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SIP / Voice gateways
Configure host, media preferences and call limits for the active SIP adapter.
Configure a gateway#
Open Voice Gateways → Add Gateway. Use the SIP provider's actual interconnection details. An SMS/SMPP gateway and a Voice/SIP gateway are different objects.
| Field | Meaning |
|---|---|
| Name / description | Provider label shown in routes and reports |
| Host / port | SIP endpoint; default port 5060 |
| Transport | The current caller supports UDP; visible TCP/TLS choices do not enable those runtime transports |
| Codec preference | AUTO, G729, PCMA or PCMU; controls offer preference |
| Username / password | Stored provider settings; digest challenge retry is not implemented |
| From domain / outbound proxy | Provider-specific SIP identity and next hop |
| Concurrent call limit | 0 means no gateway-specific limit |
| Enabled | Participation in route selection |
Network and NAT#
Set the worker's SIP bind address and the public/NAT address advertised to the provider. SIP signaling and RTP media must both be reachable according to your provider and deployment configuration. A successful INVITE transaction alone does not establish audible media.
Credentials and editing#
A configured-password badge reports presence, not the secret value. Leave the password empty when editing to keep the existing password. Never include a provider secret in screenshots or a support trace.
Verify before enabling routes#
Confirm that the provider supports the current SIP/UDP interconnection model. Then check a single route, selected codec, provider response and audio. An enabled gateway does not create route membership automatically.
Screenshots show fictional data from the public demo. Your instance may have different records and permissions.