Your line · setup

Works with the phone system you already have.

Every Odile line can get a SIP address with its own username and password, at no extra cost. Your office phone system or SIP trunk sends calls straight to it: no new phone number, no porting and no new hardware. You keep your number and your phone system.

Start free How forwarding works

Two ways in

Send calls to your SIP address

For an office phone system that can send a call to a SIP address or an outside SIP trunk with a username and password.

In your owner console, open Your phone system (SIP) and choose Get my SIP address.
Add the address in your phone system as an outside SIP trunk or destination, with its username and password. Use sign-in only, no registration.
Send unanswered or after-hours calls, or an auto-attendant option, to it.
Ring your own number from another phone to test it.

Your address looks like sip:<your line's digits>@sip.rtc.elevenlabs.io. Use TCP on port 5060 (;transport=tcp), or TLS on port 5061 (;transport=tls).

Forward to your line's number

Any phone, any carrier: forward the calls you cannot answer to your Odile line's number. If your system can only forward to phone numbers, use this route.

See the forwarding codes and setup steps.

SIP is included with After Hours ($199), Solo ($29), Odile One, Odile Bail, Odile Restaurant and developer lines on prepaid minutes. Calls are answered just like calls forwarded to your line's number, count the same way against your plan and appear in your console with the transcript.

By phone system

Choose your phone system. On a small screen, scroll the table sideways.
Phone system	How to connect	What it needs
Asterisk	Create a PJSIP endpoint with `outbound_auth` (username and password) and an `aor` whose contact is `sip:sip.rtc.elevenlabs.io:5060;transport=tcp`. In the dialplan, use `Dial(PJSIP/<your line's digits>@<endpoint>)` on no answer or after hours.	Nothing extra.
FreePBX	Connectivity > Trunks: add a PJSIP trunk with the username, secret, SIP server `sip.rtc.elevenlabs.io` and transport TCP. Add an Outbound Route for your line's digits using that trunk. Send the inbound route, IVR option or no-answer destination there, using a Misc Destination dialling your line's digits.	Nothing extra.
3CX	Add a SIP trunk with the host and authentication ID/password (outbound authentication). Add an outbound rule for your line's digits that uses it. Forward the extension or the after-hours route to that number.	Nothing extra.
Grandstream UCM	VoIP Trunks: add a SIP trunk with host, username, authenticate ID and password. Add an Outbound Route for your line's digits using it. Point the IVR's external number or the time-condition destination at it.	A UCM6xxx.
Yeastar P-Series	Extension and Trunk > Trunk > Add: create a trunk with hostname, port, transport TCP, username and password. Select it in an outbound route for your line's digits. Route after-hours inbound calls to that outbound route.	Hostname, port, TCP, username and password.
Twilio	Use a TwiML Bin with `<Dial><Sip username="…" password="…">sip:<your line's digits>@sip.rtc.elevenlabs.io:5060;transport=tcp</Sip></Dial>` on your number's voice webhook, or in the no-answer branch.	A TwiML Bin.
Telnyx	Use a TeXML app with `<Dial><Sip username="…" password="…">sip:<your line's digits>@sip.rtc.elevenlabs.io:5060;transport=tcp</Sip></Dial>`.	A TeXML app.
Zoom Phone, Microsoft Teams Phone, Cisco Webex Calling, Google Voice	Connect through your own session border controller (Zoom BYOC-P, Teams Direct Routing, Webex Local Gateway or Voice SIP Link), then route to the SIP address from there. Or forward to your line's number.	An SBC and the vendor's plan for it.
RingCentral, Aircall, Vonage Business, 8x8, Nextiva, Ooma Office, Dialpad, GoTo Connect	Forward unanswered or after-hours calls to your line's number.	Forwarding to an outside phone number.

Menu names come from each vendor's own help pages (October 2026) and can change; names are trademarks of their owners and are used only to say what they connect to.

Questions people ask

Do I need a new number?

No. You keep your own number and phone system. Send calls to your SIP address without porting or new hardware.

Does it cost extra?

No. SIP is included on every plan with a line. Calls count the same as forwarded calls, against your plan or prepaid minutes, and appear in your console with the transcript.

Is it secure?

Each address has its own username and password. You can change the password or remove the address in your console at any time. TLS is available on port 5061.

Can a developer set it up?

Yes. Use POST https://api.odilelabs.com/v1/lines/{id}/sip or the MCP tool get_sip_address. See the developer guide.