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DIDforSale has split inbound calls, no outbound calls or the wrong registration route

Trace one DIDforSale failure through interconnection model, portal state, registration or source authorization, gateway receipt, DID format, PBX route and media.

Reviewed 2026-10-22Current DIDforSale IP-interconnection, SIP-account, FreePBX PJSIP, product and customer-agreement pages plus the current Asterisk chan_sip status page were reviewed October 22, 2026. The current account, portal worksheet, order and installed PBX documentation govern where a public example differs. No DIDforSale account, portal role, IP or FQDN record, termination flag, SIP account, credential, registration, DNS answer, gateway, DID route, source packet, international-calling setting, limit, PBX, firewall, call, media stream, emergency call, bill, price, availability, compatibility or support response was inspected or testedAdvancedDocumentation based · not lab tested

Freeze one failed transaction and classify the handoff

Record local and UTC time, account, PBX and installed driver, interconnection model, portal IP or FQDN record, termination flag, SIP-account activation and registration state, DID, entered and normalized digits, caller identity, Call-ID, packet source, Request-URI, first response, matched route and media result.

Classify the first divergence: wrong interconnection model, inactive SIP account, failed registration, unapproved or changed source address, termination disabled, international calling disabled, account limit, missing provider gateway, wrong DID format, wrong PBX context or signaling success with missing media.

Prove IP authorization or SIP registration, not both by assumption

For IP interconnection, compare the observed public source with the portal-approved IP or current FQDN resolution and confirm termination is enabled for outbound use. For a SIP account, confirm activation, assigned domain, sanitized username ownership and the matching REGISTER response.

Do not treat a green SIP-account indicator as proof of IP authorization, or an allowed address as proof of registration. Correct only the configured model and remove unintended duplicate paths through the normal change process.

Trace outbound format, policy and the first provider response

Compare entered digits, PBX normalization, outbound route and the provider-leg Request-URI. DIDforSale documents 1 plus ten digits for US and Canada and 011 plus country code and number for international calls. It also says international calling is disabled by default and new accounts can have a one-call outbound limit.

Identify the responder and exact status, timeout or concurrency symptom before changing routes. A format correction will not repair a disabled service, wrong source authorization, account limit or caller-identity policy.

Correlate intermittent inbound failure with every gateway source

Filter authorized test attempts by DID, time and Call-ID, then compare the source address with the current DIDforSale gateway list. If attempts from one documented source reach the edge and another does not, inspect the narrow firewall, SBC and endpoint-identification rules for that exact source before widening anything.

If every provider attempt reaches the edge, follow the delivered called identity through context, normalization and the first loaded route. Provider selection, packet acceptance and PBX matching are three separate evidence states.

Apply version boundaries to PBX examples

The reviewed DIDforSale FreePBX PJSIP examples identify version 14.0.5.2. Other examples in the documentation set use chan_sip, which Asterisk removed in version 21. Inventory the running distribution, generated-configuration owner and loaded PJSIP objects before translating the provider handoff.

Preserve current DID formatting from the actual packet. Do not copy example secrets, addresses, object names or permissive matching rules into production without reconciling current provider and platform documentation.

Correct one boundary and retest signaling and media

Correct only the demonstrated portal record, termination flag, SIP-account activation, registration, source allow rule, number format, international setting, account limit or PBX-route mismatch. Retest one authorized non-emergency call and preserve the provider transaction, customer-edge receipt, final PBX destination, answer and packets in both media directions.

One successful call does not establish availability, compatibility, capacity, encryption, fax, pricing, emergency behavior or support outcome. No DIDforSale account, gateway, 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 DIDforSale IP-interconnection, SIP-account, FreePBX PJSIP, product and customer-agreement pages plus the current Asterisk chan_sip status page were reviewed October 22, 2026. The current account, portal worksheet, order and installed PBX documentation govern where a public example differs. No DIDforSale account, portal role, IP or FQDN record, termination flag, SIP account, credential, registration, DNS answer, gateway, DID route, source packet, international-calling setting, limit, PBX, firewall, call, media stream, emergency call, bill, price, availability, compatibility or support response was inspected or tested

DIDforSale SIP trunking: IP authorization, registration, gateways and DID routing evidence

Separate portal-approved IP or FQDN interconnection, SIP-account registration, outbound termination enablement, multiple inbound gateway sources, DID format and PBX routing.

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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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SIP trunk IP authentication

Understand source-address trust and the consequences of a WAN address change.

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SIP trunk failover and redundant calling paths

Design alternate routing without assuming active calls will survive.

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Planning emergency calling for a VoIP deployment

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 · 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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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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Hub

VoIP troubleshooting

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

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