Freeze the trunk, DID and exclusive identity mode
Record the SIP.US trunk, DID, billing plan, selected authentication mode, public address or current REGISTER binding, active gateway, primary and secondary trunk assignment, PSTN Forward, PSTN Backup, channel count and PBX owner. SIP.US documents credential registration and IP authentication as mutually exclusive on one trunk.
A normal REGISTER status belongs to credential mode. IP authentication uses a stored static public address and shows an IP Auth label instead, so the absence of a registration row is not by itself an IP-authenticated trunk failure.
Documented service facts
Reviewed 2026-10-18. Compare SIP trunk services →
Outbound authentication
DocumentedEach trunk uses either digest registration credentials or IP authentication bound to one configured static public address and port. The modes are mutually exclusive; enabling IP authentication can cause REGISTER from that public address to be ignored in favor of the IP-authenticated trunk.
SIP registration
DocumentedCredential mode exposes REGISTER status, Contact, observed source alias, gateway and expiry. IP-authenticated mode shows an IP Auth label rather than normal registration state. Preserve the effective mode and actual inbound target before diagnosing a missing registration row.
Inbound delivery
DocumentedA DID is assigned to a primary trunk and can have a separately configured secondary trunk. Credential mode sends inbound calls to the active Contact; IP authentication sends them to the stored public address. PSTN Forward, PSTN Backup and PBX routing remain separate DID controls.
Inbound failover
DocumentedCurrent gateway redundancy uses DNS SRV or gw4/gw5 proxies. Secondary trunk routing requires two trunks and a DID assignment. PSTN Forward bypasses the PBX and prevents PSTN Backup from engaging; Backup is conditional on no response or no active registration. The reviewed pages do not establish one universal timer, response matrix or active-call survival result.
Number and regional availability
Not verifiedThe glossary describes a default termination plan for the US48 and Canada. Exact DID inventory, locality, porting, international destinations, regulatory eligibility and availability for the intended account and date require direct confirmation.
Audio codecs
DocumentedSIP.US says its gateways proxy SIP while RTP can arrive from varied addresses and ports. Preserve SDP and observed media endpoints rather than treating signaling-source ranges as a universal RTP inventory.
T.38 fax
Not verifiedThe reviewed first-party technical material did not establish T.38 support, fallback behavior, number eligibility or support scope. Treat fax as a separate account and interoperability acceptance item.
Signaling and media encryption
DocumentedThe glossary describes default UDP/RTP, direct media after call setup, and optional TLS/SRTP configured through support. Direct media changes the firewall path; exact encrypted-service ports, certificates, cipher suites, media profiles and fallback behavior were not public in the reviewed sources.
Call-start capacity
DocumentedOn channel-based plans, a forwarded call uses two channels. The provider warns that PSTN Backup or failover through the same trunk can fail at the channel limit. The capacity tracker counts exceedance events but is not detailed time-series usage; confirm active-call and call-start limits separately.
Emergency calling
DocumentedSIP.US documents direct 911 dialing without an access prefix, Enhanced DIDs with dispatchable addresses, use of the Enhanced DID as emergency caller ID, 933 testing and on-site notification duties. Confirm each location, callback and notification workflow in the current account.
Pricing model
Not verifiedThe provider describes channel and minutes-package rate-plan structures, but exact channels, minute inclusions, number charges, emergency fees, international rates, taxes and support terms require a dated account quote.
Administration and support
DocumentedThe portal holds identity mode, registration details or IP Auth label, DID primary and secondary trunks, Forward, Backup, channel plan and capacity alerts. Portal state, provider attempt, edge receipt, PBX handling, media and account balance remain separate operational evidence.
SMS and MMS
Not verifiedMessaging enablement, number eligibility, registration, routing and price were not established for this SIP trunk record. Keep voice and messaging acceptance separate.
Keep registration Contact and IP-auth destination distinct
For credential mode, retain challenge, authenticated REGISTER, Contact, observed source alias, expiry and selected gateway. SIP.US says inbound traffic is sent to the Contact address and port shown in the registration status. For IP authentication, inbound calls go only to the configured public address and outbound traffic must originate from that address.
Enabling IP authentication causes the gateways to ignore REGISTER from the same configured public address and associate traffic with the IP-authenticated trunk. Do not repair that expected mode change by repeatedly rotating credentials.
Trace current gateway and DID assignment before the PBX
SIP.US says current traffic uses gw4.sip.us and gw5.sip.us, recommends DNS SRV through gw.sip.us, and says legacy gw1/gw2-only systems no longer register or receive inbound calls. Gateway selection remains separate from the DID primary and optional secondary trunk assignments.
Preserve DNS answers, selected gateway, trunk identity, provider attempt and customer-edge receipt. A second provider gateway is not a second customer trunk, and a registered secondary trunk does not prove the DID is assigned to it.
Apply Forward and Backup precedence without inventing triggers
PSTN Forward intentionally bypasses the PBX for all DID traffic. SIP.US states that PSTN Forward prevents PSTN Backup from engaging. PSTN Backup is conditional on no server response or no active registration and sends the call to a chosen telephone number. A secondary trunk is another separately assigned SIP path.
Record which control was configured, which one became eligible, the provider attempt and the observed result. The reviewed pages do not publish one universal SIP-response matrix, timer, retry count, restoration interval or established-call survival contract.
Treat channel capacity as a routing dependency
SIP.US says a forwarded call consumes an inbound and an outbound channel on channel-based plans. It also warns that PSTN Backup or failover through the same trunk can fail when that trunk is already at its channel limit. The capacity tracker counts limit events; it is not a time-series utilization record.
Preserve purchased channels, active calls, forward legs, the capacity alert and the exact failed call. Keep simultaneous-call capacity separate from call-start rate, PBX limits and another trunk's independent capacity.
Prove PBX, media, emergency and recovery outcomes separately
After edge receipt, retain the INVITE, default TO number form, endpoint identification, assigned context, first matching route, answer and SDP. Then inspect observed RTP endpoints, negotiated codec and decoded audio in both directions; SIP.US says its signaling gateways proxy SIP while RTP can arrive from varied addresses and ports.
Emergency caller identity, Enhanced DID, dispatchable address and notification workflow remain separate acceptance evidence. Change only the earliest demonstrated owner, repeat the same authorized call and verify rollback. No availability, compatibility, pricing or laboratory result is claimed.
Diagnose SIP.US DID precedence before changing the PBX
Use the SIP.US DID and PSTN Backup diagnostic to correlate Forward, Backup, identity mode, trunk assignment, channel capacity, PBX handling and media.
Sources & applicability
Primary references for the technical details above. Operational examples and planning checklists are VoIP.info editorial guidance.
- SIP.US general device configuration ↗
- SIP.US SIP glossary and operational model ↗
- SIP.US SIP Trunking product page ↗
- SIP.US Asterisk registration guide ↗
- SIP.US IP authentication guide ↗
- SIP.US registration status guide ↗
- SIP.US current gateways and interconnection requirements ↗
- SIP.US call failover options ↗
- SIP.US inbound call troubleshooting ↗
- SIP.US channel-limit guidance ↗
- SIP.US E911 information ↗
Examples require adaptation to your topology. No live PBX or hardware testing is claimed.