August 11, 2007

Opensource SoftSwitch


Asterisk is the most popular and extensible open source telephone system in the world, offering flexibility, functionality and features not available in advanced, high cost proprietary business systems. Asterisk is a complete IP telephony platform for business, and can be downloaded for free.


FreeSWITCH is an open source telephony platform designed to facilitate the creation of voice and chat driven products scaling from a soft-phone up to a soft-switch. It can be used as a simple switching engine, a media gateway or a media server to host IVR applications using simple scripts or XML to control the callflow.

We support various communication technologies such as SIP, H.323, IAX2 and GoogleTalk making it easy to interface with other open source PBX systems such as sipX, OpenPBX, Bayonne, YATE or Asterisk.


yate is a next-generation telephony engine; while currently focused on Voice over Internet Protocol (VoIP) and PSTN, its power lies in its ability to be easily extended. Voice, video, data and instant messaging can all be unified under Yate's flexible routing engine, maximizing communications efficiency and minimizing infrastructure costs for businesses.

The software is written in C++ and it supports scripting in various programming languages (such as those supported by the currently implemented embedded PHP, Python and Perl interpreters) and even any Unix shell. The PHP, Python and Perl libraries have been developed and made available in order to ease development of external functionalities for Yate.

CallWeaver (OpenPBX)

CallWeaver is a community-driven vendor-independent cross-platform open source PBX software project (formerly known as It was originally derived from Asterisk. Now it supports analog and digital PSTN telephony, multi-protocol voice over IP telephony, fax, software-fax, T.38 fax over IP and many telephony applications such as IVR, conferencing and callcenter queue management.


FreePBX is a Standardised Implementation of Asterisk that gives you a GUI to manage your system. If you've looked into Asterisk, you'd know that it doesn't come with any 'built in' programming. You can't plug a phone into it and make it work without editing configuration files, writing dialplans, and various messing about. FreePBX simplifies this by giving you a pre-written set of dialplans that allow you to have a fully functional PBX pretty much straight away.


sipX is the leading open source IP PBX in terms of scalability, robustness and ease of use. The sipX IP PBX has been successfully deployed in a lot of places. The largest known installation serves more than 5,000 users connected to one redundant (HA) system. Small installations go all the way down to a few users served by very low cost hardware.

OpenSER - open source SIP Server

OpenSER is a mature and flexible open source SIP server (RFC3261). It can be used on systems with limitted resources as well as on carrier grade servers, scaling to up to thousands call setups per second. It is written in pure C for Unix/Linux-like systems with architecture specific optimizations to offer high performances. It is customizable, being able to feature as fast load balancer; SIP server flavours: registrar, location server, proxy server, redirect server; gateway to SMS/XMPP; or advanced VoIP application server.

Asterisk Distro (Box/LiveCD)
Trixbox (Formerly Asterisk@Home - base on CentOS)
sipX Live CD (base on Fedora)

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