Does anyone know how to configure Asterisk 13.16.0 so that I can hear the RTP audio stream from a PBX server that sits behind NAT. I’ve reviewed several tickets with respect to this issue. What setting should be configured to allow a softphone client to reach the PBX server with the correct IP address when RTP communication is established. In my debugging, I found that the private IP address of the PBX server is being sent during the SIP session rather than the public IP address. Which NAT setting in 13.16.0 will allow me to pin to the correct IP address? I have the externIP (i.e. public IP) configuration setting as well as the local networks defined and nothing seems to reflect a change in the SDP header. I’ve tried a few configuration changes with no success. I have two-way audio when connecting with devices attached to the PSTN. I think this all comes down to a configuration setting under Asterisk. What configuration setting or settings is going to force the SDP header to have the correct IP address?