軟體列表

Speech synthesis for Asterisk using MS Translator text-to-speech service to synthesize speech and play it back to the user. It supports a variety of languages, local caching of voice data, and 8kHz or 16kHz sample rates to provide the best possible sound quality along with the use of wideband codecs.

jPBXLite は、Java ベースの PBX です。MySQL および Tomcat が必要です。非常にでまだ開発の初期段階。SIP 拡張機能 (ボイス ・ メール)、トランクの IVR システムをサポートしています。

Encours.org is a teleconferencing system that works directly in your Web browser. It has shared slides that are visible to all, an integrated audio system with recording, "laser" pointers for lecturers, video replays of the presentation, and a chat interface with the audience. It can be connected to a phone server.

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.

FreeSentral is an easy-to-use IP PBX based on the telephony engine Yate. Some of its features include call forward, extension groups, call logs, call hunt, call hold, Auto Attendant, call pick up, call transfer, conference, and voicemail. The demo offers a glance into how FreeSentral works. A wizard assists users on their first configuration of the IP PBX.

The fonosip-siphon ultility installs preferences, settings, and icons for using FonoSIP with the Siphon softphone.

A2Billing is a telecom switch and billing system capable of providing and billing a range of telecom products and services to customers such as calling card products, residential and wholesale VoIP termination, DID resale, and callback services.

Sintel is a VoIP application using a custom XML based protocol for communication and Ogg for transmitting media.

HOMER is a robust, carrier-grade, scalable SIP capturing system and monitoring application with hEP, IP Proto4 (IPIP) encapsulation, and port mirroring/monitoring support right out of the box, ready to process and store large amounts of signaling with instant searches, end-to-end analysis, and drill-down capabilities for ITSPs, VoIP providers, and trunk suppliers using SIP signaling.

PlugPBX is a prebuilt, ARM-based Debian system for end users to run Asterisk and FreePBX on the Marvell SheevaPlug low power platform. It includes Asterisk 1.6.1 with compiled DAHDI kernel mods, FreePBX 2.5, Apache2, MySQL, Samba, Munin, Webmin, Avahai, and OpenSSH. It is built on top of Debian Squeeze.

Open Unified Recording (OUR) is a full featured Linux-based VoIP/SIP call recording engine, indexing, and retrieval system. The system resides on the network and passively captures SIP sessions.

phpivr is an Asterisk AGI application which uses PHPAGI to organize
Interactive Voice Responses. It doesn't need extensions.conf changes
and dialplan reloading if you want to modify the IVR menu, it reads
phpivr.conf on the fly. It supports unlimited levels of IVR menus. It
is a standalone tool (you do not need to install FreePBX with all its
functionality), does not require additional exotic libraries, and is
easy to integrate into your own application.

OpenUCF is a unified communications/VoIP framework for PHP5 supporting telephony, presence, and other communication-related features. It has a Web interface for users and administrators. It features a RESTful API, rich-presence, VoIP/SIP based on Asterisk, and XMPP presence based on jabberd2.

cidalert displays desktop notifications for incoming calls on VOIP phones. It requires a VOIP phone with the ability to send actions to HTTP servers in response to incoming calls.