> ## Documentation Index
> Fetch the complete documentation index at: https://docs.famulor.io/llms.txt
> Use this file to discover all available pages before exploring further.

# SIP provider guides

> Provider-specific inbound routing, authentication, signaling IPs, and test steps

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.

<Warning>
  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.
</Warning>

## Supported guides

| Provider | Platform inbound authentication | Source of signaling addresses |
| - | - | - |
| [Easybell](/telephony/providers/easybell) | Provider source IPs | Published networks; observed addresses are labelled separately |
| [Twilio](/telephony/providers/twilio) | Provider source IPs | Global Twilio signaling CIDRs |
| [Telnyx](/telephony/providers/telnyx) | Provider source IPs | Two addresses for the selected inbound region |
| [Zadarma](/telephony/providers/zadarma) | Provider source IPs | Direct SIP and PBX use different signaling lists; [setup and API keys](/telephony/providers/zadarma). |
| [Plivo](/telephony/providers/plivo) | SIP username/password preferred | Origination-URI digest or Plivo signaling CIDRs |
| [Wavix](/telephony/providers/wavix) | Provider source IPs | Current Wavix portal gateway list |
| [Sinch](/telephony/providers/sinch) | Provider source IPs | Sinch Elastic SIP Trunking CIDRs |
| [DIDLogic](/telephony/providers/didlogic) | Provider source IPs | Current DIDLogic gateway table |
| [Vonage](/telephony/providers/vonage) | Provider source IPs | Vonage SIP source subnets |
| [Bandwidth](/telephony/providers/bandwidth) | Provider source IPs | Account-specific trunk-group addresses |
| [Fonial](/telephony/providers/fonial) | SIP-URI forwarding (no digest) | Not published by Fonial |
| [Other and custom providers](/telephony/providers/other-custom) | Confirm with the provider | Never infer from a saved hostname |

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](/telephony/providers/pbx-and-contact-center).

## 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. Outbound failures, transfer failures, and common SIP status codes are covered in [SIP troubleshooting](/telephony/sip-troubleshooting).


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