ทูลฟรีสำหรับทำตู้สาขาโทรศัพท์ในสำนักงานแบบผ่านไอพี (Open Source IP-PBX)

Open Source IP-PBX

ทูลฟรีสำหรับทำตู้สาขาโทรศัพท์ในสำนักงานแบบผ่านไอพี (IP-PBX), VOIP, IP Telephony


- Asterisk (http://www.asterisk.org) นิยม
Asterisk is an open source framework for building communications applications. Asterisk turns an ordinary computer into a communications server. Asterisk powers IP PBX systems, VoIP gateways, conference servers and other custom solutions. It is used by small businesses, large businesses, call centers, carriers and government agencies, worldwide. Asterisk is free and open source.
Asterisk is released under a dual license model, using the GNU General Public License (GPL) as a free software license and a proprietary software license to permit licensees to distribute proprietary, unpublished system components.


- CallButler (http://callbutler.codeplex.com)
CallButler is a FREE Windows-based open source PBX, IVR and Auto-Attendant Phone System built on .NET. It was a commercially available product, but is now being put into Open Source.


- Elastix (http://www.elastix.org/index.php/en/)
This is an open source Unified Communications Server software that is offered with IP PBX, email, faxing and collaboration functionality. The Web interface includes capabilities such as Call Center software with predictive dialing. Elastix integrates several software packages each including their own set of features. Elastix adds new interface for controlling and reporting. Elsastix offers Call recording, Voicemail and Voicemail-to-Email functionality, flexible IVR configurable by web interface, voice synthesis support, extension batch tool to generate large number of extensions using CSV files, integrated Echo Canceller. Phone provisioner, configurable via a web interface. It allows configuration of a large number of IP phones in a short time for supported phones. It features a web interface and includes capabilities such as call center software with predictive dialing. The Elastix functionality is based on open source projects


- FreeSWITCH (http://www.freeswitch.org)
FreeSWITCH is a scalable open source cross-platform telephony platform designed to route and interconnect popular communication protocols using audio, video, text or any other form of media. It was created in 2006 to fill the void left by proprietary commercial solutions. FreeSWITCH also provides a stable telephony platform on which many telephony applications can be developed using a wide range of free tools.


- FreePBX (http://www.freepbx.org)
FreePBX is an open source GUI (graphical user interface) that controls and manages Asterisk (PBX), FreePBX is licensed under GPL. FreePBX is a Registered Trademark of Schmooze Com Inc.


- Kamailio (http://www.kamailio.org/w/)
Kamailio used to be called OpenSER and is best known for being the “high-end” open source PABX. As its old name implies, Kamailio is predominantly used as a SIP router, but the server itself supports a number of features like instant messaging and presence. In terms of scalability, Kamailio claims to be able to handle some 5000 call setups per second and its least-cost routing can scale to handle millions of routing rules. Failover and redundancy is also included. Kamailio also supports authentication to multiple databases and extensions (about 80 are available) can be written in Perl. There is also a Java API which can be used to extend VoIP services and integrate with Web services.


- sipX (http://www.sipfoundry.org)
sipX is an open source voice over IP PBX software. The main features of sipX is software implementation of the Session Initiation Protocol(SIP) that makes an IP based communication system (IP PBX). The design of sipX deviates from the traditional Asterisk. sipX includes features such as voice mail, interactive voice response systems, auto attendants and the like. It looks like Asterisk isn’t the only open source PBX game in town anymore. sipX, as the name implies, is a SIP-only PBX project released under the LGPL. A noteworthy feature is the inclusion of an out-of-the-box


- Yate - Yet Another Telephony Engine (http://yate.null.ro/pmwiki/)
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 messenging can all be unified under Yate's flexible routing engine, maximizing communications efficiency and minimizing infrastructure costs for businesses.

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รวบรวมโดย
Aj.Arnut Ruttanatirakul
(c) SysAdmin Knowledge
http://www.sysadmin.in.th
March 11, 2014

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