v1.8.0Voicemail. Caller ID. A stronger core.

THE SELF-HOSTED COMPANY PBX

Your calls.
Your flow.
Your control.

Your office phone system.
On your own infrastructure.

Connect your SIP provider, route calls visually and give every desk a browser phone and voicemail inbox. Asterisk-powered, open source, and built for small offices of around 20 extensions.

Get MonoSIP Read the docs

Asterisk powered Docker ready MIT licensed

YOUR CALL, CONNECTED.FIG. 01

Support queue

The caller chooses Support. The queue rings an available team member.

Interactive illustration · no call is placed

ONE SYSTEM.
THE WHOLE CONVERSATION.

Asterisk 22Visual IVRBrowser WebRTCSelf-hosted DockerEnglish + Türkçe

NOW AVAILABLE / v1.8.0

More of the everyday, covered.

See what changed

The product is MonoSIP.

The console, the public site and the Docker image share the MonoSIP name. Pull ghcr.io/monosip/monosip.

A voicemail inbox that stays in view.

Waveforms, saved greetings and unheard counts stay with the voicemail inbox.

Sign-in choices stay with you.

Language and dark mode chosen on the sign-in screen are the same choices inside the admin and the portal.

01 / YOUR PHONE SYSTEM, IN FOCUS

Less switching tools.
More connecting people.

A clear view of your calls, your team and the routes in between. All in the same admin panel.

Explore the product screens

your-pbx / admin / dashboardINTERFACE CONCEPT

The whole office, at a glance.

See active calls, extension presence and queue activity in one live view. Switch to TV mode for the team.

01 / 08

Animated interface concepts with illustrative sample data.

02 / BUILT FOR THE OFFICE

The essentials.
Already on the line.

From the first ring to the final follow-up.
The everyday tools your team actually needs.

01 / THE EVERYDAY ESSENTIALS

Every desk, connected.

Extensions, live presence, forwarding and voicemail. One password for the employee portal and SIP phone. Manage the team in one place.

CSV import / exportDNDDesk phones

02 / A BETTER WAIT

Keep callers in the loop.

Ring groups and queues with six routing strategies. Position announcements, hold music, overflow and a callback option when the team is busy.

Queue callbackWrap-upOverflow routes

03 / ON YOUR SCHEDULE

Your hours. Your rules.

Route calls for open hours, lunch breaks and holidays. Override the schedule when plans change, with a destination for every state.

Weekly schedulesHolidaysManual override

04 / THE FULL PICTURE

Know how the day went.

Live call activity, detailed call records and recording playback. Open a call’s IVR timeline and see the path it took.

Call timelinesReportsTV mode

05 / YOUR EXISTING CONNECTIONS

Bring your SIP provider.

Registered or IP-based trunks, DID routing and outbound failover. Set the provider caller ID separately from the registration username.

Provider caller IDInbound DIDsFailover

06 / AFTER THE RING

Leave room for a message.

Send unanswered calls to an extension’s voicemail. Employees can see the caller, play messages and manage their own inbox in the portal.

Voicemail inboxRead stateForwarding

07 / CONNECT YOUR WORKFLOW

Put calls in context.

Use REST endpoints, signed webhooks and visual API lookups to connect your existing tools. Embed a browser call button on your website.

REST APIWebhooksCall widget

08 / OPERATE WITH CLARITY

Know when you’re ready.

A health gate checks the database, Asterisk and dialplan. Manage HTTPS, optional admin 2FA, access rules and persistent data on your infrastructure.

Readiness gateAutomatic SSLAccess controls

The complete feature reference

03 / VISUAL IVR & NEURAL SPEECH

Draw the flow.
Let the conversation follow.

A welcome, a menu, a live API lookup. Build a call experience that knows where to go next.

THE VISUAL FLOW DESIGNER ANIMATED MOCKUP

Build, test and publish. Live calls keep the flow version they started with.

Connect the right pieces.

17 node types for menus, speech, conditions, schedules, queues and more. Each exit has a clear next step.

Bring your data into the call.

Look up a caller in your API. Map JSON responses to variables. Speak the result or route to the right extension.

Find your voice.

Edge Neural voices for English and Turkish, plus EMA for offline Turkish. Prompts support variables and speech filters.

Meet the flow designer

Incoming call Welcome Menu API lookup Right destination

Illustrative call path. API nodes include success and error exits; connect a fallback before publishing.

04 / A PHONE, WHERE YOU WORK

Your browser.
Now taking calls.

A full softphone. A personal portal.
One extension to tie it all together.

Dial, answer, hold, mute and transfer from the browser. Move between portal pages while your call stays connected.

A mini phone on every portal page
Directory with live coworker presence
Your calls, recordings and voicemail inbox
Desk phone and browser can ring together

Take a closer look Explore the voicemail inbox

HTTPS and microphone access required. Reloading the browser ends the call. Mobile background calling has limitations.

YOUR EXTENSION. ANY DESK.

Browser softphone · animated interface concept

05 / KNOW WHAT YOU’RE RUNNING

Open by design.
Yours to operate.

A familiar foundation.
A simpler way to bring it together.

Asterisk handles SIP and media. The API runs your call flows and configuration. PostgreSQL stores the state. Your team works in the web app.

One prebuilt image

Asterisk, API, web, embedded PostgreSQL and speech. Persist everything under /data.

Or Docker Compose

Run the services together on a Linux VM, with nginx and coturn for the edge and remote media.

Look under the hood

DOCKER COMPOSE · SERVICE MAP

Browser / portal SIP phones / provider

YOUR LINUX SERVER

nginxTLS · ACME · WSS

Asterisk 22PJSIP · ARI · AMI

Web appAdmin + portal

Fastify APIFlows · config · jobs

PostgreSQLSettings · CDR · flows

SpeechEdge / EMA · embedded

API ↔ Asterisk over ARI / AMI · coturn relays remote WebRTC media.

Persistent volumesdatabase / recordings / config

06 / PART OF YOUR WORKFLOW

A good connection.
Beyond the phone.

Use an API lookup to recognize a customer. Originate a call from your tools. Send call events to the systems your team already uses.

Explore integrations

REST API & webhooks

Click-to-call, call records and recordings. HMAC-signed events for ringing, answered, ended and missed calls.

A call button for your website

Create an embeddable WebRTC call widget in Admin → Integrations, with an optional callback form.

Speech that works with your data

Use mapped API values in prompts. Format digits, amounts and other values with template filters.

07 / YOUR FIRST CONNECTION

One image.
A whole phone system.

Pick a Linux server. Give it a home.
Bring your office online.

01
Prepare your server.
Linux x86-64, Docker, a public hostname and persistent storage. Plan 2–4 vCPU and 4–8 GB RAM.
02
Pull, mount and run.
Replace the example hostname and IP. Mount a persistent volume at /data and configure the required ports.
03
Configure your first call.
Set strong credentials, configure HTTPS, add your SIP provider and connect an extension.

Follow the full installation guide

PREBUILT IMAGElinux/amd64

docker run

docker run -d --name monosip \
  --restart unless-stopped \
  -p 3000:3000 \
  -p 10000-10100:10000-10100/udp \
  -v monosip-data:/data \
  -e EXTERNAL_IP=203.0.113.10 \
  -e PUBLIC_HOST=pbx.example.com \
  ghcr.io/monosip/monosip:1.8.0

The image exposes the app on port 3000. Use an HTTPS proxy for your public hostname. RTP needs UDP 10000–10100; desk-phone SIP may also need port 5060. Networking details →

Release notes All install options

OWN THE SYSTEM.
UNDERSTAND THE RESPONSIBILITY.

Access, on your terms.

Optional admin TOTP, IP allowlists, login-flood protection and encrypted stored integration secrets.

Your data has a home.

Keep CDR and recordings on your infrastructure. Configure access, retention, archives and your own backups.

Operate thoughtfully

A FEW GOOD QUESTIONS

Before you
pick up.

Clear answers, straight from the project.

Find your next step

What is MonoSIP?

MonoSIP is an open-source Asterisk-based company PBX for small offices, designed around roughly 20 extensions. It includes an admin panel, visual IVR designer, browser softphone and employee portal.

What do I need to run it?

Use a Linux x86-64 server with Docker. The prebuilt image targets linux/amd64. The deployment guide recommends 2–4 vCPU and 4–8 GB RAM; neural speech runs on the server’s CPU. You also need persistent storage, a public hostname and the appropriate signalling and media ports.

Can I keep my SIP provider and desk phones?

MonoSIP supports registered and IP-based SIP trunks, multiple providers and route failover. Yealink, Fanvil and Grandstream provisioning profiles are documented. Check your provider’s transport, codecs, networking and device configuration before switching.

Does the softphone work on mobile?

The portal works in a mobile browser, but mobile browsers can suspend background tabs. Inbound calls with the screen off are unreliable. For dependable mobile reachability, use a compatible native SIP softphone with QR provisioning. MonoSIP does not ship a first-party mobile app.

Does everything work offline?

The PBX and your locally stored data run on your infrastructure. EMA provides offline Turkish speech after its models are built into the image. Edge Neural speech, including English, needs outbound internet access during generation. External SIP carriers and integrations have their own network requirements.

Can I use MonoSIP commercially or contribute?

Yes, the MIT license allows commercial use, modification and redistribution with the copyright notice. You can fork and adapt MonoSIP freely. Visit the official repository for current development and contribution guidance.

GOOD CONVERSATIONS START SOMEWHERE.

Make the next one
on your terms.

Open source. Self-hosted. Ready for your office.

Let’s get connected Explore the source

Your calls.Your flow.Your control.