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

[Asterisk Bug Tracker]

The following issue has been UPDATED. 
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https://issues.asterisk.org/view.php?id=14511 
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Reported By:                kebl0155
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   14511
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     closed
Target Version:             1.6.0.11
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:              
Resolution:                 no change required
Fixed in Version:           
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Date Submitted:             2009-02-19 12:26 CST
Last Modified:              2009-06-08 13:37 CDT
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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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Relationships       ID      Summary
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related to          0014693 Codec negotiation issue when codecs not...
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Issue History 
Date Modified    Username       Field                    Change               
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2009-06-08 13:37 lmadsen        Target Version           1.6.0.10 => 1.6.0.11
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