[asterisk-bugs] [Asterisk 0013509]: Loss of incoming audio during a phone call through a SIP trunk

Asterisk Bug Tracker noreply at bugs.digium.com
Wed Sep 17 17:39:33 CDT 2008


The following issue has been CLOSED 
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http://bugs.digium.com/view.php?id=13509 
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Reported By:                azambrano
Assigned To:                
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Project:                    Asterisk
Issue ID:                   13509
Category:                   General
Reproducibility:            random
Severity:                   major
Priority:                   normal
Status:                     closed
Asterisk Version:           1.4.22 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
Resolution:                 open
Fixed in Version:           
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Date Submitted:             2008-09-17 17:07 CDT
Last Modified:              2008-09-17 17:39 CDT
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Summary:                    Loss of incoming audio during a phone call through a
SIP trunk
Description: 
When I dial to a phone number through a sip trunk, there are some times
where I loos the incoming audio, but the call is still active, when I talk
to the other person he can hear me, but I can´t hear him, I'm sure about
that because I once I have lost the incoming audio, I have called again
that person, and he tells me that he was hearing me, also that he was
talking to me, but I couldn´t hear him. 
====================================================================== 

---------------------------------------------------------------------- 
 (0092649) blitzrage (administrator) - 2008-09-17 17:39
 http://bugs.digium.com/view.php?id=13509#c92649 
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This is most likely due to a configuration issue in sip.conf. Be sure to
disable re-invites and the like which will cause audio to be redirected
away from your Asterisk box, which will not work correctly when the system
is behind NAT (or your devices are behind NAT).

Please utilize the public help channels such as the asterisk-users mailing
list, or the #asterisk IRC channel on the Freenode network at
irc.freenode.net

Thanks for using Asterisk! 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-09-17 17:39 blitzrage      Note Added: 0092649                          
2008-09-17 17:39 blitzrage      Status                   new => closed       
======================================================================




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