Asterisk v1.4.19 - The Open Source Linux PBX
What Is Asterisk?
Asterisk is a complete PBX in software. It runs on Linux and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.
Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway). Check the Features section for a more complete list.
Asterisk needs no additional hardware for Voice over IP. For interconnection with digital and analog telephony equipment, Asterisk supports a number of hardware devices, most notably all of the hardware manufactured by Asterisk's sponsors, Digium™. Digium has single and quad span T1 and E1 interfaces for interconnection to PRI lines and channel banks as well as a single port FXO card and a one to four-port modular FXS and FXO card.
Also supported are the Internet Line Jack and Internet Phone Jack products from Quicknet.
Asterisk supports a wide range of TDM protocols for the handling and transmission of voice over traditional telephony interfaces. Asterisk supports US and European standard signalling types used in standard business phone systems, allowing it to bridge between next generation voice-data integrated networks and existing infrastructure. Asterisk not only supports traditional phone equipment, it enhances them with additional capabilities.
Using the Inter-Asterisk eXchange (IAX™) Voice over IP protocol, Asterisk merges voice and data traffic seamlessly across disparate networks. While using Packet Voice, it is possible to send data such as URL information and images in-line with voice traffic, allowing advanced integration of information.
Asterisk provides a central switching core, with four APIs for modular loading of telephony applications, hardware interfaces, file format handling, and codecs. It allows for transparent switching between all supported interfaces, allowing it to tie together a diverse mixture of telephony systems into a single switching network.
Asterisk is primarily developed on GNU/Linux for x/86. It is known to compile and run on GNU/Linux for PPC along with OpenBSD, FreeBSD, and Mac OS X Jaguar. Other platforms and standards-based UNIX-like operating systems should be reasonably easy to port for anyone with the time and requisite skill to do so. Asterisk is available in the testing and unstable Debian archives, maintained thanks to Mark Purcell.
Who Made This?
Asterisk was originally written by Mark Spencer of Digium, Inc. Code has been contributed from open source coders around the world, and testing and bug-patches from the community have provided invaluable aid to the development of this software.
Asterisk Architecture
Asterisk is carefully designed for maximum flexibility. Specific APIs are defined around a central PBX core system. This advanced core handles the internal interconnection of the PBX, cleanly abstracted from the specific protocols, codecs, and hardware interfaces from the telephony applications. This allows Asterisk to use any suitable hardware and technology available now or in the future to perform its essential functions, connecting hardware and applications.
The Asterisk core handles these items internally:
What Is Asterisk?
Asterisk is a complete PBX in software. It runs on Linux and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.
Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway). Check the Features section for a more complete list.
Asterisk needs no additional hardware for Voice over IP. For interconnection with digital and analog telephony equipment, Asterisk supports a number of hardware devices, most notably all of the hardware manufactured by Asterisk's sponsors, Digium™. Digium has single and quad span T1 and E1 interfaces for interconnection to PRI lines and channel banks as well as a single port FXO card and a one to four-port modular FXS and FXO card.
Also supported are the Internet Line Jack and Internet Phone Jack products from Quicknet.
Asterisk supports a wide range of TDM protocols for the handling and transmission of voice over traditional telephony interfaces. Asterisk supports US and European standard signalling types used in standard business phone systems, allowing it to bridge between next generation voice-data integrated networks and existing infrastructure. Asterisk not only supports traditional phone equipment, it enhances them with additional capabilities.
Using the Inter-Asterisk eXchange (IAX™) Voice over IP protocol, Asterisk merges voice and data traffic seamlessly across disparate networks. While using Packet Voice, it is possible to send data such as URL information and images in-line with voice traffic, allowing advanced integration of information.
Asterisk provides a central switching core, with four APIs for modular loading of telephony applications, hardware interfaces, file format handling, and codecs. It allows for transparent switching between all supported interfaces, allowing it to tie together a diverse mixture of telephony systems into a single switching network.
Asterisk is primarily developed on GNU/Linux for x/86. It is known to compile and run on GNU/Linux for PPC along with OpenBSD, FreeBSD, and Mac OS X Jaguar. Other platforms and standards-based UNIX-like operating systems should be reasonably easy to port for anyone with the time and requisite skill to do so. Asterisk is available in the testing and unstable Debian archives, maintained thanks to Mark Purcell.
Who Made This?
Asterisk was originally written by Mark Spencer of Digium, Inc. Code has been contributed from open source coders around the world, and testing and bug-patches from the community have provided invaluable aid to the development of this software.
Asterisk Architecture
Asterisk is carefully designed for maximum flexibility. Specific APIs are defined around a central PBX core system. This advanced core handles the internal interconnection of the PBX, cleanly abstracted from the specific protocols, codecs, and hardware interfaces from the telephony applications. This allows Asterisk to use any suitable hardware and technology available now or in the future to perform its essential functions, connecting hardware and applications.
The Asterisk core handles these items internally:
- PBX Switching - The essence of Asterisk, of course, is a Private Branch Exchange Switching system, connecting calls together between various users and automated tasks. The Switching Core transparently connects callers arriving on various hardware and software interfaces.
- Application Launcher - launches applications which perform services for uses, such as voicemail, file playback, and directory listing.
- Codec Translator - uses codec modules for the encoding and decoding of various audio compression formats used in the telephony industry. A number of codecs are available to suit diverse needs and arrive at the best balance between audio quality and bandwidth usage.
- Scheduler and I/O Manager - handles low-level task scheduling and system management for optimal performance under all load conditions.
Loadable Module APIs:
Four APIs are defined for loadable modules, facilitating hardware and protocol abstraction. Using this loadable module system, the Asterisk core does not have to worry about details of how a caller is connecting, what codecs are in use, etc.
- Channel API - the channel API handles the type of connection a caller is arriving on, be it a VoIP connection, ISDN, PRI, Robbed bit signaling, or some other technology. Dynamic modules are loaded to handle the lower layer details of these connections.
- Application API - the application API allows for various task modules to be run to perform various functions. Conferencing, Paging, Directory Listing. Voicemail, In-line data transmission, and any other task which a PBX system might perform now or in the future are handled by these separate modules.
- Codec Translator API - loads codec modules to support various audio encoding and decoding formats such as GSM, Mu-Law, A-law, and even MP3.
- File Format API - handles the reading and writing of various file formats for the storage of data in the filesystem.
Using these APIs Asterisk achieves a complete abstraction between its core functions as a PBX server system and the varied technologies existing (or in development) in the telephony arena. The modular form is what allows Asterisk to seamlessly integrate both currently implemented telephony switching hardware and the growing Packet Voice technologies emerging today. The ability to load codec modules allows Asterisk to support both the extremely compact codecs necessary for Packet Voice over slow connections such as a telephone modem while still providing high audio quality over less constricted connections.
The application API provides for flexible use of application modules to perform any function flexibly on demand, and allows for open development of new applications to suit unique needs and situations. In addition, loading all applications as modules allows for a flexible system, allowing the administrator to design the best suited path for callers on the PBX system and modify call paths to suit the changing communication needs of a going concern.
Asterisk Features
Asterisk-based telephony solutions offer a rich and flexible feature set. Asterisk offers both classical PBX functionality and advanced features, and interoperates with traditional standards-based telephony systems and Voice over IP systems. Asterisk offers the features one would expect of a large proprietary PBX system such as Voicemail, Conference Bridging, Call Queuing, and Call Detail Records.
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