Search Results for "interactive c"
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Showing 42 open source projects for "interactive c"
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Build Agents and Models on One PlatformGemini Enterprise Agent Platform is Google Cloud's comprehensive platform for developers to build, scale, govern, and optimize agents and models. Choose from Google's most advanced models and third-party models like Anthropic's Claude Model Family.
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SIPSL (SIP Service Layer) is a Programmable Session Border Controller (SBC) and SIP B2BUA, as defined in RFC3261. Written in c++ and multithreaded. The Application logic can be implemented by extending the application class and implementing the call backs.
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Activa brings the Asterisk IP PBX to the call center. Built on top of Asterisk, Activa components enable successful call center implementations adding value in areas such as computer telephony, screenpop&click2dial, agent control, automatic dialing...
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GNU Gatekeeper (GnuGk)
H.323 Gatekeeper for VoIP and videconferencing
The project has moved! Please find current versions at https://www.gnugk.org/ The GNU Gatekeeper (GnuGk) is a full featured H.323 gatekeeper under GPL license. It supports VoIP and videoconferencing and includes Radius and database support and many call routing functions. The project has moved! Please find current versions at https://www.gnugk.org/ -
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GetSmart is a download manager that supports Multi-Connection downloads. With Multi-Mirror search it can download different parts from these mirror servers simultaneously. It can 'act as proxy' to serve multiple clients. It can also works as a Daemon.
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Screen Pop for Avaya PBX
Implement Screen Pop for Avaya PBX
The ScreenPop implements screen pop function for Avaya PBX. Only web application is supported and telephony data such as AGENTID, CLI, DNIS, UCID, SPLIT, CUSTDATA and UED can be passed to the web application. -
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VoIP Honey project provides a set of tools for building an entire honeynet, thus includes honeywall and honeypot emulating VoIP environments such as Asterisk PBX or OpenSer with fully configurable connections. Voip Honey runs on GNU/Linux and Windows Systems. It can be compiled for Mac OSX as well.
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This project implements a simple STUN server and client on Windows, Linux, and Solaris. The STUN protocol (Simple Traversal of UDP through NATs) is described in the IETF RFC 3489, available at http://www.ietf.org/rfc/rfc3489.txt
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GNU Telephony
GNU Telephony Services and Applications
GNU Telephony intends anyone to use free as in freedom software for telephony, and to do real-time collaboration freely and privately over the Internet, with the freedom to do so on any platform they choose to use. -
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skiputler makes your Topcom "Butler 4012 USB VoIP" phone work on Linux and Mac OS X. It's a userspace driver which connects the phone to Skype.
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Stop Cyber Threats with VM-Series Next-Gen Firewall on AzureGain integrated visibility across all traffic in a single pass. Deploy Palo Alto Networks VM-Series to determine application identity and content while automating security policy updates via rich APIs.
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sigtran-smsc
PoC: Short Message Service Center (SMSC) over SIGTRAN
SIGTRAN based Short Message Center (SMSC) - proof of concept This is home of my thesis project which had goal of doing open source based Short Message Service Center to be connected into real mobile network (2G, 3G, ...). As being developed in very limited schedule (bachelor thesis work) it was limited to have very basic functionality: receiving message and deliver it immediately as-is to the receiver. There is no message caching nor different message charset support. It uses... -
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Solid PBX is a multi-platform Soft Switch Class 5 software targeted for home/corporate/operator level markets.
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CapiSuite is an ISDN CAPI telecommunication suite providing easy to use telecommunication functions which can be controlled from Python scripts. CapiSuite is also distributed with example scripts for call incoming handling and fax sending.
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Simu server project is aimed for IP telephony (VoIP) between classical SIP phones and Flash streaming applications.
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BladeWareVXML is a portable VoiceXML 2.1 interpreter that is an enhanced version (performance, usability and integration) of OpenVXI. A commercial version, with documentation, sample code, and support options, is available from the Commetrex Website.
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Emergency Context Resolution with Internet Technologies The project is based on a draft from files draft-ietf-ecrit-* Developed by IETF ECRIT Working Group
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This is an Interactive Voice Response System using TAPI (MS TAPI 2.x).You can use this as an answering machine.But most of all this is meant as an illustration of how you can use TAPI API for developing IVR applications in VC++
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This project has been superseded by OpalVoip (https://sourceforge.net/projects/opalvoip/) and H323Plus (https://sourceforge.net/projects/h323plus/) The OpenH323 project provides full featured, interoperable, Open Source implementation of the ITU H.323 teleconferencing protocol that can be used by personal developers and commercial users without charge.
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A Bluetooth Handsfree implementation for Linux's Bluez stack. Allows you to use your PC's microphone and speakers as a handsfree speakerphone connected to your Bluetooth phone. This contains a C/C++ library, Java bindings, and a simple Java application
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The next generation of the Openh323 Gatekeeper project that provides high availability and scalibility for carrier grade, available freely under GNU GPL.
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Asterisk Manage API and FAGI C++ Framework. Implemented with high efficient OS independent Sockets wrapper (ACE). Provide C++ API wrappers for Asterisk Manager API and FAGI. A Manage API relay(proxy) server and a FAGI socket server will be implemented.
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CVM is the CAPI Voice Mail System, designed to replace vgetty for Linux answering machine services. It aims to provide an API similar to vgetty's, but instead works via CAPI 2.0. We hope to develop a product that runs more stable than vgetty/isdn4linux.
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The Java Computer Telephony Server (JCTServer) is a hardware independent telephony server for rapid development of Interactive Voice Response (IVR) applications in Java.
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This project intends to build a complete CallerID-function for home- or small buisniess environments. It depends on simple hardware which everyone can build. It will present the incomming call's number on every workstation on the network.
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Bayonne is supported from Savanna.gnu.org. Please visit there. Sourceforge is generally depreciated except for mailing lists, but updated releases will be posted here from time to time.
