Grandstream ucm 6204 webrtc not reaching theur destinations or going out

Good day

I need help or solution

Calls are not reaching their destinations, iam using webrtc which dials from a 4rd party call center software hosted external. When I dial using webrtc it shows it ringing but to the dialed number there is np incoming call

Below are my firewall settings using Fortigate
Dip alg disbled
Port 5060,5961, 8089 opened
Firewall using public ip

SIP settings
External host i used IP address for the 3rd party software
NAT Enabled
Ports 5060, 5061
pabx ip address added

Extension
NAT Enabled

WebRTC enabled

Https

Any idea on what needed to be corrected?

This is a FreePBX forum. Though also based on Asterisk, Grandstream UCM appliances are unrelated to FreePBX; you may have better luck posting at community.asterisk.org or in a Grandstream forum.

You are describing a complex setup. You should post about the simplest thing that fails.

Are the WebRTC endpoints local to the UCM or remote?
Can they successfully call internal extensions? Outbound via the FXO ports? Outbound via a regular SIP trunk?

Can other extensions (IP phones, softphones, SIP apps) call outbound, via the 3rd party sofware or otherwise?

On a failing call, what gets logged by Asterisk? What, if anything, is logged by the 3rd party software?