I have FreePBX running on a server with a public IP address and a LAN IP address. I have been unable to get two way audio when calling from extension to extension (one on public IP one on LAN IP) using SIP softphones (I’ve tried hundreds of configurations) but I can only get two way audio when I have the two extensions dial into a conference room.
This is confusing to me because I thought SIP audio always passed through the PBX anyways. How else would recording work if it created a direct connection between extensions? So whats the difference in a conference room call vs extension to extension call?
Thanks to anyone clarifying this for me. Its quite confusing.