A fully open source, self-hosted business phone system: the everyday features of 3CX, on your own server. You bring a SIP trunk and phone numbers (the reference provider is voip.ms); SelfPhoneNG does the rest.
Status: pre-1.0 (v0.0.15), for testing. See what's left for v1.
| For | One small business or home lab: 1–100 extensions on one server |
| Runs on | Linux with Docker Compose (or Portainer); 1–2 CPUs and 2 GB of memory is plenty |
| Built on | Asterisk 24, TypeScript (Fastify + React), PostgreSQL 18 |
| You bring | A SIP trunk with at least one number, and a DNS name for the server |
| License | AGPL-3.0-or-later; no closed-source parts |
- Numbers and trunks: voip.ms trunks (registration or IP auth, UDP/TCP/TLS with SRTP) or any other provider as a generic trunk; phone numbers routed to a person, group, queue, receptionist or voicemail.
- Outbound rules with calling permissions per extension, trunk failover, and limits on outside calls at once.
- Digital receptionists (press 1 for sales …), office hours and holidays with override codes, announcements.
- Ring groups, queues, call parking and pickup, conference rooms, paging and intercom, music on hold.
- 911: always allowed, with an E911 caller ID, and an email, a ringing phone and a dashboard banner when someone dials it.
- Browser phone for every user: hold, mute, keypad, blind and attended transfer, presence, click-to-call. Nothing to install.
- Desk phones set up automatically: Yealink and Grandstream (account, busy-lamp keys on Yealink, phonebook, time, dial rules). Polycom/Poly VVX and Snom are experimental.
- Phone apps: Linphone set up by scanning a QR code. Our own Android and iPhone app is in progress.
- Anything else that speaks SIP, analog adapters included, set up by hand from the details the admin pages show.
- Voicemail with email, web playback, greetings recorded in the browser, and the light on desk phones.
- Call history for admins and for each person, with CSV export and call reports by email.
- Call recording per extension or number, with an optional announcement and retention rules.
- Local AI (optional, your own server or network only): voicemail transcripts, call transcripts and summaries, searchable.
- Live dashboard: calls in progress, who is on the phone, trunks, queues and parked calls; listen, whisper or barge in.
- Queue wallboard for a screen on the wall: waiting callers, each agent's state, today's numbers.
- Agent break reasons (Lunch, Meeting …) picked on the web or dialed from the phone, shown on the wallboard and added up in reports.
- Reports: calls per day, missed calls, busiest hours, queue service levels, agent breaks, and each person's own numbers.
- Contacts: a builtin company and personal phonebook, or Contrack. Caller names show on phones and in history.
- Spam check (off until you turn it on): outside callers who aren't contacts are looked up with IPQualityScore or the FTC complaint list and flagged. Calls are never blocked by it; blocking a number is your choice.
- Sign-in: passwords, two-step sign-in for admins, passkeys, OpenID Connect (Keycloak and others) and LDAP / Active Directory.
- Against toll fraud: calling permissions, call limits, and alerts for international or premium calls, calls your provider billed that the phone system didn't make, and a falling balance.
- Hardened by default: SIP over TLS, an option to allow desk phones only from the office, no shell-running modules in Asterisk, an audit log, and notes for fail2ban and CrowdSec.
- Install with one script; a setup wizard takes a new install to working calls.
- Updates with one command, with a backup before every database change and a roll-back when the new version doesn't start.
- Backups every night, a restore script, and health checks for the trunk, public IP and disk.
- Integrations: a REST API with tokens, and webhooks.
| Wanted | Where it stands |
|---|---|
| More SIP providers | Telnyx, Flowroute and Twilio first; today they work only as a generic trunk |
| More desk phone makes | Fanvil, Gigaset, Cisco; Polycom and Snom need trying on real phones |
| Analog adapters set up automatically | Grandstream HT802 V2 first; by hand today (how) |
| Mobile app | Being built; without push, phone apps ring only while open |
| SMS | Built for voip.ms, parked until wanted |
| More local AI | Typed prompts, type-to-speak, an AI receptionist (spec) |
| Recording prompts by phone | Planned after v1 |
Not planned: fax, video calls and team chat, several companies on one install, clustering.
You need a Linux server with Docker and the compose plugin, a DNS name pointing at it, and a voip.ms account with a number.
git clone https://git.dou.bet/spng/selfphoneng.git
cd selfphoneng
./scripts/install.shThe script asks for the host name and the first admin, then builds and starts everything (the
first build compiles Asterisk, so it takes a while). Open https://<your host name>/, sign in,
and follow the setup wizard. The admin guide covers each step, the
ports to open and what to do when a call doesn't work.
| Part | What it does |
|---|---|
| asterisk | All SIP and audio: phones, the trunk, voicemail, queues, conferences, recording |
| api | Owns the database, writes Asterisk's config from it, follows calls live, serves the web app |
| web | The admin pages, each person's portal and the browser phone (served by the api) |
| postgres | Everything you set up, call history and the audit log |
| proxy | HTTPS: the bundled Caddy, or your own nginx |
PostgreSQL is the single source of truth: every change in the web app regenerates Asterisk's config from it. Details in the spec.
- Installing and running: the admin guide, with voip.ms setup, network and firewall, updating and backups and troubleshooting.
- Features: sign-in, contacts, spam check, local AI, analog phones.
- The law: E911 and call recording.
- What changed: the changelog and the ready-for-v1 checklist.
- Writing code (people and AI alike):
AGENTS.mdfirst, then the spec and the roadmap.
pnpm install
docker compose up -d postgres asterisk
pnpm dev # api + web, reloading
pnpm lint && pnpm typecheck && pnpm test # before calling anything doneAGPL-3.0-or-later. Asterisk is GPL-2.0. Nothing closed-source is included or needed.