A FreePBX and Asterisk alternative you run from the browser

How they differ

	BitVoice	FreePBX on Asterisk
What you install	One program for Linux, macOS or Windows; SQLite inside, PostgreSQL for several servers	Asterisk, FreePBX, a web server, PHP and a database on Linux, usually from a distribution image
Where it is set up	Every setting in the web interface, checked by a test on every change	The web interface, with Asterisk's configuration files and command line underneath
Licence	Apache 2.0, no paid modules	Open source (GPL), with some modules sold commercially by Sangoma
Engine	BitVoice's own, in Go: SIP, IAX2, RTP, SRTP, WebRTC, codecs	Asterisk, in C
Track record	New: no large production sites yet	Very many sites over many years, a large community and commercial support

What comes over

On Import, upload the exports; BitVoice shows exactly what would change before anything is applied, and the import is a revision you can undo.

• FreePBX: Admin → Bulk Handler → Export → Extensions (CSV)
• Asterisk: pjsip.conf or sip.conf
• Also 3CX, Cisco Unified CM, Avaya IP Office and Communication Manager, and Mitel MiVoice Business
• Voicemail PINs, names and email addresses come over when the export has them; exports carry no SIP passwords, so each extension gets a new one, and trunks arrive switched off until you enter their passwords

The usual FreePBX feature codes work (*72, *78, *8, *97 and the rest) and can be changed. Day/night toggles (FreePBX's call flow control), PIN sets, caller ID lookup and DISA are there. IAX2 trunks join an Asterisk or FreePBX server you keep, so you can move one site or department at a time, and tools written for Asterisk's AMI and ARI can drive BitVoice too.

What BitVoice does not do yet

• SS7 beyond one link per span: no link sets, no signal transfer point, no database lookups.
• A long list of real phones, carriers and cards tried in production: see what has been tested.

The step-by-step guide to the import is in the manual.