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...e/phones/common/moh/moha/Sovereign (format 0x40004 (ulaw| h261)): No such file or directory ast_streamfile failed on SIP/user_xxx-00000012 for /home/phones/common/moh/moha/Sovereign I was under the impression that I didn't have to do anything to get slin support. when I do: module show like codec_ I get: Module Description Use codec_a_mu.so A-law and Mulaw direct Coder/Decoder 0...

Hi: Is there any software or hadware for codec conversion on asterisk ,any suggestion will be appreciated. ? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100202/ee00a520/attachment.htm

Hello, Running one asterisk server with below details. Only SIP to SIP calls. No real time configuration, no recording, no voicemail, no IVR, no codec translation. Average CPU load varies between 4 to 30 for 150 to 200 concurrent calls and we start getting problem in call quality like delay in connectivity, voice breakage etc.... Hardware: 2 Physical processor Intel(R) Xeon(R) CPU

...rted by Gregory Massel) * ASTERISK-25094 - PBX core: Investigate thread safety issues (Reported by Corey Farrell) * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute but asterisk doesn't detect it. (Reported by ibercom) * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm | adpcm | ipc10} (Reported by Badalian Vyacheslav) * ASTERISK-25100 - asterisk coredump if host has an IPv6 address that end with ::80 (Reported by Mark Petersen) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/teleph...

...rted by Gregory Massel) * ASTERISK-25094 - PBX core: Investigate thread safety issues (Reported by Corey Farrell) * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute but asterisk doesn't detect it. (Reported by ibercom) * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm | adpcm | ipc10} (Reported by Badalian Vyacheslav) * ASTERISK-25100 - asterisk coredump if host has an IPv6 address that end with ::80 (Reported by Mark Petersen) Improvements made in this release: ----------------------------------- * ASTERISK-25040 - pbx: Improve perform...

...rted by Gregory Massel) * ASTERISK-25094 - PBX core: Investigate thread safety issues (Reported by Corey Farrell) * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute but asterisk doesn't detect it. (Reported by ibercom) * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm | adpcm | ipc10} (Reported by Badalian Vyacheslav) * ASTERISK-25100 - asterisk coredump if host has an IPv6 address that end with ::80 (Reported by Mark Petersen) Improvements made in this release: ----------------------------------- * ASTERISK-25040 - pbx: Improve perform...

...ibute but asterisk doesn't detect it. (Reported by ibercom) * ASTERISK-25094 - PBX core: Investigate thread safety issues (Reported by Corey Farrell) * ASTERISK-25148 - res_pjsip NULL channel audit (Reported by Mark Michelson) * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm | adpcm | ipc10} (Reported by Badalian Vyacheslav) * ASTERISK-25137 - endpoint stasis messages are delivered twice (Reported by Vitezslav Novy) * ASTERISK-25116 - res_pjsip: Two PeerStatus AMI messages are sent for every status change (Reported by George Joseph) * A...

...ibute but asterisk doesn't detect it. (Reported by ibercom) * ASTERISK-25094 - PBX core: Investigate thread safety issues (Reported by Corey Farrell) * ASTERISK-25148 - res_pjsip NULL channel audit (Reported by Mark Michelson) * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm | adpcm | ipc10} (Reported by Badalian Vyacheslav) * ASTERISK-25131 - chan_pjsip: In-dialog authentication not handled. (Reported by Richard Mudgett) * ASTERISK-25100 - asterisk coredump if host has an IPv6 address that end with ::80 (Reported by Mark Petersen) * AST...

...tem utilities" (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-25148 - res_pjsip NULL channel audit (Reported by Mark Michelson) * ASTERISK-25131 - chan_pjsip: In-dialog authentication not handled. (Reported by Richard Mudgett) * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm | adpcm | ipc10} (Reported by Badalian Vyacheslav) * ASTERISK-25100 - asterisk coredump if host has an IPv6 address that end with ::80 (Reported by Mark Petersen) * ASTERISK-25122 - Large SIP packet received via pjsip over websocket crashes Asterisk (Reported by Ivan...

...led will not compile (Reported by David Wilcox) - [ASTERISK-28238 <https://issues.asterisk.org/jira/browse/ASTERISK-28238>] - PJSIP realtime. getcontext not working with DUNDI (Reported by Ray) - [ASTERISK-28263 <https://issues.asterisk.org/jira/browse/ASTERISK-28263>] - codec_opus: errors setting max_playback_rate and bitrate to "sdp" (Reported by Gianluca Merlo) - [ASTERISK-28250 <https://issues.asterisk.org/jira/browse/ASTERISK-28250>] - build: Cross-compilation fails for target arm-linux-gnueabihf (Reported by Jean Aunis - Prescom) - [ASTER...