[needs-packaging] sipXecs - Communications Suite, IP PBX server, VoIP
Affects | Status | Importance | Assigned to | Milestone | |
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Feisty Backports |
Invalid
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Undecided
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Unassigned | ||
Ubuntu |
Confirmed
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Wishlist
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Unassigned | ||
Bug Description
URL:
http://
http://
Description:
Our vision is quite big. But think about it. If the Internet was made up of thousands or millions of interoperable SIP proxy servers the way the Internet is filled up with interoperable email servers, we could truly communicate free and unencumbered. This is about technology of course, but first and foremost this is about providing a solution that demonstrates these capabilities while being extremely easy to use, robust and scalable. This is exactly what sipXecs was designed to do. Provide a showcase to the industry that a standards based solution can offer all the features, be easier to use than most commercial solutions, and provide the robustness and scalability for the harsh enterprise environment. Voice is a mission critical application after all.
Many thousand deployments of sipXecs around the world demonstrate that our vision is getting real. sipXecs has evolved to now being a serious competitor when companies evaluate buying a new phone system. And sipXecs does not just compete on price. The main metric is features and stability. With its load-sharing redundancy sipXecs has convinced many large companies that a standards base solution is simply the better choice.
Works with softphones (VoIP) including ;
KDE kphone, Xten X-Lite & eyeBeam Phones, SNOM 360 Softphone, Windows Messenger, LinPhone, SJPhone, sipXphone, Ekiga Phone (former GnomeMeeting), OpenWengo.
Key capabilities of sipXecs 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: sipXecs installs in hours, 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.
* All features included: For the 3.10 release alone we added more than 170 new features.
* 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 designed 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 an additional 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 dial plans 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 for all configuration. sipXecs includes integration with Microsoft AD and Exchange.
Works well alongside "FreeSwitch"
License: L-GPL
see http://
Notes:
How to install on debian, Ubuntu and platforms including AMD64
http://
Setting Up
http://
Changed in feisty-backports: | |
status: | Incomplete → Invalid |
Changed in gutsy-backports: | |
status: | Incomplete → Invalid |
Not packaged, so too early to request a backport.