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${ noResults }
5096 Commits (e276ccea11ce68ae151a2de5ab364f1343008a52)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
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 |
18 years ago |
|
|
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 |
18 years ago |
|
|
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 |
18 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 |
18 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 |
18 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 |
18 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 |
18 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 |
18 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 |
18 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 |
18 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 |
18 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 |
18 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 |
18 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 |
18 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 |
18 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 |
18 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 |
18 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 |
18 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 |
18 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 |
18 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 |
18 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 |
18 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 |
18 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 |
18 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 |
18 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 |
18 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 |
18 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 |
18 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 |
18 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 |
18 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 |
18 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 |
18 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 |
18 years ago |
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3a0f4cc933 |
Temporary revert of 111662. It's causing lots of trouble and appears to not be
the proper solution to the problem reported anyway. (related to issue #12884) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111777 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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5591c696a5 |
Merged revisions 111720 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111720 | qwell | 2008-03-28 12:55:05 -0500 (Fri, 28 Mar 2008) | 1 line Remove unimplemented softkeys. Prompted by issue #12325. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111721 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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ca8e44c051 |
The copy_request function did not take into account the necessary null terminator
for the string to be copied into. This resulted in parse_request reading invalid memory beyond the end of the string, and in some cases led to crashes. Thanks to falves11 for providing the valgrind output which led to the closure of this issue. (closes issue #12284) Reported by: falves11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111662 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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bdf4443586 |
Oops, missed one
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111185 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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438361c0b8 |
Add expiry value to the sip show subscriptions CLI command.
(closes issue #12025) Reported by: agx git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111083 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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6412a96e43 |
Large cleanup of DSP code
Per comments from dimas: 1. The code now generates DTMF_BEGIN frames in addition to DTMF_END ones. 2. "quelching" rewritten - now each detector (MF/DTMF/generic tone) may mark fragment of a frame for suppression (squelching, muting) with a call to mute_fragment. Actual muting happens only once at the very end of ast_dsp_process where all marked fragments are zeroed. This way every detector sees original data in the frame without any piece of a frame being zeroed by a detector which was run before. 3. DTMF detector tries to "mute" one block before and one block after the block where actual tone was detected. Muting of previois block is something new for this patch. Obviously this operation is not always possible - if current frame does not contain data for previous block - it is too late. But at least we make our best. Muting of next block was already done by the old code but it only affects part of the next block which is in the frame being processed. New code keeps this information in state structures so it will mute proper number of samples in the next frame(s) too. 4. Removed ast_dsp_digitdetect and ast_dsp_getdigits APIs because these are not used. 5. DSP API extended a bit - ast_dsp_was_muted() function added which returns true if DSP code was muting any fragment in the last frame. chan_zap uses this function to decide it needs to turn on confmute on the channel. This is to replace AST_FRAME_DTMF 'm'/'u' (mute/unmute) functionality. (closes issue #11968) Reported by: dimas Patches: v2-11968-dsp.patch uploaded by dimas (license 88) v4-11968-zap.patch uploaded by dimas (license 88) Tested by: dimas, qwell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111022 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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a3d7dc8903 |
Merged revisions 111020 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111020 | file | 2008-03-26 16:04:35 -0300 (Wed, 26 Mar 2008) | 4 lines If we are requested to authenticate a reinvite make sure that it contains T38 SDP if need be. (closes issue #11995) Reported by: fall ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111021 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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febd162ed2 |
Merged revisions 110628 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 lines Add an option (transmit_silence) which transmits silence during both Record() and DTMF generation. The reason this is an option is that in order to transmit silence we have to setup a translation path. This may not be needed/wanted in all cases. (closes issue #10058) Reported by: tracinet ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111017 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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ef4eff9a9b |
Add the "config reload <conffile>" command, which allows you to tell Asterisk
to reload any file that references a given configuration file. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111012 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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13787bc595 |
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@110726 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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a49b6591f5 |
Oops here too. I need to stop coding for a while...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110639 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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67efba6e50 |
Merged revisions 110635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110635 | mmichelson | 2008-03-25 10:40:33 -0500 (Tue, 25 Mar 2008) | 7 lines When reverting a commit, I accidentally left in this bit which was an experiment to see what would happen. It passed the compile test, and I didn't notice I had left this change in too. So this is a revert of a revert...sort of. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110636 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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738e4ec94e |
Add a special dialplan variable to chan_sip which will cause an audio file to be played upon completion of an attended transfer.
(closes issue #9239) Reported by: sunder git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110631 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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676d9d3303 |
Use the "Server" header when responding to SIP requests.
(closes issue #12278) Reported by: rjain Patches: chan_sip.c.diff uploaded by rjain (license 226) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110625 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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c05501d812 |
Remove the "Event: registration" header from Asterisk-generated
SIP REGISTER requests. rjain points out that RFC 3265 specifies that the Event: header is not a valid header for REGISTER requests and that the "registration" value is not defined at IANA. (closes issue #12279) Reported by: rjain Patches: chan_sip.c.diff uploaded by rjain (license 226) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110621 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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625f6bd203 |
Merged revisions 110618 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110618 | mmichelson | 2008-03-24 14:17:41 -0500 (Mon, 24 Mar 2008) | 15 lines This is a revert for revision 108288. The reason is that that revision was not for an actual bug fix per se, and so it really should not have been in 1.4 in the first place. Plus, people who compile with DO_CRASH are more likely to encounter a crash due to this change. While I think the usage of DO_CRASH in ast_sched_del is a bit absurd, this sort of change is beyond the scope of 1.4 and should be done instead in a developer branch based on trunk so that all scheduler functions are fixed at once. I also am reverting the change to trunk and 1.6 since they also suffer from the DO_CRASH potential. (closes issue #12272) Reported by: qq12345 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110619 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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c087390452 |
Merged revisions 110614 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110614 | russell | 2008-03-24 12:34:56 -0500 (Mon, 24 Mar 2008) | 2 lines Turn a NOTICE into a DEBUG message. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110615 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |