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Voxbeam source IP, route prefix, DID endpoint or Asterisk driver is wrong

Trace one Voxbeam symptom through public-source authorization, route prefix, DID override, ingress acceptance, loaded PBX objects and media.

Reviewed 2026-10-22Current Voxbeam help, FAQ, DID, service, network and codec pages plus the provider Asterisk example and current Asterisk driver guidance were reviewed October 22, 2026. The account portal, current rate deck, active DID records and installed PBX documentation govern where a public example differs. No Voxbeam account, authorized source address, trunk, route class, prefix, DID, endpoint URI, channel setting, caller identity, PBX, module, firewall, packet, call, codec, fax, 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 its direction

Record local and UTC time, account, direction, number, intended route class, normalized destination, route prefix, authorized and observed public source, signaling target, DID endpoint override, trunk default, provider source, Call-ID, Request-URI, first response, matched PBX object and media result.

Classify the first divergence: wrong source authorization, stale DNS or signaling target, missing or duplicated route prefix, wrong DID override or fallback, blocked provider source, wrong PJSIP identification or context, capacity rejection, signaling success with missing media or an unsupported service assumption.

Prove source-IP authorization and the exact outbound URI

Compare the actual translated public egress address with the address authorized in Voxbeam Settings. Resolve sbc.voxbeam.com at test time, preserve the selected address and inspect the first response to the outbound INVITE. Do not wait for a registration indicator when the current public workflow documents source-IP authentication.

Then reconstruct the Request-URI from the normalized country-code destination and exactly one selected route-class prefix. Correct only the demonstrated source, destination normalization or prefix owner; do not add global digit rewrites to compensate for one malformed route.

Trace the DID override before changing PBX routing

Confirm the DID is assigned to the expected trunk and inspect its SIP Endpoint URI. If the DID field is empty, inspect the trunk default to which the FAQ says it falls back. Preserve the provider attempt, actual source packet, delivered called identity and first loaded PBX route.

A correct default does not override a populated DID-specific destination. A current firewall rule does not prove receipt, and receipt at the edge does not prove PJSIP identification, context selection or a final extension match.

Translate legacy chan_sip fields into current loaded objects

The provider PDF uses sip.conf and chan_sip-style syntax; Asterisk removed chan_sip in version 21. On a current Asterisk or FreePBX system, inventory the running release and configuration owner, then inspect the loaded PJSIP endpoint, AOR, identify, transport and dialplan rather than pasting the legacy peer.

For inbound source matching, confirm the PJSIP IP-identifier module is loaded and the provider address matches the intended identify object. Do not carry insecure=very, example secrets, host addresses or broad ACLs into production as a translation shortcut.

Separate channel, codec, fax and media evidence

If signaling is rejected under load, compare the DID package, purchased trunk channels, any DID-specific channel cap and the provider response. Do not infer a calls-per-second value from concurrency. If signaling succeeds, preserve both SDP bodies, the selected codec and DTMF payload, RTP endpoints, NAT translation and packets in each direction.

Treat T.38 as its own qualification by number, direction and carrier path. A listed codec or fax feature, answered dialog or one-way audio packet is not proof of two-way production media or service quality.

Correct one boundary and retest without widening claims

Correct only the demonstrated authorized-source, DNS target, route prefix, DID endpoint, trunk default, ingress-source rule, PJSIP identify, context, route or media mismatch. Retest one authorized non-emergency call and preserve provider transaction, edge receipt, PBX destination, answer and both media directions.

Do not represent this result as proof of TLS, SRTP, caller-ID delivery, emergency calling, failover, capacity, price, availability, compatibility or support performance. No Voxbeam 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 Voxbeam help, FAQ, DID, service, network and codec pages plus the provider Asterisk example and current Asterisk driver guidance were reviewed October 22, 2026. The account portal, current rate deck, active DID records and installed PBX documentation govern where a public example differs. No Voxbeam account, authorized source address, trunk, route class, prefix, DID, endpoint URI, channel setting, caller identity, PBX, module, firewall, packet, call, codec, fax, media stream, emergency call, bill, price, availability, compatibility or support response was inspected or tested

Voxbeam SIP trunking: source-IP authorization, route prefixes, DID endpoints and Asterisk boundaries

Separate outbound source authorization, route-class prefixing, trunk and DID destinations, channel limits, PBX driver translation and media evidence.

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SIP trunk provider research

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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Protocol reference

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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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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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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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