do you use the qualify=yes
option for your endpoints? y. Peter Childs schrieb: > Using sipgate.co.uk, Asterisk, FreePBX and Asterisk in a Flash > > I've managed to get a basic system set up. and can now take and make > sip calls over the sip trunk I've got from sipgate.co.uk for testing > purposes.... > > Anyway I can make calls fine (if only to the testing line and other > sipgate lines as I have not set up any credit), and I can take calls > but only if someone phones me within 2 minutes of doing a "sip reload" > otherwise I just get a dead line. > > I'm thinking this is something to do with registration or Nat, but > I've set my Nat up to forward everything, and it all works for > 2minutes..... > > > > Peter. > > -- _____________________________________________________________________ -- 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
