Prepaid billing
Per-second rating from customer and provider rate cards, live balances backed by a real ledger, and margin on every call.
Self-hosted softswitch · billing · dialer
Run your own carrier-grade phone switch.
A complete softswitch, prepaid billing panel and predictive dialer you install on one Debian server. Trunks, routing, DIDs, CDRs and per-second billing — yours, on your own box.
- 1,000 downloads
- Asterisk 20
- Debian 12
- Panel in 10 languages
Live call routing
Illustration
Caller
SIP INVITE
KatoCall
rate · route · bill
Connected via B
Carrier A
Rejected
Carrier B
Answered
Carrier C
—
Destination
0:42
Answered
1,000downloads
14install steps, then a full self-check
10languages for the panel and handbooks
1server runs the whole switch
Platform
One server. The whole switch.
Every layer of KatoCall runs on a single Debian 12 server that you control — from the SIP stack to the billing ledger to the panel your customers sign in to.
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web
Admin panel & customer portal
Operators run the switch; customers see their own calls.
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PostgreSQL
Billing engine
Rate cards, per-second rating and a PostgreSQL ledger.
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AMI
Live call bridge
Follows every channel in real time and writes the CDR at hangup.
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SIP
Asterisk 20 with PJSIP
Trunks, endpoints and DIDs, generated from the panel.
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OS
Hardened Debian 12
Firewall, fail2ban and TLS, set up by the installer.
How it works
From carrier to CDR in four steps.
Set it up once in the panel. Every call after that follows the same path.
Connect your carriers
Add SIP trunks by IP or by registration, over UDP, TCP or TLS.
Add your customers
Give each account IP authentication or a softphone login, a rate card and a balance.
Route every call
Routing groups pick trunks by prefix, priority and weight, and fail over when a carrier rejects.
Bill and report
Calls are rated per second, watched live and saved as CDRs you can analyse and export.
What's inside
Everything a switch needs, in one panel.
Not a stack of parts to wire together — billing, routing, inbound, outbound and security ship as one installed system with a single admin panel.
Least-cost routing
Routing groups pick trunks by prefix, priority and weight, with failover to the next carrier when one rejects the call.
Inbound DIDs
Point a carrier's numbers at accounts and forward them anywhere, with the caller ID your carrier requires enforced on the way out.
Predictive dialer
Optional outbound campaigns and an IVR menu, pacing calls against agent availability — installed only if you want it.
Hardened by default
fail2ban on SIP, the panel and SSH with permanent bans, a firewall the installer configures, and free Let's Encrypt TLS.
Live calls & CDRs
Watch every call on the switch in real time — who's billed, the carrier, the cost as it climbs — then export the CDRs to CSV.
Call routing
Least-cost routing with automatic failover.
Each call is matched against its account's routing group. If a carrier rejects it, the next trunk in line takes the call.
- Match calls by destination prefix
- Order trunks by priority and share load by weight
- Fail over to the next carrier when one rejects
- Send the caller ID each carrier requires
SIP & VoIP
Carrier-grade SIP, configured for you.
The panel writes the PJSIP configuration, and every value is validated before it reaches Asterisk.
PJSIP on Asterisk 20
A modern SIP stack, built and configured by the installer.
UDP, TCP and TLS
Pick the transport for each trunk.
IP or registration trunks
Authenticate carriers by address or by username and password.
Carrier IP pools
Trust a carrier's whole signalling range with IPs and CIDR blocks.
Customer endpoints
IP-authenticated PBXs or registered softphones.
NAT-aware
Set the public address and local networks from the panel.
DIDs in E.164
Point inbound numbers at accounts and forward them anywhere.
Codecs and media
A codec list per endpoint, with proxied RTP or direct media.
Stuck-call protection
RTP timeouts end calls whose audio has stopped.
Analytics
See every call as it happens.
Live calls stream to the panel in real time. When they end, the CDRs are there to filter, analyse and export.
- Live calls with the billed account, carrier and running cost
- CDR statistics by period, account, supplier and prefix
- Billed minutes, revenue and margin on every call
- CDR export to CSV
Reliability
Built to keep running.
The details that matter once real traffic is flowing.
Verified install
The installer checks its own work and reports every result before you take traffic.
Update notices
The switch checks for new versions daily and tells you. It only updates itself if you opt in.
Survives restarts
Calls already in progress are picked back up after a reconnect, and billing resumes from the real answer time.
Ledger-backed balances
Every charge and top-up is recorded as a ledger entry in PostgreSQL.
Nightly backups
The database and the switch's secrets are backed up every night, and logs are rotated.
Works with what you have
- Any SIP carrier
- CSV rate import with column mapping
- CSV CDR export
- Let's Encrypt HTTPS
Security
Hardened from the first boot.
A phone switch on the public internet is scanned constantly. KatoCall ships with its defences already switched on.
- fail2ban on SIP, the panel and SSH
- Permanent bans you manage from the panel
- A firewall configured by the installer
- Free HTTPS with Let's Encrypt
- Sign-in protected against cross-site request forgery
- Every value written to Asterisk is validated
Install
One command. One box.
A fresh Debian 12 server is all you bring. The installer builds Asterisk, sets up PostgreSQL, the panel, the firewall and fail2ban, and verifies the lot.
v1.0.8 self-hosted · run it on your own server
Run this on a fresh Debian 12 server, as root:
curl -fsSL https://katocall.com/get.sh | sudo bash
It fetches everything, verifies the checksum, and runs the full install.
Then open your server in a browser and sign in — you'll set a new password right away:
useradmin
passKatocall
1,000 downloads
# one fresh Debian 12 box, one command $ curl -fsSL https://katocall.com/get.sh | sudo bash ==> 00-preflight.sh ok ==> 01-install-asterisk ok ==> … ok ==> verify.sh 50 checks passed KatoCall is installed. → http://your-server/
You'll need
- Debian 12 (Bookworm), 64-bit — a fresh VPS
- 2 CPU cores, 4 GB RAM, 20 GB disk
- A public IPv4 and root / sudo
- A SIP trunk from your carrier
- A domain name — only if you want HTTPS
Documentation
Install & setup handbooks, in ten languages.
A complete guide to installing the switch and setting up trunks, customers, DIDs and the dialer — plus a “?” beside every field in the panel itself.
FAQ
Questions, answered.
What do I need to run KatoCall?
A fresh Debian 12 (64-bit) server with 2 CPU cores, 4 GB RAM and 20 GB of disk, a public IPv4, root access and a SIP trunk from your carrier. A domain name is only needed for HTTPS.
How do I update?
Run the update command on your server. It verifies the download before installing, and your switch checks for new versions every day and tells you when one is available.
curl -fsSL https://katocall.com/update.sh | sudo bashWhich carriers does it work with?
Any carrier that gives you a SIP trunk — authenticated by IP or by registration, over UDP, TCP or TLS.
Do I have to install the dialer?
No. The predictive dialer and IVR are optional and only installed if you ask for them.
Can my customers see their own calls?
Yes. Each customer gets a portal with their own call records, DIDs, rates and transactions — never anyone else's.
Which languages are supported?
The panel and the handbooks are available in ten languages: English, Spanish, Portuguese, French, German, Russian, Chinese, Japanese, Arabic and Hindi.
Your own switch, running today.
One fresh Debian 12 server and one command. Everything else happens in the panel.
curl -fsSL https://katocall.com/get.sh | sudo bash
Contact
Get in touch.
Questions, setup help, or a bug — send a message and we'll get back to you.