A SIP engine, not a new language
Keep SIP channels, peer configuration vocabulary and existing integration surfaces. Move to chan_sofia with a documented migration checklist.
An open-source fork of Asterisk with a modern Sofia-SIP channel. Connect phones, browsers and mobile apps — while keeping familiar dialplans, AMI and realtime integration.
Use desk phones, trunks and softphones with the SIP channel, dialplan and integrations you already know.
Opt-in WebRTC audio and video, bridged to SIP. Enable BUNDLE for max-bundle browsers and negotiate a shared video codec. Video passes through; it is not transcoded.
Integrate APNs and FCM wake-up for compatible mobile apps. Enable push and supply your app credentials, token store and client integration.
From a phone on a desk to a browser on the move. Explore what the public project provides, what is optional and what your deployment needs.
Keep SIP channels, peer configuration vocabulary and existing integration surfaces. Move to chan_sofia with a documented migration checklist.
Connect SIP and WebRTC with secure media options, ICE-lite and video passthrough. Requirements and feature switches are explicit.
APNs/FCM integration, multi-device ringing and in-process HTTP/2 push delivery. Your app, credentials and provisioning complete the setup.
Build IVRs, queues, voicemail, conferencing and recording with the modules in the source tree. Select the modules and libraries your system needs.
AMI, AGI/FastAGI, PostgreSQL and other realtime engines, CDR and CEL. A modular foundation for your own telephony applications.
Inspect peers, contacts, registrations and call history. Trace SIP, SDP, ICE and DTMF, or export SIP capture to a HEP collector.
A focused release for network mobility, multi-device calls and registration behaviour. Read the changes and the conditions behind them.
Read the release notes →1.8.2 accepts authenticated ICE re-nominations when a WebRTC client changes network, and keeps the selected media pair separate from routine consent checks.
Cancelled sibling branches receive an answered-elsewhere Reason header. Compatible phones can distinguish an answered call from a missed one.
The useragent field becomes a comma-separated prefix allowlist. An empty value imposes no User-Agent restriction, avoiding the old first-registration lock.
chan_sofia preserves the SIP channel type and familiar dialplan, CLI, AMI and realtime interfaces. Review the configuration and test the call paths before a planned cutover.
; modules.conf [modules]
noload => chan_sip.so
load => chan_sofia.soThis is the driver switch, not the whole migration. Configure sofia.conf and the required modules first. The two SIP drivers cannot be loaded together; switching drivers requires a planned restart.
GABPBX is a GPLv2 PBX and telephony toolkit maintained by Germán Luis Aracil Boned. It is a fork of Asterisk, with the original authors’ notices retained. There is no hosted service or web administration panel bundled into this public engine.
This site describes the public repository, not private extensions or a commercial deployment.