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IP PBX
An IP PBX ("Internet Protocol private branch exchange") 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. Voice over IP (VoIP) gateways can be combined with traditional PBX functionality to allow businesses to use their managed intranet to help reduce long distance expenses and take advantage of the benefits of a single network for voice and data ( converged network). An IP PBX may also provide CTI features. An IP PBX can exist as a physical hardware device or as a software platform. Function IP PBX is primarily a software hosted on a regular desktop or server as per the requirement demands based on the expected traffic & criticality. Till 2019 IP PBX were deployed primarily as inbound and outbound call center solutions for l ...
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Internet Protocol
The Internet Protocol (IP) is the network layer communications protocol in the Internet protocol suite for relaying datagrams across network boundaries. Its routing function enables internetworking, and essentially establishes the Internet. IP has the task of delivering packets from the source host to the destination host solely based on the IP addresses in the packet headers. For this purpose, IP defines packet structures that encapsulate the data to be delivered. It also defines addressing methods that are used to label the datagram with source and destination information. IP was the connectionless datagram service in the original Transmission Control Program introduced by Vint Cerf and Bob Kahn in 1974, which was complemented by a connection-oriented service that became the basis for the Transmission Control Protocol (TCP). The Internet protocol suite is therefore often referred to as ''TCP/IP''. The first major version of IP, Internet Protocol Version 4 (IPv4), is the do ...
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Conference Call
A conference call is a telephone call in which someone talks to several people at the same time. The conference call may be designed to allow the called party to participate during the call or set up so that the called party merely listens into the call and cannot speak. It is sometimes called ATC (audio teleconference). Conference calls can be designed so that the calling party calls the other participants and adds them to the call; however, participants are usually able to call into the conference call themselves by dialing a telephone number that connects to a "conference bridge," which is a specialized type of equipment that links telephone lines. Companies commonly use a specialized service provider who maintains the conference bridge, or who provides the phone numbers and PIN codes that participants dial to access the meeting or conference call. These service providers can often dial-out to participants, connecting them to call and introducing them to the parties who are o ...
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Office Equipment
Office supplies are consumables and equipment regularly used in offices by businesses and other organizations, by individuals engaged in written communications, recordkeeping or bookkeeping, janitorial and cleaning, and for storage of supplies or data. The range of items classified as office supplies varies, and typically includes small, expendable, daily use items, consumable products, small machines, higher cost equipment such as computers, as well as office furniture and art. Typical products Office supplies are typically divided by type of product and general use. Some of the many different office supply products include *Blank sheet paper: various sizes from small notes to letter and poster-size; various thicknesses from tissue paper to 120 pound; construction paper; photocopier and inkjet printer paper; *Preprinted forms: time cards, tax reporting forms (1099, W-2), "while you were out" pads, desk and wall calendars; *Label and adhesive paper: name tags, file folder labe ...
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Trixbox
trixbox (formerly Asterisk@Home) was a software PBX based on Asterisk. trixbox was initially released under the name Asterisk@Home. In October 2006 it was renamed to trixbox after Digium requested that its developers cease the use of the word "Asterisk"; the renaming was further justified by the fact that the product at that time consisted of much more than just Asterisk. Versions trixbox CE (end of life) trixbox CE, the community edition founded by Kerry Garrison and Andrew Gillis, was completely free and was available under a GPLv2 license. It has been downloaded more than any other Asterisk-based PBX system according to SourceForge, with over two million downloads as of August 22, 2009. trixbox CE was last updated with a beta release, and shortly after a blog post was made by Andrew Gillis, founder & leader of development stating he was leaving. In October 2012, Fonality announced that they would no longer support or update trixbox CE, and would instead focus on trixbo ...
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Dialexia
Dialexia Communications, Inc. is a privately held Canadian corporation headquartered in Montréal, Quebec, that develops, manufactures, and sells VoIP-based Telecommunication products and services. The company specializes in the development of webRTC-enabled PBX software for organizations in specific sectors of activity, notably education & hospitality. Dialexia software enables users to connect multiple phones (e.g., extension (telephone), extensions, ring groups, etc.), share lines among several phones and implement business PBX telephone phone features such as voicemail, caller ID, call forwarding & call recording into their virtual PBX. In a November 2014 press release, Dialexia announced the addition of WebRTC capabilities to its line of PBX software products. Founded in Montreal in 2001, the company is a member of the Board of Trade of Metropolitan Montreal. History Soon after its establishment, Dialexia released Dial-Office IP-PBX, an IP PBX platform designed as an al ...
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Bicom Systems
Bicom Systems is a producer and vendor of Asterisk (PBX)-based unified communications devices for VoIP businesses. Bicom Systems uses open standards telephony. Products include all of the software and hardware components involved in building a VoIP business or ITSP. History Bicom Systems began researching and creating telecoms management software in 2003. An Asterisk pioneer, Bicom Systems launched the first ever open standards turnkey telephony system in 2004. Bicom Systems has seven products including a desktop and mobile softphone, an IP PBX telephony platform, and an IP Key Systems product. Supported hardware manufacturers include Polycom, Yealink, and others. See also * Comparison of VoIP software * List of SIP software * IP PBX An IP PBX ("Internet Protocol private branch exchange") 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 connectiv ...
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3CX Phone System
The 3CX Phone System is the software-based private branch exchange (PBX) Phone system developed and marketed by the company, 3CX. The 3CX Phone System is based on the SIP (Session Initiation Protocol) standard and enables extensions to make calls via the public switched telephone network (PSTN) or via Voice over Internet Protocol (VoIP) services on premises, in the cloud, or via a cloud service owned and operated by the 3CX company. The 3CX Phone System is available for Windows, Linux, Raspberry Pi and supports standard SIP soft/hard phones, VoIP services, faxing, voice and web meetings, as well as traditional PSTN phone lines. History The 3CX solution was developed by 3CX, an international VoIP IP PBX software development technology company as an open-standards, software-based PBX. First published as a free IP PBX product in 2006, the product was intended to provide a VoIP solution for use in a Microsoft Windows environment. The first commercial edition of the product was laun ...
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Jingle (protocol)
Jingle is an extension to XMPP (Extensible Messaging and Presence Protocol) which adds peer-to-peer (P2P) session control (signaling) for multimedia interactions such as in Voice over IP (VoIP) or videoconferencing communications. It was designed by Google and the XMPP Standards Foundation. The multimedia streams are delivered using the Real-time Transport Protocol (RTP). If needed, NAT traversal is assisted using Interactive Connectivity Establishment (ICE). , the proposed Jingle specification had not yet been approved by the XMPP Standards Foundation, but is now a Draft Standard, meaning: "''Implementations are encouraged and the protocol is appropriate for deployment in production systems, but some changes to the protocol are possible before it becomes a Final Standard.''" The libjingle library, used by Google Talk to implement Jingle, has been released to the public under a BSD license. It implements both the current standard protocol and the older, pre-standard version. ...
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Inter-Asterisk EXchange
Inter-Asterisk eXchange (IAX) is a communications protocol native to the Asterisk private branch exchange (PBX) software, and is supported by a few other softswitches, PBX systems, and softphones. It is used for transporting VoIP telephony sessions between servers and to terminal devices. The original IAX protocol is deprecated and has been superseded by a second version, commonly called IAX2. The IAX2 protocol was published as an informational (non-standards-track) RFC 5456 by discretion of the RFC Editor in February 2010. Basic properties IAX is a VoIP protocol that can be used for any type of streaming media including video, but is mainly designed for IP voice calls. IAX uses a single User Datagram Protocol (UDP) data stream between endpoints for both the session signaling and the media payloads. Thus it uses only a single UDP port number, typically 4569. This feature provides benefits for traversing network address translators on network boundaries, as it simplifies firewa ...
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Session Initiation Protocol
The Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and terminating communication sessions that include voice, video and messaging applications. SIP is used in Internet telephony, in private IP telephone systems, as well as mobile phone calling over LTE (VoLTE). The protocol defines the specific format of messages exchanged and the sequence of communications for cooperation of the participants. SIP is a text-based protocol, incorporating many elements of the Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol (SMTP). A call established with SIP may consist of multiple media streams, but no separate streams are required for applications, such as text messaging, that exchange data as payload in the SIP message. SIP works in conjunction with several other protocols that specify and carry the session media. Most commonly, media type and parameter negotiation and media setup are performed with the Session Description ...
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Speech Recognition
Speech recognition is an interdisciplinary subfield of computer science and computational linguistics that develops methodologies and technologies that enable the recognition and translation of spoken language into text by computers with the main benefit of searchability. It is also known as automatic speech recognition (ASR), computer speech recognition or speech to text (STT). It incorporates knowledge and research in the computer science, linguistics and computer engineering fields. The reverse process is speech synthesis. Some speech recognition systems require "training" (also called "enrollment") where an individual speaker reads text or isolated vocabulary into the system. The system analyzes the person's specific voice and uses it to fine-tune the recognition of that person's speech, resulting in increased accuracy. Systems that do not use training are called "speaker-independent" systems. Systems that use training are called "speaker dependent". Speech recognition ...
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Speech Synthesis
Speech synthesis is the artificial production of human speech. A computer system used for this purpose is called a speech synthesizer, and can be implemented in software or hardware products. A text-to-speech (TTS) system converts normal language text into speech; other systems render symbolic linguistic representations like phonetic transcriptions into speech. The reverse process is speech recognition. Synthesized speech can be created by concatenating pieces of recorded speech that are stored in a database. Systems differ in the size of the stored speech units; a system that stores phones or diphones provides the largest output range, but may lack clarity. For specific usage domains, the storage of entire words or sentences allows for high-quality output. Alternatively, a synthesizer can incorporate a model of the vocal tract and other human voice characteristics to create a completely "synthetic" voice output. The quality of a speech synthesizer is judged by its similarity to ...
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