Vitelity SIP & FreePBX 17 Setup

I’m running a hosted instance of FreePBX 17 and need Vitelity SIP fully integrated so I can place and receive calls without glitches. Right now basic registration works, but audio is choppy and overall call quality is poor.

Scope of work
• Create and verify the Vitelity trunk, inbound routes, outbound routes, and any required DID mapping so inbound and outbound calling is seamless.
• Diagnose and resolve the current call-quality issue—codec selection, NAT/SIP settings, QoS hints, or any other tweaks you find necessary.
• Apply best-practice security (firewall rules, fail2ban tweaks, SIP authentication hardening) to protect the PBX.
• Run test calls with me to confirm audio clarity both directions and supply a brief hand-off doc noting every change you made.

The server is reachable via SSH and the FreePBX GUI; I’ll provide credentials and a maintenance window. If you have specific experience with Vitelity and the latest FreePBX, let me know—quick, confident turnaround is a plus.

Currently we are experience 2 to 5 second delay between answer and audio flow and it rings 1.5 times after one extentions answers.