Go to sip.conf. Find the SIP Session-Timers section.
Ensure that you have this option set: session-timers=refuse This might help. If not, try other variations of the session-timers value. The default session-timer is 10 minutes - exactly half of what you claim is your duration maximums, so it seems suspiciously like that might have something to do with it. Maybe not. In any case, fire up wireshark/tethereal and watch the SIP packets for a particular call to see what's happening - distrust everything other than what you see on "the wire" and then work backwards. An understanding of SIP packet flows will be helpful here, or the "ladder view" of SIP transactions that is built into wireshark's graphical interface will certainly help as well. JT On Nov 2, 2008, at 11:07 AM, Jim Boykin wrote: > Any help. Thanks > > > On Sun, Nov 2, 2008 at 12:50 PM, Jim Boykin <[EMAIL PROTECTED]> > wrote: >> Marcin, can you elaborate. No timer has been set and call is not >> idle either. >> >> Thanks >> Jim >> >> On Fri, Oct 31, 2008 at 5:17 PM, Marcin J. Kowalczyk >> <[EMAIL PROTECTED]> wrote: >>> Jim Boykin pisze: >>>> We are running Asterisk SVN. We are facing a strange and repetable >>>> problem. All outgoing call gets terminated in approx 20 minutes. >>>> Asterisk initiates BYE message to the remote end and call >>>> terminates. >>>> >>> Sesion-timer set but not supported by sip-peers? >> --- John Todd [EMAIL PROTECTED] +1-256-428-6083 Asterisk Open Source Community Director _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
