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GoTrunk static IP, registration, POP, DID route or FreePBX driver is wrong

Trace one GoTrunk failure through endpoint identity, registration Contact, POP selection, number assignment, loaded PBX objects, CDR evidence and media.

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

Freeze one failed transaction and identify the selected model

Record local and UTC time, account, endpoint object, direction, static or dynamic address model, portal POP, provider host and resolved address, configured and observed public source, redacted credential identity, registration Contact and expiry, number, Call-ID, first response, CDR end cause, provider trace, matched PBX object, route and media result.

Classify the first divergence: wrong source authorization, stale or absent registration, wrong POP or DNS target, inactive or misassigned number, provider attempt to the wrong address, blocked source, unloaded or mismatched PJSIP object, route mismatch, signaling success with missing media or an unsupported assumption.

Prove the static-source or credential boundary before changing both

For static-IP outbound authentication, compare the endpoint's portal value with the source address observed by GoTrunk on one INVITE. For credential mode, preserve the challenged request, redacted identity and final response. Do not add credentials to a static path or widen a source rule merely because the other model exists.

For inbound static delivery, verify the configured SIP URI and provider attempt. For dynamic delivery, verify the latest successful registration Contact and its freshness. A registered contact does not prove that the DID points to that endpoint.

Trace POP, number assignment and called identity in order

Compare the endpoint's selected POP with the host and address actually used. Preserve current DNS answers and provider source packets; do not hard-code an example address or redirect all traffic because one POP is unreachable.

Then confirm the number is active and assigned to the intended endpoint, capture the provider attempt and delivered Request-URI or called identity, and compare it with the loaded inbound route. Correct the narrow number-format or route owner rather than adding a global wildcard that hides the mismatch.

Map legacy FreePBX and sip.conf intent into loaded PJSIP objects

The public GoTrunk FreePBX example uses chan_sip and its Asterisk example targets versions 11 and 13. On a current FreePBX or Asterisk system, inventory the running versions and configuration owner, then inspect the loaded PJSIP transport, registration, auth, endpoint, AOR, identify and dialplan.

Use current Asterisk CLI and startup-log evidence to prove each object loaded. If source identification is intended, verify the actual provider address matches the intended identify object. Do not change a global bind port, paste sip.conf fields into PJSIP or broaden a firewall as a translation shortcut.

Correlate CDR, call trace, PBX signaling and both media directions

Filter the GoTrunk CDR by the frozen time and identifiers, preserve the end cause and download the provider trace where available. Match it to the edge capture and PBX log using Call-ID, number and timestamps. A missing record may be a retention, filter, clock or pre-provider failure and is not proof that no attempt occurred.

If signaling completes, preserve offer and answer, codec, DTMF payload, RTP endpoints, NAT translation and packets in each direction. Provider trace presence, a 200 response or one received RTP stream is not proof of intelligible two-way media.

Correct one demonstrated boundary and keep claims narrow

Correct only the demonstrated endpoint model, authorized address, credential, registration Contact, POP target, number assignment, source rule, PJSIP object, DID normalization, route or media mismatch. Retest one authorized non-emergency call and preserve the provider transaction, CDR, edge receipt, PBX route, answer and both media directions.

Do not represent the result as proof of emergency activation, failover, fax, TLS, SRTP, capacity, rate, availability, compatibility, SLA or support performance. No GoTrunk account, PBX, call, packet or media stream was inspected or tested.

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.

Continue your research

Provider research · Current 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 tested

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.

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Hub · Provider records reviewed individually on their listed dates; account-specific terms and availability require direct confirmation

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Compare the connection model, call routing, media support and cost structure of documented SIP trunk services.

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Plan the carrier relationship that connects a self-managed PBX to outside calling.

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Understand the difference between a normal challenge, failed credentials and an unsupported algorithm.

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Design alternate routing without assuming active calls will survive.

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Assign responsibility for location, routing and outage procedures before rollout.

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Guide · Current Asterisk PJSIP object, outbound-registration, configuration-example and res_pjsip references reviewed October 1, 2026; no account, carrier, PBX, endpoint, registration, identifier, route, call, emergency workflow, media stream, capacity or failover path was tested

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Translate each carrier direction into explicit PJSIP objects, identification, restricted dialplan, retry policy and leg-by-leg acceptance evidence.

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Guide · FreePBX 17 and current Asterisk res_pjsip documentation reviewed October 5, 2026; Sangoma's generic module pages are responsibility guides rather than an exact installed field map; no FreePBX host, module, account, credential, trunk, route, registration, number, identity, call, media path, failover, capacity, emergency workflow or compatibility was tested

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Translate current carrier requirements into FreePBX-owned PJSIP trunk, route and identity controls, then verify the generated Asterisk objects and each call direction independently.

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Define management, signaling and media access separately using actual deployment requirements.

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Select interfaces, time windows and paired capture points that distinguish network loss from missing evidence.

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Troubleshooting · First-party Asterisk documentation and SIP standard reviewed September 30, 2026; FreePBX 17 terminology retained; no PBX, provider, number, route or live call test

Inbound VoIP calls do not reach the business

Trace carrier delivery, Asterisk endpoint identification, DID matching and the final destination without collapsing the boundaries.

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Troubleshooting · Current Sangoma outbound-route guidance, current Asterisk Dial and troubleshooting documentation, and SIP standards reviewed October 5, 2026; no phone, PBX, carrier or live call test

Outbound VoIP calls fail: locate the broken call leg

Classify one failed call, then follow the exact digits through the phone, PBX route, attempted trunk, remote response, answer and media.

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Hub

VoIP troubleshooting

Use symptoms and evidence to isolate registration, routing, media and device faults.

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