Open Source Telephony Software - Page 4

Zanzibar is a complete, standards based IVR. It includes an MRCPv2 Server with ASR and TTS engines as well as an voiceXML interpreter so that you can deploy and run voiceXML applications. It integrates with VOIP PBX’s (like Asterisk) using SIP and RTP.

A lightweight cross platform IP telephony client using the IAX protocol, designed for use with the asterisk open source PBX.

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

Das Fritz!Box Telefon-dingsbums ist ein Outlook-Addin, welches ein direktes Wählen der Kontakte aus dem Computer ermöglicht. Zusätzlich bietet es nützliche Funktionen, wie einen Anrufmonitor oder Rückwärtssuche.

MyPhone is Voice&Video IP Telephony Chat client. It is H323 protocol compatible, based on OpenH323 project (www.openh323.org). Client is written in VC++, using MFC/PWLib libraries, compatible with any reasonable M$Windows version. Freeware&OpenSource

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…

GNU Telephony intends anyone to use free as in freedom software for telephony, and to do real-time collaboration freely and privately over the Internet, with the freedom to do so on any platform they choose to use.

A fully perl written sofware for windows (also stand alone binary code included, so it does not require perl installation to use it) to receive short messages (SMS) sent by mobilephone.

TALK provides an efficient and scalable architecture to make CTI more easy and web oriented. TALK includes a generic JTAPI connector, a full AJAX client, a Gadget, and a library to allows your web applications to be connected to your PBX.

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).

SipLine is a modern, lightweight SIP softphone for Windows 10/11 built with .NET 9 WPF. It provides enterprise-grade VoIP communication with a focus on performance, security, and extensibility. Features Ultra-Fast: Startup in <0.5s, minimal CPU usage HD Audio: Opus and G.711 codecs with adaptive jitter buffer Security: TLS transport, SDES-SRTP encryption, Windows DPAPI credential protection Multi-Account: Up to 5 simultaneous SIP accounts Plugin SDK: C# SDK for custom integrations (CRM, call recording, automation) Plugin Marketplace: Community and premium extensions Headset Support: Jabra, Poly, Sennheiser, Logitech with HID controls Call Quality: Real-time MOS Score, Jitter, Packet Loss and RTT monitoring Enterprise: Silent MSI installer, GPO support, JSON centralized config Compatibility: 3CX, Asterisk, FreePBX, OVH, Twilio, RingCentral, and more Language packages : https://sipline.feelautom.fr/languages

PLEASE NOTE that all current AstLinux development is on GitHub: https://github.com/astlinux-project/astlinux Thanks to SourceForge for providing free hosting to the AstLinux Project for more than 10 years! Old source archive remains on SourceForge. AstLinux is a custom Linux distro centered around Asterisk, the Open Source PBX. AstLinux has many features that make it ideal for embedded and commercial Asterisk based solutions.

The project has moved! 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/

SNEP is an graphical user interface to manage Asterisk PBX. It provides high level control over routing and users administration making the PBX administration easier to non-technical users. Access our Chat community at Slack: http://snep.slack.com

Modular Debian based PBX distribution. Originally started by Federico Pereira, as DebPBX, but then joined some ex-Elastix guys.

client-server full-java chat, provided with audio&video (JMF), custom rooms, custom forums, instant- messaging and file sharing

With SIP Proxy you will have the opportunity to eavesdrop and manipulate SIP traffic. Furthermore, predefined security test cases can be executed to find weak spots in VoIP devices. Security analysts can add and execute custom test cases.

Manages phonebook and SMS in your mobile phone (cellular) from your own computer. Allows a user to edit and retrieve phonebook in SIM card or memory, send SMS or group SMS and view your phone configurations. Supports all kind of phones; supports unicode.

J2ME Polish is a next generation framework for the rapid development of J2ME applications. Features included are a powerful GUI framework, utility classes and ant-based build tools for the creation of device optimised applications.

FreePBX (formerly Asterisk Management Portal) is a project to bring together best-of-breed applications to produce a standardized implementation of Asterisk complete with web-based administrative interface.

Project moved to github: https://github.com/vloschiavo/cipit

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 reposurgeon. 2013-05-29: Maintainers of github fork "pyst2" contacted to join forces. Note that the last release here fixes the same bug as on github, otherwise github version seems to only contain cosmetic changes and examples. 2012-07-06: An offer to maintain the project has been received. 2012-01-29: There have been two or three forks that are being maintained. This is now just history unless someone cares to take up maintenance here.

UCE is H323 Telephony. UCE includes whole features of GSM phones and works with different architectures like PSTN. UCE provides voice commands,make it easy for blind users with sound server support.UCE made in TURKEY (Uraz-Cihad-Ersin) Atilim University