Local, national, toll-free, mobile
Every number type you might need — geographic local numbers, national numbers, toll-free (including US 800/888/877/866/855/844/833), and mobile numbers in supported countries.
Get local, national, toll-free, or mobile DID numbers in 80+ countries and route them to your VoIP, PBX, or softphone in minutes. A complete replacement for traditional telecom carriers — establish a real, in-country business presence anywhere in the world without a contract, a sales call, or a single piece of hardware.
Any SIP-compliant system works. If it speaks SIP — over UDP, TCP, TLS, or WebSockets (WSS) for browser/WebRTC apps — DIDHub delivers calls to it. No proprietary gateway, no special hardware, no lock-in.
DIDHub is purpose-built to give your AI voice agents real phone numbers, anywhere in the world. Pick a platform, click connect, choose your numbers — your AI is on the phone in minutes. No SBCs, no SIP wizardry, no carrier negotiations.
From London to Tokyo to São Paulo — give your business a local presence anywhere your customers are.
No sales calls. No onboarding fees. Self-serve from start to finish — most numbers are live in under 60 seconds.
Pick a country, city, area code, or ZIP. See every available number with its monthly price and capabilities (voice, SMS, MMS, fax, T.38) in real time. Filter by consecutive blocks if you need a range.
Check out with a card. Most numbers activate instantly. For regulated countries (Germany, France, etc.) we show the ID/address requirements before you pay — no surprise holds or failed activations.
Point it at your SIP URI or IP address. Set concurrent channel limits per number. Calls start flowing within seconds. Change routing anytime from the dashboard — no support ticket required.
Local, toll-free, mobile, SMS, MMS, fax, consecutive blocks, port-in — all from a single self-serve dashboard.
Every number type you might need — geographic local numbers, national numbers, toll-free (including US 800/888/877/866/855/844/833), and mobile numbers in supported countries.
Capabilities shown upfront on every number. Voice on all numbers, SMS and MMS on supported numbers, T.38 fax where available, G.711 pass-through elsewhere. Know what you're buying before you pay.
Need 10, 50, or 100 sequential numbers for a call center or extension range? Filter for consecutive blocks and buy the whole range in one transaction.
Keep the number you already have — bring it to DIDHub and route it to your VoIP. Port-in supported in most countries. Clear status tracking on every port request.
No contracts. No annual commitments. Cancel numbers individually from the dashboard. The number stays active until the end of the current period, then releases after a short grace period.
Some countries require proof of address or ID. We show those requirements before checkout — never hidden, never a surprise. Most countries have zero paperwork.
Set concurrent call limits per number — 1 for a personal line, 30+ for a queue. Change anytime. Prevents surprise overages and lets you size numbers to actual demand.
Configure a primary SIP destination and a backup. If your primary is unreachable, calls automatically fail over — no dropped calls during maintenance windows or outages.
See every call and message in real time. Filter by number, date, destination, or status. Export CSVs for accounting. No surprises at the end of the month.
Optional call recording per number — inbound, outbound, or both. Recordings stored encrypted at rest, downloadable as MP3/WAV, with configurable retention. Turn it on per number for compliance, QA, or training.
Forward inbound SMS and MMS to any email address (with picture/media as attachments), to a webhook (HTTP POST to your URL), or both. Reply from email and we'll send it back out as SMS/MMS. No custom integration required — configure per number in under a minute.
Also called 2-way voice or bidirectional termination — get a number, receive calls on it, and place outbound calls using it as your Caller ID. One account, one invoice, one routing config.
First-class support for Microsoft Teams Direct Routing — including a fully-managed Teams SBC-as-a-Service so you don't need to deploy AudioCodes, Ribbon, or your own SBC. Zoom Phone BYOC supported too. Users calling on Teams or Zoom in minutes, not weeks.
One-click integration with Vapi, Retell, Bland, ElevenLabs, Synthflow, LiveKit, Pipecat, Vocode, Cartesia, and Deepgram. HD codecs, sub-100ms regional ingress, and bulk provisioning for AI deployments at scale.
SIP over TLS 1.2/1.3 for signaling, SRTP for media, HTTPS-only API, AES-256 encryption at rest for recordings and messages. Optional on every number — required for HIPAA, MiFID II, and enterprise reviews.
Inbound spam, robocalls, and known scam patterns detected automatically using STIR/SHAKEN attestation, global reputation lists, and behavioral heuristics. Auto-divert flagged calls to voicemail, reject with SIP 603, or tag in CDRs — your choice, per number.
Connect browser-based softphones and WebRTC apps directly — SIP over Secure WebSockets (WSS) with DTLS-SRTP encrypted media. No plugins, no NAT traversal headaches, works through any corporate firewall that allows HTTPS.
Whether you need one number or two hundred, across one country or fifty, the use cases are the same: meet customers where they are.
Selling in London but headquartered in Tel Aviv? Get a +44 20 London number, route it to your existing phones, and look like a local business to UK customers.
Give each region its own toll-free number. US 800 for Americans, UK 0800 for Brits, +49 800 for Germans — all ringing into the same support queue on your PBX.
Your team is in 12 cities but the business phone system shouldn't care. Assign any number to any extension — a New York number can ring a softphone in Manila.
Spin up unique numbers for each campaign, channel, or ad. See which source drove the call. Cancel numbers when the campaign ends — no waste.
Shipping to 20 countries? Give customers a number in their own country for returns and order questions. Trust and conversion go up when the phone number isn't international.
Buy consecutive blocks for agent extensions. Set per-number channel limits to protect your trunks. Port in existing numbers so campaigns aren't disrupted.
Individual numbers per agent, per listing, or per location. Forward to a mobile or a branch office. Release unused numbers from the dashboard in a click.
Reserve toll-free and geographic numbers for your product's messaging and voice features. Scale to thousands of numbers without contracts or provisioning delays.
No setup fees on most numbers. No contracts. No minimums. Pay monthly, cancel anytime. Per-minute voice is billed separately from $0.003/min.
| Country / City | Type | Monthly | Setup |
|---|---|---|---|
| US local (any city) | Geographic | from $0.50 | $0 |
| US toll-free | Toll-free | from $1.50 | $0 |
| Canada local | Geographic | from $0.50 | $0 |
| UK London | Geographic | from $0.63 | $0 |
| UK national 0330 | National | from $1.20 | $0 |
| Germany Berlin | Geographic | from $1.10 | $0 |
| France Paris | Geographic | from $0.95 | $0 |
| Netherlands Amsterdam | Geographic | from $1.00 | $0 |
| Israel national | National | from $2.50 | $0 |
| Australia Sydney | Geographic | from $1.30 | $0 |
| Japan Tokyo | Geographic | from $2.00 | $0 |
| Brazil São Paulo | Geographic | from $1.40 | $0 |
Per-minute voice usage billed separately from $0.003/min (varies by destination). Some countries require proof of address or ID before activation — requirements are shown clearly during checkout. Full price list is visible inside the search for every country and city.
From one-person consultancies to multinational BPOs — same self-serve dashboard, same transparent pricing.
DIDHub runs on a fully distributed, multi-region SIP network. Every call ingresses to the closest healthy region, every route has automatic failover, and there's no single box, datacenter, or carrier whose outage takes you offline.
If you'd rather automate, every action in the DIDHub dashboard is also available via a clean, fully-documented REST API. Use it to provision numbers on behalf of your own customers, sync CDRs into your billing system, or build telephony into your product.
Search inventory, order numbers, configure routing, pull CDRs and recordings, send and receive SMS/MMS, manage ports — programmatically and at scale.
# 1. Search available numbers in London curl https://api.didhub.io/v1/inventory/search \ -H "Authorization: Bearer $DIDHUB_API_KEY" \ -G --data-urlencode "country=GB" \ --data-urlencode "area_code=20" \ --data-urlencode "capability=voice,sms" # 2. Order a number from the results curl -X POST https://api.didhub.io/v1/numbers \ -H "Authorization: Bearer $DIDHUB_API_KEY" \ -H "Content-Type: application/json" \ -d '{ "number": "+442012345678", "channels": 4, "routing": { "destination": "sip:pbx.example.com", "failover": "sip:backup.example.com" }, "sms_webhook": "https://app.example.com/sms/inbound" }' # 3. Done. Number is live in seconds. {"id":"did_01HXY...","status":"active","activated_at":"2026-04-24T..."}
Traditional carriers ask for contracts, minimums, and a six-week sales cycle just to get a phone number. DIDHub is different.
Everything you need to know before getting a number with DIDHub. Still have questions? Talk to us.
A DID (Direct Inward Dialing) number is a regular business phone number that terminates into a VoIP or PBX system rather than a physical phone line. From the outside it works like any other phone number — people dial it, it rings — but the calls are delivered to your phone system over the internet via SIP instead of a copper line.
DIDHub provides real, fully-functional DID numbers in 80+ countries. A US DID from DIDHub has a genuine US area code and is dialable from any phone on earth. A London DID has a +44 20 prefix that Londoners recognize as local. The numbers behave exactly like traditional phone numbers — only the delivery mechanism is modern.
No. DIDHub delivers every call via SIP over the public internet. If your phone system speaks SIP — which covers virtually every modern business phone system — there's no extra hardware to buy, install, or maintain. No trunk gateways, no adapter boxes, no on-premise appliances.
Compatible systems include 3CX, FreePBX, Asterisk, FusionPBX, Kamailio, OpenSIPS, Cisco CUCM, Grandstream UCM, Yealink Cloud, Zoom Phone BYOC, Microsoft Teams Direct Routing, and softphones like Zoiper, Linphone, MicroSIP, Bria, and Groundwire.
Any SIP-compliant system. If your platform supports RFC 3261 SIP — and nearly all modern business phone systems do — DIDHub will deliver calls to it.
Tested and supported: 3CX (all versions), FreePBX, Asterisk (any version), FusionPBX, Kamailio, OpenSIPS, Cisco CUCM, Grandstream UCM series, Yealink Cloud, Zoom Phone BYOC, Microsoft Teams Direct Routing, RingCentral BYOC, 8x8 BYOC, and every major softphone.
If you're unsure whether your system is compatible, send us the product name and we'll confirm. We haven't found one that isn't yet.
Yes. This is sometimes called 2-way voice, bidirectional termination, or inbound + outbound. DIDHub offers outbound voice termination that pairs with your DID numbers. Configure any DID as the Caller ID for your outbound traffic, and calls you make from your PBX will display the correct local number to the person you're calling.
Outbound is pay-as-you-go per minute — no monthly commitment, no per-seat fees. Rates vary by destination and start at $0.003/min for major markets. The same SIP credentials handle both inbound and outbound — one routing config, one invoice, one account.
Yes. Optional call recording is available on any number — inbound only, outbound only, or both. Enable it per number from the dashboard with one toggle.
Recordings are stored encrypted at rest, accessible from the dashboard, and downloadable as MP3 or WAV. Configurable retention (7 days, 30 days, 90 days, 1 year, or custom). Useful for compliance (PCI, HIPAA-aligned setups, MiFID II), quality assurance, sales training, and dispute resolution.
Note: customers are responsible for honoring local consent laws (e.g. two-party consent in some US states). DIDHub provides the technical capability; legal use is on you.
Yes — three options for both SMS and MMS, configurable per number:
SMS/MMS to email: inbound texts arrive as emails to any address you configure, with MMS picture/media attached as standard email attachments. Reply from your email client and we'll send it back out as SMS or MMS to the original sender. Works with any email provider (Gmail, Outlook, custom domain).
SMS/MMS to webhook: we POST inbound messages to a URL you provide as JSON. Standard format with sender, recipient, body, timestamp, and message ID — plus signed media URLs and MIME types for MMS attachments. Use it to drop messages straight into Slack, a CRM, a helpdesk, or your own app.
SMS via SIP: for systems that support SIP MESSAGE (3CX, FreePBX with modules, custom SIP platforms), we deliver SMS over the same SIP trunk as voice. (MMS is delivered via webhook or email since SIP MESSAGE doesn't carry media.)
Yes — and it's one of our fastest-growing use cases. DIDHub offers one-click integration with the major AI voice orchestration platforms: Vapi, Retell AI, Bland AI, ElevenLabs Conversational AI, Synthflow, LiveKit Agents, Pipecat (Daily), Vocode, Cartesia, and Deepgram Voice Agent. Pick your platform in the dashboard, authenticate, choose which DIDs to assign, and your AI agent is on the phone in minutes.
For platforms without a native one-click flow, every DIDHub number is plain SIP — point it at any platform's SIP endpoint and it works. No SBCs to install, no carrier negotiations.
Why teams choose DIDHub for AI voice:
• Sub-100ms regional ingress — geo-aware routing keeps audio on the shortest path so realtime AI turns feel natural, not robotic.
• HD codecs (G.722, Opus) end-to-end so STT models hear what callers actually said.
• Bulk API provisioning — spin up 10 or 10,000 numbers programmatically as your AI deployments scale.
• Local-presence outbound in 80+ countries — answer rates roughly double when the Caller ID is local.
• SIP TLS + SRTP for encrypted signaling and media — required for healthcare, finance, and enterprise reviews.
• Built-in call recording with separate inbound/outbound channels, ideal for clean diarization and training data.
Every inbound call is screened in real time before it ever rings your phone system. The screening uses three signals layered together:
• STIR/SHAKEN attestation — calls without valid A-level attestation, or with mismatched calling-party identity, are flagged. (Standard in NOAM; expanding to other regions as regulators roll it out.)
• Global reputation lists — known robocaller, scam, and spoofing source numbers, updated continuously from carrier-shared and third-party feeds.
• Behavioral heuristics — short-duration burst patterns, neighbor-spoofing (caller ID matching the called area code suspiciously often), and per-source velocity anomalies.
What happens to a flagged call — three policies, configurable per number:
1. Divert to voicemail — flagged calls go straight to voicemail, your phones don't ring. Voicemail can be transcribed and emailed/webhooked. (Default for most customers.)
2. Reject with SIP 603 — call is refused at our edge, your trunk never sees it. Cleanest for high-volume call centers that just want to drop spam.
3. Pass through but tag — call is delivered with an X-DIDHub-Spam-Score SIP header so your PBX can decide what to do. Great for IVR-based screening.
Spam filtering is on by default and free on every number. Adjust the sensitivity from "strict" (more false positives, fewer spam) to "permissive" (more permissive, may let some through) per number.
Yes. Every DIDHub number can be reached via SIP over Secure WebSockets (WSS) in addition to UDP/TCP/TLS, with DTLS-SRTP encrypted media — meaning browser-based softphones and WebRTC apps can register and place/receive calls directly without a gateway, without plugins, and without NAT traversal headaches.
Why this matters:
• Browser softphones — embed a click-to-call widget on your website using JsSIP, SIP.js, or similar. No phone app to install.
• WebRTC voice agents — AI platforms that use WebRTC for media (LiveKit, Pipecat, Daily) connect directly without a SIP-WebRTC bridge.
• Corporate firewalls — WSS is just HTTPS on port 443, so it traverses any firewall that allows web traffic.
• Mobile apps — same WSS endpoint works for browser and React Native / Capacitor apps.
WSS is enabled per number from the dashboard or API. We provide drop-in JsSIP/SIP.js configuration snippets, ICE/STUN/TURN credentials, and example HTML for click-to-call buttons.
DIDHub supports end-to-end transport encryption on every number, optional and configurable per number:
• SIP over TLS 1.2 / 1.3 for signaling — the SIP INVITE, REGISTER, and all dialog control messages between your endpoint and DIDHub are encrypted, preventing eavesdropping or tampering on signaling. Mutual TLS (mTLS) supported for enterprise deployments.
• SIP over Secure WebSockets (WSS) for browser-based softphones and WebRTC apps — same encryption guarantees as SIP TLS, delivered over the same channel as HTTPS so it traverses any firewall.
• SRTP (Secure RTP) for media — call audio is encrypted with AES-128 or AES-256, keyed via SDES or DTLS-SRTP. The voice payload is unintelligible if intercepted on the path. DTLS-SRTP is the default for WebRTC connections.
• HTTPS-only for the dashboard, API, and webhooks. We do not accept plain-HTTP API requests; webhooks reject self-signed certificates by default.
• AES-256 at rest for call recordings, SMS & MMS content (including media attachments), KYC documents, and credentials. Keys managed via a HSM-backed KMS.
Encryption is required by HIPAA-aligned setups, MiFID II for financial services, PCI-DSS where calls touch cardholder data, and most enterprise security reviews. We provide configuration guides for the common SIP stacks (Asterisk, FreePBX, 3CX, Kamailio, OpenSIPS) plus the AI voice platforms.
Yes — Teams Direct Routing is fully supported, two ways:
1. Teams SBC-as-a-Service (managed by DIDHub) — we operate the Session Border Controller for you. No AudioCodes, Ribbon, AnyNode, or Oracle SBC to deploy, license, or maintain. No public IP, no TLS certs, no PowerShell rituals. You connect your Teams tenant to our managed SBC, we hand you the carrier endpoint, and your users are dialing on DIDHub numbers within minutes. Includes media bypass, location-based routing, emergency calling configuration, and 24/7 NOC monitoring.
2. Bring your own SBC (BYO-SBC) — if you already operate AudioCodes, Ribbon, AnyNode, Oracle, or Sangoma, point it at DIDHub as the upstream SIP carrier. We provide trunk credentials, IP allowlisting, and a configuration guide for each major SBC vendor.
Either path gives you global coverage in 80+ countries at a fraction of Microsoft Calling Plans pricing — and avoids the per-seat lock-in. Calling Plans are also only available in a handful of countries; Direct Routing via DIDHub works everywhere we have numbers.
Yes — Zoom Phone BYOC (Bring Your Own Carrier) is supported. Get numbers from DIDHub, configure them as a BYOC trunk in your Zoom Phone admin, and your Zoom users are calling on DIDHub numbers across 80+ countries.
BYOC is the right choice when you need global coverage that exceeds Zoom's native country list, or when you want to keep numbers in a provider-agnostic place so you're not locked into Zoom's pricing or coverage forever.
Yes, on messaging-enabled numbers. Each number's SMS and MMS capability (inbound, outbound, or both) is shown clearly in the search results before you buy. SMS is available on most US, Canadian, UK, and Nordic numbers plus a growing list of other countries. MMS (picture and media messaging) is supported primarily on US and Canadian numbers, with growing coverage elsewhere — wherever underlying carriers allow it.
Messages are delivered to your platform via webhook (HTTP POST to a URL you provide, with signed media URLs for MMS attachments), email (with media as standard attachments), SIP MESSAGE for SMS on SIP-capable systems, or via our dashboard for manual send/receive. Long messages (concatenated SMS), group MMS, and common media types (JPEG, PNG, GIF, video) all supported.
Yes. Port-in is supported in most countries where DIDHub offers numbers. Bring your existing phone number — keep the number, keep your customers, switch only the routing.
Porting times vary by country and carrier:
• US / Canada: typically 1-3 weeks
• UK: typically 2-4 weeks
• Most EU countries: 2-6 weeks
• APAC / other: varies, usually 3-8 weeks
You'll see the estimated porting time and required documents (LOA, bill copy, etc.) before you start the port. No port-in fees on most numbers.
Cancel anytime from the dashboard — one click per number, or cancel your whole account at once. There are no early termination fees, contracts, or penalties.
When you cancel, the number stays active until the end of the current billing period so calls keep flowing. After that there's a short grace period (usually 7 days) where you can reactivate if you change your mind. Once the grace period ends, the number is released back to the pool.
Usually not. For most countries (US, Canada, UK, most of Asia-Pacific, and many others) there is zero paperwork — you check out and the number activates in seconds.
Some regulated countries — notably Germany, France, Belgium, Switzerland, Japan, and a few others — require a local address and/or a copy of ID or business registration. This is regulator-imposed, not a DIDHub rule.
What's different about DIDHub is transparency: we show every activation requirement before you check out. No charging your card and then discovering you can't use the number. If you can't meet the requirements, we don't let you buy.
Three fields: primary destination (SIP URI like sip:pbx.example.com or a static IP), optional concurrent channel limit, optional failover destination. Save and you're done.
Every change is live in seconds — no support ticket, no engineering time. You can change routing for any number at any time from the dashboard or bulk-update multiple numbers via CSV.
Advanced options: time-of-day routing (business hours vs after-hours destinations), failover on SIP timeout, regional SIP gateways for lower latency, and SIP credential authentication if you prefer username/password over IP auth.
80+ countries across North America, Europe, the Middle East, Asia-Pacific, Africa, and Latin America. New countries are added regularly.
Each country offers one or more of: local/geographic numbers (specific cities and area codes), national numbers (country-wide, no city prefix), toll-free numbers, and mobile numbers. The search shows exactly what's available in every country.
Most numbers are live in under 60 seconds. Specifically:
• Non-regulated countries (US, Canada, UK, most APAC): instant after checkout
• Regulated countries requiring document review: 1-5 business days
• Port-in from another provider: 1-6 weeks depending on country
The search result for every number shows its expected activation time so you can plan accordingly.
The maximum number of simultaneous calls a single phone number can handle. Set it per number: 1 channel for a personal extension, 10 for a small hunt group, 30+ for a call center queue.
Why it matters: setting a sensible channel limit prevents runaway costs if something goes wrong (e.g. a loop in your dialplan), and protects your PBX from being overwhelmed. It's also how you size a number to actual demand — a toll-free number expecting 200 concurrent callers needs more channels than a reception line.
Yes, in supported countries. Common block sizes: 5, 10, 20, 50, 100. Filter the search for "consecutive block" and pick the size — available ranges are shown in real time.
Typical use cases: call center agent extensions, hotel room numbers, multi-line business numbers, legacy systems that expect contiguous DID ranges.
Yes, on fax-capable numbers. T.38 capability is shown per number in search results. Where T.38 is not available we support G.711 pass-through fax (less reliable for long documents but works for most ad-hoc faxes).
Dedicated fax-over-IP numbers are available in the US, Canada, UK, and most of Europe.
Credit and debit cards (Visa, Mastercard, American Express). Billing is monthly with transparent itemized invoices. Enterprise customers with larger volumes can arrange bank transfer / invoicing via sales.
DIDHub runs on a globally distributed, fully redundant SIP network. There is no single point of failure — every component (SIP signaling, media, databases, edge) is replicated across multiple regions and providers.
The platform operates from eight geographic regions:
• NOAM — North America
• LATAM — Latin America
• EURO — Europe
• MENA — Middle East & North Africa
• AFRICA — Sub-Saharan Africa
• INDIA — Indian Subcontinent
• APAC — Asia-Pacific
• ANZAC — Australia & New Zealand
Inbound calls ingress to the nearest healthy region for lowest latency. If a region degrades, traffic automatically reroutes — no manual intervention from you. Upstream carrier diversity in every country means a single carrier's outage does not take your numbers offline. We target 99.99% platform uptime with SLA-backed service credits.
Absolutely — it's one of the most common use cases. DIDHub supports the things call centers actually need: consecutive number blocks for agent extensions, per-number concurrent channel limits, failover routing, outbound termination with Caller ID spoofing to your DIDs, detailed CDRs, and CSV bulk-management for large deployments.
Teams running on Asterisk, FreePBX, Vicidial, 3CX, Genesys Cloud BYOC, and custom SIP platforms are all using DIDHub today.
Yes — a clean, fully-documented REST API. Authenticate with a Bearer API key (rotate or revoke from the dashboard), optionally restrict by IP allowlist. Everything you can do in the UI is also available via API.
Endpoints cover:
• Search inventory by country, area code, ZIP, capabilities, consecutive-block size
• Order single numbers or blocks atomically
• Configure SIP routing, channel limits, failover destinations
• Send SMS & MMS, receive via webhook (HTTP POST JSON to your URL with signed media URLs for MMS)
• Pull CDRs (call detail records) for any time range, any number
• Download call recordings
• Submit and track port-in requests with document upload
• Subscribe to webhooks for SMS/MMS, CDR finalization, port status changes
No volume minimums — the same API is available whether you have one number or ten thousand. SaaS platforms commonly use it to provision numbers on behalf of their own customers (sub-account model supported).
Those are mostly wholesale carriers and developer-first CPaaS platforms — built for telecom engineers, carriers, and software teams that buy numbers by the thousand and integrate via API. They work well for that audience, but the typical onboarding involves credit checks, minimum commits, KYC reviews, and sales calls before you can get a single number.
DIDHub is built for the other 90% of the market — small and mid-sized businesses that already run a phone system (3CX, FreePBX, Asterisk, cloud PBX, softphones) and just want to get a handful of numbers and point them at it. Self-serve signup, card on file, flat per-number monthly pricing, cancel anytime. No minimums. No sales calls. (REST API available if you want it — just not required.)
If you need a wholesale carrier to build a CPaaS on top of, those providers are great. If you just need business phone numbers delivered to your VoIP, you don't need to shop for wholesale contracts — that's what DIDHub is for.
Yes, for most small-business use cases. DIDWW, Voxbone (now part of Bandwidth), DIDXL, Inteliquent, Telserv, and Bird all sell phone numbers globally, but they're optimized for wholesale volume and carrier/CPaaS customers. Their pricing, onboarding, and account management reflect that.
DIDHub offers the same kind of global coverage — 80+ countries, local, national, toll-free, and mobile — packaged for businesses that want to buy one number or fifty without negotiating a wholesale contract. Same underlying telephony, different buying experience.
80+ countries. Any SIP system. No contracts. Cancel anytime.