Merge GSM & SIP Conference

I need to link one incoming GSM leg with a SIP participant inside FreeSWITCH so both callers join the same live conference room without noticing any difference in media or signaling. The end result must feel like a single, seamless conversation.

Key requirements
• FreeSWITCH must act as the bridge and mixer.
• The conference room has to support call recording (start automatically and save to disk) and allow me to mute or un-mute either participant from the console or an API.

Current environment
– A working GSM gateway already delivers calls to FreeSWITCH through SIP.
– An existing SIP trunk is in place for the VoIP side.
– Root access to the FreeSWITCH server is available.

Deliverable
1. Step-by-step configuration (dial-plan snippets, conference profile, mod_conference settings).
2. Any Lua/ESL scripts needed to trigger recording and participant muting.
3. Quick validation session where we place a GSM call, add a SIP endpoint, verify audio both ways, start/stop recording, and toggle mute.
4. Brief notes on optional security add-ons (e.g., voice encryption) for future expansion.

If you have hands-on experience blending GSM gateways with FreeSWITCH conferences, let’s get this done.