[asterisk-bugs] [Asterisk 0014550]: Rtp socket are not closed after Hangup

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14550 
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Reported By:                triccyx
Assigned To:                
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Project:                    Asterisk
Issue ID:                   14550
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-02-25 04:36 CST
Last Modified:              2009-02-25 10:31 CST
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Summary:                    Rtp socket are not closed after Hangup
Description: 
After a sip call the rtp ports are never closed.


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 (0100722) triccyx (reporter) - 2009-02-25 10:31
 http://bugs.digium.com/view.php?id=14550#c100722 
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Missing 
dialog_unref(cur, "drop ref in iterator loop break");
in my code sorry again 

Issue History 
Date Modified    Username       Field                    Change               
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2009-02-25 10:31 triccyx        Note Added: 0100722                          
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