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

[Peter Katzmann (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-21172?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=209116#comment-209116 ] 

Peter Katzmann edited comment on ASTERISK-21172 at 8/14/13 9:16 AM:
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Just some new Log files because the Problem still persist with 1.8.23
You can hear clearly nothing on the caller side. Hope this will help to clear the case.
What i don't understand is that the callee rtp stream changes the ssrc and ports during call. Revert back to 1.8.17 and everything works as expected

Please examine this again
                
      was (Author: pk16208):
    Just some new Log files because the Problem still persist with 1.8.23
You can hear clearly nothing on the callee side. Hope this will help to clear the case.
What i don't understand is that the callee rtp stream changes the ssrc and ports during call. Revert back to 1.8.17 and everything works as expected
                  
> 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: SVN, 1.8.20.1, 1.8.21.0, 1.8.23.0
>         Environment: Ubuntu 10.04LTS, I686pae
>            Reporter: Peter Katzmann
>            Assignee: Matt Jordan
>         Attachments: caller-to-callee.pcap, caller-to-markus-from-asterisk-via-patton.pcap, call-from-patton-to-callee-asterisk.pcap, 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-or-to-asterisk-patton5.pcap.bz2, from-patton-to-asterisk-1817, from-patton-to-asterisk-1817-2.pcap, hotline.log.bz2, logs-1921-v2-patch.tar.bz, no-audio.log, no-audio.pcap, oneWayAudio-1.8.21, onewayaudio-1.8.21.pcap, oneWayAudio-svn, onewayaudio-svn.pcap, pcaps-1921-v2-patch.tar.bz, withandwithout.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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