[asterisk-bugs] [Asterisk 0014511]: SIP REINVITE broken in 1.6 (was working in 1.4.13)

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14511 
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Reported By:                kebl0155
Assigned To:                
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Project:                    Asterisk
Issue ID:                   14511
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.0.3 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-02-19 12:26 CST
Last Modified:              2009-02-20 04:35 CST
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Summary:                    SIP REINVITE broken in 1.6 (was working in 1.4.13)
Description: 
Hi there

We take SIP calls from a variety of different gateways from a single
provider.

With 1.4.13, we were able to use canreinvite to drop out of the audio
stream.

With 1.6.0.3, reinvite results in no audio with some of their gateways,
and not others.

There is at least one difference in the SIP conversation between the two
gateways.
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 (0100435) kebl0155 (reporter) - 2009-02-20 04:35
 http://bugs.digium.com/view.php?id=14511#c100435 
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Here's the relevant section:

[outbound]
type=peer
host=gk.magrathea.net
username=XXXXXXXX
secret=XXXXXXXX
authname=XXXXXXXX
canreinvite=yes
context=main
disallow=all
allow=alaw 

Issue History 
Date Modified    Username       Field                    Change               
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2009-02-20 04:35 kebl0155       Note Added: 0100435                          
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