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.