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${ noResults }
1724 Commits (c0a70cbb31edef6b50b2141044c847e966a8c8c7)
Author | SHA1 | Message | Date |
---|---|---|---|
|
fdd9a6f9b6 |
Merged revisions 149205 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r149205 | mmichelson | 2008-10-14 18:04:44 -0500 (Tue, 14 Oct 2008) | 20 lines Merged revisions 149204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149204 | mmichelson | 2008-10-14 18:00:01 -0500 (Tue, 14 Oct 2008) | 12 lines Add a tolerance period for sync-triggered audiohooks so that if packetization of audio is close (but not equal) we don't end up flushing the audiohooks over small inconsistencies in synchronization. Related to issue #13005, and solves the issue for most people who were experiencing the problem. However, a small number of people are still experiencing the problem on long calls, so I am not closing the issue yet ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@149206 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
e65c411c8f |
Merged revisions 149199 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r149199 | tilghman | 2008-10-14 17:38:06 -0500 (Tue, 14 Oct 2008) | 8 lines Add additional memory debugging to several core APIs, and fix several memory leaks found with these changes. (Closes issue #13505, closes issue #13543) Reported by: mav3rick, triccyx Patches: 20081001__bug13505.diff.txt uploaded by Corydon76 (license 14) Tested by: mav3rick, triccyx ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@149202 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
5f980bc229 |
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/1.6.1@148240 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
df779dd50c |
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/1.6.1@147901 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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2c1bfe7643 |
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/1.6.1@147811 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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69eb9d08ac |
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/1.6.1@146930 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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595d37e746 |
Merged revisions 146923 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r146923 | jpeeler | 2008-10-06 18:08:21 -0500 (Mon, 06 Oct 2008) | 3 lines Similar to r143204, masquerade the channel in the case of Park being called from AGI. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@146924 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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54fda83a25 |
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/1.6.1@146921 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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a22f5b3bf0 |
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/1.6.1@146874 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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bc5b69d250 |
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/1.6.1@144993 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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7c3b7122f3 |
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/1.6.1@144529 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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bc73329607 |
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/1.6.1@142678 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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8fedd0288d |
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/1.6.1@139556 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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7a636521b1 |
Fix this again so we can compile with shadow warnings enabled and IMAP chosen
in voicemail. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137112 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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b7571f835d |
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/trunk@136947 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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a40f1cc1c5 |
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/trunk@136746 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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a67af1e018 |
Merged revisions 136541 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136542 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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4fb07fb0c1 |
Merge in a few more changes. This time the include/ directory.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136402 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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29228a3afc |
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/trunk@135900 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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ff101d0b07 |
Add '+=' append operator to configuration files.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135717 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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f24d7a89f5 |
datastore inheritance is a channel feature, so move this definition back
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135681 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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aca394bf0c |
HTTP module memory leaks
(closes issue #13230) Reported by: eliel Patches: res_http_post_leak.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135476 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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6cf6d9eca5 |
Merge in changes that allow Asterisk to be built against the Hoard
memory allocator. See doc/hoard.txt for more details. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135405 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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c95460a353 |
Oops, wrong define
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134703 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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63fb8d794b |
Modify the main page of the doxygen documentation to link to a new page dedicated
to Asterisk licensing information. The licensing page includes the Asterisk license, as well as a (not yet complete) list of 3rd party libraries that may be used, as well as what license we receive them under. Help filling out this list in the format that I have started in doxyref.h would be much appreciated. :) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@133575 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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ed6323cb73 |
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/trunk@133171 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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f910cfc444 |
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/trunk@132964 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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8115a6a9bf |
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/trunk@132643 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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7c5d38ed02 |
(Step 2 of 2)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132511 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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0ecc7e302d |
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/trunk@132510 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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c87f901cfd |
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/trunk@132390 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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7575be9da1 |
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/trunk@131986 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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3fa9ad3d13 |
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/trunk@131982 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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9a08061ea3 |
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/trunk@131923 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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b46ad8b190 |
(closes issue #13089)
Reported by: murf Most of this bug was already fixed by Tilghman before I opened it; Many thanks to Tilghman for his fix in svn version 125794. That fix cleared up some of the fields in the lock_info. This commit changes the address that is stored for the lock in the lock_info struct, so that it is the same as that passed into the locking macros. This makes searching for a lock_info (as in log_show_lock()) by its lock addr possible. The lock_addr field is infinitely more useful if it is the same as what is 'publicly' available outside the lock_info code. Many thanks to kpfleming, putnopvut, and Russell for their invaluable insights earlier today. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131570 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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28534ea921 |
Swap "static" and "const", so that "static" appears at the beginning of each
declaration (suppresses a warning). (closes issue #13070) Reported by: gknispel_proformatique Patches: asterisk_trunk_const_static.patch uploaded by gknispel (license 261) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130697 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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bead8cd6f0 |
Add some debug code and add a missing release
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130232 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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b968349e19 |
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/trunk@130040 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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65710485e4 |
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/trunk@129987 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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4ff527903e |
Code wasn't ready to be merged - see -dev list discussion
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129307 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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da03cdd174 |
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/trunk@129152 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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6f400edeab |
Changing name of global api call to ast_*
My mistake, pointed out by Russell. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128378 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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45e79490ba |
Implement flags for AGI in the channel structure so taht "show channels" and
AMI commands can display that a channel is under control of an AGI. Work inspired by work at customer site, but paid for by Edvina AB git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128240 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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0a52297cf0 |
Add new SIP cli command "sip show channelstats" that displays some QoS data (if we have RTCP reports
and not use the p2p rtp bridge). I could not find a way to detect us using the p2p bridge, which would be nice. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128197 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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12e5c68622 |
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/trunk@128027 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 |