Copy your dialplan and sip debug a call.
On Sun, Nov 2, 2008 at 3:07 PM, Jim Boykin <[EMAIL PROTECTED]> 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? > >> > >> > >> > >> _______________________________________________ > >> -- 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 > -- Juan E. Rodríguez Cel. 829-886-5565 Work: 809-724-9227
_______________________________________________ -- 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
