> > hi, > > try to set the rtptimeout value in sip.conf to a resonable value - so > asterisk will kill the channels if it does not receive rtp traffic for > the specified time > > regards, > Wolfgang > I uncommeted the rtptimeout=60 value in sip.conf and did a reload. It still hasnt dropped the dead call after a couple minutes now...
Do I have to stop and start again? Was hoping it would just drop the call and continue on. Jerry _______________________________________________ -- 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
