Search Results for "VOIP SERVER"
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Showing 52 open source projects for "VOIP SERVER"
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Coturn TURN server
coturn TURN server project
coturn is a free open source implementation of TURN and STUN Server. The TURN Server is a VoIP media traffic NAT traversal server and gateway. -
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Sippts
Set of tools to audit SIP based VoIP Systems
Sippts is a set of tools to audit VoIP servers and devices using SIP protocol. Sippts is programmed in Python and it allows us to check the security of a VoIP server using SIP protocol. You can freely use, modify and distribute. If modified, please put a reference to this site. Most security tools can be used for illegal purposes, but the purpose of this tool is to check the security of your own servers and not to use to do bad things. -
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Routr
The future of programmable SIP servers
Lightweight sip proxy, location server, and registrar that provides a reliable and scalable SIP infrastructure for telephony carriers, communication service providers, and integrators. Routr provides all the tools required to deploy your VoIP network, including a command-line for remote server control. It can also be controlled via a RESTful API or a web-based GUI. Routr includes all tools for deploying your VoIP network. -
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nheko
Desktop client for Matrix using Qt and C++20
Desktop client for Matrix using Qt and C++20. The motivation behind the project is to provide a native desktop app for Matrix that feels more like a mainstream chat app (Element, Telegram etc) and less like an IRC client. -
MongoDB Atlas runs apps anywhereMongoDB Atlas gives you the freedom to build and run modern applications anywhere—across AWS, Azure, and Google Cloud. With global availability in over 115 regions, Atlas lets you deploy close to your users, meet compliance needs, and scale with confidence across any geography.
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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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Mumble PushToTalk Button
Simple GUI app for LINUX with just one clickable "PushToTalk" button
...That community has been using mumble extensively, and mumble serves as the "social hub" of LINUX Air Combat. It is commonplace for LAC users to fire up mumble, connect to the mumble server at linuxaircombat.com, and chat to arrange flights, missions, and teams. This is much better done when "PushToTalk" is available from the desktop. -
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Cerberus Content Management System 6
Cerberus Content Management System
Cerberus Content Management System is a dynamic, secure and infinitely expandable CMS designed after a Unix-Like model complete with a Unix-Like Kernel File named: Cerberus. 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,... -
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CerberusCMS5
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... -
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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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... -
11
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... -
12
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. -
13
jfSTUN
STUN/TURN Server
STUN/TURN Server. Supports enough for VoIP Phones such as jfPhone. -
14
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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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... -
18
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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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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Phonebook directory for Cisco VoIP
Phone directory for Cisco with LDAP or MySQL integration
Multiple scripts when you want to have a corporate directory on your Cisco VOIP phone. 2 are for LDAP and were tested by storing the addresses using Baikal. In one script all numbers are displayed flat, where in the other script its working with submenus. The third script is for MySQL where you can create your own database and store multiple numbers (i.e. mobile,fax,home) for each person. Installation is pretty easy and included in readme. -
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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...
