Kuwait Linux User Group

Asterisk™ v1.4.4 - The Open Source Linux PBX
Date: Wednesday, May 09, 2007 @ 03:19:10 EDT
Topic: adv


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™ 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.

Call Features
  • ADSI On-Screen Menu System
  • Alarm Receiver
  • Append Message
  • Authentication
  • Automated Attendant
  • Blacklists
  • Blind Transfer
  • Call Detail Records
  • Call Forward on Busy
  • Call Forward on No Answer
  • Call Forward Variable
  • Call Monitoring
  • Call Parking
  • Call Queuing
  • Call Recording
  • Call Retrieval
  • Call Routing (DID & ANI)
  • Call Snooping
  • Call Transfer
  • Call Waiting
  • Caller ID
  • Caller ID Blocking
  • Caller ID on Call Waiting
  • Calling Cards
  • Conference Bridging
  • Database Store / Retrieve
  • Database Integration
  • Dial by Name
  • Direct Inward System Access
  • Distinctive Ring
  • Distributed Universal Number Discovery (DUNDi™)
  • Do Not Disturb
  • E911
  • ENUM
  • Fax Transmit and Receive (3rd Party OSS Package)
  • Flexible Extension Logic
  • Interactive Directory Listing
  • Interactive Voice Response (IVR)
  • Local and Remote Call Agents
  • Macros
  • Music On Hold
    • Flexible Mp3-based System
    • Random or Linear Play
    • Volume Control
  • Predictive Dialer
  • Privacy
  • Open Settlement Protocol (OSP)
  • Overhead Paging
  • Protocol Conversion
  • Remote Call Pickup
  • Remote Office Support
  • Roaming Extensions
  • Route by Caller ID
  • SMS Messaging
  • Spell / Say
  • Streaming Media Access
  • Supervised Transfer
  • Talk Detection
  • Text-to-Speech (via Festival)
  • Three-way Calling
  • Time and Date
  • Transcoding
  • Trunking
  • VoIP Gateways
  • Voicemail
    • Visual Indicator for Message Waiting
    • Stutter Dialtone for Message Waiting
    • Voicemail to email
    • Voicemail Groups
    • Web Voicemail Interface
  • Zapateller
Computer-Telephony Integration
  • AGI (Asterisk Gateway Interface
  • Graphical Call Manager
  • Outbound Call Spooling
  • Predictive Dialer
  • TCP/IP Management Interface
Scalability
  • TDMoE (Time Division Multiplex over Ethernet)
    • Allows direct connection of Asterisk PBX
    • Zero latency
    • Uses commodity Ethernet hardware
  • Voice-over IP
    • Allows for integration of physically separate installations
    • Uses commonly deployed data connections
    • Allows a unified dialplan across multiple offices
Codecs
  • ADPCM
  • G.711 (A-Law & ?-Law)
  • G.723.1 (pass through)
  • G.726
  • G.729 (through purchase of commercial license through Digium)
  • GSM
  • iLBC
  • Linear
  • LPC-10
  • Speex
Protocols
  • IAX™ (Inter-Asterisk Exchange)
  • H.323
  • SIP (Session Initiation Protocol)
  • MGCP (Media Gateway Control Protocol
  • SCCP (Cisco? Skinny?)
Traditional Telephony Interoperability
  • E&M
  • E&M Wink
  • Feature Group D
  • FXS
  • FXO
  • GR-303
  • Loopstart
  • Groundstart
  • Kewlstart
  • MF and DTMF support
  • Robbed-bit Signaling (RBS) Types
PRI Protocols
  • 4ESS
  • BRI (ISDN4Linux)
  • DMS100
  • EuroISDN
  • Lucent 5E
  • National ISDN2
  • NFAS
Homepage and more info here:
Code:
http://www.asterisk.org/


Download Asterisk™ from here:
Code:
http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.4.tar.gz


Download Zaptel from here:
Code:
http://ftp.digium.com/pub/zaptel/releases/zaptel-1.4.2.1.tar.gz


Download Libpri from here:
Code:
http://ftp.digium.com/pub/libpri/releases/libpri-1.4.0.tar.gz


Download Asterisk-Addons from here:
Code:
http://ftp.digium.com/pub/asterisk/releases/asterisk-addons-1.4.1.tar.gz


Download Asterisk-Sounds from here:
Code:
http://ftp.digium.com/pub/asterisk/releases/asterisk-sounds-1.2.1.tar.gz


Download the Asterisk Handbook Project Draft (PDF) from here:
Code:
http://www.digium.com/handbook-draft.pdf


Check ChangeLog here:
Code:
http://ftp.digium.com/pub/telephony/asterisk/ChangeLog-1.4.4







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