sipXecs

sipXecs (Enterprise Communications Server) is an open source voice over IP telephony server. Its main feature is a software implementation of the Session Initiation Protocol (SIP), which makes it an IP based communications system (IP PBX).

sipX includes many features of a traditional private branch exchange (PBX) like voice mail, interactive voice response systems, auto attendants, integrated management and configuration of the PBX and attached phones and gateways, and more.

The software comes with a powerful Web based configuration and management application designed using the latest in Java, Web Services and Ajax technologies.

sipXecs is used by thousands of companies globally.

sipXecs 4.6.0

Price Free to download Size Multiple packages License GNU LGPL Developer SIPfoundry Website www.sipfoundry.org System Requirements SIP phones (soft or hard) SIP trunking or PSTN gateway for trunk lines Support Sites: Wiki, FAQ Selected Reviews:

Features include:

• Fully featured Unified Communications system: sipXecs is a complete IP PBX applications that has all the features you would expect from a unified communications solution
• Ease of installation: A complete sipXecs system with phones and everything installs in hours and not days or weeks, guaranteed. It runs on any standard server without the need for any special hardware
• Easy to use: No need to get expert help. With sipXecs you will be self-sufficient for all adds, moves and changes. Typically a receptionist is capable of managing the system. There are no hidden configuration files or other things that require a specialist.
• Plug & play management of everything: One - two - three clicks and you just configured a new user with a phone. The phones are auto-discovered and as soon as they are connected to the LAN pick up configuration from sipXecs and come up configured. No messing around with phone or gateway configuration ever again.
• SIP Trunking and Remote Workers: sipXecs provides a SIP trunking gateway with an optionally redundant media relay for media anchoring (which is necessary to traverse NAT). Is able to auto-detect and compensate for both far-end and near-end NATs of different types, typically making it unnecessary to buy an additional SBC
• Conferencing: sipXecs provides a voice conferencing server that is based on FreeSWITCH, fully integrated with sipXecs. It offers high performance conferencing with support for HD Voice. Every user on the system gets a personal conference bridge with dynamic Web based user controls
• Redundancy and scalability: sipXecs is unique in that it offers full load-sharing redundancy for the call control system. A server failure will not cause calls to be interrupted. sipXecs is architected as a distributed system. It scales by simply adding hardware. Need more capacity for your call center ACD application; Run it on separate hardware or add a second ACD server
• Trunk redundancy and failover: sipXecs uses external gateways for a reason. External gateways offer flexible deployment options including trunk failover and redundancy. Gateways can be deployed anywhere on the network including in different locations. You can add as many trunk lines you need not limited by how many PCI cards fit into a server chassis. Media processing does not load your CPU and media is routed peer-to-peer from the phone directly to the gateway. Gateways are plug & play managed and easy to deploy
• Interoperability: sipXecs is a truly SIP standards compliant system using native SIP call control. It is a SIP router that interoperates in a large network and routes calls. Many of the sipXecs developers actively participate in the IETF effort to standardize SIP and have authored or co-authored many of the standards
• Localization: sipXecs can be easily localized using uploadable language packs. Language packs include voice prompts, full UI translation, local dialplans and region specific call progress tone settings
• Better voice quality: sipXecs routes media peer-to-peer and not through the sipXecs server. This has many key advantages among them better voice quality, unlimited number of simultaneous calls, unlimited video calls, works with any codec supported by the end-points, and the PBX is no a single point of failure which allows load-sharing redundancy
• Web Services, SOA, and IT integration: sipXecs offers many interfaces. It is designed to integrate into an advanced IT environment including Web Services based on SOAP or REST for all configuration. sipXecs includes integration with Microsoft AD and Exchange

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Last Updated Tuesday, April 09 2013 @ 02:05 PM EDT