Debugging SIP trunks… if you are in the terminal, asterisk -r to get in the Asterisk CLI and then pjsip set logger on to get the SIP output printed to the screen. Also core set verbose 9 will get you more output from the dialplan.
What happens when you send a call to the trunk? Can you verify FreePBX is selecting the Anveo trunk (based on your Outbound Routes)? Is it sending the call in the format Anveo wants? etc. These are the troubleshooting steps I’d go with. Sorry if you already know it; I’m not trying to be condescending.
Also, from the Anveo Direct FAQ:
What IP addresses should I open on my firewall?
You need to allow the following IP addresses to reach your network: 67.212.84.21 - SIP Signaling
176.9.39.206 - SIP Signaling
50.22.102.242 - SIP Signaling
50.22.101.14 - SIP Signaling
72.9.149.25 - SIP Signaling
You should put each of these IP addresses in the Match field of your PJSIP trunk to ensure you receive incoming calls from Anveo.