Freeswitch Elevenlabs Voice Agent Integration
Budget: $30 – $250 USD
We are building a real-time voice agent using Elevenlab's WebSocket API and need an expert in Freeswitch to help us bridge our current call flow to the Elevenlab's voice agent.
Requirements:
- Proven experience with Freeswitch core and module development
- Experience with `mod_audio_fork` and `mod_audio_stream`
- Deep understanding of SIP/RTP/media flows
What you will do:
- Connect our existing Freeswitch server with Elevenlab's WebSocket-based voice agent using `mod_audio_fork` and `mod_audio_stream`, we need both to be configured.
Enable seamless, real-time, bi-directional audio between the caller and the voice agent .
Stream audio to Elevenlab in real-time and handle incoming transcription/command messages.
Maintain high availability and low latency across multiple concurrent sessions.
Ensure voice agent can:
- Execute in-call commands like:
- End the call
- Transfer the call to a human agent
- Trigger DTMF or SIP-based routing actions
- Play custom messages or handle call hold
Our FreeSWITCH is the latest version.
This project is by bid but we are also looking for a new telecom engineer who does good work at a good price, we need a lot of other things done and our OpenSIPs updated month many other things, if we work well together there will be plenty more work, we are a small startup that is getting ready to launch soon.
Requirements:
- Proven experience with Freeswitch core and module development
- Experience with `mod_audio_fork` and `mod_audio_stream`
- Deep understanding of SIP/RTP/media flows
What you will do:
- Connect our existing Freeswitch server with Elevenlab's WebSocket-based voice agent using `mod_audio_fork` and `mod_audio_stream`, we need both to be configured.
Enable seamless, real-time, bi-directional audio between the caller and the voice agent .
Stream audio to Elevenlab in real-time and handle incoming transcription/command messages.
Maintain high availability and low latency across multiple concurrent sessions.
Ensure voice agent can:
- Execute in-call commands like:
- End the call
- Transfer the call to a human agent
- Trigger DTMF or SIP-based routing actions
- Play custom messages or handle call hold
Our FreeSWITCH is the latest version.
This project is by bid but we are also looking for a new telecom engineer who does good work at a good price, we need a lot of other things done and our OpenSIPs updated month many other things, if we work well together there will be plenty more work, we are a small startup that is getting ready to launch soon.