Hi, It does not appears to be a session-timers issue. There are no SIP exchange except for BYE message initited after 20 minutes.
I increased the session timer to 3600 seconds and also tried your suggestion without any luck. Any other inputs? Thanks Jim On Mon, Nov 3, 2008 at 10:47 PM, John Todd <[EMAIL PROTECTED]> wrote: > > 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 > _______________________________________________ -- 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