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.
The Taridium ipbx is a complete software-based VoIP PBX system that replaces a traditional proprietary hardware PBX. It runs on Linux without the need for extra software licenses, and is based on the SIP standard. It supports many phones and is available for free in a 5 user version.
GROUP-E is collaboration software which integrates groupware, project management, and business server on one platform. The solution is based on a LAMP architecture (Linux, Apache, MySQL, PHP). GROUP-E offers project management, transparent Samba (file server) integration, integration of Cyrus IMAP server with administration and personal SIEVE filters, support for SyncML 1.0, LDAP-based user management with single sign-on authentication, and LDAP contact databases. GROUP-E also provides ActiveSync for synchronization.
VoiceOne is a Linux distribution that gives you the ability to install a PBX platform with an easy to use Web-based GUI. It also provides a framework for building a communication server adding various plugins. Its main features are Asterisk 1.8 with realtime configuration with MySQL, a Ubuntu 10.04 base, and support for both hard disk and Compact Flash card installation.
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.
Flite For Asterisk provides the "Flite" dialplan application, which allows you to use the Flite TTS Engine with Asterisk. This module invokes the Flite TTS engine locally, and uses it to render text to speech. It supports voice selection and 8kHz or 16kHz sample rates to provide the best possible sound quality along with the use of wideband codecs. It works with Asterisk 1.6.x, 1.8, and 10.
Asterisk eSpeak provides the "eSpeak" dialplan application, which allows you to use the eSpeak speech synthesizer with Asterisk. This module invokes the eSpeak TTS engine locally, and uses it to render text to speech. It supports the following languages: Afrikaans, Albanian, Armenian, Cantonese, Catalan, Croatian, Czech, Danish, Dutch, English, Esperanto, Estonian, Finnish, French, Georgian, German, Greek, Hindi, Hungarian, Icelandic, Indonesian, Italian, Kannada, Kurdish, Latvian, Lojban, Macedonian, Malayalam, Mandarin, Norwegian, Polish, Portuguese, Romanian, Russian, Serbian, Slovak, Spanish, Swahili, Swedish, Tamil, Turkish, Vietnamese, and Welsh. It supports 8kHz and 16kHz sample rates to provide the best possible sound quality along with the use of wideband codecs. It works with Asterisk 1.6, 1.8, and 10.
Speech synthesis for asterisk is an Asterisk AGI script that uses Google Translate to convert text to speech and play it back to the user. It supports a variety of different languages, local caching of voice data, and a choice between 8 kHz or 16 kHz sample rates to provide the best possible sound quality along with the use of wideband codecs.