Commit Graph

5523 Commits

Author SHA1 Message Date
Russell Bryant
767fa7a909 Change a verbose message to debug.
(closes issue #12514)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114604 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-24 14:55:21 +00:00
Russell Bryant
0fa42f819a Merged revisions 114587 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114588 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-23 17:18:29 +00:00
Olle Johansson
2958831a97 Merged revisions 114584 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114585 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-23 16:53:34 +00:00
Tilghman Lesher
b170c36350 Merged revisions 114571 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114572 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-22 23:58:19 +00:00
Russell Bryant
fe8b7f31db Merged revisions 114558 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-22 22:17:31 +00:00
Russell Bryant
580cb27eec Merged revisions 114537 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114538 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-22 18:04:39 +00:00
Joshua Colp
1e066813ac 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)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114529 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-22 15:54:06 +00:00
Steve Murphy
161b4abd79 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.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114520 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-22 14:38:46 +00:00
Jeff Peeler
41fd7a6a21 (closes issue #6113)
Reported by: oej
Tested by: jpeeler

This patch implements multiple parking lots for parked calls. The default parkinglot is used by default, however setting the channel variable PARKINGLOT in the dialplan will allow use of any other configured parkinglot. See configs/features.conf.sample for more details on setting up another non-default parkinglot. Also, one can (currently) set the default parkinglot to use in the driver configuration file via the parkinglot option.

Patch initially written by oej, brought up to date and finalized by mvanbaak, and then stabilized and converted to astobj2 by me.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114487 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-21 23:42:45 +00:00
Matthew Fredrickson
b43364eac8 Add support for generic name transmission (#12484) on SS7 in chan_zap
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114389 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-21 18:44:35 +00:00
Joshua Colp
a79214b5b1 Merged revisions 114322 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114323 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-21 14:40:33 +00:00
Matthew Fredrickson
264cc6ff5a SS7:Added - Generic Name / Access Transport / Redirecting Number handling. #12425
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114303 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-19 16:58:24 +00:00
Joshua Colp
b05e17fdd7 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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114271 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-18 19:35:33 +00:00
Mark Michelson
0e821d7201 Merged revisions 114257 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-18 18:03:06 +00:00
Sean Bright
e4dce85331 Merged revisions 114245 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114246 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-18 13:38:07 +00:00
Steve Murphy
5203c664de Thanks to snuff for finding these omissions
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-17 14:45:16 +00:00
Steve Murphy
5fb4b1bbe5 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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-16 23:53:27 +00:00
Kevin P. Fleming
a51fb142f9 Merged revisions 114184 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114185 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-16 20:47:30 +00:00
Olle Johansson
18866623dc Merged revisions 114148 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114151 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-15 20:39:29 +00:00
Olle Johansson
f239f24580 Adding chanvar to SIPPEER from 1.4 branch
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114150 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-15 20:31:08 +00:00
Jason Parker
b9bb0749d1 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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114141 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-15 17:21:58 +00:00
Jason Parker
6e6d6f2e10 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-14 18:34:17 +00:00
Joshua Colp
6fad8249f5 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!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-14 15:36:02 +00:00
Joshua Colp
c5d0ca23f0 Merged revisions 114103 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114104 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-14 14:53:33 +00:00
Matthew Fredrickson
5110d3bc69 Make sure linkset is locked exiting ss7_start_call
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114093 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-12 16:21:29 +00:00
Matthew Fredrickson
f9960bc748 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/trunk@114092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-12 16:13:25 +00:00
Terry Wilson
4bc75c9a55 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-11 22:48:52 +00:00
Joshua Colp
a08c4b2064 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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114049 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-10 20:28:40 +00:00
Mark Michelson
d13b45564b Merged revisions 114045 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-10 19:58:36 +00:00
Joshua Colp
4a21c5dd22 Fix spelling of existent in a few places.
(closes issue #12409)
Reported by: candlerb


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114024 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-10 13:45:45 +00:00
Joshua Colp
a4e73acaf8 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114022 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-10 13:28:30 +00:00
Mark Michelson
88cc98ea94 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113928 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-09 20:56:14 +00:00
Joshua Colp
0351ef6e6e Enable enough RTP bridging to allow P2P to work.
(closes issue #11901)
Reported by: pj


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113840 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-09 18:05:40 +00:00
Jason Parker
d314fd5336 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)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-09 17:41:09 +00:00
Joshua Colp
230d9d1465 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-09 16:52:04 +00:00
Mark Michelson
925924386a 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113682 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-09 14:41:58 +00:00
Terry Wilson
3ee1602b6a 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113597 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-09 01:36:58 +00:00
Jason Parker
b52ec53da7 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113505 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-08 18:49:21 +00:00
Jason Parker
f469ee8cf2 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113455 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-08 18:08:35 +00:00
Tilghman Lesher
fa875c0578 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-08 15:48:58 +00:00
Jeff Peeler
bb13bf705e Merged revisions 113013 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113241 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-07 21:35:48 +00:00
Jason Parker
63f574ceb4 Merged revisions 113118 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113119 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-07 18:02:51 +00:00
Jeff Peeler
566e073606 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113013 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-07 15:18:10 +00:00
Steve Murphy
f291c2af0a 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...




git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112874 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-05 01:33:13 +00:00
Philippe Sultan
71dc6a4771 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112821 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-04 19:28:49 +00:00
Philippe Sultan
db884798db 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-04 17:32:46 +00:00
Mark Michelson
3fd8236d28 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112600 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-03 14:35:47 +00:00
Tilghman Lesher
cbf80c1a3c 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)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112520 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-02 19:34:52 +00:00
Joshua Colp
b5cccfe1a4 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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112431 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-02 15:26:51 +00:00
Jeff Peeler
6699761f80 Added dnsmgr status output for sip show registry.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112360 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-01 22:55:28 +00:00