> ## 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.
# Other and custom SIP providers
> Safe setup rules for PBXs and providers without a verified global signaling allowlist
A saved hostname proves only that an endpoint was entered. It does not prove direction, authentication, source addresses, or a successful inbound call. Keep the exact DID, route it to the platform SIP FQDN, and obtain current provider-to-destination authentication details. If no stable list is published, ask support for the account's complete **SIP-signaling source CIDRs** and failover regions. Never use media ranges or another customer's addresses.
Running your own PBX or contact-center platform — 3CX, Asterisk/FreePBX, Starface, Genesys Cloud, Five9, or Aircall? See [PBX & contact center platforms](/telephony/providers/pbx-and-contact-center) for step-by-step setup, codec settings, and the failure modes specific to each.
If a provider doesn't support a static SIP trunk at all, you don't have to give up on the number. [Forward it](/telephony/call-forwarding-setup) to a number purchased through the platform for inbound calls, and place outbound calls from that platform number instead.
**Status:** `sip.1und1.de` is normally a registrar/termination host, not evidence of external FQDN forwarding. No verified global source list is included. **Action:** confirm product support, sent DID format, and all signaling sources with 1&1.
**Status:** Fonial documents SIP users and trunking, but SIP-URI forwarding is account-dependent. `sip.plusnet.de` is an outbound host, not an inbound allowlist. SIP trunking with the platform requires Fonial's paid **PLUS** plan — the free tier can't be used this way. **Action:** configure outbound separately and obtain current forwarding/source details from Fonial. Expect the SIP user to show **Offline** in Fonial's dashboard permanently; that's expected, since the platform never sends a SIP REGISTER, not a sign the integration is broken.
**Status:** a `localphone.com` endpoint alone is not a verified inbound profile. **Action:** confirm external SIP-URI routing, DID format, transport, and signaling sources with Localphone.
**Status:** the hostname can identify a PBX or reseller; no global list is verified. **Action:** identify the actual operator and obtain account-specific public signaling addresses.
**Status:** routing a DID to a SIP URI is supported, but auth and sources follow current account/trunk documentation. DIDWW uses two separate trunks — inbound routes to the platform SIP FQDN, and outbound authenticates against `out.didww.com`. **Action:** use current DIDWW sources; never reuse DIDLogic addresses. Send outbound numbers in E.164 **without** the leading `+` — DIDWW expects that exact format, and it's an easy detail to miss.
**Status:** credentials and routing are account-specific; no global list is verified here. Zadarma offers two integration paths — a PBX extension, or a direct SIP connection with **External Services → SIP URI** enabled per number. **Action:** confirm FQDN forwarding plus documented downstream Digest or current signaling CIDRs. Either way, check Zadarma's **Voice Geographic Permissions**: international outbound dialing is blocked by default per destination country, and a call to an unlisted country fails silently rather than with an obvious error.
**Status:** capabilities vary by product and region. sipcall's default trunk type registers (**Endgerät**); the platform needs the non-registering, static **Anlagenanschluss** mode instead — not the option a new sipcall customer would normally pick. **Action:** ask support for external SIP-URI routing, number format, transport, and current sources. Use Digest authentication, and set codecs to G.711 a-law with Opus and G.722 disabled (the Swiss default). If outbound calls get a 403, check that the caller ID is verified on the account.
**Status:** generic SIP/BYOC depends on the purchased product. RingCentral has no wholesale SIP trunk for retail accounts; connect by provisioning a **Generic SIP IP Phone** instead, which requires **MVP/RingEX Standard or higher** — entry plans don't expose it. **Action:** confirm external SIP destination support and request current regional signaling ranges. Always point at the global `sip.ringcentral.com` host, never a regional `sipXX.ringcentral.com` address (those are reserved for RingCentral's own desk phones). A newly created SIP endpoint needs one to two minutes to propagate through RingCentral's anti-flood protection — testing immediately produces spurious 403/404 errors, and retrying during that window can extend the lockout.
**Status:** Voice Connectors use region-specific signaling information. Chime's Voice Connector defaults to **Encryption: Enabled**, which requires TLS and SRTP — if the platform side of the trunk falls back to plain UDP on port 5060, the call fails with no media and no obvious error. **Action:** use the current data for the selected AWS region; never reuse another region's addresses. The outbound host follows `.voiceconnector.chime.aws`. Chime also gates outbound by **Calling Plan** — a destination country not explicitly enabled on the AWS account is blocked, independent of any platform-side dialing permissions.
**Status:** no verified direct external-FQDN profile is documented here. **Action:** treat it as unsupported until Placetel confirms route, number format, auth, transport, and source ranges in writing. No static trunk? Use the fallback above.
**Status:** capabilities and SBC addresses vary by country/product. **Action:** request the exact external SIP route and regional source networks for the account. No static trunk? Use the fallback above.
**Status:** Cloud PBX interconnect is product-controlled; no global list is asserted. **Action:** use a support-assisted setup and only the destination/source networks supplied by NFON. No static trunk? Use the fallback above.
**Status:** trunking and forwarding are product-specific; no global list is asserted. **Action:** confirm external SIP-URI routing and current signaling sources before saving. No static trunk? Use the fallback above.
**Status:** no current provider-to-platform authentication profile is documented here. **Action:** identify the exact product and region, then obtain the current FQDN-routing method and complete signaling sources from the provider or account portal.
**Status:** a saved hostname or credential mode is not proof of downstream Digest support. **Action:** confirm the account's external SIP destination support, DID format, transport, and current public signaling sources.
## Required confirmation
Ask whether the DID can be delivered to an external FQDN without REGISTER; the exact DID format in the Request-URI; whether the provider authenticates toward the destination with Digest and where those origination credentials are set; otherwise all signaling CIDRs and failover regions; TCP 5060 or TLS 5061; and whether SRTP is required.
Only an external call with a platform inbound record, the assigned assistant, and two-way audio confirms the integration.
Reject allowlist entries that contain prose, placeholders, hostnames, or unrelated public resolvers such as `8.8.8.8`. A provider network written without its CIDR prefix is only one host and must not be silently expanded or corrected.
Sources: [DIDWW](https://doc.didww.com/), [Fonial](https://www.fonial.de/hilfe/trunking/einrichtung-trunking), [Amazon Chime](https://docs.aws.amazon.com/chime-sdk/latest/ag/voice-connectors.html), [Zadarma](https://zadarma.com/en/support/), [sipcall](https://www.sipcall.ch), [RingCentral](https://developers.ringcentral.com/), [Placetel](https://www.placetel.de/hilfe), [Peoplefone](https://www.peoplefone.com/), [NFON](https://support.nfon.com/), [Sipgate](https://help.sipgate.de/). Accessed 22 August 2026.