Summary:	ASTERISK-26815: Call recording drops when transferring call to queue
Reporter:	Robert Hirabayashi (PCMRobertHirabayashi)	Labels:
Date Opened:	2017-02-23 12:14:54.000-0600	Date Closed:	2020-01-14 11:13:43.000-0600
Priority:	Minor	Regression?
Status:	Closed/Complete	Components:
Versions:	13.10.0	Frequency of Occurrence	Constant
Related Issues:
Environment:	Centos 6.6 FreePBX 13.0.169	Attachments:
Description:	It appears that there is an issue with Asterisk 13 ending call recording prematurely when a call is transferred. After extensive testing and troubleshooting, the FreePBX team had me switch to Asterisk 11, and the issue disappeared! What we found was that when a call was transferred using the button on the phone, the call would stop being recorded. If a call is transferred via FOP2, the call records fine. The issue is definitely with Asterisk 13, but I'm not really sure how to communicate exactly what is wrong. I'd like to think that I can upgrade to Asterisk 13, 14 and beyond and not have to be concerned that this issue still exists. Additional data: The agent is dialing an external line, placing that person on hold, then transferring the call to a queue, that fails over to a time condition, which calls a ring group. Some of this complexity was added in an attempt to bypass the very issue in question. The call will go through, but will stop recording as soon as the external line and the second agent are connected.
Comments:	By: Asterisk Team (asteriskteam) 2017-02-23 12:14:55.365-0600 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) 2017-02-24 06:24:39.592-0600 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 We need to see a console log to see where the recording is being started and to follow the events. By: Asterisk Team (asteriskteam) 2017-03-10 12:00:02.198-0600 Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines