Creating a SIP participant
To make outbound calls with SIP Service, create a SIP participant with the CreateSIPParticipant
API. It returns an SIPParticipantInfo
object that describes the participant.
Outbound calling requires at least one Outbound Trunk.
Create a
sip-participant.json
file with the following participant details:{"sip_trunk_id": "<your-trunk-id>","sip_call_to": "<phone-number-to-dial>","room_name": "my-sip-room","participant_identity": "sip-test","participant_name": "Test Caller"}Create the SIP Participant using the CLI. After you run this command, the participant makes a call to the
sip_call_to
number configured in your outbound trunk. You can monitor the call status using the SIP participant attributes. When the call is picked up by the callee, thesip.callStatus
attribute isactive
.lk sip participant create sip-participant.json
Once the user picks up, they will be connected to my-sip-room
.
Making a call with extension codes (DTMF)
To make outbound calls with fixed extension codes (DTMF tones), set dtmf
field in CreateSIPParticipant
request:
{"sip_trunk_id": "<your-trunk-id>","sip_call_to": "<phone-number-to-dial>","dtmf": "*123#ww456","room_name": "my-sip-room","participant_identity": "sip-test","participant_name": "Test Caller"}
Character w
can be used to delay DTMF by 0.5 sec.
This example will dial a specified number and will send the following DTMF tones:
*123#
- Wait 1 sec
456
Playing dial tone while the call is dialing
SIP participants emit no audio by default while the call connects. This can be changed by setting play_dialtone
field in CreateSIPParticipant
request:
{"sip_trunk_id": "<your-trunk-id>","sip_call_to": "<phone-number-to-dial>","room_name": "my-sip-room","participant_identity": "sip-test","participant_name": "Test Caller","play_dialtone": true}
If play_dialtone
is enabled, the SIP Participant plays a dial tone to the room until the phone is picked up.