Search Results for "can"
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Showing 55 open source projects for "can"
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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! Create and manage fully distributed chat rooms with no single points of control or failure. ... -
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JDA
Java wrapper for the popular chat & VOIP service
...After setting the token and other options via setters, the JDA Object is then created by calling the build() method. When build() returns, JDA might not have finished starting up. However, you can use await ready() on the JDA object to ensure that the entire cache is loaded before proceeding. -
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VoIP monitor
VoIP SIP and SKINNY quality analyzer and packet / audio recording tool
...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 SIP/RTP/RTCP/T.38/udptl protocols. VoIPmonitor can also decode audio. -
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ictagent
A WebRTC-Enabled Chrome Extension for Seamless Call Center
...Click 2 Call. Agent can access contents, contacts and communications from a single application(browser). It will make agent performance much better. It can harvest contacts from random pages / websites/ web applications. CRM Integration. Inbound and Outbound Calls. -
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5
ictdialer
SMS, Fax, Voice Broadcasting and auto dialer Software
...ICTDialer is built on top of renowned open-source technologies, including FreeSWITCH, ICTCore communications framework, and a PHP-based Angular framework. ICTDialer can be used in following faxing scenarios Voice Broadcasting Email to fax / web to fax / fax to email ATA support supporting both sending and recieving Fax over Fax machines using ATA. G.711 based Fax Origination / Termination / Gateway T.38 based Fax Origination / Termination PSTN/SS7 based Fax Origination / Termination A Single GUI is created to cover all the major communication methods and services like: Voice Broadcasting Campaigns Send Document with multiple files (optional) Fax to Email Extension Support DIDs -
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ICTFax
Open source Fax server software for inbound / outbound internet Faxing
...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. Below are Installation package link for custom projects and development work. Please Installation support services. -
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VitalPBX
Unified Communications System - PBX System
VitalPBX is a free telephone and communications PBX system for companies. It is a complete platform that can be installed on the physical hardware on the site or as a hosted application. VitalPBX acts as the upper layer interface for the Linux base and then Asterisk (one of the most popular communication toolkits in the world). For this reason, VitalPBX is the graphic user interface between you and the complex world of modern communications. -
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FusionPBX Live Medium
Project about FusionPBX on Live Medium
If you want to quickly set up a multi-tenant PBX on matter of few minutes, this is a suitable choice. - Demo environment. You can start to add extensions, gateways and dialplans, and quickly have a fully working PBX environment. - Import previous FusionPBX backup to Live environment. After this you will have a quickly working PBX on few minutes. - Install on hard disk. With this option you can install the environment to Hard Disk. Username and password for PostgreSQL is fusionpbx / fusionpbx . ... -
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SipTunnel is used to tunnel SIP UDP datagrams (and also RTP datagrams) through NAT using TCP protocol. It also can be used for many other purposes.
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Earn up to 16% annual interest with Nexo.Generate interest, borrow against your crypto, and trade a range of cryptocurrencies — all in one platform. Geographic restrictions, eligibility, and terms apply.
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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.
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Callflow Sequence Diagram Generator
Callflow Sequence Diagram Generator
The callflow sequence diagram generator is a collection of awk and shell scripts that will take a packet capture file that can be read by wireshark and produce a time sequence diagram. This is useful to view & debug SIP callflows or other network traffic -
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MumPI - Mumble PHP Interface
An administration web interface for Mumble VoIP servers in PHP
The Mumble PHP Interface (MumPI for short) allows you to manage your Mumble servers via a webinterface. Users can register, upload textures, and retrieve account data. Admins can manage virtual servers, registrations, admin accounts, online users… -
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PBXlab adds value to business thanks to its valuable contribution of technological solutions. PBXlab promotes the use of free software programs that can be tailored to the real needs of its users, speed of movement between users, at a very low cost or no cost. PBXlab offers development of add-ons and improvements to programs, leading to more full, safe and adapted to customer needs programs. We offer technological independence without having to develop products from scratch. ...
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VoIP Honey project provides a set of tools for building an entire honeynet, thus includes honeywall and honeypot emulating VoIP environments such as Asterisk PBX or OpenSer with fully configurable connections. Voip Honey runs on GNU/Linux and Windows Systems. It can be compiled for Mac OSX as well.
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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 with mobile browsers) to browse reports, perform CMS Operations and Agent Skill modification from any device having a browser... -
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NoiseGator (Noise Gate)
A simple noise gate app intended for use with VOIPs like Skype.
...Mac users can use SoundFlower. -
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Eternity smdr
perl based SMDR/CDR capture & display on Web based PHP with mysql DB
You can Capture Matrix Comsec' Eternity/NAVAN product Online SMDR/CDR using Perl Script. Store it in MYSQL Database and display it by using filter in PHP -
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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 router It is working on Debian with asterisk pbx and original captive portal system to offer a reliable Hotspot system. ... -
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baresip
Baresip is a modular SIP User-Agent with audio and video support
Baresip is a portable and modular SIP User-Agent with audio and video support. the latest source code can be found here: https://github.com/alfredh/baresip -
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Artemisa is a honeypot for VoIP (SIP) networks. It is designed to connect to a VoIP domain as a user-agent backend in order to detect malicious activity at an early stage and also adjust the policies of the enterprise in real-time.
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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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ictbilling
A client management, billing and e-commerce solution
ICTBilling contains a set of Drupal 7 modules that can be used with Ubercart E-commerce solution and ICTBroadcast, a unified communication auto dialer software. This Client and Billing solution for ITSP has following modules and rules: client module, ictservice module, ratelist module and rules for ubercart. Client module handles clients user interfaces, subscribed service details, service links, list of available addons and current balance. -
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OZEKI VoIP Service Checker
The OZEKI VoIP Service Checker allows you to check VoIP services.
The Ozeki VoIP Service Checker is a tool you can use for checking the availability of VoIP services. It can be used to verify if a give service exists, whether it can be reached from a particular network and it checks if the user details, such as the username and password to access that service is correct. This tool was developed because many mobile Internet providers block VoIP traffic and no other tool exists that makes it possible to quickly verify if a certain network support VoIP. ... -
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RightClickLync
Adds right click context menu to call a highlighted number in Lync
Adds a Right Click context menu to your browser to call number from Lync. Highlight a piece of text in IE/FF13/Chrome and the script will strip out non numerics and pass to Lync via tel: handler and Lync can normalise the number. -
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ACR - Asterisk CDR Reporter
Information on Asterisk cdr like ASR, ACD, Billing duration 'billsec',
Information on Asterisk cdr like ASR, ACD, Billing duration 'billsec', time criteria not only date, I will add more features once I have time, like channels usage and time zone shift. you are welcome to suggest ! Please feel free to download, install and/or modify the code and let me know your feedback. You can use the following CLI commands to download and install the script: cd /var/www/html wget http://citylan.dl.sourceforge.net/project/acrasteriskcdrr/rpt.tar.gz tar xzvf rpt.tar.gz then point your internet browser to http://your-ip-address/cdr.htm