Search Results for "agent"

This software suite is designed to extend the functionality of the Asterisk PBX through platform-independant web-client applications. Includes the VICIdial inbound/outbound contact center application. The suite is scalable across multiple Asterisk servers.

wxCommunicator is a cross platform SIP softphone written in C++ utilizing customized sipXtapi user agent library and wxWidgets 2.8.9 GUI library. For a list of supported features see http://wxcommunicator.sourceforge.net/features.html .

Baresip is a portable and modular SIP User-Agent with audio and video support. the latest source code can be found here: https://github.com/alfredh/baresip

Peers is a very simple softphone. It's a SIP User-Agent, written in java, it works on windows, linux and mac. It can be used with SIP servers like opensips or asterisk IPBX. It supports G711 codec (PCMU and PCMA) and telephone-events (DTMF).

Artemisa is a honeypot for VoIP (SIP) networks. It is designed to connect to a VoIP domain as a user-agent backend in order to detect malicious activity at an early stage and also adjust the policies of the enterprise in real-time.

LuaSofia is a Lua binding of Sofia-Sip library. Sofia-SIP is an open-source SIP User-Agent library, compliant with the IETF RFC3261 specification. We decided to use git, source code can be get at: git://github.com/ppizarro/luasofia.git

PartiSIPation is a sip user agent which has an easy replaceable gui. So it can be used as a softphone as well as it can be integrated in any kind of application, e.g. online games.

Asterisk based callcenter system. Administration interface to manage queues, agents, accounts, skills - track calls, create statistics. Realtime queue monitor. Agent web interface with build in IAX softphone.

A Sip live audio feeding agent. The agent captures audio from sound card and sends live audio stream(uLaw) to caller(sip phone) using RTP. It is based on Peers 0.3(http://peers.sourceforge.net/). Can be used in IP telephony to broadcast live audio.

This is a patch for Asterisk PBX that makes it usable as a call agent in PacketCable networks. It adds support of NCS messaging to Asterisk PBX. NCS, in its turn, is supported by the majority of vendors who produce EMTAs.

JOpenPhone is an opensource UA (User Agent) JAVA application that allows users to connect to a voip servers using various voip protocols such as SIP, H323.

Open_PYOSS will be a platform to develop Operational Support Systems for the Cable MSO. It will capitalized on a Distributed NxTier Computing Agent environment and WebServices core technologies to provide seamless Managed Access for MSOs or xSPs

Open Source SIP Back to Back User Agent) and Billing system. Has an Embeded TCL interpreter making session control and session mediation limitless in flexibility. Useful for: Routing Engine, Calling Cards, SBC, Peering, Wholesale and more!

A complete contact center solution for the Asterisk PBX, capable of handling inbound and outbound campaigns. Provides an agent toolbar for CTI. For managers an admin panel controls every aspect of the call center. Stunning historical and realtime stats.

This C++ library has been designed as a Chrome SIP stack. Sippet is an open-source SIP User-Agent library, compliant with the IETF RFC 3261 specification. It can be used as a building block for SIP client software for uses such as VoIP, IM, and many other real-time and P2P communication services. The main target was to enable Javascript applications to use UDP, TCP and TLS transports along WebSocket. Existing SIP solutions for the browser are forced to use the WebSockets API to send/receive SIP messages. ...