CellHub Dialer connects your team to your existing SIP/PBX infrastructure via WebRTC — no downloads, no plugins. Your SIP server stays invisible and fully protected.
Connect your existing SIP server — no reconfiguration needed on your users' end.
Works with any Asterisk, Magnus Billing, FreePBX, or compatible PBX system.
Your company gets an isolated tenant. You manage your SIP users from the Manager Panel.
No app, no plugin. Agents just open a URL, log in with their SIP credentials, and start calling.
Our Janus WebRTC Gateway bridges browser audio to your SIP server in real time.
A complete browser dialer with enterprise-grade features, ready out of the box.
Fully compatible with Chrome, Firefox, Edge, and Safari. No extensions or plugins needed.
Supports G.722 wideband, uLaw, and aLaw codecs for high-quality voice calls over WebRTC.
Users never see your SIP domain or server IP. All traffic is proxied through our WebRTC gateway — your infrastructure stays private.
Your team manager can add, remove, enable or disable SIP users independently — no need to contact support.
Full in-call DTMF support for IVR navigation. Also supports physical keyboard input for fast dialing.
Each company has a fully isolated account with its own users, SIP server config, and manager access.
Unlike traditional softphones, CellHub Dialer never exposes your SIP server address to end users.
One plan. Everything included. No hidden fees.
Per company • Unlimited users • Monthly billing
Contact us on Telegram to set up your account. Payment handled via agreement.
Reach out on Telegram and we'll have your account live within 24 hours.