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${ noResults }
1634 Commits (c74fff5ebc76c510ec255c52dbb37ad18c754385)
Author | SHA1 | Message | Date |
---|---|---|---|
|
c74fff5ebc |
Merged revisions 158540 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r158540 | russell | 2008-11-21 16:12:37 -0600 (Fri, 21 Nov 2008) | 10 lines Merged revisions 158539 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158539 | russell | 2008-11-21 16:05:55 -0600 (Fri, 21 Nov 2008) | 2 lines When compiling with DEBUG_THREADS, report the real file/func/line for ao2_lock/ao2_unlock ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@158542 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
ac0c9bbc4d |
Merged revisions 157706 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r157706 | kpfleming | 2008-11-19 06:42:19 -0600 (Wed, 19 Nov 2008) | 5 lines make some corrections to the ast_agi_register_multiple(), ast_agi_unregister_multiple() and ast_agi_fdprintf() API calls to be consistent with API guidelines also, move UPGRADE.txt to UPGRADE-1.6.txt and make the new UPGRADE.txt contain information about upgrading between Asterisk 1.6 releases ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@157738 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
35a7e1deca |
Merged revisions 157306 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r157306 | mmichelson | 2008-11-18 12:31:08 -0600 (Tue, 18 Nov 2008) | 20 lines Merged revisions 157305 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157305 | mmichelson | 2008-11-18 12:25:55 -0600 (Tue, 18 Nov 2008) | 12 lines Fix a crash in the end_bridge_callback of app_dial and app_followme which would occur at the end of an attended transfer. The error occurred because we initially stored a pointer to an ast_channel which then was hung up due to a masquerade. This commit adds a "fixup" callback to the bridge_config structure to allow for end_bridge_callback_data to be changed in the case that a new channel pointer is needed for the end_bridge_callback. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@157307 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
4fbe80cc63 |
This is the 1.6.0 version of revision 156883 of trunk.
This is different in that it preserves the case-sensitiveness of processing queues from configuration. closes issue #13703 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@156889 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 155554 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r155554 | seanbright | 2008-11-08 20:27:00 -0500 (Sat, 08 Nov 2008) | 14 lines Merged revisions 155553 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155553 | seanbright | 2008-11-08 20:08:07 -0500 (Sat, 08 Nov 2008) | 6 lines Use static functions here instead of nested ones. This requires a small change to the ast_bridge_config struct as well. To understand the reason for this change, see the following post: http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@155555 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merge revision 153709 from trunk
------------------------------------------------------------------------ r153709 | kpfleming | 2008-11-02 17:34:39 -0600 (Sun, 02 Nov 2008) | 3 lines instead of trying to forcibly load res_agi when app_stack is loaded (even if the administrator didn't want it loaded), use GCC weak symbols to determine whether it was loaded already or not; if it was loaded, then use it. ------------------------------------------------------------------------ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@153745 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 153181 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r153181 | twilson | 2008-10-31 13:55:33 -0500 (Fri, 31 Oct 2008) | 5 lines Recent CDR fixes moved execution of the 'h' exten into the bridging code, so variables that were set after ast_bridge_call was called would not show up in the 'h' exten. Added a callback function to handle setting variables, etc. from w/in the bridging code. Calls back into a nested function within the function calling ast_bridge_call (closes issue #13793) Reported by: greenfieldtech ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@153265 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 148200 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r148200 | seanbright | 2008-10-09 20:42:13 -0400 (Thu, 09 Oct 2008) | 12 lines Don't include logger.h in asterisk.h by default as it is causing problems building app_voicemail. Instead, include it where it is needed. This turned out to be a relatively minor issue because other headers include logger.h as well. Need to test -addons before merging this back to 1.6.0. (closes issue #13605) Reported by: tomo1657 Patches: 13605_seanbright.diff uploaded by seanbright (license 71) Tested by: mmichelson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@148204 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 147899 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r147899 | mvanbaak | 2008-10-09 19:48:53 +0200 (Thu, 09 Oct 2008) | 5 lines only include this for OpenBSD. At least FreeBSD is borked when including it (closes issue #13649) Reported by: ys ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@147900 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 147807 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r147807 | murf | 2008-10-09 08:17:33 -0600 (Thu, 09 Oct 2008) | 15 lines (closes issue #13557) Reported by: nickpeirson Patches: pbx.c.patch uploaded by nickpeirson (license 579) replace_bzero+bcopy.patch uploaded by nickpeirson (license 579) Tested by: nickpeirson, murf 1. replaced all refs to bzero and bcopy to memset and memmove instead. 2. added a note to the CODING-GUIDELINES 3. add two macros to asterisk.h to prevent bzero, bcopy from creeping back into the source 4. removed bzero from configure, configure.ac, autoconfig.h.in ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@147809 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 146928 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r146928 | tilghman | 2008-10-06 18:21:02 -0500 (Mon, 06 Oct 2008) | 3 lines Update documentation; AST_THREADSTORAGE() in trunk only takes a single argument. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@146929 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 146920 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r146920 | jpeeler | 2008-10-06 17:59:58 -0500 (Mon, 06 Oct 2008) | 2 lines Mvanbaak said this was needed to compile on OpenBSD, so put it in the OpenBSD section. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@146922 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 146807 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r146807 | mvanbaak | 2008-10-06 23:18:13 +0200 (Mon, 06 Oct 2008) | 2 lines make aescrypt.c compile on OpenBSD again ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@146873 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 144949-144951 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r144949 | kpfleming | 2008-09-27 10:52:56 -0500 (Sat, 27 Sep 2008) | 17 lines Merged revisions 144924-144925 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r144924 | kpfleming | 2008-09-27 10:00:48 -0500 (Sat, 27 Sep 2008) | 6 lines improve header inclusion process in a few small ways: - it is no longer necessary to forcibly include asterisk/autoconfig.h; every module already includes asterisk.h as its first header (even before system headers), which serves the same purpose - astmm.h is now included by asterisk.h when needed, instead of being forced by the Makefile; this means external modules will build properly against installed headers with MALLOC_DEBUG enabled - simplify the usage of some of these headers in the AEL-related stuff in the utils directory ........ r144925 | kpfleming | 2008-09-27 10:13:30 -0500 (Sat, 27 Sep 2008) | 2 lines fix some minor issues with rev 144924 ........ ................ r144950 | kpfleming | 2008-09-27 11:10:33 -0500 (Sat, 27 Sep 2008) | 2 lines fix bugs caused by r144949 when MALLOC_DEBUG is defined ................ r144951 | kpfleming | 2008-09-27 11:17:43 -0500 (Sat, 27 Sep 2008) | 1 line remove incorrect comment ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@144991 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 144523 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r144523 | murf | 2008-09-25 15:18:12 -0600 (Thu, 25 Sep 2008) | 13 lines I added a little verbage to hashtab for the hashtab_destroy func. It was pretty sparsely documented. This update fleshes out the pbx_lua module, to add the switch statements to the extensions in the extensions.lua file, as well as removing them when the module is unloaded. Many thanks to Matt Nicholson for his fine contribution! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@144524 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 142676 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r142676 | murf | 2008-09-11 22:50:48 -0600 (Thu, 11 Sep 2008) | 40 lines Merged revisions 142675 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142675 | murf | 2008-09-11 22:29:34 -0600 (Thu, 11 Sep 2008) | 29 lines Tested by: sergee, murf, chris-mac, andrew, KNK This is a "second attempt" to restore the previous "endbeforeh" behavior in 1.4 and up. In order to capture information concerning all the legs of transfers in all their infinite combinations, I was forced to this particular solution by a chain of logical necessities, the first being that I was not allowed to rewrite the CDR mechanism from the ground up! This change basically leaves the original machinery alone, which allows IVR and local channel type situations to generate CDR's as normal, but a channel flag can be set to suppress the normal running of the h exten. That flag would be set by the code that runs the h exten from the ast_bridge_call routine, to prevent the h exten from being run twice. Also, a flag in the ast_bridge_config struct passed into ast_bridge_call can be used to suppress the running of the h exten in that routine. This would happen, for instance, if you use the 'g' option in the Dial app. Running this routine 'early' allows not only the CDR() func to be used in the h extension for reading CDR variables, but also allows them to be modified before the CDR is posted to the backends. While I dearly hope that this patch overcomes all problems, and introduces no new problems, reality suggests that surely someone will have problems. In this case, please re-open 13251 (or 13289), and we'll see if we can't fix any remaining issues. ** trunk note: some code to suppress the h exten being run from app_queue was added; for the 'continue' option available only in trunk/1.6.x. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@142677 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 139554 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r139554 | mmichelson | 2008-08-22 14:45:41 -0500 (Fri, 22 Aug 2008) | 16 lines Merged revisions 139553 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139553 | mmichelson | 2008-08-22 14:45:19 -0500 (Fri, 22 Aug 2008) | 8 lines Fix compilation when DEBUG_THREAD_LOCALS is selected (closes issue #13298) Reported by: snuffy Patches: bug13298_20080822.diff uploaded by snuffy (license 35) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@139555 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 136947 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r136947 | tilghman | 2008-08-09 10:26:27 -0500 (Sat, 09 Aug 2008) | 18 lines Merged revisions 136946 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r136946 | tilghman | 2008-08-09 10:25:36 -0500 (Sat, 09 Aug 2008) | 10 lines Merged revisions 136945 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r136945 | tilghman | 2008-08-09 10:24:36 -0500 (Sat, 09 Aug 2008) | 2 lines Regression fixes for Solaris ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@136948 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 136746 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r136746 | murf | 2008-08-07 18:48:35 -0600 (Thu, 07 Aug 2008) | 40 lines Merged revisions 136726 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r136726 | murf | 2008-08-07 18:15:34 -0600 (Thu, 07 Aug 2008) | 32 lines (closes issue #13236) Reported by: korihor Wow, this one was a challenge! I regrouped and ran a new strategy for setting the ~~MACRO~~ value; I set it once per extension, up near the top. It is only set if there is a switch in the extension. So, I had to put in a chunk of code to detect a switch in the pval tree. I moved the code to insert the set of ~~exten~~ up to the beginning of the gen_prios routine, instead of down in the switch code. I learned that I have to push the detection of the switches down into the code, so everywhere I create a new exten in gen_prios, I make sure to pass onto it the values of the mother_exten first, and the exten next. I had to add a couple fields to the exten struct to accomplish this, in the ael_structs.h file. The checked field makes it so we don't repeat the switch search if it's been done. I also updated the regressions. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@136778 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 136542 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r136542 | kpfleming | 2008-08-07 12:44:20 -0500 (Thu, 07 Aug 2008) | 6 lines Merged revisions 136541 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@136543 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 135900 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r135900 | tilghman | 2008-08-05 22:04:01 -0500 (Tue, 05 Aug 2008) | 12 lines Merged revisions 135899 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135899 | tilghman | 2008-08-05 22:02:59 -0500 (Tue, 05 Aug 2008) | 4 lines 1) Bugfix for debugging code 2) Reduce compiler warnings for another section of debugging code (Closes issue #13237) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@135901 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 135851 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r135851 | mmichelson | 2008-08-05 19:30:53 -0500 (Tue, 05 Aug 2008) | 48 lines 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/branches/1.6.0@135852 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 135821 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r135821 | murf | 2008-08-05 17:45:32 -0600 (Tue, 05 Aug 2008) | 42 lines 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/branches/1.6.0@135822 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 134703 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r134703 | tilghman | 2008-07-30 17:38:58 -0500 (Wed, 30 Jul 2008) | 2 lines Oops, wrong define ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@134706 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 134050 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r134050 | mmichelson | 2008-07-28 11:00:19 -0500 (Mon, 28 Jul 2008) | 3 lines merging the zap_and_dahdi_trunk branch up to trunk ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@134052 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Include the licensing page in 1.6.0 as well. Now, this page exists in 1.4, trunk,
and 1.6.0. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@133982 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 133171 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r133171 | mmichelson | 2008-07-23 14:48:03 -0500 (Wed, 23 Jul 2008) | 20 lines Merged revisions 133169 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133169 | mmichelson | 2008-07-23 14:39:47 -0500 (Wed, 23 Jul 2008) | 12 lines As suggested by seanbright, the PSEUDO_CHAN_LEN in app_chanspy should be set at load time, not at compile time, since dahdi_chan_name is determined at load time. Also changed the next_unique_id_to_use to have the static qualifier. Also added the dahdi_chan_name_len variable so that strlen(dahdi_chan_name) isn't necessary. Thanks to seanbright for the suggestion. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@133172 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 132964 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r132964 | kpfleming | 2008-07-23 11:30:18 -0500 (Wed, 23 Jul 2008) | 10 lines Merged revisions 132872 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132872 | kpfleming | 2008-07-23 06:52:18 -0500 (Wed, 23 Jul 2008) | 2 lines minor optimization for stringfields: when a field is being set to a larger value than it currently contains and it happens to be the most recent field allocated from the currentl pool, it is possible to 'grow' it without having to waste the space it is currently using (or potentially even allocate a new pool) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@132965 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 132643 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r132643 | kpfleming | 2008-07-22 14:59:10 -0500 (Tue, 22 Jul 2008) | 10 lines Merged revisions 132641 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132641 | kpfleming | 2008-07-22 14:49:11 -0500 (Tue, 22 Jul 2008) | 2 lines use renamed libpri API call for controlling this feature (was improperly named before) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@132644 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 132511 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r132511 | tilghman | 2008-07-21 16:00:47 -0500 (Mon, 21 Jul 2008) | 2 lines (Step 2 of 2) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@132513 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 132510 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r132510 | tilghman | 2008-07-21 15:59:03 -0500 (Mon, 21 Jul 2008) | 5 lines Optionally build integer-based routines for FSK tone decoding (but default to the more accurate float-based routines). (Closes issue #11679) (Step 1 of 2) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@132512 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 132390 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r132390 | russell | 2008-07-21 09:47:41 -0500 (Mon, 21 Jul 2008) | 16 lines Remove libresample from the Asterisk source tree. It is now available in its own repository, and must be installed like any other library for Asterisk to use. The two modules that require it are codec_resample and app_jack. To install libresample: $ svn co http://svn.digium.com/svn/libresample/trunk libresample $ cd libresample $ ./configure $ make $ sudo make install This code is currently in our own repository because the build system did not include the appropriate targets for building a dynamic library or for installing the library. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@132391 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 131986 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r131986 | tilghman | 2008-07-18 11:48:18 -0500 (Fri, 18 Jul 2008) | 10 lines Merged revisions 131985 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131985 | tilghman | 2008-07-18 11:46:23 -0500 (Fri, 18 Jul 2008) | 2 lines Preserve ABI compatibility with last change ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@131987 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 131982 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r131982 | tilghman | 2008-07-18 11:33:56 -0500 (Fri, 18 Jul 2008) | 10 lines Merged revisions 131970 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131970 | tilghman | 2008-07-18 11:30:31 -0500 (Fri, 18 Jul 2008) | 2 lines Make the ast_assert call within ast_sched_del report something useful. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@131984 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 131923 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r131923 | kpfleming | 2008-07-18 11:16:12 -0500 (Fri, 18 Jul 2008) | 10 lines Merged revisions 131921 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131921 | kpfleming | 2008-07-18 11:15:41 -0500 (Fri, 18 Jul 2008) | 2 lines remove the dlfcn compatibility stuff, because no platforms that Asterisk currently runs on it use it, and it doesn't build anyway ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@131924 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 130145 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk Merging this rev from trunk to 1.6.0 was not simple. Why? Because we've enhanced trunk to do a [fast] merge-and-delete operation which also solved problems with contexts having entries from different registrars. Fast as in the amount of time the contexts are locked down. That *is* fast, but traversing the entire dialplan looking for priorities to delete takes more time overall. This particular fix involved pulling in those enhancements from trunk, along with all the various fixes and refinements made along the way. Merging all this from trunk into 1.6 involved: a. mergetrunk6 in the stuff from 130145; b. revert all but the prop changes c. catalog all revisions to pbx.c since 1.6.0 was forked (at rev 105596). d. catalog all revisions to pbx.c in trunk since 1.6.0 was forked, making special note of all revs that were not merged into 1.6.0. e. study each rev in trunk not applied to 1.6.0, and determine if it was involved in the merge_and_delete enhancements in trunk. 25 commits were done in 1.6.0, all but one (106306) was a merge from trunk. Trunk had 22 additional changes, of which 7 were involved in the merge_and_delete enhancements: 106757 108894 109169 116461 123358 130145 130297 f. Go to trunk and collect patches, one by one, of the changes made by each rev across the entire source tree, using svn diff -c <num> > pfile g. Apply each patch in order to 1.6.0, and resolve all failures and compilation problems before proceding to the next patch. h. test the stuff. i. profit! ........ r130145 | murf | 2008-07-11 12:24:31 -0600 (Fri, 11 Jul 2008) | 40 lines (closes issue #13041) Reported by: eliel Tested by: murf (closes issue #12960) Reported by: mnicholson In this 'omnibus' fix, I **think** I solved both the problem in 13041, where unloading pbx_ael.so caused crashes, or incomplete removal of previous registrar'ed entries. And I added code to completely remove all includes, switches, and ignorepats that had a matching registrar entry, which should appease 12960. I also added a lot of seemingly useless brackets around single statement if's, which helped debug so much that I'm leaving them there. I added a routine to check the correlation between the extension tree lists and the hashtab tables. It can be amazingly helpful when you have lots of dialplan stuff, and need to narrow down where a problem is occurring. It's ifdef'd out by default. I cleaned up the code around the new CIDmatch code. It was leaving hanging extens with bad ptrs, getting confused over which objects to remove, etc. I tightened up the code and changed the call to remove_exten in the merge_and_delete code. I added more conditions to check for empty context worthy of deletion. It's not empty if there are any includes, switches, or ignorepats present. If I've missed anything, please re-open this bug, and be prepared to supply example dialplan code. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@130946 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 129987 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r129987 | russell | 2008-07-11 09:22:44 -0500 (Fri, 11 Jul 2008) | 10 lines Merged revisions 129970 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129970 | russell | 2008-07-11 09:18:43 -0500 (Fri, 11 Jul 2008) | 2 lines add a simple ASTOBJ_TRYWRLOCK macro ... ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@130637 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 130040 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r130040 | kpfleming | 2008-07-11 10:57:17 -0500 (Fri, 11 Jul 2008) | 12 lines Merged revisions 130039 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130039 | kpfleming | 2008-07-11 10:41:56 -0500 (Fri, 11 Jul 2008) | 4 lines add support for a configuration parameter for 'inband audio during RELEASE', which is currently mandatory in libpri-1.4.4 but will become configurable in libpri-1.4.5 later today (related to issue #13042) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@130041 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 129152 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r129152 | tilghman | 2008-07-08 15:30:29 -0500 (Tue, 08 Jul 2008) | 16 lines Merged revisions 129149 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129149 | tilghman | 2008-07-08 15:27:47 -0500 (Tue, 08 Jul 2008) | 8 lines Cause SIP to return a 480 instead of a 404 when a sip peer exists, but is not registered. (closes issue #12885) Reported by: ibc Patches: 20080701__bug12885__2.diff.txt uploaded by Corydon76 (license 14) Tested by: ibc ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@129153 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 128027 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r128027 | tilghman | 2008-07-04 11:06:34 -0500 (Fri, 04 Jul 2008) | 16 lines Merged revisions 127973 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r127973 | tilghman | 2008-07-03 22:30:30 -0500 (Thu, 03 Jul 2008) | 8 lines Fix the 'dialplan remove extension' logic, so that it a) works with cidmatch, and b) completes contexts correctly when the extension is ambiguous. (closes issue #12980) Reported by: licedey Patches: 20080703__bug12980.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@128028 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 127793 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r127793 | murf | 2008-07-03 11:16:44 -0600 (Thu, 03 Jul 2008) | 38 lines 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/branches/1.6.0@127830 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 127767 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r127767 | kpfleming | 2008-07-03 11:22:02 -0500 (Thu, 03 Jul 2008) | 2 lines some minor fixes found while working on issue #12911 (and block the rev from 1.4 since the equivalent is already here) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@127778 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 126226,126513 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r126226 | seanbright | 2008-06-28 17:28:16 -0400 (Sat, 28 Jun 2008) | 8 lines Merge in changes from my cdr-tds-conversion branch. This changes the internal implementation from using the volatile libtds, to using the db-lib front end. The unintended side effect of this is that we support (at least) versions 0.62 through 0.82 of the FreeTDS distribution without any #ifdef ugliness. (closes issue #12844) Reported by: jcollie ........ r126513 | seanbright | 2008-06-30 07:57:42 -0400 (Mon, 30 Jun 2008) | 4 lines Cast a few more strings to char *, so that we can compile cleanly against FreeTDS 0.60. Update the docs to reflect that we can now compile and run against all modern releases of FreeTDS (0.60 through 0.82) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@127397 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 126574 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r126574 | russell | 2008-06-30 11:07:25 -0500 (Mon, 30 Jun 2008) | 18 lines Merged revisions 126573 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126573 | russell | 2008-06-30 11:05:08 -0500 (Mon, 30 Jun 2008) | 10 lines Fix a typo in the non-DEBUG_THREADS version of the recently added DEADLOCK_AVOIDANCE() macro. This caused the lock to not actually be released, and as a result, not avoid deadlocks at all. This resolves the issues reported in the last while about Asterisk locking up all over the place (and most commonly, in chan_iax2). (closes issue #12927) (closes issue #12940) (closes issue #12925) (potentially closes others ...) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@126575 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 125794 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r125794 | tilghman | 2008-06-27 08:54:13 -0500 (Fri, 27 Jun 2008) | 10 lines Merged revisions 125793 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125793 | tilghman | 2008-06-27 08:45:03 -0500 (Fri, 27 Jun 2008) | 2 lines In this debugging function, copy to a buffer instead of using potentially unsafe pointers. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@125795 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 125703 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r125703 | phsultan | 2008-06-27 09:28:17 +0200 (Fri, 27 Jun 2008) | 1 line Fix a compile time error that occurs if OpenSSL is not installed. Reported by Noel Morais on the users mailing list ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@125704 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 125138 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r125138 | kpfleming | 2008-06-25 18:05:28 -0500 (Wed, 25 Jun 2008) | 18 lines 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/branches/1.6.0@125146 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 124966 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r124966 | tilghman | 2008-06-24 20:08:37 -0500 (Tue, 24 Jun 2008) | 15 lines Merged revisions 124965 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124965 | tilghman | 2008-06-24 19:46:24 -0500 (Tue, 24 Jun 2008) | 7 lines Pvt deadlock causes some channels to get stuck in Reserved status. (closes issue #12621) Reported by: fabianoheringer Patches: 20080612__bug12621.diff.txt uploaded by Corydon76 (license 14) Tested by: fabianoheringer ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@124967 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 124278 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r124278 | mmichelson | 2008-06-20 11:30:18 -0500 (Fri, 20 Jun 2008) | 6 lines Change references to doc/channelvariables.txt to doc/tex/channelvariables.tex. This issue came up on the asterisk-dev mailing list. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@124279 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 123546 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r123546 | bbryant | 2008-06-17 16:46:57 -0500 (Tue, 17 Jun 2008) | 5 lines Updates all usages of ast_tcptls_session_instance to be managed by reference counts so that they only get destroyed when all threads are done using them, and memory does not get free'd causing strange issues with SIP. This code was originally written by russellb in the team/group/issue_11972/ branch. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@123547 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |