Search Results for "server voip"
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Showing 63 open source projects for "server voip"
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BlackBelt Privacy Tor/i2p+WASTE+VidVoIP
Browse on Tor/i2p, Anon p2p Chat / FileTx, Conf / Video VoIP
.... *** For those providing mirrors, please enable your users to realize this. Vidalia Based, Tor as a Service Solution. MicroSip: enables FREE PC to PC video calling with no account sign-up and no middleman server. WASTE: enables FREE Conference VoIP, chat, file transfer and support. *** AI Powered *** Tor/i2p: enables safer browsing. Tor/i2p Profile: Browse over Tor/i2p on Firefox (Vanilla, ESR), Waterfox (Classic, New), Palemoon , LibreWolf and legacy CyberFox. As with all versions of Tor - do not rely on this for strong anonymity. ... -
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ictagent
A WebRTC-Enabled Chrome Extension for Seamless Call Center
ICTAgent is a advance webphone offered as chrome extesion and when installed it will convert all available contacts as click to dial. This extension will embed WebRTC based VoIP Phone in browser which will remain connected to the main server, and will be responsible for inbound and outbound calls, transfer calls, sending DTMF as well sending Fax. It can harvest contacts from the random pages. Agent can access contents, contacts from a single application. Browser can automatically fetch required URL depending on the course of call. ... -
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queXS
Web based system for Computer Assisted Telephone Interviewing (CATI)
queXS is a web based, Open Source, CATI (Computer Assisted Telephone Interviewing) System. queXS integrates with queXML for creating questionnaires, LimeSurvey for collecting data and Asterisk for VoIP telephony. -
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ViaVoip
A portable peer to peer voice-chat/walkie-talkie.
ViaVoip is a simple Voice Over IP application that can be used when you need to talk, chat, or send files through the internet, but you can't or don't want to make use of any third party services. Its peer to peer design allows the two end points to connect directly to each other, without any central server nor account registration. It runs on Windows, Linux, Mac OS X and Android, and is portable, that is you don't need any setup, just get a copy and run it from any storage... -
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5
CalLite CRM
Software per Call Center
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... -
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cerberuscms2
Cerberus Content Management System
Cerberus Content Management System is a dynamic, secure and infinitely expandable CMS designed after a Unix-Like model. It is a custom written Web Application Framework ( W.A.F. ) with a consistent and custom written Pre-Hyper-Text-Post-Processor Programming Code Framework ( P.C.F. ). This Web Application Software Project' aim is to be the fastest and most secure Web Application Framework, Web Application Programming Code Framework, Text, Voice and Video Communications Platform and Content... -
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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. -
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Stuntman - STUN server and client
High performance, production quality STUN server and client library
New version 1.2. This is the code to STUNTMAN - an open source STUN server and client code by john selbie. Compliant with the latest RFCs including 5389, 5769, and 5780. Also includes backwards compatibility for RFC 3489. ICE and WebRTC ready. Version 1.2 compiles on Linux, MacOS, BSD, and Solaris. Supports the STUN protocol on both UDP and TCP for both IPv4 and IPv6. Windows binaries are also provided. Additional features are in development. This is a mirror of the code on... -
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MagnusBilling
Free billing to Asterisk
MagnusBilling is a VoIP server system that brings together the best IP Billing Telephony software in the market, creating a comprehensive, flexible and superior tool. Enterprises that need the freedom to build their own VoIP server. You will be able to chose your termination, giving your business larger flexibility and autonomy. Many VoIP distributors use the best of each VoIP server, termination to be able to be more competitive and to have better price and quality. ... -
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10
Accent Tool Suite
Accent interfaces and controls systems through goals and policies
Accent (Advanced Component Control Enhancing Network Technologies) is a comprehensive tool suite that interfaces a variety of communications systems and allows these systems to be controlled through goals (high-level user aims) and policies (lower-level system rules). Accent has been applied to the domains of: o Call Control: for telecommunications, particularly call control in Internet telephony o Home Care: for home automation and telecare, particularly for domestic appliances... -
11
oreka
Enterprise telephony recording and retrieval system
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... -
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The OSP Toolkit is a client side implementation of the ETSI OSP VoIP Peering protocol (ETSI TS 101 321). The OSP Toolkit project was begun in 1998 and the code has been incorporated into many commercial and open source VoIP products.
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AVAYA CMS API Server
CMS API Server with web site
Free version of AVREMO.CMS.API exposing REST Services. With this software you can browse for all CMS reports. Design a report with AVAYA CMS Supervisor designer, application automatically expose data as XML or JSON. You can also browse reports with your preferred browser (desktop or mobile) and execute CMS operations like modify dictionaries, modify agent skills, modify VDN/Vector associations and more. Application package include : - A responsive WEB server interface (compatible... -
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Mobicents is the leading Open Source VoIP Platform. It is the First and Only Open Source Certified implementation of JSLEE 1.1 (JSR 240), and SIP Servlets 1.1 (JSR 289). Mobicents also includes a powerful and extensible Media Server.
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2N VoiceBlue SMS Gateway
2n SMS Server
Full featured HTTP SMS gateway for 2n VoiceBlue GSM VoIP adapters with sending / receivig capabilities, XML outputs, POST/GET controls and a basic WEB frontend. -
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Git repo: https://github.com/asipto/siremis Web management interface for Kamailio (OpenSER) - handle subscriber profiles, access control lists, accounting, least cost routing and load balancing, monitoring charts, xmlrpc communication with SIP server
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Teamspeak 3 Love Plugin
A plugin for Teamspeak3. This plugin allows you to autofollow a user.
A plugin for Teamspeak3. This plugin allows you to follow a user while he switches through channels. For the love menu just right click any name in the server view. -
18
mumb1e
A Murmur-Server Webinterface written in PHP for Mumble
Administration Tool for Mumble -/- Murmur Servers. Mumb1e Admin Plugin, also named "MAP" is a powerful Murmur web interface application written with PHP that can create useres, edit themself and a lot more for Mumble server. MAP is developed over m -
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F-Talk P2P Encrypted Secure Voip
F-Talk cross-platform p2p encrypted secure voip software
This project connect to two peers directly ( using without any central server, stun server ) together with 128bit encrypted secure voip communication. http://ftalk.armanasci.com/guide Spefications * Peer to peer direct connection without using any central dedicated server. ( First basic part of the security ) * 128bit AES Symmetric end-to-end encryption PCM datas. ( Second basic part of the security ) * 64bit Base64 Encoding transferred server side info... -
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KAMAILIO (OpenSER) - robust, secure and scalable Open Source (GPL) SIP (RFC3261) server implementation with large features set (over 90 extension modules). As of May 2009, source code is hosted by GIT repository at http://sip-router.org
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Turn.Server is .NET TURN (Traversal Using Relay NAT) server written in C#. Compatible with [MS-TURN] extension and clients based on Microsoft UCCAPI SDK.
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Hanasu
A P2P Messenger which uses JXTA. It provides secure RSA/AES end-to-end
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. -
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ChatAnt
client-server full java chat audio-video file sharing
client-server full-java chat, provided with audio&video (JMF), custom rooms, custom forums, instant- messaging and file sharing -
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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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This is a P2P VoIP system. It support audio and video chat through kademlia-based dht overlay. The p2p module use Kademlia protocols; Audio codec is G.729. An simple Introduction here: http://www.needdo.com/2007/09/ppphonekademliap2p-voip/ . My Personal blog is http://www.idolf.cn and my email is dolfcao@gmail.com .
