Manual dialing, auto-dialer, automated IVR, live transfer, direct bridge, and phone surveys — all from any web browser. Connects to your existing Asterisk or Magnus server with zero reconfiguration.
From a single manual call to a multi-level visual IVR flow — every outbound workflow your team needs, in one browser tab.
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 and let the system dial numbers one by one automatically. When someone answers, you talk. When you hang up, the next call starts.
Fully automated. Upload your number list and an audio recording. The system calls each number server-side, plays your message, and moves on — even with your browser closed.
Combines automation with a human touch. Plays your recorded message automatically — if the customer presses your chosen key, the call instantly transfers to your live agent.
The most seamless handoff. Plays a first announcement, then when the customer presses your key, plays a second message — and bridges the live call directly to your SIP extension.
Automated phone surveys at scale. Plays your intro recording, then prompts the customer with a follow-up message — and captures every keypress they enter as their survey response.
Build multi-level interactive voice flows without any coding. Design a full call tree with menus, digit collection, live call bridges, and hang-up points — visually. Each branch leads to a different action depending on what the caller presses.
Plays an announcement and waits for a key press. Each key (1–9, *, #) branches to a different next node.
Plays a prompt, then records a sequence of digits the caller enters — ideal for account numbers, PINs, or survey answers.
Plays an optional audio clip, then bridges the live call directly to any SIP extension or phone number. Campaign pauses while connected.
Plays a closing message and ends the call cleanly. Can be attached to any branch for a graceful goodbye.
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.
When a customer presses your chosen key, the recording stops instantly and the call transfers live to your browser dialer — no extra hardware needed.
Every campaign number is tracked as Answered, Busy, No Answer, Transferred, or Failed. View 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 logs digits server-side via RFC 2833. Manual and Agent calls capture them live and display in real time.
Admins see every active call across all tenants in real time — SIP user, number, mode, and live duration. The dashboard updates every 10 seconds.
After every call, agents write a quick note and optionally schedule a callback. Notes are stored per phone number and can be reviewed anytime.
Build a Do-Not-Call list per tenant. Numbers on the blacklist are automatically filtered out when a campaign CSV is uploaded.
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.
Set allowed calling hours and a daily call limit per tenant. Managers can set individual daily limits per SIP user. Campaign pauses automatically when a limit is reached.
Receive a POST request after every call. Payload includes number, duration, result, mode, DTMF digits, and agent note — ready for n8n, Zapier, or any CRM.
Full call detail records for every level. Admins see all tenants. Managers filter by SIP user. Agents see their own calls. Filter by date range and export to CSV.
Create professional IVR announcements instantly. Type your script, choose an AI voice, preview it live, and download as MP3 — without leaving the platform.
Unlike traditional softphones, CellHub Dialer never exposes your SIP infrastructure to end users. Every credential, every IP, every domain stays server-side.
All modes included. No hidden fees, no per-user charges. Pay per company.
Contact us on Telegram to set up your account. We'll have you live within 20 minutes.
Includes Magnus server setup & full onboarding. Live within 1 business day.
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.