I am new to FreePBX and need help locating where to enter the inbound and outbound SIP information in the FreePBX 17 WebGUI. I am using Vitelity SIP trunks, and I can see that all sub-accounts are successfully registered in both the FreePBX GUI and the Vitelity website. However, there is no traffic flow from Vitelity to FreePBX, and when I call the Vitelity sub-account from my cell phone, there is no dial tone. Could someone please provide some useful information to help me resolve this issue?
Do you mean ring back tone? You don’t get dial tone on cellular phones, as they use on hook dialling, and FreePBX wouldn’t normally generate any secondary dial tone.
As to the lack of traffic. Check that you have registered with the correct contact address, then try and get logs from your router to see if the calls are reaching the router, and use sngrep, on the FreePBX machine, to see if they are reaching that.
Thanks for your help. My FreePBX is hosting at cloud Vultr.com, and I do not have access to the router.
That leaves you with two things to check, and a problem if they point to the fault being outside the Asterisk machine.