SIP
What a SIP trunk is
SIP stands for Session Initiation Protocol, and a SIP trunk is a digital connection that links a phone system you already run — an office PBX, a contact-centre platform, an Asterisk box — to the internet. Instead of physical phone lines, calls travel over your existing network connection.
For Woise, that is the point: a SIP trunk is what lets your current numbers and current telephony infrastructure hand calls to a Woise agent, and receive calls back from one, without moving those numbers to a new carrier.
If you hold an account with a cloud carrier such as Twilio or Telnyx, you want BYO telephony — that is a vendor account, not a trunk you operate. Use this page when the thing on the other end is infrastructure you run, or a SIP provider you configure endpoint-by-endpoint.
Both directions work
A SIP connection is not inbound-only. Once configured, it carries calls both ways:
| Direction | What happens | What you configure |
|---|---|---|
| Inbound | A call arrives at your phone system and is routed to Woise, which answers it with a workflow | Your side points incoming calls at the Woise endpoint |
| Outbound | An agent places a call out through your trunk, using your numbers as the caller ID | Woise holds your endpoint details and credentials, and dials through them |
This is what makes SIP worth the setup effort over a managed number: the numbers your customers already recognise keep working, and the agent simply becomes another thing that can answer or originate on them.
The two SIP providers, and their honest status
Two SIP-adjacent providers are registered in Woise's telephony registry and selectable in the telephony configuration form. They are not at the same level of readiness, and choosing between them on the assumption that they are will waste your time.
| Provider | Status | Use it when |
|---|---|---|
| Cloudonix | The fuller integration, including dedicated webhook routes for call events | This is the more complete path today; prefer it unless you specifically need Asterisk |
| Asterisk ARI | Registered and configurable, but live coverage is partial next to Cloudonix | You are connecting an Asterisk PBX you operate and can tolerate proving out your own call scenarios |
Say which of the two you are using when you ask us for help. They are not interchangeable in what is actually wired up behind them, and the first question we would otherwise have to ask is the one that determines the whole answer.
What each one asks for
Cloudonix authenticates with a Bearer Token and a Domain ID, plus an optional application name when your Cloudonix domain hosts more than one application. Numbers are supplied in E.164 form.
Asterisk ARI is configured against your own server rather than a vendor account, so the fields describe your Asterisk installation:
| Field | What it is |
|---|---|
| ARI Endpoint | The base URL of your ARI interface, e.g. http://asterisk.example.com:8088 |
| Stasis App Name | The Stasis application registered in Asterisk that Woise should attach to |
| ARI Password | The credential for that ARI user |
websocket_client.conf Name | The connection name Asterisk uses for externalMedia |
| From Extensions | SIP extensions used as the origin for outbound calls |
That last pair is where most Asterisk setups go wrong. Woise receives call audio over
externalMedia, which means your websocket_client.conf entry has to exist and be reachable
from the Asterisk host before any call will carry audio — a trunk that signals correctly but
delivers silence is almost always this.
Setting one up
A SIP connection is created exactly like any other telephony configuration: supply the endpoint details and credentials above, save it, then attach it to a number from Numbers.
From there it behaves like every other connected number. Point a workflow at it to answer inbound calls, or reference it from a campaign to place outbound ones.
Common questions
What is Woise?
Woise is a managed platform for voice AI agents. You design an agent in a drag-and-drop workflow editor, connect a phone number, and it handles real inbound and outbound calls over the phone network or in the browser. The infrastructure, carrier setup and scaling are run for you.
Do I need a phone number to start?
You bring a number you already own and import it — that is the primary path today, and it means you keep your carrier relationship and your existing number. Agents can also take and place calls in the browser over WebRTC with no number at all.
How does billing work?
You are billed for the call minutes your agents actually use, metered as credits against your account. There is no seat licence and no per-agent fee, so an agent that takes no calls costs nothing to keep.
Can an agent connect to my CRM or calendar?
Yes. Agents can read from and write to your CRM, calendar and support tools through the integrations marketplace — you authorise each connection once with OAuth, and no credentials are stored in your workflow.
Do I need engineers to build an agent?
The workflow editor is visual, so building and changing an agent needs no code. There is also a REST API, a Python SDK and a TypeScript SDK if you would rather drive the platform programmatically, and an MCP server so coding agents can build workflows for you.
What happens to call recordings and transcripts?
Recordings and transcripts belong to your workspace and are available through the dashboard and the API. Raw transcripts, recordings and phone numbers are never handed to third-party analytics or error tracking without redaction.
What infrastructure do I have to run?
None. Woise runs the speech models, the language model turn, the call recording pipeline and the telephony legs for you, and scales them with your call volume. There is nothing to provision, patch or keep running.
Still stuck? Talk to our team.