Asterisk is the world’s leading open source telephony engine and tool kit. Offering flexibility unheard of in the world of proprietary communications, Asterisk empowers developers and integrators to create advanced communication solutions...for free.

Asterisk® is released as open source under the GNU General Public License (GPL), and it is available for download free of charge. Asterisk® is the most popular open source software available, with the Asterisk Community being the top influencer in VoIP.

Asterisk as a switch (PBX)

Asterisk can be configured as the core of an IP or hybrid PBX, switching calls, managing routes, enabling features, and connecting callers with the outside world over IP, analog (POTS), and digital (T1/E1) connections.

Asterisk runs on a wide variety of operating systems including Linux, Mac OS X, OpenBSD, FreeBSD and Sun Solaris and provides all of the features you would expect from a PBX including many advanced features that are often associated with high end (and high cost) proprietary PBXs. Asterisk's architecture is designed for maximum flexibility and supports Voice over IP in many protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.

Asterisk as a Gateway

It can also be built out as the heart of a media gateway, bridging the legacy PSTN to the expanding world of IP telephony. Asterisk’s modular architecture allows it to convert between a wide range of communications protocols and media codecs.

Asterisk as a Feature/Media Server

Need an IVR? Asterisk’s got you covered. How about a conference bridge? Yep. It’s in there. What about an automated attendant? Asterisk does that too. How about a replacement for your aging legacy voicemail system? Can do. Unified messaging? No problem. Need a telephony interface for your web site? Ok.

Asterisk in the Call Center

Asterisk has been adopted by call centers around the world based on its flexibility. Call center and contact center developers have built complete ACD systems based on Asterisk. Asterisk has also added new life to existing call center solutions by adding remote IP agent capabilities, advanced skills-based routing, predictive and bulk dialing, and more.

Asterisk in the Network

Internet Telephony Service Providers (ITSPs), competitive local exchange carriers (CLECS) and even first-tier incumbents have discovered the power of open source communications with Asterisk. Feature servers, hosted services clusters, voicemail systems, pre-paid calling solutions, all based on Asterisk have helped reduce costs and enabled flexibility.

Asterisk Everywhere

Asterisk has become the basis for thousands of communications solutions. If you need to communicate, Asterisk is your answer.

Supported Platforms

Asterisk® is primarily developed on GNU/Linux for x/86 and runs on GNU/Linux for PPC along with OpenBSD, FreeBSD, and Mac OS X. 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.

Supported Hardware

Asterisk® needs no additional hardware for Voice over IP. For interconnection with digital and analog telephony equipment, Asterisk® supports a number of hardware devices.

Features

Asterisk-based telephony solutions offer a rich and flexible feature set. Asterisk® offers both classical PBX functionality and advanced features which interoperates with traditional standards-based telephony systems and Voice over IP systems.

Supported Protocols

Asterisk® supports a wide range of protocols for the handling and transmission of voice over traditional telephony interfaces including H.323, Session Initiation Protocol (SIP), Media Gateway Control Protocol (MGCP), and Skinny Client Control Protocol (SCCP).

Using the Inter-Asterisk eXchange (IAX™) Voice over IP protocol Asterisk® merges voice and data traffic seamlessly across disparate networks. The use of Packet Voice allows Asterisk® 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.

Glossary of Asterisk-related Terms

ACD

Automatic Call Distributor -- A device or system that distributes incoming calls to a specific group of terminals that agents use. It is often part of a computer telephony integration (CTI) system.

CODEC

Coder/Decoder -- A software library that contains the algorithms necessary to convert an analog signal to and from a digital one. See also: Encode, G.711, G.729, GSM

Context

The dialplan is composed of one or more extension contexts. Each extension context is itself simply a collection of extensions. Each extension context in a dialplan has a unique name associated with it. The use of contexts can be used to implement a number of important features, such as security, routing, autoattendant, multilevel menus, authentication, callback, privacy, macros, etc... See also: Dialplan

Dialplan

See also: Context

E&M

Ear & Mouth -- A type of signaling commonly used over T1 and E1 interfaces.

Encode

The process of converting an analog signal into a digital signal that can be manipulated easily by a computer. See also: CODEC

FXO

Foreign Exchange Office -- A device usually found on the customer end that is powered by the channel and can interface into the telephone company's network. Digium makes FXO modules that interface with PSTN lines using FXS signalling in either Loopstart(fxs_ls) or the more common Kewlstart(fxs_ks) modes. See also: PSTN, REN

FXS

Foreign Exchange Station -- A device usually located on the telephony company's property, a FXS device send power through a channel to a phone on the other end. Digium makes FXS modules that interface with PSTN phones using FXO signalling in either Loopstart(fxo_ls) or the more common Kewlstart(fxo_ks) modes. See also: PSTN, REN

G.711

An uncompressed codec that samples a 64kbps channel at 8 bits per sample using pulse code modulation. The Two varients of G.711 are known formally as uLaw and aLaw. See also: CODEC, G.729, GSM

G.729

The G.729 codec is an industry standard which allows for stuffing more calls in limited bandwidth to utilize IP voice in more cost effective ways. A typical call consumes 64Kbps of voice bandwidth. G.729 reduces the call to 8Kbps (normal IP overhead adds to this number). Many people are using Asterisk with G.729 to replace expensive gateways. See also: CODEC, G.711, GSM

GSM

A compressed speech codec that uses a rate of 13 kbps. See also: CODEC, G.711, G.729

H.323

A VOIP protocol that was deployed early and is widely adopted. See also: IAX, MGCP, SIP, VoIP

IAX

Inter-Asterisk eXchange -- A VOIP protocol designed to be much more NAT friendly. IAX currently only transports audio. See also: H.323, MGCP, SIP, VoIP

IVR

Interactive Voice Response -- An automated voice system that allows callers to navigate a phone system and be directed to the correct extension by pressing a series of numbers on a tuch-tone phone. (I.E. Push 1 for sales, push 2 for support, etc..)

MGCP

Media Gateway Control Protocol -- A VOIP Protocol that has both signaling and control and was designed to reduce complexity between media gateways. See also: H.323, IAX, SIP, VoIP

Open Source

Detailed definition coming soon...

PBX

Detailed definition coming soon...

PRI

Primary Rate Interface -- A PRI is a truly digital circuit, containing 24 ISDN channels. One of these channels is a dchannel and used for signaling. The rest are bchannels and used to transport audio.

PSTN

Public Switched Telephone Network -- This refers to the analog telephone network that's very common in residential networks. PSTN lines are usually much cheaper then T1 lines, but only provide one channel. Digium's FXO and FXS modules interface with this network. See also: FXO, FXS

REN

Ringer Equivalency Number -- A number which represents the ringer loading effect on a line. A ringer equivalency number of 1 represents the loading effect of a single traditional telephone set ringing circuit. Most modern telephones probably will have a REN lower than 1. The total REN expresses the total loading effect of the equipment on the ringing current generator (FXS). The REN of a Digium FXS board is 5 (representing "extension," i.e., parallel-connected telephones). The actual number of devices on the line may be greater than the REN limit, if their respective individual RENs are less than 1. See also: FXO, FXS

SIP

Detailed definition coming soon... See also: H.323, IAX, MGCP, VoIP

TDM

Time Division Multiplexing -- A processes of splitting one medium into two or more channels by using timed segments to transmit information.

Transcode

The process of converting a channel with one type of encoding to a different type of encoding in real time.

VoIP

Voice Over Internet Protocol -- A general method for transporting voice through the internet. VOIP commonly refers to the general method and not a specific protocol. SIP is currently the most widely used VOIP protocol. See also: H.323, IAX, MGCP, SIP

Zaptel

Detailed definition coming soon...