mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
5113 Commits (b7a25e15d1ed0bf6166be6d7979b75f53e110352)
Author | SHA1 | Message | Date |
---|---|---|---|
|
a79214b5b1 |
Merged revisions 114322 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114322 | file | 2008-04-21 11:39:32 -0300 (Mon, 21 Apr 2008) | 4 lines Only drop audio if we receive it without a progress indication. We allow other frames through such as DTMF because they may be needed to complete the call. (closes issue #12440) Reported by: aragon ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114323 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
264cc6ff5a |
SS7:Added - Generic Name / Access Transport / Redirecting Number handling. #12425
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114303 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
b05e17fdd7 |
Make sure ADSI is marked as unavailable on Unistim channels so voicemail does not try to do some ADSI jazz.
(closes issue #12460) Reported by: PerryB git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114271 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
0e821d7201 |
Merged revisions 114257 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114257 | mmichelson | 2008-04-18 12:44:29 -0500 (Fri, 18 Apr 2008) | 6 lines Clearing up error messages so they make a bit more sense. Also removing a redundant error message. Issue AST-15 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114259 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
e4dce85331 |
Merged revisions 114245 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114245 | seanbright | 2008-04-18 09:33:32 -0400 (Fri, 18 Apr 2008) | 1 line Only complete the SIP channel name once for 'sip show channel <channel>' ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114246 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
5203c664de |
Thanks to snuff for finding these omissions
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114201 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
5fb4b1bbe5 |
This is the scariest commit I've done in a long time. This is the astobj2-ification of chan_sip. I've tested a number of scenarios like crazy. It used to have 4x the call setup/teardown performance of trunk, but now it's roughly at parity. I will attempt to find the bottlenecks and get it back to the 4x mark. The changes made were somewhat invasive, but the value to the community of these upgrades outweighs waiting further for more testing. Every change being made to chan_sip was lousing this code up when we tried to merge. Peers, Users, Dialogs, are all now astobj2 objects, indexed via hashtables. Refcounting is used to track objects and free them at the bitter end of their lives. Please file issues on bugs.digium.com, and PLEASE, please, please be patient. One natural advantage to all the hash-table work is that loading large sip.conf files full of thousands of peers now goes much faster. One more please: PLEASE help thrash this code and test it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114190 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
a51fb142f9 |
Merged revisions 114184 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114184 | kpfleming | 2008-04-16 15:46:38 -0500 (Wed, 16 Apr 2008) | 6 lines use the ZT_SET_DIALPARAMS ioctl properly by initializing the structure to all zeroes in case it contains fields that we don't write values into (which it does as of Zaptel 1.4.10) (closes issue #12456) Reported by: fnordian ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114185 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
18866623dc |
Merged revisions 114148 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114148 | oej | 2008-04-15 22:26:05 +0200 (Tis, 15 Apr 2008) | 2 lines Handle subscribe queues in all situations... Thanks to festr_ on irc for telling me about this bug. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114151 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
f239f24580 |
Adding chanvar to SIPPEER from 1.4 branch
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114150 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
b9bb0749d1 |
Shorten the mac address pattern, since some phones use different identifiers (such as the i2050 softphone).
(closes issue #12398) Reported by: c_hans Patches: chan_unistim_svn.diff uploaded by c (license 460) Tested by: c_hans git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114141 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
6e6d6f2e10 |
Merged revisions 114120 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114120 | qwell | 2008-04-14 13:31:57 -0500 (Mon, 14 Apr 2008) | 7 lines The call_token on the pvt can occasionally be NULL, causing a crash. If it is NULL, we can skip this channel, since it can't the one we're looking for. (closes issue #9299) Reported by: vazir ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114121 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
6fad8249f5 |
During hangup it is possible for p->chan or p->owner to be NULL, so just return what the channel is bridged to instead of what they are *really* bridged to. Thanks Matt Nicholson!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114109 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
c5d0ca23f0 |
Merged revisions 114103 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114103 | file | 2008-04-14 11:52:46 -0300 (Mon, 14 Apr 2008) | 4 lines It is possible for the remote side to say they want T38 but not give any capabilities. (closes issue #12414) Reported by: MVF ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114104 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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5110d3bc69 |
Make sure linkset is locked exiting ss7_start_call
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114093 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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f9960bc748 |
Make sure we start incoming calls on SS7 with echo cancellation enabled. Also make sure when completing a COT we call ss7_start_call with the proper locks held. Lastly, make sure if we fail to get a channel from zt_new that we don't assume it's there.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114092 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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4bc75c9a55 |
Merged revisions 114083 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114083 | twilson | 2008-04-11 17:32:51 -0500 (Fri, 11 Apr 2008) | 7 lines Several places in the code called find_callno() (which releases the lock on the pvt structure) and then immediately locked the call and did things with it. Unfortunately, the call can disappear between the find_callno and the lock, causing Bad Stuff(tm) to happen. Added find_callno_locked() function to return the callno withtout unlocking for instances that it is needed. (issue #12400) Reported by: ztel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114084 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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a08c4b2064 |
A 'b' option has been added which causes chan_local to return the actual channel that is behind it when queried. This is useful for transfer scenarios as the actual channel will be transferred, not the Local channel. If you have been using Local channels as queue members and having issues when the agent did a blind transfer this option may solve the issue.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114049 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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d13b45564b |
Merged revisions 114045 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114045 | mmichelson | 2008-04-10 14:55:33 -0500 (Thu, 10 Apr 2008) | 6 lines Be sure that we're not about to set bridgepvt NULL prior to dereferencing it. (closes issue #11775) Reported by: fujin ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114046 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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4a21c5dd22 |
Fix spelling of existent in a few places.
(closes issue #12409) Reported by: candlerb git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114024 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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a4e73acaf8 |
Merged revisions 114021 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114021 | file | 2008-04-10 10:27:11 -0300 (Thu, 10 Apr 2008) | 6 lines Don't add custom URI options if they don't exist OR they are empty. (closes issue #12407) Reported by: homesick Patches: uri_options-1.4.diff uploaded by homesick (license 91) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114022 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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88cc98ea94 |
Merged revisions 113927 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113927 | mmichelson | 2008-04-09 15:54:31 -0500 (Wed, 09 Apr 2008) | 8 lines We need to set the persistant_route [sic] parameter for the sip_pvt during the initial INVITE, no matter if we're building the route set from an INVITE request or response. (closes issue #12391) Reported by: benjaminbohlmann Tested by: benjaminbohlmann ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113928 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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0351ef6e6e |
Enable enough RTP bridging to allow P2P to work.
(closes issue #11901) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113840 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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d314fd5336 |
Move all messages wrapped in skinnydebug from debug to verbose.
(closes issue #12224) Reported by: DEA Patches: chan_skinny-debug-log.txt uploaded by DEA (license 3) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113834 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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230d9d1465 |
Merged revisions 113784 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113784 | file | 2008-04-09 13:50:45 -0300 (Wed, 09 Apr 2008) | 4 lines If we receive an AUTHREQ from the remote server and we are unable to reply (for example they have a secret configured, but we do not) then queue a hangup frame on the Asterisk channel. This will cause the channel to hangup and a HANGUP to be sent via IAX2 to the remote side which is the proper thing to do in this scenario. (closes issue #12385) Reported by: viraptor ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113785 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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925924386a |
Merged revisions 113681 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113681 | mmichelson | 2008-04-09 09:40:05 -0500 (Wed, 09 Apr 2008) | 9 lines If Asterisk receives a 488 on an INVITE (not a reinvite), then we should not send a BYE. (closes issue #12392) Reported by: fnordian Patches: chan_sip.patch uploaded by fnordian (license 110) with small modification from me ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113682 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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3ee1602b6a |
Merged revisions 113596 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113596 | twilson | 2008-04-08 20:34:25 -0500 (Tue, 08 Apr 2008) | 2 lines Initialize fr->cacheable to make valgrind happy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113597 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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b52ec53da7 |
Merged revisions 113504 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113504 | qwell | 2008-04-08 13:48:55 -0500 (Tue, 08 Apr 2008) | 1 line Add a little more that is required for previously added devices. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113505 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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f469ee8cf2 |
Merged revisions 113454 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113454 | qwell | 2008-04-08 13:07:49 -0500 (Tue, 08 Apr 2008) | 4 lines Add support for several new(ish) devices - most notably, 7942/7945, 7962/7965, 7975. Thanks to Greg Oliver for providing me the required information. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113455 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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fa875c0578 |
Merged revisions 113348 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113348 | tilghman | 2008-04-08 10:39:16 -0500 (Tue, 08 Apr 2008) | 7 lines Move check for still-bridged channels out a little further, to avoid possible deadlocks. (Closes issue #12252) Reported by: callguy Patches: 20080319__bug12252.diff.txt uploaded by Corydon76 (license 14) Tested by: callguy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113349 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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bb13bf705e |
Merged revisions 113013 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r113013 | jpeeler | 2008-04-07 10:18:10 -0500 (Mon, 07 Apr 2008) | 15 lines Merged revisions 113012 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008) | 7 lines (closes issue #12362) (closes issue #12372) Reported by: vinsik Tested by: tecnoxarxa This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113241 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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63f574ceb4 |
Merged revisions 113118 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113118 | qwell | 2008-04-07 13:00:09 -0500 (Mon, 07 Apr 2008) | 8 lines Allow playback with noanswer (and add earlyrtp option). (closes issue #9077) Reported by: pj Patches: earlyrtp.diff uploaded by wedhorn (license 30) Tested by: pj, qwell, DEA, wedhorn ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113119 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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566e073606 |
Merged revisions 113012 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008) | 7 lines (closes issue #12362) (closes issue #12372) Reported by: vinsik Tested by: tecnoxarxa This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113013 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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f291c2af0a |
Found a little problem with the sip request handling that could lead to a quick crash of asterisk, and a road to a DOS attack if left unfixed.
Attaching to a running asterisk with "telnet hostname 5060", I would input "something", then hit return three times, and asterisk crashes. I traced it to handle_request_do(), which zeroes out the data (an ast_str ptr) if the string is too short. Instead of freeing the struct and nulling the pointer, it now just resets it, because this ast_str is expected by the calling routine to still be there after handle_request_do() returns. This appears to fix the crash. I assume that it was introduced with ast_str's being adopted. It's a subtle and easy-to-miss sort of problem. I also found all the places where the req.data is freed, and made sure the ptr is Nulled out as well; no good leaving bad ptrs laying around-- I didn't need to do this, but it seemed a good thing to do... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112874 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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71dc6a4771 |
Merged revisions 112820 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112820 | phsultan | 2008-04-04 21:26:15 +0200 (Fri, 04 Apr 2008) | 1 line Free newly allocated channel before returning ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112821 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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db884798db |
Merged revisions 112766 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112766 | phsultan | 2008-04-04 19:16:59 +0200 (Fri, 04 Apr 2008) | 7 lines Prevent call connections when codecs don't match. (closes issue #10604) Reported by: keepitcool Patches: branch-1.4-10604-2.diff uploaded by phsultan (license 73) Tested by: phsultan ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112785 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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3fd8236d28 |
Merged revisions 112599 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112599 | mmichelson | 2008-04-03 09:32:20 -0500 (Thu, 03 Apr 2008) | 9 lines Fix the testing of the "res" variable so that it is more logically correct and makes the correct warning and debug messages print. (closes issue #12361) Reported by: one47 Patches: chan_zap_deferred_digit.patch uploaded by one47 (license 23) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112600 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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cbf80c1a3c |
Make MISDN generate channel rename events when the name changes.
(closes issue #11142) Reported by: julianjm Patches: chan_misdn_tmpchan_trunk_v1.diff uploaded by julianjm (license 99) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112520 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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b5cccfe1a4 |
Since the SIP request structure gets reused multiple times with TCP handling we have to clear the debug state or else we will keep spitting out debug even after it has been turned off.
(closes issue #12169) Reported by: pj Patches: 12169-debugoff-2.diff uploaded by qwell (license 4) Tested by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112431 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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6699761f80 |
Added dnsmgr status output for sip show registry.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112360 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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094fc2c616 |
Fix a typo that prevented configuration of non-dynamic peers.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112351 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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e9825d7c8a |
Existing DNS manager lookups extended to check for SRV records.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112321 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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d751947b1a |
Fix last commit
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112234 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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a5cdd849e5 |
This adds DNS SRV record support to DNS manager. If there is a SRV record for a given domain, the hostname and port listed in the SRV record will be used. If no SRV record exists or a SRV lookup is not attempted, the DNS lookup on the specified domain will be performed as normal. Chan_sip has been modified to take advantage of the new SRV support.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112207 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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a8be22f9da |
Merged revisions 112204 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112204 | file | 2008-04-01 14:43:46 -0300 (Tue, 01 Apr 2008) | 4 lines Do not pass audio until the remote side has indicated they are providing early media, or if the channel has been answered. (closes issue #11823) Reported by: SDamm ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112205 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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dcf4e46d8f |
Demote a log message down to a warning.
(closes issue #12345) Reported by: caio1982 Patches: limit_msg.diff uploaded by caio1982 (license 22) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112155 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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af9c1ee0df |
Now that zaptel trunk has been removed, add the PSTN deprecation notice to chan_zap, as well.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112124 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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652ce60a6f |
I missed a place when this define was changed.
(closes issue #12334) Reported by: ovi Patches: 12334-asterisk.patch uploaded by dimas (license 88) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112071 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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76baf34555 |
This fixes a high fence violation that MALLOC_DEBUG reported to me.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111996 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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bf4893fdce |
This time the fix is proper for issue 12284. I have tested it thoroughly and found
that valgrind no longer complains and that calls do complete correctly. The fix is along the same lines as before: Make sure the final null terminator gets copied into the new sip_request's data pointer. Without it, parse_request will read and potentially write past the end of the string, causing potential crashes. (closes issue #12284...for real this time!) reported by falves11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111811 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |