Search Results for "linux voice chat server"

FusionPBX is a multi-tenant PBX and voice switch for FreeSWITCH, a highly scalable, multi-threaded, cross-platform communication platform. FusionPBX can be used either as a single or domain based PBX, and offers several functionalities: as carrier grade switch, call center server, voicemail server, fax server, conference server, VoIP server, multi-tenant appliance framework and so much more. It also offers unlimited extensions, device provisioning, music on hold, call parking, automatic call...

Matrix is an ambitious new ecosystem for open federated Instant Messaging and VoIP. Everything in Matrix happens in a room. Rooms are distributed and do not exist on any single server. Rooms can be located using convenience aliases like #matrix:matrix.org or #test:localhost:8448. Synapse is currently in rapid development, but as of version 0.5 we believe it is sufficiently stable to be run as an internet-facing service for real usage! Create and manage fully distributed chat rooms with no...

Open Source - GPLv3 inc images. A WASTE client. Download and create your own WASTE networks. Move 1000's of GB's at 100MB+ per sec. (800 Mbits per sec) FULL pause and resume capable. Voice Conference, Chat, Transfer files and Participate in Forums in a secure environment. For Windows XP 32/64, Vista 32/64, Win7 32/64, Win8 32/64, Win 10, Win 11, Linux (WINE). *** User Based Access Control - for voice, chats, file transfers and uploads. (useful in NULLNETS) *** Distributed Autonomic-Performance-Tuning - A Goal-Seeking Swarming-Semiotic AI *** AI Connect - AI Managed Connections. *** Self-Organising Anti-Spoofing Technology *** Geared Multi-threading, providing the smoothest performance possible *** Advanced Threat Detect and Manage Technology *** Voice Conferencing Over WASTE *** RNN - Recurring Neural Net - AI Noise Reduction *** Differential Files Transfer - Seriously fast data backups

JavaComm is a client–server communication platform offering chat, telephony, screen sharing, conferencing, a media player, and configurable media streams. Users can define and manage their own streams for flexible playback. All server-based communication runs over HTTPS (SSL/TLS), while file transfers are handled directly between users without passing through the server. The JavaComm Server runs on any Jakarta EE Web Profile 10 platform with the required extensions, using Tomcat 10.1.57 as...

CalLite è il software per Call Center più versatile e veloce sul mercato. Ti permette di azzerare le perdite di tempo degli utenti (operatori, agenti, supervisori, coordinatori, ecc. ecc.) per massimizzare i guadagni. Il software dispone delle funzionalità: PowerDialer, ProgressiveDialer, PredictiveDialer ed AutoDialer. Ovunque siano (da casa o in ufficio), i tuoi operatori potranno lavorare con una semplice connessione internet. Tu potrai, da ogni dispositivo fisso o mobile, amministrare e...

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.

Enterprise telephony recording and retrieval system with web based user interface. The project currently supports recording voice from VoIP SIP, Cisco Skinny (aka SCCP), raw RTP and audio sound device and runs on multiple operating systems and database systems. It can record audio from most PBX and telephony systems such as BroadWorks, Metaswitch, Asterisk, FreeSwitch, OpenSIPS, Avaya, Nortel, Mitel, Siemens, Cisco Call Manager, Cosmocom, NEC, etc... It is amongst others being used in Call...

The BOT is used in our Discord server.

The Session Initiation Protocol (SIP) is a signaling communications protocol widely used nowadays for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks. A good way to design optimization techniques for SIP deployment would be to analyze SIP traffic from existing networks. However, publicly available analyses of SIP traffic are rare and thus not a lot of knowledge exists about typical behavior of a SIP server (as opposed to, for...

An open source software project delivering converged voice and data applications. The StarPound platform enables you to rapidly develop voice and data solutions using the power of business process modeling. More info: http://www.starpound.org.

Does not work because there is no master server! A P2P Messenger which uses JXTA. It provides secure RSA/AES end-to-end encryption and VoIP-Capabilities.

SIP UMS is an Unified Messaging Server for the SEMS/IVR. It is an extendable voicemai and IVR platform written in Perl. It includes a web based adminstration tool, voice prompts (in english) and sample scripts that will work with SEMS/IVR.

ComuniQ-Wholesale Telecom Billing System. It helps you manage all interconnect agreements for voice, sms and content. Its Next Generation Wholesale Billing System that takes care of all your retail and wholesale interconnect settlements requirements

Open UC Framework (openucf) is a unified communication server which supports telephony, presence, instant messaging and contact management.

SkypeXMPP is a Karaka™ fork aimed to provide Skype to XMPP gateway It supports presence, instant messaging, multi-user chat and voice (thanks to Freeswitch) This product uses the Skype™ API but is not endorsed or certified by Skype.

The project is a preconfigured VoIP PBX VM Image based on Asterisk. But provides more advance features, such as LDAP integration, VQ monitoring via RTCP XR reports, LDAP user login, integrated billing system and telephone directory PDF generator.

MurMan, the Murmur Manager, is a user friendly, robust, and attractive web management interface for Murmur servers based on PHP, using Murmur's (server component for the Open-Source voice communication Mumble) Ice RCP.

MumurWeb will be a comprehensive webinterface for Murmur servers based on PHP. Murmur is the server component for the Open-Source voice communication Mumble.

Spark plugin for integratión with CentricCRM. This plugin for Spark XMPP client complements the integration between Asterisk PBX, Centric CRM and XMPP Server. Automatically open session on Centric CRM and redirect to contact the page

AdminsParadise VoIP and Fax server offers a FREE Enterprise grade clustered phone and fax solution for your network for ultimate reliability. Features Web based faxing, scheduled conferencing, voice-mail, roaming users, and much more. Runs Asterisk 1

IM,WebIM and Sip voip tool. Client is Win32, Server is CentOS 5. support video,audio, file trans, sip call.

eTalk is a modular and skinnable communication software for Enlightenment. Its written in C, EFL and has support for instant messaging, voice and video plugins.

VVOIP is voice and video over IP is a P2P application without any protocol like SIP (no server is required) etc.. application. It use decent codecs for video and voice encrypted streams over IP. Moreover it is also comptiable with VoIP modems, you'd con

OpenIM is a instant messenger system utilising the XMPP (Internet-standard communication protocol) and is built on the Java platform. It supports many popular systems (MSN, ICQ, AIM, Jabber, Google Talk etc.) and supports Voice (and Video) Over IP.

UCE is H323 Telephony. UCE includes whole features of GSM phones and works with different architectures like PSTN. UCE provides voice commands,make it easy for blind users with sound server support.UCE made in TURKEY (Uraz-Cihad-Ersin) Atilim University