Freeze one trunk, one direction and one failed attempt
Record the trunk identifier, timestamp and timezone, call direction, dialed number, calling identity, PBX and version, customer public source address and port, selected SIPTRUNK gateway, portal mode, portal status, number assignment, first SIP request and response, first PBX route and any failover result. Redact credentials, Authorization values and customer addresses before sharing evidence.
Classify the symptom before changing settings: no REGISTER leaves the PBX, REGISTER receives no response, authentication is rejected, portal shows registered but inbound fails, IP-authenticated traffic comes from the wrong address, outbound fails after mode change, or the call enters an unexpected treatment. These are different boundaries.
Prove the configured authentication mode before reading status
SIPTRUNK documents credential registration and IP authentication as mutually exclusive on one trunk. In registration mode, expect a REGISTER, a 401 credential challenge, an authenticated retry and a final result. In IP-authentication mode, the provider accepts outbound traffic only from the stored public address and delivers inbound calls only to the stored address and port.
Do not treat the absence of a registration indicator as an outage when the portal shows IP Auth. Conversely, do not assume a successful digest challenge proves the portal still assigns that trunk to registration mode. Preserve the portal setting and the PBX behavior from the same time window.
Trace DNS, gateway reachability and the first provider response
For registration, prove that the PBX generated REGISTER, resolved the intended gateway, sent it through the expected public address and received the provider response. SIPTRUNK publishes several gateway names; the selected DNS answer, route, source translation and return path must belong to the same attempt.
A 401 can be the expected challenge, while a later rejection, a 407 associated with the wrong IP-authentication address or port, silence, or an absent authenticated retry points to a different failure. Preserve headers without secrets rather than rotating credentials and firewall policy together.
Compare REGISTER Contact with the observed source address
In registration mode, SIPTRUNK displays the address and port from the Contact alongside the public source it observed. The provider says inbound delivery uses the Contact value. Compare both address-and-port pairs, the PBX external-media and external-signaling policy, firewall translation and the current access address.
A 200 registration result is not proof that the Contact is publicly reachable or that a number is assigned to the trunk. If the Contact is private, stale or translated to a different port, correct the earliest address owner and repeat a bounded registration before changing inbound routes.
Keep provider delivery separate from PBX routing
After provider mode and destination are correct, retain an authorized inbound INVITE and response. Compare the delivered Request-URI and To value with the first PBX rule, number format, trunk identification and final destination. SIPTRUNK documents an eleven-digit To form by default; a portal-visible registration does not prove the PBX route matches that value.
Also confirm that the DID is assigned to the intended trunk and plan. An outbound call does not prove inbound number assignment, and a received INVITE does not prove that an extension, queue, menu or voicemail destination accepted it.
Test failover and restoration as separate policies
SIPTRUNK documents gateway redundancy or SRV behavior separately from PSTN Backup, unconditional forwarding and voicemail treatments. Record which policy is configured, its destination, the provider-visible qualifying failure, the new-call result and the restoration step. Do not infer active-call survival, exact retry timers or customer-WAN resilience from those feature descriptions.
After the earliest mismatch is corrected, repeat registration or source-address acceptance, outbound authorization, inbound delivery, PBX routing, two-way media and each approved failover state. No SIPTRUNK account, device, route, failover event or call was tested for this diagnostic.
Do not use VoIP.ms registration as a DID-route test
Credential registration, IP authentication, DID POP and primary destination are separate evidence. Use the VoIP.ms diagnostic to trace them.
Apply Wavix registration evidence only to the selected model
The Wavix provider record distinguishes the digest registration path from approved IP Authentication.
Do not expect registration from Crazytel IP authentication
The Crazytel SIP trunk diagnostic branches on the selected mode before interpreting Live Registrations or source-IP evidence.
Treat MaxoTel as account-based registration in the reviewed path
The MaxoTel SIP trunk diagnostic follows portal-issued registration evidence rather than inventing direct-IP authentication.
Determine the Aussie Broadband authentication model first
The Aussie Broadband SIP trunk diagnostic leaves registration and source-address trust unassigned until the selected handoff proves one.
Determine the Symbio authentication model first
The Symbio Enterprise SIP trunk diagnostic leaves registration and source trust unassigned until the selected handoff proves one.
Use the selected Telstra identity model
The Telstra SIP Connect diagnostic leaves registration, source trust and credentials unassigned until the order and design prove them.
Use the selected Optus identity model
The Optus Evolve Voice diagnostic leaves REGISTER, source trust and credentials unassigned until OM.37683 and the application prove them.
Use the selected One NZ identity model
The One NZ SIP diagnostic leaves REGISTER, source trust and credentials unassigned until the implementation specification proves them.
Use the ordered 2degrees identity model
The 2degrees SIP diagnostic leaves REGISTER, source trust and credentials unassigned until the implementation pack proves them.
Use the ordered Vocus identity model
The Vocus SIP Voice diagnostic leaves registration, source trust and credentials unassigned until implementation evidence proves them.
Use the ordered Superloop identity model
The Superloop diagnostic leaves registration, source trust and credentials unresolved until issued evidence establishes them.
Use the ordered Enreach identity model
The Enreach diagnostic leaves registration and source trust unassigned until current evidence proves them.
Do not infer the MetTel authentication model
The MetTel SIP trunk diagnostic requires the selected order and provider values because public product records do not define one universal model.
Do not infer the NUSO authentication model
The NUSO diagnostic requires the accepted handoff because public records do not define one universal model.
Choose the exact Telesystem handoff before editing auth
The Telesystem diagnostic distinguishes registration, ESBC and direct peering.
Confirm the TPx handoff before editing authentication
The TPx diagnostic requires account-specific implementation evidence before choosing registration or IP identification.
Confirm the Lingo handoff before editing authentication
The Lingo diagnostic requires account-specific evidence before choosing registration or address identification.
Confirm the Voyant handoff before editing authentication
The Voyant diagnostic requires account-specific evidence before choosing registration or address identification.
Confirm the Windstream handoff before editing identity
The Windstream diagnostic requires account-specific evidence before choosing registration, address identification or managed edge.
Confirm the Granite handoff before editing identity
The Granite diagnostic requires order-specific evidence before choosing registration, address identification or managed edge.
Confirm the FirstLight handoff before editing identity
The FirstLight diagnostic requires order-specific evidence before choosing registration, address identification or another trust model.
Sources & applicability
Primary references for the technical details above. Operational examples and planning checklists are VoIP.info editorial guidance.
- SIPTRUNK IP authentication setup ↗
- SIPTRUNK registration connection issues ↗
- SIPTRUNK inbound-call troubleshooting ↗
- SIPTRUNK registration status ↗
- SIPTRUNK call failover options ↗
- SIPTRUNK interconnection and gateway addresses ↗
- SIPTRUNK general SIP device configuration ↗
- VoIP.ms DID routing, failover, POP and inbound format ↗
- VoIP.ms subaccount authentication and registration ↗
- VoIP.ms server and DID POP selection ↗
- VoIP.ms DID troubleshooting ↗
- VoIP.ms incoming-call troubleshooting ↗
- VoIP.ms registration troubleshooting ↗
- Wavix business SIP trunk ↗
- Wavix SIP trunk FAQ ↗
- Wavix create-a-SIP-trunk guide ↗
- Wavix SIP Trunks API overview ↗
- Wavix 3CX configuration guide ↗
- Wavix Numbers API overview ↗
- Wavix phone-number record reference ↗
- Wavix call-detail-record guide ↗
- Crazytel — Creating a Register Based SIP Trunk, updated 26 October 2026 ↗
- Crazytel — CGNAT setup recommendations, updated 26 October 2026 ↗
- Crazytel — SRTP compatibility, updated 26 October 2026 ↗
- Crazytel — Custom outbound proxy settings, updated 26 October 2026 ↗
- Crazytel — No sound on calls, 30 December 2025 ↗
- Crazytel — Real-time network status ↗
- MaxoTel — 3CX SIP Trunking, last modified 24 May 2023 ↗
- MaxoTel — Portal Guide: PBX Extensions and SIP Trunks, last modified 11 August 2023 ↗
- MaxoTel — 3CX Suddenly Loses SIP Trunk Registration, last modified 21 April 2020 ↗
- MaxoTel — SIP ALG, last modified 19 June 2019 ↗
- MaxoTel — NAT Issues, last modified 20 April 2021 ↗
- MaxoTel — Which codecs are supported?, last modified 19 April 2021 ↗
- MaxoTel — What VoIP Protocols are supported? ↗
- MaxoTel — Outage Communications Policy ↗
- Aussie Broadband — SIP Trunks ↗
- Aussie Broadband — SIP trunks Critical Information Summary, 1 July 2026 ↗
- Aussie Broadband — BYO VoIP settings, updated 14 September 2026 ↗
- Aussie Broadband — Integrated Public Number Database and emergency-service address, updated 26 August 2025 ↗
- Aussie Broadband — phone-number port timing, updated 5 May 2023 ↗
- Aussie Broadband — port forwarding and CGNAT, updated 13 August 2023 ↗
- Aussie Broadband — current, upcoming and past outages ↗
- Aussie Broadband — report a fault in MyAussie, updated 26 February 2024 ↗
- Aussie Broadband — business support and service-delivery contacts ↗
- Symbio — Enterprise SIP Trunking ↗
- Symbio — Australia voice network and carrier services ↗
- Symbio — phone numbers and porting ↗
- Symbio — SIP TLS and SRTP announcement, 24 January 2024 ↗
- Symbio — dual-carrier voice redundancy, 11 November 2025 ↗
- Symbio — security and trust ↗
- Symbio — Major Outage Register ↗
- Symbio — 24x7 Global Service Assurance, 19 December 2023 ↗
- Telstra Enterprise — SIP Connect ↗
- Telstra — Our Customer Terms, SIP Connect section, changed 3 June 2024 ↗
- Telstra — SIP Connect Customer Integration Guide ↗
- Telstra Enterprise — SIP Connect support and accredited CPE ↗
- Telstra Enterprise — Business Resumption Service + ↗
- Telstra — Business SIP Critical Information Summary ↗
- Optus Wholesale — Evolve Voice SIP trunking ↗
- Optus — Evolve Service Description, 1 November 2025 ↗
- Optus Business — Standard Form of Agreement, Evolve ↗
- Optus Business — Standard Form of Agreement ↗
- Optus — Evolve Voice factsheet (historical 2017 context) ↗
- Optus — SIP Trunking technical paper (historical context) ↗
- One NZ — SIP Advanced ↗
- One NZ — SIP Advanced Service Description ↗
- One NZ — Smart SIP ↗
- One NZ — Smart SIP Service Description ↗
- One NZ — Terms, conditions and service descriptions ↗
- One NZ — Metro Access Service Description ↗
- One NZ — About Emergency Services on 111 ↗
- 2degrees Business — SIP Trunks ↗
- 2degrees — Terms and conditions index ↗
- 2degrees — What happens in a power cut ↗
- 2degrees — Vulnerable consumers and calling 111 during a power outage ↗
- 2degrees — Privacy policy ↗
- Vocus — SIP Voice ↗
- Vocus — SIP Trunks data sheet ↗
- Vocus — SIP Voice Connect Service Schedule, 27 April 2026 ↗
- Vocus — Unified Communications Service Schedule, Australia Direct, 27 April 2026 ↗
- Vocus Support — call forwarding and diversions in BroadSoft ↗
- Vocus — Service Level Agreement ↗
- Superloop — Business VoIP and SIP ↗
- Superloop — Voice specifications data sheet ↗
- Superloop — Business Voice Service Schedule, 10 March 2026 ↗
- Superloop — business support contact ↗
- Superloop — outages and service updates ↗
- Enreach — SIP Trunks for businesses ↗
- Enreach — SIP Trunks for resellers ↗
- Enreach — PBX Trunks platform for service providers ↗
- Enreach Service Center — Enreach Trunk (DE), updated 9 April 2026 ↗
- Enreach Swyx 15 — creating a SIP trunk group ↗
- Enreach Swyx 15 — creating a SIP trunk ↗
- Enreach Control Center 15 — forwarding and number substitution ↗
- Enreach Swyx 15 — emergency call detection ↗
- Enreach Service Center — Enreach Trunk FAQ and nomadic-use boundary ↗
- MetTel — SIP Trunking datasheet, © 2025 ↗
- MetTel — VoIP and SIP Trunking terms, including 911 and traffic-policy boundaries ↗
- MetTel — support, activation, ticket and advanced-services contacts ↗
- MetTel — General Terms applicable to MetTel products ↗
- MetTel — Tariffs and product guides ↗
- NUSO — SIP Trunking product page ↗
- NUSO — Platform Services ↗
- NUSO — HIPAA compliance and SIP Trunking security responsibilities ↗
- NUSO — customer support contacts ↗
- NUSO — technical support and network-status path ↗
- NUSO — policies, programme documents and customer-service contact ↗
- Telesystem — SIP Trunks product page ↗
- Telesystem — SIP Trunk Administration for Tenant Admins ↗
- Telesystem — SIP Trunk DID Forwarding quick reference ↗
- Telesystem — E911 compliance guidance ↗
- Telesystem — network status and support paths ↗
- Telesystem — account support and escalation contacts ↗
- TPx — UCx SmartVoice SIP Trunking product page ↗
- TPx — UCx SmartVoice SIP Trunking brochure ↗
- TPx — SmartVoice PRI, CAS and SIP DID Features Guide ↗
- TPx — Introduction to Direct Trunk Overflow ↗
- TPx — 911 for SmartVoice and iPBX SIP Trunking Addendum ↗
- TPx — customer support and escalation paths ↗
- TPx — systems performance status ↗
- Lingo — Cloud SIP and PRI solutions ↗
- Lingo Knowledge Base — Cloud PRI and SIP Trunking ↗
- Lingo — Partner Product Guide 2024 ↗
- Lingo — E911 and California E911 Notice ↗
- Lingo — support resources ↗
- Lingo — service status ↗
- Voyant — communications products and SIP Trunking ↗
- Voyant — Master Services Agreement ↗
- Voyant — SIP Trunking terms and conditions ↗
- Voyant — VoIP 911 information ↗
- Voyant — Canadian 9-1-1 service notification ↗
- Sinch Community — SIP Trunk E911 bulk import in CAP, August 2024 ↗
- Sinch Community — Voyant/ANPI SIP Trunking STIR/SHAKEN caller-ID changes ↗
- Voyant — support contacts ↗
- Voyant — service status ↗
- Windstream Enterprise — SIP Trunking with Dynamic IP ↗
- Windstream Enterprise — Voice & Communications ↗
- Windstream Enterprise — Direct Trunk Overflow fact sheet ↗
- Windstream — Standard Agreement for Service Terms and Conditions ↗
- Windstream Enterprise — Dynamic IP Standard Terms and Conditions ↗
- Windstream Enterprise — Customer Responsibilities Guide ↗
- Windstream Enterprise — Data Products SLA and QoS classes ↗
- Windstream Enterprise / Uniti Solutions — U Connect portal and support ↗
- Granite — VoIP & Cloud Phone Systems ↗
- Granite — SIP Trunking brief (legacy, date-scoped) ↗
- Granite — Legal & Policy ↗
- Granite — General Terms of Service (May 2026) ↗
- Granite — VoIP Services Additional Terms and Conditions of Service ↗
- Granite — VoIP Services Service Level Agreement ↗
- Granite — Canadian VoIP 911 English ↗
- Granite — Resources and customer portals ↗
- FirstLight — Voice and SIP Trunking ↗
- FirstLight — SIP Trunk info sheet ↗
- FirstLight — Service Schedule A, Voice Services, version 4 (August 21, 2026) ↗
- FirstLight — Standard Terms and Conditions, version 5 (June 16, 2025) ↗
- FirstLight — E911 Acknowledgement, version January 14, 2019 ↗
- FirstLight — Legal, Regulatory, Terms and Conditions ↗
- FirstLight — Customer Support ↗
- FirstLight — Prevent Long Distance Fraud and PBX Hacking ↗
Examples require adaptation to your topology. No live PBX or hardware testing is claimed.