Contact person: FreeSWITCH & OpenSIPS AI Setup

Phone:Show

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Location: Riaydh, Saudi Arabia

Budget: Recommended by industry experts

Time to start: As soon as possible

Project description:
"I need a clean, source-based installation of the latest stable FreeSWITCH (with the call-center and any other modules required for a contact-centre scenario) on a Linux server. Right beside it, the latest stable OpenSIPS must be compiled and configured, including the OpenSIPS AI Voice Connector.

The topology is simple: OpenSIPS will act as the SIP gateway between FreeSWITCH and external AI voice services such as Vapi or ElevenLabs. Once everything is in place I want to:

• Place SIP calls between two regular extensions registered on FreeSWITCH.
• Route a call from a FreeSWITCH extension through OpenSIPS to the AI service and hear the synthetic voice reply.
• Confirm two-way audio and proper SIP signalling in both cases.

Feel free to rely on default sample configs for the initial build, but adjust what’s necessary to make OpenSIPS bridge FreeSWITCH and the AI endpoint cleanly (registrar settings, dispatcher tables, ACLs, etc.). I’ll provide real domain, IP and credential details at hand-over time.

Deliverables:
1. Shell script or documented command log that builds and installs FreeSWITCH and OpenSIPS on a fresh Linux image.
2. Working configuration files for FreeSWITCH (call-center module enabled) and OpenSIPS (AI Voice Connector active, acting as gateway).
3. Brief README that explains where each config lives, how to restart each service, and the SIP endpoints I can dial for the two test scenarios above.
4. Screen-recording or console logs demonstrating both test calls succeeding.

I’ll verify by reproducing the tests on my side; once audio flows in both directions the milestone is met." (client-provided description)