VoIP.info Provider research

GoTrunk SIP trunking: static-IP, registration, point-of-presence, DID routing and current-PBX boundaries

Separate static-source authorization, credential registration, POP selection, telephone-number assignment, loaded FreePBX or Asterisk objects and call evidence.

Reviewed 2026-10-23Current GoTrunk endpoint, PBX, setup, location, CDR, rate and support pages plus current Asterisk PJSIP and FreePBX 17 migration guidance were reviewed October 23, 2026. The selected account, portal values, installed PBX release and loaded objects remain authoritative where examples differ. No GoTrunk account, endpoint, credential, authorized address, registration, point of presence, number, location, route, PBX, module, firewall, packet, CDR, call trace, media stream, emergency call, rate, invoice, availability, compatibility or support response was inspected or testedAdvancedDocumentation based · not lab tested

Choose one endpoint identity model and keep directions separate

GoTrunk's endpoint page lets an account select a static or dynamic PBX address, then choose outbound and inbound authentication independently. Static outbound traffic is associated with the configured source address; credential mode generates a username and password. Preserve endpoint object, direction, configured address or redacted credential identity, observed public source and first provider response.

A saved static address, a successful REGISTER and an accepted INVITE are different evidence. Do not combine static-IP and credential examples into one permissive trunk or assume inbound and outbound share the same owner.

Documented service facts

Reviewed 2026-10-23. Compare SIP trunk services →

Authentication

Documented

GoTrunk documents direction-specific endpoint choices. A static public source can authorize outbound traffic, while credential mode generates username and password values. Inbound can also use a configured static SIP URI or credentials and registration. Preserve endpoint, direction, selected model, observed source and first response.

SIP registration

Documented

For a dynamic-address endpoint, GoTrunk says inbound calls follow the address in the most recent registration. Static-address delivery does not require inbound registration. Preserve Contact, expiry, last successful refresh and provider attempt; registration does not prove number assignment or a loaded PBX route.

Inbound delivery

Documented

Quick Start adds an inbound number after the SIP endpoint, and the FreePBX guide maps each number to an inbound route and destination. Preserve number state, endpoint assignment, provider attempt, delivered called identity, loaded route and final destination independently.

Inbound failover

Not verified

GoTrunk's network guide says its own data-centre failover can redirect traffic and tells customers to allow every listed signaling and media address. The reviewed pages do not establish ordered customer destinations, response-code triggers, timers, off-net forwarding or established-call survival. Separate provider-site continuity from customer PBX and access-path resilience, then obtain the exact account behavior before testing.

Number and regional availability

Not verified

GoTrunk says its portal offers numbers across more than 4,000 cities and supports purchases and ports, but exact inventory, portability, activation time, documents and feature combinations remain number-, account- and date-specific. Some DIDs can remain pending for support processing. Verify the exact locality, number type, status and port requirements before scheduling a migration.

Audio codecs and DTMF

Not verified

The reviewed current endpoint, network and Asterisk pages do not publish a complete customer codec and DTMF support matrix. An illustrative PJSIP template or open RTP path is not evidence of the codec, payload, packetization or digit method negotiated on a production route. Obtain the current interop profile and capture SDP and digit behavior on representative authorized calls.

T.38 fax

Not verified

The reviewed current service, endpoint, network and telephone-number pages do not establish T.38 availability, audio-fax fallback or a supported fax workflow. Ordinary voice delivery does not qualify fax. Confirm the exact number, direction, route, PBX or gateway, initial codec, redundancy mode and document set before relying on fax service.

Signaling and media encryption

Not verified

The current network guide lists TCP and UDP SIP signaling plus RTP address ranges, while the Asterisk example uses UDP. The reviewed pages do not establish a customer TLS identity, certificate-validation procedure, SRTP profile or downgrade policy. Source-IP authorization, NAT handling and firewall allowlisting are not encryption; obtain protected-transport requirements in writing and verify the negotiated leg.

Concurrent calls and burst

Not verified

The telephone-number guide says incoming channels can be increased for an additional monthly charge, but the reviewed pages do not define one account's simultaneous-call, calls-per-second, burst, queueing or abuse limits. Number count and SIP endpoints are not capacity. Obtain active limits, direction, rejection behavior, upgrade timing and billing before an authorized load test.

Emergency calling

Documented

In some locations, GoTrunk requires a verified geographic address linked to one telephone number selected as Emergency Caller Id. Confirm current country and number eligibility, validation, callback identity and provider-approved test procedure; do not infer activation from an ordinary call.

Pricing model

Not verified

GoTrunk advertises per-second billing, no setup fee and no contract, and its portal exposes destination rates. The number guide also notes a monthly charge for added incoming channels. These statements do not establish an account's current numbers, destinations, currency, billing increments, commitments, porting, emergency, support, taxes or renewal terms. Preserve a dated quote and rate export for the actual traffic mix.

Administration and support

Documented

The portal documents endpoint configuration, CDR filtering, end causes, CSV export, downloadable call traces and severity-tagged support tickets with unique references. Preserve redacted portal state, CDR and trace identifiers, matching PBX evidence and support record; none establishes media success or SLA performance.

SMS and MMS

Not verified

The reviewed current SIP-trunk, telephone-number and portal pages do not establish SMS or MMS on a GoTrunk number. Voice capability does not prove messaging eligibility, sender registration, consent controls, API delivery, throughput or fees. Verify the exact number, country and messaging program independently.

Treat registration as a current inbound destination, not service acceptance

The dynamic-address FreePBX and Asterisk examples say inbound calls follow the address in the most recent registration. Preserve the account-issued inbound identity, registrar or selected POP, Contact, expiry, last successful refresh and the destination actually attempted by the provider. A 200 response to REGISTER does not prove that a telephone number is assigned, that the next attempt reaches the edge or that a loaded inbound route accepts it.

The static-address model sends calls to the configured SIP URI without requiring an inbound registration. Diagnose the selected model rather than waiting for a registration state that is not part of it.

Bind POP, DID assignment and PBX route as separate decisions

The endpoint workflow includes a point-of-presence choice, and the PBX guides show Europe and North America host examples. Preserve the portal-selected POP, current DNS answers, address actually used, provider source and latency evidence instead of treating a documentation hostname as a permanent global default.

The quick-start workflow adds an inbound number only after the endpoint, while the FreePBX guide requires a matching DID Number and chosen destination. Keep number ownership and activation, endpoint assignment, delivered called identity, provider attempt, loaded route and final destination separate.

Translate legacy chan_sip examples into the installed platform owner

GoTrunk's public FreePBX page still instructs users to add a chan_sip trunk and discusses the historical 5160 bind-port pattern. Its Asterisk page targets versions 11 and 13 and uses sip.conf. Current Asterisk guidance describes migration to res_pjsip, while FreePBX 17 guidance says chan_sip trunks restored into a system without chan_sip are converted and must be thoroughly tested.

Treat provider pages as evidence of identity, host, route and dial-string intent, not a current paste-in template. Inventory the installed FreePBX and Asterisk releases, configuration owner and loaded PJSIP transport, registration, auth, endpoint, AOR, identify and dialplan objects before changing ports or ACLs.

Use provider CDR and trace evidence without promoting it into media success

GoTrunk documents CDR filters, end-cause indicators, CSV export and a downloadable call trace for Wireshark. Preserve time, direction, account, endpoint, provider call identifier, SIP Call-ID, number, end cause, provider trace and matching edge or PBX record. Confirm that clocks and retention cover the incident before declaring a record missing.

A provider CDR or successful SIP dialog does not establish the loaded PBX route, answer, selected codec, RTP path or intelligible two-way audio. Preserve both SDP bodies and packets in each direction when media is the symptom.

Keep emergency, rates, capacity and support within documented scope

GoTrunk documents emergency calling only in some locations, with a verified geographic address linked to one telephone number as the emergency caller ID. Confirm current country and number eligibility, address validation, callback identity and the provider-approved test method; do not place a live emergency test without authorization.

The rate page exposes termination codes and rates, while support uses severity-tagged tickets with a unique reference. These surfaces do not prove one account's current price, capacity, SLA, availability, compatibility or response time. Preserve a dated quote, active limits and actual support record instead of reproducing volatile commercial claims.

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.