[asterisk-bugs] [Asterisk 0014511]: SIP REINVITE broken in 1.6 (was working in 1.4.13)
Asterisk Bug Tracker
noreply at bugs.digium.com
Sun Mar 29 12:52:00 CDT 2009
The following issue has been set as RELATED TO issue 0014693.
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http://bugs.digium.com/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: 1.6.0.8
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:
Resolution: no change required
Fixed in Version:
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Date Submitted: 2009-02-19 12:26 CST
Last Modified: 2009-03-29 12:52 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 issue, related to 14511
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Issue History
Date Modified Username Field Change
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2009-03-29 12:52 lmadsen Relationship added related to 0014693
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