Hi, I've a problem with a cisco 827-4v and asterisk (1.0.9) acting as sip-to-pstn GW. The issue is that when a call comes in from the pstn, asterisk correctly contacts the router, which in turns send a "183 Session progress". Obviously, asterisk thinks that the telephone is not ringing (because it expects a "180 Ringing") and we have no ringback on the pstn side. Putting a ringing() in the dialplan is not an option.
Anyone has suggestions? Cheers, Simone. _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
