Commit Graph

1825 Commits

Author SHA1 Message Date
Joshua Colp
1961b57705 Add rtpdest option to SIP CHANNEL() dialplan function to return the IP address and port that RTP (be it audio/video/text) is going to.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@71988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-26 23:31:23 +00:00
Joshua Colp
d77301b8cd Tweak CLI command completion and some help text. (issue #10049 reported by IgorG)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@71613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-25 15:35:10 +00:00
Joshua Colp
76455dda03 Merged revisions 71430 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r71430 | file | 2007-06-24 21:10:06 -0400 (Sun, 24 Jun 2007) | 10 lines

Merged revisions 71414 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r71414 | file | 2007-06-24 21:02:49 -0400 (Sun, 24 Jun 2007) | 2 lines

Ignore other URIs after the first in a 300 Multiple Choice response. (issue #10041 reported by homesick)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@71434 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-25 01:11:47 +00:00
Joshua Colp
18f4920227 Merged revisions 70552 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r70552 | file | 2007-06-20 18:22:20 -0400 (Wed, 20 Jun 2007) | 10 lines

Merged revisions 70551 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r70551 | file | 2007-06-20 18:20:16 -0400 (Wed, 20 Jun 2007) | 2 lines

Don't overwrite the configured username setting upon a REGISTER. (issue #8565 reported by jsmith)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-20 22:24:47 +00:00
Russell Bryant
238b7a54cc Merged revisions 69944 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r69944 | russell | 2007-06-19 10:22:36 -0500 (Tue, 19 Jun 2007) | 10 lines

Fix a crash that could occur when handing device state changes.
When the state of a device changes, the device state thread tells the extension
state handling code that it changed.  Then, the extension state code calls the
callback in chan_sip so that it can update subscriptions to that extension.
A pointer to a sip_pvt structure is passed to this function as the call which
needs a NOTIFY sent.  However, there was no locking done to ensure that the pvt
struct didn't disappear during this process.
(issue #9946, reported by tdonahue, patch by me, patch updated to trunk to use
 the sip_pvt lock wrappers by eliel)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69945 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-19 15:27:16 +00:00
Tilghman Lesher
a67890d7a9 Merged revisions 69796 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r69796 | tilghman | 2007-06-18 14:48:17 -0500 (Mon, 18 Jun 2007) | 2 lines

Issue 10005 - Segfault with missing arguments, plus fix a missing define for SIP INFO channels

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69797 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-18 19:52:56 +00:00
Joshua Colp
9ed0563f17 Merged revisions 69794 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r69794 | file | 2007-06-18 15:00:50 -0400 (Mon, 18 Jun 2007) | 2 lines

Don't count RTP timeout when involved in a T38 fax session. (issue #9222 reported by ivoc)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69795 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-18 19:02:45 +00:00
Joshua Colp
59bc48bd05 Merged revisions 69775 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r69775 | file | 2007-06-18 14:18:12 -0400 (Mon, 18 Jun 2007) | 10 lines

Merged revisions 69765 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r69765 | file | 2007-06-18 14:13:03 -0400 (Mon, 18 Jun 2007) | 2 lines

Set the peer name on the dialog to the one configured in sip.conf and NOT the username to be used for authentication attempts. (issue #9967 reported by achauvin)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69779 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-18 18:19:54 +00:00
Joshua Colp
1dbfbe6d71 Merged revisions 69668 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r69668 | file | 2007-06-18 12:04:55 -0400 (Mon, 18 Jun 2007) | 2 lines

Don't defer the BYE till later on a transfer when the transfer itself goes kaboom and has no hope of working.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69672 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-18 16:06:17 +00:00
Joshua Colp
3e4980da79 Merged revisions 69661 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r69661 | file | 2007-06-18 11:46:32 -0400 (Mon, 18 Jun 2007) | 2 lines

Few minor transfer tweaks. We can't unlock something we never locked, and better handle a specific scenario with doing an attended transfer between two non-bridged calls.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69663 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-18 15:48:05 +00:00
Joshua Colp
981a94f023 Merged revisions 69625 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r69625 | file | 2007-06-18 09:55:00 -0400 (Mon, 18 Jun 2007) | 2 lines

Fix issue where it would be possible for the negotiated codecs to get set back to nothing. (issue #9992 reported by yehavi)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69626 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-18 13:57:33 +00:00
Russell Bryant
055d82cbce Add a massive set of changes for converting to use the ast_debug() macro.
(issue #9957, patches from mvanbaak, caio1982, critch, and dimas)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-14 19:39:12 +00:00
Russell Bryant
8e11d6c147 Merged revisions 69183 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r69183 | russell | 2007-06-13 14:57:38 -0500 (Wed, 13 Jun 2007) | 9 lines

Move the logic for destroying a call when no response is received to a BYE
outside of the block that checks for FLAG_FATAL to be set.  This flag is only
set when the packet is transmitted with the reliability set to XMIT_CRITICAL
when the original packet is transmitted.  A BYE is always sent with it set
to XMIT_RELIABLE, meaning this code could never be encountered.  This resulted
in seeing some SIP channels that would never go away with the last packet
sent being a BYE.
(part of issue #9235, patch from jcmoore)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69187 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-13 20:03:03 +00:00
Jason Parker
63535ada60 Fixes for ast_strlen_zero() janitor project.
Issue 9968, patch by eliel.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69081 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-13 17:06:53 +00:00
Russell Bryant
156d6338b2 Merged revisions 69071 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r69071 | russell | 2007-06-13 11:56:16 -0500 (Wed, 13 Jun 2007) | 3 lines

Clarify a bit of logic.  This doesn't change behavior in any way, but it is
helpful when following the logic to debug problems like 9235.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-13 16:59:42 +00:00
Russell Bryant
1d57ccb6f7 Fix a bunch of doxygen errors and document more things
(issue #9842, snuffy)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68339 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-07 23:07:25 +00:00
Olle Johansson
1f9d98016e - Doxygen updates
- Adding docs on flags to be able to clean up a bit


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68195 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-07 19:45:32 +00:00
Joshua Colp
f2cc861bcf Merged revisions 67941 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r67941 | file | 2007-06-06 20:10:48 -0400 (Wed, 06 Jun 2007) | 10 lines

Merged revisions 67938 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r67938 | file | 2007-06-06 20:09:13 -0400 (Wed, 06 Jun 2007) | 2 lines

Only notify the devicestate system of a peer state change when the peer is built from the config file. (issue #9900 reported by arkadia)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67944 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-07 00:12:21 +00:00
Russell Bryant
6aec360466 Remove our little joke that was making fun of email disclaimers which nobody
else seemed to think was very funny.  Oh well ... :)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67895 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-06 22:27:18 +00:00
Tilghman Lesher
9d05ff8ed5 Issue 9869 - replace malloc and memset with ast_calloc, and other coding guidelines changes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67864 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-06 21:20:11 +00:00
Russell Bryant
033a3df22a Merged revisions 67862 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r67862 | russell | 2007-06-06 16:14:46 -0500 (Wed, 06 Jun 2007) | 4 lines

Fix a crash when doing call pickups with SIP phones.  The code unlocked the
channel when it should not have.
(issue #9652, reported by corruptor, fixed by me)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67863 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-06 21:16:18 +00:00
Joshua Colp
e3492b9511 Merged revisions 67068 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r67068 | file | 2007-06-04 15:31:09 -0400 (Mon, 04 Jun 2007) | 2 lines

Better handle SIP devices that say they have SDP content... but really don't. (issue #9398 reported by mthomasslo)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67069 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-04 19:32:08 +00:00
Russell Bryant
93f2be2675 Fix a couple of places where "tos" was used instead of "cos".
(issue #9540, patch by IgorG)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67016 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-04 15:14:52 +00:00
Russell Bryant
3d2b58751f To satisfy some legal concerns, add an option for chan_sip to include a
disclaimer along with SIP messages in the header, X-Disclaimer.  This is off by
default.  Also, the text of the disclaimer can be customized in sip.conf.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66777 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-31 19:41:03 +00:00
Joshua Colp
9f79587d47 Merged revisions 66768 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r66768 | file | 2007-05-31 12:14:48 -0400 (Thu, 31 May 2007) | 10 lines

Merged revisions 66764 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r66764 | file | 2007-05-31 12:12:39 -0400 (Thu, 31 May 2007) | 2 lines

It is now possible for this path of execution to have the frame pointer be NULL, therefore we need to check for it before trying to access it. (issue #9836 reported by barthpbx)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-31 16:18:14 +00:00
Olle Johansson
0b2db74e5a Issue #9842 - Doxygen updates by snuffy. Thanks!
(Committed from Media Plaza in Utrecht, Netherlands - Open Source VoIP conference)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-31 10:26:55 +00:00
Olle Johansson
4c1068c136 oops. Thanks Terry.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66505 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-29 19:53:40 +00:00
Olle Johansson
ee3a0af16a Merged revisions 66503 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r66503 | oej | 2007-05-29 21:32:57 +0200 (Tue, 29 May 2007) | 2 lines

Properly handle 408 request timeout - according to the RFC, the dialog dies if a request in a dialog gets this response.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66504 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-29 19:35:43 +00:00
Olle Johansson
6d6c525b10 Merged revisions 66474 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r66474 | oej | 2007-05-29 21:02:04 +0200 (Tue, 29 May 2007) | 2 lines

Don't issue hangup on hangup on hangup on hangup (for jcmoore)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66502 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-29 19:17:49 +00:00
Olle Johansson
0b67a7d80a Merged revisions 66414 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r66414 | oej | 2007-05-29 18:07:44 +0200 (Tue, 29 May 2007) | 2 lines

Don't reset hangupcause if we already have one

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66433 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-29 16:19:53 +00:00
Olle Johansson
f4e81d7a54 Merged revisions 66404 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r66404 | oej | 2007-05-29 18:02:50 +0200 (Tue, 29 May 2007) | 2 lines

Tracking down hanging channels, killing them one by one. Issue #9235 and related

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66432 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-29 16:17:17 +00:00
Olle Johansson
36f15091bb Merged revisions 66363 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r66363 | oej | 2007-05-29 11:41:40 +0200 (Tue, 29 May 2007) | 10 lines

Merged revisions 66349 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r66349 | oej | 2007-05-29 09:53:14 +0200 (Tue, 29 May 2007) | 2 lines

Issue #9802 - Change inuse counter on CANCEL

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66367 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-29 10:02:31 +00:00
Joshua Colp
4d03b4f268 Don't try to unregister a peer using the sip unregister CLI command if they are not registered. (issue #9811 reported by eliel)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66315 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-28 23:28:52 +00:00
Joshua Colp
39e9b3112c Due to the way stringfields work the value of the url pointer will always be non-NULL so we have to use ast_strlen_zero to make sure it is not empty. (issue #9821 reported by pj)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-28 23:24:04 +00:00
Russell Bryant
4b3a3fb14c Add a new API call for creating detached threads. Then, go replace all of the
places in the code where the same block of code for creating detached threads
was replicated.  (patch from bbryant)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65968 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-24 18:30:19 +00:00
Joshua Colp
202fbe363a Merged revisions 65839 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r65839 | file | 2007-05-24 10:42:12 -0400 (Thu, 24 May 2007) | 10 lines

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/trunk@65840 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-24 14:43:49 +00:00
Olle Johansson
9b95428cce Issue #8409 and accidentally a fix to chan_sip that wasn't supposed to be there
but is still ok... Sorry. Lack of Tea, really.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-24 14:41:43 +00:00
Kevin P. Fleming
0ec502099f Yes Virginia, there is a reason why we have stringfields in the sip_pvt structure...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65814 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-24 11:38:20 +00:00
Russell Bryant
d1ba4f90de - Remove debug variable that was only used in one place
- convert string handling to the ast_str API
 - Convert strdup() to ast_strdup() and check the result
 - Minor formatting changes


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-24 03:28:39 +00:00
Mark Spencer
04e45cfda3 Add SendURL support
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-24 02:23:08 +00:00
Kevin P. Fleming
5dc23536ec Merged revisions 65683 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r65683 | kpfleming | 2007-05-23 16:51:56 -0400 (Wed, 23 May 2007) | 10 lines

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 20:53:10 +00:00
Olle Johansson
a3f9350ec2 Related to issue #9235 btw.
Merged revisions 65123 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r65123 | oej | 2007-05-18 20:16:09 +0200 (Fri, 18 May 2007) | 10 lines

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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2007-05-18 18:18:59 +00:00
Olle Johansson
1bc7fdeb6b Merged revisions 65076 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r65076 | oej | 2007-05-18 17:18:13 +0200 (Fri, 18 May 2007) | 13 lines

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/trunk@65077 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18 15:20:39 +00:00
Olle Johansson
491827a28a Merged revisions 64974 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r64974 | oej | 2007-05-18 12:37:44 +0200 (Fri, 18 May 2007) | 2 lines

Issue 9487 - stop media flows at hangup of call

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64975 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18 10:41:31 +00:00
Olle Johansson
ac343d43c8 Makeup, darling.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64973 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18 10:28:56 +00:00
Olle Johansson
f3ec447a23 Another fix for the support for recordings controlled by INFO-packets
We still lack a setting to enable/disable this per peer


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64903 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18 08:49:34 +00:00
Joshua Colp
a769766c53 Merged revisions 64754 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r64754 | file | 2007-05-17 12:10:12 -0400 (Thu, 17 May 2007) | 2 lines

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)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-17 16:11:26 +00:00
Olle Johansson
d83dcae6b1 Below patches with some re-structuring for trunk
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Merged revisions 64602 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r64602 | oej | 2007-05-16 12:38:18 +0200 (Wed, 16 May 2007) | 2 lines

Issue #9681 - Handle www-auth on BYE

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64604 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-16 10:58:22 +00:00
Olle Johansson
c472b899ef Merged revisions 64578 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r64578 | oej | 2007-05-16 12:05:47 +0200 (Wed, 16 May 2007) | 2 lines

Final part of issue #9483 - fixing transfer() of sip calls in the dial plan (twilson)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64587 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-16 10:09:42 +00:00
Olle Johansson
c352f7b0d5 Merged revisions 64543 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r64543 | oej | 2007-05-16 11:12:34 +0200 (Wed, 16 May 2007) | 10 lines

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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64570 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-16 10:02:06 +00:00