Telephony Software

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Telephony Software

Below is a listing of open source software having to do with Telephony. For a list of VoIP and business telephone providers please visit Voip-Info.org.

  • License This HTML5 Speed Test for Your ISP or Network License This HTML5 Speed Test for Your ISP or Network Icon
    License This HTML5 Speed Test for Your ISP or Network Icon

    Are you an ISP or network administrator looking for a reliable, accurate, & affordable HTML5 speed test that works on all devices?

    The SourceForge Speed Test measures Latency/Ping, Jitter, Download Speed, Upload Speed, Buffer Bloat, and Packet Loss. Upon completion, you can view detailed reports about your connection. This HTML5 speed test does not require Flash or Java, and works on all devices including tablets and smartphones. Host on your own infrastructure or use ours. For licensing, inquire today.
  • Searching for Affordable Business Phone Service? Searching for Affordable Business Phone Service? Icon
    Searching for Affordable Business Phone Service? Icon

    If you're on the hunt for business phone service that is reliable and budget-friendly, Voice over Internet Protocol (VoIP) is your answer.

    Today's small-to-medium-sized (SMB) businesses and large enterprises are saving on their monthly communications costs by making one simple decision: to switch to a VoIP service solution from their old, outdated Plain Old Telephone Service (POTS). By choosing a new VoIP service, these companies enjoy the flexibility, reliability, call features, and audio quality that only a VoIP service can provide. Plus, they cut their phone bill by up to 70%!
  • trixbox

    trixbox CE is an easy to install, VOIP phone system based on the Asterisk PBX. trixbox is designed for home or office use. trixbox CE includes CentOS linux, mysql, and all the tools needed to run a business quality phone system. (formerly asterisk@home)

  • Gammu Icon

    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

  • Elastix Icon

    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.

  • MCU Media Server Icon

    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. .

  • VoIP monitor Icon

    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.

  • Compare Cell Phone Plans at Wirefly Compare Cell Phone Plans at Wirefly Icon
    Compare Cell Phone Plans at Wirefly Icon

    You’re probably paying too much for cell phone service. Wirefly compares hundreds of plans to help you save.

    You’re probably paying too much for cell phone service. Wirefly compares hundreds of plans to help you save. Enter what you need (minutes, data, texts) into Wirefly’s innovative plan comparison tools and see your savings instantly.
  • Asterisk GUI client, VICIdial Icon

    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.

  • STUN Client and Server

    This project implements a simple STUN server and client on Windows, Linux, and Solaris. The STUN protocol (Simple Traversal of UDP through NATs) is described in the IETF RFC 3489, available at http://www.ietf.org/rfc/rfc3489.txt

  • Java SIP softphone Icon

    Java SIP softphone

    Peers is a very simple softphone. It's a SIP User-Agent, written in java, it works on windows, linux and mac. It can be used with SIP servers like opensips or asterisk IPBX. It supports G711 codec (PCMU and PCMA) and telephone-events (DTMF).

  • BlackBelt Privacy - Tor+WASTE+Video VoIP Icon

    BlackBelt Privacy - Tor+WASTE+Video VoIP

    Simple to install and use Tor client / server with WASTE and VideoVoIP

    *** PLEASE NOTE: There are now 2 seperate versions here. *** One is Pre Firefox 57. The other is Post Firefox 57. *** For those providing mirrors, please enable your users to realize this. MicroSip: enables FREE PC to PC video calling with no account sign-up and no middleman server. WASTE: enables chat, file transfer and support. Tor: enables safer browsing. Tor Firefox Profile: Browse over Tor. As with all versions of Tor - do not rely on this for strong anonymity. A usability enhanced Privacy Pack. An installer, for : Windows XP 32/64, Vista 32/64, Win7 32/64, Win8 32/64, Win10 32/64.

  • CyanogenMod 7 for Samsung Galaxy S

    CyanogenMod 7 for Samsung Galaxy S Series Phones Sources: http://github.com/CyanogenMod

  • anyremote Icon

    anyremote

    Remote control software for applications using Bluetooth or Wi-Fi.

    Remote control software for applications using Bluetooth or Wi-Fi. anyRemote supports wide range of modern cell phones line Nokia, SonyEricsson, Motorola, majority of Android phones and others.

  • Chan-SCCP channel driver for Asterisk Icon

    Chan-SCCP channel driver for Asterisk

    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.

  • NbuExplorer Icon

    NbuExplorer

    Nokia NBU, NBF, NFB, NFC and ARC backup file parser, extractor and viewer. It can help you to check content of backup or extract files from it. Requires MS .Net Framework 2

  • qtmoko Icon

    qtmoko

    QtMoko is distribution and software stack for Linux based SmartPhones

  • libsrtp

    The libSRTP project has been moved to https://github.com/cisco/libsrtp

  • t38modem

    What is t38modem? From your application view point it's a fax/voice modem pool. From IP network view point it's a H.323/SIP endpoint with T.38 fax support. From your view point it's a gateway between an application and IP network. Works with HylaFAX.

  • AppKonference

    AppKonference , a high-performance Asterisk conferencing module, is a fork of AppConference focused on voice.

  • OpenSIPS/OpenSER-a versatile SIP Server Icon

    OpenSIPS/OpenSER-a versatile SIP Server

    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

  • Kablink Icon

    Kablink

    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

  • The ASN.1 Compiler

    Go to github.com/vlm/asn1c for the latest version.

    This ASN.1 compiler turns ASN.1 specifications into C code. The asn1c is shipped together with conformant BER/DER/XER/PER codecs. The X.509, GSM TAP3, MEGACO, RRC and LDAP encoding and decoding examples are part of the source code distribution. NOTE: THE asn1c PROJECT HAS LARGELY MOVED TO GITHUB: http://github.com/vlm/asn1c

  • 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 Centers and Contact Centers for Quality monitoring (QM) purposes.

  • AstTapi

    AstTapi, an opensource Asterisk Tapi driver for windows. This allows users of TAPI compliant applications such as Outlook and Act to dial contacts directly from the application using an Asterisk PBX Server.

  • Disc-OS

    Vem ai o Disc-OS 3.0 Aguardem.. Modificações: Asterisk 14 ou Asterisk 15 Beta-1 Nova Interface UI Ubuntu 17.04 (Zesty Zapus) PHP 7.1 Disc-OS é uma distribuição de um PABX IP baseado em software livre. Desenvolvido para o mercado brasileiro com interfaces em português, de fácil instalação e configuração, contendo Linux customizado, software Asterisk 1.4 e configurador Disc.

  • MonAst :: The Asterisk Monitor

    Asterisk Monitor is a HTML interface that acts a operator pannel for asterisk to display user/peer status and calls. This uses a reverse AJAX, PHP and Python to originate, transfer and hangup calls, manage queues and meetme rooms.

  • Java SMPP API

    Java implementation of the SMPP API

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