Yate - Yet Another Telephony Engine!

MicroSIP is a free portable SIP softphone for Windows based on PJSIP stack. It allowing to do high quality VoIP calls (P2P or on regular telephones) via open SIP protocol. From cloud of SIP providers you can choose best for you, register account and use it with MicroSIP. You'll get free p2p calls and cheap international calls.

FreePBX is a web-based open source GUI (graphical user interface) that controls and manages Asterisk (PBX), an open source communication server. FreePBX is licensed under the GNU General Public License (GPL), an open source license. FreePBX can be installed manually or as part of the pre-configured FreePBX Distro that includes the system OS, Asterisk, FreePBX GUI and assorted dependencies.

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.

FreeSWITCH is a scalable open source cross-platform telephony platform designed to route and interconnect popular communication protocols using audio, video, text or any other form of media. It was created in 2006 to fill the void left by proprietary commercial solutions. FreeSWITCH also provides a stable telephony platform on which many telephony applications can be developed using a wide range of free tools.

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

FreeSentral is a full IP PBX consisting of a Linux Distribution, an IP PBX and a Web Graphical User Interface for easy configuration

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.

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:

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.

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).