[asterisk-bugs] [JIRA] (ASTERISK-21317) Attended transfer to a Queue stops music on hold

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-21317?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=205165#comment-205165 ] 

Rusty Newton commented on ASTERISK-21317:
-----------------------------------------

{quote}
Any chance I could restrict access to the files?
{quote}

Unfortunately no. I'll go ahead and restrict the whole issue to just you and the bug marshals if that helps.

The symptoms in this appear the same as ASTERISK-17833 , marking as duplicate. We don't have much information in the old issue (which was reported on Asterisk 1.8.4 according to comments)

I see that you didn't include SIP debug "sip set debug on" in your Asterisk log. Can you get a new set of data, and add that to the Asterisk log?  It really helps to coordinate information between the PCAP and Asterisk log.

Can you also attach the relevant dialplan for the original extension dialed by the caller and the queue extension it dials into?

We could also use the sip.conf and queues.conf  (sanitized) attached as well.


                
> Attended transfer to a Queue stops music on hold
> ------------------------------------------------
>
>                 Key: ASTERISK-21317
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-21317
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Applications/app_queue
>    Affects Versions: 1.8.18.0
>         Environment: Linux 3.2.0-686 SMP, Asterisk 1.8.18.0
>            Reporter: Filip Jenicek
>            Assignee: Filip Jenicek
>         Attachments: dump.pcap, log.txt
>
>
> Attended transfer to a Queue stops music on hold.
> Steps to reproduce:
> 1. Answer an incoming call on a Cisco SPA508G
> 2. Press the xfer button (attended transfer) and transfer the call to a queue
> <Caller hears music on hold>
> <Callee hears music from the queue>
> 3. Hangup the Cisco to finish the transfer
> <Caller is waiting in the Queue in silence>
> I was able to find a discussion on asterisk-dev, but I failed to locate an issue in jira.
> @see http://lists.digium.com/pipermail/asterisk-dev/2010-January/041662.html
> Is it safe to move "ast_quiet_chan(peerb);" after the "if(peerd)" as suggested by Gabriel Ortiz Lour?

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