What is Asterisk?

Asterisk, the world's most popular open source communications project, is free, open source software that converts an ordinary computer into a feature-rich voice communications server. Asterisk makes it simple to create and deploy a wide range of telephony applications and services, including IP PBXs, VoIP gateways, call center ACDs and IVR systems. 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 leading open source telephony project and the Asterisk community has been ranked as a key factor in the growth of VoIP. What Does Asterisk Do? Asterisk is like an erector set or a box of Legos for people who want to create communications applications. That's why we refer to it as a "tool-kit" or "development platform". Asterisk includes all the building blocks needed to create a PBX system, an IVR system or virtually any other kind of communications solution. The "blocks" in the kit include:Drivers for various VoIP protocols. Drivers for PSTN interface cards and devices. Routing and call handling for incoming calls. Outbound call generation and routing. Media management functions (record, play, generate tone, etc.). Call detail recording for accounting and billing. Transcoding (conversion from one media format to another). Protocol conversion (conversion from one protocol to another). Database integration for accessing information on relational databases. Web services integration for accessing data using standard internet protocols. LDAP integration for accessing corporate directory systems. Single and mult-party call bridging. Call recording and monitoring functions. Integrated "Dialplan" scripting language for call processing. External call management in any programming or scripting language through Asterisk Gateway Interface (AGI) Event notification and CTI integration via the Asterisk Manager Interface (AMI). Speech synthesis (aka "text-to-speech") in various languages and dialects using third party engines. Speech recognition in various languages using third party recognition engines.

This combination of components allows an integrator or developer to quickly create voice-enabled applications. The open nature of Asterisk means that there is no fixed limit on what it can be made to do. Asterisk integrators have built everything from very small IP PBX systems to massive carrier media servers. Asterisk As A 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. Asterisks 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? Asterisks got you covered. How about a conference bridge? Yep. Its 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 Public 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 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).

Asterisk is available in Debian Stable and is maintained by the Debian VoIP Team. Using the Inter-Asterisk eXchange (IAX) Voice over IP protocol Asterisk merges voice and data Supported hardware Asterisk needs no additional traffic seamlessly across disparate networks. The use hardware for Voice over IP. For interconnection with of Packet Voice allows Asterisk to send data such digital and analog telephony equipment, Asterisk as URL information and images in-line with voice supports a number of hardware devices, most notably traffic, allowing advanced integration of information. all of the hardware manufactured by Digium, the creator of Asterisk. Asterisk provides a central switching core, with four APIs for modular loading of telephony Features Asterisk-based telephony solutions offer a applications, hardware interfaces, file format rich and flexible feature set. Asterisk offers both handling, and codecs. It allows for transparent classical PBX functionality and advanced features switching between all supported interfaces, allowing which interoperates with traditional standards-based it to tie together a diverse mixture of telephony telephony systems and Voice over IP systems. systems into a single switching network.

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. Examples: 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... DAHDI (Digium Asterisk Hardware Device Interface) - A high density kernel telephony interface for PSTN hardware.

Dialplan - A dial plan establishes the expected number and pattern of digits for a telephone number. This includes country codes, access codes, area codes and all combinations of digits dialed. For instance, the North American public switched telephone network (PSTN) uses a 10-digit dial plan that includes a 3-digit area code and a 7-digit telephone number. Most PBXs support variable-length dial plans that use 3 to 11 digits. Dial plans must comply with the telephone networks to which they connect. 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. 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. 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. 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. 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. GSM - A compressed speech codec that uses a rate of 13 kbps. H.323 - A VOIP protocol that was deployed early and is widely adopted. IAX (Inter-Asterisk eXchange) - A VOIP protocol designed to be much more NAT friendly. IAX currently only transports audio. 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. PRI (Primary Rate Interface) - A PRI is a truly digital circuit, containing 24 ISDN channels. One of these channels is a D channel and used for signaling. The rest are B channels and used to transport audio. PSTN (Public Switched Telephone Network) - Originally a network of fixed-line analog telephone systems, the PSTN is now almost entirely digital and includes mobile as well as fixed telephones. The network works in much the same way that the Internet is the network of the world's public IP-based packet-switched networks. 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. SIP (Session Initiation Protocol) - A signaling protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP). SIP adoption amongst hardware and software vendors continues to expand. 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. Zaptel - The Zaptel project has been renamed 'DAHDI' as of May 2008. DAHDI is a series of drivers for telephony hardware devices.

Codecs ADPCM G.711 (A-Law & -Law) G.719 (pass through) G.722 G.722.1 licensed from Polycom G.722.1 Annex C licensed from Polycom G.723.1 (pass through) G.726 G.729a GSM iLBC Linear LPC-10 Speex ISDN Protocols AT&T 4ESS EuroISDN PRI and BRI Lucent 5ESS National ISDN 1 National ISDN 2 NFAS Nortel DMS100 Q.SIG

VoIP Protocols Google Talk H.323 IAX (Inter-Asterisk eXchange) Jingle/XMPP MGCP (Media Gateway Control Protocol SCCP (Cisco Skinny) SIP (Session Initiation Protocol) Skype UNIStim

Traditional Telephony Protocols E&M E&M Wink Feature Group D FXS FXO GR-303 Loopstart Groundstart Kewlstart MF and DTMF support Robbed-bit Signaling (RBS) Types MFC-R2 (Not supported. However, a patch is available)