SipRegistrar – turn your firewall into a lightweight SIP PBX/registrar for small offices.

humaxoid

Hi everyone,

I'd like to share a package I've been working on for a while: SipRegistrar.
It solves a fairly niche but real-world problem—merging the roles of a classic edge router and a SIP registrar into a single, low-maintenance box for small business environments.

If you've ever deployed a separate on-premises PBX (Asterisk/FreeSWITCH) just to let 5–50 desk phones call each other by short extensions, or to connect a couple of analog gateways to the outside world, you probably know the feeling: overkill. One more VM or Raspberry Pi to maintain, another set of updates, another point of failure.
This package eliminates that middleman.

What it does

The package installs Kamailio 6.1.1 on pfSense 2.7.2 and turns your existing firewall into a vendor‑neutral SIP registrar.
Any RFC 3261‑compliant device—Grandstream, Yealink, Cisco, Fanvil, Snom, or softphones like Zoiper/MicroSIP—can register and start calling each other using short, custom extensions (2–5 digits).

It also handles:

Outbound routing to external SIP trunks (e.g. Zadarma, Rostelecom, or any provider) and analog FXO/FXS gateways (Yeastar, Grandstream, AudioCodes, etc.)
Prefix‑based routing – send calls with 9 to one trunk, 8 to another
Incoming DID routing with simple business‑hours / after‑hours rules
Ring/hunt groups (parallel ring‑all or sequential with timeouts)
A clean pfSense‑style GUI under Services → SIP Registrar with:
live registration status
active calls view
call history (CDR) with sorting and filtering

No media proxy – RTP flows directly peer‑to‑peer between phones, keeping latency low and CPU usage minimal. That's why it's ideal for LAN‑only deployments (offices, workshops, warehouses, home labs).

Why this exists

pfSense is rock‑solid for routing and firewalling. Kamailio is a world‑class SIP router. But there's no official package to glue them together in a simple, GUI‑driven way—especially not one that fits into the pfSense ecosystem without turning your router into a full‑blown PBX.

This package tries to fill that gap. It's not a replacement for Asterisk/FreeSWITCH—you won't get voicemail, IVRs, conference bridges, or B2BUA features. But if your core need is extension‑to‑extension dialing + sane gateway/trunk routing, this does exactly that with almost zero maintenance.

Quick reality check (read before installing)

ASLR must be disabled – the installer does this automatically, but it's worth knowing.
All of that is documented in SECURITY.ru.md and INSTALL.ru.md inside the repo.

Getting started

Download the package and copy it to your pfSense in the /tmp directory
Connect via SSH to your pfSense and run the installation script with the sh command in the /tmp/offline directory install.sh
Add a firewall rule that allows UDP traffic to port 5060 on your LAN interface.
Configure phones, gateways, trunks, and routing through the web interface (Services → SIP Registar).
Point your SIP phones to the pfSense LAN IP — that's it.

Detailed screenshots and step‑by‑step instructions are in the repo.

Links

GitHub: https://github.com/humaxoid/SipRegistar

License: BSD 2‑Clause

I'm actively maintaining this and welcome bug reports, feature requests, and constructive feedback. If you've been running a small office with a separate PBX and want to simplify your stack, give it a try and let me know how it works for you.

Good luck with the installation.