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395 projects in result set
Dernière Mise à Jour: 2005-08-04 01:55

Open VXI VoiceXML Interpreter

開く VXI VoiceXML インタプリタは VoiceXML ダイアログのマークアップ言語を解釈する携帯用のオープン ソース ライブラリです。それは、VoiceXML のマークアップを実行可能性がありますどのように理解する上で興味がある党のための参照として使用する設計されています。

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Dernière Mise à Jour: 2020-01-27 10:04

Network Caller ID

NCID は、呼び出し元の ID (CID)、ネットワーク上に分散です。NCID パッケージには、サーバー、ゲートウェイ、および出力モジュールとクライアントが含まれています。追加パッケージもございます。LCDncid クライアント パッケージ液晶ディスプレイ上の発信者 ID を表示する LCDproc を使用します。

(Traduction automatique)
Statut de développement: 5 - Production / écurie
Utilisateurs cibles: Advanced End Users, End Users/Desktop
Langage de programmation: C, Perl, Tcl, Coque Unix
Interface utilisateur: Tk
Dernière Mise à Jour: 2006-01-11 11:05

SIP Express Router

SER or SIP Express Router is a very fast and flexible SIP (RFC3261) server. It can handle thousands of calls per second on low-budget hadware. A C shell-like scripting language provides full control over the server's behavior. Its modular architecture allows only required functionality to be loaded. The following modules are available: accounting, digest authentication, CPL scripts, ENUM support, instant messaging, MySQL support, PostgreSQL support, a presence agent, Radius authentication and accounting, Diameter authentication, record routing, an SMS gateway, a Jabber gateway, NAT traversal support transaction module, a registrar, and user location.

Dernière Mise à Jour: 2014-04-12 12:27

libre

libre is a generic library for real-time communications with asynchronous I/O support. It is written in portable POSIX source code that conforms to the ANSI C89 and ISO C99 standards. It is robust and fast, with a low memory footprint. It also features RFC compliance and support for IPv4 and IPv6. Protocol implementations include SIP, SDP, RTP/RTCP, BFCP, DNS, STUN/TURN/ICE, HTTP, and WebSockets.

Dernière Mise à Jour: 2014-04-28 02:06

Teamspeak 3 Love Plugin

Teamspeak3 のためのプラグイン。このプラグインは彼を通じてスイッチがユーザーに従うことができます。だけの愛のメニューを右サーバー ビューの任意の名前をクリックします。

(Traduction automatique)
Statut de développement: 5 - Production / écurie
Utilisateurs cibles: End Users/Desktop
Langage naturel: English
Langage de programmation: C
Interface utilisateur: Plugins
Dernière Mise à Jour: 2010-12-02 02:45

Wammu

Wammu is a mobile phone manager that uses Gammu as its backend. It works with any phone that Gammu supports, including many models from Nokia, Siemens, and Alcatel. It has complete support (read, edit, delete, copy) for contacts, todo, and calendar. It can read, save, and send SMS. It includes an SMS composer for multi-part SMS messages, and it can display SMS messages that include pictures. Currently, only text and predefined bitmaps or sounds can be edited in the SMS composer. It can export messages to an IMAP4 server (or other email storage).

Dernière Mise à Jour: 2002-05-22 22:34

CPhone

CPhone is a cross-platform GUI for the OpenH323 VOIP libraries.

Dernière Mise à Jour: 2012-04-22 12:29

restund

restund is a modular STUN/TURN server that is designed around the principle of a lightweight core and server modules that extend its functionality. Both UDP and TCP are supported, along with IPv6 and IPv4. Supported modules include STUN, TURN, MySQL database, syslog, and status monitoring.

Dernière Mise à Jour: 2007-08-28 14:26

freePBX

The freePBX (formerly Asterisk Management Portal)
project brings together best-of-breed applications
to produce a standardized implementation of
Asterisk, complete with Web-based administrative
interface.

(Traduction automatique)
Dernière Mise à Jour: 2009-05-13 14:22

Gnu Gatekeeper ACD

The GnuGk ACD does automatic call distribution for the Gnu Gatekeeper. It is the first step to a VOIP call-center solution based on GnuGk. Incoming calls are routed to agents based on different distribution algorithms.

(Traduction automatique)
Dernière Mise à Jour: 2009-05-24 07:00

sip-redirect

sip-redirect is a tiny SIP redirect server. It supports IPv4 and IPv6, but the IPv6 support is optional. The RFC 3261 was the base for this simple and very configurable implementation. There is neither TCP nor multicast support programmed in.

Dernière Mise à Jour: 2013-06-22 22:15

pyst

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.

(Traduction automatique)
Dernière Mise à Jour: 2011-04-18 21:53

Appkonference

Appkonference is a high performance voice/video conferencing system for Asterisk. It is a fork of appconference, and it focuses on reliability and scalability. Appkonference has been tested on both Asterisk 1.4 and 1.6.X. Both Asterisk installations supported more than 1200 participants at a time.

Dernière Mise à Jour: 2014-06-14 04:08

astGUIclient

astGUIclient is part of a suite of programs that are designed to
interact with the Asterisk PBX Phone system at a client computer level
to extend the functionality of your phone and system. The main GUI
application, astGUIclient, is a set of PHP Web-based scripts utilizing
Javascript and XMLHTTPRequest functions that work through a browser to
give realtime information and functionality with nothing more than an
Internet browser on the client computer. Another component included with
the astGUIclient package is the VICIDIAL auto-dialer, a list dialer
which can dial one-call-at-a-time or be put in auto-dial mode to act as
a predictive dialer. It can function as an ACD for inbound calls or for
Closer calls coming from VICIDIAL fronters, and allows for remote agents
that may only have a phone.

(Traduction automatique)
Dernière Mise à Jour: 2011-12-26 22:34

mediastreamer

Mediastreamer is a portable C library that allows you to create and run audio and video streams. It is designed for any kind of voice over IP applications. It features RTP connectivity, audio codecs (Speex, iLBC, G711, GSM, and AMR), video codecs (MPEG4, H263, H264, and Theora), sound card I/O, wav file streaming, webcam video capture, echo-cancellation, conferencing, parametric equalization, and various other utilities. It has a modular design that makes it extensible through plugins. This is the media-streaming component of linphone, a GPL SIP video phone.