_register_application(app_rcvfax_name, rcvfax_exec, app_rcvfax_synopsis, app_rcvfax_desc);
/* The default SPAN message handler prints to stderr. It is something we do not want */
span_set_message_handler(NULL);
return res;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Simple FAX Application",
.load = load_module,
.unload = unload_module,
);

View File

@@ -27,6 +27,10 @@
* \ingroup applications
*/
/*** MODULEINFO
<depend>working_fork</depend>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
@@ -38,6 +42,10 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include <signal.h>
#include <fcntl.h>
#include <ctype.h>
#include <errno.h>
#ifdef HAVE_CAP
#include <sys/capability.h>
#endif /* HAVE_CAP */
#include "asterisk/file.h"
#include "asterisk/channel.h"
@@ -122,6 +130,9 @@ static int send_waveform_to_fd(char *waveform, int length, int fd)
char c;
#endif
sigset_t fullset, oldset;
#ifdef HAVE_CAP
cap_t cap;
#endif
sigfillset(&fullset);
pthread_sigmask(SIG_BLOCK, &fullset, &oldset);
@@ -133,6 +144,15 @@ static int send_waveform_to_fd(char *waveform, int length, int fd)
pthread_sigmask(SIG_SETMASK, &oldset, NULL);
return res;
}
#ifdef HAVE_CAP
cap = cap_from_text("cap_net_admin-eip");
if (cap_set_proc(cap)) {
/* Careful with order! Logging cannot happen after we close FDs */
ast_log(LOG_WARNING, "Unable to remove capabilities.\n");
}
cap_free(cap);
#endif
for (x = 0; x < 256; x++) {
if (x != fd)
close(x);
@@ -148,7 +168,11 @@ static int send_waveform_to_fd(char *waveform, int length, int fd)
*(waveform + x) = c;
}
#endif
write(fd, waveform, length);
if (write(fd, waveform, length) < 0) {
ast_log(LOG_WARNING, "write() failed: %s\n", strerror(errno));
}
close(fd);
exit(0);
}
@@ -327,11 +351,21 @@ static int festival_exec(struct ast_channel *chan, void *vdata)
if (!(cachedir = ast_variable_retrieve(cfg, "general", "cachedir"))) {
cachedir = "/tmp/";
}
data = ast_strdupa(vdata);
AST_STANDARD_APP_ARGS(args, data);
if (!(festivalcommand = ast_variable_retrieve(cfg, "general", "festivalcommand"))) {
festivalcommand = "(tts_textasterisk \"%s\" 'file)(quit)\n";
const char *startcmd = "(tts_textasterisk \"";
const char *endcmd = "\" 'file)(quit)\n";
strln = strlen(startcmd) + strlen(args.text) + strlen(endcmd) + 1;
newfestivalcommand = alloca(strln);
snprintf(newfestivalcommand, strln, "%s%s%s", startcmd, args.text, endcmd);
festivalcommand = newfestivalcommand;
} else { /* This else parses the festivalcommand that we're sent from the config file for \n's, etc */
int i, j;
newfestivalcommand = alloca(strlen(festivalcommand) + 1);
newfestivalcommand = alloca(strlen(festivalcommand) + strlen(args.text) + 1);
for (i = 0, j = 0; i < strlen(festivalcommand); i++) {
if (festivalcommand[i] == '\\' && festivalcommand[i + 1] == 'n') {
@@ -340,6 +374,10 @@ static int festival_exec(struct ast_channel *chan, void *vdata)
} else if (festivalcommand[i] == '\\') {
newfestivalcommand[j++] = festivalcommand[i + 1];
i++;
} else if (festivalcommand[i] == '%' && festivalcommand[i + 1] == 's') {
sprintf(&newfestivalcommand[j], "%s", args.text); /* we know it is big enough */
j += strlen(args.text);
i++;
} else
newfestivalcommand[j++] = festivalcommand[i];
}
@@ -347,9 +385,6 @@ static int festival_exec(struct ast_channel *chan, void *vdata)
festivalcommand = newfestivalcommand;
}
data = ast_strdupa(vdata);
AST_STANDARD_APP_ARGS(args, data);
if (args.interrupt && !strcasecmp(args.interrupt, "any"))
args.interrupt = AST_DIGIT_ANY;
@@ -410,17 +445,25 @@ static int festival_exec(struct ast_channel *chan, void *vdata)
writecache = 1;
strln = strlen(args.text);
ast_debug(1, "line length : %d\n", strln);
write(fdesc, &strln, sizeof(strln));
write(fdesc, args.text, strln);
if (write(fdesc,&strln,sizeof(int)) < 0) {
ast_log(LOG_WARNING, "write() failed: %s\n", strerror(errno));
}
if (write(fdesc,data,strln) < 0) {
ast_log(LOG_WARNING, "write() failed: %s\n", strerror(errno));
}
seekpos = lseek(fdesc, 0, SEEK_CUR);
ast_debug(1, "Seek position : %d\n", seekpos);
}
} else {
read(fdesc, &strln, sizeof(strln));
if (read(fdesc,&strln,sizeof(int)) != sizeof(int)) {
ast_log(LOG_WARNING, "read() failed: %s\n", strerror(errno));
}
ast_debug(1, "Cache file exists, strln=%d, strlen=%d\n", strln, (int)strlen(args.text));
if (strlen(args.text) == strln) {
ast_debug(1, "Size OK\n");
read(fdesc, &bigstring, strln);
if (read(fdesc,&bigstring,strln) != strln) {
ast_log(LOG_WARNING, "read() failed: %s\n", strerror(errno));
}
bigstring[strln] = 0;
if (strcmp(bigstring, args.text) == 0) {
readcache = 1;
@@ -440,7 +483,8 @@ static int festival_exec(struct ast_channel *chan, void *vdata)
} else {
ast_debug(1, "Passing text to festival...\n");
fs = fdopen(dup(fd), "wb");
fprintf(fs, festivalcommand, args.text);
fprintf(fs, "%s", festivalcommand);
fflush(fs);
fclose(fs);
}
@@ -449,7 +493,9 @@ static int festival_exec(struct ast_channel *chan, void *vdata)
if (writecache == 1) {
ast_debug(1, "Writing result to cache...\n");
while ((strln = read(fd, buffer, 16384)) != 0) {
write(fdesc, buffer, strln);
if (write(fdesc,buffer,strln) < 0) {
ast_log(LOG_WARNING, "write() failed: %s\n", strerror(errno));
}
}
close(fd);
close(fdesc);

View File

@@ -18,7 +18,7 @@
/*! \file
*
* \brief App to flash a zap trunk
* \brief App to flash a DAHDI trunk
*
* \author Mark Spencer <markster@digium.com>
*
@@ -26,14 +26,15 @@
*/
/*** MODULEINFO
<depend>zaptel</depend>
<depend>dahdi</depend>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/zapata.h"
#include <sys/ioctl.h>
#include <dahdi/user.h>
#include "asterisk/lock.h"
#include "asterisk/file.h"
@@ -45,22 +46,22 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
static char *app = "Flash";
static char *synopsis = "Flashes a Zap Trunk";
static char *synopsis = "Flashes a DAHDI Trunk";
static char *descrip =
"Performs a flash on a zap trunk. This can be used\n"
"Performs a flash on a DAHDI trunk. This can be used\n"
"to access features provided on an incoming analogue circuit\n"
"such as conference and call waiting. Use with SendDTMF() to\n"
"perform external transfers\n";
static inline int zt_wait_event(int fd)
static inline int dahdi_wait_event(int fd)
{
/* Avoid the silly zt_waitevent which ignores a bunch of events */
/* Avoid the silly dahdi_waitevent which ignores a bunch of events */
int i,j=0;
i = ZT_IOMUX_SIGEVENT;
if (ioctl(fd, ZT_IOMUX, &i) == -1) return -1;
if (ioctl(fd, ZT_GETEVENT, &j) == -1) return -1;
i = DAHDI_IOMUX_SIGEVENT;
if (ioctl(fd, DAHDI_IOMUX, &i) == -1) return -1;
if (ioctl(fd, DAHDI_GETEVENT, &j) == -1) return -1;
return j;
}
@@ -68,23 +69,23 @@ static int flash_exec(struct ast_channel *chan, void *data)
{
int res = -1;
int x;
struct zt_params ztp;
struct dahdi_params dahdip;
if (strcasecmp(chan->tech->type, "Zap")) {
ast_log(LOG_WARNING, "%s is not a Zap channel\n", chan->name);
if (strcasecmp(chan->tech->type, "DAHDI")) {
ast_log(LOG_WARNING, "%s is not a DAHDI channel\n", chan->name);
return -1;
}
memset(&ztp, 0, sizeof(ztp));
res = ioctl(chan->fds[0], ZT_GET_PARAMS, &ztp);
memset(&dahdip, 0, sizeof(dahdip));
res = ioctl(chan->fds[0], DAHDI_GET_PARAMS, &dahdip);
if (!res) {
if (ztp.sigtype & __ZT_SIG_FXS) {
x = ZT_FLASH;
res = ioctl(chan->fds[0], ZT_HOOK, &x);
if (dahdip.sigtype & __DAHDI_SIG_FXS) {
x = DAHDI_FLASH;
res = ioctl(chan->fds[0], DAHDI_HOOK, &x);
if (!res || (errno == EINPROGRESS)) {
if (res) {
/* Wait for the event to finish */
zt_wait_event(chan->fds[0]);
dahdi_wait_event(chan->fds[0]);
}
res = ast_safe_sleep(chan, 1000);
ast_verb(3, "Flashed channel %s\n", chan->name);

View File

@@ -146,7 +146,6 @@ AST_APP_OPTIONS(followme_opts, {
});
static int ynlongest = 0;
static time_t start_time, answer_time, end_time;
static const char *featuredigittostr;
static int featuredigittimeout = 5000; /*!< Feature Digit Timeout */
@@ -659,6 +658,9 @@ static struct ast_channel *wait_for_winner(struct findme_user_listptr *findme_us
case AST_CONTROL_VIDUPDATE:
ast_verb(3, "%s requested a video update, passing it to %s\n", winner->name, caller->name);
break;
case AST_CONTROL_SRCUPDATE:
ast_verb(3, "%s requested a source update, passing it to %s\n", winner->name, caller->name);
break;
case AST_CONTROL_PROCEEDING:
ast_verb(3, "%s is proceeding passing it to %s\n", winner->name,caller->name);
break;
@@ -767,7 +769,6 @@ static void findmeexec(struct fm_args *tpargs)
while (nm) {
ast_debug(2, "Number %s timeout %ld\n", nm->number,nm->timeout);
time(&start_time);
number = ast_strdupa(nm->number);
ast_debug(3, "examining %s\n", number);
@@ -779,9 +780,9 @@ static void findmeexec(struct fm_args *tpargs)
}
if (!strcmp(tpargs->context, ""))
sprintf(dialarg, "%s", number);
snprintf(dialarg, sizeof(dialarg), "%s", number);
else
sprintf(dialarg, "%s@%s", number, tpargs->context);
snprintf(dialarg, sizeof(dialarg), "%s@%s", number, tpargs->context);
tmpuser = ast_calloc(1, sizeof(*tmpuser));
if (!tmpuser) {
@@ -849,7 +850,7 @@ static void findmeexec(struct fm_args *tpargs)
if (winner)
break;
if (!caller) {
if (!caller || ast_check_hangup(caller)) {
tpargs->status = 1;
ast_free(findme_user_list);
return;
@@ -874,9 +875,35 @@ static void findmeexec(struct fm_args *tpargs)
}
static void end_bridge_callback(void *data)
{
char buf[80];
time_t end;
struct ast_channel *chan = data;
time(&end);
ast_channel_lock(chan);
if (chan->cdr->answer.tv_sec) {
snprintf(buf, sizeof(buf), "%ld", end - chan->cdr->answer.tv_sec);
pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", buf);
}
if (chan->cdr->start.tv_sec) {
snprintf(buf, sizeof(buf), "%ld", end - chan->cdr->start.tv_sec);
pbx_builtin_setvar_helper(chan, "DIALEDTIME", buf);
}
ast_channel_unlock(chan);
}
static void end_bridge_callback_data_fixup(struct ast_bridge_config *bconfig, struct ast_channel *originator, struct ast_channel *terminator)
{
bconfig->end_bridge_callback_data = originator;
}
static int app_exec(struct ast_channel *chan, void *data)
{
struct fm_args targs;
struct fm_args targs = { 0, };
struct ast_bridge_config config;
struct call_followme *f;
struct number *nm, *newnm;
@@ -886,7 +913,6 @@ static int app_exec(struct ast_channel *chan, void *data)
int duration = 0;
struct ast_channel *caller;
struct ast_channel *outbound;
static char toast[80];
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(followmeid);
@@ -949,6 +975,11 @@ static int app_exec(struct ast_channel *chan, void *data)
}
ast_mutex_unlock(&f->lock);
/* Answer the call */
if (chan->_state != AST_STATE_UP) {
ast_answer(chan);
}
if (ast_test_flag(&targs.followmeflags, FOLLOWMEFLAG_STATUSMSG))
ast_stream_and_wait(chan, targs.statusprompt, "");
@@ -994,7 +1025,10 @@ static int app_exec(struct ast_channel *chan, void *data)
ast_set_flag(&(config.features_callee), AST_FEATURE_REDIRECT);
ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMON);
ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMON);
config.end_bridge_callback = end_bridge_callback;
config.end_bridge_callback_data = chan;
config.end_bridge_callback_data_fixup = end_bridge_callback_data_fixup;
ast_moh_stop(caller);
/* Be sure no generators are left on it */
ast_deactivate_generator(caller);
@@ -1005,13 +1039,7 @@ static int app_exec(struct ast_channel *chan, void *data)
ast_hangup(outbound);
goto outrun;
}
time(&answer_time);
res = ast_bridge_call(caller,outbound,&config);
time(&end_time);
snprintf(toast, sizeof(toast), "%ld", (long)(end_time - start_time));
pbx_builtin_setvar_helper(caller, "DIALEDTIME", toast);
snprintf(toast, sizeof(toast), "%ld", (long)(end_time - answer_time));
pbx_builtin_setvar_helper(caller, "ANSWEREDTIME", toast);
if (outbound)
ast_hangup(outbound);
}

View File

@@ -34,6 +34,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/cdr.h"
#include "asterisk/app.h"
#include "asterisk/module.h"
static char *app = "ForkCDR";
@@ -41,12 +42,118 @@ static char *synopsis =
"Forks the Call Data Record";
static char *descrip =
" ForkCDR([options]): Causes the Call Data Record to fork an additional\n"
"cdr record starting from the time of the fork call\n"
"cdr record starting from the time of the fork call. This new cdr record will\n"
"be linked to end of the list of cdr records attached to the channel. The original CDR\n"
"has a LOCKED flag set, which forces most cdr operations to skip it, except\n"
"for the functions that set the answer and end times, which ignore the LOCKED\n"
"flag. This allows all the cdr records in the channel to be 'ended' together\n"
"when the channel is closed.\n"
"The CDR() func (when setting CDR values) normally ignores the LOCKED flag also,\n"
"but has options to vary its behavior. The 'T' option (described below), can\n"
"override this behavior, but beware the risks.\n"
"\n"
"Detailed Behavior Description:\n"
"First, this app finds the last cdr record in the list, and makes\n"
"a copy of it. This new copy will be the newly forked cdr record.\n"
"Next, this new record is linked to the end of the cdr record list.\n"
"Next, The new cdr record is RESET (unless you use an option to prevent this)\n"
"This means that:\n"
" 1. All flags are unset on the cdr record\n"
" 2. the start, end, and answer times are all set to zero.\n"
" 3. the billsec and duration fields are set to zero.\n"
" 4. the start time is set to the current time.\n"
" 5. the disposition is set to NULL.\n"
"Next, unless you specified the 'v' option, all variables will be\n"
"removed from the original cdr record. Thus, the 'v' option allows\n"
"any CDR variables to be replicated to all new forked cdr records.\n"
"Without the 'v' option, the variables on the original are effectively\n"
"moved to the new forked cdr record.\n"
"Next, if the 's' option is set, the provided variable and value\n"
"are set on the original cdr record.\n"
"Next, if the 'a' option is given, and the original cdr record has an\n"
"answer time set, then the new forked cdr record will have its answer\n"
"time set to its start time. If the old answer time were carried forward,\n"
"the answer time would be earlier than the start time, giving strange\n"
"duration and billsec times.\n"
"Next, if the 'd' option was specified, the disposition is copied from\n"
"the original cdr record to the new forked cdr.\n"
"Next, if the 'D' option was specified, the destination channel field\n"
"in the new forked CDR is erased.\n"
"Next, if the 'e' option was specified, the 'end' time for the original\n"
"cdr record is set to the current time. Future hang-up or ending events\n"
"will not override this time stamp.\n"
"Next, If the 'A' option is specified, the original cdr record will have\n"
"it ANS_LOCKED flag set, which prevent future answer events\n"
"from updating the original cdr record's disposition. Normally, an\n"
"'ANSWERED' event would mark all cdr records in the chain as 'ANSWERED'.\n"
"Next, if the 'T' option is specified, the original cdr record will have\n"
"its 'DONT_TOUCH' flag set, which will force the cdr_answer, cdr_end, and\n"
"cdr_setvar functions to leave that cdr record alone.\n"
"And, last but not least, the original cdr record has its LOCKED flag\n"
"set. Almost all internal CDR functions (except for the funcs that set\n"
"the end, and answer times, and set a variable) will honor this flag\n"
"and leave a LOCKED cdr record alone.\n"
"This means that the newly created forked cdr record will be affected\n"
"by events transpiring within Asterisk, with the previously noted\n"
"exceptions.\n"
" Options:\n"
" v - If the option is passed all cdr variables will be passed along also.\n";
" a - update the answer time on the NEW CDR just after it's been inited.\n"
" The new CDR may have been answered already. The reset that forkcdr\n"
" does will erase the answer time. This will bring it back, but\n"
" the answer time will be a copy of the fork/start time. It will\n"
" only do this if the initial cdr was indeed already answered.\n"
" A - Lock the original CDR against the answer time being updated.\n"
" This will allow the disposition on the original CDR to remain the same.\n"
" d - Copy the disposition forward from the old cdr, after the \n"
" init.\n"
" D - Clear the dstchannel on the new CDR after reset.\n"
" e - end the original CDR. Do this after all the neccessary data\n"
" is copied from the original CDR to the new forked CDR.\n"
" R - do NOT reset the new cdr.\n"
" s(name=val) - Set the CDR var 'name' in the original CDR, with value\n"
" 'val'.\n"
" T - Mark the original CDR with a DONT_TOUCH flag. setvar, answer, and end\n"
" cdr funcs will obey this flag; normally they don't honor the LOCKED\n"
" flag set on the original CDR record.\n"
" Beware-- using this flag may cause CDR's not to have their end times\n"
" updated! It is suggested that if you specify this flag, you might\n"
" wish to use the 'e' flag as well!\n"
" v - When the new CDR is forked, it gets a copy of the vars attached\n"
" to the current CDR. The vars attached to the original CDR are removed\n"
" unless this option is specified.\n";
static void ast_cdr_fork(struct ast_channel *chan)
enum {
OPT_SETANS = (1 << 0),
OPT_SETDISP = (1 << 1),
OPT_RESETDEST = (1 << 2),
OPT_ENDCDR = (1 << 3),
OPT_NORESET = (1 << 4),
OPT_KEEPVARS = (1 << 5),
OPT_VARSET = (1 << 6),
OPT_ANSLOCK = (1 << 7),
OPT_DONTOUCH = (1 << 8),
};
enum {
OPT_ARG_VARSET = 0,
/* note: this entry _MUST_ be the last one in the enum */
OPT_ARG_ARRAY_SIZE,
};
AST_APP_OPTIONS(forkcdr_exec_options, {
AST_APP_OPTION('a', OPT_SETANS),
AST_APP_OPTION('A', OPT_ANSLOCK),
AST_APP_OPTION('d', OPT_SETDISP),
AST_APP_OPTION('D', OPT_RESETDEST),
AST_APP_OPTION('e', OPT_ENDCDR),
AST_APP_OPTION('R', OPT_NORESET),
AST_APP_OPTION_ARG('s', OPT_VARSET, OPT_ARG_VARSET),
AST_APP_OPTION('T', OPT_DONTOUCH),
AST_APP_OPTION('v', OPT_KEEPVARS),
});
static void ast_cdr_fork(struct ast_channel *chan, struct ast_flags optflags, char *set)
{
struct ast_cdr *cdr;
struct ast_cdr *newcdr;
@@ -61,27 +168,72 @@ static void ast_cdr_fork(struct ast_channel *chan)
return;
ast_cdr_append(cdr, newcdr);
ast_cdr_reset(newcdr, &flags);
if (!ast_test_flag(&optflags, OPT_NORESET))
ast_cdr_reset(newcdr, &flags);
if (!ast_test_flag(cdr, AST_CDR_FLAG_KEEP_VARS))
ast_cdr_free_vars(cdr, 0);
if (!ast_strlen_zero(set)) {
char *varname = ast_strdupa(set), *varval;
varval = strchr(varname,'=');
if (varval) {
*varval = 0;
varval++;
ast_cdr_setvar(cdr, varname, varval, 0);
}
}
if (ast_test_flag(&optflags, OPT_SETANS) && !ast_tvzero(cdr->answer))
newcdr->answer = newcdr->start;
if (ast_test_flag(&optflags, OPT_SETDISP))
newcdr->disposition = cdr->disposition;
if (ast_test_flag(&optflags, OPT_RESETDEST))
newcdr->dstchannel[0] = 0;
if (ast_test_flag(&optflags, OPT_ENDCDR))
ast_cdr_end(cdr);
if (ast_test_flag(&optflags, OPT_ANSLOCK))
ast_set_flag(cdr, AST_CDR_FLAG_ANSLOCKED);
if (ast_test_flag(&optflags, OPT_DONTOUCH))
ast_set_flag(cdr, AST_CDR_FLAG_DONT_TOUCH);
ast_set_flag(cdr, AST_CDR_FLAG_CHILD | AST_CDR_FLAG_LOCKED);
}
static int forkcdr_exec(struct ast_channel *chan, void *data)
{
int res = 0;
char *argcopy = NULL;
struct ast_flags flags = {0};
char *opts[OPT_ARG_ARRAY_SIZE];
AST_DECLARE_APP_ARGS(arglist,
AST_APP_ARG(options);
);
if (!chan->cdr) {
ast_log(LOG_WARNING, "Channel does not have a CDR\n");
return 0;
}
if (!ast_strlen_zero(data))
ast_set2_flag(chan->cdr, strchr(data, 'v'), AST_CDR_FLAG_KEEP_VARS);
argcopy = ast_strdupa(data);
AST_STANDARD_APP_ARGS(arglist, argcopy);
opts[OPT_ARG_VARSET] = 0;
if (!ast_strlen_zero(arglist.options))
ast_app_parse_options(forkcdr_exec_options, &flags, opts, arglist.options);
ast_cdr_fork(chan);
if (!ast_strlen_zero(data))
ast_set2_flag(chan->cdr, ast_test_flag(&flags, OPT_KEEPVARS), AST_CDR_FLAG_KEEP_VARS);
ast_cdr_fork(chan, flags, opts[OPT_ARG_VARSET]);
return res;
}

View File

@@ -42,7 +42,7 @@ static char *synopsis = "Get ADSI CPE ID";
static char *descrip =
" GetCPEID(): Obtains and displays ADSI CPE ID and other information in order\n"
"to properly setup zapata.conf for on-hook operations.\n";
"to properly setup dahdi.conf for on-hook operations.\n";
static int cpeid_setstatus(struct ast_channel *chan, char *stuff[], int voice)

View File

@@ -27,6 +27,10 @@
* \ingroup applications
*/
/*** MODULEINFO
<depend>working_fork</depend>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
@@ -34,6 +38,9 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include <signal.h>
#include <fcntl.h>
#include <sys/time.h>
#ifdef HAVE_CAP
#include <sys/capability.h>
#endif /* HAVE_CAP */
#include "asterisk/paths.h" /* use ast_config_AST_CONFIG_DIR */
#include "asterisk/lock.h"
@@ -44,8 +51,8 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/module.h"
#include "asterisk/translate.h"
#define ICES "/usr/bin/ices"
#define LOCAL_ICES "/usr/local/bin/ices"
#define path_BIN "/usr/bin/"
#define path_LOCAL "/usr/local/bin/"
static char *app = "ICES";
@@ -54,7 +61,9 @@ static char *synopsis = "Encode and stream using 'ices'";
static char *descrip =
" ICES(config.xml) Streams to an icecast server using ices\n"
"(available separately). A configuration file must be supplied\n"
"for ices (see examples/asterisk-ices.conf). \n";
"for ices (see contrib/asterisk-ices.xml). \n"
"\n"
"- ICES version 2 cient and server required.\n";
static int icesencode(char *filename, int fd)
@@ -62,6 +71,9 @@ static int icesencode(char *filename, int fd)
int res;
int x;
sigset_t fullset, oldset;
#ifdef HAVE_CAP
cap_t cap;
#endif
sigfillset(&fullset);
pthread_sigmask(SIG_BLOCK, &fullset, &oldset);
@@ -78,6 +90,16 @@ static int icesencode(char *filename, int fd)
signal(SIGPIPE, SIG_DFL);
pthread_sigmask(SIG_UNBLOCK, &fullset, NULL);
#ifdef HAVE_CAP
cap = cap_from_text("cap_net_admin-eip");
if (cap_set_proc(cap)) {
/* Careful with order! Logging cannot happen after we close FDs */
ast_log(LOG_WARNING, "Unable to remove capabilities.\n");
}
cap_free(cap);
#endif
if (ast_opt_high_priority)
ast_set_priority(0);
dup2(fd, STDIN_FILENO);
@@ -85,13 +107,23 @@ static int icesencode(char *filename, int fd)
if ((x != STDIN_FILENO) && (x != STDOUT_FILENO))
close(x);
}
/* Most commonly installed in /usr/local/bin */
execl(ICES, "ices", filename, (char *)NULL);
/* But many places has it in /usr/bin */
execl(LOCAL_ICES, "ices", filename, (char *)NULL);
/* As a last-ditch effort, try to use PATH */
/* Most commonly installed in /usr/local/bin
* But many places has it in /usr/bin
* As a last-ditch effort, try to use PATH
*/
execl(path_LOCAL "ices2", "ices", filename, (char *)NULL);
execl(path_BIN "ices2", "ices", filename, (char *)NULL);
execlp("ices2", "ices", filename, (char *)NULL);
ast_debug(1, "Couldn't find ices version 2, attempting to use ices version 1.");
execl(path_LOCAL "ices", "ices", filename, (char *)NULL);
execl(path_BIN "ices", "ices", filename, (char *)NULL);
execlp("ices", "ices", filename, (char *)NULL);
ast_log(LOG_WARNING, "Execute of ices failed\n");
ast_log(LOG_WARNING, "Execute of ices failed, could not find command.\n");
close(fd);
_exit(0);
}
@@ -151,7 +183,6 @@ static int ices_exec(struct ast_channel *chan, void *data)
if (c)
*c = '\0';
res = icesencode(filename, fds[0]);
close(fds[0]);
if (res >= 0) {
pid = res;
for (;;) {
@@ -182,6 +213,7 @@ static int ices_exec(struct ast_channel *chan, void *data)
ast_frfree(f);
}
}
close(fds[0]);
close(fds[1]);
if (pid > -1)

View File

@@ -28,11 +28,15 @@
*
* \arg http://www.jackaudio.org/
*
* \note To install libresample, check it out of the following repository:
* <code>$ svn co http://svn.digium.com/svn/thirdparty/libresample/trunk</code>
*
* \ingroup applications
*/
/*** MODULEINFO
<depend>jack</depend>
<depend>resample</depend>
***/
#include "asterisk.h"
@@ -44,16 +48,17 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include <jack/jack.h>
#include <jack/ringbuffer.h>
#include <libresample.h>
#include "asterisk/module.h"
#include "asterisk/channel.h"
#include "asterisk/strings.h"
#include "asterisk/lock.h"
#include "asterisk/libresample.h"
#include "asterisk/app.h"
#include "asterisk/pbx.h"
#include "asterisk/audiohook.h"
#define RESAMPLE_QUALITY 0
#define RESAMPLE_QUALITY 1
#define RINGBUFFER_SIZE 16384

View File

@@ -42,43 +42,46 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
/* special result value used to force macro exit */
#define MACRO_EXIT_RESULT 1024
#define WAITEXTENWARNING "Use of the application WaitExten within a macro will not function as expected.\n" \
"Please use the Read application in order to read DTMF from a channel currently\n" \
"executing a macro.\n"
static char *descrip =
" Macro(macroname,arg1,arg2...): Executes a macro using the context\n"
"'macro-<macroname>', jumping to the 's' extension of that context and\n"
"executing each step, then returning when the steps end. \n"
"The calling extension, context, and priority are stored in ${MACRO_EXTEN}, \n"
"The calling extension, context, and priority are stored in ${MACRO_EXTEN},\n"
"${MACRO_CONTEXT} and ${MACRO_PRIORITY} respectively. Arguments become\n"
"${ARG1}, ${ARG2}, etc in the macro context.\n"
"If you Goto out of the Macro context, the Macro will terminate and control\n"
"will be returned at the location of the Goto.\n"
"If ${MACRO_OFFSET} is set at termination, Macro will attempt to continue\n"
"at priority MACRO_OFFSET + N + 1 if such a step exists, and N + 1 otherwise.\n"
"Extensions: While a macro is being executed, it becomes the current context.\n"
" This means that if a hangup occurs, for instance, that the macro\n"
" will be searched for an 'h' extension, NOT the context from which\n"
" the macro was called. So, make sure to define all appropriate\n"
" extensions in your macro! (Note: AEL does not use macros)\n"
"WARNING: Because of the way Macro is implemented (it executes the priorities\n"
" contained within it via sub-engine), and a fixed per-thread\n"
" memory stack allowance, macros are limited to 7 levels\n"
" of nesting (macro calling macro calling macro, etc.); It\n"
" may be possible that stack-intensive applications in deeply nested macros\n"
" could cause asterisk to crash earlier than this limit. It is advised that\n"
" if you need to deeply nest macro calls, that you use the Gosub application\n"
" (now allows arguments like a Macro) with explict Return() calls instead.\n";
" contained within it via sub-engine), and a fixed per-thread memory\n"
" stack allowance, macros are limited to 7 levels of nesting (macro\n"
" calling macro calling macro, etc.). It may be possible that\n"
" stack-intensive applications in deeply nested macros could cause\n"
" Asterisk to crash earlier than this limit. It is advised that if you\n"
" need to deeply nest macro calls, that you use the Gosub application\n"
" (now allows arguments like a Macro) with explict Return() calls\n"
" instead.\n"
WAITEXTENWARNING;
static char *if_descrip =
" MacroIf(<expr>?macroname_a[,arg1][:macroname_b[,arg1]])\n"
"Executes macro defined in <macroname_a> if <expr> is true\n"
"(otherwise <macroname_b> if provided)\n"
"Arguments and return values as in application Macro()\n";
"Arguments and return values as in application Macro()\n"
WAITEXTENWARNING;
static char *exclusive_descrip =
" MacroExclusive(macroname,arg1,arg2...):\n"
"Executes macro defined in the context 'macro-macroname'\n"
"Only one call at a time may run the macro.\n"
"(we'll wait if another call is busy executing in the Macro)\n"
"Arguments and return values as in application Macro()\n";
"Arguments and return values as in application Macro()\n"
WAITEXTENWARNING;
static char *exit_descrip =
" MacroExit():\n"
@@ -97,6 +100,30 @@ static char *if_synopsis = "Conditional Macro Implementation";
static char *exclusive_synopsis = "Exclusive Macro Implementation";
static char *exit_synopsis = "Exit From Macro";
static void macro_fixup(void *data, struct ast_channel *old_chan, struct ast_channel *new_chan);
struct ast_datastore_info macro_ds_info = {
.type = "MACRO",
.chan_fixup = macro_fixup,
};
static void macro_fixup(void *data, struct ast_channel *old_chan, struct ast_channel *new_chan)
{
int i;
char varname[10];
pbx_builtin_setvar_helper(new_chan, "MACRO_DEPTH", "0");
pbx_builtin_setvar_helper(new_chan, "MACRO_CONTEXT", NULL);
pbx_builtin_setvar_helper(new_chan, "MACRO_EXTEN", NULL);
pbx_builtin_setvar_helper(new_chan, "MACRO_PRIORITY", NULL);
pbx_builtin_setvar_helper(new_chan, "MACRO_OFFSET", NULL);
for (i = 1; i < 100; i++) {
snprintf(varname, sizeof(varname), "ARG%d", i);
while (pbx_builtin_getvar_helper(new_chan, varname)) {
/* Kill all levels of arguments */
pbx_builtin_setvar_helper(new_chan, varname, NULL);
}
}
}
static struct ast_exten *find_matching_priority(struct ast_context *c, const char *exten, int priority, const char *callerid)
{
@@ -152,18 +179,32 @@ static int _macro_exec(struct ast_channel *chan, void *data, int exclusive)
const char *inhangupc;
int offset, depth = 0, maxdepth = 7;
int setmacrocontext=0;
int autoloopflag, dead = 0, inhangup = 0;
int autoloopflag, inhangup = 0;
char *save_macro_exten;
char *save_macro_context;
char *save_macro_priority;
char *save_macro_offset;
struct ast_datastore *macro_store = ast_channel_datastore_find(chan, &macro_ds_info, NULL);
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "Macro() requires arguments. See \"core show application macro\" for help.\n");
return -1;
}
do {
if (macro_store) {
break;
}
if (!(macro_store = ast_channel_datastore_alloc(&macro_ds_info, NULL))) {
ast_log(LOG_WARNING, "Unable to allocate new datastore.\n");
break;
}
/* Just the existence of this datastore is enough. */
macro_store->inheritance = DATASTORE_INHERIT_FOREVER;
ast_channel_datastore_add(chan, macro_store);
} while (0);
/* does the user want a deeper rabbit hole? */
s = pbx_builtin_getvar_helper(chan, "MACRO_RECURSION");
if (s)
@@ -304,15 +345,9 @@ static int _macro_exec(struct ast_channel *chan, void *data, int exclusive)
case MACRO_EXIT_RESULT:
res = 0;
goto out;
case AST_PBX_KEEPALIVE:
ast_debug(2, "Spawn extension (%s,%s,%d) exited KEEPALIVE in macro %s on '%s'\n", chan->context, chan->exten, chan->priority, macro, chan->name);
ast_verb(2, "Spawn extension (%s, %s, %d) exited KEEPALIVE in macro '%s' on '%s'\n", chan->context, chan->exten, chan->priority, macro, chan->name);
goto out;
break;
default:
ast_debug(2, "Spawn extension (%s,%s,%d) exited non-zero on '%s' in macro '%s'\n", chan->context, chan->exten, chan->priority, chan->name, macro);
ast_verb(2, "Spawn extension (%s, %s, %d) exited non-zero on '%s' in macro '%s'\n", chan->context, chan->exten, chan->priority, chan->name, macro);
dead = 1;
goto out;
}
}
@@ -385,31 +420,30 @@ static int _macro_exec(struct ast_channel *chan, void *data, int exclusive)
chan->priority++;
}
out:
/* Don't let the channel change now. */
ast_channel_lock(chan);
/* Reset the depth back to what it was when the routine was entered (like if we called Macro recursively) */
snprintf(depthc, sizeof(depthc), "%d", depth);
if (!dead) {
pbx_builtin_setvar_helper(chan, "MACRO_DEPTH", depthc);
ast_set2_flag(chan, autoloopflag, AST_FLAG_IN_AUTOLOOP);
}
pbx_builtin_setvar_helper(chan, "MACRO_DEPTH", depthc);
ast_set2_flag(chan, autoloopflag, AST_FLAG_IN_AUTOLOOP);
for (x = 1; x < argc; x++) {
/* Restore old arguments and delete ours */
snprintf(varname, sizeof(varname), "ARG%d", x);
if (oldargs[x]) {
if (!dead)
pbx_builtin_setvar_helper(chan, varname, oldargs[x]);
pbx_builtin_setvar_helper(chan, varname, oldargs[x]);
ast_free(oldargs[x]);
} else if (!dead) {
} else {
pbx_builtin_setvar_helper(chan, varname, NULL);
}
}
/* Restore macro variables */
if (!dead) {
pbx_builtin_setvar_helper(chan, "MACRO_EXTEN", save_macro_exten);
pbx_builtin_setvar_helper(chan, "MACRO_CONTEXT", save_macro_context);
pbx_builtin_setvar_helper(chan, "MACRO_PRIORITY", save_macro_priority);
}
pbx_builtin_setvar_helper(chan, "MACRO_EXTEN", save_macro_exten);
pbx_builtin_setvar_helper(chan, "MACRO_CONTEXT", save_macro_context);
pbx_builtin_setvar_helper(chan, "MACRO_PRIORITY", save_macro_priority);
if (save_macro_exten)
ast_free(save_macro_exten);
if (save_macro_context)
@@ -417,13 +451,13 @@ static int _macro_exec(struct ast_channel *chan, void *data, int exclusive)
if (save_macro_priority)
ast_free(save_macro_priority);
if (!dead && setmacrocontext) {
if (setmacrocontext) {
chan->macrocontext[0] = '\0';
chan->macroexten[0] = '\0';
chan->macropriority = 0;
}
if (!dead && !strcasecmp(chan->context, fullmacro)) {
if (!strcasecmp(chan->context, fullmacro)) {
/* If we're leaving the macro normally, restore original information */
chan->priority = oldpriority;
ast_copy_string(chan->context, oldcontext, sizeof(chan->context));
@@ -443,8 +477,7 @@ static int _macro_exec(struct ast_channel *chan, void *data, int exclusive)
}
}
if (!dead)
pbx_builtin_setvar_helper(chan, "MACRO_OFFSET", save_macro_offset);
pbx_builtin_setvar_helper(chan, "MACRO_OFFSET", save_macro_offset);
if (save_macro_offset)
ast_free(save_macro_offset);
@@ -456,6 +489,7 @@ static int _macro_exec(struct ast_channel *chan, void *data, int exclusive)
res = 0;
}
}
ast_channel_unlock(chan);
return res;
}

File diff suppressed because it is too large Load Diff

View File

@@ -25,19 +25,28 @@
* \ingroup applications
*/
/*** MODULEINFO
<depend>res_indications</depend>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/module.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
static char *app = "Milliwatt";
static char *synopsis = "Generate a Constant 1000Hz tone at 0dbm (mu-law)";
static char *synopsis = "Generate a Constant 1004Hz tone at 0dbm (mu-law)";
static char *descrip =
"Milliwatt(): Generate a Constant 1000Hz tone at 0dbm (mu-law)\n";
" Milliwatt([options]): Generate a Constant 1004Hz tone at 0dbm.\n"
"Previous versions of this application generated the tone at 1000Hz. If for\n"
"some reason you would prefer that behavior, supply the 'o' option to get the\n"
"old behavior.\n"
"";
static char digital_milliwatt[] = {0x1e,0x0b,0x0b,0x1e,0x9e,0x8b,0x8b,0x9e} ;
@@ -55,7 +64,7 @@ static void milliwatt_release(struct ast_channel *chan, void *data)
static int milliwatt_generate(struct ast_channel *chan, void *data, int len, int samples)
{
unsigned char buf[AST_FRIENDLY_OFFSET + 640];
const int maxsamples = sizeof (buf) / sizeof (buf[0]);
const int maxsamples = ARRAY_LEN(buf);
int i, *indexp = (int *) data;
struct ast_frame wf = {
.frametype = AST_FRAME_VOICE,
@@ -74,6 +83,7 @@ static int milliwatt_generate(struct ast_channel *chan, void *data, int len, int
ast_log(LOG_WARNING, "Only doing %d samples (%d requested)\n", maxsamples, samples);
samples = maxsamples;
}
len = samples * sizeof (buf[0]);
wf.datalen = len;
wf.samples = samples;
@@ -92,35 +102,58 @@ static int milliwatt_generate(struct ast_channel *chan, void *data, int len, int
return 0;
}
static struct ast_generator milliwattgen =
{
static struct ast_generator milliwattgen = {
alloc: milliwatt_alloc,
release: milliwatt_release,
generate: milliwatt_generate,
};
static int milliwatt_exec(struct ast_channel *chan, void *data)
static int old_milliwatt_exec(struct ast_channel *chan)
{
ast_set_write_format(chan, AST_FORMAT_ULAW);
ast_set_read_format(chan, AST_FORMAT_ULAW);
if (chan->_state != AST_STATE_UP)
if (chan->_state != AST_STATE_UP) {
ast_answer(chan);
}
if (ast_activate_generator(chan,&milliwattgen,"milliwatt") < 0) {
ast_log(LOG_WARNING,"Failed to activate generator on '%s'\n",chan->name);
return -1;
}
while(!ast_safe_sleep(chan, 10000));
while (!ast_safe_sleep(chan, 10000))
;
ast_deactivate_generator(chan);
return -1;
}
static int milliwatt_exec(struct ast_channel *chan, void *data)
{
const char *options = data;
struct ast_app *playtones_app;
int res = -1;
if (!ast_strlen_zero(options) && strchr(options, 'o')) {
return old_milliwatt_exec(chan);
}
if (!(playtones_app = pbx_findapp("Playtones"))) {
ast_log(LOG_ERROR, "The Playtones application is required to run Milliwatt()\n");
return -1;
}
res = pbx_exec(chan, playtones_app, "1004/1000");
while (!res) {
res = ast_safe_sleep(chan, 10000);
}
return res;
}
static int unload_module(void)
{
return ast_unregister_application(app);

View File

@@ -960,7 +960,7 @@ static int sendmail(struct minivm_template *template, struct minivm_account *vmu
return -1;
}
/* Allocate channel used for chanvar substitution */
ast = ast_channel_alloc(0, AST_STATE_DOWN, 0, 0, "", "", "", 0, 0);
ast = ast_channel_alloc(0, AST_STATE_DOWN, 0, 0, "", "", "", 0, "%s", "");
snprintf(dur, sizeof(dur), "%d:%02d", duration / 60, duration % 60);
@@ -1031,7 +1031,7 @@ static int sendmail(struct minivm_template *template, struct minivm_account *vmu
}
ast_debug(4, "-_-_- Fromstring now: %s\n", ast_strlen_zero(passdata) ? "-default-" : passdata);
fprintf(p, "Message-ID: <Asterisk-%d-%s-%d-%s>\n", (unsigned int)rand(), vmu->username, (int)getpid(), who);
fprintf(p, "Message-ID: <Asterisk-%d-%s-%d-%s>\n", (unsigned int)ast_random(), vmu->username, (int)getpid(), who);
len_passdata = strlen(vmu->fullname) * 2 + 3;
passdata2 = alloca(len_passdata);
if (!ast_strlen_zero(vmu->email))
@@ -1064,7 +1064,7 @@ static int sendmail(struct minivm_template *template, struct minivm_account *vmu
fprintf(p, "MIME-Version: 1.0\n");
/* Something unique. */
snprintf(bound, sizeof(bound), "voicemail_%s%d%d", vmu->username, (int)getpid(), (unsigned int)rand());
snprintf(bound, sizeof(bound), "voicemail_%s%d%d", vmu->username, (int)getpid(), (unsigned int)ast_random());
fprintf(p, "Content-Type: multipart/mixed; boundary=\"%s\"\n\n\n", bound);
@@ -1560,10 +1560,10 @@ static int leave_voicemail(struct ast_channel *chan, char *username, struct leav
duration < global_vmminmessage ? "IGNORED" : "OK",
vmu->accountcode
);
fprintf(txt, logbuf);
fprintf(txt, "%s", logbuf);
if (minivmlogfile) {
ast_mutex_lock(&minivmloglock);
fprintf(minivmlogfile, logbuf);
fprintf(minivmlogfile, "%s", logbuf);
ast_mutex_unlock(&minivmloglock);
}
@@ -1636,7 +1636,7 @@ static int minivm_notify_exec(struct ast_channel *chan, void *data)
ast_log(LOG_ERROR, "Out of memory\n");
return -1;
}
argc = ast_app_separate_args(tmpptr, ',', argv, sizeof(argv) / sizeof(argv[0]));
argc = ast_app_separate_args(tmpptr, ',', argv, ARRAY_LEN(argv));
if (argc == 2 && !ast_strlen_zero(argv[1]))
template = argv[1];
@@ -1706,7 +1706,7 @@ static int minivm_record_exec(struct ast_channel *chan, void *data)
ast_log(LOG_ERROR, "Out of memory\n");
return -1;
}
argc = ast_app_separate_args(tmp, ',', argv, sizeof(argv) / sizeof(argv[0]));
argc = ast_app_separate_args(tmp, ',', argv, ARRAY_LEN(argv));
if (argc == 2) {
if (ast_app_parse_options(minivm_app_options, &flags, opts, argv[1])) {
return -1;
@@ -1768,7 +1768,7 @@ static int minivm_greet_exec(struct ast_channel *chan, void *data)
ast_log(LOG_ERROR, "Out of memory\n");
return -1;
}
argc = ast_app_separate_args(tmpptr, ',', argv, sizeof(argv) / sizeof(argv[0]));
argc = ast_app_separate_args(tmpptr, ',', argv, ARRAY_LEN(argv));
if (argc == 2) {
if (ast_app_parse_options(minivm_app_options, &flags, opts, argv[1]))
@@ -1962,7 +1962,6 @@ static int minivm_accmess_exec(struct ast_channel *chan, void *data)
{
int argc = 0;
char *argv[2];
int res = 0;
char filename[PATH_MAX];
char tmp[PATH_MAX];
char *domain;
@@ -1987,7 +1986,7 @@ static int minivm_accmess_exec(struct ast_channel *chan, void *data)
ast_log(LOG_ERROR, "Out of memory\n");
error = TRUE;
} else
argc = ast_app_separate_args(tmpptr, ',', argv, sizeof(argv) / sizeof(argv[0]));
argc = ast_app_separate_args(tmpptr, ',', argv, ARRAY_LEN(argv));
}
if (argc <=1) {
@@ -2004,8 +2003,10 @@ static int minivm_accmess_exec(struct ast_channel *chan, void *data)
error = TRUE;
}
if (error)
if (error) {
pbx_builtin_setvar_helper(chan, "MINIVM_ACCMESS_STATUS", "FAILED");
return -1;
}
ast_copy_string(tmp, argv[0], sizeof(tmp));
username = tmp;
@@ -2016,13 +2017,14 @@ static int minivm_accmess_exec(struct ast_channel *chan, void *data)
}
if (ast_strlen_zero(domain) || ast_strlen_zero(username)) {
ast_log(LOG_ERROR, "Need username@domain as argument. Sorry. Argument 0 %s\n", argv[0]);
pbx_builtin_setvar_helper(chan, "MINIVM_ACCMESS_STATUS", "FAILED");
return -1;
}
if(!(vmu = find_account(domain, username, TRUE))) {
/* We could not find user, let's exit */
ast_log(LOG_WARNING, "Could not allocate temporary memory for '%s@%s'\n", username, domain);
pbx_builtin_setvar_helper(chan, "MINIVM_NOTIFY_STATUS", "FAILED");
pbx_builtin_setvar_helper(chan, "MINIVM_ACCMESS_STATUS", "FAILED");
return -1;
}
@@ -2053,10 +2055,10 @@ static int minivm_accmess_exec(struct ast_channel *chan, void *data)
if(ast_test_flag(vmu, MVM_ALLOCED))
free_user(vmu);
pbx_builtin_setvar_helper(chan, "MINIVM_ACCMESS_STATUS", "SUCCESS");
/* Ok, we're ready to rock and roll. Return to dialplan */
return res;
return 0;
}
/*! \brief Append new mailbox to mailbox list from configuration file */
@@ -2474,7 +2476,7 @@ static int load_config(int reload)
static char *handle_minivm_list_templates(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct minivm_template *this;
char *output_format = "%-15s %-10s %-10s %-15.15s %-50s\n";
#define HVLT_OUTPUT_FORMAT "%-15s %-10s %-10s %-15.15s %-50s\n"
int count = 0;
switch (cmd) {
@@ -2497,10 +2499,10 @@ static char *handle_minivm_list_templates(struct ast_cli_entry *e, int cmd, stru
AST_LIST_UNLOCK(&message_templates);
return CLI_FAILURE;
}
ast_cli(a->fd, output_format, "Template name", "Charset", "Locale", "Attach media", "Subject");
ast_cli(a->fd, output_format, "-------------", "-------", "------", "------------", "-------");
ast_cli(a->fd, HVLT_OUTPUT_FORMAT, "Template name", "Charset", "Locale", "Attach media", "Subject");
ast_cli(a->fd, HVLT_OUTPUT_FORMAT, "-------------", "-------", "------", "------------", "-------");
AST_LIST_TRAVERSE(&message_templates, this, list) {
ast_cli(a->fd, output_format, this->name,
ast_cli(a->fd, HVLT_OUTPUT_FORMAT, this->name,
this->charset ? this->charset : "-",
this->locale ? this->locale : "-",
this->attachment ? "Yes" : "No",
@@ -2540,7 +2542,7 @@ static char *complete_minivm_show_users(const char *line, const char *word, int
static char *handle_minivm_show_users(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct minivm_account *vmu;
char *output_format = "%-23s %-15s %-15s %-10s %-10s %-50s\n";
#define HMSU_OUTPUT_FORMAT "%-23s %-15s %-15s %-10s %-10s %-50s\n"
int count = 0;
switch (cmd) {
@@ -2565,14 +2567,14 @@ static char *handle_minivm_show_users(struct ast_cli_entry *e, int cmd, struct a
AST_LIST_UNLOCK(&minivm_accounts);
return CLI_FAILURE;
}
ast_cli(a->fd, output_format, "User", "E-Template", "P-template", "Zone", "Format", "Full name");
ast_cli(a->fd, output_format, "----", "----------", "----------", "----", "------", "---------");
ast_cli(a->fd, HMSU_OUTPUT_FORMAT, "User", "E-Template", "P-template", "Zone", "Format", "Full name");
ast_cli(a->fd, HMSU_OUTPUT_FORMAT, "----", "----------", "----------", "----", "------", "---------");
AST_LIST_TRAVERSE(&minivm_accounts, vmu, list) {
char tmp[256] = "";
if ((a->argc == 3) || ((a->argc == 5) && !strcmp(a->argv[4], vmu->domain))) {
count++;
snprintf(tmp, sizeof(tmp), "%s@%s", vmu->username, vmu->domain);
ast_cli(a->fd, output_format, tmp, vmu->etemplate ? vmu->etemplate : "-",
ast_cli(a->fd, HMSU_OUTPUT_FORMAT, tmp, vmu->etemplate ? vmu->etemplate : "-",
vmu->ptemplate ? vmu->ptemplate : "-",
vmu->zonetag ? vmu->zonetag : "-",
vmu->attachfmt ? vmu->attachfmt : "-",
@@ -2588,7 +2590,7 @@ static char *handle_minivm_show_users(struct ast_cli_entry *e, int cmd, struct a
static char *handle_minivm_show_zones(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct minivm_zone *zone;
char *output_format = "%-15s %-20s %-45s\n";
#define HMSZ_OUTPUT_FORMAT "%-15s %-20s %-45s\n"
char *res = CLI_SUCCESS;
switch (cmd) {
@@ -2607,10 +2609,10 @@ static char *handle_minivm_show_zones(struct ast_cli_entry *e, int cmd, struct a
AST_LIST_LOCK(&minivm_zones);
if (!AST_LIST_EMPTY(&minivm_zones)) {
ast_cli(a->fd, output_format, "Zone", "Timezone", "Message Format");
ast_cli(a->fd, output_format, "----", "--------", "--------------");
ast_cli(a->fd, HMSZ_OUTPUT_FORMAT, "Zone", "Timezone", "Message Format");
ast_cli(a->fd, HMSZ_OUTPUT_FORMAT, "----", "--------", "--------------");
AST_LIST_TRAVERSE(&minivm_zones, zone, list) {
ast_cli(a->fd, output_format, zone->name, zone->timezone, zone->msg_format);
ast_cli(a->fd, HMSZ_OUTPUT_FORMAT, zone->name, zone->timezone, zone->msg_format);
}
} else {
ast_cli(a->fd, "There are no voicemail zones currently defined\n");

View File

@@ -55,12 +55,17 @@ static const char *desc = ""
"Records the audio on the current channel to the specified file.\n"
"If the filename is an absolute path, uses that path, otherwise\n"
"creates the file in the configured monitoring directory from\n"
"asterisk.conf.\n\n"
"asterisk.conf. Use of StopMixMonitor is required to guarantee\n"
"the audio file is available for processing during dialplan execution.\n\n"
"Valid options:\n"
" a - Append to the file instead of overwriting it.\n"
" b - Only save audio to the file while the channel is bridged.\n"
" Note: Does not include conferences or sounds played to each bridged\n"
" party.\n"
" Note: If you utilize this option inside a Local channel, you must\n"
" make sure the Local channel is not optimized away. To do this,\n"
" be sure to call your Local channel with the '/n' option.\n"
" For example: Dial(Local/start@mycontext/n)\n"
" v(<x>) - Adjust the heard volume by a factor of <x> (range -4 to 4)\n"
" V(<x>) - Adjust the spoken volume by a factor of <x> (range -4 to 4)\n"
" W(<x>) - Adjust the both heard and spoken volumes by a factor of <x>\n"
@@ -75,8 +80,7 @@ static const char *stop_app = "StopMixMonitor";
static const char *stop_synopsis = "Stop recording a call through MixMonitor";
static const char *stop_desc = ""
" StopMixMonitor():\n"
"Stops the audio recording that was started with a call to MixMonitor()\n"
"on the current channel.\n"
"Stop recording a call through MixMonitor, and free the recording's file handle.\n"
"";
struct module_symbols *me;
@@ -89,7 +93,7 @@ struct mixmonitor {
char *post_process;
char *name;
unsigned int flags;
struct ast_channel *chan;
struct mixmonitor_ds *mixmonitor_ds;
};
enum {
@@ -115,6 +119,67 @@ AST_APP_OPTIONS(mixmonitor_opts, {
AST_APP_OPTION_ARG('W', MUXFLAG_VOLUME, OPT_ARG_VOLUME),
});
/* This structure is used as a means of making sure that our pointer to
* the channel we are monitoring remains valid. This is very similar to
* what is used in app_chanspy.c.
*/
struct mixmonitor_ds {
struct ast_channel *chan;
/* These condition variables are used to be sure that the channel
* hangup code completes before the mixmonitor thread attempts to
* free this structure. The combination of a bookean flag and a
* ast_cond_t ensure that no matter what order the threads run in,
* we are guaranteed to never have the waiting thread block forever
* in the case that the signaling thread runs first.
*/
unsigned int destruction_ok;
ast_cond_t destruction_condition;
ast_mutex_t lock;
/* The filestream is held in the datastore so it can be stopped
* immediately during stop_mixmonitor or channel destruction. */
int fs_quit;
struct ast_filestream *fs;
};
static void mixmonitor_ds_close_fs(struct mixmonitor_ds *mixmonitor_ds)
{
ast_mutex_lock(&mixmonitor_ds->lock);
if (mixmonitor_ds->fs) {
ast_closestream(mixmonitor_ds->fs);
mixmonitor_ds->fs = NULL;
mixmonitor_ds->fs_quit = 1;
ast_verb(2, "MixMonitor close filestream\n");
}
ast_mutex_unlock(&mixmonitor_ds->lock);
}
static void mixmonitor_ds_destroy(void *data)
{
struct mixmonitor_ds *mixmonitor_ds = data;
ast_mutex_lock(&mixmonitor_ds->lock);
mixmonitor_ds->chan = NULL;
mixmonitor_ds->destruction_ok = 1;
ast_cond_signal(&mixmonitor_ds->destruction_condition);
ast_mutex_unlock(&mixmonitor_ds->lock);
}
static void mixmonitor_ds_chan_fixup(void *data, struct ast_channel *old_chan, struct ast_channel *new_chan)
{
struct mixmonitor_ds *mixmonitor_ds = data;
ast_mutex_lock(&mixmonitor_ds->lock);
mixmonitor_ds->chan = new_chan;
ast_mutex_unlock(&mixmonitor_ds->lock);
}
static struct ast_datastore_info mixmonitor_ds_info = {
.type = "mixmonitor",
.destroy = mixmonitor_ds_destroy,
.chan_fixup = mixmonitor_ds_chan_fixup,
};
static int startmon(struct ast_channel *chan, struct ast_audiohook *audiohook)
{
struct ast_channel *peer = NULL;
@@ -133,10 +198,22 @@ static int startmon(struct ast_channel *chan, struct ast_audiohook *audiohook)
#define SAMPLES_PER_FRAME 160
static void mixmonitor_free(struct mixmonitor *mixmonitor)
{
if (mixmonitor) {
if (mixmonitor->mixmonitor_ds) {
ast_mutex_destroy(&mixmonitor->mixmonitor_ds->lock);
ast_cond_destroy(&mixmonitor->mixmonitor_ds->destruction_condition);
ast_free(mixmonitor->mixmonitor_ds);
}
ast_free(mixmonitor);
}
}
static void *mixmonitor_thread(void *obj)
{
struct mixmonitor *mixmonitor = obj;
struct ast_filestream *fs = NULL;
struct ast_filestream **fs = NULL;
unsigned int oflags;
char *ext;
int errflag = 0;
@@ -145,7 +222,9 @@ static void *mixmonitor_thread(void *obj)
ast_audiohook_lock(&mixmonitor->audiohook);
while (mixmonitor->audiohook.status == AST_AUDIOHOOK_STATUS_RUNNING) {
fs = &mixmonitor->mixmonitor_ds->fs;
while (mixmonitor->audiohook.status == AST_AUDIOHOOK_STATUS_RUNNING && !mixmonitor->mixmonitor_ds->fs_quit) {
struct ast_frame *fr = NULL;
ast_audiohook_trigger_wait(&mixmonitor->audiohook);
@@ -156,27 +235,34 @@ static void *mixmonitor_thread(void *obj)
if (!(fr = ast_audiohook_read_frame(&mixmonitor->audiohook, SAMPLES_PER_FRAME, AST_AUDIOHOOK_DIRECTION_BOTH, AST_FORMAT_SLINEAR)))
continue;
if (!ast_test_flag(mixmonitor, MUXFLAG_BRIDGED) || ast_bridged_channel(mixmonitor->chan)) {
ast_mutex_lock(&mixmonitor->mixmonitor_ds->lock);
if (!ast_test_flag(mixmonitor, MUXFLAG_BRIDGED) || (mixmonitor->mixmonitor_ds->chan && ast_bridged_channel(mixmonitor->mixmonitor_ds->chan))) {
/* Initialize the file if not already done so */
if (!fs && !errflag) {
if (!*fs && !errflag && !mixmonitor->mixmonitor_ds->fs_quit) {
oflags = O_CREAT | O_WRONLY;
oflags |= ast_test_flag(mixmonitor, MUXFLAG_APPEND) ? O_APPEND : O_TRUNC;
if ((ext = strrchr(mixmonitor->filename, '.')))
*(ext++) = '\0';
else
ext = "raw";
if (!(fs = ast_writefile(mixmonitor->filename, ext, NULL, oflags, 0, 0644))) {
if (!(*fs = ast_writefile(mixmonitor->filename, ext, NULL, oflags, 0, 0644))) {
ast_log(LOG_ERROR, "Cannot open %s.%s\n", mixmonitor->filename, ext);
errflag = 1;
}
}
/* Write out frame */
if (fs)
ast_writestream(fs, fr);
/* Write out the frame(s) */
if (*fs) {
struct ast_frame *cur;
for (cur = fr; cur; cur = AST_LIST_NEXT(cur, frame_list)) {
ast_writestream(*fs, cur);
}
}
}
ast_mutex_unlock(&mixmonitor->mixmonitor_ds->lock);
/* All done! free it. */
ast_frame_free(fr, 0);
@@ -187,22 +273,55 @@ static void *mixmonitor_thread(void *obj)
ast_audiohook_unlock(&mixmonitor->audiohook);
ast_audiohook_destroy(&mixmonitor->audiohook);
ast_verb(2, "End MixMonitor Recording %s\n", mixmonitor->name);
if (fs)
ast_closestream(fs);
mixmonitor_ds_close_fs(mixmonitor->mixmonitor_ds);
if (mixmonitor->post_process) {
ast_verb(2, "Executing [%s]\n", mixmonitor->post_process);
ast_safe_system(mixmonitor->post_process);
}
ast_free(mixmonitor);
ast_mutex_lock(&mixmonitor->mixmonitor_ds->lock);
if (!mixmonitor->mixmonitor_ds->destruction_ok) {
ast_cond_wait(&mixmonitor->mixmonitor_ds->destruction_condition, &mixmonitor->mixmonitor_ds->lock);
}
ast_mutex_unlock(&mixmonitor->mixmonitor_ds->lock);
ast_verb(2, "End MixMonitor Recording %s\n", mixmonitor->name);
mixmonitor_free(mixmonitor);
return NULL;
}
static int setup_mixmonitor_ds(struct mixmonitor *mixmonitor, struct ast_channel *chan)
{
struct ast_datastore *datastore = NULL;
struct mixmonitor_ds *mixmonitor_ds;
if (!(mixmonitor_ds = ast_calloc(1, sizeof(*mixmonitor_ds)))) {
return -1;
}
ast_mutex_init(&mixmonitor_ds->lock);
ast_cond_init(&mixmonitor_ds->destruction_condition, NULL);
if (!(datastore = ast_channel_datastore_alloc(&mixmonitor_ds_info, NULL))) {
ast_mutex_destroy(&mixmonitor_ds->lock);
ast_cond_destroy(&mixmonitor_ds->destruction_condition);
ast_free(mixmonitor_ds);
return -1;
}
/* No need to lock mixmonitor_ds since this is still operating in the channel's thread */
mixmonitor_ds->chan = chan;
datastore->data = mixmonitor_ds;
ast_channel_lock(chan);
ast_channel_datastore_add(chan, datastore);
ast_channel_unlock(chan);
mixmonitor->mixmonitor_ds = mixmonitor_ds;
return 0;
}
static void launch_monitor_thread(struct ast_channel *chan, const char *filename, unsigned int flags,
int readvol, int writevol, const char *post_process)
{
@@ -236,7 +355,10 @@ static void launch_monitor_thread(struct ast_channel *chan, const char *filename
/* Copy over flags and channel name */
mixmonitor->flags = flags;
mixmonitor->chan = chan;
if (setup_mixmonitor_ds(mixmonitor, chan)) {
mixmonitor_free(mixmonitor);
return;
}
mixmonitor->name = (char *) mixmonitor + sizeof(*mixmonitor);
strcpy(mixmonitor->name, chan->name);
if (!ast_strlen_zero(postprocess2)) {
@@ -253,7 +375,7 @@ static void launch_monitor_thread(struct ast_channel *chan, const char *filename
return;
}
ast_set_flag(&mixmonitor->audiohook, AST_AUDIOHOOK_TRIGGER_WRITE);
ast_set_flag(&mixmonitor->audiohook, AST_AUDIOHOOK_TRIGGER_SYNC);
if (readvol)
mixmonitor->audiohook.options.read_volume = readvol;
@@ -269,7 +391,6 @@ static void launch_monitor_thread(struct ast_channel *chan, const char *filename
}
ast_pthread_create_detached_background(&thread, NULL, mixmonitor_thread, mixmonitor);
}
static int mixmonitor_exec(struct ast_channel *chan, void *data)
@@ -355,6 +476,14 @@ static int mixmonitor_exec(struct ast_channel *chan, void *data)
static int stop_mixmonitor_exec(struct ast_channel *chan, void *data)
{
struct ast_datastore *datastore = NULL;
/* closing the filestream here guarantees the file is avaliable to the dialplan
* after calling StopMixMonitor */
if ((datastore = ast_channel_datastore_find(chan, &mixmonitor_ds_info, NULL))) {
mixmonitor_ds_close_fs(datastore->data);
}
ast_audiohook_detach_source(chan, mixmonitor_spy_type);
return 0;
}

View File

@@ -25,12 +25,19 @@
* \ingroup applications
*/
/*** MODULEINFO
<depend>working_fork</depend>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include <sys/time.h>
#include <signal.h>
#ifdef HAVE_CAP
#include <sys/capability.h>
#endif /* HAVE_CAP */
#include "asterisk/lock.h"
#include "asterisk/file.h"
@@ -58,6 +65,9 @@ static int mp3play(char *filename, int fd)
int res;
int x;
sigset_t fullset, oldset;
#ifdef HAVE_CAP
cap_t cap;
#endif
sigfillset(&fullset);
pthread_sigmask(SIG_BLOCK, &fullset, &oldset);
@@ -69,6 +79,15 @@ static int mp3play(char *filename, int fd)
pthread_sigmask(SIG_SETMASK, &oldset, NULL);
return res;
}
#ifdef HAVE_CAP
cap = cap_from_text("cap_net_admin-eip");
if (cap_set_proc(cap)) {
/* Careful with order! Logging cannot happen after we close FDs */
ast_log(LOG_WARNING, "Unable to remove capabilities.\n");
}
cap_free(cap);
#endif
if (ast_opt_high_priority)
ast_set_priority(0);
signal(SIGPIPE, SIG_DFL);
@@ -76,8 +95,7 @@ static int mp3play(char *filename, int fd)
dup2(fd, STDOUT_FILENO);
for (x=STDERR_FILENO + 1;x<256;x++) {
if (x != STDOUT_FILENO)
close(x);
close(x);
}
/* Execute mpg123, but buffer if it's a net connection */
if (!strncasecmp(filename, "http://", 7)) {
@@ -106,7 +124,7 @@ static int timed_read(int fd, void *data, int datalen, int timeout)
struct pollfd fds[1];
fds[0].fd = fd;
fds[0].events = POLLIN;
res = poll(fds, 1, timeout);
res = ast_poll(fds, 1, timeout);
if (res < 1) {
ast_log(LOG_NOTICE, "Poll timed out/errored out with %d\n", res);
return -1;

View File

@@ -25,6 +25,10 @@
* \ingroup applications
*/
/*** MODULEINFO
<depend>working_fork</depend>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
@@ -33,6 +37,9 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include <sys/time.h>
#include <sys/socket.h>
#include <signal.h>
#ifdef HAVE_CAP
#include <sys/capability.h>
#endif /* HAVE_CAP */
#include "asterisk/lock.h"
#include "asterisk/file.h"
@@ -63,6 +70,9 @@ static int NBScatplay(int fd)
int res;
int x;
sigset_t fullset, oldset;
#ifdef HAVE_CAP
cap_t cap;
#endif
sigfillset(&fullset);
pthread_sigmask(SIG_BLOCK, &fullset, &oldset);
@@ -77,6 +87,15 @@ static int NBScatplay(int fd)
signal(SIGPIPE, SIG_DFL);
pthread_sigmask(SIG_UNBLOCK, &fullset, NULL);
#ifdef HAVE_CAP
cap = cap_from_text("cap_net_admin-eip");
if (cap_set_proc(cap)) {
/* Careful with order! Logging cannot happen after we close FDs */
ast_log(LOG_WARNING, "Unable to remove capabilities.\n");
}
cap_free(cap);
#endif
if (ast_opt_high_priority)
ast_set_priority(0);
@@ -98,7 +117,7 @@ static int timed_read(int fd, void *data, int datalen)
struct pollfd fds[1];
fds[0].fd = fd;
fds[0].events = POLLIN;
res = poll(fds, 1, 2000);
res = ast_poll(fds, 1, 2000);
if (res < 1) {
ast_log(LOG_NOTICE, "Selected timed out/errored out with %d\n", res);
return -1;

View File

@@ -26,7 +26,7 @@
*/
/*** MODULEINFO
<depend>zaptel</depend>
<depend>dahdi</depend>
<depend>app_meetme</depend>
***/
@@ -73,17 +73,17 @@ AST_APP_OPTIONS(page_opts, {
AST_APP_OPTION('s', PAGE_SKIP),
});
#define MAX_DIALS 128
static int page_exec(struct ast_channel *chan, void *data)
{
char *options, *tech, *resource, *tmp;
char *options, *tech, *resource, *tmp, *tmp2;
char meetmeopts[88], originator[AST_CHANNEL_NAME], *opts[0];
struct ast_flags flags = { 0 };
unsigned int confid = ast_random();
struct ast_app *app;
int res = 0, pos = 0, i = 0;
struct ast_dial *dials[MAX_DIALS];
struct ast_dial **dial_list;
unsigned int num_dials;
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "This application requires at least one argument (destination(s) to page)\n");
@@ -108,6 +108,21 @@ static int page_exec(struct ast_channel *chan, void *data)
snprintf(meetmeopts, sizeof(meetmeopts), "MeetMe,%ud,%s%sqxdw(5)", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m"),
(ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
/* Count number of extensions in list by number of ampersands + 1 */
num_dials = 1;
tmp2 = tmp;
while (*tmp2) {
if (*tmp2 == '&') {
num_dials++;
}
tmp2++;
}
if (!(dial_list = ast_calloc(num_dials, sizeof(struct ast_dial *)))) {
ast_log(LOG_ERROR, "Can't allocate %ld bytes for dial list\n", (long)(sizeof(struct ast_dial *) * num_dials));
return -1;
}
/* Go through parsing/calling each device */
while ((tech = strsep(&tmp, "&"))) {
int state = 0;
@@ -124,9 +139,14 @@ static int page_exec(struct ast_channel *chan, void *data)
}
/* Ensure device is not in use if skip option is enabled */
if (ast_test_flag(&flags, PAGE_SKIP) && (state = ast_device_state(tech)) != AST_DEVICE_NOT_INUSE) {
ast_log(LOG_WARNING, "Destination '%s' has device state '%s'.\n", tech, devstate2str(state));
continue;
if (ast_test_flag(&flags, PAGE_SKIP)) {
state = ast_device_state(tech);
if (state == AST_DEVICE_UNKNOWN) {
ast_log(LOG_WARNING, "Destination '%s' has device state '%s'. Paging anyway.\n", tech, devstate2str(state));
} else if (state != AST_DEVICE_NOT_INUSE) {
ast_log(LOG_WARNING, "Destination '%s' has device state '%s'.\n", tech, devstate2str(state));
continue;
}
}
*resource++ = '\0';
@@ -147,7 +167,7 @@ static int page_exec(struct ast_channel *chan, void *data)
ast_dial_run(dial, chan, 1);
/* Put in our dialing array */
dials[pos++] = dial;
dial_list[pos++] = dial;
}
if (!ast_test_flag(&flags, PAGE_QUIET)) {
@@ -164,7 +184,7 @@ static int page_exec(struct ast_channel *chan, void *data)
/* Go through each dial attempt cancelling, joining, and destroying */
for (i = 0; i < pos; i++) {
struct ast_dial *dial = dials[i];
struct ast_dial *dial = dial_list[i];
/* We have to wait for the async thread to exit as it's possible Meetme won't throw them out immediately */
ast_dial_join(dial);

View File

@@ -95,15 +95,17 @@ static int parkandannounce_exec(struct ast_channel *chan, void *data)
timeout = atoi(args.timeout) * 1000;
if (ast_strlen_zero(args.dial)) {
ast_log(LOG_WARNING, "PARK: A dial resource must be specified i.e: Console/dsp or Zap/g1/5551212\n");
ast_log(LOG_WARNING, "PARK: A dial resource must be specified i.e: Console/dsp or DAHDI/g1/5551212\n");
return -1;
}
dialtech = strsep(&args.dial, "/");
ast_verb(3, "Dial Tech,String: (%s,%s)\n", dialtech, args.dial);
if (!ast_strlen_zero(args.return_context))
if (!ast_strlen_zero(args.return_context)) {
ast_clear_flag(chan, AST_FLAG_IN_AUTOLOOP);
ast_parseable_goto(chan, args.return_context);
}
ast_verb(3, "Return Context: (%s,%s,%d) ID: %s\n", chan->context, chan->exten, chan->priority, chan->cid.cid_num);
if (!ast_exists_extension(chan, chan->context, chan->exten, chan->priority, chan->cid.cid_num)) {
@@ -113,7 +115,9 @@ static int parkandannounce_exec(struct ast_channel *chan, void *data)
/* we are using masq_park here to protect * from touching the channel once we park it. If the channel comes out of timeout
before we are done announcing and the channel is messed with, Kablooeee. So we use Masq to prevent this. */
ast_masq_park_call(chan, NULL, timeout, &lot);
res = ast_masq_park_call(chan, NULL, timeout, &lot);
if (res == -1)
return res;
ast_verb(3, "Call Parking Called, lot: %d, timeout: %d, context: %s\n", lot, timeout, args.return_context);
@@ -144,9 +148,12 @@ static int parkandannounce_exec(struct ast_channel *chan, void *data)
ast_verb(4, "Announce Template:%s\n", args.template);
for (looptemp = 0, tmp[looptemp++] = strsep(&args.template, ":");
looptemp < sizeof(tmp) / sizeof(tmp[0]);
tmp[looptemp++] = strsep(&args.template, ":"));
for (looptemp = 0; looptemp < ARRAY_LEN(tmp); looptemp++) {
if ((tmp[looptemp] = strsep(&args.template, ":")) != NULL)
continue;
else
break;
}
for (i = 0; i < looptemp; i++) {
ast_verb(4, "Announce:%s\n", tmp[i]);

View File

@@ -55,6 +55,8 @@ static char *descrip =
"This application sets the following channel variable upon completion:\n"
" PLAYBACKSTATUS The status of the playback attempt as a text string, one of\n"
" SUCCESS | FAILED\n"
"See Also: Background (application) -- for playing soundfiles that are interruptible\n"
" WaitExten (application) -- wait for digits from caller, optionally play music on hold\n"
;
@@ -198,13 +200,13 @@ static int do_say(say_args_t *a, const char *s, const char *options, int depth)
ast_debug(2, "doing [%s]\n", fn);
/* locate prefix and data, if any */
fmt = index(fn, ':');
fmt = strchr(fn, ':');
if (!fmt || fmt == fn) { /* regular filename */
ret = s_streamwait3(a, fn);
continue;
}
fmt++;
data = index(fmt, ':'); /* colon before data */
data = strchr(fmt, ':'); /* colon before data */
if (!data || data == fmt) { /* simple prefix-fmt */
ret = do_say(a, fn, options, depth);
continue;
@@ -217,14 +219,14 @@ static int do_say(say_args_t *a, const char *s, const char *options, int depth)
if (*p == '\'') {/* file name - we trim them */
char *y;
strcpy(fn2, ast_skip_blanks(p+1)); /* make a full copy */
y = index(fn2, '\'');
y = strchr(fn2, '\'');
if (!y) {
p = data; /* invalid. prepare to end */
break;
}
*y = '\0';
ast_trim_blanks(fn2);
p = index(p+1, '\'');
p = strchr(p+1, '\'');
ret = s_streamwait3(a, fn2);
} else {
int l = fmt-fn;

File diff suppressed because it is too large Load Diff

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@@ -86,7 +86,7 @@ static int read_exec(struct ast_channel *chan, void *data)
int tries = 1, to = 0, x = 0;
double tosec;
char *argcopy = NULL;
struct ind_tone_zone_sound *ts = NULL;
struct tone_zone_sound *ts = NULL;
struct ast_flags flags = {0};
const char *status = "ERROR";
@@ -185,11 +185,11 @@ static int read_exec(struct ast_channel *chan, void *data)
}
} else {
res = ast_app_getdata(chan, arglist.filename, tmp, maxdigits, to);
if (res == 0)
if (res == AST_GETDATA_COMPLETE || res == AST_GETDATA_EMPTY_END_TERMINATED)
status = "OK";
else if (res == 1)
else if (res == AST_GETDATA_TIMEOUT)
status = "TIMEOUT";
else if (res == 2)
else if (res == AST_GETDATA_INTERRUPTED)
status = "INTERRUPTED";
}
if (res > -1) {

View File

@@ -79,7 +79,7 @@ static int readexten_exec(struct ast_channel *chan, void *data)
int maxdigits = sizeof(exten) - 1;
int timeout = 0, digit_timeout = 0, x = 0;
char *argcopy = NULL, *status = "";
struct ind_tone_zone_sound *ts = NULL;
struct tone_zone_sound *ts = NULL;
struct ast_flags flags = {0};
AST_DECLARE_APP_ARGS(arglist,

View File

@@ -36,7 +36,6 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/channel.h"
#include "asterisk/dsp.h" /* use dsp routines for silence detection */
static char *app = "Record";
static char *synopsis = "Record to a file";
@@ -51,6 +50,7 @@ static char *descrip =
"or 0 there is no maximum.\n"
"- 'options' may contain any of the following letters:\n"
" 'a' : append to existing recording rather than replacing\n"
" 'k' : keep recorded file upon hangup\n"
" 'n' : do not answer, but record anyway if line not yet answered\n"
" 'q' : quiet (do not play a beep tone)\n"
" 's' : skip recording if the line is not yet answered\n"
@@ -72,11 +72,13 @@ enum {
OPTION_SKIP = (1 << 3),
OPTION_STAR_TERMINATE = (1 << 4),
OPTION_IGNORE_TERMINATE = (1 << 5),
FLAG_HAS_PERCENT = (1 << 6),
OPTION_KEEP = (1 << 6),
FLAG_HAS_PERCENT = (1 << 7),
};
AST_APP_OPTIONS(app_opts,{
AST_APP_OPTION('a', OPTION_APPEND),
AST_APP_OPTION('k', OPTION_KEEP),
AST_APP_OPTION('n', OPTION_NOANSWER),
AST_APP_OPTION('q', OPTION_QUIET),
AST_APP_OPTION('s', OPTION_SKIP),
@@ -325,6 +327,9 @@ static int record_exec(struct ast_channel *chan, void *data)
if (!f) {
ast_debug(1, "Got hangup\n");
res = -1;
if (!ast_test_flag(&flags, OPTION_KEEP)) {
ast_filedelete(args.filename, NULL);
}
}
if (gotsilence) {

View File

@@ -116,7 +116,7 @@
*/
/*** MODULEINFO
<depend>zaptel</depend>
<depend>dahdi</depend>
<depend>tonezone</depend>
<defaultenabled>no</defaultenabled>
***/
@@ -218,6 +218,8 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include <sys/ioctl.h>
#include <sys/io.h>
#include <math.h>
#include <dahdi/user.h>
#include <dahdi/tonezone.h>
#include <netinet/in.h>
#include <arpa/inet.h>
@@ -236,8 +238,6 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/localtime.h"
#include "asterisk/app.h"
#include "asterisk/zapata.h"
static char *app = "Rpt";
static char *synopsis = "Radio Repeater/Remote Base Control System";
@@ -888,12 +888,8 @@ static void load_rpt_vars(int n, int init)
ast_free(rpt_vars[n].p.tailmsgbuf);
memset(&rpt_vars[n].p, 0, sizeof(rpt_vars[n].p));
if (init) {
char *cp;
int savearea = (char *)&rpt_vars[n].p - (char *)&rpt_vars[n];
cp = (char *) &rpt_vars[n].p;
memset(cp + sizeof(rpt_vars[n].p), 0,
sizeof(rpt_vars[n]) - (sizeof(rpt_vars[n].p) + savearea));
/* clear all the fields in the structure after 'p' */
memset(&rpt_vars[n].p + sizeof(rpt_vars[0].p), 0, sizeof(rpt_vars[0]) - sizeof(rpt_vars[0].p) - offsetof(typeof(rpt_vars[0]), p));
rpt_vars[n].tele.next = &rpt_vars[n].tele;
rpt_vars[n].tele.prev = &rpt_vars[n].tele;
rpt_vars[n].rpt_thread = AST_PTHREADT_NULL;
@@ -1638,13 +1634,13 @@ static int send_morse(struct ast_channel *chan, const char *string, int speed, i
ast_stopstream(chan);
/*
* Wait for the zaptel driver to physically write the tone blocks to the hardware
* Wait for the dahdi driver to physically write the tone blocks to the hardware
*/
for (i = 0; i < 20 ; i++) {
flags = ZT_IOMUX_WRITEEMPTY | ZT_IOMUX_NOWAIT;
res = ioctl(chan->fds[0], ZT_IOMUX, &flags);
if (flags & ZT_IOMUX_WRITEEMPTY)
flags = DAHDI_IOMUX_WRITEEMPTY | DAHDI_IOMUX_NOWAIT;
res = ioctl(chan->fds[0], DAHDI_IOMUX, &flags);
if (flags & DAHDI_IOMUX_WRITEEMPTY)
break;
if ( ast_safe_sleep(chan, 50)) {
res = -1;
@@ -1689,13 +1685,13 @@ static int send_tone_telemetry(struct ast_channel *chan, const char *tonestring)
ast_stopstream(chan);
/*
* Wait for the zaptel driver to physically write the tone blocks to the hardware
* Wait for the dahdi driver to physically write the tone blocks to the hardware
*/
for (i = 0; i < 20 ; i++) {
flags = ZT_IOMUX_WRITEEMPTY | ZT_IOMUX_NOWAIT;
res = ioctl(chan->fds[0], ZT_IOMUX, &flags);
if (flags & ZT_IOMUX_WRITEEMPTY)
flags = DAHDI_IOMUX_WRITEEMPTY | DAHDI_IOMUX_NOWAIT;
res = ioctl(chan->fds[0], DAHDI_IOMUX, &flags);
if (flags & DAHDI_IOMUX_WRITEEMPTY)
break;
if (ast_safe_sleep(chan, 50)) {
res = -1;
@@ -1896,7 +1892,7 @@ static void wait_interval(struct rpt *myrpt, int type, struct ast_channel *chan)
static void *rpt_tele_thread(void *this)
{
ZT_CONFINFO ci; /* conference info */
struct dahdi_confinfo ci; /* conference info */
int res = 0, haslink, hastx, hasremote, imdone = 0, unkeys_queued, x;
struct rpt_tele *mytele = (struct rpt_tele *)this;
struct rpt_tele *tlist;
@@ -1940,9 +1936,9 @@ static void *rpt_tele_thread(void *this)
ci.confno = (((mytele->mode == ID) || (mytele->mode == IDTALKOVER) || (mytele->mode == UNKEY) ||
(mytele->mode == TAILMSG)) ?
myrpt->txconf : myrpt->conf);
ci.confmode = ZT_CONF_CONFANN;
ci.confmode = DAHDI_CONF_CONFANN;
/* first put the channel on the conference in announce mode */
if (ioctl(mychannel->fds[0], ZT_SETCONF, &ci) == -1) {
if (ioctl(mychannel->fds[0], DAHDI_SETCONF, &ci) == -1) {
ast_log(LOG_WARNING, "Unable to set conference mode to Announce\n");
rpt_mutex_lock(&myrpt->lock);
remque((struct qelem *)mytele);
@@ -2122,9 +2118,9 @@ static void *rpt_tele_thread(void *this)
/* set for all to hear */
ci.chan = 0;
ci.confno = myrpt->conf;
ci.confmode = ZT_CONF_CONFANN;
ci.confmode = DAHDI_CONF_CONFANN;
/* first put the channel on the conference in announce mode */
if (ioctl(mychannel->fds[0], ZT_SETCONF, &ci) == -1) {
if (ioctl(mychannel->fds[0], DAHDI_SETCONF, &ci) == -1) {
ast_log(LOG_WARNING, "Unable to set conference mode to Announce\n");
rpt_mutex_lock(&myrpt->lock);
remque((struct qelem *)mytele);
@@ -2149,9 +2145,9 @@ static void *rpt_tele_thread(void *this)
/* set for all to hear */
ci.chan = 0;
ci.confno = myrpt->txconf;
ci.confmode = ZT_CONF_CONFANN;
ci.confmode = DAHDI_CONF_CONFANN;
/* first put the channel on the conference in announce mode */
if (ioctl(mychannel->fds[0], ZT_SETCONF, &ci) == -1) {
if (ioctl(mychannel->fds[0], DAHDI_SETCONF, &ci) == -1) {
ast_log(LOG_WARNING, "Unable to set conference mode to Announce\n");
rpt_mutex_lock(&myrpt->lock);
remque((struct qelem *)mytele);
@@ -2529,7 +2525,7 @@ static void rpt_telemetry(struct rpt *myrpt, int mode, void *data)
static void *rpt_call(void *this)
{
ZT_CONFINFO ci; /* conference info */
struct dahdi_confinfo ci; /* conference info */
struct rpt *myrpt = (struct rpt *)this;
int res;
struct ast_frame wf;
@@ -2545,10 +2541,10 @@ static void *rpt_call(void *this)
}
ci.chan = 0;
ci.confno = myrpt->conf; /* use the pseudo conference */
ci.confmode = ZT_CONF_REALANDPSEUDO | ZT_CONF_TALKER | ZT_CONF_LISTENER
| ZT_CONF_PSEUDO_TALKER | ZT_CONF_PSEUDO_LISTENER;
ci.confmode = DAHDI_CONF_REALANDPSEUDO | DAHDI_CONF_TALKER | DAHDI_CONF_LISTENER
| DAHDI_CONF_PSEUDO_TALKER | DAHDI_CONF_PSEUDO_LISTENER;
/* first put the channel on the conference */
if (ioctl(mychannel->fds[0], ZT_SETCONF, &ci) == -1) {
if (ioctl(mychannel->fds[0], DAHDI_SETCONF, &ci) == -1) {
ast_log(LOG_WARNING, "Unable to set conference mode to Announce\n");
ast_hangup(mychannel);
myrpt->callmode = 0;
@@ -2563,10 +2559,10 @@ static void *rpt_call(void *this)
}
ci.chan = 0;
ci.confno = myrpt->conf;
ci.confmode = ZT_CONF_REALANDPSEUDO | ZT_CONF_TALKER | ZT_CONF_LISTENER
| ZT_CONF_PSEUDO_TALKER | ZT_CONF_PSEUDO_LISTENER;
ci.confmode = DAHDI_CONF_REALANDPSEUDO | DAHDI_CONF_TALKER | DAHDI_CONF_LISTENER
| DAHDI_CONF_PSEUDO_TALKER | DAHDI_CONF_PSEUDO_LISTENER;
/* first put the channel on the conference */
if (ioctl(genchannel->fds[0], ZT_SETCONF, &ci) == -1) {
if (ioctl(genchannel->fds[0], DAHDI_SETCONF, &ci) == -1) {
ast_log(LOG_WARNING, "Unable to set conference mode to Announce\n");
ast_hangup(mychannel);
ast_hangup(genchannel);
@@ -2588,7 +2584,7 @@ static void *rpt_call(void *this)
pthread_exit(NULL);
}
/* start dialtone if patchquiet is 0. Special patch modes don't send dial tone */
if ((!myrpt->patchquiet) && (tone_zone_play_tone(mychannel->fds[0], ZT_TONE_DIALTONE) < 0)) {
if ((!myrpt->patchquiet) && (tone_zone_play_tone(mychannel->fds[0], DAHDI_TONE_DIALTONE) < 0)) {
ast_log(LOG_WARNING, "Cannot start dialtone\n");
ast_hangup(mychannel);
ast_hangup(genchannel);
@@ -2624,7 +2620,7 @@ static void *rpt_call(void *this)
if (!congstarted) {
congstarted = 1;
/* start congestion tone */
tone_zone_play_tone(mychannel->fds[0], ZT_TONE_CONGESTION);
tone_zone_play_tone(mychannel->fds[0], DAHDI_TONE_CONGESTION);
}
}
res = ast_safe_sleep(mychannel, MSWAIT);
@@ -2693,10 +2689,10 @@ static void *rpt_call(void *this)
/* set appropriate conference for the pseudo */
ci.chan = 0;
ci.confno = myrpt->conf;
ci.confmode = (myrpt->p.duplex == 2) ? ZT_CONF_CONFANNMON :
(ZT_CONF_CONF | ZT_CONF_LISTENER | ZT_CONF_TALKER);
ci.confmode = (myrpt->p.duplex == 2) ? DAHDI_CONF_CONFANNMON :
(DAHDI_CONF_CONF | DAHDI_CONF_LISTENER | DAHDI_CONF_TALKER);
/* first put the channel on the conference in announce mode */
if (ioctl(myrpt->pchannel->fds[0], ZT_SETCONF, &ci) == -1) {
if (ioctl(myrpt->pchannel->fds[0], DAHDI_SETCONF, &ci) == -1) {
ast_log(LOG_WARNING, "Unable to set conference mode to Announce\n");
ast_hangup(mychannel);
ast_hangup(genchannel);
@@ -2716,7 +2712,7 @@ static void *rpt_call(void *this)
myrpt->callmode = 4;
rpt_mutex_unlock(&myrpt->lock);
/* start congestion tone */
tone_zone_play_tone(genchannel->fds[0], ZT_TONE_CONGESTION);
tone_zone_play_tone(genchannel->fds[0], DAHDI_TONE_CONGESTION);
rpt_mutex_lock(&myrpt->lock);
}
}
@@ -2748,10 +2744,10 @@ static void *rpt_call(void *this)
/* set appropriate conference for the pseudo */
ci.chan = 0;
ci.confno = myrpt->conf;
ci.confmode = ((myrpt->p.duplex == 2) || (myrpt->p.duplex == 4)) ? ZT_CONF_CONFANNMON :
(ZT_CONF_CONF | ZT_CONF_LISTENER | ZT_CONF_TALKER);
ci.confmode = ((myrpt->p.duplex == 2) || (myrpt->p.duplex == 4)) ? DAHDI_CONF_CONFANNMON :
(DAHDI_CONF_CONF | DAHDI_CONF_LISTENER | DAHDI_CONF_TALKER);
/* first put the channel on the conference in announce mode */
if (ioctl(myrpt->pchannel->fds[0], ZT_SETCONF, &ci) == -1) {
if (ioctl(myrpt->pchannel->fds[0], DAHDI_SETCONF, &ci) == -1) {
ast_log(LOG_WARNING, "Unable to set conference mode to Announce\n");
}
pthread_exit(NULL);
@@ -2809,7 +2805,7 @@ static int function_ilink(struct rpt *myrpt, char *param, char *digits, int comm
char digitbuf[MAXNODESTR];
struct rpt_link *l;
int reconnects = 0;
ZT_CONFINFO ci; /* conference info */
struct dahdi_confinfo ci; /* conference info */
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(s1);
AST_APP_ARG(s2); /* XXX Never used. Scratch? XXX */
@@ -2963,9 +2959,9 @@ static int function_ilink(struct rpt *myrpt, char *param, char *digits, int comm
/* make a conference for the pseudo-one */
ci.chan = 0;
ci.confno = myrpt->conf;
ci.confmode = ZT_CONF_CONF | ZT_CONF_LISTENER | ZT_CONF_TALKER;
ci.confmode = DAHDI_CONF_CONF | DAHDI_CONF_LISTENER | DAHDI_CONF_TALKER;
/* first put the channel on the conference in proper mode */
if (ioctl(l->pchan->fds[0], ZT_SETCONF, &ci) == -1) {
if (ioctl(l->pchan->fds[0], DAHDI_SETCONF, &ci) == -1) {
ast_log(LOG_WARNING, "Unable to set conference mode to Announce\n");
ast_hangup(l->chan);
ast_hangup(l->pchan);
@@ -3074,9 +3070,9 @@ static int function_ilink(struct rpt *myrpt, char *param, char *digits, int comm
/* make a conference for the tx */
ci.chan = 0;
ci.confno = myrpt->conf;
ci.confmode = ZT_CONF_CONF | ZT_CONF_LISTENER | ZT_CONF_TALKER;
ci.confmode = DAHDI_CONF_CONF | DAHDI_CONF_LISTENER | DAHDI_CONF_TALKER;
/* first put the channel on the conference in proper mode */
if (ioctl(l->pchan->fds[0], ZT_SETCONF, &ci) == -1) {
if (ioctl(l->pchan->fds[0], DAHDI_SETCONF, &ci) == -1) {
ast_log(LOG_WARNING, "Unable to set conference mode to Announce\n");
ast_hangup(l->chan);
ast_hangup(l->pchan);
@@ -3321,7 +3317,7 @@ static int function_macro(struct rpt *myrpt, char *param, char *digitbuf, int co
return DC_ERROR;
}
myrpt->macrotimer = MACROTIME;
strncat(myrpt->macrobuf, val, sizeof(myrpt->macrobuf) - 1);
strncat(myrpt->macrobuf, val, sizeof(myrpt->macrobuf) - strlen(myrpt->macrobuf) - 1);
rpt_mutex_unlock(&myrpt->lock);
return DC_COMPLETE;
}
@@ -3369,7 +3365,7 @@ static int function_gosub(struct rpt *myrpt, char *param, char *digitbuf, int co
return DC_ERROR;
}
myrpt->gosubtimer = GOSUBTIME;
strncat(myrpt->gosubbuf, val, sizeof(myrpt->gosubbuf) - 1);
strncat(myrpt->gosubbuf, val, sizeof(myrpt->gosubbuf) - strlen(myrpt->gosubbuf) - 1);
rpt_mutex_unlock(&myrpt->lock);
return DC_COMPLETE;
}
@@ -3931,18 +3927,18 @@ static void rbi_out_parallel(struct rpt *myrpt, unsigned char *data)
static void rbi_out(struct rpt *myrpt, unsigned char *data)
{
struct zt_radio_param r = { 0, };
struct dahdi_radio_param r = { 0, };
r.radpar = ZT_RADPAR_REMMODE;
r.data = ZT_RADPAR_REM_RBI1;
r.radpar = DAHDI_RADPAR_REMMODE;
r.data = DAHDI_RADPAR_REM_RBI1;
/* if setparam ioctl fails, its probably not a pciradio card */
if (ioctl(myrpt->rxchannel->fds[0], ZT_RADIO_SETPARAM, &r) == -1) {
if (ioctl(myrpt->rxchannel->fds[0], DAHDI_RADIO_SETPARAM, &r) == -1) {
rbi_out_parallel(myrpt, data);
return;
}
r.radpar = ZT_RADPAR_REMCOMMAND;
r.radpar = DAHDI_RADPAR_REMCOMMAND;
memcpy(&r.data, data, 5);
if (ioctl(myrpt->rxchannel->fds[0], ZT_RADIO_SETPARAM, &r) == -1) {
if (ioctl(myrpt->rxchannel->fds[0], DAHDI_RADIO_SETPARAM, &r) == -1) {
ast_log(LOG_WARNING, "Cannot send RBI command for channel %s\n", myrpt->rxchannel->name);
return;
}
@@ -3951,7 +3947,7 @@ static void rbi_out(struct rpt *myrpt, unsigned char *data)
static int serial_remote_io(struct rpt *myrpt, unsigned char *txbuf, int txbytes, char *rxbuf, int rxmaxbytes, int asciiflag)
{
int i;
struct zt_radio_param prm;
struct dahdi_radio_param prm;
char *buf = alloca(30 + txbytes * 3);
int len;
@@ -3962,18 +3958,18 @@ static int serial_remote_io(struct rpt *myrpt, unsigned char *txbuf, int txbytes
strcat(buf + len, "\n");
ast_debug(1, "%s", buf);
prm.radpar = ZT_RADPAR_REMMODE;
prm.radpar = DAHDI_RADPAR_REMMODE;
if (asciiflag)
prm.data = ZT_RADPAR_REM_SERIAL_ASCII;
prm.data = DAHDI_RADPAR_REM_SERIAL_ASCII;
else
prm.data = ZT_RADPAR_REM_SERIAL;
if (ioctl(myrpt->rxchannel->fds[0], ZT_RADIO_SETPARAM, &prm) == -1)
prm.data = DAHDI_RADPAR_REM_SERIAL;
if (ioctl(myrpt->rxchannel->fds[0], DAHDI_RADIO_SETPARAM, &prm) == -1)
return -1;
prm.radpar = ZT_RADPAR_REMCOMMAND;
prm.radpar = DAHDI_RADPAR_REMCOMMAND;
prm.data = rxmaxbytes;
memcpy(prm.buf, txbuf, txbytes);
prm.index = txbytes;
if (ioctl(myrpt->rxchannel->fds[0], ZT_RADIO_SETPARAM, &prm) == -1)
if (ioctl(myrpt->rxchannel->fds[0], DAHDI_RADIO_SETPARAM, &prm) == -1)
return -1;
if (rxbuf) {
*rxbuf = 0;
@@ -4585,9 +4581,9 @@ static int rmt_telem_start(struct rpt *myrpt, struct ast_channel *chan, int dela
static int rmt_telem_finish(struct rpt *myrpt, struct ast_channel *chan)
{
struct zt_params par;
struct dahdi_params par;
if (ioctl(myrpt->txchannel->fds[0], ZT_GET_PARAMS, &par) == -1) {
if (ioctl(myrpt->txchannel->fds[0], DAHDI_GET_PARAMS, &par) == -1) {
return -1;
}
@@ -5661,7 +5657,7 @@ static void *rpt(void *this)
const char *idtalkover;
int ms = MSWAIT, i, lasttx=0, val, remrx=0, identqueued, othertelemqueued, tailmessagequeued, ctqueued;
struct ast_channel *who;
ZT_CONFINFO ci; /* conference info */
struct dahdi_confinfo ci; /* conference info */
time_t t;
struct rpt_link *l, *m;
struct rpt_tele *telem;
@@ -5783,9 +5779,9 @@ static void *rpt(void *this)
/* make a conference for the tx */
ci.chan = 0;
ci.confno = -1; /* make a new conf */
ci.confmode = ZT_CONF_CONF | ZT_CONF_LISTENER;
ci.confmode = DAHDI_CONF_CONF | DAHDI_CONF_LISTENER;
/* first put the channel on the conference in proper mode */
if (ioctl(myrpt->txchannel->fds[0], ZT_SETCONF, &ci) == -1) {
if (ioctl(myrpt->txchannel->fds[0], DAHDI_SETCONF, &ci) == -1) {
ast_log(LOG_WARNING, "Unable to set conference mode to Announce\n");
rpt_mutex_unlock(&myrpt->lock);
ast_hangup(myrpt->pchannel);
@@ -5800,10 +5796,10 @@ static void *rpt(void *this)
/* make a conference for the pseudo */
ci.chan = 0;
ci.confno = -1; /* make a new conf */
ci.confmode = ((myrpt->p.duplex == 2) || (myrpt->p.duplex == 4)) ? ZT_CONF_CONFANNMON :
(ZT_CONF_CONF | ZT_CONF_LISTENER | ZT_CONF_TALKER);
ci.confmode = ((myrpt->p.duplex == 2) || (myrpt->p.duplex == 4)) ? DAHDI_CONF_CONFANNMON :
(DAHDI_CONF_CONF | DAHDI_CONF_LISTENER | DAHDI_CONF_TALKER);
/* first put the channel on the conference in announce mode */
if (ioctl(myrpt->pchannel->fds[0], ZT_SETCONF, &ci) == -1) {
if (ioctl(myrpt->pchannel->fds[0], DAHDI_SETCONF, &ci) == -1) {
ast_log(LOG_WARNING, "Unable to set conference mode to Announce\n");
rpt_mutex_unlock(&myrpt->lock);
ast_hangup(myrpt->pchannel);
@@ -5830,9 +5826,9 @@ static void *rpt(void *this)
/* make a conference for the tx */
ci.chan = 0;
ci.confno = myrpt->txconf;
ci.confmode = ZT_CONF_CONF | ZT_CONF_TALKER ;
ci.confmode = DAHDI_CONF_CONF | DAHDI_CONF_TALKER ;
/* first put the channel on the conference in proper mode */
if (ioctl(myrpt->txpchannel->fds[0], ZT_SETCONF, &ci) == -1) {
if (ioctl(myrpt->txpchannel->fds[0], DAHDI_SETCONF, &ci) == -1) {
ast_log(LOG_WARNING, "Unable to set conference mode to Announce\n");
rpt_mutex_unlock(&myrpt->lock);
ast_hangup(myrpt->txpchannel);
@@ -6804,7 +6800,7 @@ static void *rpt_master(void *config)
}
}
usleep(2000000);
sleep(2);
}
ast_config_destroy(cfg);
pthread_exit(NULL);
@@ -6820,8 +6816,8 @@ static int rpt_exec(struct ast_channel *chan, void *data)
struct ast_channel *who;
struct ast_channel *cs[20];
struct rpt_link *l;
ZT_CONFINFO ci; /* conference info */
ZT_PARAMS par;
struct dahdi_confinfo ci; /* conference info */
struct dahdi_params par;
int ms, elap;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(node);
@@ -7059,9 +7055,9 @@ static int rpt_exec(struct ast_channel *chan, void *data)
/* make a conference for the tx */
ci.chan = 0;
ci.confno = myrpt->conf;
ci.confmode = ZT_CONF_CONF | ZT_CONF_LISTENER | ZT_CONF_TALKER;
ci.confmode = DAHDI_CONF_CONF | DAHDI_CONF_LISTENER | DAHDI_CONF_TALKER;
/* first put the channel on the conference in proper mode */
if (ioctl(l->pchan->fds[0], ZT_SETCONF, &ci) == -1) {
if (ioctl(l->pchan->fds[0], DAHDI_SETCONF, &ci) == -1) {
ast_log(LOG_WARNING, "Unable to set conference mode to Announce\n");
pthread_exit(NULL);
}
@@ -7074,7 +7070,7 @@ static int rpt_exec(struct ast_channel *chan, void *data)
if (chan->_state != AST_STATE_UP) {
ast_answer(chan);
}
return AST_PBX_KEEPALIVE;
return 0;
}
rpt_mutex_lock(&myrpt->lock);
/* if remote, error if anyone else already linked */
@@ -7183,13 +7179,13 @@ static int rpt_exec(struct ast_channel *chan, void *data)
/* if we are on 2w loop and are a remote, turn EC on */
if (myrpt->remote && (myrpt->rxchannel == myrpt->txchannel)) {
i = 128;
ioctl(myrpt->rxchannel->fds[0], ZT_ECHOCANCEL, &i);
ioctl(myrpt->rxchannel->fds[0], DAHDI_ECHOCANCEL, &i);
}
if (chan->_state != AST_STATE_UP) {
ast_answer(chan);
}
if (ioctl(myrpt->txchannel->fds[0], ZT_GET_PARAMS, &par) != -1) {
if (ioctl(myrpt->txchannel->fds[0], DAHDI_GET_PARAMS, &par) != -1) {
if (par.rxisoffhook) {
ast_indicate(chan, AST_CONTROL_RADIO_KEY);
myrpt->remoterx = 1;

View File

@@ -40,11 +40,13 @@ static char *app = "SendDTMF";
static char *synopsis = "Sends arbitrary DTMF digits";
static char *descrip =
" SendDTMF(digits[,timeout_ms]): Sends DTMF digits on a channel. \n"
" Accepted digits: 0-9, *#abcd, w (.5s pause)\n"
" SendDTMF(digits[,[timeout_ms][,duration_ms]]): Sends DTMF digits on a channel. \n"
" Accepted digits: 0-9, *#abcd, (default .25s pause between digits)\n"
" The application will either pass the assigned digits or terminate if it\n"
" encounters an error.\n";
" encounters an error.\n"
" Optional Params: \n"
" timeout_ms: pause between digits.\n"
" duration_ms: duration of each digit.\n";
static int senddtmf_exec(struct ast_channel *chan, void *vdata)
{
@@ -90,7 +92,7 @@ static int manager_play_dtmf(struct mansession *s, const struct message *m)
astman_send_error(s, m, "Channel not specified");
return 0;
}
if (!digit) {
if (ast_strlen_zero(digit)) {
astman_send_error(s, m, "No digit specified");
ast_channel_unlock(chan);
return 0;

View File

@@ -42,7 +42,7 @@ static const char *app = "SendText";
static const char *synopsis = "Send a Text Message";
static const char *descrip =
" SendText(text[,options]): Sends text to current channel (callee).\n"
" SendText(text): Sends text to current channel (callee).\n"
"Result of transmission will be stored in the SENDTEXTSTATUS\n"
"channel variable:\n"
" SUCCESS Transmission succeeded\n"
@@ -58,24 +58,23 @@ static int sendtext_exec(struct ast_channel *chan, void *data)
char *parse = NULL;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(text);
AST_APP_ARG(options);
);
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "SendText requires an argument (text[,options])\n");
/* NOT ast_strlen_zero, because some protocols (e.g. SIP) MUST be able to
* send a zero-length message. */
if (!data) {
ast_log(LOG_WARNING, "SendText requires an argument (text)\n");
return -1;
} else
parse = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, parse);
if (args.options) {
}
ast_channel_lock(chan);
if (!chan->tech->send_text) {
ast_channel_unlock(chan);
/* Does not support transport */
pbx_builtin_setvar_helper(chan, "SENDTEXTSTATUS", status);
return 0;
}
status = "FAILURE";

View File

@@ -68,7 +68,11 @@ static int setcallerid_pres_exec(struct ast_channel *chan, void *data)
deprecated = 1;
ast_log(LOG_WARNING, "SetCallerPres is deprecated. Please use Set(CALLERPRES()=%s) instead.\n", (char *)data);
}
pres = ast_parse_caller_presentation(data);
/* For interface consistency, permit the argument to be specified as a number */
if (sscanf(data, "%d", &pres) != 1 || pres < 0 || pres > 255 || (pres & 0x9c)) {
pres = ast_parse_caller_presentation(data);
}
if (pres < 0) {
ast_log(LOG_WARNING, "'%s' is not a valid presentation (see 'show application SetCallerPres')\n",

View File

@@ -528,9 +528,10 @@ static void packdate(unsigned char *o, time_t w)
}
/*! \brief unpack a date and return */
static time_t unpackdate(unsigned char *i)
static struct timeval unpackdate(unsigned char *i)
{
struct tm t;
struct ast_tm t;
t.tm_year = 100 + (i[0] & 0xF) * 10 + (i[0] >> 4);
t.tm_mon = (i[1] & 0xF) * 10 + (i[1] >> 4) - 1;
t.tm_mday = (i[2] & 0xF) * 10 + (i[2] >> 4);
@@ -542,7 +543,8 @@ static time_t unpackdate(unsigned char *i)
t.tm_min += 15 * ((i[6] & 0x7) * 10 + (i[6] >> 4));
else
t.tm_min -= 15 * ((i[6] & 0x7) * 10 + (i[6] >> 4));
return mktime(&t);
return ast_mktime(&t, NULL);
}
/*! \brief unpacks bytes (7 bit encoding) at i, len l septets,
@@ -743,7 +745,9 @@ static void sms_log(sms_t * h, char status)
}
*p++ = '\n';
*p = 0;
write(o, line, strlen(line));
if (write(o, line, strlen(line)) < 0) {
ast_log(LOG_WARNING, "write() failed: %s\n", strerror(errno));
}
close(o);
}
*h->oa = *h->da = h->udl = 0;
@@ -1058,7 +1062,7 @@ static unsigned char sms_handleincoming (sms_t * h)
p += unpackaddress(h->oa, h->imsg + p);
h->pid = h->imsg[p++];
h->dcs = h->imsg[p++];
h->scts.tv_sec = unpackdate(h->imsg + p);
h->scts = unpackdate(h->imsg + p);
p += 7;
p += unpacksms(h->dcs, h->imsg + p, h->udh, &h->udhl, h->ud, &h->udl, h->udhi);
h->rx = 1; /* received message */
@@ -1740,7 +1744,7 @@ static int sms_exec(struct ast_channel *chan, void *data)
sms_t h = { 0 };
/* argument parsing support */
struct ast_flags sms_flags;
char *parse, *sms_opts[OPTION_ARG_ARRAY_SIZE];
char *parse, *sms_opts[OPTION_ARG_ARRAY_SIZE] = { 0, };
char *p;
AST_DECLARE_APP_ARGS(sms_args,
AST_APP_ARG(queue);

View File

@@ -363,6 +363,7 @@ static int speech_create(struct ast_channel *chan, void *data)
pbx_builtin_setvar_helper(chan, "ERROR", "1");
return 0;
}
pbx_builtin_setvar_helper(chan, "ERROR", NULL);
datastore->data = speech;
ast_channel_datastore_add(chan, datastore);
@@ -696,7 +697,7 @@ static int speech_background(struct ast_channel *chan, void *data)
}
time(&start);
snprintf(tmp, sizeof(tmp), "%c", f->subclass);
strncat(dtmf, tmp, sizeof(dtmf));
strncat(dtmf, tmp, sizeof(dtmf) - strlen(dtmf) - 1);
/* If the maximum length of the DTMF has been reached, stop now */
if (max_dtmf_len && strlen(dtmf) == max_dtmf_len)
done = 1;
@@ -727,6 +728,7 @@ static int speech_background(struct ast_channel *chan, void *data)
speech->results->text = ast_strdup(dtmf);
speech->results->grammar = ast_strdup("dtmf");
}
ast_speech_change_state(speech, AST_SPEECH_STATE_NOT_READY);
}
/* See if it was because they hung up */

View File

@@ -25,6 +25,10 @@
* \ingroup applications
*/
/*** MODULEINFO
<use>res_agi</use>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
@@ -35,6 +39,11 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/manager.h"
#include "asterisk/channel.h"
/* usage of AGI is optional, so indicate that to the header file */
#define ASTERISK_AGI_OPTIONAL
#include "asterisk/agi.h"
static const char *app_gosub = "Gosub";
static const char *app_gosubif = "GosubIf";
static const char *app_return = "Return";
@@ -61,7 +70,6 @@ static const char *pop_descrip =
" StackPop():\n"
"Removes last label on the stack, discarding it.\n";
static void gosub_free(void *data);
static struct ast_datastore_info stack_info = {
@@ -92,28 +100,25 @@ static int frame_set_var(struct ast_channel *chan, struct gosub_stack_frame *fra
}
}
if (!ast_strlen_zero(value)) {
if (!found) {
variables = ast_var_assign(var, "");
AST_LIST_INSERT_HEAD(&frame->varshead, variables, entries);
pbx_builtin_pushvar_helper(chan, var, value);
} else
pbx_builtin_setvar_helper(chan, var, value);
manager_event(EVENT_FLAG_DIALPLAN, "VarSet",
"Channel: %s\r\n"
"Variable: LOCAL(%s)\r\n"
"Value: %s\r\n"
"Uniqueid: %s\r\n",
chan->name, var, value, chan->uniqueid);
if (!found) {
variables = ast_var_assign(var, "");
AST_LIST_INSERT_HEAD(&frame->varshead, variables, entries);
pbx_builtin_pushvar_helper(chan, var, value);
} else {
pbx_builtin_setvar_helper(chan, var, value);
}
manager_event(EVENT_FLAG_DIALPLAN, "VarSet",
"Channel: %s\r\n"
"Variable: LOCAL(%s)\r\n"
"Value: %s\r\n"
"Uniqueid: %s\r\n",
chan->name, var, value, chan->uniqueid);
return 0;
}
static void gosub_release_frame(struct ast_channel *chan, struct gosub_stack_frame *frame)
{
unsigned char i;
char argname[15];
struct ast_var_t *vardata;
/* If chan is not defined, then we're calling it as part of gosub_free,
@@ -122,14 +127,6 @@ static void gosub_release_frame(struct ast_channel *chan, struct gosub_stack_fra
* that frame, so that we re-expose the variables from the previous frame
* that were hidden by this one.
*/
if (chan) {
for (i = 1; i <= frame->arguments && i != 0; i++) {
snprintf(argname, sizeof(argname), "ARG%hhd", i);
pbx_builtin_setvar_helper(chan, argname, NULL);
}
}
/* Delete local variables */
while ((vardata = AST_LIST_REMOVE_HEAD(&frame->varshead, entries))) {
if (chan)
pbx_builtin_setvar_helper(chan, ast_var_name(vardata), NULL);
@@ -234,7 +231,7 @@ static int gosub_exec(struct ast_channel *chan, void *data)
);
if (ast_strlen_zero(data)) {
ast_log(LOG_ERROR, "%s requires an argument: %s([[context|]exten|]priority[(arg1[|...][|argN])])\n", app_gosub, app_gosub);
ast_log(LOG_ERROR, "%s requires an argument: %s([[context,]exten,]priority[(arg1[,...][,argN])])\n", app_gosub, app_gosub);
return -1;
}
@@ -274,8 +271,9 @@ static int gosub_exec(struct ast_channel *chan, void *data)
/* Create the return address, but don't save it until we know that the Gosub destination exists */
newframe = gosub_allocate_frame(chan->context, chan->exten, chan->priority + 1, args2.argc);
if (!newframe)
if (!newframe) {
return -1;
}
if (ast_parseable_goto(chan, label)) {
ast_log(LOG_ERROR, "Gosub address is invalid: '%s'\n", (char *)data);
@@ -283,6 +281,16 @@ static int gosub_exec(struct ast_channel *chan, void *data)
return -1;
}
if (!ast_exists_extension(chan, chan->context, chan->exten, ast_test_flag(chan, AST_FLAG_IN_AUTOLOOP) ? chan->priority + 1 : chan->priority, chan->cid.cid_num)) {
ast_log(LOG_ERROR, "Attempt to reach a non-existent destination for gosub: (Context:%s, Extension:%s, Priority:%d)\n",
chan->context, chan->exten, chan->priority);
ast_copy_string(chan->context, newframe->context, sizeof(chan->context));
ast_copy_string(chan->exten, newframe->extension, sizeof(chan->exten));
chan->priority = newframe->priority;
ast_free(newframe);
return -1;
}
/* Now that we know for certain that we're going to a new location, set our arguments */
for (i = 0; i < args2.argc; i++) {
snprintf(argname, sizeof(argname), "ARG%d", i + 1);
@@ -391,8 +399,113 @@ static struct ast_custom_function local_function = {
.read = local_read,
};
static int handle_gosub(struct ast_channel *chan, AGI *agi, int argc, char **argv)
{
int old_priority, priority;
char old_context[AST_MAX_CONTEXT], old_extension[AST_MAX_EXTENSION];
struct ast_app *theapp;
char *gosub_args;
if (argc < 4 || argc > 5) {
return RESULT_SHOWUSAGE;
}
ast_debug(1, "Gosub called with %d arguments: 0:%s 1:%s 2:%s 3:%s 4:%s\n", argc, argv[0], argv[1], argv[2], argv[3], argc == 5 ? argv[4] : "");
if (sscanf(argv[3], "%d", &priority) != 1 || priority < 1) {
/* Lookup the priority label */
if ((priority = ast_findlabel_extension(chan, argv[1], argv[2], argv[3], chan->cid.cid_num)) < 0) {
ast_log(LOG_ERROR, "Priority '%s' not found in '%s@%s'\n", argv[3], argv[2], argv[1]);
ast_agi_send(agi->fd, chan, "200 result=-1 Gosub label not found\n");
return RESULT_FAILURE;
}
} else if (!ast_exists_extension(chan, argv[1], argv[2], priority, chan->cid.cid_num)) {
ast_agi_send(agi->fd, chan, "200 result=-1 Gosub label not found\n");
return RESULT_FAILURE;
}
/* Save previous location, since we're going to change it */
ast_copy_string(old_context, chan->context, sizeof(old_context));
ast_copy_string(old_extension, chan->exten, sizeof(old_extension));
old_priority = chan->priority;
if (!(theapp = pbx_findapp("Gosub"))) {
ast_log(LOG_ERROR, "Gosub() cannot be found in the list of loaded applications\n");
ast_agi_send(agi->fd, chan, "503 result=-2 Gosub is not loaded\n");
return RESULT_FAILURE;
}
/* Apparently, if you run ast_pbx_run on a channel that already has a pbx
* structure, you need to add 1 to the priority to get it to go to the
* right place. But if it doesn't have a pbx structure, then leaving off
* the 1 is the right thing to do. See how this code differs when we
* call a Gosub for the CALLEE channel in Dial or Queue.
*/
if (argc == 5) {
if (asprintf(&gosub_args, "%s,%s,%d(%s)", argv[1], argv[2], priority + 1, argv[4]) < 0) {
ast_log(LOG_WARNING, "asprintf() failed: %s\n", strerror(errno));
gosub_args = NULL;
}
} else {
if (asprintf(&gosub_args, "%s,%s,%d", argv[1], argv[2], priority + 1) < 0) {
ast_log(LOG_WARNING, "asprintf() failed: %s\n", strerror(errno));
gosub_args = NULL;
}
}
if (gosub_args) {
int res;
ast_debug(1, "Trying gosub with arguments '%s'\n", gosub_args);
ast_copy_string(chan->context, "app_stack_gosub_virtual_context", sizeof(chan->context));
ast_copy_string(chan->exten, "s", sizeof(chan->exten));
chan->priority = 0;
if ((res = pbx_exec(chan, theapp, gosub_args)) == 0) {
struct ast_pbx *pbx = chan->pbx;
struct ast_pbx_args args;
memset(&args, 0, sizeof(args));
args.no_hangup_chan = 1;
/* Suppress warning about PBX already existing */
chan->pbx = NULL;
ast_agi_send(agi->fd, chan, "100 result=0 Trying...\n");
ast_pbx_run_args(chan, &args);
ast_agi_send(agi->fd, chan, "200 result=0 Gosub complete\n");
if (chan->pbx) {
ast_free(chan->pbx);
}
chan->pbx = pbx;
} else {
ast_agi_send(agi->fd, chan, "200 result=%d Gosub failed\n", res);
}
ast_free(gosub_args);
} else {
ast_agi_send(agi->fd, chan, "503 result=-2 Memory allocation failure\n");
return RESULT_FAILURE;
}
/* Restore previous location */
ast_copy_string(chan->context, old_context, sizeof(chan->context));
ast_copy_string(chan->exten, old_extension, sizeof(chan->exten));
chan->priority = old_priority;
return RESULT_SUCCESS;
}
static char usage_gosub[] =
" Usage: GOSUB <context> <extension> <priority> [<optional-argument>]\n"
" Cause the channel to execute the specified dialplan subroutine, returning\n"
" to the dialplan with execution of a Return()\n";
struct agi_command gosub_agi_command =
{ { "gosub", NULL }, handle_gosub, "Execute a dialplan subroutine", usage_gosub , 0 };
static int unload_module(void)
{
if (ast_agi_unregister) {
ast_agi_unregister(ast_module_info->self, &gosub_agi_command);
}
ast_unregister_application(app_return);
ast_unregister_application(app_pop);
ast_unregister_application(app_gosubif);
@@ -404,6 +517,13 @@ static int unload_module(void)
static int load_module(void)
{
/* usage of AGI is optional, so check to see if the ast_agi_register()
function is available; if so, use it.
*/
if (ast_agi_register) {
ast_agi_register(ast_module_info->self, &gosub_agi_command);
}
ast_register_application(app_pop, pop_exec, pop_synopsis, pop_descrip);
ast_register_application(app_return, return_exec, return_synopsis, return_descrip);
ast_register_application(app_gosubif, gosubif_exec, gosubif_synopsis, gosubif_descrip);

View File

@@ -33,6 +33,10 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/module.h"
#include "asterisk/app.h"
#include "asterisk/channel.h" /* autoservice */
#include "asterisk/strings.h"
#include "asterisk/threadstorage.h"
AST_THREADSTORAGE(buf_buf);
static char *app = "System";
@@ -62,6 +66,7 @@ static char *descrip2 =
static int system_exec_helper(struct ast_channel *chan, void *data, int failmode)
{
int res = 0;
struct ast_str *buf = ast_str_thread_get(&buf_buf, 16);
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "System requires an argument(command)\n");
@@ -72,7 +77,9 @@ static int system_exec_helper(struct ast_channel *chan, void *data, int failmode
ast_autoservice_start(chan);
/* Do our thing here */
res = ast_safe_system((char *)data);
ast_str_get_encoded_str(&buf, 0, (char *) data);
res = ast_safe_system(ast_str_buffer(buf));
if ((res < 0) && (errno != ECHILD)) {
ast_log(LOG_WARNING, "Unable to execute '%s'\n", (char *)data);
pbx_builtin_setvar_helper(chan, chanvar, "FAILURE");

View File

@@ -83,11 +83,11 @@ static int background_detect_exec(struct ast_channel *chan, void *data)
tmp = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, tmp);
if ((sscanf(args.silence, "%d", &x) == 1) && (x > 0))
if (!ast_strlen_zero(args.silence) && (sscanf(args.silence, "%d", &x) == 1) && (x > 0))
sil = x;
if ((sscanf(args.min, "%d", &x) == 1) && (x > 0))
if (!ast_strlen_zero(args.min) && (sscanf(args.min, "%d", &x) == 1) && (x > 0))
min = x;
if ((sscanf(args.max, "%d", &x) == 1) && (x > 0))
if (!ast_strlen_zero(args.max) && (sscanf(args.max, "%d", &x) == 1) && (x > 0))
max = x;
ast_debug(1, "Preparing detect of '%s', sil=%d, min=%d, max=%d\n", args.filename, sil, min, max);

View File

@@ -109,7 +109,7 @@ static int measurenoise(struct ast_channel *chan, int ms, char *who)
return (noise / samples);
}
static int sendnoise(struct ast_channel *chan, int ms)
static int sendnoise(struct ast_channel *chan, int ms)
{
int res;
res = ast_tonepair_start(chan, 1537, 2195, ms, 8192);
@@ -117,7 +117,7 @@ static int sendnoise(struct ast_channel *chan, int ms)
res = ast_waitfordigit(chan, ms);
ast_tonepair_stop(chan);
}
return res;
return res;
}
static int testclient_exec(struct ast_channel *chan, void *data)
@@ -127,41 +127,41 @@ static int testclient_exec(struct ast_channel *chan, void *data)
char fn[80];
char serverver[80];
FILE *f;
/* Check for test id */
if (ast_strlen_zero(testid)) {
ast_log(LOG_WARNING, "TestClient requires an argument - the test id\n");
return -1;
}
if (chan->_state != AST_STATE_UP)
res = ast_answer(chan);
/* Wait a few just to be sure things get started */
res = ast_safe_sleep(chan, 3000);
/* Transmit client version */
if (!res)
res = ast_dtmf_stream(chan, NULL, "8378*1#", 0, 0);
ast_debug(1, "Transmit client version\n");
/* Read server version */
ast_debug(1, "Read server version\n");
if (!res)
if (!res)
res = ast_app_getdata(chan, NULL, serverver, sizeof(serverver) - 1, 0);
if (res > 0)
res = 0;
ast_debug(1, "server version: %s\n", serverver);
if (res > 0)
res = 0;
if (!res)
res = ast_safe_sleep(chan, 1000);
/* Send test id */
if (!res)
res = ast_dtmf_stream(chan, NULL, testid, 0, 0);
if (!res)
res = ast_dtmf_stream(chan, NULL, "#", 0, 0);
if (!res)
res = ast_dtmf_stream(chan, NULL, testid, 0, 0);
if (!res)
res = ast_dtmf_stream(chan, NULL, "#", 0, 0);
ast_debug(1, "send test identifier: %s\n", testid);
if ((res >=0) && (!ast_strlen_zero(testid))) {
@@ -175,7 +175,7 @@ static int testclient_exec(struct ast_channel *chan, void *data)
fprintf(f, "CLIENTTEST ID: %s\n", testid);
fprintf(f, "ANSWER: PASS\n");
res = 0;
if (!res) {
/* Step 1: Wait for "1" */
ast_debug(1, "TestClient: 2. Wait DTMF 1\n");
@@ -186,8 +186,9 @@ static int testclient_exec(struct ast_channel *chan, void *data)
else
res = -1;
}
if (!res)
if (!res) {
res = ast_safe_sleep(chan, 1000);
}
if (!res) {
/* Step 2: Send "2" */
ast_debug(1, "TestClient: 2. Send DTMF 2\n");
@@ -203,7 +204,7 @@ static int testclient_exec(struct ast_channel *chan, void *data)
fprintf(f, "WAIT 1 SEC: %s\n", (res < 0) ? "FAIL" : "PASS");
if (res > 0)
res = 0;
}
}
if (!res) {
/* Step 4: Measure noise */
ast_debug(1, "TestClient: 4. Measure noise\n");
@@ -249,7 +250,7 @@ static int testclient_exec(struct ast_channel *chan, void *data)
}
if (!res) {
/* Step 9: Measure noise */
ast_debug(1, "TestClient: 6. Measure tone\n");
ast_debug(1, "TestClient: 9. Measure tone\n");
res = measurenoise(chan, 4000, "TestClient");
fprintf(f, "MEASURETONE: %s (%d)\n", (res < 0) ? "FAIL" : "PASS", res);
if (res > 0)
@@ -257,7 +258,7 @@ static int testclient_exec(struct ast_channel *chan, void *data)
}
if (!res) {
/* Step 10: Send "7" */
ast_debug(1, "TestClient: 7. Send DTMF 7\n");
ast_debug(1, "TestClient: 10. Send DTMF 7\n");
res = ast_dtmf_stream(chan, NULL, "7", 0, 0);
fprintf(f, "SEND DTMF 7: %s\n", (res < 0) ? "FAIL" : "PASS");
if (res > 0)
@@ -273,6 +274,9 @@ static int testclient_exec(struct ast_channel *chan, void *data)
else
res = -1;
}
if (!res) {
res = ast_safe_sleep(chan, 1000);
}
if (!res) {
/* Step 12: Hangup! */
ast_debug(1, "TestClient: 12. Hangup\n");
@@ -301,7 +305,7 @@ static int testserver_exec(struct ast_channel *chan, void *data)
res = ast_answer(chan);
/* Read version */
ast_debug(1, "Read client version\n");
if (!res)
if (!res)
res = ast_app_getdata(chan, NULL, testid, sizeof(testid) - 1, 0);
if (res > 0)
res = 0;
@@ -315,8 +319,8 @@ static int testserver_exec(struct ast_channel *chan, void *data)
if (res > 0)
res = 0;
if (!res)
res = ast_app_getdata(chan, NULL, testid, sizeof(testid) - 1, 0);
if (!res)
res = ast_app_getdata(chan, NULL, testid, sizeof(testid) - 1, 0);
ast_debug(1, "read test identifier: %s\n", testid);
/* Check for sneakyness */
if (strchr(testid, '/'))
@@ -368,7 +372,6 @@ static int testserver_exec(struct ast_channel *chan, void *data)
if (res > 0)
res = 0;
}
if (!res) {
/* Step 5: Wait one second */
ast_debug(1, "TestServer: 5. Wait one second\n");
@@ -377,7 +380,6 @@ static int testserver_exec(struct ast_channel *chan, void *data)
if (res > 0)
res = 0;
}
if (!res) {
/* Step 6: Measure noise */
ast_debug(1, "TestServer: 6. Measure tone\n");
@@ -386,7 +388,6 @@ static int testserver_exec(struct ast_channel *chan, void *data)
if (res > 0)
res = 0;
}
if (!res) {
/* Step 7: Send "5" */
ast_debug(1, "TestServer: 7. Send DTMF 5\n");
@@ -395,14 +396,13 @@ static int testserver_exec(struct ast_channel *chan, void *data)
if (res > 0)
res = 0;
}
if (!res) {
/* Step 8: Transmit tone noise */
ast_debug(1, "TestServer: 8. Transmit tone\n");
res = sendnoise(chan, 6000);
fprintf(f, "SENDTONE: %s\n", (res < 0) ? "FAIL" : "PASS");
}
if (!res || (res == '7')) {
/* Step 9: Wait for "7" */
ast_debug(1, "TestServer: 9. Wait DTMF 7\n");
@@ -414,8 +414,9 @@ static int testserver_exec(struct ast_channel *chan, void *data)
else
res = -1;
}
if (!res)
if (!res) {
res = ast_safe_sleep(chan, 1000);
}
if (!res) {
/* Step 10: Send "8" */
ast_debug(1, "TestServer: 10. Send DTMF 8\n");
@@ -451,7 +452,7 @@ static int unload_module(void)
res = ast_unregister_application(testc_app);
res |= ast_unregister_application(tests_app);
return res;
return res;
}
static int load_module(void)

View File

@@ -48,15 +48,22 @@ static char *descrip =
static int userevent_exec(struct ast_channel *chan, void *data)
{
char *parse, buf[2048] = "";
int x, buflen = 0;
char *parse;
int x;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(eventname);
AST_APP_ARG(extra)[100];
);
struct ast_str *body = ast_str_create(16);
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "UserEvent requires an argument (eventname,optional event body)\n");
ast_free(body);
return -1;
}
if (!body) {
ast_log(LOG_WARNING, "Unable to allocate buffer\n");
return -1;
}
@@ -65,13 +72,11 @@ static int userevent_exec(struct ast_channel *chan, void *data)
AST_STANDARD_APP_ARGS(args, parse);
for (x = 0; x < args.argc - 1; x++) {
ast_copy_string(buf + buflen, args.extra[x], sizeof(buf) - buflen - 2);
buflen += strlen(args.extra[x]);
ast_copy_string(buf + buflen, "\r\n", 3);
buflen += 2;
ast_str_append(&body, 0, "%s\r\n", args.extra[x]);
}
manager_event(EVENT_FLAG_USER, "UserEvent", "UserEvent: %s\r\n%s", args.eventname, buf);
manager_event(EVENT_FLAG_USER, "UserEvent", "UserEvent: %s\r\n%s", args.eventname, body->str);
ast_free(body);
return 0;
}

File diff suppressed because it is too large Load Diff

View File

@@ -69,7 +69,7 @@ static char *descrip =
"TIMEOUT - if exited without silence detected after timeout\n";
static int do_waiting(struct ast_channel *chan, int silencereqd, time_t waitstart, int timeout) {
struct ast_frame *f;
struct ast_frame *f = NULL;
int dspsilence = 0;
static int silencethreshold = 128;
int rfmt = 0;
@@ -78,48 +78,47 @@ static int do_waiting(struct ast_channel *chan, int silencereqd, time_t waitstar
time_t now;
rfmt = chan->readformat; /* Set to linear mode */
res = ast_set_read_format(chan, AST_FORMAT_SLINEAR);
if (res < 0) {
if ((res = ast_set_read_format(chan, AST_FORMAT_SLINEAR)) < 0) {
ast_log(LOG_WARNING, "Unable to set channel to linear mode, giving up\n");
return -1;
}
sildet = ast_dsp_new(); /* Create the silence detector */
if (!sildet) {
/* Create the silence detector */
if (!(sildet = ast_dsp_new())) {
ast_log(LOG_WARNING, "Unable to create silence detector :(\n");
return -1;
}
ast_dsp_set_threshold(sildet, silencethreshold);
/* Await silence... */
f = NULL;
for(;;) {
for (;;) {
/* Start with no silence received */
dspsilence = 0;
res = ast_waitfor(chan, silencereqd);
/* Must have gotten a hangup; let's exit */
if (res <= 0) {
f = NULL;
if (res < 0) {
pbx_builtin_setvar_helper(chan, "WAITSTATUS", "HANGUP");
break;
}
/* We waited and got no frame; sounds like digital silence or a muted digital channel */
if (!res) {
if (res == 0) {
dspsilence = silencereqd;
} else {
/* Looks like we did get a frame, so let's check it out */
f = ast_read(chan);
if (!f)
if (!(f = ast_read(chan))) {
pbx_builtin_setvar_helper(chan, "WAITSTATUS", "HANGUP");
break;
if (f && f->frametype == AST_FRAME_VOICE) {
ast_dsp_silence(sildet, f, &dspsilence);
ast_frfree(f);
}
if (f->frametype == AST_FRAME_VOICE) {
ast_dsp_silence(sildet, f, &dspsilence);
}
ast_frfree(f);
}
ast_verb(3, "Got %dms silence< %dms required\n", dspsilence, silencereqd);
ast_verb(6, "Got %dms silence< %dms required\n", dspsilence, silencereqd);
if (dspsilence >= silencereqd) {
ast_verb(3, "Exiting with %dms silence >= %dms required\n", dspsilence, silencereqd);
@@ -154,7 +153,9 @@ static int waitforsilence_exec(struct ast_channel *chan, void *data)
int iterations = 1, i;
time_t waitstart;
res = ast_answer(chan); /* Answer the channel */
if (chan->_state != AST_STATE_UP) {
res = ast_answer(chan); /* Answer the channel */
}
if (!data || ( (sscanf(data, "%d,%d,%d", &silencereqd, &iterations, &timeout) != 3) &&
(sscanf(data, "%d|%d", &silencereqd, &iterations) != 2) &&

View File

@@ -160,11 +160,16 @@ static int _while_exec(struct ast_channel *chan, void *data, int end)
return -1;
}
#if 0
/* dont want run away loops if the chan isn't even up
this is up for debate since it slows things down a tad ......
Debate is over... this prevents While/EndWhile from working
within the "h" extension. Not good.
*/
if (ast_waitfordigit(chan,1) < 0)
return -1;
#endif
for (x=0;;x++) {
if (get_index(chan, prefix, x)) {

243
autoconf/acx_pthread.m4 Normal file
View File

@@ -0,0 +1,243 @@
dnl @synopsis ACX_PTHREAD([ACTION-IF-FOUND[, ACTION-IF-NOT-FOUND]])
dnl
dnl @summary figure out how to build C programs using POSIX threads
dnl
dnl This macro figures out how to build C programs using POSIX threads.
dnl It sets the PTHREAD_LIBS output variable to the threads library and
dnl linker flags, and the PTHREAD_CFLAGS output variable to any special
dnl C compiler flags that are needed. (The user can also force certain
dnl compiler flags/libs to be tested by setting these environment
dnl variables.)
dnl
dnl Also sets PTHREAD_CC to any special C compiler that is needed for
dnl multi-threaded programs (defaults to the value of CC otherwise).
dnl (This is necessary on AIX to use the special cc_r compiler alias.)
dnl
dnl NOTE: You are assumed to not only compile your program with these
dnl flags, but also link it with them as well. e.g. you should link
dnl with $PTHREAD_CC $CFLAGS $PTHREAD_CFLAGS $LDFLAGS ... $PTHREAD_LIBS
dnl $LIBS
dnl
dnl If you are only building threads programs, you may wish to use
dnl these variables in your default LIBS, CFLAGS, and CC:
dnl
dnl LIBS="$PTHREAD_LIBS $LIBS"
dnl CFLAGS="$CFLAGS $PTHREAD_CFLAGS"
dnl CC="$PTHREAD_CC"
dnl
dnl In addition, if the PTHREAD_CREATE_JOINABLE thread-attribute
dnl constant has a nonstandard name, defines PTHREAD_CREATE_JOINABLE to
dnl that name (e.g. PTHREAD_CREATE_UNDETACHED on AIX).
dnl
dnl ACTION-IF-FOUND is a list of shell commands to run if a threads
dnl library is found, and ACTION-IF-NOT-FOUND is a list of commands to
dnl run it if it is not found. If ACTION-IF-FOUND is not specified, the
dnl default action will define HAVE_PTHREAD.
dnl
dnl Please let the authors know if this macro fails on any platform, or
dnl if you have any other suggestions or comments. This macro was based
dnl on work by SGJ on autoconf scripts for FFTW (www.fftw.org) (with
dnl help from M. Frigo), as well as ac_pthread and hb_pthread macros
dnl posted by Alejandro Forero Cuervo to the autoconf macro repository.
dnl We are also grateful for the helpful feedback of numerous users.
dnl
dnl @category InstalledPackages
dnl @author Steven G. Johnson <stevenj@alum.mit.edu>
dnl @version 2006-05-29
dnl @license GPLWithACException
AC_DEFUN([ACX_PTHREAD],
[
AC_REQUIRE([AC_CANONICAL_HOST])
AC_LANG_SAVE
AC_LANG_C
acx_pthread_ok=no
# We used to check for pthread.h first, but this fails if pthread.h
# requires special compiler flags (e.g. on True64 or Sequent).
# It gets checked for in the link test anyway.
# First of all, check if the user has set any of the PTHREAD_LIBS,
# etcetera environment variables, and if threads linking works using
# them:
if test x"$PTHREAD_LIBS$PTHREAD_CFLAGS" != x; then
save_CFLAGS="$CFLAGS"
CFLAGS="$CFLAGS $PTHREAD_CFLAGS"
save_LIBS="$LIBS"
LIBS="$PTHREAD_LIBS $LIBS"
AC_MSG_CHECKING([for pthread_join in LIBS=$PTHREAD_LIBS with CFLAGS=$PTHREAD_CFLAGS])
AC_TRY_LINK_FUNC(pthread_join, acx_pthread_ok=yes)
AC_MSG_RESULT($acx_pthread_ok)
if test x"$acx_pthread_ok" = xno; then
PTHREAD_LIBS=""
PTHREAD_CFLAGS=""
fi
LIBS="$save_LIBS"
CFLAGS="$save_CFLAGS"
fi
# We must check for the threads library under a number of different
# names; the ordering is very important because some systems
# (e.g. DEC) have both -lpthread and -lpthreads, where one of the
# libraries is broken (non-POSIX).
# Create a list of thread flags to try. Items starting with a "-" are
# C compiler flags, and other items are library names, except for "none"
# which indicates that we try without any flags at all, and "pthread-config"
# which is a program returning the flags for the Pth emulation library.
acx_pthread_flags="pthreads none -Kthread -kthread lthread -pthread -pthreads -mthreads pthread --thread-safe -mt pthread-config"
# The ordering *is* (sometimes) important. Some notes on the
# individual items follow:
# pthreads: AIX (must check this before -lpthread)
# none: in case threads are in libc; should be tried before -Kthread and
# other compiler flags to prevent continual compiler warnings
# -Kthread: Sequent (threads in libc, but -Kthread needed for pthread.h)
# -kthread: FreeBSD kernel threads (preferred to -pthread since SMP-able)
# lthread: LinuxThreads port on FreeBSD (also preferred to -pthread)
# -pthread: Linux/gcc (kernel threads), BSD/gcc (userland threads)
# -pthreads: Solaris/gcc
# -mthreads: Mingw32/gcc, Lynx/gcc
# -mt: Sun Workshop C (may only link SunOS threads [-lthread], but it
# doesn't hurt to check since this sometimes defines pthreads too;
# also defines -D_REENTRANT)
# ... -mt is also the pthreads flag for HP/aCC
# pthread: Linux, etcetera
# --thread-safe: KAI C++
# pthread-config: use pthread-config program (for GNU Pth library)
case "${host_cpu}-${host_os}" in
*solaris*)
# On Solaris (at least, for some versions), libc contains stubbed
# (non-functional) versions of the pthreads routines, so link-based
# tests will erroneously succeed. (We need to link with -pthreads/-mt/
# -lpthread.) (The stubs are missing pthread_cleanup_push, or rather
# a function called by this macro, so we could check for that, but
# who knows whether they'll stub that too in a future libc.) So,
# we'll just look for -pthreads and -lpthread first:
acx_pthread_flags="-pthreads pthread -mt -pthread $acx_pthread_flags"
;;
esac
if test x"$acx_pthread_ok" = xno; then
for flag in $acx_pthread_flags; do
case $flag in
none)
AC_MSG_CHECKING([whether pthreads work without any flags])
;;
-*)
AC_MSG_CHECKING([whether pthreads work with $flag])
PTHREAD_CFLAGS="$flag"
;;
pthread-config)
AC_CHECK_PROG(acx_pthread_config, pthread-config, yes, no)
if test x"$acx_pthread_config" = xno; then continue; fi
PTHREAD_CFLAGS="`pthread-config --cflags`"
PTHREAD_LIBS="`pthread-config --ldflags` `pthread-config --libs`"
;;
*)
AC_MSG_CHECKING([for the pthreads library -l$flag])
PTHREAD_LIBS="-l$flag"
;;
esac
save_LIBS="$LIBS"
save_CFLAGS="$CFLAGS"
LIBS="$PTHREAD_LIBS $LIBS"
CFLAGS="$CFLAGS $PTHREAD_CFLAGS"
# Check for various functions. We must include pthread.h,
# since some functions may be macros. (On the Sequent, we
# need a special flag -Kthread to make this header compile.)
# We check for pthread_join because it is in -lpthread on IRIX
# while pthread_create is in libc. We check for pthread_attr_init
# due to DEC craziness with -lpthreads. We check for
# pthread_cleanup_push because it is one of the few pthread
# functions on Solaris that doesn't have a non-functional libc stub.
# We try pthread_create on general principles.
AC_TRY_LINK([#include <pthread.h>],
[pthread_t th; pthread_join(th, 0);
pthread_attr_init(0); pthread_cleanup_push(0, 0);
pthread_create(0,0,0,0); pthread_cleanup_pop(0); ],
[acx_pthread_ok=yes])
LIBS="$save_LIBS"
CFLAGS="$save_CFLAGS"
AC_MSG_RESULT($acx_pthread_ok)
if test "x$acx_pthread_ok" = xyes; then
break;
fi
PTHREAD_LIBS=""
PTHREAD_CFLAGS=""
done
fi
# Various other checks:
if test "x$acx_pthread_ok" = xyes; then
save_LIBS="$LIBS"
LIBS="$PTHREAD_LIBS $LIBS"
save_CFLAGS="$CFLAGS"
CFLAGS="$CFLAGS $PTHREAD_CFLAGS"
# Detect AIX lossage: JOINABLE attribute is called UNDETACHED.
AC_MSG_CHECKING([for joinable pthread attribute])
attr_name=unknown
for attr in PTHREAD_CREATE_JOINABLE PTHREAD_CREATE_UNDETACHED; do
AC_TRY_LINK([#include <pthread.h>], [int attr=$attr; return attr;],
[attr_name=$attr; break])
done
AC_MSG_RESULT($attr_name)
if test "$attr_name" != PTHREAD_CREATE_JOINABLE; then
AC_DEFINE_UNQUOTED(PTHREAD_CREATE_JOINABLE, $attr_name,
[Define to necessary symbol if this constant
uses a non-standard name on your system.])
fi
AC_MSG_CHECKING([if more special flags are required for pthreads])
flag=no
case "${host_cpu}-${host_os}" in
*-aix* | *-freebsd* | *-darwin*) flag="-D_THREAD_SAFE";;
*solaris* | *-osf* | *-hpux*) flag="-D_REENTRANT";;
esac
AC_MSG_RESULT(${flag})
if test "x$flag" != xno; then
PTHREAD_CFLAGS="$flag $PTHREAD_CFLAGS"
fi
LIBS="$save_LIBS"
CFLAGS="$save_CFLAGS"
# More AIX lossage: must compile with xlc_r or cc_r
if test x"$GCC" != xyes; then
AC_CHECK_PROGS(PTHREAD_CC, xlc_r cc_r, ${CC})
else
PTHREAD_CC=$CC
fi
else
PTHREAD_CC="$CC"
fi
AC_SUBST(PTHREAD_LIBS)
AC_SUBST(PTHREAD_CFLAGS)
AC_SUBST(PTHREAD_CC)
# Finally, execute ACTION-IF-FOUND/ACTION-IF-NOT-FOUND:
if test x"$acx_pthread_ok" = xyes; then
ifelse([$1],,AC_DEFINE(HAVE_PTHREAD,1,[Define if you have POSIX threads libraries and header files.]),[$1])
:
else
acx_pthread_ok=no
$2
fi
AC_LANG_RESTORE
])dnl ACX_PTHREAD

View File

@@ -0,0 +1,31 @@
# Check if a given expression will compile using a certain header.
# AST_C_COMPILE_CHECK([package], [expression], [header file], [version], [description])
AC_DEFUN([AST_C_COMPILE_CHECK],
[
if test "x${PBX_$1}" != "x1" -a "${USE_$1}" != "no"; then
if test "x$5" != "x"; then
AC_MSG_CHECKING([for $5])
else
AC_MSG_CHECKING([if "$2" compiles using $3])
fi
saved_cppflags="${CPPFLAGS}"
if test "x${$1_DIR}" != "x"; then
$1_INCLUDE="-I${$1_DIR}/include"
fi
CPPFLAGS="${CPPFLAGS} ${$1_INCLUDE}"
AC_COMPILE_IFELSE(
[ AC_LANG_PROGRAM( [#include <$3>],
[ $2; ]
)],
[ AC_MSG_RESULT(yes)
PBX_$1=1
AC_DEFINE([HAVE_$1], 1, [Define if your system has the $1 headers.])
AC_DEFINE([HAVE_$1_VERSION], $4, [Define $1 headers version])
],
[ AC_MSG_RESULT(no) ]
)
CPPFLAGS="${saved_cppflags}"
fi
])

View File

@@ -0,0 +1,33 @@
# Check if a given macro is defined in a certain header.
# AST_C_DEFINE_CHECK([package], [macro name], [header file], [version])
AC_DEFUN([AST_C_DEFINE_CHECK],
[
if test "x${PBX_$1}" != "x1"; then
AC_MSG_CHECKING([for $2 in $3])
saved_cppflags="${CPPFLAGS}"
if test "x${$1_DIR}" != "x"; then
$1_INCLUDE="-I${$1_DIR}/include"
fi
CPPFLAGS="${CPPFLAGS} ${$1_INCLUDE}"
AC_COMPILE_IFELSE(
[ AC_LANG_PROGRAM( [#include <$3>],
[#if defined($2)
int foo = 0;
#else
int foo = bar;
#endif
0
])],
[ AC_MSG_RESULT(yes)
PBX_$1=1
AC_DEFINE([HAVE_$1], 1, [Define if your system has the $1 headers.])
AC_DEFINE([HAVE_$1_VERSION], $4, [Define $1 headers version])
],
[ AC_MSG_RESULT(no) ]
)
CPPFLAGS="${saved_cppflags}"
fi
AC_SUBST(PBX_$1)
])

View File

@@ -0,0 +1,20 @@
AC_DEFUN([AST_CHECK_GNU_MAKE], [AC_CACHE_CHECK([for GNU make], [ac_cv_GNU_MAKE],
ac_cv_GNU_MAKE='Not Found' ;
ac_cv_GNU_MAKE_VERSION_MAJOR=0 ;
ac_cv_GNU_MAKE_VERSION_MINOR=0 ;
for a in make gmake gnumake ; do
if test -z "$a" ; then continue ; fi ;
if ( sh -c "$a --version" 2> /dev/null | grep GNU 2>&1 > /dev/null ) ; then
ac_cv_GNU_MAKE=$a ;
ac_cv_GNU_MAKE_VERSION_MAJOR=`$ac_cv_GNU_MAKE --version | grep "GNU Make" | cut -f3 -d' ' | cut -f1 -d'.'`
ac_cv_GNU_MAKE_VERSION_MINOR=`$ac_cv_GNU_MAKE --version | grep "GNU Make" | cut -f2 -d'.' | cut -c1-2`
break;
fi
done ;
) ;
if test "x$ac_cv_GNU_MAKE" = "xNot Found" ; then
AC_MSG_ERROR( *** Please install GNU make. It is required to build Asterisk!)
exit 1
fi
AC_SUBST([GNU_MAKE], [$ac_cv_GNU_MAKE])
])

View File

@@ -0,0 +1,23 @@
# Check whether any of the mandatory modules are not present, and
# print error messages in case. The mandatory list is built using
# --with-* arguments when invoking configure.
AC_DEFUN([AST_CHECK_MANDATORY],
[
AC_MSG_CHECKING([for mandatory modules: ${ac_mandatory_list}])
err=0;
for i in ${ac_mandatory_list}; do
eval "a=\${PBX_$i}"
if test "x${a}" = "x1" ; then continue; fi
if test ${err} = "0" ; then AC_MSG_RESULT(fail) ; fi
AC_MSG_RESULT()
eval "a=\${${i}_OPTION}"
AC_MSG_NOTICE([***])
AC_MSG_NOTICE([*** The $i installation appears to be missing or broken.])
AC_MSG_NOTICE([*** Either correct the installation, or run configure])
AC_MSG_NOTICE([*** including --without-${a}.])
err=1
done
if test $err = 1 ; then exit 1; fi
AC_MSG_RESULT(ok)
])

View File

@@ -0,0 +1,132 @@
AC_DEFUN([AST_CHECK_OPENH323], [
OPENH323_INCDIR=
OPENH323_LIBDIR=
AC_LANG_PUSH([C++])
if test "${OPENH323DIR:-unset}" != "unset" ; then
AC_CHECK_HEADER(${OPENH323DIR}/version.h, HAS_OPENH323=1, )
fi
if test "${HAS_OPENH323:-unset}" = "unset" ; then
AC_CHECK_HEADER(${PWLIBDIR}/../openh323/version.h, OPENH323DIR="${PWLIBDIR}/../openh323"; HAS_OPENH323=1, )
if test "${HAS_OPENH323:-unset}" != "unset" ; then
OPENH323DIR="${PWLIBDIR}/../openh323"
saved_cppflags="${CPPFLAGS}"
CPPFLAGS="${CPPFLAGS} -I${PWLIB_INCDIR}/openh323 -I${PWLIB_INCDIR}"
AC_CHECK_HEADER(${OPENH323DIR}/include/h323.h, , OPENH323_INCDIR="${PWLIB_INCDIR}/openh323"; OPENH323_LIBDIR="${PWLIB_LIBDIR}", [#include <ptlib.h>])
CPPFLAGS="${saved_cppflags}"
else
saved_cppflags="${CPPFLAGS}"
CPPFLAGS="${CPPFLAGS} -I${HOME}/openh323/include -I${PWLIB_INCDIR}"
AC_CHECK_HEADER(${HOME}/openh323/include/h323.h, HAS_OPENH323=1, , [#include <ptlib.h>])
CPPFLAGS="${saved_cppflags}"
if test "${HAS_OPENH323:-unset}" != "unset" ; then
OPENH323DIR="${HOME}/openh323"
else
saved_cppflags="${CPPFLAGS}"
CPPFLAGS="${CPPFLAGS} -I/usr/local/include/openh323 -I${PWLIB_INCDIR}"
AC_CHECK_HEADER(/usr/local/include/openh323/h323.h, HAS_OPENH323=1, , [#include <ptlib.h>])
CPPFLAGS="${saved_cppflags}"
if test "${HAS_OPENH323:-unset}" != "unset" ; then
OPENH323DIR="/usr/local/share/openh323"
OPENH323_INCDIR="/usr/local/include/openh323"
if test "x$LIB64" != "x" && test -d "/usr/local/lib64"; then
OPENH323_LIBDIR="/usr/local/lib64"
else
OPENH323_LIBDIR="/usr/local/lib"
fi
else
saved_cppflags="${CPPFLAGS}"
CPPFLAGS="${CPPFLAGS} -I/usr/include/openh323 -I${PWLIB_INCDIR}"
AC_CHECK_HEADER(/usr/include/openh323/h323.h, HAS_OPENH323=1, , [#include <ptlib.h>])
CPPFLAGS="${saved_cppflags}"
if test "${HAS_OPENH323:-unset}" != "unset" ; then
OPENH323DIR="/usr/share/openh323"
OPENH323_INCDIR="/usr/include/openh323"
if test "x$LIB64" != "x" && test -d "/usr/local/lib64"; then
OPENH323_LIBDIR="/usr/lib64"
else
OPENH323_LIBDIR="/usr/lib"
fi
fi
fi
fi
fi
fi
if test "${HAS_OPENH323:-unset}" != "unset" ; then
if test "${OPENH323_INCDIR:-unset}" = "unset"; then
OPENH323_INCDIR="${OPENH323DIR}/include"
fi
if test "${OPENH323_LIBDIR:-unset}" = "unset"; then
OPENH323_LIBDIR="${OPENH323DIR}/lib"
fi
OPENH323_LIBDIR="`cd ${OPENH323_LIBDIR}; pwd`"
OPENH323_INCDIR="`cd ${OPENH323_INCDIR}; pwd`"
OPENH323DIR="`cd ${OPENH323DIR}; pwd`"
AC_SUBST([OPENH323DIR])
AC_SUBST([OPENH323_INCDIR])
AC_SUBST([OPENH323_LIBDIR])
fi
AC_LANG_POP([C++])
])
AC_DEFUN([AST_CHECK_OPENH323_BUILD], [
if test "${HAS_OPENH323:-unset}" != "unset"; then
AC_MSG_CHECKING(OpenH323 build option)
OPENH323_SUFFIX=
prefixes="h323_${PWLIB_PLATFORM}_ h323_ openh323"
for pfx in $prefixes; do
#files=`ls -l /usr/local/lib/lib${pfx}*.so* 2>/dev/null`
files=`ls -l ${OPENH323_LIBDIR}/lib${pfx}*.so* 2>/dev/null`
if test -z "$files"; then
# check the default location
files=`ls -l /usr/local/lib/lib${pfx}*.so* 2>/dev/null`
fi
libfile=
if test -n "$files"; then
for f in $files; do
if test -f $f -a ! -L $f; then
libfile=`basename $f`
break;
fi
done
fi
if test -n "$libfile"; then
OPENH323_PREFIX=$pfx
break;
fi
done
if test "${libfile:-unset}" != "unset"; then
OPENH323_SUFFIX=`eval "echo ${libfile} | sed -e 's/lib${OPENH323_PREFIX}\(@<:@^.@:>@*\)\..*/\1/'"`
fi
case "${OPENH323_SUFFIX}" in
n)
OPENH323_BUILD="notrace";;
r)
OPENH323_BUILD="opt";;
d)
OPENH323_BUILD="debug";;
*)
if test "${OPENH323_PREFIX:-undef}" = "openh323"; then
notrace=`eval "grep NOTRACE ${OPENH323DIR}/openh323u.mak | grep = | sed -e 's/@<:@A-Z0-9_@:>@*@<:@ @:>@*=@<:@ @:>@*//'"`
if test "x$notrace" = "x"; then
notrace="0"
fi
if test "$notrace" -ne 0; then
OPENH323_BUILD="notrace"
else
OPENH323_BUILD="opt"
fi
OPENH323_LIB="-l${OPENH323_PREFIX}"
else
OPENH323_BUILD="notrace"
fi
;;
esac
AC_MSG_RESULT(${OPENH323_BUILD})
AC_SUBST([OPENH323_SUFFIX])
AC_SUBST([OPENH323_BUILD])
fi
])

263
autoconf/ast_check_pwlib.m4 Normal file
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@@ -0,0 +1,263 @@
AC_DEFUN([AST_CHECK_PWLIB_PLATFORM], [
PWLIB_OSTYPE=
case "$host_os" in
linux*) PWLIB_OSTYPE=linux ;
;;
freebsd* ) PWLIB_OSTYPE=FreeBSD ;
;;
openbsd* ) PWLIB_OSTYPE=OpenBSD ;
ENDLDLIBS="-lossaudio" ;
;;
netbsd* ) PWLIB_OSTYPE=NetBSD ;
ENDLDLIBS="-lossaudio" ;
;;
solaris* | sunos* ) PWLIB_OSTYPE=solaris ;
;;
darwin* ) PWLIB_OSTYPE=Darwin ;
;;
beos*) PWLIB_OSTYPE=beos ;
STDCCFLAGS="$STDCCFLAGS -D__BEOS__"
;;
cygwin*) PWLIB_OSTYPE=cygwin ;
;;
mingw*) PWLIB_OSTYPE=mingw ;
STDCCFLAGS="$STDCCFLAGS -mms-bitfields" ;
ENDLDLIBS="-lwinmm -lwsock32 -lsnmpapi -lmpr -lcomdlg32 -lgdi32 -lavicap32" ;
;;
* ) PWLIB_OSTYPE="$host_os" ;
AC_MSG_WARN("OS $PWLIB_OSTYPE not recognized - proceed with caution!") ;
;;
esac
PWLIB_MACHTYPE=
case "$host_cpu" in
x86 | i686 | i586 | i486 | i386 ) PWLIB_MACHTYPE=x86
;;
x86_64) PWLIB_MACHTYPE=x86_64 ;
P_64BIT=1 ;
LIB64=1 ;
;;
alpha | alphaev56 | alphaev6 | alphaev67 | alphaev7) PWLIB_MACHTYPE=alpha ;
P_64BIT=1 ;
;;
sparc ) PWLIB_MACHTYPE=sparc ;
;;
powerpc ) PWLIB_MACHTYPE=ppc ;
;;
ppc ) PWLIB_MACHTYPE=ppc ;
;;
powerpc64 ) PWLIB_MACHTYPE=ppc64 ;
P_64BIT=1 ;
LIB64=1 ;
;;
ppc64 ) PWLIB_MACHTYPE=ppc64 ;
P_64BIT=1 ;
LIB64=1 ;
;;
ia64) PWLIB_MACHTYPE=ia64 ;
P_64BIT=1 ;
;;
s390x) PWLIB_MACHTYPE=s390x ;
P_64BIT=1 ;
LIB64=1 ;
;;
s390) PWLIB_MACHTYPE=s390 ;
;;
* ) PWLIB_MACHTYPE="$host_cpu";
AC_MSG_WARN("CPU $PWLIB_MACHTYPE not recognized - proceed with caution!") ;;
esac
PWLIB_PLATFORM="${PWLIB_OSTYPE}_${PWLIB_MACHTYPE}"
AC_SUBST([PWLIB_PLATFORM])
])
AC_DEFUN([AST_CHECK_PWLIB], [
PWLIB_INCDIR=
PWLIB_LIBDIR=
AC_LANG_PUSH([C++])
if test "${PWLIBDIR:-unset}" != "unset" ; then
AC_CHECK_HEADER(${PWLIBDIR}/version.h, HAS_PWLIB=1, )
fi
if test "${HAS_PWLIB:-unset}" = "unset" ; then
if test "${OPENH323DIR:-unset}" != "unset"; then
AC_CHECK_HEADER(${OPENH323DIR}/../pwlib/version.h, HAS_PWLIB=1, )
fi
if test "${HAS_PWLIB:-unset}" != "unset" ; then
PWLIBDIR="${OPENH323DIR}/../pwlib"
else
AC_CHECK_HEADER(${HOME}/pwlib/include/ptlib.h, HAS_PWLIB=1, )
if test "${HAS_PWLIB:-unset}" != "unset" ; then
PWLIBDIR="${HOME}/pwlib"
else
AC_CHECK_HEADER(/usr/local/include/ptlib.h, HAS_PWLIB=1, )
if test "${HAS_PWLIB:-unset}" != "unset" ; then
AC_PATH_PROG(PTLIB_CONFIG, ptlib-config, , /usr/local/bin$PATH_SEPARATOR/usr/local/share/pwlib/make)
PWLIB_INCDIR="/usr/local/include"
PWLIB_LIBDIR=`${PTLIB_CONFIG} --pwlibdir 2>/dev/null`
if test "${PWLIB_LIBDIR:-unset}" = "unset"; then
PWLIB_LIBDIR=`${PTLIB_CONFIG} --ptlibdir 2>/dev/null`
fi
if test "${PWLIB_LIBDIR:-unset}" = "unset"; then
if test "x$LIB64" != "x"; then
PWLIB_LIBDIR="/usr/local/lib64"
else
PWLIB_LIBDIR="/usr/local/lib"
fi
fi
PWLIB_LIB=`${PTLIB_CONFIG} --ldflags --libs`
PWLIB_LIB="-L${PWLIB_LIBDIR} `echo ${PWLIB_LIB}`"
else
AC_CHECK_HEADER(/usr/include/ptlib.h, HAS_PWLIB=1, )
if test "${HAS_PWLIB:-unset}" != "unset" ; then
AC_PATH_PROG(PTLIB_CONFIG, ptlib-config, , /usr/bin$PATH_SEPARATOR/usr/share/pwlib/make)
PWLIB_INCDIR="/usr/include"
PWLIB_LIBDIR=`${PTLIB_CONFIG} --pwlibdir 2>/dev/null`
if test "${PWLIB_LIBDIR:-unset}" = "unset"; then
PWLIB_LIBDIR=`${PTLIB_CONFIG} --ptlibdir 2>/dev/null`
fi
if test "${PWLIB_LIBDIR:-unset}" = "unset"; then
if test "x$LIB64" != "x"; then
PWLIB_LIBDIR="/usr/lib64"
else
PWLIB_LIBDIR="/usr/lib"
fi
fi
PWLIB_LIB=`${PTLIB_CONFIG} --ldflags --libs`
PWLIB_LIB="-L${PWLIB_LIBDIR} `echo ${PWLIB_LIB}`"
fi
fi
fi
fi
fi
#if test "${HAS_PWLIB:-unset}" = "unset" ; then
# echo "Cannot find pwlib - please install or set PWLIBDIR and try again"
# exit
#fi
if test "${HAS_PWLIB:-unset}" != "unset" ; then
if test "${PWLIBDIR:-unset}" = "unset" ; then
if test "${PTLIB_CONFIG:-unset}" != "unset" ; then
PWLIBDIR=`$PTLIB_CONFIG --prefix`
else
echo "Cannot find ptlib-config - please install and try again"
exit
fi
fi
if test "x$PWLIBDIR" = "x/usr" -o "x$PWLIBDIR" = "x/usr/"; then
PWLIBDIR="/usr/share/pwlib"
PWLIB_INCDIR="/usr/include"
if test "x$LIB64" != "x"; then
PWLIB_LIBDIR="/usr/lib64"
else
PWLIB_LIBDIR="/usr/lib"
fi
fi
if test "x$PWLIBDIR" = "x/usr/local" -o "x$PWLIBDIR" = "x/usr/"; then
PWLIBDIR="/usr/local/share/pwlib"
PWLIB_INCDIR="/usr/local/include"
if test "x$LIB64" != "x"; then
PWLIB_LIBDIR="/usr/local/lib64"
else
PWLIB_LIBDIR="/usr/local/lib"
fi
fi
if test "${PWLIB_INCDIR:-unset}" = "unset"; then
PWLIB_INCDIR="${PWLIBDIR}/include"
fi
if test "${PWLIB_LIBDIR:-unset}" = "unset"; then
PWLIB_LIBDIR="${PWLIBDIR}/lib"
fi
AC_SUBST([PWLIBDIR])
AC_SUBST([PWLIB_INCDIR])
AC_SUBST([PWLIB_LIBDIR])
fi
AC_LANG_POP([C++])
])
AC_DEFUN([AST_CHECK_PWLIB_VERSION], [
if test "x$7" != "x"; then
VNAME="$7"
else
VNAME="$2_VERSION"
fi
if test "${HAS_$2:-unset}" != "unset"; then
$2_VERSION=`grep "$VNAME" ${$2_INCDIR}/$3 | sed -e 's/[[[:space:]]]\{1,\}/ /g' | cut -f3 -d ' ' | sed -e 's/"//g'`
$2_MAJOR_VERSION=`echo ${$2_VERSION} | cut -f1 -d.`
$2_MINOR_VERSION=`echo ${$2_VERSION} | cut -f2 -d.`
$2_BUILD_NUMBER=`echo ${$2_VERSION} | cut -f3 -d.`
let $2_VER=${$2_MAJOR_VERSION}*10000+${$2_MINOR_VERSION}*100+${$2_BUILD_NUMBER}
let $2_REQ=$4*10000+$5*100+$6
AC_MSG_CHECKING(if $1 version ${$2_VERSION} is compatible with chan_h323)
if test ${$2_VER} -lt ${$2_REQ}; then
AC_MSG_RESULT(no)
unset HAS_$2
else
AC_MSG_RESULT(yes)
fi
fi
])
AC_DEFUN([AST_CHECK_PWLIB_BUILD], [
if test "${HAS_$2:-unset}" != "unset"; then
AC_MSG_CHECKING($1 installation validity)
saved_cppflags="${CPPFLAGS}"
saved_libs="${LIBS}"
if test "${$2_LIB:-unset}" != "unset"; then
LIBS="${LIBS} ${$2_LIB} $7"
else
LIBS="${LIBS} -L${$2_LIBDIR} -l${PLATFORM_$2} $7"
fi
CPPFLAGS="${CPPFLAGS} -I${$2_INCDIR} $6"
AC_LANG_PUSH([C++])
AC_LINK_IFELSE(
[AC_LANG_PROGRAM([$4],[$5])],
[ AC_MSG_RESULT(yes)
ac_cv_lib_$2="yes"
],
[ AC_MSG_RESULT(no)
ac_cv_lib_$2="no"
]
)
AC_LANG_POP([C++])
LIBS="${saved_libs}"
CPPFLAGS="${saved_cppflags}"
if test "${ac_cv_lib_$2}" = "yes"; then
if test "${$2_LIB:-undef}" = "undef"; then
if test "${$2_LIBDIR}" != "" -a "${$2_LIBDIR}" != "/usr/lib"; then
$2_LIB="-L${$2_LIBDIR} -l${PLATFORM_$2}"
else
$2_LIB="-l${PLATFORM_$2}"
fi
fi
if test "${$2_INCDIR}" != "" -a "${$2_INCDIR}" != "/usr/include"; then
$2_INCLUDE="-I${$2_INCDIR}"
fi
PBX_$2=1
AC_DEFINE([HAVE_$2], 1, [$3])
fi
fi
])

90
autoconf/ast_ext_lib.m4 Normal file
View File

@@ -0,0 +1,90 @@
# Helper function to setup variables for a package.
# $1 -> the package name. Used in configure.ac and also as a prefix
# for the variables ($1_DIR, $1_INCLUDE, $1_LIB) in makeopts
# $3 -> option name, used in --with-$3 or --without-$3 when calling configure.
# $2 and $4 are just text describing the package (short and long form)
# AST_EXT_LIB_SETUP([package], [short description], [configure option name], [long description])
AC_DEFUN([AST_EXT_LIB_SETUP],
[
$1_DESCRIP="$2"
$1_OPTION="$3"
AC_ARG_WITH([$3], AC_HELP_STRING([--with-$3=PATH],[use $2 files in PATH $4]),
[
case ${withval} in
n|no)
USE_$1=no
;;
y|ye|yes)
ac_mandatory_list="${ac_mandatory_list} $1"
;;
*)
$1_DIR="${withval}"
ac_mandatory_list="${ac_mandatory_list} $1"
;;
esac
])
PBX_$1=0
AC_SUBST([$1_LIB])
AC_SUBST([$1_INCLUDE])
AC_SUBST([$1_DIR])
AC_SUBST([PBX_$1])
])
# Check for existence of a given package ($1), either looking up a function
# in a library, or, if no function is supplied, only check for the
# existence of the header files.
# AST_EXT_LIB_CHECK([package], [library], [function], [header],
# [extra libs], [extra cflags], [version])
AC_DEFUN([AST_EXT_LIB_CHECK],
[
if test "x${PBX_$1}" != "x1" -a "${USE_$1}" != "no"; then
pbxlibdir=""
# if --with-$1=DIR has been specified, use it.
if test "x${$1_DIR}" != "x"; then
if test -d ${$1_DIR}/lib; then
pbxlibdir="-L${$1_DIR}/lib"
else
pbxlibdir="-L${$1_DIR}"
fi
fi
pbxfuncname="$3"
if test "x${pbxfuncname}" = "x" ; then # empty lib, assume only headers
AST_$1_FOUND=yes
else
AC_CHECK_LIB([$2], [${pbxfuncname}], [AST_$1_FOUND=yes], [AST_$1_FOUND=no], ${pbxlibdir} $5)
fi
# now check for the header.
if test "${AST_$1_FOUND}" = "yes"; then
$1_LIB="${pbxlibdir} -l$2 $5"
# if --with-$1=DIR has been specified, use it.
if test "x${$1_DIR}" != "x"; then
$1_INCLUDE="-I${$1_DIR}/include"
fi
$1_INCLUDE="${$1_INCLUDE} $6"
if test "x$4" = "x" ; then # no header, assume found
$1_HEADER_FOUND="1"
else # check for the header
saved_cppflags="${CPPFLAGS}"
CPPFLAGS="${CPPFLAGS} ${$1_INCLUDE}"
AC_CHECK_HEADER([$4], [$1_HEADER_FOUND=1], [$1_HEADER_FOUND=0])
CPPFLAGS="${saved_cppflags}"
fi
if test "x${$1_HEADER_FOUND}" = "x0" ; then
$1_LIB=""
$1_INCLUDE=""
else
if test "x${pbxfuncname}" = "x" ; then # only checking headers -> no library
$1_LIB=""
fi
PBX_$1=1
# XXX don't know how to evaluate the description (third argument) in AC_DEFINE_UNQUOTED
AC_DEFINE_UNQUOTED([HAVE_$1], 1, [Define this to indicate the ${$1_DESCRIP} library])
AC_DEFINE_UNQUOTED([HAVE_$1_VERSION], [$7], [Define to indicate the ${$1_DESCRIP} library version])
fi
fi
fi
])

View File

@@ -0,0 +1,43 @@
# Check for a package using $2-config. Similar to AST_EXT_LIB_CHECK,
# but use $2-config to determine cflags and libraries to use.
# $3 and $4 can be used to replace --cflags and --libs in the request
# AST_EXT_TOOL_CHECK([package], [tool name], [--cflags], [--libs], [includes], [expression])
AC_DEFUN([AST_EXT_TOOL_CHECK],
[
if test "x${PBX_$1}" != "x1" -a "${USE_$1}" != "no"; then
PBX_$1=0
AC_CHECK_TOOL(CONFIG_$1, $2-config, No)
if test ! "x${CONFIG_$1}" = xNo; then
if test x"$3" = x ; then A=--cflags ; else A="$3" ; fi
$1_INCLUDE=$(${CONFIG_$1} $A)
if test x"$4" = x ; then A=--libs ; else A="$4" ; fi
$1_LIB=$(${CONFIG_$1} $A)
if test x"$5" != x ; then
saved_cppflags="${CPPFLAGS}"
if test "x${$1_DIR}" != "x"; then
$1_INCLUDE="-I${$1_DIR}/include"
fi
CPPFLAGS="${CPPFLAGS} ${$1_INCLUDE}"
saved_libs="${LIBS}"
LIBS="${$1_LIB}"
AC_LINK_IFELSE(
[ AC_LANG_PROGRAM( [ $5 ],
[ $6; ]
)],
[ PBX_$1=1
AC_DEFINE([HAVE_$1], 1, [Define if your system has the $1 headers.])
],
[]
)
CPPFLAGS="${saved_cppflags}"
LIBS="${saved_libs}"
else
PBX_$1=1
AC_DEFINE([HAVE_$1], 1, [Define if your system has the $1 libraries.])
fi
fi
fi
])

61
autoconf/ast_func_fork.m4 Normal file
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@@ -0,0 +1,61 @@
# AST_FUNC_FORK
# -------------
AN_FUNCTION([fork], [AST_FUNC_FORK])
AN_FUNCTION([vfork], [AST_FUNC_FORK])
AC_DEFUN([AST_FUNC_FORK],
[AC_REQUIRE([AC_TYPE_PID_T])dnl
AC_CHECK_HEADERS(vfork.h)
AC_CHECK_FUNCS(fork vfork)
if test "x$ac_cv_func_fork" = xyes; then
_AST_FUNC_FORK
else
ac_cv_func_fork_works=$ac_cv_func_fork
fi
if test "x$ac_cv_func_fork_works" = xcross; then
case $host in
*-*-amigaos* | *-*-msdosdjgpp* | *-*-uclinux* | *-*-linux-uclibc* )
# Override, as these systems have only a dummy fork() stub
ac_cv_func_fork_works=no
;;
*)
ac_cv_func_fork_works=yes
;;
esac
AC_MSG_WARN([result $ac_cv_func_fork_works guessed because of cross compilation])
fi
ac_cv_func_vfork_works=$ac_cv_func_vfork
if test "x$ac_cv_func_vfork" = xyes; then
_AC_FUNC_VFORK
fi;
if test "x$ac_cv_func_fork_works" = xcross; then
ac_cv_func_vfork_works=$ac_cv_func_vfork
AC_MSG_WARN([result $ac_cv_func_vfork_works guessed because of cross compilation])
fi
if test "x$ac_cv_func_vfork_works" = xyes; then
AC_DEFINE(HAVE_WORKING_VFORK, 1, [Define to 1 if `vfork' works.])
else
AC_DEFINE(vfork, fork, [Define as `fork' if `vfork' does not work.])
fi
if test "x$ac_cv_func_fork_works" = xyes; then
AC_DEFINE(HAVE_WORKING_FORK, 1, [Define to 1 if `fork' works.])
PBX_WORKING_FORK=1
AC_SUBST(PBX_WORKING_FORK)
fi
])# AST_FUNC_FORK
# _AST_FUNC_FORK
# -------------
AC_DEFUN([_AST_FUNC_FORK],
[AC_CACHE_CHECK(for working fork, ac_cv_func_fork_works,
[AC_RUN_IFELSE(
[AC_LANG_PROGRAM([AC_INCLUDES_DEFAULT],
[
/* By Ruediger Kuhlmann. */
return fork () < 0;
])],
[ac_cv_func_fork_works=yes],
[ac_cv_func_fork_works=no],
[ac_cv_func_fork_works=cross])])]
)# _AST_FUNC_FORK

View File

@@ -0,0 +1,31 @@
# Helper function to check for gcc attributes.
# AST_GCC_ATTRIBUTE([attribute name], [attribute syntax], [attribute scope])
AC_DEFUN([AST_GCC_ATTRIBUTE],
[
AC_MSG_CHECKING(for compiler 'attribute $1' support)
saved_CFLAGS="$CFLAGS"
CFLAGS="$CFLAGS -Wall -Wno-unused -Werror"
if test "x$2" = "x"
then
AC_COMPILE_IFELSE(
AC_LANG_PROGRAM([$3 void __attribute__(($1)) *test(void *muffin, ...) {return (void *) 0;}],
[]),
AC_MSG_RESULT(yes)
AC_DEFINE_UNQUOTED([HAVE_ATTRIBUTE_$1], 1, [Define to 1 if your GCC C compiler supports the '$1' attribute.]),
AC_MSG_RESULT(no)
)
else
AC_COMPILE_IFELSE(
AC_LANG_PROGRAM([$3 void __attribute__(($2)) *test(void *muffin, ...) {return (void *) 0;}],
[]),
AC_MSG_RESULT(yes)
AC_DEFINE_UNQUOTED([HAVE_ATTRIBUTE_$1], 1, [Define to 1 if your GCC C compiler supports the '$1' attribute.]),
AC_MSG_RESULT(no)
)
fi
CFLAGS="$saved_CFLAGS"
]
)

View File

@@ -0,0 +1,11 @@
# AST_PROG_EGREP
# -------------
m4_ifndef([AST_PROG_EGREP], [AC_DEFUN([AST_PROG_EGREP],
[AC_CACHE_CHECK([for egrep], [ac_cv_prog_egrep],
[if echo a | (grep -E '(a|b)') >/dev/null 2>&1
then ac_cv_prog_egrep='grep -E'
else ac_cv_prog_egrep='egrep'
fi])
EGREP=$ac_cv_prog_egrep
AC_SUBST([EGREP])
])]) # AST_PROG_EGREP

83
autoconf/ast_prog_ld.m4 Normal file
View File

@@ -0,0 +1,83 @@
# AST_PROG_LD
# ----------
# find the pathname to the GNU or non-GNU linker
AC_DEFUN([AST_PROG_LD],
[AC_ARG_WITH([gnu-ld],
[AC_HELP_STRING([--with-gnu-ld],
[assume the C compiler uses GNU ld @<:@default=no@:>@])],
[test "$withval" = no || with_gnu_ld=yes],
[with_gnu_ld=no])
AC_REQUIRE([AST_PROG_SED])dnl
AC_REQUIRE([AC_PROG_CC])dnl
AC_REQUIRE([AC_CANONICAL_HOST])dnl
AC_REQUIRE([AC_CANONICAL_BUILD])dnl
ac_prog=ld
if test "$GCC" = yes; then
# Check if gcc -print-prog-name=ld gives a path.
AC_MSG_CHECKING([for ld used by $CC])
case $host in
*-*-mingw*)
# gcc leaves a trailing carriage return which upsets mingw
ac_prog=`($CC -print-prog-name=ld) 2>&5 | tr -d '\015'` ;;
*)
ac_prog=`($CC -print-prog-name=ld) 2>&5` ;;
esac
case $ac_prog in
# Accept absolute paths.
[[\\/]]* | ?:[[\\/]]*)
re_direlt='/[[^/]][[^/]]*/\.\./'
# Canonicalize the pathname of ld
ac_prog=`echo $ac_prog| $SED 's%\\\\%/%g'`
while echo $ac_prog | grep "$re_direlt" > /dev/null 2>&1; do
ac_prog=`echo $ac_prog| $SED "s%$re_direlt%/%"`
done
test -z "$LD" && LD="$ac_prog"
;;
"")
# If it fails, then pretend we aren't using GCC.
ac_prog=ld
;;
*)
# If it is relative, then search for the first ld in PATH.
with_gnu_ld=unknown
;;
esac
elif test "$with_gnu_ld" = yes; then
AC_MSG_CHECKING([for GNU ld])
else
AC_MSG_CHECKING([for non-GNU ld])
fi
AC_CACHE_VAL(lt_cv_path_LD,
[if test -z "$LD"; then
lt_save_ifs="$IFS"; IFS=$PATH_SEPARATOR
for ac_dir in $PATH; do
IFS="$lt_save_ifs"
test -z "$ac_dir" && ac_dir=.
if test -f "$ac_dir/$ac_prog" || test -f "$ac_dir/$ac_prog$ac_exeext"; then
lt_cv_path_LD="$ac_dir/$ac_prog"
# Check to see if the program is GNU ld. I'd rather use --version,
# but apparently some variants of GNU ld only accept -v.
# Break only if it was the GNU/non-GNU ld that we prefer.
case `"$lt_cv_path_LD" -v 2>&1 </dev/null` in
*GNU* | *'with BFD'*)
test "$with_gnu_ld" != no && break
;;
*)
test "$with_gnu_ld" != yes && break
;;
esac
fi
done
IFS="$lt_save_ifs"
else
lt_cv_path_LD="$LD" # Let the user override the test with a path.
fi])
LD="$lt_cv_path_LD"
if test -n "$LD"; then
AC_MSG_RESULT($LD)
else
AC_MSG_RESULT(no)
fi
test -z "$LD" && AC_MSG_ERROR([no acceptable ld found in \$PATH])
AST_PROG_LD_GNU
])# AST_PROG_LD

View File

@@ -0,0 +1,16 @@
# AST_PROG_LD_GNU
# --------------
AC_DEFUN([AST_PROG_LD_GNU],
[AC_REQUIRE([AST_PROG_EGREP])dnl
AC_CACHE_CHECK([if the linker ($LD) is GNU ld], lt_cv_prog_gnu_ld,
[# I'd rather use --version here, but apparently some GNU lds only accept -v.
case `$LD -v 2>&1 </dev/null` in
*GNU* | *'with BFD'*)
lt_cv_prog_gnu_ld=yes
;;
*)
lt_cv_prog_gnu_ld=no
;;
esac])
with_gnu_ld=$lt_cv_prog_gnu_ld
])# AST_PROG_LD_GNU

21
autoconf/ast_prog_sed.m4 Normal file
View File

@@ -0,0 +1,21 @@
# AST_PROG_SED
# -----------
# Check for a fully functional sed program that truncates
# as few characters as possible. Prefer GNU sed if found.
AC_DEFUN([AST_PROG_SED],
[AC_CACHE_CHECK([for a sed that does not truncate output], ac_cv_path_SED,
[dnl ac_script should not contain more than 99 commands (for HP-UX sed),
dnl but more than about 7000 bytes, to catch a limit in Solaris 8 /usr/ucb/sed.
ac_script=s/aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/bbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbb/
for ac_i in 1 2 3 4 5 6 7; do
ac_script="$ac_script$as_nl$ac_script"
done
echo "$ac_script" | sed 99q >conftest.sed
$as_unset ac_script || ac_script=
ifdef([_AC_PATH_PROGS_FEATURE_CHECK], [_AC_PATH_PROGS_FEATURE_CHECK], [_AC_PATH_PROG_FEATURE_CHECK])(SED, [sed gsed],
[_AC_FEATURE_CHECK_LENGTH([ac_path_SED], [ac_cv_path_SED],
["$ac_path_SED" -f conftest.sed])])])
SED="$ac_cv_path_SED"
AC_SUBST([SED])dnl
rm -f conftest.sed
])# AST_PROG_SED

View File

@@ -40,7 +40,7 @@ check_for_app aclocal${MY_AM_VER}
echo "Generating the configure script ..."
aclocal${MY_AM_VER} 2>/dev/null
aclocal${MY_AM_VER} -I autoconf
autoconf${MY_AC_VER}
autoheader${MY_AC_VER}
automake${MY_AM_VER} --add-missing --copy 2>/dev/null

View File

@@ -0,0 +1,21 @@
<category name="MENUSELECT_CFLAGS" displayname="Compiler Flags - Development">
<member name="MTX_PROFILE" displayname="Enable Code Profiling Using TSC Counters">
</member>
<member name="DEBUG_SCHEDULER" displayname="Enable Scheduler Debugging Output">
</member>
<member name="DEBUG_THREADLOCALS" displayname="Enable Thread-Local-Storage Debugging">
</member>
<member name="DETECT_DEADLOCKS" displayname="Detect Deadlocks">
<depend>DEBUG_THREADS</depend>
</member>
<member name="DUMP_SCHEDULER" displayname="Dump Scheduler Contents for Debugging">
</member>
<member name="TRACE_FRAMES" displayname="Trace Frame Allocations">
</member>
<member name="DO_CRASH" displayname="Crash on fatal errors">
</member>
<member name="THREAD_CRASH" displayname="Crash on mutex errors">
</member>
<member name="CHANNEL_TRACE" displayname="Enable CHANNEL(trace) function">
</member>
</category>

View File

@@ -1,6 +1,10 @@
<category name="MENUSELECT_CFLAGS" displayname="Compiler Flags" positive_output="yes" remove_on_change=".lastclean">
<member name="DONT_OPTIMIZE" displayname="Disable Optimizations by the Compiler">
</member>
<member name="ENABLE_UPLOADS" displayname="Enable HTTP uploads">
<defaultenabled>yes</defaultenabled>
<depend>gmime</depend>
</member>
<member name="DEBUG_THREADS" displayname="Enable Thread Debugging">
</member>
<member name="STATIC_BUILD" displayname="Build static binaries">
@@ -8,11 +12,11 @@
<member name="LOADABLE_MODULES" displayname="Runtime module loading">
<defaultenabled>yes</defaultenabled>
</member>
<member name="DEBUG_FD_LEAKS" displayname="Enable File Descriptor Leak Detection">
</member>
<member name="LOW_MEMORY" displayname="Optimize for Low Memory Usage">
</member>
<member name="LOTS_OF_SPANS" displayname="More than 32 Zaptel spans">
</member>
<member name="MTX_PROFILE" displayname="Enable Code Profiling Using TSC Counters">
<member name="LOTS_OF_SPANS" displayname="More than 32 DAHDI spans">
</member>
<member name="RADIO_RELAX" displayname="Relax DTMF for Radio Applications">
</member>
@@ -20,7 +24,6 @@
<defaultenabled>no</defaultenabled>
</member>
<member name="G711_REDUCED_BRANCHING" displayname="New ulaw/alaw codec, reduced branching (might help it run faster in some architectures)">
<defaultenabled>yes</defaultenabled>
<depend>G711_NEW_ALGORITHM</depend>
</member>
<member name="TEST_CODING_TABLES" displayname="New ulaw/alaw codec, turn on table tests on init">
@@ -31,23 +34,8 @@
</member>
<member name="DEBUG_CHANNEL_LOCKS" displayname="Debug Channel Locking">
</member>
<member name="DEBUG_SCHEDULER" displayname="Enable Scheduler Debugging Output">
</member>
<member name="DEBUG_THREADLOCALS" displayname="Enable Thread-Local-Storage Debugging">
</member>
<member name="DETECT_DEADLOCKS" displayname="Detect Deadlocks">
<depend>DEBUG_THREADS</depend>
</member>
<member name="DUMP_SCHEDULER" displayname="Dump Scheduler Contents for Debugging">
</member>
<member name="MALLOC_DEBUG" displayname="Keep Track of Memory Allocations">
</member>
<member name="TRACE_FRAMES" displayname="Trace Frame Allocations">
</member>
<member name="DO_CRASH" displayname="Crash on fatal errors">
</member>
<member name="THREAD_CRASH" displayname="Crash on mutex errors">
</member>
<member name="BUSYDETECT_TONEONLY" displayname="Enable additional comparision of only the tone duration not the silence part">
<conflict>BUSYDETECT_COMPARE_TONE_AND_SILENCE</conflict>
<defaultenabled>no</defaultenabled>
@@ -59,6 +47,8 @@
<member name="BUSYDETECT_DEBUG" displayname="Enable additional busy detection debugging">
<defaultenabled>no</defaultenabled>
</member>
<member name="CHANNEL_TRACE" displayname="Enable CHANNEL(trace) function">
<member name="IAX_OLD_FIND" displayname="Use the old, slow method of searching for IAX callnos">
</member>
<member name="INTEGER_CALLERID" displayname="Use the (less accurate) integer-based method for decoding FSK tones (for embedded systems)">
</member>
</category>

View File

@@ -5,10 +5,10 @@
<member name="CDR" displayname="Call Detail Recording" remove_on_change="cdr/*.o">
<depend>gnu_ld</depend>
</member>
<member name="CHANNELS" displayname="Channels" remove_on_change="channels/*.o channels/misdn/*.o">
<member name="CHANNELS" displayname="Channels" remove_on_change="channels/*.o channels/*/*.o">
<depend>gnu_ld</depend>
</member>
<member name="CODECS" displayname="Coders/Decoders" remove_on_change="codecs/*.o codecs/gsm/src/*.o codecs/ilbc/*.o codecs/lpc10/*.o codecs/gsm/lib/libgsm.a codecs/lpc10/liblpc10.a codecs/ilbc/libilbc.a codecs/g722/libg722.a">
<member name="CODECS" displayname="Coders/Decoders" remove_on_change="codecs/*.o codecs/*/*.o codecs/*/*/*.o codecs/*/*.a codecs/*/*/*.a">
<depend>gnu_ld</depend>
</member>
<member name="FORMATS" displayname="File Formats" remove_on_change="formats/*.o">
@@ -17,10 +17,13 @@
<member name="FUNCS" displayname="Dialplan Functions" remove_on_change="funcs/*.o">
<depend>gnu_ld</depend>
</member>
<member name="PBX" displayname="PBX Functionality" remove_on_change="pbx/*.o pbx/ael/*.o">
<member name="PBX" displayname="PBX Functionality" remove_on_change="pbx/*.o pbx/*/*.o">
<depend>gnu_ld</depend>
</member>
<member name="RES" displayname="Resource Modules" remove_on_change="res/*.o res/snmp/*.o">
<member name="RES" displayname="Resource Modules" remove_on_change="res/*.o res/*/*.o">
<depend>gnu_ld</depend>
</member>
<member name="TEST" displayname="Test Modules" remove_on_change="tests/*.o">
<depend>gnu_ld</depend>
</member>
</category>

View File

@@ -1,3 +1,3 @@
/\/\*\*\* MAKEOPTS/ {printit=1; next}
/\*\*\*\// {if (printit) exit}
// {if (printit) print}
/.*/ {if (printit) print}

View File

@@ -1,3 +1,3 @@
/\/\*\*\* MODULEINFO/ {printit=1; next}
/\*\*\*\// {if (printit) exit}
// {if (printit) print}
/.*/ {if (printit) print}

View File

@@ -7,31 +7,28 @@ cat << END
*/
END
TMP=`${GREP} MENUSELECT_CFLAGS menuselect.makeopts | sed 's/MENUSELECT_CFLAGS\=//g' | sed 's/-D//g'`
TMP=`${GREP} -e "^MENUSELECT_CFLAGS" menuselect.makeopts | sed 's/MENUSELECT_CFLAGS\=//g' | sed 's/-D//g'`
for x in ${TMP}; do
echo "#define ${x} 1"
if test "x${BUILDOPTS}" != "x" ; then
BUILDOPTS="${BUILDOPTS}, ${x}"
else
BUILDOPTS="${x}"
fi
done
TMP=`${GREP} MENUSELECT_BUILD_DEPS menuselect.makeopts | sed 's/MENUSELECT_BUILD_DEPS\=//g'`
TMP=`${GREP} -e "^MENUSELECT_BUILD_DEPS" menuselect.makeopts | sed 's/MENUSELECT_BUILD_DEPS\=//g'`
for x in ${TMP}; do
x2=`echo ${x} | tr a-z A-Z`
echo "#define AST_MODULE_${x2} 1"
done
if ${GREP} AST_DEVMODE makeopts | ${GREP} -q yes
if ${GREP} "AST_DEVMODE" makeopts | ${GREP} -q "yes"
then
echo "#define AST_DEVMODE 1"
TMP="${TMP} AST_DEVMODE"
fi
case ${OSARCH} in # actually we should check build_os
*BSD|mingw|darwin*)
BUILDSUM=`echo ${TMP} | md5 | cut -c1-32`
;;
SunOS)
BUILDSUM=`echo ${TMP} | digest -a md5 | cut -c1-32`
;;
*)
BUILDSUM=`echo ${TMP} | md5sum | cut -c1-32`
;;
esac
BUILDSUM=`echo ${TMP} | ${MD5} | cut -c1-32`
echo "#define AST_BUILDOPT_SUM \"${BUILDSUM}\""
echo "#define AST_BUILDOPTS \"${BUILDOPTS}\""

View File

@@ -10,7 +10,12 @@ elif [ -d .svn ]; then
REV=`svnversion -c ${1} | cut -d: -f2`
BASE=`LANG=C svn pg svnmerge-integrated ${1} | cut -d: -f1`
INTEGRATED=`LANG=C svn pg automerge-propname ${1}`
if [ -z "${INTEGRATED}" ] ; then
INTEGRATED=svnmerge-integrated
fi
BASE=`LANG=C svn pg ${INTEGRATED} ${1} | cut -d: -f1`
if [ "${PARTS}" = "trunk" ] ; then
echo SVN-trunk-r${REV}
@@ -30,7 +35,7 @@ elif [ -d .svn ]; then
fi
if [ ${BRANCH} != 0 ] ; then
if [ -z ${RESULT} ] ; then
if [ -z "${RESULT}" ] ; then
RESULT="${PART}"
else
RESULT="${RESULT}-${PART}"
@@ -39,7 +44,7 @@ elif [ -d .svn ]; then
fi
if [ ${TEAM} != 0 ] ; then
if [ -z ${RESULT} ] ; then
if [ -z "${RESULT}" ] ; then
RESULT="${PART}"
else
RESULT="${RESULT}-${PART}"

View File

@@ -1,9 +1,9 @@
#!/bin/sh
if [ ! -f ../.flavor ]; then
if [ ! -f .flavor ]; then
EXTRA=""
else
aadkver=`cat ../.version`
aadkflavor=`cat ../.flavor`
aadkver=`cat .version`
aadkflavor=`cat .flavor`
EXTRA=" (${aadkflavor} ${aadkver})"
fi
cat << END

View File

@@ -1,5 +1,5 @@
#!/bin/sh
if [ ! -f ../.flavor ]; then
if [ ! -f .flavor ]; then
cat << END
/*
* version.h
@@ -10,8 +10,8 @@ if [ ! -f ../.flavor ]; then
END
else
aadkver=`cat ../.version`
aadkflavor=`cat ../.flavor`
aadkver=`cat .version`
aadkflavor=`cat .flavor`
cat << END
/*
* version.h

View File

@@ -1,7 +1,9 @@
ASOUND=@PBX_ALSA@
CRYPTO=@PBX_CRYPTO@
CURL=@PBX_CURL@
DAHDI=@PBX_DAHDI@
FREETDS=@PBX_FREETDS@
GMIME=@PBX_GMIME@
GNU_LD=@GNU_LD@
GSM=@PBX_GSM@
GTK2=@PBX_GTK2@
@@ -28,9 +30,12 @@ PGSQL=@PBX_PGSQL@
POPT=@PBX_POPT@
PORTAUDIO=@PBX_PORTAUDIO@
PRI=@PBX_PRI@
RESAMPLE=@PBX_RESAMPLE@
RADIUS=@PBX_RADIUS@
SPANDSP=@PBX_SPANDSP@
SPEEX=@PBX_SPEEX@
SPEEXDSP=@PBX_SPEEXDSP@
SPEEX_PREPROCESS=@PBX_SPEEX_PREPROCESS@
SQLITE3=@PBX_SQLITE3@
SQLITE=@PBX_SQLITE@
SS7=@PBX_SS7@
@@ -42,7 +47,5 @@ USB=@PBX_USB@
VORBIS=@PBX_VORBIS@
VPBAPI=@PBX_VPB@
WINARCH=@WINARCH@
ZAPTEL=@PBX_ZAPTEL@
ZAPTEL_TRANSCODE=@PBX_ZAPTEL_TRANSCODE@
ZAPTEL_VLDTMF=@PBX_ZAPTEL_VLDTMF@
WORKING_FORK=@PBX_WORKING_FORK@
ZLIB=@PBX_ZLIB@

View File

@@ -8,6 +8,16 @@
make -C sounds MENUSELECT_CORE_SOUNDS=CORE-SOUNDS-EN-GSM MENUSELECT_MOH=MOH-FREEPLAY-WAV WGET=wget DOWNLOAD=wget all
make AWK=awk GREP=grep menuselect-tree
make_tex_docs() {
# make backup of asterisk.tex because we are going to alter it
cp asterisk.tex asterisk.tex.orig
sed -e "s/ASTERISKVERSION/${VERSION}/" asterisk.tex > asterisk_local.tex
mv asterisk_local.tex asterisk.tex
rubber --pdf asterisk.tex
latex2html asterisk.tex
# restore backup of asterisk.tex
mv asterisk.tex.orig asterisk.tex
}
VERSION=`cat .version`
sed -i -e "s/ASTERISKVERSION/${VERSION}/" doc/tex/asterisk.tex
cd doc/tex && rubber --pdf asterisk.tex && latex2html asterisk.tex
cd doc/tex && make_tex_docs

View File

@@ -30,7 +30,7 @@ esac
case "${OSARCH}" in
linux-gnu|FreeBSD)
nm ${1} | ${GREP} -e "$TEXTSYM" | cut -d" " -f3 | ${FILTER} > striplist
sed -e "s/^/-N /" striplist | xargs ${STRIP} ${1}
sed -e "s/^/-N /" striplist | xargs -n 40 ${STRIP} ${1}
rm -f striplist
;;
*)

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