[asterisk-bugs] [Asterisk 0012815]: BRIDGEPEER variable not updated on attended transfer when codec translation is used.

[noreply at bugs.digium.com]

The following issue has been SUBMITTED. 
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http://bugs.digium.com/view.php?id=12815 
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Reported By:                ramonpeek
Assigned To:                
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Project:                    Asterisk
Issue ID:                   12815
Category:                   Channels/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.20.1 
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:             06-09-2008 07:24 CDT
Last Modified:              06-09-2008 07:24 CDT
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Summary:                    BRIDGEPEER variable not updated on attended transfer
when codec translation is used.
Description: 
When a call is attended transferred from extension A to C by extension B,
and extension A uses a different codec than B or C (for example g.729,
instead of G.711) the BRIDGEPEER variable is not updated.

I've tested this using a simple setup with SIP channels that have
canreinvite set to no.

I noticed that the BRIDGEPEER variable is set correctly on the first
bridge between extension A and B, but when B attended transfers A to C the
BRIDGEPEER variable is not update upon transfer.
If all peers use G.711 this does happen!


I've attached two traces;
One in which the BRIDGEPEER is set correctly because we are using G.711 on
all peers and the other one where the BRIDGEPEER variable is not updated
because peer A uses G.729.
In this trace SIP/400 is peer A, SIP/401 is peer B and SIP/402 is peer C.

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Issue History 
Date Modified   Username       Field                    Change               
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06-09-08 07:24  ramonpeek      Asterisk Version          => 1.4.20.1        
06-09-08 07:24  ramonpeek      SVN Branch (only for SVN checkou => N/A          
  
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