Your own SIP trunk

A voice AI platform for your own SIP trunk and phone numbers

VoiceDock works with your own SIP trunk and your own phone numbers. Your number stays with your carrier and nothing is ported: calls reach the voice agent over the trunk, and the agent can call out and transfer callers over the same connection. We set the trunk up with you and your provider, and read the SIP logs ourselves when a call does not connect. Vapi, Retell AI, Bland AI and Synthflow also support your own trunk; the comparison further down shows how they differ.

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Read the docs

Your number, your carrier · inbound, outbound and transfers · € 0,07/min orchestration · telephony per leg

What you get

What changes when calls run over your own trunk

Your phone numbers and your contract with your carrier stay as they are. The only change is where the calls for the chosen number are sent.

Your number stays where it is

The number stays registered with your own carrier and in your own name. Nothing is ported and there is no second subscription. If you ever leave, you point the traffic somewhere else. Phone numbers in the docs.

No call forwarding, no double call time

With call forwarding, a call stays on your line as an outbound call for as long as it lasts, and a transfer to a colleague adds a second connection. Over a trunk, the call arrives at the voice agent directly.

Inbound, outbound and transfers on one connection

The same trunk carries incoming calls and, if your carrier allows it, outbound calls and transfers. With a trunk assigned to the number, warm and cold transfers run through your own phone system. Call transfers in the docs.

Someone who reads the SIP logs with you

A first call on a new trunk often fails on something small: a codec, a number format, a firewall rule. We set the trunk up with you and your provider and look at the signalling ourselves. Our article on why voice agent calls fail shows how we go about it.

How it works

Connect your trunk in four steps

The steps are the same for every carrier that offers a real SIP trunk. Most of the work sits with whoever runs your telephony, and we do it together.

01
Tell us the number and who runs your telephony
That can be you, your carrier or your IT partner. Inbound SIP is set up per connection, so this is where we start talking. Get in touch.
02
Exchange the connection details
We send the address the trunk should deliver to, the port and how the connection is verified. You send the fixed outgoing addresses of your trunk and the number of calls it may carry at the same time. We accept inbound SIP from fixed addresses on an allowlist.
03
Add the number and choose an assistant
Register the number in the dashboard or with POST /numbers, and assign the assistant that should answer. The caller's number arrives in E.164 format, so the assistant can recognise the caller and look them up in your own system. SIP trunks in the docs.
04
Test with real calls, then go live
Call the number from a mobile phone and from a landline, test a transfer to a real destination, and only then route your live traffic to it. A transfer that works in a demo can still fail on a real line, so that test comes before going live.

Compared

Your own SIP trunk on five voice AI platforms

All five connect to your own carrier. The difference is mostly who does the setup and what your trunk has to support. The table is taken from each vendor's own documentation, read on 2 November 2026: Vapi, Retell AI, Bland AI, Synthflow and VoiceDock. For the full picture beyond SIP, see our pages on Vapi, Retell AI, Bland AI and Synthflow.

	VoiceDock	Vapi	Retell AI	Bland AI	Synthflow
Calls coming in	Your trunk delivers to the address we give you for that connection	Point your trunk at `sip.vapi.ai`, or `sip.eu.vapi.ai` for its EU region	Set the origination URI to `sip:sip.retellai.com`, or dial a SIP URI	Point your carrier or PBX at Bland	Connect your carrier, PBX or contact center and import the number
Calls going out	Over your trunk, including transfers	Over a `byo-sip-trunk` credential you create with the API	Through the termination URI of your provider, entered when you import the number	Through your SIP provider	Over the same connection, on the numbers you already have
Setup	With us, per connection	Yourself, with the API	Yourself, by importing the number	Yourself, with a setup wizard or the API, including connection tests and live SIP tracing	Provider guides; some setups need its Professional Services
Addresses	Fixed outgoing addresses of your trunk go on our allowlist	Published signalling IPs per region; media on UDP 40000-60000	Published IP blocks; media on UDP 10000-65535	Published IP ranges to allow-list in your firewall	Covered per provider in its guides
Signalling	UDP or TCP on port 5060, or TLS on port 5061 if your side supports it	Port 5060, or TLS on port 5061	TCP, UDP, TLS or mutual TLS	TLS 1.2 or higher on port 5061 only; SRTP for media	Covered in its SIP guide

Vendor documentation as read on 2 November 2026. Details change; the linked pages are leading.

Fair to say

Where another platform may suit your trunk better

Each of these is a real advantage, and worth weighing before you choose.

You want to do it all yourself, today

Vapi, Retell AI and Bland AI publish their SIP addresses and let you connect a trunk without talking to anyone. On VoiceDock, your account and API are ready at once, but inbound SIP is set up with us per connection.

Your provider cannot send from fixed addresses

We accept inbound SIP from fixed addresses and do not register with your provider as if we were a phone. Some cloud phone systems only work that way or offer no trunk at all; ask your provider before you plan around it.

You need encrypted media by default

Bland AI requires TLS and SRTP on every connection, and Retell AI supports mutual TLS. On VoiceDock, signalling can run over TLS, but media encryption on the trunk is not on by default. If your environment requires it, tell us up front so we can arrange it with you.

You run a contact center platform

Synthflow lists guides for Five9, Genesys Cloud, Talkdesk, Zoom Phone and others, and tests part of them itself. If your agents have to sit inside one of those stacks, that groundwork is already written down.

Telecom and VoIP providers

Offer a voice assistant on the lines you already provide, with an organisation per customer and your own price.

Warm transfer

Hand a caller to a person over your trunk, and keep the caller with the assistant when nobody answers.

Reliability

How your carrier keeps reaching one address when the platform moves between datacenters.

FAQ

Questions about connecting your own SIP trunk

Which voice AI platform works with my own SIP trunk and phone numbers?

VoiceDock does, and so do Vapi, Retell AI, Bland AI and Synthflow. On VoiceDock your number stays with your carrier, nothing is ported, and calls reach the voice agent over your trunk. We set that up with you and your provider. The table above shows how the five differ.

Do I have to port my phone number?

No. The number stays with your carrier and in your own name. Only the incoming traffic for that number is routed over the trunk to the assistant.

What do you need from my carrier or IT partner?

A SIP trunk for incoming traffic, delivered from fixed addresses. SIP over UDP or TCP on port 5060, or over TLS on port 5061 if your side supports it. The caller's number in E.164 format, the number of simultaneous calls, and, if the assistant should transfer or call out, outbound traffic on the same trunk. We send the connection details directly, because they differ per connection.

What if my provider does not offer a SIP trunk?

Some cloud phone systems for small businesses have no trunk to an outside SIP address. Call forwarding then works with almost every provider, but the call stays on your line as an outbound call for its full length, and a transfer adds a second connection. Moving the number to a provider with a real trunk avoids that. Tell us what you have and we will look at the route with you.

Does it work with 3CX, FreePBX or Asterisk?

For transfers, yes: you create a trunk to your phone system with its host, port and credentials, and assign it to the number. The SIP trunk documentation has examples for FreePBX, Asterisk and 3CX. Inbound calls from your phone system are set up with us per connection.

Can the voice agent call out and transfer over my trunk?

Yes, if your carrier allows outbound traffic on the trunk. Your own numbers support inbound and outbound calls, and with a trunk assigned, transfers run through your phone system. Managed numbers from us are for testing and take inbound calls only.

What does it cost?

Orchestration is € 0,07 per minute, the same as on any other number, and model usage is at cost or on your own keys. Telephony is billed per leg at the carrier's rate; on your own trunk, that is the rate you already have with your carrier. Pricing and what a voice agent costs per minute show the whole sum.

Is a call over my trunk processed in the EU?

Orchestration, dashboard, API and storage run on EU infrastructure. Where the audio is processed depends on the model you choose; the Trust Center lists it per model, and a standard data processing agreement is available.

Connect your own number to a voice agent

Plan a call with your telephony details at hand, or create an account and build the assistant first. Are you a provider yourself? The page for telecom providers covers offering this to your customers.