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${ noResults }
242 Commits (79d555f70c7a1c4d7239cbc678a34b5776503afc)
Author | SHA1 | Message | Date |
---|---|---|---|
|
79d555f70c |
Merged revisions 163579 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r163579 | file | 2008-12-12 12:55:15 -0400 (Fri, 12 Dec 2008) | 4 lines Since chan_sip is callback devicestate driven do not pass in actual states, pass in unknown so we get asked. Additionally do not pass in an actual device state value in ast_setstate since the channel may be callback driven. (closes issue #13525) Reported by: pj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@163581 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
85ad905561 |
Merged revisions 163449 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r163449 | russell | 2008-12-12 07:55:30 -0600 (Fri, 12 Dec 2008) | 34 lines Merged revisions 163448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163448 | russell | 2008-12-12 07:44:08 -0600 (Fri, 12 Dec 2008) | 26 lines Resolve issues that could cause DTMF to be processed out of order. These changes come from team/russell/issue_12658 1) Change autoservice to put digits on the head of the channel's frame readq instead of the tail. If there were frames on the readq that autoservice had not yet read, the previous code would have resulted in out of order processing. This required a new API call to queue a frame to the head of the queue instead of the tail. 2) Change up the processing of DTMF in ast_read(). Some of the problems were the result of having two sources of pending DTMF frames. There was the dtmfq and the more generic readq. Both were used for pending DTMF in various scenarios. Simplifying things to only use the frame readq avoids some of the problems. 3) Fix a bug where a DTMF END frame could get passed through when it shouldn't have. If code set END_DTMF_ONLY in the middle of digit emulation, and a digit arrived before emulation was complete, digits would get processed out of order. (closes issue #12658) Reported by: dimas Tested by: russell, file Review: http://reviewboard.digium.com/r/85/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@163515 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
39a8f051b7 |
Merged revisions 163171 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r163171 | russell | 2008-12-11 14:07:47 -0600 (Thu, 11 Dec 2008) | 16 lines Fix the "failed" extension for outgoing calls. The conversion to use ast_check_hangup() everywhere instead of checking the softhangup flag directly introduced this problem. The issue is that ast_check_hangup() checked for tech_pvt to be NULL. Unfortunately, this will be NULL is some valid circumstances, such as with a dummy channel. The fix is simple. Don't check tech_pvt. It's pointless, because the code path that sets this to NULL is when the channel hangup callback gets called. This happens inside of ast_hangup(), which is the same function responsible for freeing the channel. Any code calling ast_check_hangup() better not be calling it after that point, and if so, we have a bigger problem at hand. (closes issue #14035) Reported by: erogoza ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@163173 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
9edc58f39e |
Merged revisions 154687 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r154687 | murf | 2008-11-05 09:11:11 -0700 (Wed, 05 Nov 2008) | 9 lines Merged revisions 154685 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154685 | murf | 2008-11-05 09:06:53 -0700 (Wed, 05 Nov 2008) | 1 line This fix was prompted by communication from user, who was seeing thousands of error logs... looks like EAGAIN. Made such uninteresting. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@154690 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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1036849a42 |
import gcc 4.3.2 warning fixes from trunk, with a few changes specific to this branch
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@153710 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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50c690fb59 |
Merged revisions 141949 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r141949 | russell | 2008-09-08 20:47:56 -0500 (Mon, 08 Sep 2008) | 9 lines Modify ast_answer() to not hold the channel lock while calling ast_safe_sleep() or when calling ast_waitfor(). These are inappropriate times to hold the channel lock. This is what has caused "could not get the channel lock" messages from chan_sip and has likely caused a negative impact on performance results of SIP in Asterisk 1.6. Thanks to file for pointing out this section of code. (closes issue #13287) (closes issue #13115) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@141951 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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aaff8a2a20 |
Merged revisions 141157 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r141157 | murf | 2008-09-05 08:18:43 -0600 (Fri, 05 Sep 2008) | 9 lines Merged revisions 141156 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141156 | murf | 2008-09-05 08:15:43 -0600 (Fri, 05 Sep 2008) | 1 line A small change to prevent double-posting of CDR's; thanks to Daniel Ferrer for bringing it to our attention ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@141159 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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08146f95bc |
Merged revisions 140692 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r140692 | murf | 2008-09-02 16:55:12 -0600 (Tue, 02 Sep 2008) | 13 lines Merged revisions 140690 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140690 | murf | 2008-09-02 16:40:13 -0600 (Tue, 02 Sep 2008) | 1 line After reconsidering, with respect to 13409, ast_cdr_detach should be OK, better in fact, than ast_cdr_free, which generates lots of useless warnings that will undoubtably generate complaints. Hmmm. It doesn't hush the useless warnings, but it does allow control of posting via the detach and post routines, for those possible situations, where you'd want to post single-channel cdrs. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@140695 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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0c8172a97c |
Merged revisions 140691 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r140691 | murf | 2008-09-02 16:50:59 -0600 (Tue, 02 Sep 2008) | 22 lines Merged revisions 140670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140670 | murf | 2008-09-02 16:15:57 -0600 (Tue, 02 Sep 2008) | 14 lines (closes issue #13409) Reported by: tomaso Patches: asterisk-1.6.0-rc2-cdrmemleak.patch uploaded by tomaso (license 564) I basically spent the day, verifying that this patch solves the problem, and doesn't hurt in non-problem cases. Why valgrind did not plainly reveal this leak absolutely mystifies and stuns me. Many, many thanks to tomaso for finding and providing the fix. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@140693 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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fdd92290af |
Convert deprecated routines to the new names.
(closes issue #13297) Reported by: snuffy Patches: bug13297_20080814.diff uploaded by snuffy (license 35) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137456 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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790fde68d9 |
Another batch of files from RSW. The remaining apps and a few more
files from main/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137089 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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b69c8e6ab5 |
Another big chunk of changes from the RSW branch. Bunch of stuff from main/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137082 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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9b5b8246c5 |
Fix a calculation error I had made in the poll. The poll
would reset to 500 ms every time a non-voice frame was received. The total time we poll should be 500 ms, so now we save the amount of time left after the poll returned and use that as our argument for the next call to poll git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136633 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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ed4e6bf52b |
Scrap the 500 ms delay when Asterisk auto-answers a channel.
Instead, poll the channel until receiving a voice frame. The cap on this poll is 500 ms. The optional delay is still allowable in the Answer() application, but the delay has been moved back to its original position, after the call to the channel's answer callback. The poll for the voice frame will not happen if a delay is specified when calling Answer(). (closes issue #12708) Reported by: kactus git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136631 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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700d4501b8 |
Merged revisions 135949 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135949 | tilghman | 2008-08-05 22:53:36 -0500 (Tue, 05 Aug 2008) | 4 lines Fix a longstanding bug in channel walking logic, and fix the explanation to make sense. (Closes issue #13124) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135950 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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89c2844242 |
Merged revisions 135841,135847,135850 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135841 | mmichelson | 2008-08-05 19:25:10 -0500 (Tue, 05 Aug 2008) | 27 lines Merging the issue11259 branch. The purpose of this branch was to take into account "burps" which could cause jitterbuffers to misbehave. One such example is if the L option to Dial() were used to inject audio into a bridged conversation at regular intervals. Since the audio here was not passed through the jitterbuffer, it would cause a gap in the jitterbuffer's timestamps which would cause a frames to be dropped for a brief period. Now ast_generic_bridge will empty and reset the jitterbuffer each time it is called. This causes injected audio to be handled properly. ast_generic_bridge also will empty and reset the jitterbuffer if it receives an AST_CONTROL_SRCUPDATE frame since the change in audio source could negatively affect the jitterbuffer. All of this was made possible by adding a new public API call to the abstract_jb called ast_jb_empty_and_reset. (closes issue #11259) Reported by: plack Tested by: putnopvut ........ r135847 | mmichelson | 2008-08-05 19:27:54 -0500 (Tue, 05 Aug 2008) | 4 lines Revert inadvertent changes to app_skel that occurred when I was testing for a memory leak ........ r135850 | mmichelson | 2008-08-05 19:29:54 -0500 (Tue, 05 Aug 2008) | 3 lines Remove properties that should not be here ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135851 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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5eaf8450d6 |
Merged revisions 135799 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135799 | murf | 2008-08-05 17:13:20 -0600 (Tue, 05 Aug 2008) | 34 lines (closes issue #12982) Reported by: bcnit Tested by: murf I discovered that also, in the previous bug fixes and changes, the cdr.conf 'unanswered' option is not being obeyed, so I fixed this. And, yes, there are two 'answer' times involved in this scenario, and I would agree with you, that the first answer time is the time that should appear in the CDR. (the second 'answer' time is the time that the bridge was begun). I made the necessary adjustments, recording the first answer time into the peer cdr, and then using that to override the bridge cdr's value. To get the 'unanswered' CDRs to appear, I purposely output them, using the dial cmd to mark them as DIALED (with a new flag), and outputting them if they bear that flag, and you are in the right mode. I also corrected one small mention of the Zap device to equally consider the dahdi device. I heavily tested 10-sec-wait macros in dial, and without the macro call; I tested hangups while the macro was running vs. letting the macro complete and the bridge form. Looks OK. Removed all the instrumentation and debug. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135821 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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7df8b8b848 |
make datastore creation and destruction a generic API since it is not really channel related, and add the ability to add/find/remove datastores to manager sessions
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135680 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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6291cd19bf |
remove remaining Zaptel references in various places
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134086 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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0c23159464 |
Deprecate *_device_state_* APIs in favor of *_devstate_* APIs
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@133860 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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c780a443bf |
Merged revisions 133649 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133649 | tilghman | 2008-07-25 12:19:39 -0500 (Fri, 25 Jul 2008) | 8 lines Fix some errant device states by making the devicestate API more strict in terms of the device argument (only without the unique identifier appended). (closes issue #12771) Reported by: davidw Patches: 20080717__bug12771.diff.txt uploaded by Corydon76 (license 14) Tested by: davidw, jvandal, murf ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@133665 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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d185405755 |
Janitor project to convert sizeof to ARRAY_LEN macro.
(closes issue #13002) Reported by: caio1982 Patches: janitor_arraylen5.diff uploaded by caio1982 (license 22) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129045 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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bc2cfb3e81 |
Merged revisions 127663 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r127663 | murf | 2008-07-02 18:16:25 -0600 (Wed, 02 Jul 2008) | 30 lines The CDRfix4/5/6 omnibus cdr fixes. (closes issue #10927) Reported by: murf Tested by: murf, deeperror (closes issue #12907) Reported by: falves11 Tested by: murf, falves11 (closes issue #11849) Reported by: greyvoip As to 11849, I think these changes fix the core problems brought up in that bug, but perhaps not the more global problems created by the limitations of CDR's themselves not being oriented around transfers. Reopen if necc, but bug reports are not the best medium for enhancement discussions. We need to start a second-generation CDR standardization effort to cover transfers. (closes issue #11093) Reported by: rossbeer Tested by: greyvoip, murf ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127793 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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37db658b1f |
Place the delay in __ast_answer prior to the channel-specific answer
callback. This change differs from commit 127113 in that now the channel is not set to AST_STATE_UP until after the answer callback. (closes issue #12924) Reported by: snyfer git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127157 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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8cdb1f7f41 |
change the process of inserting a delay into the ast_answer() path so that we don't tell the calling channel that it has been answered unutil after the delay; for a single-thread call this won't matter all, but for a dual-thread call (using chan_local) this may fix the problem in issue 12924
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127113 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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02b1317d0f |
- add get_max_rate timing API call
- change ast_settimeout() to honor max rate in edge cases of file playback (this will make some warning messages go away at the end of playing back a file) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125332 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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fd4a60c459 |
Merged revisions 125132 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125132 | kpfleming | 2008-06-25 17:21:30 -0500 (Wed, 25 Jun 2008) | 10 lines allow tonezone to live in a different place than DAHDI/Zaptel, since dahdi-tools and dahdi-linux are now separate packages and can be installed in different places don't include tonezone.h in dahdi_compat.h, because only a couple of modules need it get app_rpt building again after the DAHDI changes (closes issue #12911) Reported by: tzafrir ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125138 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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56654fc0f2 |
Merged revisions 123930 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123930 | tilghman | 2008-06-19 11:58:19 -0500 (Thu, 19 Jun 2008) | 5 lines Change informative messages to use the _multiple variant when multiple formats are possible. (Closes issue #12848) Reported by klaus3000 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123931 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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e27a98ce5a |
- Fix a typo in a timing API call
- Convert the last part of channel.c over to use the timing API. This would not have made a difference when using the dahdi timing module. I noticed it when trying to use another timing source. Oops. :) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122923 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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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 |
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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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1af7ea2df1 |
Merged revisions 122130 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122130 | tilghman | 2008-06-12 10:11:30 -0500 (Thu, 12 Jun 2008) | 4 lines Occasionally, the alertpipe loses its nonblocking status, so detect and correct that situation before it causes a deadlock. (Reported and tested by ctooley via #asterisk-dev) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122131 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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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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610faf1132 |
Merged revisions 121442 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r121442 | file | 2008-06-10 09:52:06 -0300 (Tue, 10 Jun 2008) | 4 lines Update BRIDGEPEER variable before we do a generic bridge in case we just broke out of a native bridge and fell through to generic. (closes issue #12815) Reported by: ramonpeek ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121444 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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85a631f767 |
arbitrary formatting change to test a mantis change
(closes issue #12824) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121285 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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0bee0c65b2 |
Minor formatting change to test a mantis change ...
(closes issue #12824) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121284 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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90cb28f89c |
Merged revisions 121280 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r121280 | russell | 2008-06-09 11:35:40 -0500 (Mon, 09 Jun 2008) | 10 lines Do not attempt to do emulation if an END digit is received and the length is less than the defined minimum digit length, and the other end only wants END digits (SIP INFO, for example). (closes issue #12778) Reported by: tsearle Patches: 12778.rev1.txt uploaded by russell (license 2) Tested by: tsearle ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121282 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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51051ce949 |
Add lock tracking for rwlocks. Previously, lock.h only had the ability to
hold tracking information for mutexes. Now, the "core show locks" output will output information about who is holding a rwlock when a thread is waiting on it. (closes issue #11279) Reported by: ys Patches: trunk_lock_utils.v8.diff uploaded by ys (license 281) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120064 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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f1e9371da8 |
- revert change to ast_queue_hangup and create ast_queue_hangup_with_cause
- make data member of the ast_frame struct a named union instead of a void Recently the ast_queue_hangup function got a new parameter, the hangupcause Feedback came in that this is no good and that instead a new function should be created. This I did. The hangupcause was stored in the seqno member of the ast_frame struct. This is not very elegant, and since there's already a data member that one should be used. Problem is, this member was a void *. Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone wants to store another type in there in the future. This commit is so massive, because all ast_frame.data uses have to be altered to ast_frame.data.data Thanks russellb and kpfleming for the feedback. (closes issue #12674) Reported by: mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117802 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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7d2d373d6f |
Minor formatting change to test a mantis change ...
(issue #12674) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117212 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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08f91c1192 |
Merged revisions 116463 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116463 | russell | 2008-05-14 16:32:00 -0500 (Wed, 14 May 2008) | 4 lines Add ast_assert(), which can be used to handle fatal errors. It is only compiled in if dev-mode is enabled, and only aborts if DO_CRASH is defined. (inspired by issue #12650) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116469 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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b6aef57619 |
Merged revisions 116088 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116088 | mmichelson | 2008-05-13 18:47:49 -0500 (Tue, 13 May 2008) | 12 lines A change to the way channel locks are handled when DEBUG_CHANNEL_LOCKS is defined. After debugging a deadlock, it was noticed that when DEBUG_CHANNEL_LOCKS is enabled in menuselect, the actual origin of channel locks is obscured by the fact that all channel locks appear to happen in the function ast_channel_lock(). This code change redefines ast_channel_lock to be a macro which maps to __ast_channel_lock(), which then relays the proper file name, line number, and function name information to the core lock functions so that this information will be displayed in the case that there is some sort of locking error or core show locks is issued. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116089 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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b5a127daac |
Modify TIMEOUT() to be accurate down to the millisecond.
(closes issue #10540) Reported by: spendergrass Patches: 20080417__bug10540.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115076 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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08e674bce0 |
Pass the hangup cause all the way to the calling app/channel.
(closes issue #11328) Reported by: rain Patches: 20071207__pass_cause_in_hangup_control_frame.diff.txt uploaded by Corydon76 (license 14) brought up-to-date to trunk by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114637 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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96638d7bb5 |
re-add a fix that got lost with a recent change
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114548 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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efb767e432 |
Convert several DEBUG logs into ast_debug.
(closes issue #12444) Reported by: IgorG Patches: channel_c_debug.diff uploaded by IgorG (license 20) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114131 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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0270776ca5 |
Merged revisions 114117 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114117 | mmichelson | 2008-04-14 12:41:03 -0500 (Mon, 14 Apr 2008) | 11 lines Increase the retry count when attempting to show channels. This apparently cleared an issue someone was seeing when attempting to show channels when the load was high. (closes issue #11667) Reported by: falves11 Patches: 11677.txt uploaded by russell (license 2) Tested by: falves11 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114118 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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9ddc843fbe |
Merged revisions 114106 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114106 | mmichelson | 2008-04-14 09:58:02 -0500 (Mon, 14 Apr 2008) | 5 lines Save a local copy of the generate callback prior to unlocking the channel in case the generate callback goes NULL on us after the channel is unlocked. Thanks to Russell for pointing this need out to me. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114107 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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be02a94138 |
Merged revisions 113065 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113065 | mmichelson | 2008-04-07 11:08:45 -0500 (Mon, 07 Apr 2008) | 13 lines This fix prevents a deadlock that was experienced in chan_local. There was deadlock prevention in place in chan_local, but it would not work in a specific case because the channel was recursively locked. By unlocking the channel prior to calling the generator's generate callback in ast_read_generator_actions(), we prevent the recursive locking, and therefore the deadlock. (closes issue #12307) Reported by: callguy Patches: 12307.patch uploaded by putnopvut (license 60) Tested by: callguy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113066 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 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 |
17 years ago |