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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 ...
Asterisk Gateway Interface (AGI) is a software interface and communications protocol for application level control of selected features of the Asterisk PBX. AGI allows an external, user-written program, launched from the Asterisk dial plan via pipes to control telephony operations on its associated control and voice channels.
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 is a web-based open-source graphical user interface (GUI) that manages Asterisk, a voice over IP (VoIP) and telephony server. [2]FreePBX is licensed under the GNU General Public License version 3, [3] with commercial modules available under their own licenses.
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 voice over IP telephony sessions between servers and to terminal devices.
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 ...
A SIP address is written in user@domain.tld format in a similar fashion to an email address.An address like: sip:1-999-123-4567@voip-provider.example.net. instructs a SIP client to use the NAPTR and SRV schemes to look up the SIP server associated with the DNS name voip-provider.example.net and connect to that server.
SIP trunking is a voice over Internet Protocol (VoIP) technology and streaming media service based on the Session Initiation Protocol (SIP) by which Internet telephony service providers (ITSPs) deliver telephone services and unified communications to customers equipped with SIP-based private branch exchange (IP-PBX) and unified communications facilities. [1]