[asterisk-bugs] [Asterisk 0014519]: 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=14519 
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Reported By:                triccyx
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   14519
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.1-rc1 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-02-20 09:59 CST
Last Modified:              2009-02-23 10:34 CST
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Summary:                    Rtp socket are not closed after Hangup
Description: 
After a sip call the rtp ports are never closed.
Please try with "netstat -lnp | grep asterisk"


Perhaps dialog_unlink_all() need
if (dialog->rtp) 
    ast_rtp_destroy(dialog->rtp);
...


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Relationships       ID      Summary
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related to          0014522 sip channels stay open
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 (0100557) file (administrator) - 2009-02-23 10:34
 http://bugs.digium.com/view.php?id=14519#c100557 
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I'm afraid I'm going to need much more information to try to track this
down. I've tried many scenarios with both 1.6.1-rc1 and 1.6.1 SVN and they
all work fine. Can you provide console output, dialplan, SIP traces, SIP
history, everything that could possibly help? 

Issue History 
Date Modified    Username       Field                    Change               
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2009-02-23 10:34 file           Note Added: 0100557                          
2009-02-23 10:34 file           Status                   assigned => feedback
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