Enabling SIP/PSTN Dial-in & Dial-out for your Domain

  • Ensure your Daily account is a paid account (add a credit card in the dashboard)
  • Add an Advanced Support Plan to your account
  • Reach out to help@daily.co to ensure your account is enabled for SIP/PSTN
    • For Dial-out, Daily support will provision a number manually on your domain.
    • Once approved, you will be able to set enable_dialout on your rooms.

PSTN Dial-In

Create & provision the room for dial-in

First, create a room with your desired room configuration using the /rooms REST API endpoint.

Phone Numbers

Phone numbers that are associated with dial-in.


All PSTN enabled rooms must have an expiry date exp (link) shorter than 365 days. If you’re using an owner token, we also recommend using an eject flag.

Example to create a room with an expiry

Add PSTN dial-in to a specific room

Now that your room is created, you can configure it for PSTN dial-in by adding the dialin property. dialin property has 2 fields

display_name : when PSTN participant join the room, it is displayed as given name. wait_for_meeting_start: If PSTN participant join before the start of room, should PSTN participant wait in conference or should the call disconnect. When wait_for_meeting_start is set, PSTN participant listen the hold music till the room starts.

Here's an example REST API call showing how to configure PSTN dial-in:

Example API Return

The API will return a Dial-In Code that is unique to your room.

How to test

  1. Join the call with the room URL: https://yourDomain.daily.co/your-room-name
  2. To join via phone, call, +12095038039,,12345678987 (example case)
  3. Enjoy!

SIP Dial-In

Provision the room for SIP dial-in

To enable SIP dial-in on a room, use the room properties REST API to set the "sip" property. As follows:

When this is done, a new read-only "sip_uri" property will be added to your room, as can be seen in the response:

You can disable SIP on your room by setting the "sip" property to null.

Dialing in

Once the meeting session has started for the given room, your SIP client may dial the provided "sip_uri".

  • When the SIP call is established, a new participant with the previously provided "display_name" will join the call
  • If the SIP call ends, that participant will leave the call
  • The SIP client may dial again to join the call again
  • Currently, only one SIP client may join at a time
  • If the SIP call fails for some reason, the SIP client will need to re-establish the call
  • When the Daily meeting session ends, the SIP call will be terminated
  • If possible, we recommend that SIP clients use the Opus codec for optimal quality and latency

PSTN & SIP Dial-out

Links to our documentation on the Dial-out methods: