[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=205383#comment-205383 ] 

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

So, as far as the PCAP goes, I can't identify a one-way audio issue. That is - I see RTP to and from endpoints on both sides of Asterisk (from 1003 to asterisk to 1000 and 1000 to 1003). Using wireshark the RTP appears to have audio inside it (ringing, blowing in handset, etc)

In the accompanying logs I don't see anything obviously wrong, as in  any unusual errors or debug that would indicate an obvious issue.

I'd say that the "Server: OpenStage_60_V3 R0.73.0" is likely receiving the RTP, but possibly dropping it for one reason or another..

Your next step is to look at the log on the Siemens unit and interpret what it is doing with the received RTP and then let us know. If there is a bug here, then it would mean that Asterisk is doing something goofy with the RTP and the Siemens unit doesn't like it. Otherwise this may just be a misconfiguration on the Siemens side of things.

Summary: 

* Lack of any errors or unusual behavior in logs shows that Asterisk doesn't know that it is doing anything wrong. 
* A cursory analysis of the PCAP doesn't show anything obviously wrong. 
* You are saying that a user behind the endpoint at 10.0.0.100 in your PCAP isn't hearing anything so we need to know:
** Does that endpoint (OpenStage) think something is wrong with the RTP it is receiving from Asterisk?
** What does the endpoint (OpenStage) think is wrong with the RTP?
* Alternatively you may swap out the OpenStage endpoint with a any other SIP phone (Polycom or even Digium would be easy for us to reproduce with if you can show it happens with those)

At this point this doesn't appear to be a bug. I'll leave this in Waiting on Feedback for a couple weeks to give you a chance to look at the Siemens. 


                
> 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
>         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-1.8.21, onewayaudio-1.8.21.pcap, oneWayAudio-svn, onewayaudio-svn.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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