FreeSWITCH!

Wazo is mainly an IPBX that provides communication services such audio, video, and instant messaging based on the use of standard network technologies such as the TCP/IP protocol. It is interoperable with most market telephony systems and gives users access to flexible and advanced services without requiring specialized hardware.

3CXPhone is a free VoIP phone available for Windows, iPhone or any Android based smart phone such as Google Nexus, Sony Xperia, Motorola Droid or Samsung Galaxy. Connect 3CXPhone to a VoIP provider or to a VoIP PBX (for example 3CX Phone System) to make calls to any VoIP, mobile or landline number.

Zent.IO is a Contact Center Cloud Software Multichannel that unify chat, phone and e-mail. All the communication channels will be answered from a single interface. The agent will only need a web browser

Elastix is an open source Unified Communications Server software that brings together IP PBX, email, IM, faxing and collaboration functionality. It has a web interface and includes capabilities such as a Call Center software with predictive dialing.

XiVO is a scalable platform based on Asterisk that offers a free and complete PBX solution. It is licensed under the GPLv3.

uSipSwitch a micro sip switch (sip proxy) perfect for edge SIP SBC work (e.g. hosted PBX business). Comes with web console and DNS health tools. Uses modern clustering NoSQL technology for advanced HA.

OpenSIPS (Open SIP Server) is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. OpenSIPS, as a SIP server, is the core component of any SIP-based VoIP solution. With a very flexible and customizable routing engine, OpenSIPS unifies voice, video, IM and presence services in a highly efficient way, thanks to its scalable (modular) design.What OpenSIPS has to offer, comes in a reliable and high-performance flavour - OpenSIPS is one of the fastest SIP servers, with a throughput that confirms it as a solution up to enterprise or carrier-grade class. »

Starfish PBX aims to be an open source alternative to Digium’s Switchvox. The goal is to have a software equivalent to which we can add additional functionality like:

Kamailio® (successor of former OpenSER and SER) is an Open Source SIP Server released under GPL, able to handle thousands of call setups per second. Among features: asynchronous TCP, UDP and SCTP, secure communication via TLS for VoIP (voice, video); WebSocket support for WebRTC; IPv4 and IPv6; SIMPLE instant messaging and presence with embedded XCAP server and MSRP relay; asynchronous operations; IMS extensions; ENUM; DID and least cost routing; load balancing; routing fail-over; accounting, authentication and authorization; support for many backend systems such as MySQL, Postgres, Oracle, Radius, LDAP, Redis, Cassandra, MongoDB, Memcached; XMLRPC control interface, SNMP monitoring. It can be used to build large VoIP servicing platforms or to scale up SIP-to-PSTN gateways, PBX systems or media servers like Asterisk™, FreeSWITCH™ or SEMS. Kamailio and the SIP Express Router (SER) teamed up for the integration of the two applications and new development. »

AsteriskNOW is the fastest way to get started building custom telephony solutions with Asterisk. Simply download the .iso file, burn it to a CD, drop it into the CD or DVD drive on the target computer and in less than 30 minutes you will have a full functional Asterisk system ready for your custom telephony application.

VoipSwitch is a SoftSwitch System with routing and billing, with SIP protocol support (in older versions also H323).

PBX in a Flash is a PBX system based on Asterisk, FreePBX and CentOS, coming as an ISO download. A two stage installation process allows for more flexibility for the installation of the PBX specific components. The Asterisk components are compiled from source during the install process which allows for easier adaptions later on. The system is update-able without re-install. A script collection on pbxinaflash.org allows for easy extension of the base system.

Yate is a softswitch (concept of separating the network hardware from network software).

Flexisip is a SIP proxy server implementation compliant to RFC3261, written in C++ 11.