Skip to main content
Use these guides when a carrier sends an existing number to the platform by SIP. Provider settings and signaling networks can change; the source pages below were checked on 22 August 2026. Re-check the linked provider source before changing a production trunk.

​
The rule that prevents most inbound failures

The platform SIP FQDN is the destination for the provider’s INVITE. It is not an authentication method. Configure both sides:
  1. Copy the platform SIP URI from Settings → Numbers → Add a number → Add SIP integration.
  2. Route the provider’s number to that FQDN.
  3. Create the platform trunk with the exact DID the provider sends. Use +E.164 unless the provider guide explicitly says otherwise. A SIP extension is not a wildcard.
  4. Choose SIP username/password only when the provider documents digest authentication toward the destination. Otherwise choose Provider source IPs and enter the provider’s current SIP-signaling IPs or CIDRs.
  5. Keep outbound credentials and the provider termination host separate from inbound authentication.
Never use RTP/media ranges as Provider source IPs, never copy an IP list from another carrier, and never remove the exact DID to make matching broader.

​
Supported guides

Connecting your own PBX or contact-center platform instead of a carrier — 3CX, Asterisk/FreePBX, Starface, Genesys Cloud, Five9, or Aircall? See PBX & contact center platforms.

​
Prove the setup

Call the exact DID from an external phone. The setup is only confirmed when a new inbound item appears in History, the assigned assistant joins, and two-way audio works. Ringing, a delay, or a provider-side call duration alone does not prove that the call reached the platform. Provider-specific sources are linked from each guide. Check the provider’s current documentation before changing a production trunk.