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

[Asterisk Bug Tracker]

The following issue has been CLOSED 
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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:                     closed
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:              
Resolution:                 open
Fixed in Version:           
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Date Submitted:             2009-07-20 05:03 CDT
Last Modified:              2009-09-17 15:41 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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 (0110899) lmadsen (administrator) - 2009-09-17 15:41
 https://issues.asterisk.org/view.php?id=15532#c110899 
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Closing this issue as it appears to be a configuration/interop issue with a
specific provider.

I don't believe it is possible to set different defaultexpiry's for
individual peers. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-09-17 15:41 lmadsen        Note Added: 0110899                          
2009-09-17 15:41 lmadsen        Status                   new => closed       
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