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395 projects in result set
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: 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: 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: 2018-09-03 04:11

VSCP Protocol & Friends

マシンツーマシン(m2m:Machine-to-Machine communication)およびインターネットのための拡張性の高いプロトコルとソフトウェア ツールのコレクションです。ここでのプログラムは Windows と Linux で動作し、VSCP の周りは、非常にシンプルな制御プロトコルに基づいています。

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: 2006-03-01 07:29

SCMxx

SCMxx is a console program that allows you to
exchange certain types of data with mobile phones
made by Siemens. Some of the data types that can
be exchanged are logos, ring tones, vCalendars,
vCards, phonebook entries, and SMS messages. It
works with the following phones: S25, C35i, M35i,
S35i, ME45, S45, SL45, M50, and probably some
others, too. It basically uses the AT command set
published by Siemens (with some other additional
resources).

(Traduction automatique)
Dernière Mise à Jour: 2004-12-21 10:19

asterisk-oh323

asterisk-oh323 adds H.323 support to the ASTERISK soft PBX. It does this
by interfacing the OpenH323 library to ASTERISK through a loadable
module. The package provides the channel driver as well as a wrapper in
a shared library form. It is able to initiate and receive calls to and
from H.323 endpoints, and has been successfully tested with the H.323
terminals on the OpenH323 site (ohphone, openphone), GnomeMeeting,
Microsoft NetMeeting, Cisco routers, H.323 Snom phones, and other hardware and software H.323 IP phones.

(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: 2007-03-03 14:11

1VideoConference

1VideoConverence is Web2.0 audio-video conference call software for Asterisk with support for Web, phone, MSN, Skype, Yahoo, and Jabber clients. This VoIP and VVoIP conferencing app for business, government, education, and health care is based on C#, WinFX, XAML, and .NET 3.0.

Dernière Mise à Jour: 2007-04-27 08:34

Sipp

Sipp is a performance testing tool for the SIP protocol. It includes a few basic SipStone user agent scenarios (UAC & UAS) and establishes and releases multiple calls with the INVITE and BYE methods. It also reads XML scenario files describing any performance testing configuration. It features the dynamic display of statistics about running tests, periodic CSV statistics dumps, TCP, UDP, or TLS over IPv4 or IPv6 over multiple sockets or multiplexed with retransmission management, regular expressions and variables in scenario files, conditional branching, and dynamically-adjustable call rates. RTP play (voice, video, and RFC2833 DTMFs) is also supported.

(Traduction automatique)
Dernière Mise à Jour: 2009-02-01 16:53

Jiplet Container

Jiplet Container (Java SIP Servlet) is a servlet-like development and runtime environment for SIP applications. The SIP protocol is widely used for voice services over IP networks. This product enables developers to create server-side SIP applications using a component-based model similar to that envisioned by the J2EE architecture. The Jiplet container runs as a standalone server as well as a JBOSS service.

(Traduction automatique)
Dernière Mise à Jour: 2003-11-27 07:56

tsemgr

tsemgr is a GTK+ application to manage the SonyEricsson T68 mobile phone. It allows you to read and send short messages (sms), view and edit the phonebook, upload files via IrDA and Bluetooth, and turn your phone into a remote control for your Linux box.

(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: 2012-01-08 00:16

SIPFwd

The SIP forwarding daemon (implemented as a stateless SIP proxy) allows you to seamlessly forward SIP requests to other SIP servers. Its main purpose is to enable users to use their own domain name in SIP URIs without the hassle of having to run a full-blown SIP server (by forwarding SIP requests to third-party SIP servers). Configuration information is stored in an SQLite database, and low resource consumption is a main priority for the project.

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)