Search Results for "stable"
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56 programs for "stable" with 2 filters applied:
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Paligo is an end-to-end Component Content Management System (CCMS) solution for technical documentation, policies and procedures, knowledge management, and more.
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YesEvents offers a comprehensive suite of services that spans the entire conference lifecycle and ensures every detail is executed with precision. Our commitment to exceptional customer service extends beyond conventional boundaries, consistently exceeding expectations and enriching both organizer and attendee experiences.
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Open Phone Abstraction Library (OPAL) is a C++ multi-platform, multi-protocol library for Fax, Video & Voice over IP and other networks. Also included is the Portable Tool Library (PTLib) which is a C++ multi-platform abstraction library and collection o
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Kablink open team collaboration software uses social networking to unify team workspaces w/ real-time web conferencing. Collaboration for knowledge networking, program management, communities-of-practice, telework, other business process/functional areas
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Replacement for the SCCP channel driver in Asterisk. Extended features include Shared Lines, Presence / BLF, customizable Feature Buttons, and Custom Device State. Visit our discussion mailing list for help and join us as a developer if you like. The project moved to https://github.com/chan-sccp/chan-sccp
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Elastix
Unified Communications Server
Elastix is a software-based PBX powered by 3CX and based on Debian. An open-standards solution, Elastix is an easy to install and manage UC system compatible with popular IP phones, gateways and SIP trunks. Elastix is complete with unified communications features such as integrated WebRTC video conferencing, chat, presence and softphones and smartphone clients for Windows, Mac, iOS and Android. -
With Twilio Voice, you can build unique phone call experiences with one API, to create, receive, control and monitor calls with just a few lines of code. Create an engaging voice experience that you can quickly scale and modify with a wide array of customization options and resources.
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Mumble
Low-latency, high quality voice chat for gamers
Mumble is an open source, low-latency, high quality voice chat software primarily intended for use while gaming. It includes game linking, so voice from other players comes from the direction of their characters, and has echo cancellation so the sound from your loudspeakers won't be audible to other players. -
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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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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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cyber-wrt PBX
Cyber-wrt pbx is embedded system in DD-Wrt router
############################################### #--- CYBER-WRT by Christian Zeler / Comdif Telecom ---# #----------------- http://cyberhotel.comdif.com ------------------# ############################################## You can find 2015 news Cyber-hotel versions on http://cyberhotel.comdif.com including cyber-hotel on freepbx - Cyber-hotel decktop. . Cyber-wrt is a complet Asterisk IPBX and Hotspot system embedded in DD-Wrt router. Last version is build for ASUS RT-AC56U... -
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Voxy - the simplest way to integrate VoiceXML applications in Asterisk. Configure your dial plan with the URL of your VoiceXML application and it's done.
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Yooz provides the smartest, most powerful, and easiest-to-use cloud-based E-invoicing and Purchase-to-Pay automation solution. It delivers unmatched savings, speed, and security with affordable zero-risk subscriptions to more than 5,000 customers and 300,000 users worldwide.
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Sipp is a performance testing tool for the SIP protocol. Its main features are basic SIPStone scenarios, TCP/UDP transport, customizable (XML-based) scenarios, dynamic adjustment of call-rate and a comprehensive set of real-time statistics.
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One of the first secure telephony programs. Not limited to VOIP but can also directly encrypt a telephone line by using modems. Calls are encrypted using Diffie-Hellmann key exchange and several symmetric ciphers. Download is possible from Germany only, see homepage
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Voice over IP (VoIP) software specifically for ham radio operators. The primary purpose is to use the Internet to link Ham repeaters with a secondary purpose of providing remote access for users. Compatibility with EchoLink and IRLP is a goal.
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asterCRM is a call center software for asterisk based VoIP system, also it has some CRM functions. It provide useful features such as pop-up, predictive dialer, click to call, extension status .... astercrm could work with all asterisk based system.
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CafeSip - Look what Java and SIP can do
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. -
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Maintained version of old asterisk applications ported to 1.4 that for copyright reasons cannot be included into official Digium's releases and that for some reason the author is not keeping up-to-date. The project also require Zaptel instead of DAHDI.
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Mission Control is a Telepathy component implementing the Account Manager and Channel Dispatcher specifications. It stores IM account settings, brings accounts online as necessary, and hands communication channels to the relevant applications.
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OpenBTS
GSM L1-L3 stack with SIP network interfaces
OpenBTS is an implementation of the GSM air interface (Um) that allows cellular handsets to be used directly as SIP endpoints. It uses a software-defined radio to generate its air interface and uses Asterisk or yate as its network interface. -
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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.
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trixbox CE is an easy to install, VOIP phone system based on the Asterisk PBX. trixbox is designed for home or office use. trixbox CE includes CentOS linux, mysql, and all the tools needed to run a business quality phone system. (formerly asterisk@home)
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Telepathy-SofiaSIP is a SIP connection manager (protocol plugin) for the Telepathy framework (http://telepathy.freedesktop.org). It's based on the Sofia-SIP stack.
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VoIP's project with Speex codec, speech detecting and crypto coding with traffic counting. All code is a pure Java, so completely cross-platforms. Traffic is about 9 MB per hour in both ends, sources: http://www.open-source-soft.narod.ru/arrow.7z
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FreePBX is the most widely deployed Asterisk based PBX application with over 3 million direct and indirect downloads. It transforms your LAMPA stack into a powerful PBX. See http://www.freepbx.org for Forums, Docs, Bug Tracker, Project Site and more.