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FusionPBX gateway setup: add a ZuteSIP number

Updated 2026-09-30 · By the ZuteSIP team (ZuteSIP Inc.)

Short answer: In FusionPBX a SIP trunk is called a gateway. Open Accounts > Gateways, click Add, and enter your ZuteSIP SIP username and password, sip.zutesipdialer.com as the proxy, Register set to true and the external profile. Save and start the gateway, then add an inbound route for your number and an outbound route using the gateway.

FusionPBX is a web interface for FreeSWITCH, so gateways, routes and codecs all live in the GUI. The setup below uses registration with the per-number credentials ZuteSIP provides.

What you need

  • A ZuteSIP number and the SIP username and password from its page in the dashboard.
  • FusionPBX admin or superadmin access, and the domain you want the number to ring in.
  • Firewall rules allowing SIP to sip.zutesipdialer.com on 5060 (or 5061 for TLS) and FreeSWITCH's RTP range.

Settings to enter

SettingValue
SIP server / registrar / domainsip.zutesipdialer.com
Transport and portUDP or TCP 5060; TLS 5061
Username<SIP_USERNAME> (from the number's page in the dashboard)
Password<SIP_PASSWORD> (from the same page)
AuthenticationDigest (username + password), registration based
CodecsG.711 µ-law (PCMU) and G.711 A-law (PCMA)
DTMFRFC 2833 (RTP events) or SIP INFO
Inbound DID<YOUR_NUMBER> in E.164 format

Step-by-step: add the gateway

Field names match the FusionPBX documentation for Gateways as of September 2026.

  1. Sign in to FusionPBX, open the Accounts menu and choose Gateways.
  2. Click Add.
  3. Gateway: a name such as zutesip. Username: <SIP_USERNAME>. Password: <SIP_PASSWORD>.
  4. From User: <SIP_USERNAME>. From Domain and Proxy: sip.zutesipdialer.com. Realm can stay empty or be set to the same host.
  5. Register: true. Expire Seconds: leave the default unless support advises otherwise.
  6. Context: public. Profile: external. Enabled: true.
  7. Under the advanced options, set Register Transport to udp (or tls with the proxy on port 5061) and, if shown, Codec Preferences to PCMU,PCMA.
  8. Save, then use the Start button on the gateway list and confirm the status shows it as registered (REGED).

Inbound route: send your DID to a destination

Go to Dialplan > Inbound Routes (or Destinations on newer builds) and add a route whose destination number is your ZuteSIP number. Point it at an extension, ring group, IVR or call center queue in the right domain.

Because the gateway uses the public context, the route must match the number exactly as it arrives. Check the FreeSWITCH log or the CDR entry of a test call for the format, with or without the plus.

Outbound route and caller ID

Open Dialplan > Outbound Routes, select the zutesip gateway and pick or write a dial pattern, for example international numbers dialed with a leading plus.

Set the outbound caller ID number for your extensions (or on the route) to your ZuteSIP number. Match it to the number whose credentials the gateway registers with. Outgoing calls count against the number's 1,000 included minutes per month.

Several numbers or several tenants

Every ZuteSIP number has its own SIP username and password, so each number becomes its own gateway. Name them clearly, for example zutesip-us-sales and zutesip-gb-support, so outbound routes and logs stay readable.

In a multi-tenant FusionPBX, create the gateway in the domain that will use it, or as a global gateway if you want several domains to dial out through it. Inbound routes are always tied to one domain, so a single DID rings in exactly one tenant.

Test the gateway

  1. In Status > SIP Status (or the fs_cli command sofia status gateway zutesip), confirm the gateway state is REGED.
  2. Call your ZuteSIP number from another phone and confirm the inbound destination rings with two-way audio.
  3. Press digits in an IVR to check DTMF, then place an outbound call and check the caller ID the other side sees.

Troubleshooting

  • Gateway shows FAIL_WAIT or keeps retrying: usually 401/403 from wrong credentials. Re-copy them from the dashboard and restart the gateway.
  • Registered, but inbound calls go nowhere: the inbound route number does not match the arriving DID, or it was created in a different domain.
  • One-way audio: set the external RTP and SIP IP variables for your public address, forward the RTP range and disable SIP ALG on the router.
  • Calls rejected for codec reasons: make sure PCMU and PCMA are allowed on the external profile and the gateway.
  • IVR digits not recognized: use RFC 2833 DTMF, which FreeSWITCH uses by default; SIP INFO is the fallback.

Frequently asked questions

Is a FusionPBX gateway the same as a SIP trunk?

Yes. FusionPBX calls the connection to a SIP provider a gateway; it does the same job as a trunk in other PBXs.

Should Register be true or false?

True. ZuteSIP authenticates each number by username and password, which requires registration.

Which SIP profile should the gateway use?

Use the external profile, which FusionPBX intends for provider connections.

Can I share one ZuteSIP number across several FusionPBX domains?

A number has one set of credentials, so register it once and route it to one domain; use separate numbers for separate tenants.

Is there a monthly minute allowance?

Each number includes 1,000 outgoing minutes and 500 outgoing SMS per month, with unlimited incoming calls and SMS.

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