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SIPTRUNK is registered but inbound calls or audio fail

Trace the selected authentication mode, registered Contact, provider attempt, PBX route and direct-media RTP without treating portal registration as end-to-end proof.

Reviewed 2026-10-19Current first-party SIPTRUNK device, registration, inbound-call, IP-authentication, failover, registration-status, interconnection and direct-media documentation was reviewed October 19, 2026. Credential registration and static-IP authentication remain different service paths; portal registration, Contact address, observed public source, inbound route, signaling gateways, direct-media RTP sources and failover treatments remain separate evidence. No SIPTRUNK account, trunk, credential, registration, number, route, gateway, DNS answer, PBX, firewall, packet, media stream, failover event, price, availability, compatibility or live call was inspected or testedAdvancedDocumentation based · not lab tested

Freeze one failed inbound call and one bounded control

Record timestamp and timezone, ordered trunk, authentication mode, DID, provider gateway, portal registration, Contact and observed source, provider attempt, edge receipt, PBX response, route destination, answer, SDP, RTP endpoints and user-visible audio. Classify the earliest gap: no current destination, wrong Contact, no attempt, blocked signaling, unmatched route, rejected call, no RTP or one-way RTP.

Redact credentials, numbers, addresses and call content. A working outbound call or green registration indicator is useful control evidence, not proof of inbound delivery or media.

Confirm the selected authentication and destination model

If the trunk uses credential registration, preserve the 401 challenge, authenticated REGISTER, 200 response, accepted expiry and current portal row. If it uses IP authentication, verify the stored public IP and port instead of diagnosing the intentional absence of REGISTER.

Do not enable both paths or change credentials, NAT and inbound routes together. Correct the selected service boundary first and retain a rollback.

Compare Contact with the public source observed by SIPTRUNK

For the registration path, compare the Contact IP and port with the alias or public source the provider reports. SIPTRUNK documents that inbound traffic is attempted toward Contact. A private, stale or remapped Contact can coexist with a successful registration and still send an inbound INVITE to the wrong place.

Confirm DNS, selected gateway, NAT ownership, port mapping and registration refresh. Do not guess a public address from a different node or time interval.

Trace the provider attempt into the loaded inbound route

Join the portal or support attempt to an edge capture by timestamp. Preserve source gateway, transport, Request-URI, To and From. SIPTRUNK documents an eleven-digit To user by default for inbound North American calls, while From normally uses ten digits. Compare the actual To with endpoint identification, context and the first loaded Asterisk or FreePBX match.

If nothing reaches the edge, remain at provider routing, Contact or network policy. If the PBX responds, retain the first final SIP result before changing destinations or number normalization.

Trace direct-media RTP independently of SIP

After answer, read the actual SDP addresses and ports and identify the first RTP packet in each direction. SIPTRUNK documents that RTP can arrive from addresses other than its signaling gateways. A firewall limited to gateway addresses can therefore pass SIP while dropping direct media.

Change only the demonstrated media rule within the approved security design. Do not expose an arbitrary UDP range merely because the provider article mentions varied sources; preserve segmentation, active-call validation, state, rate controls and the PBX’s configured RTP range.

Correct the earliest mismatch and test recovery separately

Correct only the first proven divergence: wrong service mode, expired binding, stale Contact, port translation, DNS or gateway choice, DID assignment, blocked SIP, endpoint identification, context, number match, SDP address or RTP policy. Repeat the same call and confirm answer plus both audio directions.

Then test only the authorized Secondary Trunk, PSTN Backup, PSTN Forward or voicemail treatment, recording trigger and restoration separately. Do not infer active-call survival, recovery time or future availability. No SIPTRUNK account, registration, PBX, firewall, packet, media stream, failover event or live call was inspected for this diagnostic.

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 first-party SIPTRUNK device, registration, inbound-call, IP-authentication, failover, registration-status, interconnection and direct-media documentation was reviewed October 19, 2026. Credential registration and static-IP authentication remain different service paths; portal registration, Contact address, observed public source, inbound route, signaling gateways, direct-media RTP sources and failover treatments remain separate evidence. No SIPTRUNK account, trunk, credential, registration, number, route, gateway, DNS answer, PBX, firewall, packet, media stream, failover event, price, availability, compatibility or live call was inspected or tested

SIPTRUNK registration, inbound Contact routing and direct-media evidence

Keep credential registration or IP authentication, Contact delivery, inbound number matching, gateway redundancy, direct-media RTP and failover treatment independently observable.

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

SIP trunk provider research

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

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Guide

SIP trunking: numbers, capacity and operational control

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

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Protocol reference

SIP REGISTER explained

Understand contact bindings, authentication challenges and why registration does not prove calling works.

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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

Plan and configure an Asterisk SIP trunk

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

Configure and verify a FreePBX 17 PJSIP trunk

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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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 · Asterisk 22 LTS · FreePBX 17 / Debian 12 checks

VoIP call connects but has no audio

Separate endpoint audio selection from a media path that fails in both directions.

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Troubleshooting · Asterisk 22 LTS · FreePBX 17 / Debian 12 checks

One-way VoIP audio

Trace the missing direction from advertised media address to receiving endpoint.

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Guide

Firewall rules for VoIP

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

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Guide · Wireshark 4.x / current upstream tool documentation; inspect installed versions

VoIP packet-capture methodology

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

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Guide · Wireshark 4.x / current upstream tool documentation; inspect installed versions

Find RTP streams and interpret loss and jitter

Separate captured packet statistics from what the receiving listener actually experienced.

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Protocol reference

SDP for VoIP: read addresses, ports and media direction

Interpret the session description without confusing an offered parameter with a working network path.

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