search for: app_confbridge

...:Currently asterisk core dumped as soon as app2 load . I want to load module as after join confbridge by any user , using MyConfbridgeCount() to know no of conference user count . file: app/app_confbrige.c ========================================================================== partial code of app_confbridge.c: ========================================================================= static const char *const app2 ="MyConfbridgeCount"; static int count_exec(struct ast_channel *chan, const char *data) { int res = 0; struct conference_bridge *conf=NULL; int count; char *localdata; char val[80]...

...ridge application, but it throws "Failed to find a bridge technology to satisfy capabilities 0x4 (ulaw)" error. Please see console output below. -- Executing [501 at services:9] ConfBridge("SIP/OpenSER-00000005", "1001") in new stack [May 19 13:36:05] DEBUG[7452]: app_confbridge.c:404 join_conference_bridge: Trying to find conference bridge '1001' [May 19 13:36:05] DEBUG[7452]: bridging.c:475 ast_bridge_new: Failed to find a bridge technology to satisfy capabilities 0x4 (ulaw) [May 19 13:36:05] DEBUG[7452]: app_confbridge.c:368 destroy_conference_bridge: Destroying...

...nfig] 2. MyConfbridgeCount(4000,count) [pbx_config] 3. verbose(3,"== ${count} ====") [pbx_config] issue:Currently asterisk core dumped as soon as app2 load . file: app/app_confbrige.c ===================================================================== partial code of app_confbridge.c: ==================================================================== static const char *const app2 ="MyConfbridgeCount"; static int load_module(void) { ast_verb(3 ,"==Inside load_module=="); ast_verb(3 ,"\n ==Inside load_module==\n &q...

...running on Asterisk 1.4.2 using MeetMe and it's been pretty much rock solid since it was installed. We do around 460,000 minutes on it monthly and peak at about 150 simultaneous sip channels. I'm adding a second bridge for redundancy purposes into another facility and would like to go with app_confbridge and Solaris. Since it's a relatively new app I have to question if it's been put to the test and whether I can expect the same kind of stability. Does anyone use confbridge in a large installation and can provide feedback on its stability, quality in comparison to MeetMe? I use a sango...

...ridge On Wed, May 6, 2009 at 7:38 AM, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote: >> Can someone please tell me in which file the code for the application to >> be found? I was not able to find a file named app_bridge.c in the folder >> apps. > > app_bridge.c ? app_confbridge.c ? What are you looking for, exactly? > > The apps folder: > > http://svn.digium.com/svn/asterisk/trunk/apps/ So I have a question with regard to app_confbridge, which provides an application called ConfBridge. I did find a bug here that discusses bridging and confbridge, but not re...

...:Currently asterisk core dumped as soon as app2 load . I want to load module as after join confbridge by any user , using MyConfbridgeCount() to know no of conference user count . file: app/app_confbrige.c ========================================================================== partial code of app_confbridge.c: ========================================================================= static const char *const app2 ="MyConfbridgeCount"; static int count_exec(struct ast_channel *chan, const char *data) { int res = 0; struct conference_bridge *conf=NULL; int count; char *localdata; char val[80]...

http://stackoverflow.com/questions/21015596/failed-to-get-160-samples-from-read-factory- asterisk-11-5-1-app-confbridge-c

...esent. The sample config looks like this: -- ;announce_only_user=yes ;Sets if the only user announcement should be played when a channel enters a empty conference. On by default. -- But - disabling it (announce_only_user=no) doesn't take effect. And looking at the source asterisk-11.8.1/apps/app_confbridge.c I can't even find this option. Any clues? Also - setting quiet=yes still plays join/leave sound. My current work-around is: sound_join=silence/1 sound_leave=silence/1 But this seems a bit ineffective... In Meetme the quiet-flag also disabled join/leave sounds. Is this by design or an ove...

...y bridge_ modules? Asterisk 1.6.2.0~rc2-0ubuntu1.2 -- Executing [521 at outbound:1] Answer("SIP/109-b877a8c8", "") in new stack -- Executing [521 at outbound:2] ConfBridge("SIP/109-b877a8c8", "conf") in new stack [Mar 19 03:16:33] ERROR[2294]: app_confbridge.c:434 join_conference_bridge: Conference bridge '521' could not be created. dial plan: exten => _52X,1,Answer() exten => _52X,n,ConfBridge(${EXTEN}) Thanks, Kelvin

...07 - iostreams: Potential DoS when client connection closed prematurely (Reported by Sean Bright) Bugs fixed in this release: ----------------------------------- * ASTERISK-27783 - res_pjsip_pubsub: apparent crash on shutdown (Reported by Kevin Harwell) * ASTERISK-27870 - app_confbridge: Conference bridge and announcer channels are not removed if conference is ended as soon as it starts (Reported by Robert Mordec) * ASTERISK-27943 - AMI: Action SendText needs to use the correct thread. (Reported by Richard Mudgett) * ASTERISK-27942 - res_pjsip_messa...

Hi Everyone. I was referred here by malcolmd of the Asterisk forums. What follows is a copy of this question: http://forums.asterisk.org/viewtopic.php?f=1&t=92007? I've recently upgraded from Asterisk 11 to Asterisk 13. Most of it went smoothly thanks to the documentation detailing how to upgrade to 12 and then how to upgrade to 13. The only thing that didn't work correctly was

...documentation that says > "MOH will no longer work in softmix, you should use simple". This has me > concerned that I've done something wrong elsewhere in my config that is > causing softmix to not work correctly.? > The bridging technology bridge_softmix is only used by app_confbridge in Asterisk v11. Nothing else in v11 uses the bridging framework. Unless you were using app_confbridge, you were not using bridge_softmix in v11. The various bridging technology modules in v12 and later are for different scenarios. The bridging framework is smart enough to pick the best bridging...

...cery_config_internal_load: Unable to load config file 'pjsip.conf' Jan 21 19:36:47 asterisk.riverside asterisk[1411]: [Jan 21 19:36:47] ERROR[1411]: config_options.c:710 aco_process_config: Unable to load config file 'confbridge.conf' asterisk[1411]: [Jan 21 19:36:47] ERROR[1411]: app_confbridge.c:4351 load_module: Unable to load config. Not loading module. asterisk[1411]: [Jan 21 19:36:47] WARNING[1411]: loader.c:2381 load_modules: Some non-required modules failed to load. asterisk[1411]: [Jan 21 19:36:47] ERROR[1411]: loader.c:2396 load_modules: res_stun_monitor declined to load. aste...

...9; 548 bytes in 2 allocations in file 'features.c' 80 bytes in 10 allocations in file 'channel.c' 324 bytes in 3 allocations in file 'cdr.c' 84 bytes in 4 allocations in file 'timing.c' 484 bytes in 1 allocations in file 'app_confbridge.c' 779 bytes in 13 allocations in file 'taskprocessor.c' 4752 bytes in 22 allocations in file 'file.c' 14656 bytes in 162 allocations in file 'sched.c' 2128 bytes in 79 allocations in file 'manager.c' 1400 bytes in 6 allocations...

...l 'nat' setting and do not set 'nat' per peer/user. !!! (config category='0000FFFF0002' global force_rport='No' peer/user force_rport='Yes') SIP channel loading... No IAX provisioning configuration found, IAX provisioning disabled. Adding default_menu menu to app_confbridge Failed to load configuration file. Module not activated. No mappings found in cel_custom.conf. Not logging CEL to custom CSVs. Failed to load configuration file. *CLI> Asterisk Ready. Another thing, everywhere in doc?s and references I see that I can use ?exit? on CLI. If I try to type ?exit? -...

...* ASTERISK-27818 - Username bruteforce is possible when using ACL with PJSIP (Reported by John) Bugs fixed in this release: ----------------------------------- * ASTERISK-27783 - res_pjsip_pubsub: apparent crash on shutdown (Reported by Kevin Harwell) * ASTERISK-27870 - app_confbridge: Conference bridge and announcer channels are not removed if conference is ended as soon as it starts (Reported by Robert Mordec) * ASTERISK-27909 - cdr: Deadlock with submit_scheduled_batch and submit_unscheduled_batch (Reported by Denis Lebedev) * ASTERISK-26987 -...

...in this release: New Features made in this release: ----------------------------------- * ASTERISK-27704 - Add cache_pools debug option to pjproject.conf (Reported by Richard Mudgett) Bugs fixed in this release: ----------------------------------- * ASTERISK-27804 - bridge_softmix / app_confbridge: Add support for combining REMB reports (Reported by Joshua Colp) * ASTERISK-27809 - [patch] utils/pval: Add -lBlocksRuntime for compiler clang conditionally. (Reported by Alexander Traud) * ASTERISK-27806 - BASIC-RETRANS: Implement send (Reported by Benj...

...thing in the upgrade documentation that says "MOH will no longer work in softmix, you should use simple". This has me concerned that I've done something wrong elsewhere in my config that is causing softmix to not work correctly.? The bridging technology bridge_softmix is only used by app_confbridge in Asterisk v11. Nothing else in v11 uses the bridging framework. Unless you were using app_confbridge, you were not using bridge_softmix in v11. The various bridging technology modules in v12 and later are for different scenarios. The bridging framework is smart enough to pick the best bridging...

...channel video source not set to NULL after unref (Reported by Richard Kenner) Improvements made in this release: ----------------------------------- * ASTERISK-27683 - [patch] BuildSystem: Allow newer autotools on OpenBSD. (Reported by Alexander Traud) * ASTERISK-27651 - app_confbridge: Add Muted to ConfbridgeJoin and channel snapshot headers to ConfbridgeList AMI events (Reported by Richard Mudgett) * ASTERISK-27647 - app_confbridge/bridge_softmix: When channel muted report talking stopped if was talking. (Reported by Richard Mudgett) * ASTE...

...by Alexander Traud) * ASTERISK-27944 - res_pjsip_t38: Crash receiving 1xx responses other than 100 before 200 for T.38 reINVITE (Reported by Joshua Elson) * ASTERISK-27783 - res_pjsip_pubsub: apparent crash on shutdown (Reported by Kevin Harwell) * ASTERISK-27870 - app_confbridge: Conference bridge and announcer channels are not removed if conference is ended as soon as it starts (Reported by Robert Mordec) * ASTERISK-27909 - cdr: Deadlock with submit_scheduled_batch and submit_unscheduled_batch (Reported by Denis Lebedev) * ASTERISK-26987 -...