[asterisk-bugs] [JIRA] (ASTERISK-21172) One way audio when external Call forwarded to queue member

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-21172?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=204966#comment-204966 ] 

Rusty Newton commented on ASTERISK-21172:
-----------------------------------------

Peter,

 Does this only happen with the one queue member 5333 ? 

 Does this only occur with the one particular external caller?

 Does this happen with any caller, regardless of which queue member they get transferred too?

 For the caller or queue member this happens with - does it happen on *every* call or is it random?

 Is there a chance you could test with the latest SVN snapshot of the 1.8 branch? (and report which revision you tried)

 Can you provide an example PCAP, along with the Asterisk log file (DEBUG,VERBOSE) and SIP debug enabled? The SIP debug in the Asterisk log helps to correlate what's happening in the PCAP to the Asterisk log.

 If you can help to narrow down the most minimal configuration it would require to reproduce this issue - that would be extremely useful. I'm having issues debugging your packet captures - wireshark doesn't like some of the RTP and is crashing. If we can reproduce the issue that would be easier.

                
> One way audio when external Call forwarded to queue member
> ----------------------------------------------------------
>
>                 Key: ASTERISK-21172
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-21172
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Applications/app_queue, Channels/chan_local, Channels/chan_sip/General
>    Affects Versions: 1.8.20.1
>         Environment: Ubuntu 10.04LTS, I686pae
>            Reporter: Peter Katzmann
>            Assignee: Peter Katzmann
>         Attachments: digium-info_20130226.tar.gz, flow-queue, flow-user, from-asterisk-to-queue1817.cap, from-asterisk-to-queuemember-1817, from-asterisk-to-queue-member.pcap.bz2, from-ext-to-asterisk.pcap.bz2, from-patton-to-asterisk-1817, from-patton-to-asterisk-1817-2.pcap, onewayAudio, onewayaudio.pcap
>
>
> When i call from external an colleague and he is not available the call is forwarded to the hotline queue.
> On queue operator pick up the call, ringing stops on caller side ba he can't here the agent but the agent can hear the caller.
> When i try the same scenario internal audio ist ok
> Asterisk 1.8.17 is working fine
> Core dump during one call is available so if there is some information inside give me some information how to retrieve them.

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