We have a Freeswitch instance setup and it is working for calls using SIP to carriers. The current solution we have uses a NAT server to present a single IP to carriers. The issue we are facing now is a new carrier that required us to connect via VPN and then handle RTP over the internet via a NAT. Currently we have setup the new requirement and can make outbound calls and play and hear media, however, DTMF is not working. Also when a call is transferred no audio is present on either side. One major difference I can see between using our current carriers that are woking and the new one is that I cannot see in the Freeswitch log 'RTP Correct ip/port confirmed'. Not too sure if this indicates there is an issue or not. Our Freeswitch configs are vanilla except that we set the NAT external IP for SIP and RTP in the vars.xml I have attached a Freeswitch log with additional SIP tracing, any help in resolving this issue would be greatly appreciated. Thanks, Kieran