Fix FreePBX One-Way Audio

Incoming calls that arrive through our SIP trunk suffer classic “one-way audio”: the caller hears us, yet we get silence. Outbound calls and all internal extension traffic are perfectly clear, so the problem is isolated to inbound SIP traffic only.

I need someone comfortable with Asterisk, FreePBX, RTP/NAT behaviour, and typical firewall or trunk mis-configurations to jump in, trace what is happening, and implement a permanent remedy without interrupting live service. Remote SSH or VPN access to the PBX and edge firewall can be provided immediately, along with console access to the SIP provider’s portal for any trunk changes.

Acceptance means:
• Two-way audio is restored on every new incoming call test.
• No regression on outbound or extension-to-extension calls.
• A brief changelog explaining what you adjusted (trunk settings, NAT rules, port forwards, codecs, etc.) so I can document and roll back if ever required.

If you’ve solved one-way audio on FreePBX before, this should be a quick win.