Logiciel Télécharger Carte

The vomit utility converts a Cisco IP phone conversation into a wav file that can be played with ordinary sound players. The phone conversation can either be played directly from the network or from a tcpdump output file. Vomit is also capable of inserting wavfiles into ongoing telephone conversations. It can be used as a network debugging tool, a speaker phone, etc.

SuperShaper-SOHO is a traffic shaping setup for DSL connections which prioritizes VoIP and interactive traffic and makes sure P2P traffic doesn't saturate your uplink.

JVOIPLIB is an object-oriented Voice-over-IP (VoIP) library, written in C++. Its purpose is to make it easy to set up and control VoIP sessions. The session's characteristics can be changed during a VoIP call.

Asterisell is a Web application for rating, showing to customers, and billing of Asterisk VoIP calls.

USB Phone Connector is a framework for connecting USB "phones" (typically a device incorporating a USB-addressable speaker, microphone, LCD display, and like peripherals in a telephone-like form factor) to a softphone application. Newer versions support Skype as well as the linphone SIP phone.

SipExchange is a softswitch that provides standard SIP services like location, proxy, and presence. Using the SipExchange application, service providers can offer VoIP telephone services to their subscribers as well as other services based on voice, video, and instant messaging. SipExchange supports many of the standard subscriber features offered by the traditional telephone exchanges and PBXs. In addition, SipExchange supports external call control capabilities which service providers and software developers can use to create new features and services rapidly and plug them into the SipExchange application. SipExchange works with standard SIP phones that adhere to the SIP protocol standards. Its software architecture is flexible, scalable, and easily extensible. It runs as an enterprise application inside the JBoss server and takes advantage of many services that a J2EE server provides. SipExchange provides a portal-based user interface with which system administrators can manage subscribers and features as well as perform other routine operations. From the portal, subscribers can manage their profiles, view the call detail records, and customize the features to which they have subscribed. Service providers can easily add additional content to the portal and customize the look and feel.

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.

AstBill is Web-based billing, routing, and
management software for Asterisk and MySQL based
on Drupal. It provides pre- and post-paid billing
services. It completely automates Asterisk billing
from start to finish. Key benefits are the central
Web-based installation, credit control on outgoing
calls, and the call routing module.

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.