> ## Documentation Index
> Fetch the complete documentation index at: https://docs.verbeo.ai/llms.txt
> Use this file to discover all available pages before exploring further.

# Agent SIP URI

> Every Verbeo agent has its own SIP address. Point a softphone, PBX, SIP trunk, or carrier at it to route calls straight to that agent's AI — no phone number or outbound API needed.

Every agent in Verbeo automatically gets a unique **SIP URI** — a dialable address that connects an incoming call directly to that agent. Point any SIP‑compatible system at it — a softphone, a PBX, a SIP trunk, or your own carrier — and the agent answers in real time over Verbeo's voice infrastructure.

There's nothing to enable and no credentials to provision: the SIP URI exists for every agent the moment it's created.

<Note>
  A SIP URI handles **inbound** calls *into* an agent (something dials the agent). If you want to give an agent a real phone number on the public network instead, see [Import a phone number from Twilio](/guides/import-twilio-number).
</Note>

## Anatomy of the SIP URI

Every agent's SIP URI has the same shape:

```
sip:<agent-id>@1mq55afrqof.sip.livekit.cloud
     │             │
     │             └── Verbeo SIP host — the same for every account
     └── unique per agent — this is what routes the call to the right AI
```

* The **host** `1mq55afrqof.sip.livekit.cloud` is shared by all Verbeo accounts.
* The **user part** (e.g. `a1b2c3d4e5f67890abcdef1234567890`) is unique to one agent and is what selects which agent answers.

<Warning>
  Treat the full SIP URI like a secret. There's no password on it by default, so anyone who has the URI can call your agent and consume your credits. See [Keep your SIP URI secure](#keep-your-sip-uri-secure).
</Warning>

## Where to find your SIP URI

<Steps>
  <Step title="Open the agent">
    In the [Verbeo dashboard](https://dashboard.verbeo.ai), go to **Agents** in the left sidebar and click the agent you want to connect.
  </Step>

  <Step title="Open the SIP Configuration section">
    In the agent editor, expand the **SIP Configuration** panel.
  </Step>

  <Step title="Copy the SIP URI">
    Click the copy icon next to the **SIP URI** field. It looks like:

    ```
    sip:a1b2c3d4e5f67890abcdef1234567890@1mq55afrqof.sip.livekit.cloud
    ```

    This is the value you paste into your softphone, PBX, or trunk.
  </Step>
</Steps>

<Tip>
  The user part of the URI is just the agent's UUID with the dashes removed — so agent `a1b2c3d4-e5f6-7890-abcd-ef1234567890` is reachable at `sip:a1b2c3d4e5f67890abcdef1234567890@…`.
</Tip>

## Connect a system to your agent

Use the full `sip_uri` as the destination. The exact steps depend on what's placing the call.

<Tabs>
  <Tab title="Softphone">
    Any SIP softphone (Zoiper, Linphone, MicroSIP, etc.) can dial the agent directly.

    1. Add a SIP account with **Host / Domain** = `1mq55afrqof.sip.livekit.cloud`. A username/password is **not** required — registration will show as failed, which is expected and fine.
    2. Set the transport (see [Transport](#transport) below).
    3. Dial the full URI:

    ```
    sip:a1b2c3d4e5f67890abcdef1234567890@1mq55afrqof.sip.livekit.cloud
    ```
  </Tab>

  <Tab title="PBX / SIP trunk">
    From a PBX or SIP switch (Asterisk, FreeSWITCH, 3CX, Jambonz…), add a route or dialplan entry that sends the call to the agent's SIP URI. For example, in Asterisk PJSIP:

    ```ini theme={null}
    exten => 100,1,Dial(PJSIP/a1b2c3d4e5f67890abcdef1234567890@verbeo-agent)
    ```

    …where the `verbeo-agent` endpoint points at `1mq55afrqof.sip.livekit.cloud` over TLS. This lets you bridge any extension, IVR menu, or queue into a Verbeo agent.
  </Tab>

  <Tab title="Twilio">
    Forward a Twilio call to the agent with a TwiML `<Dial><Sip>`:

    ```xml theme={null}
    <Response>
      <Dial answerOnBridge="true">
        <Sip>sip:a1b2c3d4e5f67890abcdef1234567890@1mq55afrqof.sip.livekit.cloud</Sip>
      </Dial>
    </Response>
    ```

    Point a Twilio number's Voice webhook at a TwiML Bin containing this, and calls to that number reach the agent. Twilio negotiates TLS to Verbeo for you.
  </Tab>
</Tabs>

## Transport

Verbeo's SIP host accepts **TLS** (recommended), as well as TCP and UDP.

<Note>
  Use **TLS** when your client supports it — it encrypts the call setup. Some free softphone tiers gate TLS behind a paid plan; in that case use **TCP**, or place the call through a service that handles TLS for you (such as the Twilio bridge above).
</Note>

## Test your SIP URI

<Steps>
  <Step title="Copy the agent's SIP URI">
    Grab it from the agent's **SIP Configuration** panel (or the API).
  </Step>

  <Step title="Place a call">
    Dial the URI from a softphone or your PBX/trunk. The agent should pick up and start the conversation.
  </Step>

  <Step title="Confirm in Call History">
    Open **Call History** in the dashboard. You'll see a new **Inbound** call to the agent's SIP URI with status **Completed** and the duration of your test.
  </Step>
</Steps>

## Keep your SIP URI secure

The SIP URI is open by design (so any carrier can reach it without credentials), and the unguessable agent ID is what protects it. A few good practices:

<AccordionGroup>
  <Accordion title="Treat the URI as a secret">
    Don't embed the SIP URI in client‑side code or public repositories. Anyone with the full address can call the agent and use your credits.
  </Accordion>

  <Accordion title="Prefer TLS transport">
    Where your client or PBX supports it, connect over **TLS** rather than plain UDP/TCP so the call signaling is encrypted in transit.
  </Accordion>

  <Accordion title="Restrict source addresses on your side">
    If the calls originate from your own PBX or SIP trunk, use that system's IP allowlisting so only your trusted infrastructure can reach the agent.
  </Accordion>

  <Accordion title="Need stricter access control?">
    If you require per‑agent authentication or an IP allowlist on Verbeo's side, contact [support@verbeo.ai](mailto:support@verbeo.ai).
  </Accordion>
</AccordionGroup>


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