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.
Related ProjectsMPC-BE, Dumpper v.60.3, Nucleus日本語版, FreeStyleWiki, Win32 Disk Imager |
ZRTP Protocol Library is an implementation of Phil Zimmermann's ZRTP protocol, created based on and interoperable with Zfone beta 2. Combined with the GNU RTP Stack (ccrtp), this offers the ability to create communication services that natively support the ZRTP protocol.
Related ProjectsOpenTween, DeSmuME, Darik's Boot and Nuke, iReport-Designer for JasperReports, PukiWiki |
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.
Related ProjectsHinemos, Darik's Boot and Nuke, DeSmuME, SmillaEnlarger, OpenTween |
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.
Related ProjectsRealTerm: Serial/TCP Terminal, PukiWiki, Darik's Boot and Nuke, SharpDevelop-jp, OpenTween |
trixbox CEは、インストールが容易なAsterrisk PBXベースのVOIP電話システムです。trixboxは、自宅やオフィスで使用する目的で設計されています。trixbox CEにはCentOS Linux、Mysql、ビジネス品質の電話システムに必要な各種ツールが含まれています。
Related ProjectsSkype4Java (a.k.a Skype API for Java), The Lovely Sister Database, LetterFix for Mac OS X Mail.app, Win32 Disk Imager, MPC-BE |
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.
Related ProjectsSmillaEnlarger, Darik's Boot and Nuke, iReport-Designer for JasperReports, Skype4Java (a.k.a Skype API for Java), PukiWiki |
Asterisell is a Web application for rating, showing to customers, and billing of Asterisk VoIP calls.
Related ProjectsNucleus日本語版, PukiWiki, Skype4Java (a.k.a Skype API for Java), iReport-Designer for JasperReports, FreeStyleWiki |
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.
Related ProjectsDarik's Boot and Nuke, CrystalDiskInfo, PukiWiki, Virtual Floppy Drive, iReport-Designer for JasperReports |
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.
Related ProjectsDeSmuME, Darik's Boot and Nuke, RealTerm: Serial/TCP Terminal, Nucleus日本語版, PukiWiki |
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.
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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.
Related ProjectsJulius, Hinemos, Clonezilla, MPC-BE, MeCab |
pjsip is a multimedia communication library based on the SIP protocol. It is integrated with a rich media and a NAT traversal library supporting the ICE protocol. It is very portable and has a small footprint for embedded use.
Related ProjectsRealTerm: Serial/TCP Terminal, SmillaEnlarger, Skype4Java (a.k.a Skype API for Java), FreeStyleWiki, OpenTween |
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.
Related ProjectsBathyScaphe, Open Jane, bbs2chreader, RealTerm: Serial/TCP Terminal, SmillaEnlarger |
Elastixは、AsteriskベースのPBXに使いやすいインターフェースを付加するためにベストなツール群を統合化したアプライアンスソフトウェアです。また、オープンソースのテレフォニーのためのベストなソフトウェアパッケージとするために、独自のユーティリティの設定も追加されています。
Related ProjectsWin32 Disk Imager, LetterFix for Mac OS X Mail.app, Dumpper v.60.3, postLDAPadmin, MPC-BE |
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.
Related ProjectsDeSmuME, SmillaEnlarger, Nucleus日本語版, FreeStyleWiki, iReport-Designer for JasperReports |