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SIP over IPv4 and SIP over IPv6 are different worlds, making it impossible to register IPv4-only phones on IPv6-only services, and so on. SIPproxy64 is a proxy that translates between the two worlds. It is small enough to run inside a router.
GoAutoDial is an all-in-one open source Vicidial administrator utility that enables you to manage your Vicidial - CentOS based server. Package includes Vicidial web admin, GoAutoDial admin, NEW AgentUI. For more info, please visit www.goautodial.com.
CAGI is an API for writing AGI applications in the open source Asterisk PBX software. It was developed to integrate a medical dictation system in to Asterisk in 2005 and has been used in a production environment since then.
BCMSlib is a web user interface software that monitors the status and performance of a call center. It uses data collected in the Basic Call Management System (BCMS) on a Avaya DEFINITY System or a S8300/S8400/S8500/S8700 system.
ComuniQ-Wholesale Telecom Billing System. It helps you manage all interconnect agreements for voice, sms and content. Its Next Generation Wholesale Billing System that takes care of all your retail and wholesale interconnect settlements requirements
rates4linux provides a mostly current charge rates database for the isdn4linux isdnlog tool. This project was formerly know as the "Tarif Datenbank Crew".
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Gemini Enterprise Agent Platform lets you rapidly build, scale, govern and optimize production-ready agents grounded in your organization's data. The platform enables developers to build custom or pre-built agents for virtually any use case. New customers get $300 in free credits.
Googsystray is a system tray app for Google Voice, GMail, Google Calendar, Google Reader, and Google Docs. It notifies on new messages, alerts, etc., and provides basic services quickly.
SEMS is a GPLv2+ media and application server for SIP based VoIP services. Basic applications like announcement, pre-call announcement, RBT, conference, voicemail, mailbox etc. and lots of example applications available.
SkypeXMPP is a Karaka™ fork aimed to provide Skype to XMPP gateway It supports presence, instant messaging, multi-user chat and voice (thanks to Freeswitch) This product uses the Skype™ API but is not endorsed or certified by Skype.
Fluent is a suite of tools for the management of diverse VoIP networks. Fluent RTE, a reporting and templating engine, is at the core. Add-on modules are available to manage provisioning, CDR, feature sets, routing and more.
Castadiva is a test bed based on low-cost off-the-shelf devices, which is used to test protocols developed for MANETs. Castadiva is completely compatible with the file format used by the ns-2 simulator.
Lwazi is a robust telephony platform aiming to facilitate speedy development of experimental applications without sacrificing power by combining Asterisk with the MobilIVR Python interface bundled into a single build with a unified control interface.
Live-cd com Servidor SIP baseado em opensips. O Servidor de Telefone Livre transformará um simples PC numa central SIP multidominio. Estamos construindo a documentacao do projeto na wiki do sistema trac.
Simple interface for browsing sms from PalmDatabase.db3 and IPhone .mddata backup files, including advanced search patterns / filtering, reverse phone number lookup from vcard files (and vcard browsing), CSV export. Can handle 10'000+ sms easily.
Callblast is an open source application for Asterisk. It's designed to deliver voice messages to a group of people. It has support for multiple clients with multiple call lists.
SafiServer is an engine that powers the applications (called Saflets) created using SafiWorkshop. SafiServer uses these applications to control communications (i.e. Asterisk) along with a multitude of other applications (database, email, etc).
The OpenSIPStack Library is an implementation of the Session Initiation Protocol as described in RFC 3261. Applications: * OpenSBC (B2BUA) * OSSPhone (Softphone) Source code is available at http://www.opensipstack.org.
Asterisk Dialplan application, which allows you to use Lia_Phon and Mbrola as a French speech synthesizer. Application du plan de numérotation d'Asterisk, qui permet d'utiliser Lia_Phon et Mbrola comme synthétiseur vocal français sous Asterisk.
trixbox CE is an easy to install, VOIP phone system based on the Asterisk PBX. trixbox is designed for home or office use. trixbox CE includes CentOS linux, mysql, and all the tools needed to run a business quality phone system. (formerly asterisk@home)
The project is a preconfigured VoIP PBX VM Image based on Asterisk. But provides more advance features, such as LDAP integration, VQ monitoring via RTCP XR reports, LDAP user login, integrated billing system and telephone directory PDF generator.