[asterisk-bugs] [Asterisk 0012815]: BRIDGEPEER variable not updated on attended transfer when codec translation is used.
noreply at bugs.digium.com
noreply at bugs.digium.com
Tue Jun 10 07:45:30 CDT 2008
The following issue has been RESOLVED.
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http://bugs.digium.com/view.php?id=12815
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Reported By: ramonpeek
Assigned To: file
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Project: Asterisk
Issue ID: 12815
Category: Channels/General
Reproducibility: always
Severity: minor
Priority: normal
Status: resolved
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:
Resolution: fixed
Fixed in Version:
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Date Submitted: 06-09-2008 07:24 CDT
Last Modified: 06-10-2008 07:45 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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06-10-08 07:45 svnbot Status assigned => resolved
06-10-08 07:45 svnbot Resolution open => fixed
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