[asterisk-bugs] [JIRA] (ASTERISK-23812) One Way Audio on REFER with Jitterbuffer On

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-23812?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=220722#comment-220722 ] 

Rusty Newton edited comment on ASTERISK-23812 at 8/20/14 5:10 PM:
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Maybe I'm sleepy. Where is 525 in the pcap?
[Edit - Nevermind, I see the full call in the pcap. Must have been looking at another pcap.]

Can you also include an Asterisk debug log along with the pcap? 
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information


was (Author: rnewton):
Maybe I'm sleepy. Where is 525 in the pcap?

Can you also include an Asterisk debug log along with the pcap? 
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

> One Way Audio on REFER with Jitterbuffer On
> -------------------------------------------
>
>                 Key: ASTERISK-23812
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23812
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/Transfers
>    Affects Versions: 11.3.0, 11.9.0
>            Reporter: JoshE
>            Assignee: Rusty Newton
>            Severity: Critical
>         Attachments: 226answer_atttransto_525.pcap
>
>
> We are seeing one way audio on transferred calls with the jitterbuffer running.  This affects ALL versions of Asterisk 11 from 11.3 to 11.9.
> Inbound external call treated with:
> Set(JITTERBUFFER(fixed)=200)
> Answered by a SIP extension on Asterisk.  This is transferred to another SIP extension on the same PBX.  When the transfer is completed, and this can be either attended or blind, you will have one way audio.  The internal party's audio makes it out to the original inbound call, but the person to whom the call was tranferred cannot hear.
> Turning off the jitterbuffer 100% resolves the issue.  This could possibly be related to ASTERISK-21144.



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