Ok I have used FreePBX for many years but I am no expert. Friends company paid for someone to get a new VoiP system up and running but they have walked away mid project and the company wants to know if it is salvageable.
The basic setup seems like it should work. They setup the system on a DigitalOcean droplet. Then they put the proper firewall rules in place to only allow connections from their SIP trunk provider and their offices. Trunks are connected and calls can be made outbound. Inbound calls seem to come in but then get disconnected after 20-30 seconds.
I have tried changing from UDP to TCP and it seems to help audio issues but it does not fix the call dropping. I also tried to run the endpoint via the VPN using the internal vpn IP but I am getting the same issue.
All endpoints will be remote to the PBX. All locations have the network firewall configured to port forward all SIP 5060/5061 UDP/TCP and 10000-20000 TCP/UDP to the phone or ATA. SIP ALG is turned off. Any ideas where to start on this?