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.
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.
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.
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.
The CommuniGate Pro Community Edition provides a single address and identity for all forms of Internet communications. You can check your e-mail, chat with colleagues via Instant Messaging and VoIP, view multiple calendars (and other presence features), maintain address books, and view streaming RSS news feeds, all through the powerful Flash-based interface.