[asterisk-bugs] [Asterisk 0015532]: Getting one way audio after blind transfer from a SIP trunk call.

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Jul 20 19:37:45 CDT 2009


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=15532 
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Reported By:                olivier1010
Assigned To:                
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Project:                    Asterisk
Issue ID:                   15532
Category:                   Core/RTP
Reproducibility:            sometimes
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.25.1 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-07-20 05:03 CDT
Last Modified:              2009-07-20 19:37 CDT
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Summary:                    Getting one way audio after blind transfer from a
SIP trunk call.
Description: 

Quite often when calling with a SIP trunk to outside, we loose outgoing
audio.

But we can still hear audio from called party.

We are using G711 alaw for the outgoing SIP trunk.

This appeared with a recent asterisk update to version 1.4.25.1

This is an Elastix 1.5.2 distribution, and the machine is a pentium 4
monocore.


This does not occur with internal calls.


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---------------------------------------------------------------------- 
 (0108004) olivier1010 (reporter) - 2009-07-20 19:37
 https://issues.asterisk.org/view.php?id=15532#c108004 
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This is in pure SIP environnement. We have an ISDN gateway connected to
this server but through an Ethernet SIP trunk.

Anyway, the calls where we are experiencing problems do not go through
this gateway.

It seems that there is more chance to have one way audio if the calling
party stay more than about 30 seconds in music on hold state. If the
transfer is done fastly, there is less chance to get one way audio.

It is difficult to track down this problem as we do not see specific error
reporting in the logs. I will try to investigate more, but i think that it
was important at first to alert the developpers about this. If it is really
an asterisk problem, other reports should follow. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-07-20 19:37 olivier1010    Note Added: 0108004                          
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