[asterisk-bugs] [JIRA] (ASTERISK-25171) Early completion of feature code attended transfer results in intermittent one-way audio, "ghost ringing" and robotic sound for some phones and not others.

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25171?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=226549#comment-226549 ] 

Rusty Newton commented on ASTERISK-25171:
-----------------------------------------

I noticed that in both tests with chan_sip and chan_pjsip that the WriteFormat and WriteTranscode values in 'core show channel' seem odd for the B party. With blind transfers and normal attended transfers it is all ulaw. I only see transcoding come in during the early completion on the attended transfer.

{noformat}
newtonr-laptop*CLI> core show channel SIP/BOB-00000007 
<snip>
          State: Up (6)
  NativeFormats: (ulaw)
    WriteFormat: slin
     ReadFormat: ulaw
 WriteTranscode: Yes (slin at 8000)->(ulaw at 8000)
  ReadTranscode: No 
 Time to Hangup: 0
   Elapsed Time: 0h0m26s
{noformat}

{noformat}
newtonr-laptop*CLI> core show channel SIP/CATHY-00000008 
<snip>
          State: Up (6)
  NativeFormats: (ulaw)
    WriteFormat: ulaw
     ReadFormat: ulaw
 WriteTranscode: No 
  ReadTranscode: No 
 Time to Hangup: 0
   Elapsed Time: 0h1m31s
{noformat}

> Early completion of feature code attended transfer results in intermittent one-way audio, "ghost ringing" and robotic sound for some phones and not others.
> -----------------------------------------------------------------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-25171
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25171
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Bridges/bridge_native_rtp, Channels/chan_sip/General, Resources/res_pjsip
>    Affects Versions: SVN, 13.4.0
>         Environment: * Asterisk GIT-13-723a9d4 built by rnewton @ newtonr-laptop on a x86_64 running Linux on 2015-05-27 16:13:50 UTC
> * pjsip 2.3
>            Reporter: Rusty Newton
>         Attachments: asterisk-25171.pcap, extensions_25171.txt, full_25171.txt, messages_25171.txt, pjsip_25171.txt
>
>
> This occurs with chan_sip and chan_pjsip.
> Three phones, A, B and C. 
> A and B are Digium D40's. I've tested with C being an ATCOM AT-610PR or a Yealink SIP-T28P. If I swap C out with a Digium D40 the problem does not occur. It only occurs with the Yealink and ATCOM as C, and possibly with other phones that I don't have on my desk.
> REPRODUCTION:
> * A calls B
> * A initiates feature code attended transfer and dials C
> * While C is ringing, A hangs up. "blonde transfer"
> * C answers and is bridged with B.
> The result is that B, hears muffled and static-ridden ringing that goes on seemingly forever. Typically B does not receive audio from C, but every few calls B will be able to hear C. The received audio when available will be quite robotic sounding.
> The audio from B to C is fine. C can hear B clearly.
> Debug to be attached shortly.



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