Asmaa, You're getting ahead of yourself. How do you expect audio to work if your firewall/NAT settings aren't even configured correctly to establish SIP sessions?
Go back and read the message that I sent yesterday. Fix the SIP three-way handshake problem. That is step 1 and you'll know you have it right when you stop seeing 'Retransmission timeout reached on transmission' errors. You still won't have audio but that's step 2. It requires properly configuring Asterisk's NAT settings and the firewall(s) between the phones and the server to allow RTP traffic to flow, but don't worry about it until step 1 is complete. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
