Elastix 
Elastix is an appliance software that integrates the best tools available for Asterisk-based PBXs into a single, easy-to-use interface. It also adds its own set of utilities and allows for the creation of third party modules to make it the best software package available for open source telephony. For more information, please visit www.elastix.org
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Trixbox CE
trixbox is a business class IP PBX system based on Digum's Asterisk Open Source PBX Software. trixbox CE is the original free, fully featured, open source PBX application platform system. Beginning in 2004 as Asterisk@Home, the trixbox ® Community Edition (CE) telephony application platform is the open source software that has quickly become the most popular Asterisk ® -based distribution in the world. trixbox CE combines the best of the open source telephony tools into one easy-to-install package, along with the trixbox dashboard which provides a web-based interface to configure and manage a complete IP-PBX system. The most flexible and customizable communications platform available, trixbox CE averages over 65,000 downloads a month.
For more information, please visit:
http://www.trixbox.com/products/trixbox-ce/
http://click4pbx.com/trixbox/index.html
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Endpoint manager |
Extensions |
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IVR |
Queues |
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DID's |
Trunks |
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Music on Hold |
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AsteriskNow
Asterisk® in minutes. AsteriskNOW is an open source Software Appliance; a customized Linux distribution that includes Asterisk (the leading open source telephony engine and tool kit), the AsteriskGUI™, and all other software needed for an Asterisk system. AsteriskNOW is easy to install, and offers flexibility, functionality and features not available in advanced, high-cost proprietary business systems.
For more information, please visit http://www.asterisknow.org
A2billing
The A2Billing Platform has been deployed in a number of commercial environments by both traditional TDM based telecoms companies wishing to move into the VoIP market, and calling card and call-shop businesses. Additionally, there has been a lot of interest from IT and networking companies who are beginning to deploy VoIP PBXs in addition to their traditional business, and wish to enjoy an ongoing income by terminating their customer's calls using A2Billing as their Wholesale Billing Platform.
For more informations, please visit http://click4pbx.com/a2billing.html
PBXware
PBXware is the World's First and most mature Open Standards Turnkey Telephony Platform. Since 2003 PBXware has deployed flexible, reliable and scalable New Generation Communication Systems to SMBs, Enterprises and Governments worldwide by unifying the most Advanced of Latest Technologies.
PBXware supports a wide range of PSTN and VoIP technologies. Creation of Enhanced Voicemail, ACD Queues, IVR Auto Attendants, Conference Bridges, Music on Hold, Least Cost Routing, national/global voice networks, and much more... all deployable as a single unit or redundant network. A real cost saving solution!!!
For more information, please visit http://click4pbx.com/pbxware/index.html
Opensips
OpenSIPS ( Open SIP S erver) 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 .
For more information, please visit http://www.opensips.org
OpenSIPS Control Panel (OpenSIPS-CP) is a web interface (PHP based) for provisioning your OpenSIPS system. OpenSIPS-CP is not designed for provisioning users, but for provisioning the operational side of OpenSIPS (like DB, MI operations).
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Openser/Kamailio
KAMAILIO (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.
For more information, please visit http://www.kamailio.org
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Vicidial
VICIDIAL is a set of programs that are designed to interact with the Asterisk Open-Source PBX Phone system to act as a complete inbound/outbound call center suite.
The agent interface is an interactive set of web pages that work through a web browser to give real-time information and functionality with nothing more than an internet browser on the client computer.
The manager interface is also web-based and offers the ability to view many real-time and summary reports as well as many detailed campaign and agent options and settings.
For more information, please visit http://astguiclient.sourceforge.net/
Freeswitch
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 PBX, 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.
FreeSWITCH supports many advanced SIP features such as presence/BLF/SLA as well as TCP TLS and sRTP. It also can be used as a transparent proxy with and without media in the path to act as a SBC (session border controller) and proxy T.38 and other end to end protocols.
For more information, please visit http://www.freeswitch.org
sipXecs
sipXecs Enterprise Communications System (ECS) is a highly robust and scalable, enterprise-grade, and fully featured unified communications solution with integrated voicemail, unified messaging, presence, call center, multiple auto attendants, paging and intercom services and a powerful plug & play web-based configuration and management system. sipXecs is based entirely on the Session Initiation Protocol (SIP) standard and operates on standard Linux operating system. It interoperates with a large number of third party phones, PSTN/IP gateways and applications such as Microsoft Exchange 2007 without compromising ease of use.
For more information, please visit http://www.sipfoundry.org |
