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.
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 Presence Panel is a simple application that allows the user to monitor the status of extensions on multiple Asterisk based PBX systems. The application connects to the Asterisk server using the manager interface. When it starts, it uses a manager command to probe the status of the extension, after which it relies on the extension status messages in the manager session to update its status. The application features contact groups, the ability to click to dial a contact, computer driven dialling for any number, the ability to connect to multiple Asterisk servers, and integration with the system tray on supported platforms.
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.
DeforaOS Phone is a GTK+ application that has an interchangeable telephony backend, complete with call, contact, and message management. With its backends for GSM modems, it can be used as the telephony application on phones including the Openmoko Freerunner, Nokia N900, and some HTC phones running custom Linux distributions, and as a tethering application that provides Internet access to a desktop computer. Support for VoIP protocols is under development, with a backend based on the sofia-sip library.
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.
Ip phone Scanning Made Easy (ISME) scans a VOIP environment, adapts to enterprise VOIP, and exploits the possibilities of being connected directly to an IP Phone VLAN. It seeks to get the phone's configuration file directly from a TFTP server, enable SIP/SIPS (TCP/UDP), communicate with an embedded Web server and Web server banner, identify the editor by MAC address, and identify potential default login/password combinations which should be changed.
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.