Asterisk (PBX)

Last updated
Asterisk
Developer(s) Sangoma Technologies Corporation
Stable release(s) [±]
21.3.1 (17 May 2024;8 months ago (2024-05-17) [1] )

20.5.0 LTS (18 October 2023;15 months ago (2023-10-18) [2] )
19.8.1 (7 July 2023;18 months ago (2023-07-07) [3] )

18.20.0 LTS (18

Contents

 October 2023;15 months ago (2023-10-18) [4] )
Written in C
Type Voice over Internet Protocol
License GPLv2 with additional licenses available from Digium [5]
Website www.asterisk.org   OOjs UI icon edit-ltr-progressive.svg

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]

Features

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.

Internationalization

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]

Derived products

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]

See also

Related Research Articles

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.

<span class="mw-page-title-main">Analog telephone adapter</span> Type of telephone adapter

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<span class="mw-page-title-main">Business telephone system</span> Telephone system typically used in business environments

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.

<span class="mw-page-title-main">Skype for Business Server</span> Real-time communications server software

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<span class="mw-page-title-main">SipXecs</span>

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.

<span class="mw-page-title-main">FreePBX</span> Open-source graphical user interface

FreePBX is a web-based open-source graphical user interface (GUI) that manages Asterisk, a voice over IP (VoIP) and telephony server.

snom German brand of VoIP telephones

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.

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<span class="mw-page-title-main">Elastix</span> Linux distribution

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<span class="mw-page-title-main">AskoziaPBX</span> Telephone system firmware

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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.

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<span class="mw-page-title-main">FreePBX Distro</span> Software system

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.

References

  1. ↑ "ChangeLog-21-current" . Retrieved 22 May 2024.
  2. ↑ "ChangeLog-20-current" . Retrieved 27 Oct 2023.
  3. ↑ "ChangeLog-19-current" . Retrieved 27 Oct 2023.
  4. ↑ "ChangeLog-18-current" . Retrieved 27 Oct 2023.
  5. ↑ "Asterisk LICENSE". GitHub . Retrieved 2020-02-13.
  6. ↑ Olejniczak, Stephen P.; Kirby, Brady (2007). Asterisk For Dummies . John Wiley & Sons. ISBN   9780470098547.
  7. ↑ Van Meggelen, Jim; Smith, Jared; Madsen, Leif (2007). Asterisk: The Future of Telephony. O'Reilly Media, Inc. ISBN   9780596510480.
  8. ↑ The README for version 0.1.0 states: "Currently, the Asterisk Open Source PBX is only known to run on the Linux OS, although it may be portable to other UNIX-like operating systems as well." See here Archived 2017-02-12 at the Wayback Machine
  9. ↑ "Asterisk on OpenWrt". 6 June 2018. Retrieved 2018-06-10.
  10. ↑ AstLinux: Boot via USB Flash Storage
  11. ↑ download page of sound files for Asterisk
  12. ↑ ץ "You are not the next caller in line" Archived 2016-03-04 at the Wayback Machine , parody on-hold message where a pre-recorded Allison Smith sheepishly confesses (at 0:00:45) that the caller is actually *not* next in queue and would be lucky to get a response at 11:30pm from the cleaning lady after other workers had left for the day.
  13. ↑ "Zombie-Proof Your Phone System". Go.digium.com. Archived from the original on 2016-03-05. Retrieved 2016-01-04.
  14. ↑ "Asterisk Voice Recordings". The IVR Voice. Archived from the original on 2015-01-18. Retrieved 2015-01-17.
  15. ↑ astlinux-project.org
  16. ↑ Madsen, Leif; Jim Van Meggelen; Russell Bryant (2013). Asterisk: The Definitive Guide, 4th Edition (4th ed.). O'Reilly Media. p. 800. ISBN   978-1-4493-3242-6. 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.
  17. ↑ "astTECS - THE IP-PBX PEOPLE". VoIP-Info. 12 July 2008.
  18. ↑ "Datasheet_UCM6300_Audio_Series_English.pdf" (PDF). Retrieved 2025-01-21. Based on Asterisk* version 16 open source telephony operating system
  19. ↑ "E-MetroTel Green Migration" . Retrieved 2025-01-22.
  20. ↑ "Allison On Demand". AsteriskExchange.com.
  21. ↑ "Asterisk Software Add-Ons". Digium. 2015-12-29. Archived from the original on 2015-03-16. Retrieved 2016-01-04.
  22. ↑ Riddell, Matt (2009-08-08). "35 Great free Asterisk applications". Venturevoip.com. Archived from the original on 2015-04-26. Retrieved 2016-01-04.
  23. ↑ "The 5-Minute PBX: Incredible PBX 11 and Incredible Fax Get a Facelift". Nerd Vittles. 2013-06-18. Retrieved 2016-01-04.
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