# Dial

> The Dial node forwards the call to a phone number, a SIP address or a registered user, with caller ID and announcement.

Source: https://edesy.in/docs/ivr/nodes/dial

The Dial node forwards the call to a phone number, a SIP address (for example an AI voice platform or your PBX) or a registered user. It can set the caller ID the other side sees and speak an announcement before dialling. It is a richer version of Transfer. The call leaves the flow at this node.

## Settings

| Field | Required | Default | Notes |
|---|---|---|---|
| Dial to | No | Phone number | Phone number, SIP address or Registered user |
| Destination | Yes | | `+919800000000`, `sip:agent@sip.example.com`, or a username |
| Caller ID (optional) | No | Your IVR number | Shown to the person you dial |
| Announce message (optional) | No | | Spoken to the caller before dialling |
| Ring timeout (seconds) | No | 30 | 5 to 120 |

Exits: none. The flow ends at Dial.

## Dial to a SIP address

Use a SIP address to hand the caller to an AI voice agent platform (such as LiveKit Cloud, Vapi, Retell or Bolna) or to your own PBX. See [SIP trunking](https://edesy.in/docs/sip-trunking) and [SIP trunk for AI voice agents](https://edesy.in/docs/sip-trunk-for-ai-voice-agents) for setting up the far end.

## Flow checks

- "Dial node has no destination." (error)