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Voipfone SIP trunks: registration, number routing and fallback evidence

Separate the account registration, routed telephone number, PBX edge, inbound match, call diverts, PSTN fallback, media and caller outcome.

Reviewed 2026-10-17Current first-party Voipfone registration, third-party PBX, number-routing, call-divert, porting, emergency-address and terms documentation reviewed October 17, 2026. Account-number registration, telephone-number routing, PBX delivery and customer-side fallback remain separate controls. Portal labels and legacy chan_sip examples do not establish a current PJSIP profile, one universal delivered DID form or the behavior of an uninspected account. No account, credential, number, port, route, divert, extension, PBX, registration, provider attempt, packet, media stream, emergency workflow or live call was inspected or testedAdvancedDocumentation based · not lab tested

Inventory the account registration and telephone numbers separately

Voipfone’s Asterisk and FreePBX examples register an account number and password against sip.voipfone.net. The FreePBX guide explicitly says the account number used in trunk configuration is not one of the account’s Voipfone telephone numbers. Keep the registrar identity, authentication identity, Contact, account number, each telephone number, port state and routing owner as separate records.

The published examples are chan_sip-era starting points, not a current PJSIP security profile or proof of the fields active on one account. Preserve the raw REGISTER request, challenge, successful response, expiry and effective Contact without exposing credentials.

Documented service facts

Reviewed 2026-10-17. Compare SIP trunk services →

SIP authentication

Documented

Voipfone’s Asterisk and FreePBX examples use an account number and password at sip.voipfone.net. The account number is not one of the account’s telephone numbers; preserve authentication, number and routing identities separately.

SIP registration

Documented

The published examples show account-number registration at sip.voipfone.net. They are chan_sip-era starting points, not a current PJSIP profile or proof of the expiry, Contact, concurrent-registration or source policy active on one account.

Inbound delivery

Documented

The control panel routes individual telephone numbers to destinations including extensions, voicemail, mobiles, queues, groups and IVRs. Provider delivery and the PBX’s first matching inbound rule remain separate evidence.

Routing and failover

Documented

Voipfone documents Always, closed, busy, no-answer and PSTN-failover controls. The reviewed pages do not publish one complete trigger, timer, retry, precedence or restoration contract for every account.

Number and regional availability

Documented

Voipfone advertises UK geographic, 03 and 0800 numbers and says it can port many UK landline numbers but not mobile numbers. This does not establish current inventory, portability, identity rights, serviceability or regulatory eligibility for a particular address or number. Preserve dated order and port responses for the exact requirement.

Audio codecs and DTMF

Documented

The legacy FreePBX example lists A-law and RFC2833 DTMF, while the Asterisk page also labels RFC2833 for its chan_sip peer. Those examples do not establish a current PJSIP codec, packetization, RTP-address, DTMF or transcoding profile. Obtain the current interface record and preserve both SDP bodies, negotiated payloads and decoded digit evidence.

T.38 fax

Not verified

Voipfone markets fax services separately, but the reviewed trunk and PBX pages do not establish T.38 or G.711 pass-through entitlement, switchover, redundancy, gateway support or page limits for a self-hosted PBX trunk. Treat fax as a separately ordered and accepted workflow tied to the exact service, gateway and destination.

Signaling and media encryption

Not verified

The reviewed trunk pages and PBX examples do not publish SIP TLS, certificate validation, SRTP, keying or a protected end-to-end path. The provider's warning about internet-facing PBX compromise is not an encryption profile. Secure the PBX independently and obtain the current trunk protection specification before representing signaling or media as protected.

Concurrent calls and call starts

Documented

Voipfone markets its SIP trunks as free and unlimited and says simultaneous calls are limited only by bandwidth to the service. That public statement does not establish an account call-start rate, burst policy, destination control, PBX or SBC ceiling, codec bandwidth or successful load result. Keep active calls, attempted starts, access bandwidth and system resources as separate measured quantities.

Emergency calling

Documented

Voipfone says customers can call 999 and 112 and can register a service address against inbound numbers. Its terms and emergency page warn that calls may fail during power or broadband loss, and the customer must keep the location current. Confirm every site's numbers and addresses, use the provider's approved validation process and maintain an independently powered alternative calling method.

Pricing model

Documented

Voipfone calls the SIP trunk connection free and sells monthly plans, Flex components, numbers, calling credit and usage. That marketing label is not a complete account price. Use a dated order or quote for numbers, porting, extensions, minutes, usage, options, hardware, broadband, support, taxes, term, renewal and cancellation.

Administration and support

Documented

The control panel owns number routing and call-divert controls, while the PBX owns registration and inbound matching. Preserve portal changes, effective times, call identifiers, edge evidence and rollback separately.

SMS and MMS

Not verified

Voipfone advertises SMS and mobile services elsewhere in its portfolio, but the reviewed trunk pages do not establish SMS or MMS entitlement on self-hosted-PBX trunk numbers, sender registration, API access, throughput, webhook security, retention or price. Treat messaging as a separately ordered service and workflow.

Treat number routing as a portal control before PBX delivery

Voipfone’s current control-panel documentation says each inbound telephone number can be routed to a selected destination such as an extension, voicemail, mobile, call queue, call group or IVR. That number-level selection is distinct from the account registration and from the PBX rule that handles an INVITE after delivery.

Record the called number, effective portal destination, timestamp, provider call identifier, first provider target, edge receipt, delivered Request-URI and To identity, normalized DID, first matching PBX route and final destination. A registered trunk does not prove that a particular number is assigned, ported, routed or delivered to the intended PBX object.

Keep PBX inbound matching narrower than provider routing

The current FreePBX example creates an inbound route using the Voipfone account number, while the portal documentation describes separately routed telephone numbers. Preserve that source distinction instead of assuming every account delivers the same identifier or format.

Inspect the actual INVITE and the first matching context or inbound route. Caller identity, line-presentation labels, an account number and the dialed telephone number can appear in different fields; none should be silently substituted for the PBX match key. Provider delivery does not prove the extension, queue, group, IVR or voicemail answered.

Model diverts and fallback as explicit downstream controls

Voipfone documents Always, When Busy, No Answer, When Closed and PSTN-failover choices in its call-divert controls. It describes PSTN failover as a destination used when the phone or PBX loses connectivity. These are configured call-routing controls, not evidence that an unobserved outage triggered the intended path.

Record which rule was enabled, its destination, precedence, charge owner, trigger evidence, provider attempt, any PBX response and caller outcome. An Always divert can bypass PBX delivery; a no-answer divert requires a delivered and unanswered attempt; a connectivity fallback is not an established-call survival guarantee.

Trace signaling success and media success independently

After a delivered INVITE, retain the provider source, transport, Request-URI, called-number form, PBX context, matched rule, response sequence and release cause. Then compare both SDP bodies, negotiated codec, RTP endpoints, NAT and firewall decisions, packet flow in both directions and an authorized listening result.

The legacy PBX examples list A-law and RFC2833 DTMF but do not establish a universal current codec, DTMF, TLS, SRTP or packetization contract. A registered account, correct number route or answered extension does not prove two-way media.

Bind porting, emergency and commercial state to dated evidence

A port request, successful registration and portal routing are separate lifecycle records. Preserve the port order, losing-provider acceptance, completion notice, number inventory and authorized acceptance call. For emergency calling, keep each number’s current address, site ownership, approved validation method and an independently powered alternative because Voipfone warns that power or broadband failure can prevent 999 or 112 calls.

Use a dated account record for numbers, extensions, options, usage, support, taxes, credits and diversion charges. No Voipfone service, route, fallback, emergency workflow, price, availability, compatibility or live call was tested for this guide.

Trace one Voipfone number from portal routing outward

Use the Voipfone inbound-number diagnostic to prove number ownership, portal destination, account registration, provider attempt, PBX match, divert precedence and media.

Sources & applicability

Primary references for the technical details above. Operational examples and planning checklists are VoIP.info editorial guidance.

Examples require adaptation to your topology. No live PBX or hardware testing is claimed.