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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 ...
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.
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.
Servers can optionally send this response to indicate a call is being forwarded. [1]: §21.1.3 182 Queued Indicates that the destination was temporarily unavailable, so the server has queued the call until the destination is available. A server may send multiple 182 responses to update progress of the queue. [1]: §21.1.4 183 Session Progress
Creacode SIP Application Server Real-time SIP call controller and IVR product for carrier-class VoIP networks; Dialogic Corporation Powermedia Media Servers, audio and video SIP IVR, media and conferencing servers for Enterprise and Carriers. Dialexia VoIP Softswitches, IP PBX for medium and enterprise organizations, billing servers.
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.
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