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Simwood Hosted SIP Trunks: inbound registration and routing evidence

Separate outbound digest success from inbound registration, Contact-user addressing, called-number routing, PBX identification and current service-specific network policy.

Reviewed 2026-10-19Current first-party Simwood Hosted SIP Trunks, interop, trunk overview, Asterisk 16 sample, hosted TLS/SRTP, Partner and Direct whitelisting and portal material were reviewed October 19, 2026. The batch keeps Hosted, Partner, Direct, Wholesale and Carrier Services scopes separate; outbound digest authorization separate from inbound registration; Contact user separate from the called-number value in To; configured routes separate from observed delivery; and the version-scoped Asterisk 16 example separate from current FreePBX or Asterisk configuration. No Simwood account, trunk, credential, number, registration, PBX, FreePBX route, DNS answer, firewall, packet, media stream, failover event, price, availability, compatibility, capacity or live call was inspected or testedIntermediateDocumentation based · not lab tested

Keep Hosted and carrier service scopes separate

Simwood publishes Hosted SIP Trunks alongside Carrier Services, Wholesale, Partner and Direct material. Start with the ordered product, account portal, assigned server and authentication model. Do not use a Carrier Services or Wholesale template to relax the boundary for a Hosted trunk, and do not copy Partner or Direct firewall data into another service without current confirmation.

The Hosted overview describes per-trunk credentials and allocated Trunk Channels. These public properties do not establish one account's current credentials, number assignment, concurrency, price or availability.

Documented service facts

Reviewed 2026-10-19. Compare SIP trunk services →

Authentication

Documented

Hosted outbound can answer a 401 digest challenge without a REGISTER, while Hosted inbound requires a current registration. Keep outbound authorization, registration and Carrier, Wholesale, Partner or Direct account models separate.

SIP registration

Documented

Hosted documentation shows REGISTER, challenge and authenticated REGISTER. The Contact user can shape the inbound Request-URI user. Preserve the assigned server, transport, accepted expiry, Contact and provider-observed binding; a working outbound call is not inbound registration evidence.

Inbound delivery

Documented

The Hosted interop guide distinguishes a Contact-derived Request-URI user from the called number in To. Preserve both values, the provider attempt, source, PJSIP identification, context and first loaded route match before normalizing a dialed identity.

Inbound failover

Documented

The current product page lists time-of-day routing, ring timeouts, failover options and forwarding. Public sources reviewed do not establish one universal trigger matrix, timer, destination order, restoration interval or established-call survival result.

Number and regional availability

Not verified

Simwood publishes UK number allocation and porting procedures for Carrier Services, while the reviewed Hosted SIP Trunk pages do not prove inventory, portability or service eligibility for a requested account, number type and address. Obtain a dated orderability and port result for the exact service rather than transferring wholesale API coverage to a hosted account.

Audio codecs and DTMF

Not verified

Simwood's hosted Asterisk 16 example permits PCMU and PCMA, but that dated sample does not establish the full current codec set, DTMF mode, packetization or transcoding behavior for every trunk. Confirm the current account profile and inspect the actual SDP offer and answer and telephone-event payload on authorized calls.

T.38 fax

Not verified

The reviewed Hosted SIP Trunk product, interop and sample-configuration pages do not establish T.38 support or a fax acceptance workflow. Qualify fax separately using the exact number, direction, codec, gateway, document set, switchover behavior and fallback; ordinary voice success is not fax evidence.

Signaling and media encryption

Documented

Hosted documentation describes TLS on 5061 and SDES SRTP settings. Verify the assigned host, certificate validation, negotiated signaling transport, SDP crypto result, media endpoints and downgrade policy. Configuration or an open port is not proof of a protected call.

Concurrent calls and burst

Documented

The hosted trunk overview says simultaneous calls are governed by allocated Trunk Channels. The reviewed public pages do not establish an account's channel count, calls-per-second limit, burst behavior, rejection response or fair-use terms. Preserve active-call capacity and call-start rate as different controls and obtain both from the current account and contract.

Emergency calling

Not verified

The reviewed Hosted SIP Trunk pages do not establish emergency-calling eligibility, address registration, outbound identity, callback routing or a provider-approved test procedure. Confirm every element for the exact account, number and location, retain an independently tested fallback, and do not place a live emergency test without provider and local authorization.

Pricing model

Not verified

The service page describes SIP trunking but does not publish a complete current account quote. Obtain dated terms for channels, inbound and outbound traffic, destinations, numbers, porting, failover forwarding, emergency service, taxes, support, commitments, renewal and termination. Do not transfer wholesale or deprecated hosted rate documents to the current product.

Administration and support

Documented

The Asterisk example is explicitly scoped to Asterisk 16, supplied as-is and unsupported by Simwood. Its older short source-address list is not a current firewall policy; the current Partner and Direct page lists more ranges, excludes Wholesale and says its port list is not all-inclusive. Obtain current Hosted policy.

SMS and MMS

Not verified

Simwood documents messaging elsewhere, but the reviewed Hosted SIP Trunk pages do not establish SMS or MMS entitlement on trunk numbers. Voice service does not prove number eligibility, sender registration, consent controls, API delivery, throughput, retention or fees. Verify a separately identified messaging product and exact number before representing support.

Outbound success does not prove an inbound binding

The hosted interop guide documents outbound INVITE authorization through a 401 digest challenge without requiring registration. The same guide says registration is required for inbound calls. Preserve the outbound challenge and result as a bounded control, then separately capture the assigned registrar, REGISTER challenge, authenticated REGISTER, accepted expiry, Contact and current binding.

A working outbound call proves neither that the provider has a current inbound Contact nor that the called number selects the expected trunk. Use the paired diagnostic to locate the first missing boundary.

Separate Contact-user addressing from the called number

For Hosted trunks, the interop guide says the registered Contact user can shape the user in the inbound INVITE Request-URI. It separately says the To user carries the callee number that should be used for routing. Record Request-URI, To, source, transport, endpoint identification, context and the first matching FreePBX or Asterisk route.

Do not broaden source trust or create a catch-all route simply because the two user values differ. First prove whether the mismatch belongs to the registration Contact, provider number route, PJSIP endpoint, context or dialplan normalization.

Treat the Asterisk 16 sample as version-scoped evidence

Simwood labels its sample as an Asterisk 16 PJSIP example, provides it as-is and says it does not support Asterisk. Its registration, auth, AoR, endpoint, identify and header-parsing pattern can explain object ownership, but it is not a current universal recipe for Asterisk or FreePBX. Compare it with the installed branch, generated configuration and loaded objects before adapting anything.

The sample contains an older, short source-address list. Simwood's current whitelisting page, updated October 31, 2025, lists substantially more ranges, limits itself to Partner and Direct, excludes Wholesale and says the ports are not all-inclusive. Neither page establishes a complete Hosted firewall policy; obtain current service-specific values.

Keep signaling, media and failover evidence distinct

The hosted secure-calling article documents TLS on 5061 and SRTP settings using SDES. Confirm the assigned host, certificate identity and chain, peer validation, negotiated transport, SDP crypto result, RTP endpoints and fallback policy. An open TLS port or configured SRTP option is not proof that a completed call protected both signaling and media.

The current product page advertises ring timeouts, time-of-day routing, failover options and forwarding. The reviewed public material does not establish one universal response-code matrix, timer, destination order, restoration interval or established-call survival result. Freeze the portal rule and test only an authorized new-call condition.

Run one bounded acceptance trace

For one authorized number and time window, record product scope, portal route, registrar and binding, provider attempt, customer-edge receipt, Request-URI, To, endpoint identification, context, route match, answer, SDP and each media direction. Preserve timestamps and redacted correlation identifiers rather than credentials or call content.

Confirm current number eligibility, portability, emergency workflow, messaging, fax, codecs, DTMF, channel allocation, call-start rate, pricing and support for the exact order. No account, PBX, registration, packet, media stream or live call was inspected for this guide.

Diagnose outbound success with missing inbound delivery

Use the Simwood inbound diagnostic to join Hosted registration, Contact-user addressing, called-number routing and loaded PBX state.

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.