[asterisk-bugs] [Asterisk 0018837]: [patch] Deadlock with attended transfer of SIP call

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18837 
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Reported By:                alecdavis
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18837
Category:                   Core/RTP
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           1.8.2.3 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2011-02-17 19:25 CST
Last Modified:              2011-02-18 03:50 CST
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Summary:                    [patch] Deadlock with attended transfer of SIP call
Description: 
3 SIP phones.

A calls B, and B answers.
B puts A on hold by selecting line2.
B calls C, and C answers.
B initiates transfer of line1 to line2, phone uses REFER.


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 (0132117) Irontec (reporter) - 2011-02-18 03:50
 https://issues.asterisk.org/view.php?id=18837#c132117 
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Sorry, but with this config we havn't got any lock.. (We can't reproduce
the lock)

The problems are with dtmfmode=info:
-With directmedia=yes there are two Invites from Asterisk.
http://lists.digium.com/pipermail/asterisk-dev/2011-February/047827.html

-With directmedia=update we have deadlocks. See 0018734


I'll try to reproduce your deadlock without the patch. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-02-18 03:50 Irontec        Note Added: 0132117                          
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