[asterisk-bugs] [Asterisk 0011545]: SIP REINVITE sometimes broken.
noreply at bugs.digium.com
noreply at bugs.digium.com
Sat Dec 15 03:49:50 CST 2007
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=11545
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Reported By: kebl0155
Assigned To:
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Project: Asterisk
Issue ID: 11545
Category: Channels/chan_sip/Interoperability
Reproducibility: sometimes
Severity: minor
Priority: normal
Status: new
Asterisk Version: SVN
SVN Branch (only for SVN checkouts, not tarball releases): trunk
SVN Revision (number only!): 92855
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 12-13-2007 18:17 CST
Last Modified: 12-15-2007 03:49 CST
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Summary: SIP REINVITE sometimes broken.
Description:
We sometimes get calls disconnected when reinvite is enabled.
Our provider suggested a patch (attached).
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oej - 12-15-07 03:49
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There are related bug reports about not updating the route table after the
initial call setup. This is a duplicate.
A very good catch though. Propably needs fixing in 1.4 too.
Issue History
Date Modified Username Field Change
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12-15-07 03:49 oej Note Added: 0075456
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