User XML element

Nest <User> inside <Dial> to create a SIP or registered WebRTC/SIP B-leg.

<?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` string	Optional DTMF sequence scheduled after the user answers. Use digits and `w` pause characters. There is no default.
`sendOnPreanswer` boolean	When `true`, schedules `sendDigits` on the early-media path instead of the post-answer path. Default: `false`.
`sipHeaders` 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 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:

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 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.

XML Overview

User XML element

Attributes

Send metadata to the SIP leg

Send digits after answer

Caller ID and caller name

Next steps

​Attributes

​Send metadata to the SIP leg

​Send digits after answer

​Caller ID and caller name

​Next steps

Attributes

Send metadata to the SIP leg

Send digits after answer

Caller ID and caller name

Next steps