I have a SIP communication scenario between two asterisk servers, on the main server I have a trunk and on the other I have an extension to which I register this trunk. It turns out that the outgoing IP of these servers are different, for example: on my trunk on the main server, I put the IP 200.9.221.x as “host”, but on the second server, this is not the valid outgoing IP, both are behind a NAT and the calls are going mute.
host= destination IP
fromdomain= destination IP
Question: Is there a way to solve the problem of mute calls when establishing a SIP communication between two servers that operate behind a NAT?