Search Results for "VOIP SERVER"
Sort By:
Showing 105 open source projects for "VOIP SERVER"
View related business solutions-
Our Free Plans just got better! | Auth0You asked, we delivered! Auth0 is excited to expand our Free and Paid plans to include more options so you can focus on building, deploying, and scaling applications without having to worry about your security. Auth0 now, thank yourself later.
-
$300 Free Credits for Your Google Cloud ProjectsLaunch your next project with $300 in free Google Cloud credits—no strings attached. Test, build, and deploy without risk. Use your credits across the entire Google Cloud platform to find what works best for your needs. After your credits are used, continue with always-free tier services. Only pay when you're ready to scale. Sign up in minutes and start exploring.
-
1
VoIP monitor
VoIP SIP and SKINNY quality analyzer and packet / audio recording tool
VoIPmonitor is open source network packet sniffer with commercial frontend for SIP SKINNY MGCP RTP and RTCP VoIP protocols running on linux. VoIPmonitor is designed to analyze quality of VoIP call based on network parameters - delay variation and packet loss according to ITU-T G.107 E-model which predicts quality on MOS scale. Calls with all relevant statistics are saved to MySQL or ODBC database. Optionally each call can be saved to pcap file with either only SIP / SKINNY protocol or... -
2
Synapse
Matrix reference homeserver
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! -
3
FusionPBX
A domain based multi-tenant PBX & voice switch for FreeSWITCH
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 distribution and many other superb features. Give your company the business functionality and corporate level phone system it needs with FusionPBX! -
4
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. ... -
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.
-
5
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. ... -
6
ViciStack - VICIdial Call Center Tools
VICIdial call center optimization tools, monitoring utilities, and per
VICIdial call center optimization tools, scripts, and monitoring utilities for system administrators and call center managers. Includes performance tuning guides, AMD (Answering Machine Detection) analysis tools, compliance automation scripts, and real-time monitoring dashboards for VICIdial-based contact centers running on Asterisk PBX. Key features: - Server performance monitoring and alerting - AMD accuracy tuning and analysis - Campaign optimization scripts - Compliance and DNC... -
7
ICTFax
Open source Fax server software for inbound / outbound internet Faxing
Note: for binaries and installation instruction please visit official website. http://ictfax.org/ ICTFax is Open Source Fax Server software to send and receive Fax over Internet & Emails developed over Freeswitch, ICTCore and Angular Framework , It works with T.38 and G.711 pass through and feature email to fax, fax to email, web to fax, fax to web and also support fax machines to send and recieve Fax using ATA. Installation Guide https://www.ictfax.org/ictfax-installation-guide/ Administration Guide https://www.ictfax.org/ictfax-admin-guide/ User Guide https://www.ictfax.org/ictfax-user-guide/ Commercial Support If you do not posses Linux / VoIP expertise to install and setup ICTFax or you are busy, We can help you in installation, configuration and testing to make sure ICTFax setup is functional and tested. ... -
8
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. -
9
ictcore
ICTCore: Unified Communications Framework for web developers
ICTCore is an open-source Communications Platform as a Service (CPaaS) designed to empower developers and system integrators to build, deploy, and manage communication-enabled applications with ease. With support for voice, SMS, email, and fax channels, ICTCore provides a programmable communication layer that enables rapid development of ICT-based solutions using standard development skills.Following are few projects developed over ICTCore communications framework ICTFax open source fax... -
Compliant and Reliable File Transfers Backed by Top Security CertificationsStop relying on non-certified, legacy file transfer tools that creak under the weight of modern security demands. Get full audit trails, advanced access controls and more supported by an award-winning team of experts. Start your free 25-day trial today.
-
10
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... -
11
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... -
12
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... -
13
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. -
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... -
15
Kazoo Live Medium
Kazoo Live Medium. From Null to a Full Featured PBX on minutes
From Null to a Full Featured PBX in matter of minutes!! If you want: - To boot or install quickly Kazoo on your server. - To test about how it works, without installing on your hard drive. - To install on cluster environment - To install if your previous platform go downtime. This is a suitable option. What is included? Monster-UI Open Source Apps, like SmartPBX, CallFlows, PBX Connector, Voicemails, Faxes, Accounts and Number Manager Kazoo v4.3.x Kamailio ... -
16
OpenSIPS (former OpenSER) is an GPL implementation of a multi-functionality SIP Server that targets to deliver a high-level technical solution (performance, security and quality) to be used in professional SIP server platforms. IMPORTANT: this is no longer the main hosting for the project. This was moved on GITHUB - https://github.com/OpenSIPS/opensips
-
17
mumsys
Ein Mumble-Server-Webinterface, das in PHP geschrieben wurd
Mumble Admin Pulgin optimiert für php 7.x.x und ZeroC-Ice 3.7 Ursprünglich wurde das Webinterface von Michael Koch Geschrieben der das Projekt am 23.05.2016 Stilllegte. Features create, edit, delete Server User create, edit, delete virtual Murmur/Mumble Server edit Murmur Configs show Channels, edit ACL start, stop, restart virtual Murmur/Mumble Server Channelviewer, internal and external Serverhosting Fully accessable Pluginusers with wide... -
18
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. ... -
19
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... -
20
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... -
21
SIPSL (SIP Service Layer) is a Programmable Session Border Controller (SBC) and SIP B2BUA, as defined in RFC3261. Written in c++ and multithreaded. The Application logic can be implemented by extending the application class and implementing the call backs.
-
22
MCU Media Server
SIP Video Multiconference Media Server with WebRTC support.
REPOSITORY MOVED TO GITHUB!! https://github.com/medooze/media-server Video Multiconference Media Server with WebRTC support. Provide Multiconference and video broadcasting services to any SIP service. Supports VP8, H264, MP4V-ES, H263 and H263P, continuous presence, RTMP flash broadcasting, adhoc conferences, load balancing and administrative WEB interface. JSR309 driver implementation under development. . -
23
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.
-
24
RADIUS to OSP
Connverts RADIUS acccounting messages to ETSI OSP CDRs
The RADIUS to OSP project is a module for the FreeRADIUS server which converts RADIUS accounting records into ETSI OSP Usage Indication messages. -
25
VoiceOne gives you the ability to install and compleatly configure a pbx platform based on Asterisk 1.8 with an easy to use web GUI, which would be a framework to build a communication server adding various plugins.
