Update an existing voice route’s configuration including SIP trunk settings, forwarding numbers, IVR menus, and workgroup assignments. Changes apply immediately to all numbers using this route.
Enter the token obtained from the Authentication endpoint
Call Route ID to update (CRID) - Required
"CRID-509de94f-79cc-429b-a317-2102654dabbb"
Friendly name for voice route
"Main SIP Trunk"
Type of voice routing
Trunk, URL, Forward "Trunk"
"https://domain.com"
GET, POST "GET"
"20.87.87.87"
"20.87.87.86"
"5061"
"5060"
UDP, TCP, TLS "UDP"
UDP, TCP, TLS "TCP"
true
false
true
Authentication type for voice URL
Basic, Bearer "Basic"
Username for voice URL authentication
"admin"
Password or bearer token for voice URL authentication
"passwordORbearerToken"
Callback URL for voice events
"https://apiendpoint.com"
HTTP method for callback voice URL
GET, POST "POST"
URI for primary SIP trunk
"https://pstn.joonto.com"
URI for secondary SIP trunk
"https://pstn2.joonto.com"
Priority for primary trunk (lower = higher priority)
10
Weight for primary trunk load balancing
10
Priority for secondary trunk (lower = higher priority)
20
Weight for secondary trunk load balancing
20
Header manipulation rule identifier
"HMI8802029"
Enable SIP REFER for call transfers
true
Caller ID to use for transferred calls
Transferee, Transferor, Transferee/Transferor "Transferee/Transferor"
Enable symmetric RTP for NAT traversal
true
Enable TLS/SRTP for secure trunk communication
true
Enable PSTN call transfers
true
List of whitelisted IP addresses
[
{ "IpAddress": "20.20.10.10" },
{ "IpAddress": "20.20.10.20" }
]Enable CNAM lookup for caller ID name
true
Subdomain for SIP termination URI
"client"
Enable real-time call audio streaming
true
WebSocket Secure URI for call streaming
"wss://192.67.88.2"
Enable intelligent routing features
true
Update Voice Route
The response is of type object.