Open Source Telephony Software - Page 2
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SIM Explorer
Sim Card explorer and decoder
This software uses your smart card reader/terminal to navigate through the SIM Card directory tree. The contents of the files are decoded and showed to the user. Other features, such as CHV management and SIM backup, are also available. -
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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 source code was released under a very flexible and liberal license, which gathers a large and active community around the project. You too can contribute to it no matter if you are a professional programmer or software development is just your hobby. Download YetiForce for free and find out why you should become one of more than 250 000 people who already use it. Testing versions: GitStable: https://gitstable.yetiforce.com/ GitDeveloper: https://gitdeveloper.yetiforce.com/ -
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008
Open-source event-driven AI powered Softphone
008 is an open-source event-driven AI powered WebRTC Softphone compatible with macOS, Windows, and Linux. It is also accessible on the web (though official support for browser-related issues is not provided). The name '008' or 'agent 008' reflects our ambition: beyond crafting the premier Open Source Softphone, we aim to introduce a programmable, event-driven AI agent. This agent utilizes embedded artificial intelligence models operating directly on the softphone, ensuring efficiency and reduced operational costs. This project is a WebRTC softphone, and communication is achieved via SIP over a socket. Leading PBX systems like Asterisk or Freeswitch support socket connections. If your provider does not offer this feature, consider using a SIP proxy such as Kamailio, Opensip or Routr. The softphone is internally configured using a JSON definition. The configuration file can be loaded from either a server or a local file. 008 reads the file only once. -
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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).
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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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International Telephone Input
Plugin for entering and validating international telephone numbers
A JavaScript plugin for entering and validating international telephone numbers. It adds a flag dropdown to any input, detects the user's country, displays a relevant placeholder and provides formatting/validation methods. Lots of initialization options for customization, as well as public methods for interaction. The user types their national number and the plugin gives you the full standardized international number. Navigate the country dropdown by typing a country's name, or using up/down keys. Automatically set the input placeholder to an example number for the selected country. Automatically select the user's current country using an IP lookup. Access the country data to create a separate country dropdown for an address form, and then listen for change events to keep the two dropdowns in sync. Use Twilio's API to build phone verification, SMS 2FA, appointment reminders, marketing notifications and so much more. -
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BlackBelt WASTE - ipv4/Tor/i2p +AI+Voice
Modern, AI-Smart, WASTE p2p for ipv4, Tor and i2p + Voice Conference.
Open Source - GPLv3 inc images. A WASTE client. Download and create your own WASTE networks. Move 1000's of GB's at 100MB+ per sec. (800 Mbits per sec) FULL pause and resume capable. Voice Conference, Chat, Transfer files and Participate in Forums in a secure environment. For Windows XP 32/64, Vista 32/64, Win7 32/64, Win8 32/64, Win 10, Win 11, Linux (WINE). *** User Based Access Control - for voice, chats, file transfers and uploads. (useful in NULLNETS) *** Distributed Autonomic-Performance-Tuning - A Goal-Seeking Swarming-Semiotic AI *** AI Connect - AI Managed Connections. *** Self-Organising Anti-Spoofing Technology *** Geared Multi-threading, providing the smoothest performance possible *** Advanced Threat Detect and Manage Technology *** Voice Conferencing Over WASTE *** RNN - Recurring Neural Net - AI Noise Reduction *** Differential Files Transfer - Seriously fast data backups -
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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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jphonelite
Java VoIP Softphone (SIP) (replaced by jfPhone)
jphonelite is a Java SIP VoIP SoftPhone for Desktops (Windows, Linux, Mac) and Android. Features 6 lines with transfer, hold, conference (up to all 6 lines), g711 u/a, g722, g729a, and video (video support in Linux or Windows only and includes H263/H264/VP8). Applet includes full JavaScript support. STUN/TURN/ICE supported. Encrypt media with SRTP. DTLS Key Exchange. Video conferencing with jPBXLite/0.12+ NOTE : THIS PROJECT HAS BEEN SUPERSEDED BY jfPhone. Please go to jfphone.sourceforge.net to get latest version. -
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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 Centers and Contact Centers for Quality monitoring (QM) purposes. -
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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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openLTE
An open source 3GPP LTE implementation.
OpenLTE is an open source implementation of the 3GPP LTE specifications. Currently, octave code is available for test and simulation of downlink transmit and receive functionality and uplink PRACH transmit and receive functionality. In addition, GNU Radio applications are available for downlink transmit and receive to and from a file, downlink receive using rtl-sdr, HackRF, or USRP B2X0, LTE I/Q file recording using rtl-sdr, HackRF, or USRP B2X0, and a simple eNodeB using USRP B2X0. The current focus is on extending the capabilities of the GNU Radio applications and adding capabilities to the simple base station application (LTE_fdd_enodeb). -
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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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A TAPI driver for SIP. SIPTAPI gives you a click2dial feature with any TAPI enabled application (e.g. MS Outlook) and SIP PBX/proxy. This is the "free" version of SIPTAPI. There is also an enhanced commercial version available at www.ipcom.at.
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Amazon Connect connect-rtc-js
Provide softphone support to AmazonConnect customers
connect-rtc.js provides softphone support to AmazonConnect customers when they choose to directly integrate with AmazonConnect API and not use the AmazonConnect web application. It implements Amazon Connect WebRTC signaling protocol and integrates with browser WebRTC APIs to provide a simple contact session interface that can seamlessly integrate with Amazon Connect StreamJS. In a typical amazon-connect-streams integration, connect-rtc-js is not required on parent page. Softphone call handling is done by embedded CCP. Load connect-rtc-js along with amazon-connect-streams on parent page. In the gh-pages branch prebuilt ready-to-use files can be downloaded/linked directly. -
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The Open VXI VoiceXML interpreter is a portable open source library that interprets the VoiceXML dialog markup language. It is designed to serve as a reference for parties interested in understanding how VoiceXML markup might be executed.
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mod_MumbleLink
Positional Audio Communication for Minecraft with Mumble
A Mod so that Minecraft now natively supports Mumble's positional audio feature. This means: Directional and positionally attenuated VOIP in relation to the game world. Please visit the Forum for information about the newest Version! Main Forum-Thread: http://www.minecraftforum.net/forums/mapping-and-modding/minecraft-mods/1272675 TheSkorm's Fork on GitHub: https://github.com/TheSkorm/mod_mumblelink Mumble: http://mumble.sourceforge.net Donations are greatly appreciated: https://sourceforge.net/donate/?user_id=2370023 -
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FMA is a SMS Manager, Mobile Phone Monitor, Remote Control Agent, Phonebook Manager, Organizer, Fun and much more; whatever you want it to be, it is whatever a mobile phone should have :-) (Currently based on Sony Ericsson features set).
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Vaak (Elastix-Persian)
http://www.vaakpbx.com
سيستم تلفني واك مبتني بر استريسك و محيط گرافيكي الستيكس، تنها سيستم تلفني ويپ فارسي دنيا . -
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TurnServer is a implementation of Traversal Using Relay around NAT (TURN) protocol. This protocol allows a client to obtain IP addresses and ports from such a relay.
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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.
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Modem Manager GUI
Frontend for ModemManager 0.6/0.7, Wader and oFono
Modem Manager GUI is a simple GTK based graphical interface compatible with Modem manager, Wader and oFono system services able to control EDGE/3G/4G broadband modem specific functions. You can check balance of your SIM card, send or receive SMS messages, control mobile traffic consuption and more using Modem Manager GUI. -
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jpbxlite
Java VoIP/SIP PBX system (replaced by jfPBX)
jPBXLite is a VoIP/SIP PBX. Supports SIP extensions, voicemail, trunks, conferences, queues (ACD) and an IVR system. Support video conferencing with jPhoneLite/1.4.0. NOTE:THIS PROJECT WAS RENAMED AND IS NOW jfPBX. Please go to jfpbx.sourceforge.net
