I’m testing a FreePBX install. I can make internal calls (extension to extension). However, inbound and outbound calls connect, but have no audio. I had a 3CX install working, but their call parking didn’t work for us.
I’ve tried a trial SIPSTATION trunk and a trial SIP us trunk, with the same results. No audio. I figure I’ve skipped some important step, but I’m not sure what or how to debug it.
If you have ringing and are able to connect signaling is likely fine and the RTP ports are the issue. Maybe track the traffic with a wireshark and/or set the SIP Debug on in asterisk and capture the logs.
In Asterisk SIP Settings, confirm that External Address and Local Networks are correctly set. If you change these, after Submit and Apply Config you must restart Asterisk.
In your router/firewall, confirm that UDP ports 10000-20000 from any source address are forwarded to the LAN address of the PBX.
If you still have trouble, post make/model of router/firewall and any VoIP-related settings. Confirm that router has a public IP address on its WAN interface.