Manual dialing, auto-dialer with agent, automated IVR campaigns, and smart IVR-to-agent transfer — all from any web browser. Connects to your existing Asterisk or Magnus server. No app, no plugin, no exposure of your SIP infrastructure. Complete with CDR reports, live call monitoring, blacklist, callback scheduler, webhooks, and more.
Every SIP user can switch between modes depending on their workflow — no extra setup required.
Classic browser-based SIP dialer. Type a number, hit call. Full DTMF support for IVR navigation and keyboard shortcuts for speed.
Upload a CSV list of numbers and let the system dial them one by one automatically. When someone answers, you talk. When you hang up, the next call starts after a configurable delay.
Fully automated. Upload your number list and an audio recording. The system calls each number server-side, plays your message when answered, and moves to the next — even with your browser closed.
Combines automation with human touch. Plays your recorded message automatically — if the customer is interested and presses your chosen key, the call instantly transfers to your live agent.
Connect your existing SIP server — no reconfiguration needed on your users' end.
Works with any Asterisk, Magnus Billing, FreePBX, or compatible PBX system. We run a one-command setup script.
Your company gets an isolated account. SIP users are imported automatically from your Magnus server.
No app, no plugin. Agents log in with their SIP credentials, choose a mode, and start calling immediately.
Our Janus WebRTC Gateway bridges browser audio to your SIP server in real time. Your SIP infrastructure stays completely hidden.
A complete outbound calling platform with enterprise-grade features, ready out of the box.
Fully compatible with Chrome, Firefox, Edge, and Safari. No extensions or plugins — just open a URL and call.
Create unlimited campaigns per user. Upload CSV lists, set your delay between calls, track progress in real time, and pause or stop at any moment.
IVR campaigns run directly on the server — no browser or agent required. Upload any MP3 or WAV file and let it run 24/7 unattended.
Each company has a manager who can add, remove, enable, or disable SIP users independently — no support ticket needed.
Users never see your SIP domain or server IP. All traffic proxied through our WebRTC gateway — your infrastructure stays private.
Every company runs in complete isolation. Separate users, SIP servers, campaigns, and manager access. Unlimited users per tenant.
IVR campaigns that convert. When a customer presses your chosen key, the recording stops instantly and the call transfers live to your browser dialer — no separate hardware needed.
Every campaign number is tracked as Answered, Busy, No Answer, Transferred, or Failed. View progress and stats live as the campaign runs.
Paste one command into your Magnus/Asterisk server — it configures everything automatically. No manual SIP configuration needed.
Capture keypresses entered by the called party — across all modes. IVR campaigns log digits server-side via RFC 2833. Manual and Agent calls capture them live via Janus WebRTC events and display them on the dialer screen in real time. All logs are viewable per call and auto-purged every 4 hours.
Admins see every active call across all tenants in real time — SIP user, number, mode, and live duration. Managers see their own agents. The dashboard counter updates every 10 seconds automatically.
After every call, agents are prompted to write a quick note and optionally schedule a callback. Notes are stored per phone number and can be reviewed anytime. IVR campaign notes are managed from the campaign dashboard.
Build a Do-Not-Call list per tenant. Numbers on the blacklist are automatically filtered out when a campaign CSV is uploaded — they never get dialed. Manage the list manually from the campaign page at any time.
Schedule a follow-up call for any number directly from the post-call screen. A badge on the dialer shows pending callbacks at a glance. Mark callbacks as done to keep the list clean.
Admins can set allowed calling hours and a daily call limit per tenant. Managers can set individual daily limits per SIP user. When a limit is reached the campaign pauses automatically and the agent is notified.
Configure a webhook URL and receive a POST request after every call. The payload includes phone number, duration, result, call mode, DTMF digits pressed, and the agent's note — ready to pipe into n8n, Zapier, or any CRM.
Full call detail records for every level. Admins see all tenants with company name. Managers filter by SIP user. Agents see only their own calls. Filter by date range and status. Export to CSV in one click.
Create professional IVR announcements instantly. Type your script, choose an AI voice (6 OpenAI voices), preview it live, and download as MP3 — ready to use in your IVR campaigns without leaving the platform.
Unlike traditional softphones, CellHub Dialer never exposes your SIP infrastructure to end users.
One plan. All four modes included. No hidden fees, no per-user charges.
Per company • Unlimited users • All 4 modes included
Contact us on Telegram to set up your account. We'll have you live within 20 minutes.
Full source code. Deploy on your own server, under your own brand.
Source code delivered after payment. Includes one-time setup support.
Contact us on Telegram — we'll configure your account and import your SIP users the same day.