On Wed, 25 Feb 2009 10:18:02 you wrote: > use wireshark or somethink like it (tcpdump) and see if the "bye" is > reaching asterisk. > if this is the problem you can use rtptimeout option in the sip.conf or > iax.conf. > David > > 2009/2/23 Michael <[email protected]> > > > I am running Asterisk 1.4.22.2, though I have also found this problem > > with 1.4.23.x > > > > Sometimes after I hang up the system continues to spew packets to my > > phone causing it to become unusable until I restart Asterisk. > > > > Michael
It usually seems to happen after I use voice mail. I have no 'rtptimeout' set, what should this be? Michael _______________________________________________ -- 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
