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Atera - an All-in-one platform for IT management
Ideal for IT departments and MSPs (managed service providers)
Your IT essentials, integrated & elevated. Take your IT management from automated to autonomous, download Atera's agent to start your free trial!
A suite of open-source tools and frameworks for creating SIP apps
Session Initialtion Protocol (SIP) is widely used for telephone services over the Internet. CafeSip provides a suite of open-source tools and applications for creating customized SIP services and applications using the Java.
Give your IT, operations, and business teams the ability to deliver exceptional services—without the complexity.
Freshservice is an intuitive, AI-powered platform that helps IT, operations, and business teams deliver exceptional service without the usual complexity. Automate repetitive tasks, resolve issues faster, and provide seamless support across the organization. From managing incidents and assets to driving smarter decisions, Freshservice makes it easy to stay efficient and scale with confidence.
Simple testing tool to generate RTP data packets and send it via netwok interface or save into pcap file. Primarily intended for use with SIPp application to test speech quality with different codecs.
Lwazi is a robust telephony platform aiming to facilitate speedy development of experimental applications without sacrificing power by combining Asterisk with the MobilIVR Python interface bundled into a single build with a unified control interface.
An Asterisk configuration/dial plan manager written Object Pascal (FreePascal/Freevision). Supports multiple trunks and call groups. This is a multi-Tenant system requiring very little configuration to have a fully working PBX system.
VoIPER is a VoIP security testing toolkit incorporating several VoIP fuzzers and auxilliary tools to assist the auditor. It can currently generate over 200,000 SIP tests and H.323/IAX modules are in development. It's also a damn cool project name ;)
IHU is a VoIP application for Linux (using Qt and Speex), with low latency, crypted stream, minimal use of bandwith, and without intermediary servers. It is the easiest way to talk real-time with your friends (like phone) on the internet or LAN.
Transform your applications and workflows into powerful agentic systems at global scale.
Gemini Enterprise Agent Platform lets you rapidly build, scale, govern and optimize production-ready agents grounded in your organization's data. The platform enables developers to build custom or pre-built agents for virtually any use case. New customers get $300 in free credits.
CONFIANCE stands for CONFerencing IMS-enabled Architecture for Next-generation Communication Experience: an implementation of the IETF XCON (Centralized Conferencing) framework and of the BFCP (Binary Floor Control Protocol)
This is a simple software, comprised of a server and a client, that can be used together with any IP-phone using the SIP protocol (TELE2 and Bredbandsbolget in sweden). It also retrives information, sush as name, adress, etc about the number from eniro.s
Send and receive faxes using spandsp library, record, playback and connect calls (conference) with this commandline linux application for capi the CAPI 2.0 interface. Also runs on Embedded Systems (e.g. AR7 or German FritzBox).
Tipic Inc. is proud to announce the born of the first Open Source, P2P, VoIP and Video conferencing solution for XMPP/Jabber. The target of this project is to develop a new portable set of codes using high quality libraries to add VoIP and Video to XMPP
Phone is a simple IP telephony program allowing voice communication over IP network.
Phone is written in C and can be built on UNIX/Linux and MS Windows platforms.
Speech profile of Person is created that contains elementary sounds uttered. Profile is 1 time download for listeners. The actual audio sample is encoded based on profile. Decode using the profile stored earlier by User, and the audio can be regenerated.
Simple commandline IP telephonly application for Linux, that doesn't conform to either H.323 or SIP, doesn't use RTP and is not interoperable with any other program. It does work well though, and has quite good audio quality.
UVOIP is a privacy-enhancing technology for unobservable voice communication between two parties on the Internet. M.Knezevic, V.Velichkov, "Demonstration of Unobservable Voice Over IP", ADAMUS'08
JabbahVoice is an open-source framework which allows developers to easily create and extend voice portals using Asterisk (www.asterisk.org), the famous open-source IP PBX.
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!