[asterisk-bugs] [Asterisk 0009690]: rtptimeout parameter non-functional on sip-to-sip

[noreply at bugs.digium.com]

The following issue has been REOPENED. 
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http://bugs.digium.com/view.php?id=9690 
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Reported By:                mattv
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   9690
Category:                   Core/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.4 
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        No 
Request Review:              
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Date Submitted:             05-08-2007 15:47 CDT
Last Modified:              08-29-2007 06:03 CDT
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Summary:                    rtptimeout parameter non-functional on sip-to-sip
Description: 
I am using a very vanilla version of Asterisk with the 'rtptimeout=5'
parameter set in sip.conf under the global section and for respective
peers.  I just want to be able to unplug an MTA box connected to the
asterisk server and have Asterisk drop the call (and the other bridged leg)
after 10 seconds of no data.

Unfortunately, this isn't working.  A little more detail about the
enviornment:

I am using MTA boxes to call Asterisk, which is bridging the call to
another SIP end point.  There is no authentication or registration
required: Asterisk follows a simple dial plan that routes anything sent to
it right on to the next SIP device.  That functionality is sound, and
normal call functions work perfectly. 

However, physically disrupting the ethernet connection on either side of
the bridged call does not close the connections, in spite of the rtptimeout
parameter specified.  All documentation and troubleshooting  I have found
on the web and in the bug tracker have turned up nothing. 
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---------------------------------------------------------------------- 
 mattv - 08-29-07 06:03  
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I added the canreinvite directive and now a different problem has arisen. 
Now I get the following new notices:
    
-- Packet2Packet bridging SIP/10.0.0.210-0838b278 and
SIP/LA_IAD_1-08386b00

[Aug 29 03:24:30] NOTICE[1995]: chan_sip.c:15306 do_monitor:
'SIP/10.0.0.210-0838b278' will not be disconnected in 11 seconds because it
is directly bridged to another RTP stream

Why are we directly bridging if we have disabled canreinvite?  Shouldn't
that be going through Asterisk? Monitoring RTP packets it appears that way,
so I guess why is it throwing this error and keeping me from doing a hangup
of the call?

Thank you in advance! 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
08-29-07 06:03  mattv          Resolution               no change required =>
reopened
08-29-07 06:03  mattv          Note Added: 0069596                          
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