search for: ast_writestream

..., "") in new stack -- Executing Record("Zap/27-1", "soundexampless:mp3") in new stack -- Playing 'beep' WARNING[360468]: File translate.c, Line 128 (ast_translator_build_path): No translator path from UNKN to ULAW WARNING[360468]: File file.c, Line 218 (ast_writestream): Unable to translate to format mp3, source format ALAW WARNING[360468]: File app_record.c, Line 166 (record_exec): Problem writing frame Segmentation fault I guess this is pretty explanatory. Regards Alex -------------- next part -------------- An HTML attachment was scrubbed... URL: htt...

...elcome.g729 Failed to convert /tmp/welcome.gsm to /tmp/welcome.g729! Command 'file convert /tmp/welcome.gsm /tmp/welcome.g729' failed. [Dec 20 17:24:18] WARNING[2221]: translate.c:256 ast_translator_build_path: No translator path from g723 to alaw [Dec 20 17:24:18] WARNING[2221]: file.c:184 ast_writestream: Unable to translate to format g729, source format gsm Even though I have the module format_g729.so. Do I need to have licensed G729 codec for this? or codec_g729.so? Kindly let me know how to convert the file. Regards -------------- next part -------------- An HTML attachment was scrubbed... UR...

Hi all, i want to monitor channels with the "Monitor" app in mp3 formatted files. But, if i use mp3 as first argument for the Monitor app, errors occurs.... WANRING[28689]: File file.c, Line 193 (ast_writestream): Unable to translate to format mp3, source format 64 ...following more "Warnings" ..... Is it not possible to monitor to mp3-files? What's wrong? Thanks for Help, Thomas.

Hi What are the disk IOPS required for Asterisk call recording? I am trying to find out number of disks required in RAID array to record 500 calls. Is there any formula to calculate IOPS required by Asterisk call recording? This will help me to find IOPS for different scale. If I assume that Asterisk will write data on disk every second for each call, I will need disk array to support minimum

...@@ -35,7 +35,7 @@ #include "asterisk.h" -ASTERISK_FILE_VERSION(__FILE__, "$Revision: 201423 $") +ASTERISK_FILE_VERSION(__FILE__, "$Revision: 204012 $") #include <stdlib.h> #include <stdio.h> @@ -273,8 +273,9 @@ ast_writestream(*fs, cur); } } + } else { + ast_mutex_unlock(&mixmonitor->mixmonitor_ds->lock); } - ast_mutex_unlock(&mixmonitor->mixmonitor_ds->lock); /* All done! free it. */ ast_frame_fre...

...arse multiple Supported: headers (Reported by Olle Johansson) * ASTERISK-24190 - IMAP voicemail causes segfault (Reported by Nick Adams) * ASTERISK-24432 - Install refcounter.py when REF_DEBUG is enabled (Reported by Corey Farrell) * ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream leaks (Reported by Corey Farrell) * ASTERISK-24307 - Unintentional memory retention in stringfields (Reported by Etienne Lessard) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.32.0 Thank you f...

...arse multiple Supported: headers (Reported by Olle Johansson) * ASTERISK-24190 - IMAP voicemail causes segfault (Reported by Nick Adams) * ASTERISK-24432 - Install refcounter.py when REF_DEBUG is enabled (Reported by Corey Farrell) * ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream leaks (Reported by Corey Farrell) * ASTERISK-24307 - Unintentional memory retention in stringfields (Reported by Etienne Lessard) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.32.0 Thank you f...

I just check the source code, Monitor uses ast_writestream and it eventurally goes down to au_write, g723_write, etc. They don't commit to the disk. So, in effect, if you have a lot of ram, the audio should stay in ram until it gets swap out or the file is closed. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asteris...

...uses segfault (Reported by Nick Adams) * ASTERISK-24466 - app_queue: fix a couple leaks to struct call_queue (Reported by Corey Farrell) * ASTERISK-24432 - Install refcounter.py when REF_DEBUG is enabled (Reported by Corey Farrell) * ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream leaks (Reported by Corey Farrell) * ASTERISK-24307 - Unintentional memory retention in stringfields (Reported by Etienne Lessard) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.14.0 Thank you fo...

...uses segfault (Reported by Nick Adams) * ASTERISK-24466 - app_queue: fix a couple leaks to struct call_queue (Reported by Corey Farrell) * ASTERISK-24432 - Install refcounter.py when REF_DEBUG is enabled (Reported by Corey Farrell) * ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream leaks (Reported by Corey Farrell) * ASTERISK-24307 - Unintentional memory retention in stringfields (Reported by Etienne Lessard) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.14.0 Thank you fo...

...ted by Corey Farrell) * ASTERISK-24432 - Install refcounter.py when REF_DEBUG is enabled (Reported by Corey Farrell) * ASTERISK-24411 - [patch] Status of outbound registration is not changed upon unregistering. (Reported by John Bigelow) * ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream leaks (Reported by Corey Farrell) * ASTERISK-24480 - res_http_websockets: Module reference decrease below zero (Reported by Corey Farrell) * ASTERISK-24482 - func_talkdetect: Fix stasis message leak in audiohook callback (Reported by Corey Farrell) * ASTERISK-24487 - configurat...

...ted by Corey Farrell) * ASTERISK-24432 - Install refcounter.py when REF_DEBUG is enabled (Reported by Corey Farrell) * ASTERISK-24411 - [patch] Status of outbound registration is not changed upon unregistering. (Reported by John Bigelow) * ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream leaks (Reported by Corey Farrell) * ASTERISK-24480 - res_http_websockets: Module reference decrease below zero (Reported by Corey Farrell) * ASTERISK-24482 - func_talkdetect: Fix stasis message leak in audiohook callback (Reported by Corey Farrell) * ASTERISK-24487 - configurat...

...ted by Corey Farrell) * ASTERISK-24432 - Install refcounter.py when REF_DEBUG is enabled (Reported by Corey Farrell) * ASTERISK-24411 - [patch] Status of outbound registration is not changed upon unregistering. (Reported by John Bigelow) * ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream leaks (Reported by Corey Farrell) * ASTERISK-24487 - configuration: sections should be loadable as template even when not marked (Reported by Scott Griepentrog) * ASTERISK-24307 - Unintentional memory retention in stringfields (Reported by Etienne Lessard) For a full list of ch...

...ted by Corey Farrell) * ASTERISK-24432 - Install refcounter.py when REF_DEBUG is enabled (Reported by Corey Farrell) * ASTERISK-24411 - [patch] Status of outbound registration is not changed upon unregistering. (Reported by John Bigelow) * ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream leaks (Reported by Corey Farrell) * ASTERISK-24487 - configuration: sections should be loadable as template even when not marked (Reported by Scott Griepentrog) * ASTERISK-24307 - Unintentional memory retention in stringfields (Reported by Etienne Lessard) For a full list of ch...

...by Scott Griepentrog) * ASTERISK-24482 - func_talkdetect: Fix stasis message leak in audiohook callback (Reported by Corey Farrell) * ASTERISK-24480 - res_http_websockets: Module reference decrease below zero (Reported by Corey Farrell) * ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream leaks (Reported by Corey Farrell) * ASTERISK-24411 - [patch] Status of outbound registration is not changed upon unregistering. (Reported by John Bigelow) * ASTERISK-24432 - Install refcounter.py when REF_DEBUG is enabled (Reported by Corey Farrell) * ASTERISK-24466 - app_queue...

...) - [ASTERISK-24480 <https://issues.asterisk.org/jira/browse/ASTERISK-24480>] - res_http_websockets: Module reference decrease below zero (Reported by Corey Farrell) - [ASTERISK-24476 <https://issues.asterisk.org/jira/browse/ASTERISK-24476>] - main/app.c / app_voicemail: ast_writestream leaks (Reported by Corey Farrell) - [ASTERISK-22409 <https://issues.asterisk.org/jira/browse/ASTERISK-22409>] - Local channels in a ConfBridge w/ jitterbuffer=yes leak ast_frame's after masquerade (Reported by Corey Farrell) - [ASTERISK-24411 <https://issues.asterisk.org...