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SIP trunking

A SIP trunk lets you keep your phone numbers exactly where they are — with your current carrier or on your own PBX — and still have a Callable agent answer them.

Every other telephony option asks you to move numbers onto an account Callable can reach. SIP works the other way round: we give you an address, and you point your existing system at it. Nothing about your numbers, your carrier contract, or your dial plan has to change.

When to use it

Use a SIP trunk when:

  • Your numbers are with a carrier Callable doesn't integrate with directly.
  • You run your own PBX (Asterisk, FreeSWITCH, 3CX, Avaya, Cisco) and want to hand only some calls to an agent.
  • Porting numbers is slow, contractual, or politically impossible.

If your numbers are already on Telnyx, Twilio, SignalWire or Vonage, use that provider instead — it's fewer moving parts. See Phone numbers.

How a call flows

Your PBX / carrier
        │  SIP INVITE
        ▼
your-subdomain.sip.telnyx.com     ← the address we give you
        │  SIP + RTP terminated here
        ▼
Callable answers, agent speaks

Your system sends the call over SIP. It's terminated on Callable's carrier edge, handed to the agent, and answered — the same agent, prompt, knowledge base and tools you'd get on any other number.

Setting it up

  1. Go to Integrations and open SIP Trunk.
  2. Optionally type a subdomain. Leave it blank and one is generated for you.
  3. Save. You'll get back a hostname that looks like your-company.sip.telnyx.com.
  4. On your PBX or carrier portal, route the numbers you want answered to that host:
sip:[email protected]
  1. Open the agent that should answer, and set its phone number to the number your callers dial.

Step 5 matters. Callable decides which agent answers by looking at the number that was dialled — the same rule used for every other inbound call. If no agent has that number set, the caller hears "this number isn't connected to an assistant yet" and the call ends.

Tip: Route one low-traffic number first and call it yourself. Once you hear the agent answer, move the rest.

Checking it's live

Press Test connection on the SIP Trunk card. It reads the trunk back from the carrier and shows you the exact address to configure. If it says no trunk is provisioned, save the integration again.

Subdomains

Your subdomain is globally unique — if someone else has already taken the one you asked for, you'll be told to pick another. Choose something recognisable, like your company name; you'll be typing it into your PBX config.

Re-saving the integration doesn't create a second trunk. It reconciles the one you already have, which is also how the address stays valid if anything about the platform moves.

Disconnecting

Disconnecting the integration removes the trunk and releases the subdomain. Calls sent to that address afterwards will fail, so update your PBX first.

What SIP doesn't change

  • Billing for your numbers stays with your existing carrier. Callable bills only for the agent's usage — the AI, speech-to-text and text-to-speech, metered in credits like any other call.
  • Your dial plan is yours. Send Callable one number or a hundred, business hours only, or overflow when your team doesn't pick up.
  • Everything downstream is identical: transcripts, recordings, summaries, monitoring and analytics all work exactly as they do for numbers bought on the platform.

Next steps

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