[asterisk-bugs] [JIRA] (ASTERISK-26815) Call recording drops when transferring call to queue

[Joshua Colp (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26815?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=235425#comment-235425 ] 

Joshua Colp commented on ASTERISK-26815:
----------------------------------------

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.

> Call recording drops when transferring call to queue
> ----------------------------------------------------
>
>                 Key: ASTERISK-26815
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26815
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 13.10.0
>         Environment: Centos 6.6
> FreePBX 13.0.169
>            Reporter: Robert Hirabayashi
>            Severity: Minor
>
> 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.



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