1750 Commits (f08ab8278c86d084d4d98bef164bb0090dcba674)

Author	SHA1	Message	Date
Sean Bright	7add06a4d6	Fix a problem found while building res_snmp. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154919 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	0d25ddd366	Add LISTFILTER dialplan function, along with supporting documentation. See ... documentation for more information on how to use it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154915 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Fredrickson	5250201d8b	Make compilation of chan_dahdi so that it does not require the new pri_progress_with_cause function to have libpri support work. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154875 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	086a52d9d1	Introduce a new API call ast_channel_search_locked, which iterates through the ... channel list calling a caller-defined callback. The callback returns non-zero if a match is found. This should speed up some of the code that I committed earlier today in chan_sip (which is also updated by this commit). Reviewed by russellb and kpfleming via ReviewBoard: http://reviewboard.digium.com/r/28/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154429 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	2cc8e25222	Slightly optimize ast_devstate_str and rename global functions devstate2str and config_text_file_save to have an ast_ prefix ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154260 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	a67790c6f5	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/trunk@153709 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	585899dbc0	Merged revisions 153651 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r153651 | russell | 2008-11-02 13:51:17 -0600 (Sun, 02 Nov 2008) | 2 lines features.h depends on linkedlists.h, so include it ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153652 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	5b168ee34b	Merge changes from team/group/appdocsxml ... This commit introduces the first phase of an effort to manage documentation of the interfaces in Asterisk in an XML format. Currently, a new format is available for applications and dialplan functions. A good number of conversions to the new format are also included. For more information, see the following message to asterisk-dev: http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153365 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	d521ad9696	* Fixed timeout logic in the dialing API as setting timeouts ... had no effect * Updated dialing API documentation to indicate that timeouts are specified in milliseconds * Added a new timeout argument to the Page application. If time expires, any endpoints which have not answered will be hung up. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	5fe37e47c6	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/trunk@153181 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	d61cc3e1f6	Add a todo for a new timing API implementation that would work for Linux systems ... as of kernel 2.6.25 and glibc 2.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152990 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	694dd34413	Fix a bug in AST_SCHED_REPLACE_UNREF(). The reference count of the object ... _must_ be increased before creating the scheduler entry. Otherwise, you create a race condition where the reference count may hit zero and the object can disappear out from under you. This could also would have incorrectly decreased the reference count in the case that the scheduler add failed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152887 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	4c83309c54	try to get this committed before the buildbot complains about a broken tree ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152810 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	6fad66dfb3	Merged revisions 152535 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines The magic trick to avoid this crash is not to try to find the channel by name in the list, which is slow and resource consuming, but rather to pay attention to the result codes from the ast_bridge_call, to which I added the AST_PBX_NO_HANGUP_PEER_PARKED value, which now are returned when a channel is parked. Why? because CDR's aren't generated via parking, so nothing is needed, but if a transfer occurred, there are critical things I need. If you get AST_PBX_KEEPALIVE, then don't touch the channel pointer. If you get AST_PBX_NO_HANGUP_PEER, or AST_PBX_NO_HANGUP_PEER_PARKED, then don't touch the peer pointer. Updated the several places where the results from a bridge were not being properly obeyed, and fixed some code I had introduced so that the results of the bridge were not overridden (in trunk). All the places that previously tested for AST_PBX_NO_HANGUP_PEER now have to check for both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED. I tested this against the 4 common parking scenarios: 1. A calls B; B answers; A parks B; B hangs up while A is getting the parking slot announcement, immediately after being put on hold. 2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but before the park times out. 3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold. 4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out. No crash. I also ran the scenarios above against valgrind, and accesses looked good. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152536 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	1ddc834b39	cleaup of the TCP/TLS socket API: ... 1) rename 'struct server_args' to 'struct ast_tcptls_session_args', to follow coding guidelines 2) make ast_make_file_from_fd() static and rename it to something that indicates what it really is for (again coding guidelines) 3) rename address variables inside 'struct ast_tcptls_session_args' to be more descriptive (dare i say it... coding guidelines) 4) change ast_tcptls_client_start() to use the new 'remote_address' field of the session args for the destination of the connection, and use the 'local_address' field to bind() the socket to the proper source address, if one is supplied 5) in chan_sip, ensure that we pass in the PP address we are bound to when creating outbound (client) connections, so that our connections will appear from the correct address git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@151101 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jason Parker	ae0a736353	Merge codec_consistency branch. This should make sample usage much happier. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150729 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	b3bb9564d3	Fix the FRACK! warnings in chan_iax2 when POKE/LAGRQ packets are not answered. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	29a8fe20c8	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/trunk@149205 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	d5837ba8c2	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/trunk@149199 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	c1351ad237	Merge realtime_update2 branch, which adds a new realtime API call named ... 'update2', which permits updates which match across multiple columns, instead of requiring all tables to have a single unique identifier. All of the other API calls with the exception of 'update' already had the ability to match on multiple fields, so it was a missing and very desireable feature that an API call implementing an update should have this, too. This does not change any outward performance of Asterisk, but it should make life easier for application developers who use the RealTime framework. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148570 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	1dedb785ab	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/trunk@148200 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	3ed062f810	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/trunk@147899 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	e235a07376	(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/trunk@147807 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	9335b3ad34	Allow people to select the old console behavior of white text on a black ... background, by using the startup flag '-B'. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	eaa1b73fcf	Update documentation; AST_THREADSTORAGE() in trunk only takes a single ... argument. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146928 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	4560279c69	All ODBC parts can now use either unixodbc or iodbc. ... This allows for the ODBC parts to work on OpenBSD as well. 99.99% of the work is done by seanbright (bow, bow) and I actually did nothing but test and yell at him that it still didn't work :) Thanks for helping out ! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146925 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	2ec290b09d	Similar to r143204, masquerade the channel in the case of Park being called from AGI. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146923 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	623bc9d82a	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/trunk@146920 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	cd4829706a	make aescrypt.c compile on OpenBSD again ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146807 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	529874de7b	Add schedule extensions to app_meetme. In addition, the reporter found a ... problem within strptime(3), which we are correcting here with ast_strptime(). (closes issue #11040) Reported by: DEA Patches: 20080910__bug11040.diff.txt uploaded by Corydon76 (license 14) Tested by: DEA git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145649 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	fa2f4776a2	fix bugs caused by r144949 when MALLOC_DEBUG is defined ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@144950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	629861a705	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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@144949 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	38028fa641	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/trunk@144523 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	08af5bb312	Create a new config file status, CONFIG_STATUS_FILEINVALID for differentiating ... when a file is invalid from when a file is missing. This is most important when we have two configuration files. Consider the following example: Old system: sip.conf users.conf Old result New result ======== ========== ========== ========== Missing Missing SIP doesn't load SIP doesn't load Missing OK SIP doesn't load SIP doesn't load Missing Invalid SIP doesn't load SIP doesn't load OK Missing SIP loads SIP loads OK OK SIP loads SIP loads OK Invalid SIP loads incompletely SIP doesn't load Invalid Missing SIP doesn't load SIP doesn't load Invalid OK SIP doesn't load SIP doesn't load Invalid Invalid SIP doesn't load SIP doesn't load So in the case when users.conf doesn't load because there's a typo that disrupts the syntax, we may only partially load users, instead of failing with an error, which may cause some calls not to get processed. Worse yet, the old system would do this with no indication that anything was even wrong. (closes issue #10690) Reported by: dtyoo Patches: 20080716__bug10690.diff.txt uploaded by Corydon76 (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142992 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	67f7ac0499	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/trunk@142676 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Bradley Latus	7eb7696239	Minor fix to doco ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142000 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	8953b0f359	(closes issue #13366) ... Reported by: erousseau This was a reasonable enhancement request, which was easy to implement. Since it's an enhancement, it could only be applied to trunk. Basically, for accounting where "initiated" seconds are billed for, if the microseconds field on the end time is greater than the microseconds field for the answer time, add one second to the billsec field. The implementation was requested by erousseau, and I've implemented it as requested. I've updated the CHANGES, the cdr.conf.sample, and the .h files accordingly, to accept and set a flag for the corresponding new option. cdr.c adds in the extra second based on the usec fields if the option is set. Tested, seems to be working fine. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@140057 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	6dd5b8609f	Optional light colored background, for those who use black on white terminals. ... (closes issue #13306) Reported by: Corydon76 Patches: 20080814__bug13306__3.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, pkempgen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@139981 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	c0754e89ad	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/trunk@139554 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	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
Tilghman Lesher	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
Steve Murphy	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
Kevin P. Fleming	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
Sean Bright	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
Tilghman Lesher	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
Mark Michelson	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
Steve Murphy	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
Tilghman Lesher	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
Kevin P. Fleming	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
Kevin P. Fleming	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