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.
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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)
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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
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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
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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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2007-06-25 01:11:47 +00:00
Joshua Colp
18f4920227
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r70552 | file | 2007-06-20 18:22:20 -0400 (Wed, 20 Jun 2007) | 10 lines
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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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2007-06-20 22:24:47 +00:00
Russell Bryant
238b7a54cc
Merged revisions 69944 via svnmerge from
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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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2007-06-19 15:27:16 +00:00
Tilghman Lesher
a67890d7a9
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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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2007-06-18 19:52:56 +00:00
Joshua Colp
9ed0563f17
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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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2007-06-18 19:02:45 +00:00
Joshua Colp
59bc48bd05
Merged revisions 69775 via svnmerge from
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r69775 | file | 2007-06-18 14:18:12 -0400 (Mon, 18 Jun 2007) | 10 lines
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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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2007-06-18 18:19:54 +00:00
Joshua Colp
1dbfbe6d71
Merged revisions 69668 via svnmerge from
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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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2007-06-18 16:06:17 +00:00
Joshua Colp
3e4980da79
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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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2007-06-18 15:48:05 +00:00
Joshua Colp
981a94f023
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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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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.
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(issue #9957 , patches from mvanbaak, caio1982, critch, and dimas)
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2007-06-14 19:39:12 +00:00
Russell Bryant
8e11d6c147
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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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2007-06-13 20:03:03 +00:00
Jason Parker
63535ada60
Fixes for ast_strlen_zero() janitor project.
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Issue 9968, patch by eliel.
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2007-06-13 17:06:53 +00:00
Russell Bryant
156d6338b2
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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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2007-06-13 16:59:42 +00:00
Russell Bryant
1d57ccb6f7
Fix a bunch of doxygen errors and document more things
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(issue #9842 , snuffy)
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2007-06-07 23:07:25 +00:00
Olle Johansson
1f9d98016e
- Doxygen updates
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- Adding docs on flags to be able to clean up a bit
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2007-06-07 19:45:32 +00:00
Joshua Colp
f2cc861bcf
Merged revisions 67941 via svnmerge from
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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
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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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2007-06-07 00:12:21 +00:00
Russell Bryant
6aec360466
Remove our little joke that was making fun of email disclaimers which nobody
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else seemed to think was very funny. Oh well ... :)
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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
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2007-06-06 21:20:11 +00:00
Russell Bryant
033a3df22a
Merged revisions 67862 via svnmerge from
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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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2007-06-06 21:16:18 +00:00
Joshua Colp
e3492b9511
Merged revisions 67068 via svnmerge from
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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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2007-06-04 19:32:08 +00:00
Russell Bryant
93f2be2675
Fix a couple of places where "tos" was used instead of "cos".
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(issue #9540 , patch by IgorG)
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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
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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.
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2007-05-31 19:41:03 +00:00
Joshua Colp
9f79587d47
Merged revisions 66768 via svnmerge from
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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
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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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2007-05-31 16:18:14 +00:00
Olle Johansson
0b2db74e5a
Issue #9842 - Doxygen updates by snuffy. Thanks!
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(Committed from Media Plaza in Utrecht, Netherlands - Open Source VoIP conference)
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2007-05-31 10:26:55 +00:00
Olle Johansson
4c1068c136
oops. Thanks Terry.
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2007-05-29 19:53:40 +00:00
Olle Johansson
ee3a0af16a
Merged revisions 66503 via svnmerge from
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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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2007-05-29 19:35:43 +00:00
Olle Johansson
6d6c525b10
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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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2007-05-29 19:17:49 +00:00
Olle Johansson
0b67a7d80a
Merged revisions 66414 via svnmerge from
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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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2007-05-29 16:19:53 +00:00
Olle Johansson
f4e81d7a54
Merged revisions 66404 via svnmerge from
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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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2007-05-29 16:17:17 +00:00
Olle Johansson
36f15091bb
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r66363 | oej | 2007-05-29 11:41:40 +0200 (Tue, 29 May 2007) | 10 lines
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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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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)
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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)
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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
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places in the code where the same block of code for creating detached threads
was replicated. (patch from bbryant)
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2007-05-24 18:30:19 +00:00
Joshua Colp
202fbe363a
Merged revisions 65839 via svnmerge from
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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
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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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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
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but is still ok... Sorry. Lack of Tea, really.
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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...
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2007-05-24 11:38:20 +00:00
Russell Bryant
d1ba4f90de
- Remove debug variable that was only used in one place
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- convert string handling to the ast_str API
- Convert strdup() to ast_strdup() and check the result
- Minor formatting changes
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2007-05-24 03:28:39 +00:00
Mark Spencer
04e45cfda3
Add SendURL support
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2007-05-24 02:23:08 +00:00
Kevin P. Fleming
5dc23536ec
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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
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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.
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r65123 | oej | 2007-05-18 20:16:09 +0200 (Fri, 18 May 2007) | 10 lines
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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
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r65076 | oej | 2007-05-18 17:18:13 +0200 (Fri, 18 May 2007) | 13 lines
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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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2007-05-18 15:20:39 +00:00
Olle Johansson
491827a28a
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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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2007-05-18 10:41:31 +00:00
Olle Johansson
ac343d43c8
Makeup, darling.
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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
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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
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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
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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
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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