[asterisk-bugs] [Asterisk 0014519]: 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=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 10:43 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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---------------------------------------------------------------------- 
 (0100561) file (administrator) - 2009-02-23 10:43
 http://bugs.digium.com/view.php?id=14519#c100561 
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If you truly are using 1.6.1-rc1 then that is not true, as the issue in
14522 seems to be isolated to changes only made in trunk. If you are not
using 1.6.1-rc1 then it is important to know exactly what you are indeed
using. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-02-23 10:43 file           Note Added: 0100561                          
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