Hi, I'm setting up the sip phone device behind the nat and trying to DNAT incoming sip call, but I can't establish the call (ringing was arrived but no sound). I sniffed the sip traffic with Ethereal and found that "200 OK" response does not be NATed properly... Does sip_contrack_nat has capability to DNAT incoming SIP call? Regards, Masayuki Tanaka -------------------------------------- Know more about Breast Cancer http://pr.mail.yahoo.co.jp/pinkribbon/
