United States: SIP providers and connecting a number to the PBX

We do not issue numbers: the venue buys a US number from a carrier or a DID provider and brings it into its own PBX as a trunk. The common steps — trunk modes, form fields, network and testing — are covered in the overview "Connecting a provider number to the PBX"; this page covers only what is specific to the USA: where to get the number, what the provider issues and what of it goes into the admin panel.

The data on this page was collected on 9 September 2026 from provider documentation and federal rules (eCFR); the links are in "Sources".

In brief#

• A number with a SIP trunk is bought in the USA on your own, by card, within hours. The venue needs no licence: the regulatory burden (STIR/SHAKEN, USF contributions) is on the provider, you are simply its customer. This is the densest trunk market on our country list: with most providers both the server address and the authentication mode are configured in the account, with no account manager and no paper contract.
• The first candidate is Sangoma SIPStation. It is a trunk from the very vendor that makes Asterisk and FreePBX: the Asterisk guide is written by the vendor itself, the mode is registration to trunk1.freepbx.com, and the number arrives in the incoming call as ten digits. Next come DIDForSale (the only one that publishes both modes and static addresses in full) and SIP.US / SIPTRUNK.com (registration to gw4.sip.us, the login being the trunk number).
• Choose registration rather than IP authentication. Incoming calls are recognised by the SIP host of the trunk, while US carriers often send calls from a whole pool of addresses that does not reduce to a single host (Sangoma Carrier Services — a /20 subnet, DIDForSale — eight addresses, Skyetel — dynamic ones). With registration the provider sends calls to wherever the registration came from, and the question goes away.
• The main local pitfall is the last 7 digits. A line is identified by the last seven digits of the number, and in the US numbering plan the last seven digits are not unique: +1 212 555 1234 and +1 646 555 1234 are one and the same line for the PBX. With a single number this goes unnoticed; with several numbers in different area codes the routing rules will start to misfire.
• 911 is the duty of whoever operates the PBX. Direct 911 dialling without a prefix, notification of a responsible person and the delivery of a precise dispatchable location are required by 47 CFR 9.16 (Kari's Law and the RAY BAUM'S Act). On top of that, the E911 address must be filled in for every purchased number in the provider's account: while it is empty the number works, but a 911 call goes to the wrong place.
• Documents. For a local number providers ask for a US service address (which is also the E911 address); for a toll-free 8xx number an address is usually not needed; a foreign company can buy a US number, but each provider has its own terms. In any case the trunk only works on a dedicated PBX with a static IP — how the two trunk modes work is described in the overview.

Numbers in the country#

The country code is +1, and the national number is always 10 digits: area code NPA (3) + NXX (3) + line number (4).

• Landline and mobile numbers look the same: the same area code is used for both. You cannot tell "is this a mobile" by the look of the number.
• Toll-free (free for the caller): codes 800, 833, 844, 855, 866, 877, 888. They are administered by SOMOS through authorised RESPORG carriers.
• Dialling: inside the country — 1 + 10 digits; international from the USA — 011 + country code + number. Many areas require mandatory 10-digit local dialling (after the launch of the 988 crisis line), so seven-digit patterns must not be left in the outgoing rules.
• Service numbers: 911 — emergency, 988 — crisis line, 611, 211, 411 — service and directory.
• How the number arrives in an incoming call. Three schools: 10 digits (2125551234 — that is what SIPStation sends), 11 digits with a leading 1, and E.164 with +1 (Commio, Voxtelesys, and Skyetel when the format is +1NPANXXXXXX); Sangoma Carrier Services lets you choose the format in its portal. Any of them suits the PBX: it matches the last seven digits.

⚠️ What this means for several numbers. The last seven digits in US numbering are not unique — the area code is not part of the match. If a venue has two numbers whose NXX-XXXX coincide (quite possible for numbers in different cities), a call to the second number may follow the rule of the first one. When ordering a second number, ask the provider for one with a different ending — in DID providers' portals the number is picked from a list.

Who issues a number with a SIP trunk#

The first nine rows are providers whose connection parameters are either published or issued in the portal straight away; for them there are "What to enter in the admin panel" tables below. The rest mean talking to a sales department — or are not a source of a trunk at all.

Sangoma SIPStation#

A retail SIP trunk from the very vendor that develops FreePBX and Asterisk: sold per channel in the SIPStation Store, activated in minutes, and numbers (local and toll-free) are bought there too. The Asterisk connection guide is written by the vendor itself — among the providers we checked, this is the most reliable source of parameters.

What to enter in the admin panel (Telephony → "Trunks & pairings" → “Add provider” → the "SIP trunk (your own provider)" card):

What to set up at the provider: buy the trunk and the numbers in the SIPStation Store → take the SIP Username and SIP Password from My Account → Account Configuration → make sure the purchased numbers point at the account → fill in the E911 address for every number.

Pitfalls ⚠️ The vendor asks you to keep two trunks: the second one to the backup host trunk2.freepbx.com with the same login and password, otherwise incoming calls will not move over when the first node is unavailable. Its guide relies on the number being passed in the P-Asserted-Identity / Remote-Party-ID header, and our PBX does not send those by default: the “Show as” field must contain a number of your own SIPStation account, or the outgoing call will leave with the trunk login. The port is not named in the guide — the standard 5060 is given here.

Source: Setting up Standard SIPStation Manually in FreePBX, Trunk FAQ — 09.09.2026.

DIDForSale#

A Californian DID aggregator with numbers in 13,000+ rate centers and fully open documentation: both connection modes and the gateway addresses are published, and you do not need an account to read them. It supports both IP authentication (with static addresses, unlike most) and an ordinary SIP account with registration.

What to set up at the provider: in the portal — Interconnection → Manage IP → Add IP Address with the Enable Termination checkbox for outgoing calls (IP mode), or SIP Accounts → Add New SIP Account (registration mode) → buy a number → point it at that IP or SIP account in Configuration of Phone Numbers → fill in the E911 address in the number's card.

Pitfalls ⚠️ The outgoing dialling format is mandatory: 1 + 10 digits for the USA and Canada (19499300360), 011 + country code for international calls; in any other format outgoing calls are rejected, and the prefix is set in an outgoing rule. One public IP can be used in only one DIDForSale account — two venues on a single dedicated PBX will not fit in IP mode. And in that mode incoming calls arrive from eight addresses (66.209.76.70–.73 and .101–.104) while the trunk recognises the provider by a single host: one more argument for registration.

Source: Asterisk Interconnection Guide, Adding IP Address, Creating SIP Account — 09.09.2026.

SIP.US and SIPTRUNK.com#

The same platform in two storefronts: SIP.US sells trunks directly (a 60-minute trial without a card, payment per channel), while SIPTRUNK.com sells the same service as a reseller platform where trunks and numbers are provisioned from the partner panel. In both cases the login is the number of your SIP trunk, ten digits.

What to set up at the provider: register an account (the trial trunk is issued immediately) → take the trunk number and password → buy or port a number → point it at the trunk → fill in the E911 address.

Pitfalls ⚠️ The vendor asks you to create both registrations for redundancy: a second trunk to gw5.sip.us (for SIPTRUNK.com the backup gateways are gw2, gw4, gw5.siptrunk.com) with the same credentials. The port and the transport are not stated explicitly in the support articles — the standard 5060/UDP is given here; if registration does not come up, ask their support.

Source: Asterisk PBX (SIP.US), SIP Trunking for FreePBX, Asterisk PBX (SIPTRUNK) — 09.09.2026.

Skyetel#

A carrier popular with the FreePBX community: cheap inbound with per-minute billing, an account you register yourself and numbers bought in the portal.

What to set up at the provider: SIP Registration → SIP Credentials → +Add Credentials (Routing Method — "Route using the username") → Phone Numbers → Buy Phone Numbers → in the settings of the purchased number set the SIP format to +1NPANXXXXXX and select the credentials you created in Call Routing → SIP Credentials → fill in the E911 address.

Pitfalls ⚠️ The generic address reg.skyetel.com does not belong in the SIP host field: it is an SRV record without an ordinary A record (checked on 09.09.2026), while the PBX addresses the host with an explicit port — enter a regional node instead. The provider recommends IP authentication, but it does not suit us: the addresses it calls from change from call to call. Also check on activation whether the purchased number arrives in the call itself (and not just the credential login) — that determines whether the PBX can tell several lines of one trunk apart.

Source: SIP Registration, IP Auth vs SIP Registration, Skyetel IP Addresses — 09.09.2026.

Voxtelesys#

A carrier from North Dakota aimed at call centres and PBX integrators: it has its own guides for FreePBX, VitalPBX, Issabel and 3CX, and supports both modes.

What to set up at the provider: create a connection profile and take the domain, login and password from SIP Registrations (for IP mode, give them the static IP of your PBX instead and set Authentication = None) → assign a Trunk Group and a Route Profile to the number in Phone Numbers → Numbers → fill in the E911 address.

Pitfalls ⚠️ The number arrives in the incoming call in E.164 (+18882335678) — in “Lines” you may enter it in any format. The limit of simultaneous calls is set by the provider per account: if the second call drops, that is the reason, not the trunk settings.

Source: FreePBX 16 SIP Trunk Setup, FreePBX 16 SIP Trunk Setup with IP Authentication, 3CX: Voxtelesys — 09.09.2026.

Callcentric#

A New York carrier operating since 2001 and aimed at small business: it has the longest series of guides for Asterisk specifically, including the modern PJSIP stack. The account is created right at sign-up.

What to set up at the provider: sign up (the trunk is created automatically) → enable Outbound Calling and pick a rate plan, otherwise there will be no outgoing calls → buy a number and point it at the account → fill in the E911 address.

Pitfalls ⚠️ In the “Show as” field the provider accepts only numbers bought from itself. A call here is addressed to the account 1777XXXXXXX, and the purchased number may travel in a separate header: check on activation whether the number arrives in the call itself — if it does not, the PBX will not tell several numbers on one trunk apart, so route them with a single common rule.

Source: Asterisk 17 PJSIP Configuration, 3CX: Callcentric — 09.09.2026.

IP-authenticated trunks: Sangoma Carrier Services and Commio#

These two providers have only one mode — IP authentication; there is no login or password at all. Pick them deliberately: the rates are wholesale, but incoming calls arrive from a pool of addresses while the trunk recognises the provider by a single SIP host.

Sangoma Carrier Services (formerly VoIP Innovations)#

The wholesale arm of the same vendor as SIPStation (its former brand was VoIP Innovations): cheaper per minute, but designed for volume.

What to set up at the provider: in the Back Office add the public static IP of your PBX (the "Server" tab in the Telephony section) and enable termination for outgoing calls → in Endpoints → Endpoint Group Management choose the number format and specify your address as an endpoint → point the purchased numbers at that group → fill in the E911 address.

Pitfalls ⚠️ Calls arrive not only from the address in the SIP host field, but from the whole 64.136.160.0/20 subnet and six more separate addresses. The trunk recognises the provider by its own host, so some incoming calls may go unrecognised — for this provider that is the main risk; if it matters to you, pick a provider with registration.

Source: Wholesale SIP Trunking — Asterisk Trunk Config, carrierservices.sangoma.com — 09.09.2026.

Commio (thinQ)#

A carrier from North Carolina with multi-carrier routing: outgoing calls are spread across dozens of carriers by price, while incoming calls come to your own numbers. The mode is IP authentication.

What to set up at the provider: Products → Outbound → Routing Profiles — create a profile with a sensible price cap → Trunks — create a trunk, specify the public IP of your PBX and the channel limit → Products → Inbound → Routing Profiles and Configure Numbers — point the purchased numbers at the inbound profile → fill in the E911 address.

Pitfalls ⚠️ The outbound profile has a Max Rate — the maximum price per minute. If no carrier fits into it, the call fails with a 503 error and this looks like a broken trunk. Raise the cap in their portal. The provider's codecs are G.711 in both flavours and G.729, so it is compatible with our PBX.

Source: Elastic SIP Trunking (Commio), the VitalPBX guide to connecting Commio — 09.09.2026.

Providers that work by contract, and those that will not do#

International providers with US numbers#

All major international providers have US numbers, so there is little country-specific detail here. Their parameters come from the portal, and the Telnyx and Twilio cards are already in the catalog when adding a provider — the server address and the mode are filled in for you.

• Telnyx — registration to sip.telnyx.com:5060 or a domain-based connection; local and toll-free numbers, activated within hours. The fastest path with no correspondence.
• Twilio Elastic SIP Trunking — IP or credential-list authentication only, with an address of the form <name>.pstn.<edge>.twilio.com; calls arrive from the address pool of the chosen edge.
• Bandwidth — a US carrier with its own numbering; it supports both registration and IP routes, but the registration port differs from the standard 5060 — take it from their portal.
• Flowroute, VoIP.ms, SignalWire — these specialise precisely in the USA and Canada; the parameters are issued in the portal.
• DIDWW, Plivo Zentrunk, Vonage, Sinch, Infobip, didlogic — all of them cover the USA. With DIDWW the inbound and outbound trunks are different hosts, which means two trunks in the admin panel.
• Documents. For a local number providers require a US service address (which doubles as the E911 address); for toll-free an address is usually not needed. A foreign company can buy a number: a "local address" here means the service address of the number, not the registration of a legal entity — but every provider has its own rules, so ask before you pay.

Number porting#

• Porting between providers (LNP) is guaranteed by the rules: a simple port takes one business day, a non-simple one four (47 CFR 52.35). The request is filed by the new provider under your authorisation, and the old one may not demand data beyond the prescribed list.
• What actually delays a port: the account name and address do not match the old carrier's record, the balance is not settled, or a pending order sits on the number. Check the account details before filing.
• Toll-free 8xx numbers are ported differently — through SOMOS and an authorised RESPORG carrier; without the consent of the current RESPORG there will be no port. Alcazar Networks holds RESPORG status itself, which makes this faster with them.
• ⚠️ The E911 address does not travel with the port. It is entered anew at the new provider; until it is filled in, a 911 call from that number goes to the wrong place.
• The caller name (CNAM) does not travel either: it lives in separate databases and is registered with the new provider as a separate service.

Rules and documents#

If it does not work#

1. You work by IP and some incoming calls do not arrive. US carriers send calls from a pool of addresses (Sangoma Carrier Services — a whole subnet, DIDForSale — eight addresses, Skyetel — dynamic ones), while the trunk recognises the provider by a single SIP host. Switch the trunk to registration — all of the providers listed above except Commio and Carrier Services support it.
2. Registration with Skyetel does not come up. The generic address reg.skyetel.com has ended up in the SIP host field — it is an SRV record without an ordinary A record. Enter nw1.reg.skyetel.com or se1.reg.skyetel.com, port 5060.
3. Outgoing calls through Commio fail with a 503 error. The trunk is not to blame, the Max Rate price cap in the outbound profile is: no carrier fitted into it. Raise the cap in their portal.
4. Outgoing calls are rejected by DIDForSale and other US providers. The number must be given in their format: 1 + 10 digits for the USA and Canada, 011 + country code for international calls. Set the prefix in an outgoing rule.
5. A call to the second number follows the scenario of the first one. The last seven digits of the two numbers coincide — see "Numbers in the country". Give them different endings when ordering, or route both with a single common rule.
6. Guests see "Spam Likely" or an unfamiliar number instead of yours. The "Show as" field contains a number that does not belong to this provider: it is either replaced with the trunk number or given a low STIR/SHAKEN attestation. Use a number bought from the same provider.
7. The number works, but 911 goes to the wrong place. The E911 address of the number is empty in the provider's account. Check it whenever you add a number and after every port.

Causes that do not depend on the country are in “If calls do not go through”; the network and codec requirements are in “Network and compatibility”. A SIP trace in the trunk menu helps to dissect a particular call.

FAQ#

Can I get a US number if my company is registered outside the USA?#

As a rule, yes: for a local number the provider asks for a US service address — the very one that becomes the E911 address for a 911 call — and not for the registration of a legal entity. Toll-free 8xx numbers are usually issued without an address at all. Terms differ from provider to provider, so ask before you pay.

Registration or IP authentication — which to choose?#

Registration. US providers send calls from address pools, while incoming calls are recognised by the single SIP host of the trunk: in IP mode some calls may go unrecognised. With registration the provider sends calls to wherever the registration came from. The exceptions are Sangoma Carrier Services and Commio: their only mode is IP, and that is a deliberate trade-off for wholesale rates.

We use RingCentral (or 8x8, Zoom Phone) — can we bring its number into the PBX?#

No. That is a cloud PBX sold by the seat; it does not give a trunk to the outside, and forwarding its numbers to a third-party station is prohibited by the acceptable use policy. There are two options: port the number to a trunk provider (1–4 business days) or buy a new number and gradually move your advertising to it.

How long does connection take?#

Signing up yourself with SIPStation, SIP.US, Skyetel or DIDForSale takes hours: the account is created immediately and the number is bought by card. Porting an existing number takes 1–4 business days plus the preparation of the request. An enterprise contract with an access carrier (Verizon, AT&T, Comcast, Lumen) takes weeks — and still does not suit a cloud PBX.

Sources#

All links checked on 09.09.2026.

US communications rules (eCFR and the US Code)

• 47 CFR 9.16 — direct 911 dialling, notification and dispatchable location for multi-line telephone systems: https://www.ecfr.gov/current/title-47/chapter-I/subchapter-A/part-9/subpart-B/section-9.16
• 47 CFR 9.11 — E911 and Registered Location for interconnected VoIP: https://www.ecfr.gov/current/title-47/section-9.11
• 47 CFR 64.1604 — Truth in Caller ID, the ban on misleading caller ID substitution: https://www.ecfr.gov/current/title-47/part-64/section-64.1604
• 47 CFR 64.6305 — robocall mitigation (STIR/SHAKEN, the Robocall Mitigation Database): https://www.ecfr.gov/current/title-47/part-64/section-64.6305
• 47 CFR 52.35 — number porting intervals (1 and 4 business days): https://www.ecfr.gov/current/title-47/part-52/section-52.35
• 18 U.S.C. § 2511 — the federal one-party consent rule for recording: https://www.law.cornell.edu/uscode/text/18/2511

Providers

• Sangoma — setting up a SIPStation trunk manually in FreePBX: https://sangomakb.atlassian.net/wiki/spaces/SAF/pages/15925762 · Trunk FAQ: https://sangomakb.atlassian.net/wiki/spaces/SAF/pages/15794504/Trunk+FAQ
• Sangoma Carrier Services — Asterisk trunk config and the list of addresses: https://sangomakb.atlassian.net/wiki/spaces/WCS/pages/663486613/Wholesale+SIP+Trunking+-+Asterisk+Trunk+Config
• DIDForSale — connecting Asterisk: https://docs.didforsale.com/sip-trunking/sip-trunking/asterisk · adding an IP address: https://docs.didforsale.com/sip-trunking/ip-configuration · SIP account: https://docs.didforsale.com/sip-trunking/sip-account
• SIP.US — Asterisk PBX: https://support.sip.us/hc/en-us/articles/201914177-Asterisk-PBX · SIPTRUNK.com — Asterisk PBX: https://support.siptrunk.com/hc/en-us/articles/38749675990171-Asterisk-PBX
• Skyetel — SIP Registration: https://support.skyetel.com/hc/en-us/articles/360056914834-SIP-Registration · IP Auth vs SIP Registration: https://support.skyetel.com/hc/en-us/articles/4409130351255-IP-Auth-vs-SIP-Registration
• Voxtelesys — FreePBX 16 SIP trunk setup: https://voxtelesys.com/tutorial/freepbx-16-sip-trunk-setup
• Callcentric — Asterisk 17 PJSIP configuration: https://www.callcentric.com/support/device/asterisk/17_pjsip
• Commio — Elastic SIP Trunking: https://www.commio.com/products/inbound-voice/sip-trunking/elastic-sip-trunking/
• Nextiva — SIP Trunking for a third-party PBX: https://www.nextiva.com/products/pbx-sip-trunking · Alcazar Networks: https://www.alcazarnetworks.com/ · IntelePeer: https://www.intelepeer.com/products/sip-trunking/
• Verizon Business — IP Trunking: https://www.verizon.com/business/products/voice-collaboration/voip/ip-trunking/ · RingCentral — Bring Your Own Carrier: https://www.ringcentral.com/features/bring-your-own-carrier.html

Secondary sources

• VitalPBX — a step-by-step guide to connecting Commio: https://vitalpbx.com/blog/commio-sip-trunk-guide-vitalpbx/
• 3CX — configuring Voxtelesys, Callcentric and Skyetel trunks: https://www.3cx.com/docs/sip-trunk/voxtelesys-usa/ · https://www.3cx.com/docs/sip-trunk/callcentric-usa/ · https://www.3cx.com/docs/sip-trunk/skyetel-usa/

Related pages#

• Connecting a provider number to the PBX — the common steps, trunk modes, network, connection test.
• Setting up SIP phones — how to connect desk phones and DECT to extensions.