[asterisk-bugs] [Asterisk 0016014]: Asterisk cuts audio to the internal extension

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16014 
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Reported By:                mrmrmrmr
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16014
Category:                   Channels/chan_sip/CodecHandling
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           Older 1.6.0 
JIRA:                        
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:             2009-10-02 15:37 CDT
Last Modified:              2009-10-05 13:39 CDT
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Summary:                    Asterisk cuts audio to the internal extension
Description: 
Hi, 

I have a problem with one call scenario on my Asterisk server. 
The call is coming from an external SIP proxy to my server. Asterisk sends
the call to one internal extension. 
Extension rings and then answers. 
The user at extension hears audio for 0.5 seconds. Then audio in that
direction is cut. (no RTP is sent) 
Audio in other direction continues. 

I got network traces for this call scenario. Everything seems normal in
SIP messaging and as RTP. 
I can also see that external audio continues even after the internal
extension's audio is cut.

This problem occurs with one type of CPE on the external side. With other
CPEs this problem is not reproduced.

I believe Asterisk cuts the internal extension's audio because it doesn't
like something in the RTP stream of external CPE.

I will add the network trace for unsuccessful call together with RTP debug
on Asterisk.
On both of these files it is clearly seen that one way audio is cut in a
very short time after call is established.


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 (0111864) mrmrmrmr (reporter) - 2009-10-05 13:39
 https://issues.asterisk.org/view.php?id=16014#c111864 
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Ok; I got the sip trace along with the RTP trace on Asterisk console and
attached it.
I also attached a text file which includes all of my sip configuration.

Please let me know if anything else is required. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-10-05 13:39 mrmrmrmr       Note Added: 0111864                          
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