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

# User XML element

> Dial a SIP URI or registered Vobiz WebRTC/SIP user inside Dial, with optional DTMF and X-VH metadata.

Nest `<User>` inside [`<Dial>`](/docs/xml/dial) to create a SIP or registered WebRTC/SIP B-leg.

```xml theme={null}
<?xml version="1.0" encoding="UTF-8"?>
<Response>
  <Dial
      timeout="30"
      timeLimit="1800"
      callbackUrl="https://example.com/dial-events"
      action="https://example.com/dial-result"
      redirect="false">
    <User>sip:agent@example.com</User>
  </Dial>
  <Hangup/>
</Response>
```

The user must resolve to a reachable SIP or registered Vobiz WebRTC endpoint. The B-leg uses the same `DialAnswer`, `DialConnected`, `DialHangup`, and final action lifecycle as a `<Number>` B-leg.

## Attributes

| Attribute                          | Description                                                                                                                                  |
| ---------------------------------- | -------------------------------------------------------------------------------------------------------------------------------------------- |
| `sendDigits` <br /> *string*       | Optional DTMF sequence scheduled after the user answers. Use digits and `w` pause characters. There is no default.                           |
| `sendOnPreanswer` <br /> *boolean* | When `true`, schedules `sendDigits` on the early-media path instead of the post-answer path. Default: `false`.                               |
| `sipHeaders` <br /> *string*       | Comma-separated `key=value` metadata pairs attached to the SIP B-leg. Vobiz exposes them with an `X-VH-` prefix in Dial lifecycle callbacks. |

## Send metadata to the SIP leg

```xml theme={null}
<?xml version="1.0" encoding="UTF-8"?>
<Response>
  <Dial
      timeout="30"
      callbackUrl="https://example.com/dial-events"
      callbackMethod="POST">
    <User sipHeaders="Reason=transfer,Reference=abc123">
      sip:agent@example.com
    </User>
  </Dial>
</Response>
```

The Dial callbacks include fields such as:

```text theme={null}
X-VH-Reason=transfer
X-VH-Reference=abc123
```

These are application-defined metadata fields. Do not use `sipHeaders` to set the displayed PSTN caller ID; use the parent `callerId` when that applies to your routing flow.

Vobiz does not expand placeholders in these values. Render values such as account references or caller identifiers in your application before returning the XML.

## Send digits after answer

```xml theme={null}
<?xml version="1.0" encoding="UTF-8"?>
<Response>
  <Dial timeout="30">
    <User sendDigits="wwww2410">sip:agent@example.com</User>
  </Dial>
</Response>
```

Vobiz schedules the sequence on the post-answer DTMF path. The destination must support and interpret telephone-event DTMF.

## Caller ID and caller name

Set `callerId` and `callerName` on the parent `<Dial>` when your SIP routing flow requires them. Caller-name display remains dependent on the destination endpoint, carrier, and device.

## Next steps

* [Review the Dial element](/docs/xml/dial)
* [Handle Dial callbacks](/docs/xml/dial/dial-status-reporting)
* [Dial a PSTN number](/docs/xml/dial/number)