Search Results for "call"
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Showing 37 open source projects for "call"
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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 automation - Asterisk dialplan utilities - Real-time agent and queue dashboards Visit https://vicistack.com for documentation, guides, and community resources. -
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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 SIP/RTP/RTCP/T.38/udptl protocols. VoIPmonitor can also decode audio. -
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VOIP-VOICE-TO-TEXT&ANALYS
Convert VoIP calls to text and analyze them with AI
The VoIP voice-to-text software for Issabel is an intelligent, AI-based solution that converts calls into accurate Persian text. After each call, the audio file is sent to the GPT-4O AI engine, producing editable transcripts. The software also provides AI-powered call analysis, extracting key points, customer requests, satisfaction levels, and sensitive topics, all stored in the database. This helps sales and support teams make faster decisions, improve response quality, and enhance customer experience. ... -
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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. -
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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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. VitalPBX will help you implement a secure... -
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pcapsipdump is libpcap-based SIP sniffer with per-call sorting capabilities. It writes SIP/RTP sessions to disk in a same format, as "tcpdump -w", but one file per SIP session (even if there is thousands of concurrent SIP sessions). Getting started: http://pcapsipdump.sf.net/
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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 and devices to support home care o Sensor Networks: sensor networks, particularly managing wind farms However, the approach is generic and extensible for managing other kinds of systems. ... -
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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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SkypeInYourLanguage
Skype language files for additonal languages (details below)
...You cannot run a version older than Skype 7 and even installing 7-8 is fiendishly complicated because it forces everyone into the app unless you run it in compatibility mode. Time to call it a day. I will back up the current files and then shut down the project at the end of the month. Please make your own backups from SourceForge because I till take that down too. If anyone wishes to take it on to handle just the Linux part, please give me a shout. Thank you for all your hard work over the years and really sorry to see this go but it's probably time to find an alternative to Skype that's localizable and OS. -
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GNU Gatekeeper (GnuGk)
H.323 Gatekeeper for VoIP and videconferencing
...Please find current versions at https://www.gnugk.org/ The GNU Gatekeeper (GnuGk) is a full featured H.323 gatekeeper under GPL license. It supports VoIP and videoconferencing and includes Radius and database support and many call routing functions. The project has moved! Please find current versions at https://www.gnugk.org/ -
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Asterisk CDR import
Simple web interface for mapping CDR from VoIP provider to Asterisk
This is a simple tool to map information for the cost of phone calls given by your VoIP provider, to the call log inside your Asterisk. As a result, you will know how much each of your employees has spent for phone calls. Additionally, this tool shows information about the assigned DID and CLIP numbers in Asterisk. This is helpful to identify obsolete, duplicate and multiple assignments. -
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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...
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AVAYA CMS API Server
CMS API Server with web site
... - Full Documentation - The server application, including administration GUI - Dashboard : a .NET consumer, used to test server and get REST URLs - CMS Operations (modify Dictionaries, Call Center Administration, etc) - Agent skill modification. - Basic security. -
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cdr-view
Webface witch extract CDR (Call Detail Records) from MySQL base.
Webface witch extract CDR (Call Detail Records ) directly from MySQL base. May be used as SOHO billing system. -
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cyber-wrt PBX
Cyber-wrt pbx is embedded system in DD-Wrt router
...It's first an original IPBX GUI with all features needed for company PBX. it include if you need a Hotel GUI offering what a Hotel PBX must do billing, check-in check-out, wake-up, housekeeping, direct call in room and many more. It include also a captive portal system for internet access controle using one lan port on your router. Try it, you will not belie -
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vTiger Freeswitch PBX Integration NYFON
vTiger Freeswitch PBX Integration
vTiger Freeswitch Integration by NYFON. Allows to perform outbound (Click to Call) and incoming (wip) calls from vTiger 6.1+ interface. Modified PBXManager allows to choose between Asterisk and Freeswitch for PBX integration. Same (and extended in the future) functionality as Asterisk interface. -
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OSDial
OpenSource Call Center and Dialer System
An Open Source (predictive) Dialer. OSDial is a full featured GPL dialer which does Manual, Power and Predictive dialing out of the box. Its easily installed through one of our .iso images. Install manual available at http://store.osdial.com/ Full details available at web site (http://osdial.com). -
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PHP 2 Way Webcam Video Chat
1 on 1 Webcam Videochat Script with P2P Support
...It's a solution for conducting easy to setup face to face meetings without leaving your office or home. It's the easiest and most cost-effective way to meet somebody and discuss one on one, to make a video call just by providing a private room access link. -
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VoIPGMap
Graph active VoIP calls on Google Maps.
A bash shell script which will read active call destinations in E.164 format from a source (such as Asterisk), then run the called number through some plain text and MySQL databases to find the best latitude/longitude match. It then creates a .html file with Google Maps API JavaScript code which will display a single call origination point and the active destinations (calls) on a Google Maps world map. -
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asterCRM is a call center software for asterisk based VoIP system, also it has some CRM functions. It provide useful features such as pop-up, predictive dialer, click to call, extension status .... astercrm could work with all asterisk based system.
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This project aim to provide a web based application to plan Conference Call using Asterisk application MeetMe(). The engine is based on Vaadin language to improve the portability of the application, Hibernate and some AGI script based on PHP.
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SIP Inspector is a tool written in JAVA to simulate different SIP messages and scenarios. You can create your own SIP signaling scenarios, customize SIP messages and monitor incoming and outgoing messages. The tool can play RTP streams from a pcap fi
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SEMS is a GPLv2+ media and application server for SIP based VoIP services. Basic applications like announcement, pre-call announcement, RBT, conference, voicemail, mailbox etc. and lots of example applications available.
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The web application offers to import call protocols of phone calls and lists of billed calls from different telephone providers. They can be matched to existing call protocol entries to generate cross-provider phone bills.
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Open Unified Recording (OUR) is a full featured Linux based Open Source VoIP/SIP Call Recording engine, indexing and retrieval system. The system resides on the network and passively captures SIP sessions. Project is sponsored by UnifiedRecording.com
