Russell Bryant
e7cb53e8b2
Fix loading persistent queue members when using realtime configuration for queues.
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Also, remove an unneeded leading slash for the astdb family.
(issue #9911 , patch by atis)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@68280 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Jason Parker
51f1d28d2d
Fix an issue with newer phones which require packets be padded out to the correct length.
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Issue 9887, patch by DEA.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@68249 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
2d491b0eae
Disable chan_features by default in menuselect
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@68196 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
b58a48d672
Merged revisions 67938 via svnmerge from
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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/branches/1.4@67941 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Russell Bryant
6595debbc5
Fix a crash when doing call pickups with SIP phones. The code unlocked the
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channel when it should not have.
(issue #9652 , reported by corruptor, fixed by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67862 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Russell Bryant
83744c244a
Suppress a bunch of debug output unless option_debug is on
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67457 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Jason Parker
4a7ae2ebde
Correctly update date/time on devices throughout the life of the device, instead of just at registration.
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Issue 9152, yet another patch by DEA.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67421 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Christian Richter
619cdb016a
Merged revisions 67307 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r67307 | crichter | 2007-06-05 17:42:03 +0200 (Di, 05 Jun 2007) | 1 line
briding is a bool, fixed copy and paste issue.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67334 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Christian Richter
b0dd31cb52
Merged revisions 67306 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r67306 | crichter | 2007-06-05 17:39:43 +0200 (Di, 05 Jun 2007) | 1 line
simplified the EVENT_SETUP handling in the cb_events function a lot. Commented the different possibilities a bit and made functions of shared code. When the dialed extension does not exist in the extensions.conf we'll jump into the 'i' extension if this does exist, else we disconnect the call with the cause:1 = No Route to Destination.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
24922406f4
Only muck with the thread structure if an idle one was found/created.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Kevin P. Fleming
5a5d549b1d
ensure that a burst of full frames (AST_FRAME_DTMF being the prime example) will not be processed out of order... this is a brute force fix, but seems to be the safest fix for now (thanks to the Digium PQ department for finding this bug)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Christian Richter
9f749257ef
Merged revisions 67209 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r67209 | crichter | 2007-06-05 12:05:45 +0200 (Di, 05 Jun 2007) | 1 line
added possibility to deactivate bridging per port
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67210 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Russell Bryant
3b5bd316ed
Fix up a bunch of places where the iax2 pvt structure can disappear and the
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code did not account for it and crashes.
(issues #9642 , #9569 , #9666 , probably others ... based on the work by
stevedavies and mihai, with additional changes from me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Jason Parker
0b7607ad55
Fix for skinny keepalives.
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If there is no traffic from the phone for (keep_alive * 1100) ms (arbitrarily
adding 10% for network issues, etc), unregister the device.
Issue 8394, patch by DEA.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67156 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Jason Parker
6b150d7b9c
Fixes for dtmf/dialing with mgcp (similar to the recent fix for chan_skinny)
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Issue 9855, patch by DEA.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Russell Bryant
885d846739
Add comments for two functions that get called with the appropriate call locked,
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but perform operations that could result in the pvt structure getting destroyed
before returning again, causing numerous seg faults all over the module.
(inspired by issues #9642 , #9569 , and #9666 , and the work done by stevedavies
and mihai)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67119 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
01456184f9
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/branches/1.4@67068 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Russell Bryant
0b7719e9a1
Resolve a deadlock in chan_iax2. When handling an implicit ACK to a frame that
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was marked as the final transmission for a call, don't call iax2_destroy() for
that call while the global frame queue is still locked. There is a very nice
explanation of the deadlock in the report.
(issue #9663 , thorough report and patch from stevedavies, additional positive
test reports from mihai and joff_oconnell)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67020 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Russell Bryant
0ecb2ecf42
Changes to the way DTMF is handled in the core broke dialing in chan_skinny.
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This patch makes chan_skinny usable again. I did not end up testing this,
but there are multiple positive test reports listed in the bug report.
(issue #9596 , reported by pj, testing by pj and mvanbaak, and the fix was
written by DEA)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66881 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
c7112015ba
Merged revisions 66764 via svnmerge from
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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/branches/1.4@66768 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
612f61a9b2
Silly me for having out of date source! Oh well... I'm still leaving my comment.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66639 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
3d8d697e34
When calling some peer/host that may not exist/reply back... don't keep the dialog in memory for all of eternity.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
7905fca326
Change how channel names are generated a bit. (issue #9825 reported by eldadran)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66602 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
332eabcc07
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/branches/1.4@66503 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
8d06f379fe
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/branches/1.4@66474 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
9f15005143
Don't reset hangupcause if we already have one
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66414 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
ff9e2751c6
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/branches/1.4@66404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
bab6473879
Merged revisions 66349 via svnmerge from
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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/branches/1.4@66363 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
3a5a649415
Make the usedistinctiveringdetection option work again. (issue #9823 reported by premeau)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66312 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Jason Parker
a8ab3e5738
I don't know what this was trying to do, but it's clearly incorrect.
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Issues 9808 and 9809.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66244 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Kevin P. Fleming
5f9897ba56
handle the GNUTLS library properly in the configure script and build system
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don't build in OSP support unless we have found and are allowed to use SSL support
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66157 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Kevin P. Fleming
0b14e93d00
oops, use #ifdef instead of #if
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65967 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Kevin P. Fleming
e49f14e6cb
don't reference GnuTLS headers and functions unless the configure script found it
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65966 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
4eab565b25
Issue 7672 - fix by zandbelt - Asterisk core dump since the GnuTLS interface did not support multithreading correctly.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65901 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
e8addde3f1
Issue 8193 - NAT issues with gtalk/STUN. Patch by phsultan. Thanks!
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65892 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
abc8683ce7
Issue 7686, fix by phsultan, NAT issues when calling from gtalk to SIP over nat.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65857 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
e4f9cc15e6
Issue #8536 - Caller ID not set in CDR for jingle
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65841 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
0df2a42f96
Merged revisions 65837 via svnmerge from
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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
19 years ago
Olle Johansson
6cfe6a550e
Issue 8409 - phsultan - Fix "login" as component to jabber server.
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...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
19 years ago
Christian Richter
9f54cd55af
Merged revisions 65767 via svnmerge from
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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
19 years ago
Kevin P. Fleming
e1518f42ae
start the delayed PBX when receive voice or video full frames as well, and comment this delayed-PBX activity
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65685 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Kevin P. Fleming
ca6b421be4
Merged revisions 65682 via svnmerge from
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65683 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Kevin P. Fleming
b89faf596b
clear the 'delay PBX' flag when we are ready to start the PBX
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65680 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65679 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Kevin P. Fleming
f608d90283
Merged revisions 65676 via svnmerge from
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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
19 years ago
Russell Bryant
7398856cea
Merged revisions 65588 via svnmerge from
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Russell Bryant
c556cc222c
List res_smdi as a dependency for app_voicemail and chan_zap
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(Thanks to mnicholson for pointing it out)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65501 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Christian Richter
e7355ec53b
Merged revisions 65328 via svnmerge from
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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
19 years ago
Olle Johansson
4483fa12e8
Merged revisions 65122 via svnmerge from
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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
19 years ago
Olle Johansson
7fe3608300
Merged revisions 65075 via svnmerge from
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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
19 years ago
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.
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- 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
19 years ago
Christian Richter
012fe116fa
Merged revisions 65007 via svnmerge from
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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
19 years ago
Olle Johansson
73d0ba053b
Issue 9487 - stop media flows at hangup of call
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64974 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Christian Richter
d682a74e26
Merged revisions 64902 via svnmerge from
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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
19 years ago
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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64754 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
4ae20ba8e4
Fix auth on BYE. (Different patch than for 1.2)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64605 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
5acc63d688
Issue #9681 - Handle www-auth on BYE
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64602 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
374b52f717
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/branches/1.4@64578 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
70ad556544
Issue #9439 - properly handle username parameters in SIP uri.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
bf1a15b9bf
Merged revisions 64535 via svnmerge from
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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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19 years ago
Olle Johansson
56af259505
Merged following patch with a lot of changes for 1.4
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Merged revisions 64514 via svnmerge from
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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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19 years ago
Christian Richter
d17174cfc1
Merged revisions 64513 via svnmerge from
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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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19 years ago
Olle Johansson
1f2afa0ff1
Change -2 to XMIT_ERROR to clarify a bit more
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19 years ago
Russell Bryant
b340fcd7d0
Properly handle AST_CONTROL_PROGRESS by just ignoring it. An unknown indication
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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
19 years ago
Olle Johansson
e041d57175
Handle network errors, like host or network unreachable, in a better way. This means that
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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.
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19 years ago
Steve Murphy
8c635fb2c1
As per 9570, worrisome CDR warnings have been removed, that are either not helpful, or not relevant.
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19 years ago
Joshua Colp
9a73c07fb1
This concludes my final adventure with bitmasks and the onhold flag. Would anyone care for some peanuts?
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19 years ago
Joshua Colp
699aa6ad74
Tweak hold flags some more. They can be of three states when active: active, inactive, one direction.
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19 years ago
Joshua Colp
026000bae8
Ensure the onhold flag is set no matter what when being put on hold.
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19 years ago
Jason Parker
c2c91c7fef
Merged revisions 63828 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r63828 | qwell | 2007-05-10 18:14:55 -0500 (Thu, 10 May 2007) | 4 lines
Fix an issue with trying to kill a thread before it gets created.
Issue 9709, patch by nic_bellamy.
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19 years ago
Olle Johansson
ca1ae5e81a
Merged revisions 63748 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r63748 | oej | 2007-05-10 22:38:54 +0200 (Thu, 10 May 2007) | 4 lines
Do not allocate SIP pvt's for PEERs we can not reach.
This was seen as a lot of dialogs being created then immediately destroyed at reload/restart of the SIP channel.
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19 years ago
Joshua Colp
67a0bbaa83
Do not prematurely go on hold if sendonly was not actually set.
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19 years ago
Matthew Fredrickson
a181c6a777
Merged revisions 63653 via svnmerge from
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r63653 | mattf | 2007-05-09 12:20:20 -0500 (Wed, 09 May 2007) | 2 lines
Make sure we only create a DSP if it's requested on SUB_REAL
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19 years ago
Joshua Colp
c83c0072bb
Merged revisions 63610 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r63610 | file | 2007-05-09 12:51:03 -0400 (Wed, 09 May 2007) | 2 lines
Properly handle hints that point to multiple devices in chan_sip. Why chan_sip is even doing this I have no idea but I would rather not go into a rant. (issue #9536 reported by rlister)
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19 years ago
Christian Richter
3e63c9d542
Merged revisions 62945,63402,63519 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r62945 | crichter | 2007-05-03 17:39:21 +0200 (Do, 03 Mai 2007) | 1 line
when we're in state WAITING4DIGS, we use the asterisk tone-generator which prods us, so we can't just return -1 in misdn_write in this case. Added a MISDN_KEYPAD channel variable, and fixed the sending of keypad. this enables us to modify the call forward parameters in the switch.
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r63402 | crichter | 2007-05-08 17:07:37 +0200 (Di, 08 Mai 2007) | 1 line
added application misdn_check_l2l1 which tries to pull up the L1/L2 on all ports that have the layers down in a group. It waits then for a timeout. This helps for scenarios where multiple PMP BRIs are grouped together, or where a provider has a faulty PTP Implementation, that looses the L2 after a while.
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r63519 | crichter | 2007-05-09 13:26:16 +0200 (Mi, 09 Mai 2007) | 1 line
release_chan frees ch, so we should never touch ch after release_chan, this may cause segfaults.
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19 years ago
Olle Johansson
d30faa1dc2
Don't retransmit 200 OK's on ignore status. (Reported on asterisk-users)
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19 years ago
Joshua Colp
b50b92ab77
Minor backport of revision 59083 in trunk. Don't queue an unhold frame up if the call was never on hold to begin with.
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19 years ago
Joshua Colp
95de3fbf0c
Merged revisions 62987 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r62987 | file | 2007-05-03 13:42:19 -0300 (Thu, 03 May 2007) | 2 lines
When a peer is seeded or built tell the devicestate core to update it's status. This is easier then having chan_sip load before pbx_config. (issue #9658 reported by dlynes)
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19 years ago
Christian Richter
460e677ea6
Merged revisions 61357,61770,62885 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r61357 | crichter | 2007-04-11 14:05:57 +0200 (Mi, 11 Apr 2007) | 1 line
some fixes for PMP Hold/Retrieve, it should work now, when briding=no
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r61770 | crichter | 2007-04-24 15:50:05 +0200 (Di, 24 Apr 2007) | 1 line
added lock for sending messages to avoid double sending. shuffled some empty_chans after the cb_event calls, this avoids that a release_complete from a quite different call releases a fresh created setup by accident.
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r62885 | crichter | 2007-05-03 15:59:00 +0200 (Do, 03 Mai 2007) | 1 line
fixed the problem that misdn_write did not return -1 when called with 0 samples in a frame this resultet in a deadlock in some circumstances, when the call ended because of a busy extension. added encoding of keypad.
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19 years ago
Tilghman Lesher
8bcfcfca88
Merged revisions 62691 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r62691 | tilghman | 2007-05-02 12:38:16 -0500 (Wed, 02 May 2007) | 4 lines
Issue 9638 - if a text frame is sent with no terminating NULL through a bridged
IAX connection, the remote end will receive garbage characters tacked onto the
end.
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19 years ago
Steve Murphy
55f4eb3e3d
a)In chan_zap, set the clid, src fields in channel_alloc call. b)in the channel_alloc func, set the cid_num and name fields from the arglist[blush]. c) don't update the channel app & app data fields if you are in the 'h' extension. d)the load_module func in cdr_radius needs to return DECLINE, SUCCESS.
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19 years ago
Olle Johansson
d5fda03428
Don't unlock a channel that we already know does not exist (propably isue 8228)
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19 years ago
Russell Bryant
3595c5fed1
Merged revisions 62417 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r62417 | russell | 2007-04-30 10:57:26 -0500 (Mon, 30 Apr 2007) | 4 lines
This patch fixes an issue where depending on the cause code, when the network
sends a PRI disconnect, the call may not be properly hung up.
(issue #9588 , reported and patched by softins)
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19 years ago
Russell Bryant
8936804377
Fix a bug that made the "language" setting in zapata.conf not
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functional. (issue #9626 , reported and fixed by sergee)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@62331 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Russell Bryant
6b76a95f09
Fix a weird problem where when a caller talking to someone sitting behind an
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agent channel sent a digit, the digit would be played to the agent for forever.
This is because chan_agent always returned -1 from its send_digit_begin and _end
callbacks. This non-zero return value indicates to the Asterisk core that it
would like an inband DTMF generator put on the channel. However, this is the
wrong thing to do. It should *always* return 0, instead. When the digit begin
and end functions are called on the proxied channel, the underlying channel
will indicate whether inband DTMF is needed or not, and the generator will be
put on that one, and not the Agent channel.
(issue #9615 , #9616 , reported by jiddings and BigJimmy, and fixed by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@62218 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
34f9e0c4dd
Merged revisions 62126 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r62126 | oej | 2007-04-27 15:57:45 +0200 (Fri, 27 Apr 2007) | 4 lines
Issue #7351 - SIP Cancel fails due to the wrong contact uri. Reported by PPYY, failed to fix by OEJ
final fix by wojtekka - THANKS!!!! THis was a hard one to catch.
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19 years ago
Olle Johansson
b0121ca8a3
Issue #9608 - fix some annoying DEBUG messages not controlled by option_debug (DEA). Thanks!
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19 years ago
Joshua Colp
0525347958
Merged revisions 62037 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r62037 | file | 2007-04-26 12:30:57 -0400 (Thu, 26 Apr 2007) | 2 lines
Revert previous fix for when the IAX2 channel goes funky (that's the technical term). This is causing legit calls to be prematurely hung up. (issue #9600 reported by justdave)
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19 years ago
Kevin P. Fleming
0e525813a8
Merged revisions 61913 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r61913 | kpfleming | 2007-04-25 17:24:59 -0500 (Wed, 25 Apr 2007) | 2 lines
handle a very bizarre race condition with channels being redirected before a simple switch can be started on them (issue #9286 )
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19 years ago
Russell Bryant
5574da0c12
Merged revisions 61866 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r61866 | russell | 2007-04-25 16:55:23 -0500 (Wed, 25 Apr 2007) | 2 lines
If the callerid= option is specified, but empty, clear any previous data.
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19 years ago
Russell Bryant
e6aa1145ed
Merged revisions 61862 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r61862 | russell | 2007-04-25 16:06:22 -0500 (Wed, 25 Apr 2007) | 2 lines
Ensure that callerid settings are reset on a reload.
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19 years ago
Russell Bryant
77c52fd8bf
Merged revisions 61798 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r61798 | russell | 2007-04-25 11:20:38 -0500 (Wed, 25 Apr 2007) | 3 lines
Fix a typo where cid_num got copied instead of cid_ani.
(issue #9587 , reported and patched by xrg)
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19 years ago
Dwayne M. Hubbard
db55615090
Merged revisions 61777 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r61777 | dhubbard | 2007-04-24 13:20:31 -0500 (Tue, 24 Apr 2007) | 1 line
removed #if 0 block from chan_phone, chan_zap, and chan_modem restart_monitor()
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19 years ago
Joshua Colp
b271179161
Merged revisions 61771 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r61771 | file | 2007-04-24 12:05:06 -0400 (Tue, 24 Apr 2007) | 2 lines
Allow RFC2833 to be sent in the response SDP when an INVITE comes in without SDP. (issue #9546 reported by mcrawford)
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19 years ago
Olle Johansson
32be6621f6
Merged revisions 61685 via svnmerge from
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r61685 | oej | 2007-04-19 09:56:21 +0200 (Thu, 19 Apr 2007) | 3 lines
Send NOTIFY to Contact: in SUBSCRIBE
- as reported by Intertex and Citel. Fixed during SIPit 20 in Antwerp.
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19 years ago
Olle Johansson
dc6d309bdb
#9483 , half of patch by twilson to solve 302 redirect issues
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19 years ago
Joshua Colp
eb631c00a3
For those very verbose SIP implementations that attach tons of info to the Contact header... let's increase our variable sizes. (issue #9535 reported by jeffg)
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19 years ago
Steve Murphy
8a386a9816
A fix for chan_oss that resulted from the CDR changes; it helps to use the right info.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@61644 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
c06e063e1a
Don't assume the callid of a dialog will be set, as in some circumstances it may not. (issue #9534 reported by tecnoxarxa)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@61641 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Russell Bryant
5858778f40
Merged revisions 61476 via svnmerge from
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r61476 | russell | 2007-04-11 11:01:25 -0500 (Wed, 11 Apr 2007) | 5 lines
If someone sets the "useragent" option in sip.conf to be empty, then don't add
the User-Agent header at all. It is an optional header, anyway. Also, the bug
report says that some of Japan's SIP providers don't allow it for some weird
reason. (issue #9488 , reported by makoto, fixed by me)
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19 years ago