[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
Target Version:             1.6.0.8
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:              2009-03-05 06:41 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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 (0101256) kebl0155 (reporter) - 2009-03-05 06:41
 http://bugs.digium.com/view.php?id=14511#c101256 
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Doh...

When we upgraded to 1.6.0.3 it overwrote the

disallow=all
allow=alaw

section in the general section of our sip.conf

Restoring these lines does seem to have fixed the problem.  Sorry, this is
looking like my bad.

I'll try this with a live customer tomorrow (just to be sure) and report
back. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-03-05 06:41 kebl0155       Note Added: 0101256                          
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