Asif RiazBusiness phone systems are essential for many organizations, but finding a solution that offers...
Business phone systems are essential for many organizations, but finding a solution that offers flexibility, control, and accessible source code can be challenging.
I'd like to introduce OpenWebPBX, an open-source, self-hosted PBX project built on FreeSWITCH and FusionPBX technologies.
I'm helping introduce this project to the developer community with the project's permission.
OpenWebPBX is a multi-tenant business phone system with a web-based administration interface inspired by the Windows 11 desktop experience.
The project aims to make everyday PBX administration easier while keeping the software self-hosted and its source code publicly available.
OpenWebPBX includes:
Version 1.0.16 is available for the Debian 13 server and Android app.
This release includes Meet audio/video conferencing, voicemail access through extension 121, and optional native video calling between colleagues on Android.
The project also provides installation documentation, release downloads, and feature-coverage reports.
Some areas still require additional real-world verification, including physical phone behavior, carrier cutovers, certain network conditions, and broader device compatibility. Full feature parity with 3CX remains a roadmap goal rather than a completed claim.
The project states that no OpenWeb software license fee is charged.
However, self-hosting infrastructure, SIP carriers, and optional third-party services may have separate costs.
The project retains the applicable licenses and notices of its underlying components. Its native Android client is licensed under AGPL-3.0-or-later.
You can explore the project here:
Official website: https://openwebpbx.com/
GitHub repository: https://github.com/embire2/openwebpbx
Version 1.0.16 downloads: https://github.com/embire2/openwebpbx/releases/tag/v1.0.16
If you're considering a deployment, review the installation guide and feature-coverage documentation, and test your own SIP provider and phones before moving production calls.
I'd be interested to hear from developers, system administrators, and VoIP enthusiasts:
Feel free to explore the repository and share constructive feedback.
Thanks for reading!