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Dialexia VoIP Softswitches, IP PBX for medium and enterprise organizations, billing servers. IBM WebSphere Application Server - Converged HTTP and SIP container JEE Application Server; Interactive Intelligence Windows-based IP PBX for small, medium and enterprise organizations; Kerio Operator, IP PBX for small and medium enterprises
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
Dial-Office IP-PBX is a SIP-based IP-PBX business phone system, [1] first released in 2003 by Canadian telecommunications software provider Dialexia.The software allows users to connect multiple phones (e.g., 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 ...
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]
An application instantiates the session with the Session Description Protocol (SDP) over Session Initiation Protocol (SIP) or other rendezvous methods. The MSRP protocol is defined in RFC 4975. [1] MSRP messages can also be transmitted by using intermediaries peers, by using the relay extensions defined in RFC 4976. [2]
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 ...
FreeSWITCH is a free and open-source telephony software for real-time communication protocols using audio, video, text and other forms of media. The software has applications in WebRTC, voice over Internet Protocol (VoIP), video transcoding, Multipoint Control Unit (MCU) functionality and supports Session Initiation Protocol (SIP) features.
SipXecs is designed as a software-only, distributed cloud application.It runs on the Linux operating system CentOS or RHEL on either virtualized or physical servers. A minimum configuration allows running all of the sipXecs components on a single server, including database, all available services, and the sipXecs management.