[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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