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

[noreply at bugs.digium.com]

The following issue has been ASSIGNED. 
====================================================================== 
http://bugs.digium.com/view.php?id=9690 
====================================================================== 
Reported By:                mattv
Assigned To:                file
====================================================================== 
Project:                    Asterisk
Issue ID:                   9690
Category:                   Core/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     assigned
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:              
====================================================================== 
Date Submitted:             05-08-2007 15:47 CDT
Last Modified:              08-29-2007 08:55 CDT
====================================================================== 
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. 
====================================================================== 

---------------------------------------------------------------------- 
 svnbot - 08-29-07 08:55  
---------------------------------------------------------------------- 
Repository: asterisk
Revision: 81331

------------------------------------------------------------------------
r81331 | file | 2007-08-29 08:55:49 -0500 (Wed, 29 Aug 2007) | 4 lines

(closes issue http://bugs.digium.com/view.php?id=9690)
Reported by: mattv
Make rtp timeouts work even if two RTP streams are directly bridged in the
RTP stack.

------------------------------------------------------------------------ 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
08-29-07 08:55  svnbot         Checkin                                      
08-29-07 08:55  svnbot         Note Added: 0069598                          
08-29-07 08:55  svnbot         Status                   feedback => assigned
======================================================================