[asterisk-bugs] [Asterisk 0013745]: Recordings out of sync when using chanspy

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=13745 
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Reported By:                geoffs
Assigned To:                mmichelson
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Project:                    Asterisk
Issue ID:                   13745
Category:                   Applications/app_mixmonitor
Reproducibility:            sometimes
Severity:                   major
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           1.4.22 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2008-10-20 09:31 CDT
Last Modified:              2009-05-25 01:59 CDT
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Summary:                    Recordings out of sync when using chanspy
Description: 
Inbound and outbound tracks will get out of sync by several seconds when a
call is monitored using chanspy.  
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Relationships       ID      Summary
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related to          0012837 [patch] Chanspy audio is delayed or lost
related to          0011945 MixMonitor - Out Of Sync Audio With Zap...
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 (0105356) alanredo (reporter) - 2009-05-25 01:59
 https://issues.asterisk.org/view.php?id=13745#c105356 
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I'm running a Druid OSE 2.0 install on our voip server, Asterisk 1.4.23.1
and DAHDI, we have been experiencing this problem for a long time now. Even
stranger is the problem that has turned up since the upgrade to 1.4.23.1,
whenever someone is done listening in on a channel using chanspy and do a
hangup the source channel stays "online" and can be seen using 'sip show
channels'. This results in max channels being meet and after a few days the
SIP account is no longer capable of making calls or taking calls.

The solution to this is that i restart the asterisk service on a daily
basis just to be sure. I think it has something to do with the chanspy bugg
but i can't really prove it, the box is very busy so turning on sip
debugging is not really an option. Any tips on where i might be able to
find some more information about this is appreciated. I'll see if i can
make something happen on a second PBX we have here and in that case i'll
post again. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-05-25 01:59 alanredo       Note Added: 0105356                          
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