AI-generated Answer for f3rch00361
Short answer: ElevenLabs doesn’t publish FreePBX/Asterisk–specific instructions. However, the transfer_to_number tool works when your calls run over SIP trunking and you configure either Conference transfer or SIP REFER as outlined below. If any of these prerequisites aren’t met, transfers will fail.
What to set up on the ElevenLabs side
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Use SIP trunking for your phone number so calls between your PBX and ElevenLabs are SIP. Inbound from your PBX should target ElevenLabs’ origination URI sip:sip.rtc.elevenlabs.io:5060;transport=tcp. Signaling is TCP/TLS; media (RTP) is UDP. Allow TCP 5060 or TLS 5061 for SIP and open RTP UDP on your firewall. ElevenLabs’ SIP side runs at 48 kHz; your PBX must support or resample.
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In your agent, add the system tool “Transfer to Human” (transfer_to_number). The tool only works on phone calls (not the chat widget).
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Define transfer rules:
- Transfer Type: choose Conference (default) or SIP REFER.
- Number Type: Phone (E.164) or SIP URI.
- Destination: E.164 phone number or a SIP URI like sip:1001@your-pbx.example.com.
- Condition: e.g., “User explicitly asks to speak to a human.”
Choosing the right transfer method
- SIP REFER (direct handoff): Works only when the live call itself is over SIP and your SIP trunk/PBX allows REFER. Required if you want to transfer to a SIP URI (e.g., an internal FreePBX extension or queue).
- Conference transfer: ElevenLabs calls the destination, bridges the caller and the human, then removes the AI. Use this if REFER isn’t supported. Note that only SIP-trunked numbers can transfer to external numbers; Twilio-hosted numbers can transfer only to other Twilio numbers unless you use Twilio Elastic SIP Trunking via SIP.
Example: rule and tool call
- Rule (in the Agent UI): Transfer Type = SIP REFER; Number Type = SIP URI; SIP URI = sip:1001@pbx.yourdomain.com; Condition = “When the user asks for a human agent.”
- The LLM must call the tool with a transfer_number that matches a configured rule and include client/agent messages, for example:
{
"type": "function",
"function": {
"name": "transfer_to_number",
"arguments": "{"reason":"Escalation","transfer_number":"sip:1001@pbx.yourdomain.com","client_message":"Connecting you to a human agent now.","agent_message":"Customer requested escalation to support."}"
}
}
Troubleshooting checklist
- Ensure your number is connected via SIP trunking (not just the Twilio native integration if you need off‑Twilio transfers).
- If using SIP REFER, confirm your PBX/trunk accepts REFER and that the live call leg is SIP.
- Verify SIP/RTP networking: TCP 5060 or TLS 5061 open for SIP; RTP UDP allowed; certificates valid if using TLS; and sample-rate handling at 48 kHz.
If you need FreePBX-specific dialplan or REFER settings, they’re not covered in the ElevenLabs docs. If you can share your current SIP trunking setup (signaling transport, TLS, and whether REFER is enabled), I can cross-check it against the documented ElevenLabs requirements above.
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