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

[Asterisk Bug Tracker]

The following issue requires your FEEDBACK. 
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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 08:46 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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 (0100707) file (administrator) - 2009-02-25 08:46
 http://bugs.digium.com/view.php?id=14550#c100707 
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I just spent some time trying to reproduce this without the configs
provided and with the configs provided. Both worked fine and the dialog was
destroyed and RTP sockets closed. After looking at your full log I see that
you have made some considerable changes to chan_sip. I'm going to have to
ask you to try things on a vanilla installation of Asterisk with no
modifications. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-02-25 08:46 file           Note Added: 0100707                          
2009-02-25 08:46 file           Status                   new => feedback     
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