Self-Hosted SIP PBX Systems
Open-source telephony platforms for managing voice communications and internal phone extensions within small office environments.
Self-Hosted SIP PBX Systems
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- signalwire/freeswitch
signalwire/freeswitch
4,637View on GitHubFreeSWITCH is an open-source software telephony switch that routes and processes voice, video, and messaging calls over IP networks using standard protocols such as SIP and RTP. It serves as both a call routing and switching engine and a programmable telephony API, enabling developers to build custom telephony applications with event-driven call control and media manipulation. As a SIP media server, it handles media streaming, transcoding, and conferencing, and functions as a unified communications platform integrating voice, video, messaging, and conferencing into a single switchable infrastructure. The platform is built on a modular architecture with an asynchronous event bus that distributes all signalling, state, and media events to subscribed modules and scripts. An embedded scripting runtime supports Lua and JavaScript for IVR and call control logic within the same process, eliminating the need for external CGI or IPC. The Sofia-SIP library handles low-level SIP parsing and registration, while a state machine driven call control model manages each call leg, emitting events at every transition. Codec transparency and an in-memory media bypass engine optimize media handling by redirecting streams between compatible endpoints. FreeSWITCH enables deployment of a telephony switch on standard hardware to manage voice calls and media streaming, replacing proprietary telecom switches. Capabilities include configuring call routing rules for inbound calls, managing SIP endpoints with profiles for registration and media negotiation, and integrating cloud-based voice and messaging services through a unified API. The platform supports cloud telephony integration, inbound call routing, and endpoint configuration, making it extensible for various telephony needs.
FreeSWITCH is a powerful, programmable telephony engine that provides the core SIP, WebRTC, and call routing capabilities required for a PBX, though it functions as a developer-focused toolkit rather than a pre-configured, out-of-the-box business phone system.
CCall Traffic RoutingTelephony SoftswitchesVoIP and Telephony - 2600hz/kazoo
2600hz/kazoo
The core of an open-source, distributed, highly scalable platform designed to provide robust telecom services
Kazoo is a distributed, carrier-grade open-source PBX platform that provides the full suite of required features including SIP, WebRTC, IVR, and multi-extension management for scalable voice and video communications.
ErlangVoIP and Telephony - fusionpbx/fusionpbx
fusionpbx/fusionpbx
1,011View on GitHubOfficial FusionPBX - A full-featured domain based multi-tenant PBX and voice switch for FreeSwitch.
FusionPBX is a comprehensive, multi-tenant PBX platform built on FreeSwitch that provides all the requested features, including SIP support, IVR, voicemail, and PSTN gateway integration for business communication.
PHPVoIP and Telephony - asterisk/asterisk
asterisk/asterisk
3,307View on GitHubThe official Asterisk Project repository.
Asterisk is the industry-standard, open-source framework for building a full-featured PBX, providing comprehensive support for SIP, WebRTC, IVR, call routing, and PSTN connectivity.
CVoIP and Telephony - fonoster/fonoster
fonoster/fonoster
7,997View on GitHubFonoster is a conversational AI framework and multi-tenant communications platform as a service. It serves as a programmable voice gateway and SIP telephony platform, enabling the creation of voice-based assistants and automated communication workflows using large language models. The project distinguishes itself through a vendor-agnostic speech integration engine that abstracts speech-to-text and text-to-speech providers. It features a multi-tenant architecture that isolates telephony resources and user identities into distinct organizational workspaces. The system covers a broad range of telephony capabilities, including SIP trunk configuration, bidirectional audio streaming, and PBX functionality. It provides tools for call flow logic control, real-time call status monitoring, and the programmatic origination of outbound calls. Security is handled through role-based access control, token-based session authentication, and API key management. The communication stack can be deployed on private infrastructure or orchestrated using Docker containers.
Fonoster is a programmable communications platform that provides the core SIP telephony and PBX functionality required for managing voice workflows, though it is designed more as a developer-focused framework for building custom communication services than a traditional out-of-the-box office PBX.
TypeScriptCall Traffic RoutingPSTN IntegrationsPSTN Connectivity - belledonnecommunications/flexisip
BelledonneCommunications/flexisip
Linphone.org mirror for flexisip (git://git.linphone.org/flexisip.git)
Flexisip is a high-performance SIP proxy and server that provides the core infrastructure for managing voice and video communications, though it functions as a SIP server rather than a full-featured PBX with a built-in IVR or management GUI.
C++SIP Servers
