Search Results for "x86_64-linux-gnu"
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648 programs for "x86_64-linux-gnu" with 2 filters applied:
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Deploy Red Hat Ansible Automation Platform on Microsoft Azure for a strategic automation solution that allows you to orchestrate, govern and operationalize your Azure environment.
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The Pipeliner CRM experience is so different from traditional CRM which is why we drive the highest adoption rates, lower Total Cost of Ownership, and rapid Return On Investment. Plus with fast onboarding and how easy it is to learn to use and/or administer the system, there is little to no business interruption when deploying.
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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... -
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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... -
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Asterisk GUI client, VICIdial
VICIdial Contact Center Suite
This software suite is designed to extend the functionality of the Asterisk PBX through platform-independant web-client applications. Includes the VICIdial inbound/outbound contact center application. The suite is scalable across multiple Asterisk servers. -
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Kalkun is an open source web-based SMS (Short Message Service) manager. It uses Gammu SMSD (part of gammu family) as SMS gateway engine to deliver and retrieve messages from your phone/modem. Documentation is on the wiki: https://github.com/kalkun-sms/Kalkun/wiki Development and source code can be found on github: https://github.com/kalkun-sms/Kalkun/
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An all-in-one cloud-based management solution for a variety of health and fitness centers, Shapenet delivers a full suite of integrated technology solutions.
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IAXmodem is a software modem written in C that uses an IAX channel (commonly provided by an Asterisk PBX system) instead of a traditional phone line and uses a DSP library instead of DSP hardware chipsets.
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Open Phone Abstraction Library (OPAL) is a C++ multi-platform, multi-protocol library for Fax, Video & Voice over IP and other networks. Also included is the Portable Tool Library (PTLib) which is a C++ multi-platform abstraction library and collection o
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YetiForce CRM
The most affordable CRM system according to Capterra
Are you looking for a system to boost your sales? Do you want to improve your relationships with customers? Do you want to contribute to the fastest growing open source project on GitHub? You can use YetiForce completely free of charge. With YetiForce’s help you will manage 12 business processes in your company, no matter the size. The vast number of features available for free helped us win the first place in Capterra’s “Most Affordable CRM Software” ranking. YetiForce’s open... -
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... for Android devices NCIDdisplay - a homebrew client that displays on large LED modules NCIDpop - a popup client for Windows, Mac, and Linux
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Pyst consists of a set of interfaces and libraries to allow programming of Asterisk from python. The library currently supports AGI, AMI, and the parsing of Asterisk configuration files. The library also includes debugging facilities for AGI. 2014-04-17: Moved the version control to GIT. To check out see the tab "Code". Note that the whole history including ancient CVS, then some time in monotone, then subversion was united into one GIT repository thanks to ESR's...
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Our CPQ covers your entire sales-to-production process. Configure products, calculate pricing, send quotes to customers, secure and order, and generate engineering documents. All with Configure One Cloud.
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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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A C++ framework utilizing Design Patterns for creating Linux and Windows communications applications that contain Dialogic® products. Includes media and network classes (analog, digital, SIP, H323), multithreaded event handling, distributed app support.
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Kablink open team collaboration software uses social networking to unify team workspaces w/ real-time web conferencing. Collaboration for knowledge networking, program management, communities-of-practice, telework, other business process/functional areas
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AppKonference , a high-performance Asterisk conferencing module, is a fork of AppConference focused on voice.
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Replacement for the SCCP channel driver in Asterisk. Extended features include Shared Lines, Presence / BLF, customizable Feature Buttons, and Custom Device State. Visit our discussion mailing list for help and join us as a developer if you like. The project moved to https://github.com/chan-sccp/chan-sccp
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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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Generic JTAPI and Jain Jcc and Jcat is a framework for the rapid development of Java telephony spec. implementations. It does this by reducing the "service provider" coding requirements by an order of magnitude and by providing common framework.
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Gammu
Cellular manager for mobile phones/modems
Gammu is a cellular manager for mobile phones/modems. It contains libraries and functions for ringtones,logos,phonebook,SMS,etc. (used by external software), a command line version (with backup/restore) and SMS gateway (with MySQL and PostgreSQL supp -
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Siproxd is a proxy/masquerading daemon for the SIP protocol. It allows SIP clients (softphones & hardphones) to work behind an IP masquerading firewall or router.
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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 ... -
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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 https... -
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jfBroadcast
VoIP/SIP Autodialer
VoIP/SIP AutoDialer. Broadcasts a message with the option to transfer person to another number. Now includes new survey options. -
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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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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
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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...