RSS 150 projects tagged "Telephony"

Download Website Updated 17 Mar 2014 Yet Another Telephony Engine

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Pop 869.31
Vit 57.58

Yate is a next-generation telephony engine. While currently focused on Voice over Internet Protocol (VoIP) and PSTN, its power lies in its ability to be easily extended. It supports SIP, H.323, IAX, MGCP, Jingle, Jabber, E1, T1, analogic, robbed bit, ISDN PRI, BRI, and SS7. YateClient is an Instant Messenger and Voice application for Jabber/Jingle IM, SIP, H.323, and IAX VoIP protocols.

No download No website Updated 22 Sep 2012 OTRS

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Pop 803.94
Vit 12.52

OTRS is a platform independent Web-based help desk system that supports service organization of any kind (e.g. IT service, customer and technical product service, complaint management, public services, etc.) to increase their efficiency. It increases transparency as well as service quality and lowers your total cost of ownership. It has been certified ITIL V3 compatible by PinkVERIFY for incident, problem, change, service asset and configuration, request fulfillment, and knowledge management. Other ITIL processes like service catalog and service level management are supported as well.

Download Website Updated 02 Feb 2010 Asterisk

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Pop 672.61
Vit 8.04

Asterisk is a hybrid TDM and packet voice PBX (Private Branch eXchange) and IVR platform with ACD functionality. It acts as middleware between the Internet (IAX, SIP, MGCP, Skinny, H.323), telephony channels (like Zaptel, T1, PRI, E1, FXO, FXS, VoIP, VoFR, ISDN, modems, Internet Phone Jack, etc.), and applications (like voice-mail, conferencing, directories, MP3 players, intercoms, etc.). It has many advanced features such as a codec translation API. The base distribution includes several channel backends, as well as applications. However, the beauty of Asterisk is its ability to be extended using its APIs, dynamic module loader, and AGI scripting interface. End users can even write their own applications that run on the system in C or any scripting language of their choice.

Download Website Updated 02 Dec 2011 gnokii

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Pop 572.73
Vit 20.08

gnokii is a multisystem tool suite for mobile phones. It provides a library to communicate with a phone hiding the communication protocol. The library handles SMS, phonebook, calendar, phone calls, and other mobile phone capabilities. It supports Nokia-FBUS mobiles, AT-capable phones (most of the mobiles), as well as Symbian-based phones.

Download Website Updated 26 Dec 2011 linphone

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Pop 567.71
Vit 17.53

Linphone is an audio and video Internet phone with GTK+ and console interfaces. It uses the SIP protocol, and is compatible with most SIP clients and gateways. It can use various audio and video codecs such as Speex, GSM, G711, G722, ilbc, amr, Theora, H263-1998, MPEG4, H264, VP8, and snow.

Download Website Updated 19 Jul 2011 Gammu

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Pop 537.28
Vit 29.19

Gammu (formerly known as MyGnokii2) is a cellular manager for various mobile phones/modems. It supports a wide variety of Nokia, Symbian, and AT devices (Siemens, Alcatel, Falcom, WaveCom, IPAQ, Samsung, SE, and others) over cables, infrared, or BlueTooth. It contains libraries with functions for ringtones, phonebook, SMS, logos, WAP, date/time, alarm, calls, and more (used by external applications like Wammu). It also includes a command line utility that can make many things (including backups) and an SMS gateway with full MySQL and PostgreSQL support from the PHP interface.

Download No website Updated 14 Jan 2014 SFLphone

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Pop 507.98
Vit 30.46

SFLphone is an SIP/IAX2 compatible softphone. The goal is to create a robust enterprise-class desktop phone. While it can serve home users very well, it is designed for intensive corporate use.

Download Website Updated 03 Jan 2014 GNU Gatekeeper

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Pop 414.72
Vit 49.07

The GNU Gatekeeper is a free H.323 gatekeeper based on the OpenH323 project. You can use it to manage a Voice-over-IP network and let endpoints (e.g., Netmeeting) communicate through symbolic names. It also has an external interface for billing and other applications. It runs on a number of Unix versions (including Linux and Solaris) and Windows.

Download Website Updated 06 Mar 2014 Kamailio

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Pop 410.11
Vit 50.48

Kamailio (formerly OpenSER) is a high-performance SIP (RFC3261) server with a flexible architecture and many extensions. The server implements proxy, registrar, redirect, and location SIP/VoIP services. It has support for UDP, TCP, TLS, and SCTP transport layers, DNSsec, ENUM, AAA via database, RADIUS, DIAMETER, gateways to SMS and XMPP, least cost routing, load balancing, NAT traversal, and call processing language. Kamailio implements SIMPLE presence and instant messaging extensions, and includes an embedded XCAP server and MSRP relay. It can be also used as a routing SIP sever for WebRTC via WebSocket.

Download Website Updated 06 Feb 2013 Skype

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Pop 360.37
Vit 10.66

Skype uses P2P (peer-to-peer) technology to provide voice- and video-based communication with other Internet users. The technology is extremely advanced, but easy to use. It features excellent sound quality, end-to-end encryption, and automatic negotiation of firewalls or routers. Among major features are SkypeOut and SkypeIn, adding the possibility to make low-cost calls to land line phones and having a fixed number to be able to receive calls from land line phones.

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