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...but I was not able to have a succeed call (it was established, but no voice running, and then the call disconnected) if Asterisk behind NAT and Polycom behind NAT. When Asterisk behind NAT and Polycom behind NAT, I forwarded the 5060 UDP to asterisk (at asterisk router) and to Polycom IP Phone at polycomg router site, but the problem stayed. Also I was use nat=yes in the sip.conf Also I forwarded the udp rtp ports (that configured in rtp.conf) to the asterisk IP address, and did not succeed. What else I have to do? Any advise? Regards Bilal