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:[email protected], 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 and SIP trunk for AI voice agents for setting up the far end.
Flow checks#
- "Dial node has no destination." (error)