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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.
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 ...
SIP requests and responses may be generated by any SIP user agent; user agents are divided into clients (UACs), which initiate requests, and servers (UASes), which respond to them. [ 1 ] : §8 A single user agent may act as both UAC and UAS for different transactions: [ 1 ] : p26 for example, a SIP phone is a user agent that will be a UAC when ...
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]
A SIP server, also known as a SIP proxy, manages all SIP calls within a network and takes responsibility for receiving requests from user agents for the purpose of placing and terminating calls. Asterisk
Most of the IP PBX installation uses Asterisk (PBX) for its telephony support, built on LAMP (Linux-Apache-MySQL-PHP). With telecom service providers across the world is slowly preferring SIP Trunks over Primary Rate Interface as main enterprise communication delivery, the IP PBXs will now be in demand extensively.
The SIP URI scheme is a Uniform Resource Identifier (URI) scheme for the Session Initiation Protocol (SIP) multimedia communications protocol. A SIP address is a URI that addresses a specific telephone extension on a voice over IP system. Such a number could be a private branch exchange or an E.164 telephone number dialled through a specific ...
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.