Asterisk & FreePBX

A WebRTC Softphone That Works with Asterisk, FreePBX, and More

A drop-in browser softphone for your existing Asterisk or FreePBX setup. No new servers. No rip-and-replace. Users open a link and the phone works.

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Works with your existing stack

No rip-and-replace. No new PBX.

Stylish Softphone is a front-end layer. It connects to your existing PBX over SIP via WebSocket (WSS). If your system runs chan_pjsip with a WSS transport, it works.

Asterisk FreePBX FusionPBX VitalPBX OpenSIPS
Connection overview
Browser
User's Chrome / Firefox
→
SIP over WSS
DTLS-SRTP audio
→
Asterisk / FreePBX
chan_pjsip, port 8089
Requirements

What you need on your PBX

The only Asterisk-side requirement is a working chan_pjsip endpoint with WebRTC enabled and a WSS transport. Most modern Asterisk setups already have this.

; pjsip.conf — WSS transport
[transport-wss]
type=transport
protocol=wss
bind=0.0.0.0:8089
[your-extension]
type=endpoint
webrtc=yes
dtls_auto_generate_cert=yes

FreePBX users: WebRTC is available under Admin → Advanced Settings. Enable WebRTC and configure a WSS transport on the PJSIP settings page.

DTLS-SRTP for audio

WebRTC requires encrypted media. Stylish Softphone negotiates DTLS-SRTP on every call. Audio never travels unencrypted.

ICE for NAT traversal

ICE handles NAT. If your Asterisk is behind a firewall, configure STUN or a TURN server in your pjsip transport settings. The softphone handles the rest.

Codec support

Works with G.711 (ulaw/alaw), G.722 wideband, and Opus. Enable the codecs you want in your Asterisk endpoint config.

No SIP config for end users

SIP credentials are generated in the admin portal and stored encrypted. Users never see a SIP server field or registration URL.

Setup

Four steps from PBX to working softphone

Step 01

Enable WebRTC on your PBX

Add a WSS transport to pjsip.conf and set webrtc=yes on your endpoints. Takes 10 minutes on a standard Asterisk setup.

Step 02

Enter your SIP domain

In the admin portal, enter your Asterisk or FreePBX domain. Stylish Softphone connects over WSS and validates the connection.

Step 03

Provision users

Add users, assign extensions, and generate SIP credentials in the admin portal. Send users their login link. No SIP config for them to enter.

Flat-rate pricing for your whole team

No per-seat fees until you genuinely scale. Starter from $49/mo covers small deployments.

View pricing →
$49
Starter
$99
Growth
$199
Pro

Try it against your Asterisk setup

Free 14-day trial. Bring your own PBX. No credit card required.

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