Hi, I have one problem, i´ve a trunk sip Asterisk----------- Cisco 2600. Call inbound work very good, but call outbound don´t work. Call progress but no audio. Canreinvite=no , no Nat, No problem Codec.
Any idea??? Thanks in advance, D _______________________________________________ -- 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
