chan-sccp-b is an extension of the original chan-sccp implementation for the Asterisk soft-PBX. It lets you hook up a Cisco/SCCP Phone to your Asterisk server using the SCCP protocol, which works a lot better than the SIP firmware on the same phone. It provides full phone functionality instead of just a simple SIP channel provider. It offers functionality like shared lines, hotline functionality, guest login, dynamic speeddials, private line automatic ring-down (PLAR), personal softkey configurations, Dundi support, SCCP extended dialplan functions, manager support, and custom device state buttons.
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.
SafiWorkshop allows you to design, test, debug, and deploy advanced process flow applications (like IVRs) from a single, unified development environment. For database work, it includes a fully featured SQL query editor. SafiWorkshop also includes some Asterisk PBX specific features like an audio prompt editor.
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.
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.
SafiServer is an engine that powers the applications called Saflets that control one or more Asterisk PBXs. These can be used for IVR applications such as information, account management, nearest dealer routing, and more. It also acts as an application repository for multi-developer environments.
Appkonference is a high performance voice/video conferencing system for Asterisk. It is a fork of appconference, and it focuses on reliability and scalability. Appkonference has been tested on both Asterisk 1.4 and 1.6.X. Both Asterisk installations supported more than 1200 participants at a time.
MeetmeAutoMute is a utility for Asterisk that can automatically mute participants in a Meetme conference. This can be handy for when you are using the conference to broadcast some audio (like during a speech or presentation, and do not want to have people interfere with the speaker. It is written in Python and uses the AMI through the pyst library. Everything is configurable through a config file, and can be controlled at run time through the use of "signals". An "init.d" style bash script is included that will start the app as a background daemon.