Asterisk - free solution to computer telephony (including, VoIP ) to the open source code from Digium , originally developed by Mark Spencer . The application works on the operating systems Linux , FreeBSD , OpenBSD , and Solaris . The project name comes from the character "*" ( the English. asterisk - "star").
Asterisk in combination with the necessary equipment has all the features of classical PBX , supports many VoIP -protocols and provides a feature rich call control, including:
Voice mail .
Conference.
Interactive voice response ( IVR ).
Call Center (statement calls in queue and distribution by agents using different algorithms).
Record ( Call Detail Record ).
Asterisk was originally created as the engine for a PBX system (in fact, many refer to it as the Asterisk PBX) and includes all of the components necessary to build a powerful, scalable business phone system. These include advanced features that usually cost extra on a commercial phone system: things like voicemail, automated attendant, call queueing, conference bridging, parking, paging, and intercom calling.
Asterisk is technology and protocol agnostic, which means that you can connect it to the outside world using VoIP or traditional telephony technologies. It also means that you can use virtually any standards-based IP phone: Asterisk includes drivers for SIP and other protocols. That being said, Digium offers a line of IP phones that were specifically designed to compliment Asterisk and take advantage of a number of key productivity features.
Asterisk is future-proof. Unlike traditional phone systems that are generally upgraded using a forklift, Asterisk continues to evolve. Phone systems based on Asterisk see significant improvements each year as new features are included.
To create additional functionality, you can use their own language Asterisk dial plan for writing, writing a module in the language C , or by using the AGI - flexible and versatile interface for integration with external data processing systems. Modules that run through AGI , can be written in any programming language.
Asterisk is distributed under a dual license , by which time the main source, distributed under an open license GNU GPL , you can create private modules containing the licensed code, for example, a module to support the codec G.729 .
Thanks to the free license Asterisk actively developed and supported by thousands of people from all over the world. To get away from the problems created by dual licensing, was created a fork of the project, now called CallWeaver .
History
Mark Spencer, creator of the program, founded Linux Support Services . Spencer wanted to organize a 24-hour service voice support, but an initial budget of $ 4,000 is not allowed to buy extremely expensive systems Call-centers . In 2001, due to the dotcom crisis in Linux Support Services started having problems, and Spencer started to think that the development of software PBX open source can be more interesting than the customer support Linux at all. Jim Dixon of the Zapata Telephony proposed business model for Asterisk . At the same time, and changed the company's name - from Linux Support Services at Digium .
Versions
In Asterisk version numbers adhere to the principle: versions in development - odd, stable - even.
1.0 - released on 23 September 2004.
1.2 - Released November 15, 2005.
1.4 - released December 26, 2006.
1.6 - Released October 2, 2008.
Starting with version 1.6, Asterisk no longer supports Zaptel, leaving only support DAHDI.
1.8 - Released October 21, 2010
Unlike previous versions: Support for SRTP , support for IPv6 in SIP-driver, the integration of the calendar, a new system of logging calls «Channel Event Logging» (CEL), support «Advice of Charge» - service for information about the cost of the call; integration Google Talk and Google Voice ; support the pitch change.
10.0 - Released December 15, 2011
Despite the change of numbering, the tenth version will not be major changes. Adds support for high quality audio, up to 192 kHz , with applications ConfBridge adds support videoconferencing ; Asterisk server can now be text messaging protocols SIP and XMPP , supporting the work of the gateway for the transmission of facsimile communications protocol T.38 ; codec support SILK and CELT.
11 - released on October 31, 2012
Added support for WebRTC , which allows you to make calls directly from your browser , without using any plug-ins in the browser , the new driver supports the chan_motif Google Talk and Jingle, including video, enhanced support IPv6.
Features
The Asterisk software includes many features available in proprietary PBX systems: voice mail, conference calling, interactive voice response (phone menus), and automatic call distribution. Users can create new functionality by writing dial plan scripts in several of Asterisk's own extensions languages, by adding custom loadable modules written in C, or by implementing Asterisk Gateway Interface (AGI) programs using any programming language capable of communicating via the standard streams system (stdin and stdout) or by network TCP sockets.
Special hardware must be installed in Asterisk servers to attach
traditional analog telephones, or to connect to PSTN lines. Digium and a
number of other firms sell PCI cards to attach telephones, telephone lines, T1 and E1 lines, and other analog and digital phone services to a server.
Asterisk supports a wide range of video and Voice over IP protocols, including the Session Initiation Protocol (SIP), the Media Gateway Control Protocol (MGCP), and H.323. Asterisk can interoperate with most SIP telephones, acting both as registrar and as a gateway between IP phones and the PSTN. The Inter-Asterisk eXchange (IAX2), a native protocol in Asterisk provides efficient trunking of
calls among Asterisk PBXes, in addition to distributed configuration
logic, and call completion to VoIP service providers who support it.
Some telephones support the IAX2 protocol directly (see Comparison of VoIP software for examples).
By supporting a mix of traditional and VoIP telephony services, Asterisk
allows deployers to build new telephone systems, or gradually migrate
existing systems to new technologies. Some sites are using Asterisk
servers to replace proprietary PBXes; others to provide additional
features (such as voice mail or voice response menus, or virtual call shops) or to reduce costs by carrying long-distance calls over the Internet (toll bypass).
Asterisk was one of the first open source PBX software packages.
In addition to VoIP protocols, Asterisk supports many traditional
circuit-switching protocols such as ISDN and SS7. This requires
appropriate hardware interface cards supporting such protocols, marketed
by third-party vendors. Each protocol requires the installation of
software modules such as Zaptel, Libpri, Libss7, chanss7, wanpipe and
others. With these features, Asterisk provides a wide spectrum of
communications options.
Equipment
Asterisk can work with analog lines ( FXO / FXS modules) and digital ( ISDN , BRI and PRI - flows T1 / E1 ). With certain computer motherboards (the most well-known manufacturers which are Digium , Sangoma , OpenVox , Rhino , AudioCodes ) Asterisk can connect to a high-capacity lines, T1 / E1 , which can work in parallel with tens and hundreds of telephone connections. A complete list of supported hardware for connection to the public telephone network is determined by hardware support in kernel modules, for example:
DAHDI , acronym «Digium Asterisk Hardware Device Interface» (formerly known as Zaptel ), is being developed in parallel with Asterisk by Digium.
mISDN, developed by Karsten Keil from the team SuSE and company Beronet .
CAPI .
Protocols
It supports the following protocols:
SIP ,
H.323 ,
IAX2 ,
MGCP ,
Skinny / SCCP ,
XMPP ( Google Talk ),
Unistim ,
Skype , a commercial channel.
May transmit text and video signals (for example, use the videophone ). In addition, the realized work with other computer protocols:
DUNDi - protocol, also developed by Digium .
OSP .
T.38 , supports faxing.
Supports a wide range of computer equipment and protocols can host a huge number of scenarios of interaction networks, obtaining and processing information.
Programming
Configuring and programming produced by several mechanisms:
dialplan , which is written in a special language. Available as an older version, and the new - AEL , and the language Lua .
AGI .
AMI .
The configuration of the databases.
Expansion of their functions as possible by writing in C language of the new module, which is possible thanks to the detailed Doxygen -documentation.
Specialized distributions
For ease of installation and use, there are several predefined distributions that include the operating system, the compiled Asterisk, the modules and the standard configuration.
AsteriskNow - distribution from the company Digium includes 2 web interface options to choose from: Asterisk GUI and FreePBX.
AstLinux.
AstPbx - Russian distro preconfigured with rich functionality with a focus on . conf -files.
Elastix - distribution from the company PaloSanto Solutions OpenSource implementation of the ideology of unified communications with localized WEB-interface.
FreePBX - a web-based interface for configuring Asterisk.
PBX in a Flash .
PoundKey .
Starfish PBX - has not been updated since 2009.
Switchvox - a product of Digium.
Thirdlane PBX .
Trixbox , formerly Asterisk @ Home, is based on CentOS .
VirtualPBX - solution for hosted virtual PBX and IVR, feature-rich Voice 2.0
No comments:
Post a Comment