Create a new voice route (call route) to configure how inbound calls are handled. Define SIP trunk settings, forwarding numbers, IVR menus, workgroups, and schedules. Each route is identified by a unique CRID (Call Route ID).
Enter the token obtained from the Authentication endpoint
Friendly name for voice route
"Main SIP Trunk"
Type of voice routing
Trunk, URL, Forward "Trunk"
Webhook URL for voice events
"https://domain.com"
HTTP method for voice URL
GET, POST "POST"
Authentication type
None, Basic, Bearer Username for auth
"admin"
Password or token
"passwordORbearerToken"
Callback URL
"https://apiendpoint.com"
GET, POST "POST"
Primary SIP trunk IP
"20.87.87.87"
Secondary SIP trunk IP
"20.87.87.86"
Primary trunk port
"5061"
Secondary trunk port
"5060"
Primary transport protocol
UDP, TCP, TLS "UDP"
Secondary transport protocol
UDP, TCP, TLS "TCP"
"https://pstn.joonto.com"
"https://pstn2.joonto.com"
Enable primary trunk
true
Enable secondary trunk
false
Priority for trunk 1
10
Weight for trunk 1
10
Priority for trunk 2
20
Weight for trunk 2
20
Header manipulation ID
"HMI8802029"
Enable SIP REFER
true
Caller ID for transfers
"Transferee/Transferor"
Enable symmetric RTP
true
Enable secure trunk
true
Enable PSTN transfer
true
IP whitelist
Enable CNAM lookup
true
Termination URI subdomain
"client"
Enable call streaming
true
WebSocket secure URI
"wss://192.67.88.2"
Enable/disable route
true
Create Voice Route
The response is of type object.