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 firstname.lastname@example.org 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_dialouton your rooms.
First, create a room with your desired room configuration using the
/rooms REST API endpoint.
Phone numbers that are associated with dial-in.
Example to create a room with an expiry
Now that your room is created, you can configure it for PSTN dial-in by adding the
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.
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:
The API will return a Dial-In Code that is unique to your room.
- Join the call with the room URL: https://yourDomain.daily.co/your-room-name
- To join via phone, call, +12095038039,,12345678987 (example case)
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
Once the meeting session has started for the given room, your SIP client may dial the provided
- 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: