![]() | |
Developer(s) | Sangoma Technologies Corporation |
---|---|
Stable release(s) [±] | |
21.3.1 (17 May 2024 [1] ) 20.5.0 LTS (18 October 2023 [2] ) ContentsOctober 2023 [4] ) | |
Preview release(s) [±] | |
Repository | |
Written in | C |
Type | Voice over Internet Protocol |
License | GPLv2 with additional licenses available from Digium [5] |
Website | www![]() |
Asterisk is a software implementation of a private branch exchange (PBX). In conjunction with suitable telephony hardware interfaces and network applications, Asterisk is used to establish and control telephone calls between telecommunication endpoints such as customary telephone sets, destinations on the public switched telephone network (PSTN) and devices or services on voice over Internet Protocol (VoIP) networks. Its name comes from the asterisk (*) symbol for a signal used in dual-tone multi-frequency (DTMF) dialing.
Asterisk was created in 1999 by Mark Spencer of Digium, which, since 2018, has been a division of Sangoma Technologies Corporation. [6] [7] Originally designed for Linux, [8] Asterisk runs on a variety of operating systems, including NetBSD, OpenBSD, FreeBSD, macOS, and Solaris, and can be installed in embedded systems based on OpenWrt. [9] [10]
The Asterisk software includes many features available in commercial and proprietary PBX systems: voice mail, conference calling, interactive voice response (phone menus), and automatic call distribution. Users can create new functionality by writing dial plan scripts in several of Asterisk's own extensions languages by adding custom loadable modules written in PHP or C or by implementing Asterisk Gateway Interface (AGI) programs using any programming language capable of communicating via the standard streams system (stdin and stdout) or by network TCP sockets.
Asterisk supports several standard VOIP protocols, including the Session Initiation Protocol (SIP), the Media Gateway Control Protocol (MGCP), and H.323. Asterisk supports most SIP telephones, acting both as registrar and back-to-back user agent. It can serve as a gateway between IP phones and the PSTN via T- or E-carrier interfaces or analog FXO cards. The Inter-Asterisk eXchange (IAX) protocol, RFC 5456, native to Asterisk, provides efficient trunking of calls between Asterisk PBX systems in addition to distributing some configuration logic. Many VoIP service providers support it for call completion into the PSTN, often because they themselves have deployed Asterisk or offer it as a hosted application. Some telephones also support the IAX protocol.
By supporting a variety of traditional and VoIP telephony services, Asterisk allows deployers to build telephone systems, or migrate existing systems to new technologies. Some sites are using Asterisk to replace proprietary PBXes, others provide additional features, such as voice mail or voice response menus, or virtual call shops, or to reduce cost by carrying both local and long-distance calls over the Internet.
In addition to VoIP protocols, Asterisk supports traditional circuit-switching protocols such as ISDN and SS7. This requires appropriate hardware interface cards, marketed by third-party vendors. Each protocol requires the installation of software modules. In Asterisk release 14 the Opus audio codec is supported.
While initially developed in the United States, Asterisk has become a popular VoIP PBX worldwide. It allows having multiple sets of voice prompts identified by language (and even multiple sets of prompts for each language) as well as support for time formats in different languages. Several sets of prompts for the interactive voice response and voice mail features are included with Asterisk: American, British, and Australian English, Canadian French, Japanese, Russian, Mexican Spanish and Swedish. [11] A few novelty prompts are offered, such as jokes [12] and a themed "zombie apocalypse" message for Halloween. [13] Additionally, voice sets are offered for commercial sale in various languages, dialects, and genders.
The default set of English-language Asterisk prompts are recorded by professional telephone voice Allison Smith. [14]
Asterisk is a core component in many commercial products and open-source projects. Some of the commercial products are hardware and software bundles, for which the manufacturer supports and releases the software with an open-source distribution model.
Various add-on products, often commercial, are available that extend Asterisk features and capabilities.
The standard voice prompts included with the system are free. A business can purchase matching voice announcements of its company name, IVR menu options and employee or department names (as a library of live recordings of common names [20] or a set of fully customised prompts recorded by the same professional voice talent) at additional cost for seamless integration into the system.
Other add-ons provide fax support, text-to-speech, additional codecs and new features. [21] Some third-party add-ons are free; [22] a few even support embedded platforms such as the Raspberry Pi. [23]
Voice over Internet Protocol (VoIP), also known as IP telephony, refers to a set of technologies used for voice communication sessions over Internet Protocol (IP) networks, such as the Internet. VoIP enables voice calls to be transmitted as data packets, facilitating various methods of voice communication, including traditional applications like Skype, Microsoft Teams, Google Voice, and VoIP phones. Regular telephones can also be used for VoIP by connecting them to the Internet via analog telephone adapters (ATAs), which convert traditional telephone signals into digital data packets that can be transmitted over IP networks.
An analog telephone adapter (ATA) or FXS gateway is a device for connecting traditional analog telephones, fax machines, and similar customer-premises devices to a digital telephone system or a voice over IP telephone network.
A business telephone system is a telephone system typically used in business environments, encompassing the range of technology from the key telephone system (KTS) to the private branch exchange (PBX).
Direct inward dialing (DID), also called direct dial-in (DDI) in Europe and Oceania, is a telecommunication service offered by telephone companies to subscribers who operate private branch exchange (PBX) systems. The feature provides service for multiple telephone numbers over one or more analog or digital physical circuits to the PBX, and transmits the dialed telephone number to the PBX so that a PBX extension is directly accessible for an outside caller, possibly by-passing an auto-attendant.
Skype for Business Server is real-time communications server software that provides the infrastructure for enterprise instant messaging, presence, VoIP, ad hoc and structured conferences and PSTN connectivity through a third-party gateway or SIP trunk. These features are available within an organization, between organizations and with external users on the public internet or standard phones.
SipXecs is a free software enterprise communications system. It was initially developed by Pingtel Corporation in 2003 as a voice over IP telephony server located in Boston, MA. The server was later extended with additional collaboration capabilities as part of the SIPfoundry project. Since its extension, sipXecs now acts as a software implementation of the Session Initiation Protocol (SIP), making it a full IP-based communications system.
Mobile VoIP or simply mVoIP is an extension of mobility to a voice over IP network. Two types of communication are generally supported: cordless telephones using DECT or PCS protocols for short range or campus communications where all base stations are linked into the same LAN, and wider area communications using 3G or 4G protocols.
Text over IP is a means of providing a real-time text (RTT) service that operates over IP-based networks. It complements Voice over IP (VoIP) and Video over IP.
FreePBX is a web-based open-source graphical user interface (GUI) that manages Asterisk, a voice over IP (VoIP) and telephony server.
Snom Technology GmbH is a German company which manufactures Voice over Internet Protocol (VoIP) telephones, based on the IETF standard Session Initiation Protocol (SIP). Snom's products are targeted at the small- to medium-sized business sector, home offices, Internet service providers, carriers, and original equipment manufacturers. The company, founded in 1996 and headquartered in Berlin, is a wholly owned subsidiary of VTech Holdings Limited, since 2016.
Unified communications (UC) is a business and marketing concept describing the integration of enterprise communication services such as instant messaging (chat), presence information, voice, mobility features, audio, web & video conferencing, fixed-mobile convergence (FMC), desktop sharing, data sharing, call control and speech recognition with non-real-time communication services such as unified messaging. UC is not necessarily a single product, but a set of products that provides a consistent unified user interface and user experience across multiple devices and media types.
An IP PBX is a system that connects telephone extensions to the public switched telephone network (PSTN) and provides internal communication for a business. An IP PBX is a PBX system with IP connectivity and may provide additional audio, video, or instant messaging communication utilizing the TCP/IP protocol stack.
Elastix is a unified communications server software that brings together IP PBX, email, IM, faxing and collaboration functionality. It has a Web interface and includes capabilities such as a call center software with predictive dialing.
A softphone is a software program for making telephone calls over the Internet using a general purpose computer rather than dedicated hardware. The softphone can be installed on a piece of equipment such as a desktop, mobile device, or other computer and allows the user to place and receive calls without requiring an actual telephone set. Often, a softphone is designed to behave like a traditional telephone, sometimes appearing as an image of a handset with a display panel and buttons with which the user can interact. A softphone is usually used with a headset connected to the sound card of the PC or with a USB phone.
Aculab is a privately held, UK-based limited company that was founded in 1978. With headquarters and R&D facilities located in Milton Keynes, UK, and its branch office is in Norwood, Massachusetts, USA.
AskoziaPBX is a closed source telephone system firmware. It is a fork of the m0n0wall project and uses the Asterisk private branch exchange (PBX) software to realize all telephony functions.
VaxTele SIP Server SDK is a complete development toolkit, which allows software vendors and Internet telephony service providers (ITSP) to develop SIP Server and (SIP) Session Initiation Protocol based VoIP systems for Microsoft Windows to install computer to computer voice chat, chat rooms, IVR systems, call center services, calling card services, dial/receive computer to PSTN and mobile phone calling services.
The 1100-series IP phones are 6 different desktop IP clients manufactured by Avaya for Unified communications which can operate on the SIP or UNIStim protocols. The SIP Firmware supports presence selection and notification along with secure instant messaging.
The FreePBX Distro was a freeware unified communications software system that consisted of a graphical user interface (GUI) for configuring, controlling and managing Asterisk PBX software. The FreePBX Distro included packages that offer VoIP, PBX, Fax, IVR, voice-mail and email functions.
FreePBX, the juggernaut of the Asterisk community. This interface (which is at the heart of many of the most popular Asterisk distributions, such as AsteriskNOW, Elastix, the FreePBX Distro, and PBX in a Flash), is unarguably a very large part of why Asterisk has been as successful as it has. With the FreePBX interface, you can configure and manage many aspects of an Asterisk system without touching a single configuration file. While we purists may like everyone to work only with the config files, we recognize that for many, learning Linux and editing these files by hand is simply not going to happen. For those folks, there is FreePBX, and it has our respect for the important contributions it has made to the success of Asterisk.
Based on Asterisk* version 16 open source telephony operating system