Fix FreeSWITCH Airtel Outbound Calls

My in-house contact-centre stack is almost ready: a React softphone running [login to view URL] hands WebRTC traffic to FreeSWITCH, which in turn should hand the call to an Airtel SIP trunk for delivery to the PSTN. In the lab everything looks clean, yet the moment we hit real Indian numbers the call stalls—no ringing, no media—so we know the problem sits squarely in the SIP signalling with Airtel, not in registration or dialplan execution. I already have: • Stable WebRTC leg from the browser to FreeSWITCH • Dialplans and Lua scripts that work for internal tests • Airtel trunk registered and authenticated What I need is an engineer who can jump onto the server, fire up sngrep / sofia trace, spot what Airtel is rejecting (the call currently never connects), and patch the dialplan or SIP headers accordingly. Familiarity with Indian trunk quirks, especially Airtel or Tata, will help you move fast; experience with platforms like Exotel, Ozonetel or Servetel is a bonus because you’ll recognise the usual gotchas. Deliverable A working outbound route that reaches Indian PSTN numbers consistently. I’ll consider it done once we can place ten consecutive calls without failure and the SIP traces show clean 200 OK/ACK exchanges. You’ll have SSH access to the FreeSWITCH box (Ubuntu 22.04) and a test Airtel trunk. I’m available on Slack for quick feedback loops. If you speak Lua and can read SIP in your sleep, this should be a quick win for both of us.

Project ID: 40411495