Summary:	ASTERISK-26215: app_mixmonitor: No sound when using mixmonitor
Reporter:	TMIMI fakhreddine (symbian001)	Labels:
Date Opened:	2016-07-19 12:44:57	Date Closed:	2016-07-22 00:53:44
Priority:	Major	Regression?
Status:	Closed/Complete	Components:	Applications/app_mixmonitor
Versions:	13.9.1	Frequency of Occurrence
Related Issues:
Environment:	Ubuntu 14.0.4 LTS 64bits	Attachments:
Description:	When I start a call usung mixmonitor there is no sound. No one hear the other but when I play the record file the conversation is recorded correctly. Debug msg : audiohook.c:329 audiohook_read_frame_both: Failed to get 160 samples from write factory 0x7f829403ed38
Comments:	By: Asterisk Team (asteriskteam) 2016-07-19 12:44:58.236-0500 Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report. Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process]. By: Joshua C. Colp (jcolp) 2016-07-20 11:41:51.027-0500 Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. Please read over the Asterisk Issue Guidelines [1] which discusses the information necessary for your issue to be resolved and the format that information needs to be in. We would be grateful if you would then provide a more complete description of the problem. At a minimum, we need: 1. The specific steps or actions you took that caused you to encounter the problem. 2. The behavior you expected and the location of documentation that led you to that expectation. 3. The behavior you actually encountered. To demonstrate the issue in detail, please include Asterisk log files generated per the instructions on the wiki [2]. If applicable, please ensure that protocol-level trace debugging is enabled, e.g., 'sip set debug on' if the issue involves chan_sip, and configuration information such as dialplan and channel configuration. Thanks! [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines [2] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information By: TMIMI fakhreddine (symbian001) 2016-07-21 19:46:59.992-0500 Thanks for your support. unfortunately I had to install an older version (1.8) because of time lack and everything is ok When it's possible I will test again and post more details.