DNAT incoming SIP call
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
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