Hi there, I need your help. Please le me know if it is possible to have following implementation in place:
Asterisk server #1 (ast1) has server SIP clients with extensions 17XX Asterisk server #2 (ast2) has server SIP clients with extensions 16XX All I need that extensions from ast1 be able to call extensions to ast2. But asterisk servers need to be used only for call signaling setup. RTP must go directly between SIP endpoints. Is it possible to do? What is the best way to do it? I.N. _______________________________________________ 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
