SIP

TrixBox

Posted by PrabhasPokharel on Aug 21, 2009
TrixBox data sheet 4008 Views
Organization that developed the Tool: 
Main Contact: 
Andrew Gillis
Problem or Need: 

Asterisk is a powerful PBX system, but requires a lot of effort to just install. There was a need for something that was easier to install and administer, thus the creation of TrixBox.

Main Contact Email : 
Brief Description: 

TrixBox offers two options, TrixBox CE and TrixBox Pro. TrixBox CE is an open telephony platform that combines the best of the open source telephony tools into one easy-to-install package. Based on an enhanced LAAMP (an open source bundle of Linux®, Apache™, Asterisk®, mySQL®, and PHP), the TrixBox dashboard provides easy to use, Web-based interfaces to setup, manage, maintain, and support a complete IP-PBX system. TrixBox Pro is an enhanced version that comes with more support than TrixBox CE.

Tool Category: 
App resides and runs on a server
Key Features : 

Unlimited Extensions, Voicemail and fax Support, VoiceMail to email and web, IVR Menu System, Ring Groups, Call Queues, Conferencing, Time-Based Routing, Music On Hold, Paging and Intercom, Admin Status Screen, Package Manager for easy updates, Network Settings and Phone Provisioning Tool, Opern Source Echo Cancellation.

Main Services: 
Interactive Voice Response (IVR)
Tool Maturity: 
Currently deployed
Platforms: 
Linux/UNIX
Current Version: 
2.4
Program/Code Language: 
C/C++
Support Forums: 
http://www.trixbox.com/support-and-training/community
http://help.trixbox.com/
Languages supported: 
English
Is the Tool's Code Available?: 
No
Is an API available to interface with your tool?: 
Yes
Global Regions: 
Featured?: 
Yes

FreeSWITCH

Posted by PrabhasPokharel on Aug 21, 2009
FreeSWITCH data sheet 3989 Views
Organization that developed the Tool: 
Main Contact: 
Anthony Minessale II
Problem or Need: 

Open Source tool needed for managing a softswitch that is modular, and uses simple scripts to manage workflow.

Main Contact Email : 
Brief Description: 

FreeSWITCH is an open source telephony platform designed to facilitate the creation of voice and chat driven products. 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.

Tool Category: 
App resides and runs on a server
Key Features : 

FreeSWITCH includes many modules which provide applications by default including conferencing, XML-RPC control of live calls, Interactive voice response (IVR), TTS/ASR (text to speech/automatic speech recognition), Public switched telephone network (PSTN) interconnection ability supporting both analogue and digital circuits, Voice over IP protocols including SIP, Inter-Asterisk eXchange, H.323, Jabber, GoogleTalk and others. Applications using the FreeSWITCH library can be written in C/C++, Python, Perl, Lua, JavaScript using Mozilla's SpiderMonkey engine, Java and Microsoft .NET via Microsoft's CLR or via Mono. FreeSWITCH is designed to be modular, easy to use with scripting done entirely in XML, and more stable than Asterisk.

Main Services: 
Interactive Voice Response (IVR)
Other
Tool Maturity: 
Currently deployed
Release Date: 
2008-05
Platforms: 
Linux/UNIX
Windows
Current Version: 
1
Program/Code Language: 
C/C++
Other
Support Forums: 
http://wiki.freeswitch.org
Is the Tool's Code Available?: 
Yes
URL for license: 
http://www.mozilla.org/MPL/MPL-1.1.html
Is an API available to interface with your tool?: 
Yes
Featured?: 
Yes

Asterisk

Posted by PrabhasPokharel on Aug 21, 2009
Asterisk data sheet 1955 Views
Organization that developed the Tool: 
Main Contact: 
Mark Spencer
Problem or Need: 

Open-source system needed for managing a telephone PBX system.

Main Contact Email : 
Brief Description: 

Asterisk is a software implementation of a telephone private branch exchange (PBX). Like any PBX, it allows attached telephones to make calls to one another, and to connect to other telephone services including the public switched telephone network (PSTN) and Voice over Internet Protocol (VoIP) services. See the wikipedia article for more.

Tool Category: 
App resides and runs on a server
Key Features : 

Voice mail, conference calling, interactive voice response (phone menus), and automatic call distribution. With hardware, can attach traditional analogue telephones to an Asterisk installation. Direct support of VoIP protocols, including SIP, MGCP and H.323. Large Userbase, and a large collection of proprietary and free add-ons and features.

Main Services: 
Interactive Voice Response (IVR)
Other
Tool Maturity: 
Currently deployed
Release Date: 
2004-09
Platforms: 
Linux/UNIX
Windows
Other
Current Version: 
1.61008
Program/Code Language: 
C/C++
Support Forums: 
http://forums.digium.com/
http://www.asterisk.org/community
http://asterisktutorials.com/
Is the Tool's Code Available?: 
Yes
URL for license: 
http://en.wikipedia.org/wiki/GNU_General_Public_License
Is an API available to interface with your tool?: 
Yes
Featured?: 
Yes