 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 & µ-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 |
|
