What phone? What firmware version? and have you set qualify=yes for the phones in question? On 1/4/06, Jason <[EMAIL PROTECTED]> wrote: > > 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 >
-- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 _______________________________________________ --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
