> Thank you for your reply. There is a wealth of information on the > wiki, etc. I turned on RTP debug and the SPA is not sending it's > public IP it is sending it's NAT IP (192.168.1.100) so *'s RTP > packets are going nowhere...
Do I understand your question correctly: You have an SPA behind NAT1 and * and a second SIP device behind NAT2. Both devices register, but calls between the devices result in no audio? If that is the case, you can do one of two things: - set canreinvite=no for the devices' sip.conf entries, or - teach both devices to *stop* using their internal IPs for all communications and remove nat=yes from the entry for the SIP device inside NAT2. To set the SPA to give the correct IP, enable STUN, add a STUN server, and say Yes to "Substitue VIA Addr". -- Nabeel Jafferali X2 Networks www.x2n.ca T: 1.647.722.6900 1.877.VOIP.X2N F: 1.866.655.6698 _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users