mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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1.6.0.8
1.6.0.9
1.6.1-beta1
1.6.1-beta2
1.6.1-beta3
1.6.1-beta4
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1.6.1.0
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1.6.1.0-rc4
1.6.1.0-rc5
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10.5.0-digiumphones-rc1
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10.5.1-digiumphones
10.5.2
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10.6.0
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10.6.0-digiumphones-rc1
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10.7.0
10.7.0-digiumphones
10.7.0-digiumphones-rc1
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10.8.0
10.8.0-digiumphones
10.8.0-digiumphones-rc1
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10.9.0
10.9.0-digiumphones
10.9.0-digiumphones-rc1
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10.9.0-digiumphones-rc3
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11.0.0
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${ noResults }
5264 Commits (c23e5fea96135f5b7de00ec0eb8cf651a6aff08d)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Tilghman Lesher
|
bb0dbfa520 |
Merged revisions 123391 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123391 | tilghman | 2008-06-17 13:56:53 -0500 (Tue, 17 Jun 2008) | 3 lines Fix 3 more places where failure to lock the structure could cause the wrong lock to be unlocked. (Closes issue #12795) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123392 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Mark Michelson
|
8c6184f0da |
Merged revisions 123333 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123333 | mmichelson | 2008-06-17 13:09:16 -0500 (Tue, 17 Jun 2008) | 11 lines Cisco BTS sends SIP responses with a tab between the Cseq number and SIP request method in the Cseq: header. Asterisk did not handle this properly, but with this patch, all is well. (closes issue #12834) Reported by: tobias_e Patches: 12834.patch uploaded by putnopvut (license 60) Tested by: tobias_e ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123334 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Jeff Peeler
|
c837af5e28 |
Fix some (more) variables that were forgotten to be renamed, related to 117658
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123238 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Doug Bailey
|
2690378411 |
Clean up code that handles fsk mwi message generation by pulling it from do_monitor and creating its own thread.
Added RP-AS mwi message generation using patches from meneault as a basis. (closes issue #8587) Reported by: meneault Tested by: meneault git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123203 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
ddfac77569 |
Oopsie, breakage
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123201 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Jeff Peeler
|
121c046ba5 |
Fix some variables that were forgotten to be renamed, related to 117658
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123166 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
596f8b5186 |
Merged revisions 123113 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123113 | tilghman | 2008-06-16 14:50:12 -0500 (Mon, 16 Jun 2008) | 2 lines Port "hasvoicemail" change from SIP to other channel drivers ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123114 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
ba07bd38b7 |
Merged revisions 123110 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123110 | tilghman | 2008-06-16 14:21:58 -0500 (Mon, 16 Jun 2008) | 8 lines People expect that if "hasvoicemail" is set in users.conf, even if "mailbox" isn't set, that SIP will detect a mailbox. (closes issue #12855) Reported by: PLL Patches: 20080614__bug12855__2.diff.txt uploaded by Corydon76 (license 14) Tested by: PLL ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123111 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Joshua Colp
|
523532204a |
Merged revisions 122919 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122919 | file | 2008-06-16 09:31:09 -0300 (Mon, 16 Jun 2008) | 6 lines Only compare the first 15 characters so that even if the charset is specified we still accept it as SDP. (closes issue #12803) Reported by: lanzaandrea Patches: chan_sip.c.diff uploaded by lanzaandrea (license 496) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122920 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
|
1c8f33b0d6 |
Merged revisions 122869 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122869 | file | 2008-06-16 09:08:28 -0300 (Mon, 16 Jun 2008) | 6 lines Don't send a BYE on a dialog that is already gone during a REFER. (closes issue #12865) Reported by: flefoll Patches: chan_sip.c.br14.121495.patch-ALREADYGONE uploaded by flefoll (license 244) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122870 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
b2ef18dab4 |
Add some more IAX2-specific information about the channel to the CHANNEL()
function and begin the transition from SIPCHANINFO() to just using CHANNEL(). (closes issue #12856) Reported by: mostyn Patches: iax_and_sip_channel_info.patch uploaded by mostyn (license 398) (with some additional cleanup by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122802 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Russell Bryant
|
b6457ecf4c |
Merge changes from timing branch
- Convert chan_iax2 to use the timing API - Convert usage of timing in the core to use the timing API instead of using DAHDI directly - Make a change to the timing API to add the set_rate() function - change the timing core to use a rwlock - merge a timing implementation, res_timing_dahdi Basic testing was successful using res_timing_dahdi git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122523 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Michiel van Baak
|
a68a160b75 |
Implement call parking in chan_skinny.
(closes issue #11342) Reported by: DEA Patches: chan_skinny-park.txt uploaded by DEA (license 3) chan_skinny-park-v2.diff.txt uploaded by mvanbaak (license 7) Tested by: DEA, mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122493 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Russell Bryant
|
968cf2b578 |
Merged revisions 122259 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122259 | russell | 2008-06-12 13:22:44 -0500 (Thu, 12 Jun 2008) | 3 lines Fix some race conditions that cause ast_assert() to report that chan_iax2 tried to remove an entry that wasn't in the scheduler ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122262 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Jeff Peeler
|
d9faeb6177 |
Fix pseudo channel allocation errors on startup when using SS7. (from mattf r121914, moving from chan_zap to chan_dahdi)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122244 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Jeff Peeler
|
c8b2bd8817 |
Make sure we hangup any calls we have and NULL out the ss7call value when we get a reset circuit message. Fixes crash bug. (from mattf r121857, moving from chan_zap to chan_dahdi)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122243 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Jeff Peeler
|
ef3b214728 |
Goodbye Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI. Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122234 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
|
6ac8ccaba4 |
Make this build under dev mode
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122232 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Matthew Fredrickson
|
78e593eecb |
Fix pseudo channel allocation errors on startup when using SS7
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121914 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
|
467c6f5f90 |
Merged revisions 121861 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r121861 | tilghman | 2008-06-11 13:18:16 -0500 (Wed, 11 Jun 2008) | 3 lines Make calls to ast_assert() actually test something, so that the error message printed is not nonsensical (reported by mvanbaak via #asterisk-bugs). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121867 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Matthew Fredrickson
|
4639347531 |
Make sure we hangup any calls we have and NULL out the ss7call value when we get a reset circuit message. Fixes crash bug
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121857 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Christian Richter
|
06a8effb23 |
Merged revisions 121751 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r121751 | crichter | 2008-06-11 11:28:04 +0200 (Mi, 11 Jun 2008) | 1 line fixed issue with previous commit, the find_free_channel test for channels which where inuse was broken. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121770 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Joshua Colp
|
7025da48e5 |
Fix issue where session timer headers were present when they should not have been.
(closes issue #12706) Reported by: falves11 Patches: chan_sip.c.diff uploaded by rjain (license 226) Tested by: falves11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121503 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Joshua Colp
|
51602928e3 |
Merged revisions 121495 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r121495 | file | 2008-06-10 10:34:27 -0300 (Tue, 10 Jun 2008) | 4 lines If we are destroying a dialog only set the MWI dialog pointer on the related peer to NULL if it is the dialog currently being destroyed. (closes issue #12828) Reported by: ramonpeek ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121496 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
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74932445b7 |
Bump up the debug level of a couple of messages
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121407 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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53459f86b2 |
Expand RQ_INTEGER type out to multiple types, one for each precision
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121367 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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ba622c3431 |
Add storage of the useragent in the realtime database.
(Closes AST-38) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121334 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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f1d9962610 |
Merged revisions 121229 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 (Note that this is being merged to trunk/1.6.0 because it may affect non-callback agents with ackcall set) ........ r121229 | mmichelson | 2008-06-09 10:02:37 -0500 (Mon, 09 Jun 2008) | 16 lines A unique situation of timeouts brought forth a failure situation for autologoff in chan_agent. If using AgentCallbackLogin-style agents, then if the timeout specified by the Dial() to reach the agent's phone was shorter than the timeout specified in queues.conf, then autologoff would only work if the caller hung up while the agent's phone was ringing. This patch allows autologoff to work in this situation when the call in queue transfers to the next available agent (as it would have if the timeout in queues.conf were less than the timeout in the Dial()). (closes issue #12754) Reported by: Rodrigo Patches: 12754.patch uploaded by putnopvut (license 60) Tested by: Rodrigo ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121230 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Jeff Peeler
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33b3d38a3c |
This was accidentally reverted.
Fixes a bug where if a stream monitor thread was not created (caused from failure of opening or starting the stream) pthread_cancel was called with an invalid thread ID. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121163 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
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96233af5dc |
Merged revisions 121078 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r121078 | russell | 2008-06-07 09:10:56 -0500 (Sat, 07 Jun 2008) | 7 lines Don't run LIST_HEAD_DESTROY on a STATIC list (closes issue #12807) Reported by: ys Patches: chan_agent_local.diff uploaded by ys (license 281) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121079 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
|
07265a5033 |
Added a facility for sending arbitrary SIP notify commands from AMI.
(closes issue #12562) Reported by: michael-fig Patches: 20080515__bug12562.diff.txt uploaded by Corydon76 (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121042 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Jeff Peeler
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c7da6df5e1 |
Merged revisions 120959 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120959 | jpeeler | 2008-06-06 13:29:14 -0500 (Fri, 06 Jun 2008) | 1 line add another LOW_MEMORY define I forgot ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120960 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Jeff Peeler
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0bc65f7465 |
Merged revisions 120908 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120908 | jpeeler | 2008-06-06 13:05:15 -0500 (Fri, 06 Jun 2008) | 1 line only define thread storage variable if necessary for LOW_MEMORY ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120909 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Jeff Peeler
|
5934801d84 |
Merged revisions 120863,120885 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120863 | jpeeler | 2008-06-06 10:33:15 -0500 (Fri, 06 Jun 2008) | 3 lines This fixes a crash when LOW_MEMORY is turned on. Two allocations of the ast_rtp struct that were previously allocated on the stack have been modified to use thread local storage instead. ........ r120885 | jpeeler | 2008-06-06 11:39:20 -0500 (Fri, 06 Jun 2008) | 2 lines Correction to commmit 120863, make sure proper destructor function is called as well define two thread storage local variables. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120906 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
|
9471b87d27 |
Merge the adaptive realtime branch, which will make adding new required fields
to realtime less painful in the future. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120789 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Brett Bryant
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c1451b5537 |
This patch adds more detailed statistics for RTP channels, and provides an API call to access it, including maximums, minimums, standard deviatinos,
and normal deviations. Currently this is implemented for chan_sip, but could be added to the func_channel_read callbacks for the CHANNEL function for any channel that uses RTP. (closes issue #10590) Reported by: gasparz Patches: chan_sip_c.diff uploaded by gasparz (license 219) rtp_c.diff uploaded by gasparz (license 219) rtp_h.diff uploaded by gasparz (license 219) audioqos-trunk.diff uploaded by snuffy (license 35) rtpqos-trunk-r119891.diff uploaded by sergee (license 138) Tested by: jsmith, gasparz, snuffy, marsosa, chappell, sergee git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120635 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
|
c73972e48d |
Merged revisions 120425 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120425 | tilghman | 2008-06-04 13:35:47 -0500 (Wed, 04 Jun 2008) | 6 lines If we fail to setup the PRI request channel, don't continue, exit with an error. (closes issue #11989) Reported by: Corydon76 Patches: 20080213__zap_memleak.diff.txt uploaded by Corydon76 (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120426 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
|
5866b0dfe8 |
Merged revisions 120168 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120168 | russell | 2008-06-03 16:34:55 -0500 (Tue, 03 Jun 2008) | 4 lines Fix another place where peer->callno could change at a very bad time, and also fix a place where a peer was used after the reference was released. (inspired by rev 120001) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120169 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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f3a62ab27e |
Merged revisions 120001 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120001 | tilghman | 2008-06-03 11:10:53 -0500 (Tue, 03 Jun 2008) | 9 lines Save the callno when we're poking, because our peer structure could change during destruction (and thus we unlock the wrong callno, causing a cascade failure). (closes issue #12717) Reported by: gewfie Patches: 20080525__bug12717.diff.txt uploaded by Corydon76 (license 14) Tested by: gewfie ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120012 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
|
16e401cc68 |
Merged revisions 119926 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119926 | file | 2008-06-03 11:46:24 -0300 (Tue, 03 Jun 2008) | 2 lines Treat ECONNREFUSED as an error that will stop further retransmissions. (issue #AST-58, patch from Switchvox) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119927 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
|
d6240ac21e |
Merged revisions 119838 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119838 | russell | 2008-06-02 15:08:04 -0500 (Mon, 02 Jun 2008) | 7 lines Revert a change made for issue #12479. This change caused a regression such that a dial string such as (IAX2/foo) did not automatically fall back to dialing the 's' extension anymore. (closes issue #12770) Reported by: dagmoller ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119839 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Philippe Sultan
|
001c95b595 |
Do not link the guest account with any configured XMPP client (in
jabber.conf). The actual connection is made when a call comes in Asterisk. Apply this fix to Jingle too. Fix the ast_aji_get_client function that was not able to retrieve an XMPP client from its JID. (closes issue #12085) Reported by: junky Tested by: phsultan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119741 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
|
2feb90d511 |
Merged revisions 119687 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119687 | russell | 2008-06-02 07:30:17 -0500 (Mon, 02 Jun 2008) | 3 lines Even of the first PING or LAGRQ doesn't get sent because it comes up too soon, make sure to reschedule so it gets sent later. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119688 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Christian Richter
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0f4bebac81 |
Merged revisions 119636 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119636 | crichter | 2008-06-02 11:29:21 +0200 (Mo, 02 Jun 2008) | 1 line fixed compile issue when dev-mode is enabled ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119637 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Christian Richter
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c47a3f89eb |
Merged revisions 119585 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119585 | crichter | 2008-06-02 10:35:28 +0200 (Mo, 02 Jun 2008) | 1 line Added counter for unhandled_bmsg Print, this prevents the logs to be flooded to fast and save CPU in this error scenario. Added 'last_used' element to bc structure, when a bchannel changes from used to free this exact time will be marked in last_used. When a new channel is requested the find_free_chan function will check if the new empty channel was used within the last second, if yes it will search for the next channel, if no it will return this channel. This simple mechanism has prooven to prevent race conditions where the NT and TE tried to allocate the exact same channel at the same time (RELEASE cause: 44). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119586 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
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ea3c47e7df |
Merged revisions 119533 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119533 | russell | 2008-06-01 20:06:09 -0500 (Sun, 01 Jun 2008) | 2 lines Change a debug message to an actual debug message ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119534 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
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87c9b6fc25 |
Merged revisions 119238 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r119238 | russell | 2008-05-30 07:55:36 -0500 (Fri, 30 May 2008) | 15 lines Merged revisions 119237 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r119237 | russell | 2008-05-30 07:49:39 -0500 (Fri, 30 May 2008) | 7 lines - Instead of only enforcing destination call number checking on an ACK, check all full frames except for PING and LAGRQ, which may be sent by older versions too quickly to contain the destination call number. (As suggested by Tim Panton on the asterisk-dev list) - Merge changes from team/russell/iax2-frame-race, which prevents PING and LAGRQ from being sent before the destination call number is known. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119239 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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dbf9f31446 |
Merged revisions 119071 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119071 | tilghman | 2008-05-29 15:24:11 -0500 (Thu, 29 May 2008) | 7 lines Call waiting tone occurs too often, because it's getting serviced by both subchannels. (closes issue #11354) Reported by: cahen Patches: 20080512__bug11354.diff.txt uploaded by Corydon76 (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119072 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
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9397f04294 |
Merged revisions 119009 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r119009 | russell | 2008-05-29 13:49:12 -0500 (Thu, 29 May 2008) | 16 lines Merged revisions 119008 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r119008 | russell | 2008-05-29 13:45:21 -0500 (Thu, 29 May 2008) | 7 lines Merge changes from team/russell/iax2-another-fix-to-the-fix As described in the following post to the asterisk-dev mailing list, only enforce destination call numbers when processing an ACK. http://lists.digium.com/pipermail/asterisk-dev/2008-May/033217.html (closes issue #12631) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119010 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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6e5d843a71 |
Merged revisions 118953 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118953 | tilghman | 2008-05-29 12:20:16 -0500 (Thu, 29 May 2008) | 3 lines Add some debugging code that ensures that when we do deadlock avoidance, we don't lose the information about how a lock was originally acquired. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118955 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |