[asterisk-bugs] [Asterisk 0016036]: Asterisk unable to bridge RTP when a peer server performs a call transfer when canreinvite is enabled
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Oct 20 10:53:54 CDT 2009
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=16036
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Reported By: jehanzeb
Assigned To:
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Project: Asterisk
Issue ID: 16036
Category: Features/Parking
Reproducibility: always
Severity: major
Priority: normal
Status: new
Asterisk Version: Older 1.4
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2009-10-07 10:50 CDT
Last Modified: 2009-10-20 10:53 CDT
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Summary: Asterisk unable to bridge RTP when a peer server
performs a call transfer when canreinvite is enabled
Description:
Hi, I am having problems with the following scenario.
the call process is setup as follows.
handset A <---> server A <----> asterisk <---> server B <---> handset B
and C
when a call is placed by handset A to handset B, all works fine until
handset B tries to transfer the call to handset C. At this point when the
server B tries to update asterisk with the ip address to which asterisk
should route the RTP to. Asterisk acknowledges the change but does not
update handset A of the change. therefore after the call has been
transfered even though Handset A is able to hear handset C, Handset C
cannot hear handset A. I will attach the sip configuration, dialplan and a
sip debug file to illustrate my point. Please let me know if there is any
other information i should provide.
I am currently running Asterisk 1.4.21.2~dfsg-3
Regards,
Jehanzeb
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(0112472) jehanzeb (reporter) - 2009-10-20 10:53
https://issues.asterisk.org/view.php?id=16036#c112472
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Hi Guys,
Any ideas for this one yet
Regards,
Jehanzeb
Issue History
Date Modified Username Field Change
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2009-10-20 10:53 jehanzeb Note Added: 0112472
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