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OnSIP trunk has the wrong 000 prefix, registration, inbound failover or number transfer

Trace one OnSIP call through trunking-user state, registration, outbound digit transformation or inbound number state, provider routing, PBX match and media.

Reviewed 2026-10-22Current OnSIP trunking, enablement, inbound-call-preference, E911 and Asterisk support pages plus the current Asterisk chan_sip status page were reviewed October 22, 2026. OnSIP account state and the applicable current PBX documentation govern where a public example differs. No OnSIP account, role, trunking user, credential, registration, DNS answer, number transfer, route, ring timer, external destination, PBX, dialplan, firewall, packet, call, media stream, capacity event, emergency call, bill, price, availability, compatibility or support response was inspected or testedAdvancedDocumentation based · not lab tested

Freeze one call and classify the first divergence

Record local and UTC time, account and PBX, trunking user, installed channel driver, registration state, exact entered digits, normalized digits, caller identity, number-transfer state, saved ring timer and external destination. For the matching transaction retain the provider Request-URI, Call-ID, first response, edge receipt, PBX route and media result.

Classify the first divergence: trunking flag or registration, missing or duplicate 000 transformation, wrong identity, number not transferred, provider attempt absent, PBX rule pre-empts failover, wrong PBX destination, or signaling success with missing media.

Confirm the loaded driver before using a configuration example

Do not paste OnSIP's sip.conf sample into a current PJSIP or FreePBX deployment. It documents legacy chan_sip objects and commands, while Asterisk removed chan_sip in version 21. Inventory the running Asterisk or distribution version, loaded channel driver, generated configuration owner and effective objects.

Translate the provider-assigned username, authentication username, domain, proxy and secret into the current driver model. Then confirm that the intended outbound registration is loaded and that a sanitized REGISTER and response match the expected account; never expose reusable credentials in captures or tickets.

Trace the 000 flag without changing user-facing numbering

Compare the user-entered number, dialplan normalization, outbound-route match and provider-leg Request-URI. OnSIP documents 000 as a required trunking flag, and its Asterisk checklist separates that flag from successful authentication and caller identity. Apply it once at the OnSIP trunk boundary and preserve the ordinary destination for user-facing records.

If a call fails, identify the responder and exact first status or timeout. Do not attribute every rejection to the prefix; registration, authentication, route selection, caller identity and destination policy remain independent.

Prove transfer state and inbound PBX matching in order

OnSIP warns that transferring an existing PSTN Gateway number immediately changes where it rings. Compare the number inventory, transfer action and timestamp with the provider attempt. If the attempt reaches the edge, follow its called identity through endpoint identification, context, digit normalization and first matching inbound route.

A saved number, an accepted firewall packet and a matching route are three different evidence states. Correct the earliest demonstrated mismatch rather than adding broad catch-all routes.

Separate ring-time failover from local forwarding

Preserve the OnSIP ring-time and external-number selection, then compare the initial PBX attempt, provisional and final responses, local forwarding behavior and any later external attempt. OnSIP explicitly warns that PBX inbound rules may conflict with or pre-empt the provider preference.

Do not assume a universal timer, response-code trigger or restoration sequence from the public article. Guard against loops and use only an authorized, non-emergency test scenario.

Correct one boundary and retest signaling and media

Correct only the demonstrated account, registration, digit transformation, caller identity, transfer, failover or PBX-route mismatch. Retest an authorized outbound call and inbound number, preserving registration, provider transaction, PBX destination, answer and packets in both media directions.

Keep Hosted Trunking emergency calling out of this test: OnSIP says its E911 feature is unavailable to Hosted Trunking subscribers. One successful call does not establish availability, compatibility, capacity, encryption, fax, pricing, emergency behavior or support outcome. No OnSIP account, 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 OnSIP trunking, enablement, inbound-call-preference, E911 and Asterisk support pages plus the current Asterisk chan_sip status page were reviewed October 22, 2026. OnSIP account state and the applicable current PBX documentation govern where a public example differs. No OnSIP account, role, trunking user, credential, registration, DNS answer, number transfer, route, ring timer, external destination, PBX, dialplan, firewall, packet, call, media stream, capacity event, emergency call, bill, price, availability, compatibility or support response was inspected or tested

OnSIP Trunking: registration, 000 prefix, number cutover and failover evidence

Separate the trunking-user registration, required 000 outbound flag, immediate number-transfer cutover, inbound ring timer, PBX routing and Hosted Trunking E911 exclusion.

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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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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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SIP trunk failover and redundant calling paths

Design alternate routing without assuming active calls will survive.

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Business number porting: a cutover checklist

Keep ownership records, dependencies and a fallback plan aligned with the gaining provider’s process.

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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

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 · Asterisk 22 LTS

Asterisk dialplan: contexts, patterns and loaded-call evidence

Build a restricted Asterisk 22 call path, then prove its context, match, priorities and runtime behavior separately.

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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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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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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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VoIP troubleshooting

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

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