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Author SHA1 Message Date
Russell Bryant
f32662deb4 Fix the calculation of the RTT for RTCP. The previous code would result in 
oscillating and incorrect data.  Additionally, the RTT would sometimes report
negative values due to incorrect calculations.
(issue #9601, patch from davetroy)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65842 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-24 14:49:05 +00:00
Olle Johansson
e4f9cc15e6 Issue #8536 - Caller ID not set in CDR for jingle 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65841 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-24 14:48:55 +00:00
Joshua Colp
0df2a42f96 Merged revisions 65837 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r65837 | file | 2007-05-24 10:40:38 -0400 (Thu, 24 May 2007) | 2 lines

Allow RFC2833 to be negotiated when an INVITE comes in without SDP and is not matched to a user or peer. (issue #9546 reported by mcrawford)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65839 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-24 14:42:12 +00:00
Olle Johansson
6cfe6a550e Issue 8409 - phsultan - Fix "login" as component to jabber server. 
...and, by accident, fix a bug in chan_sip for stopping a loop on retransmits
   of BYE requests.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65836 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-24 14:38:30 +00:00
Christian Richter
9f54cd55af Merged revisions 65767 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r65767 | crichter | 2007-05-24 11:19:58 +0200 (Do, 24 Mai 2007) | 1 line

we should only activate the generator in chan_misdn, when asterisk hask not yet taken the call (WAITING4DIGS state). Alerting audio will be generated fomr asterisk for example.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65768 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-24 09:37:32 +00:00
Kevin P. Fleming
e1518f42ae start the delayed PBX when receive voice or video full frames as well, and comment this delayed-PBX activity 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65685 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-23 20:59:19 +00:00
Kevin P. Fleming
ca6b421be4 Merged revisions 65682 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r65682 | kpfleming | 2007-05-23 16:46:22 -0400 (Wed, 23 May 2007) | 2 lines

ensure that variables are set on a newly created channel before we start a PBX on it

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 2007-05-23 20:51:56 +00:00
Kevin P. Fleming
b89faf596b clear the 'delay PBX' flag when we are ready to start the PBX 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65680 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-23 20:35:50 +00:00
Kevin P. Fleming
09dc4253d8 don't start a PBX on a new incoming IAX2 channel until we have some sort of response to our ACCEPT (ACK or anything else) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65679 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-23 20:30:24 +00:00
Kevin P. Fleming
f608d90283 Merged revisions 65676 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r65676 | kpfleming | 2007-05-23 16:06:13 -0400 (Wed, 23 May 2007) | 2 lines

if we are going to set variables on a newly created channel, it should be done *before* we start the PBX on it

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65677 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-23 20:07:59 +00:00
Russell Bryant
7398856cea Merged revisions 65588 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r65588 | russell | 2007-05-23 08:06:17 -0500 (Wed, 23 May 2007) | 3 lines

Revert revision 62417 as someone reported problems with it to Mark.  This was
related to issue #9588.

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 2007-05-23 13:07:13 +00:00
Kevin P. Fleming
d0be58c8e1 when building a version string for a developer branch, include the base branch in the version string 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-22 20:25:41 +00:00
Russell Bryant
c556cc222c List res_smdi as a dependency for app_voicemail and chan_zap 
(Thanks to mnicholson for pointing it out)


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 2007-05-22 18:40:38 +00:00
Joshua Colp
aae1e3e377 Remove a double const. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-22 15:04:46 +00:00
BJ Weschke
288d51d94f Fix a problem with flag recognition. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65408 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-22 14:02:56 +00:00
Russell Bryant
ece55e3de7 Merged revisions 65389 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r65389 | russell | 2007-05-22 08:07:03 -0500 (Tue, 22 May 2007) | 4 lines

Fix a memory leak that I just noticed in the device state handling in app_queue.
On most device state changes, it would leak roughly 8 to 64 bytes (the length of
the name of the device).

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65394 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-22 13:09:34 +00:00
Christian Richter
e7355ec53b Merged revisions 65328 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r65328 | crichter | 2007-05-22 09:46:39 +0200 (Di, 22 Mai 2007) | 1 line

we stop the tones only when we're in the pre-call phase, otherwise e.g. when in CONNECTED state we should not stop tones when we receive an Information Message
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65342 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-22 08:12:20 +00:00
Joshua Colp
0a992b39de res_agi needs to export two symbols (ast_agi_register and ast_agi_unregister) for usage by others. (issue #9755 reported by mnicholson) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65250 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-20 17:59:58 +00:00
Steve Murphy
2419b6cfb8 Ugh. The svnmerge didn't catch the shift from cdr.c to main/cdr.c, and neither did I. This is the remainder of the 9717 patch, the fix for the run-away FAIL status for a call 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-18 22:26:51 +00:00
Steve Murphy
dfee354cfa Merged revisions 65172 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1 line

This update will fix the situation that occurs as described by 9717, where when several targets are specified for a dial, if any one them reports FAIL, the whole call gets FAIL, even though others were ringing OK. I rearranged the priorities, so that a new disposition, NULL, is at the lowest level, and the disposition get init'd to NULL. Then, next up is FAIL, and next up is BUSY, then NOANSWER, then ANSWERED. All the related set routines will only do so if the disposition value to be set to is greater than what's already there. This gives the intended effect. So, if all the targets are busy, you'd get BUSY for the call disposition. If all get BUSY, but one, and that one rings is not answered, you get NOANSWER. If by some freak of nature, the NULL value doesn't get overridden, then the disp2str routine will report NOANSWER as before.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-18 22:06:27 +00:00
Olle Johansson
4483fa12e8 Merged revisions 65122 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r65122 | oej | 2007-05-18 20:10:46 +0200 (Fri, 18 May 2007) | 2 lines

Not getting an ACK to a 200 OK in the initial invite is critical to the call.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65123 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-18 18:16:09 +00:00
Olle Johansson
7fe3608300 Merged revisions 65075 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r65075 | oej | 2007-05-18 17:12:09 +0200 (Fri, 18 May 2007) | 5 lines

Issue 9235 - part of the problem, maybe not all. Please retry with this patch (and no
other patch) if you have problems with hanging SIP channels. Thank you.

A special Thank You to WeBRainstorm that gave me access to his system.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65076 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-18 15:18:13 +00:00
Olle Johansson
50f79ba4b2 - Adding support for putting calls OFF hold with a re-invite with blank SDP. This was a bug found while doing tests at SIPit in Antwerp. 
- In order to not duplicate code, I restructured some of the code for putting calls on/off hold.

Thanks DEA for reminding me. This fix has been asleep in the videocaps branch until now.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65041 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-18 12:58:39 +00:00
Christian Richter
012fe116fa Merged revisions 65007 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r65007 | crichter | 2007-05-18 13:23:11 +0200 (Fr, 18 Mai 2007) | 1 line

fixed a warning regarding Keypad encoding. encode the IE sending_complete at the right position.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65039 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-18 12:40:46 +00:00
Olle Johansson
73d0ba053b Issue 9487 - stop media flows at hangup of call 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64974 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-18 10:37:44 +00:00
Christian Richter
d682a74e26 Merged revisions 64902 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r64902 | crichter | 2007-05-18 10:24:08 +0200 (Fr, 18 Mai 2007) | 1 line

we *need* to send a PROCEEDING when sending_complete is set, even if need_more_infos is requested.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64904 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-18 08:58:51 +00:00
Russell Bryant
88f6591b24 Fix a small bug I noticed while working on something else. app_queue did not 
unregister its device state monitoring callback in unload_module().  So, this
would make Asterisk crash on the first device state change after you 
unload the module.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64868 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-18 02:48:51 +00:00
Tilghman Lesher
002214d84f Merged revisions 64819 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r64819 | tilghman | 2007-05-17 16:14:36 -0500 (Thu, 17 May 2007) | 2 lines

How is it that we never caught that this is returning the opposite of our documentation, until now?

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64820 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-17 21:19:34 +00:00
Jason Parker
954f082462 Merged revisions 64758 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r64758 | qwell | 2007-05-17 11:52:38 -0500 (Thu, 17 May 2007) | 4 lines

If we have a negative current message, we shouldn't go back even further...

Issue 9727.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64761 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-17 16:53:27 +00:00
Russell Bryant
c3d7124e49 Remove script that is no longer functional since the build system was redone. 
(issue #9340, reported by junky)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64759 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-17 16:52:53 +00:00
Russell Bryant
78989c792f Increase the size of a buffer to support longer dial strings for channels. 
(issue #9291, reported and fix suggested by meni)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64756 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-17 16:47:29 +00:00
Joshua Colp
7a8ca54257 Even more direct RTP setup fixes! Don't allow a codec that isn't supported to creep into the SDP of either side. (issue #9446 reported by marcelbarbulescu) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64754 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-17 16:10:12 +00:00
Joshua Colp
5e41785d73 Fix authuser support. (issue #9740 reported by xmarksthespot) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64720 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-17 13:48:44 +00:00
Russell Bryant
b1f6d367e8 Update the main README to reflect the new build process for 1.4 and above. 
(issue #9725, patch by eliel)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64686 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-17 06:13:53 +00:00
Olle Johansson
2a7e311efe Blocking patch already in this code 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64609 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-16 11:01:11 +00:00
Olle Johansson
4ae20ba8e4 Fix auth on BYE. (Different patch than for 1.2) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64605 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-16 10:59:28 +00:00
Olle Johansson
5acc63d688 Issue #9681 - Handle www-auth on BYE 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64602 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-16 10:38:18 +00:00
Olle Johansson
374b52f717 Final part of issue #9483 - fixing transfer() of sip calls in the dial plan (twilson) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64578 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-16 10:05:47 +00:00
Olle Johansson
70ad556544 Issue #9439 - properly handle username parameters in SIP uri. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-16 09:57:22 +00:00
Olle Johansson
bf1a15b9bf Merged revisions 64535 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r64535 | oej | 2007-05-16 11:08:22 +0200 (Wed, 16 May 2007) | 2 lines

Support SIP uri's starting with SIP: and sip: (reported by Tony Mountfield on the mailing list. Thanks!)

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 2007-05-16 09:12:34 +00:00
Olle Johansson
56af259505 Merged following patch with a lot of changes for 1.4 
------

Merged revisions 64514 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r64514 | oej | 2007-05-16 10:25:56 +0200 (Wed, 16 May 2007) | 6 lines

Issue #9726 - rlister - Better logging for ACL denials

While at it, also added better logging and handling of peers that are not supposed to register.

My patch, stole the issue report from Russell. My apologies, Russell :-)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64516 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-16 08:46:18 +00:00
Christian Richter
d17174cfc1 Merged revisions 64513 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r64513 | crichter | 2007-05-16 10:23:42 +0200 (Mi, 16 Mai 2007) | 1 line

in the case immediate=yes, we directly jump into the dialplan, where people can use PlayTones to indicate a Dialtone, so we don't need to to that by ourself. also we should not do a dialtone_indicate for incoming calls on a TE port in overlapdialmode.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64515 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-16 08:44:51 +00:00
Russell Bryant
69cd2cb213 Properly fix a problem that occurs when you set PARKINGEXTEN to an exten where 
a call is already parked.  (issue #9723, patch by me)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64426 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-15 19:52:18 +00:00
Russell Bryant
39ebbfe69d When someone requests a specific parking space using the PARKINGEXTEN variable, 
ensure that no other caller is already there.
(issue #9723, reported by mdu113, patch by me)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64353 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-14 21:16:39 +00:00
Olle Johansson
1f2afa0ff1 Change -2 to XMIT_ERROR to clarify a bit more 
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 2007-05-14 19:26:50 +00:00
Russell Bryant
b340fcd7d0 Properly handle AST_CONTROL_PROGRESS by just ignoring it. An unknown indication 
will trigger an error and cause sounds to stop, which in this case, is ringing.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64306 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-14 19:13:00 +00:00
Olle Johansson
e041d57175 Handle network errors, like host or network unreachable, in a better way. This means that 
calls to hosts or qualify (OPTION) messages will fail quicker if the TCP/IP stack tells us
that there is an issue.

Since this is an unconnected UDP socket, we will not get error messages directly
in most cases, but maybe on the second and third try.

This is already implemented in trunk.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64280 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-14 18:52:09 +00:00
Joshua Colp
e694a86517 Properly set datalen field when doing PLC in codec_speex. (issue #9722 reported by mihai) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64278 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-14 18:48:33 +00:00
Joshua Colp
656739534c Merged revisions 64275 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r64275 | file | 2007-05-14 14:34:06 -0400 (Mon, 14 May 2007) | 2 lines

Only perform stripping of - strings from the channel name for Zap channels. Anywhere else we might remove a legitimate part of a device name. (issue #9668 reported by stevedavies)

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 2007-05-14 18:36:34 +00:00
Joshua Colp
aff63fba76 Fix scenario where if a phone that simply called Echo() put itself on hold it could never get off hold. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64240 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-14 17:23:51 +00:00