I have remote users that are setup to sip into the Asterisk server. Problem is that if you call there extension after they have been registered For a while there phones don't ring. If I do a sip show peers they can be seen as registered in. Also the user can dial out. If they reset the phone they can receive calls. This seems to be more of an issue with the Grand stream phones.
The Grandstream has these two settings I am un sure of. NAT Traversal (STUN): currently set to no SUBSCRIBE for MWI: currently set to no Any ideas? -Jason _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
