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

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26815?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=235415#comment-235415 ] 

Asterisk Team commented on ASTERISK-26815:
------------------------------------------

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].

> 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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