[asterisk-bugs] [Asterisk 0014522]: sip channels stay open

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14522 
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Reported By:                pj
Assigned To:                russell
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Project:                    Asterisk
Issue ID:                   14522
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     assigned
Asterisk Version:           SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 177944 
Request Review:              
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Date Submitted:             2009-02-21 12:28 CST
Last Modified:              2009-02-23 11:29 CST
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Summary:                    sip channels stay open
Description: 
I have currently only 7 peers online, after uptime about 10min, and no call
processing, I have 84 sip channels open, and this number quickly
increases.
It seems, that each sip qualify message causes another sip channel to stay
open.
Asterisk SVN-trunk-r175699 (that I used before) doesn't have this issue.
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Relationships       ID      Summary
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related to          0014519 Rtp socket are not closed after Hangup
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 (0100567) svnbot (reporter) - 2009-02-23 11:29
 http://bugs.digium.com/view.php?id=14522#c100567 
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Repository: asterisk
Revision: 178022

U   trunk/main/sched.c
U   trunk/tests/test_sched.c

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r178022 | russell | 2009-02-23 11:29:16 -0600 (Mon, 23 Feb 2009) | 6 lines

Fix a regression in scheduler entry ordering, and add a regression test
for it.

(closes issue http://bugs.digium.com/view.php?id=14522)
Reported by: pj
Tested by: russell

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http://svn.digium.com/view/asterisk?view=rev&revision=178022 

Issue History 
Date Modified    Username       Field                    Change               
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2009-02-23 11:29 svnbot         Checkin                                      
2009-02-23 11:29 svnbot         Note Added: 0100567                          
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