v2.10 · self-hosted · source-available on GitHub

The call center you run.
Not the one you rent.

Dialix is a self-hosted PBX and call center: queues, IVR, a visual call-flow designer, and a mobile app that is the desk phone — all in one Rust binary, on your server, with your SIP trunk. No per-seat licensing.

curl -fsSL .../install.sh | sudo bash
calling.yourcompany.com/dashboard
Dialix operations dashboard showing live calls, queue stats, and call volume over the last 7 days
~27 MBsingle Rust binary
AnySIP trunk or carrier API
€4VPS is enough to run it
€10per tenant, per month — not per seat

Built the way engineers wish telephony worked

Every screenshot below is a real screen from a running installation — nothing here is a mockup.

Setup

One-click telephony setup — verified, not assumed

“⚙ Configure automatically” writes the Asterisk config, generates passwords, arms the SIP firewall, installs the TTS engine, and reloads. Then it checks every step and shows you a pass/fail list — with a fix hint when something isn't green.

Dialix setup wizard showing organization mode selection: single organization or multi-tenant
Scheduling

Describe a schedule in English or Dutch. It becomes an exact window.

Type "vanaf vrijdag 24 juli 17:00 tot en maandag 24 augustus 7:30" into a time group and Dialix parses it into a precise active-from/active-until range, rings the right teammates, and falls back to voicemail when the window closes.

Dialix new time group dialog with natural-language schedule parsing and member selection
Mobile

The app is the desk phone

Scan a QR code on-screen and an Android phone becomes a fully provisioned SIP extension in seconds — no manual SIP config, ever. Inbound calls wake it with push even when the app is closed; outbound shows the company caller ID.

Dialix auto-connect dialog showing a QR code and enrollment code for zero-config device pairing

Everything a contact center needs. Nothing you have to pay per seat for.

Queues, IVR, recording, transcription, reporting, CRM screen-pop, click-to-call — the full catalog, shipping today.

Call flow designer

Drag-and-drop canvas with IVR, queues, time rules, switches and webhooks. The route a caller would take right now is painted green and re-evaluated every 5 seconds.

Switches & feature codes

Named on/off switches — "Night mode", "Emergency" — toggled from the web, the app, a flow, or by dialing *21 from any phone.

Voice & audio

Server-side TTS with automatic per-prompt language detection, telephony-grade audio pipeline, and barge-in with buffered early keypresses.

Native mobile app

A real SIP engine on Android, not a callback trick. QR sign-in, push-woken ringing, transfer, DND, voicemail — all in-app.

Live supervisor console

Listen, whisper, barge, hold, transfer and record any call in progress. Race-proof call parking with self-cleaning slots.

Resilient inbound routing

Per-number routing to a flow, queue, hunt list or forward. Push-woken handsets get a retry budget instead of a dropped call.

Security by default

fail2ban-armed SIP firewall bans scanners after 5 failures, bcrypt/argon2 auth, TLS via ACME, and a full admin audit log.

Multi-tenant & white-label

Organizations exist because their database exists. Per-tenant trunks, branding, custom domains with automatic per-domain TLS.

Admin & observability

One-click "configure automatically" with a verified pass/fail checklist, live SIP logs, one-file backup/restore.

Any phone. Any trunk. One directory.

Zero-touch provisioning across eight phone vendors, an edge SBC for branch offices, and a company directory that shows live presence — click to call.

SIP / SBC devices
Dialix SIP / SBC devices screen listing a reception phone and a branch Raspberry Pi bridge with live status
Directory
Dialix company directory showing teammates and contacts with click-to-call buttons
YealinkGrandstreamFanvilSnomPolyCisco MPPHtekGigaset DECT
Dialix Desk custom SIP phone, currently in development
Coming soonFull reveal once the first production units ship
Hardware · In development

Dialix Desk — a modern, cordless-first SIP phone, built for Dialix

We're building our own desk phone from the ground up, optimized end-to-end for the Dialix stack: a single PoE cable to the base, a USB-C connected 0–9 keypad module you can dock on any side of the unit, a wireless handset with a wired option, a dedicated speaker button, a physical D-pad and three soft keys for driving the on-screen UI hands-free, plus dedicated call, history and phone book buttons — with firmware updates pushed the same way as everything else on your server.

Key features
  • Just one PoE cable
    The base runs on a single Ethernet cable — the keypad module draws its power over USB-C, so nothing else touches an outlet
  • USB-C keypad module
    0–9 extension board docks top, left or right of the screen and connects with one USB-C cable — no second power brick
  • Wireless handset, wired if you want it
    Ships cordless on a magnetic charging cradle by default, with a wired handset available as an option
  • Dedicated speaker button
    Drop into hands-free speakerphone with a single press
  • Up / down / left / right / OK
    A physical D-pad drives the on-screen UI without ever touching the display
  • 3 soft keys under the display
    Context-aware buttons that open whatever menu the screen is showing
  • Dedicated call & end-call keys
    Answer, hang up or hit redial without hunting through menus
  • One-touch call history
    Jump straight to recents from a dedicated hardware button
  • One-touch phone book
    Open the directory instantly and dial by name
  • Zero-touch provisioning
    Boots straight into your org — no config file to upload
Android · GPLv3

Your agents' phones are the desk phones

No hardware to provision, no VPN to configure. An agent scans a code and their phone registers as a real SIP extension — with the same call timer, transfer options and caller ID as a desk phone at the office.

  • Native SIP engine (liblinphone) — real audio, no callback tricks
  • QR sign-in with zero manual SIP configuration
  • Push wake: inbound calls ring even when the app is closed or dozing
  • Blind transfer to extensions, numbers and SIP URIs
  • Queue join/leave, org switches, DND/presence, voicemail PIN — all in-app
  • Play-Store-ready build (16 KB page size, R8 full mode); APK on every release
Dialix Android app call history screen
Dialix Android app keypad and dialer screen
Dialix Android app contacts screen

One command. One binary. Your server.

Nothing is seeded — you create the first administrator yourself, in the browser, so no generated credential is ever left unclaimed in a log. Migrations run at boot, and upgrading is dropping in a new binary and restarting.

install.sh
# fresh Debian / Ubuntu server
git clone https://github.com/jarno9981/Dialix.git
cd Dialix
sudo ./install.sh service

# or: everything on one machine —
# MariaDB, database plane, SuperCenter
sudo ./install.sh --selfhost
  • Seeds no account — you create the first admin, in the browser
  • Clustering: shared DB + Redis, weighted nodes, one node adopts the secret
  • TLS via a real Let’s Encrypt cert (Cloudflare DNS-01) or self-signed
Technical fact sheet
Server
Single Rust binary (axum), ~27 MB, x86-64 + arm64
Media engine
Asterisk 20+, driven live over ARI
Database
SQLite by default; MySQL/MariaDB for scale
Web console
Vanilla JS, no build step, self-served by the binary
Codecs
Opus, G.722, PCMU/PCMA with server-side transcoding
Deploy target
Any Debian/Ubuntu VM — a €4 VPS is enough
For MSPs

Host a fleet. Bill it. Keep every customer isolated.

Run several organizations on one install, each with its own users, numbers, trunk and branding on its own subdomain — or hand a customer a fully custom domain with correct TLS out of the box.

  • An organization exists because its database exists — no registry row can conjure or hide one
  • Per-tenant SIP trunks: each org can run its own carrier account, or fall back to yours
  • Custom branding — logo, colors, login background — reflected in web, mobile and email
  • One certificate per domain via built-in SNI, so every tenant’s domain gets correct TLS
Head panel · Tenantsoperator
DE
default
calling.yourco.com
operator
AC
acme
acme.calling.yourco.com
tenant
VA
vanrijt
vpbx.vanrijtit.nl
tenant · custom domain
Pricing

Free forever, or ten euros a tenant a month

No per-seat license fees, no minute billing. Community edition is free and self-hosted forever. Unlock AI and multi-tenant for ten euros per tenant, per month — priced by tenant, not by seat, so a 20-customer install is 20 × €10 = €200/month, however many users each customer has.

Community
€0forever

The full PBX, self-hosted on your own server. No AI features, single organization only.

  • Full PBX — calls, queues, IVR, voicemail
  • Web console, mobile app, desk phone support
  • Self-hosted, unlimited extensions & minutes
  • AI features — transcription, summaries, natural-language scheduling
  • Multi-tenant — multiple organizations on one install
Get Community, free
All features
€10/ tenant / month

Everything Community has, plus AI and unlimited multi-tenant hosting — billed monthly, per tenant you host. Run 20 customer tenants? That's 20 × €10 = €200/month total, with unlimited seats inside each one.

  • Everything in Community
  • AI transcription, call summaries & voicemail-to-text
  • Natural-language scheduling ("vanaf vrijdag 17:00 tot maandag 7:30")
  • Multi-tenant — unlimited organizations, one install
  • Per-tenant SIP trunks, branding & custom domains
  • Priority security & upgrade support
Unlock all features

€10 per tenant, per month — billed by tenant count, not seats. 1 tenant = €10/month, 20 tenants = €200/month. No seat limits, no minute limits, no card required for Community.

Example

One tenant, 10 extensions

3CX and Yeastar bill per user/extension, so cost climbs with headcount. Dialix bills the tenant, not the seat — 10 extensions costs the same as 1.

Dialix
€10/ mo
€10 flat, per tenant
3CX
€35–100/ mo
~€3.5–€10 / user / mo
Yeastar
€40–150/ mo
~€4–€15 / user / mo

3CX and Yeastar figures are illustrative street pricing for a 10-user tenant and vary by plan, region and promotion — check their current price lists for exact quotes.

Migrating from 3CX

Export your extensions. Import them. Repoint your phones.

Export extensions from 3CX Admin Console as CSV, preview the import, then apply it — each row becomes a registrable Dialix SIP extension, reusing the existing auth password where possible.

Where Dialix matches or beats 3CX

  • Inbound/outbound calling, queues, hunt groups, IVR, voicemail + transcription
  • Multi-vendor provisioning, native desktop softphone, mobile app, web client
  • Edge SBC / Raspberry-Pi bridge with WAN-outage survivability
  • Self-hosted, no per-extension licensing

Where 3CX is still ahead

  • Built-in WhatsApp / live-chat / website-chat omnichannel
  • Deeper Microsoft 365 / Teams integration
  • Breadth of CRM connectors and years of on-device firmware testing
8/10

Honest self-assessment as a 3CX replacement for a typical SMB or contact-center deployment — core telephony, provisioning and one-command install on par; omnichannel is the remaining gap.

Your PBX. Your server. Your data.

Read the docs, run the installer, and create your first administrator in the browser — in the time it takes to provision a VM.