[asterisk-bugs] [Asterisk 0010667]: Audio problems (one direction) with 2 SIP peers when using queues and DTMF tones

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=10667 
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Reported By:                daphi
Assigned To:                
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Project:                    Asterisk
Issue ID:                   10667
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.11  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             09-07-2007 08:08 CDT
Last Modified:              09-07-2007 09:15 CDT
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Summary:                    Audio problems (one direction) with 2 SIP peers when
using queues and DTMF tones
Description: 
Given the standard sip.conf, which is was only modified as specified
below.

The sipgate peer is used for incoming calls only and is routed into a
queue. The main problems is, when the caller sends a dtmf tone the callee
cannot hear the caller anymore, although the caller can hear the callee.

If the sipgate extension is routed directly to the sip phone (321) without
going into the queue first there is no problem with dtmf tones. After the
dtmf tone is transmitted both parties can hear each other again. So the
problem must be related to queue.

We have tested the same setup on Asterisk 1.2.10 without any issues.

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---------------------------------------------------------------------- 
 putnopvut - 09-07-07 09:15  
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I could not reproduce this issue locally using the 1.4's latest SVN
checkout. 

Issue History 
Date Modified   Username       Field                    Change               
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09-07-07 09:15  putnopvut      Note Added: 0070097                          
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