[asterisk-bugs] [Asterisk 0016224]: [patch] Thousands of Invites never discarded in sip channels

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16224 
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Reported By:                falves11
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16224
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           SVN 
JIRA:                       SWP-400 
Regression:                 Yes 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases):  1.4  
SVN Revision (number only!): 229360 
Request Review:              
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Date Submitted:             2009-11-11 14:27 CST
Last Modified:              2009-12-03 00:54 CST
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Summary:                    [patch] Thousands of Invites never discarded in sip
channels
Description: 
The scenario is very simple
We receive an INVITE
we respond Hangup(34)

after a few hours, if you type "ship show channels" you get thousand of
those failed invites, all identica, llike this:
208.X.X.X   7134239089  65745230093  00102/00000  0x0 (nothing)    No     
 Init: INVITE              
208.X.X.X   8633996336  41250481070  00102/00000  0x0 (nothing)    No     
 Init: INVITE              
208.X.X.X   4046670409  0403812114e  00102/00000  0x0 (nothing)    No     
 Init: INVITE              
208.X.X.X   7134239089  06060dc345a  00102/00000  0x0 (nothing)    No     
 Init: INVITE 
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 (0114640) ili4 (reporter) - 2009-12-03 00:54
 https://issues.asterisk.org/view.php?id=16224#c114640 
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Yes, I miss a trailing '}'. I've updated patch at link
http://portal.orn.ru/files/asterisk-1.4-rescheduled-destruction.patch .

falves11, did You compiled chan_sip.so with my patch/changes?

If recompiled chan_sip.so causes same problems, try to set in sip.conf
this settings:

rtptimeout=60 
session-timers=originate
session-expires=600
session-minse=90
session-refresher=uas


Look at /var/log/asterisk/debug. Is there a "Re-scheduled destruction of
SIP call" mesages? In my debug log I found many those messages with same
call-id. Do you have same? 

Issue History 
Date Modified    Username       Field                    Change               
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2009-12-03 00:54 ili4           Note Added: 0114640                          
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