The SIP forwarding daemon (implemented as a stateless SIP proxy) allows you to seamlessly forward SIP requests to other SIP servers. Its main purpose is to enable users to use their own domain name in SIP URIs without the hassle of having to run a full-blown SIP server (by forwarding SIP requests to third-party SIP servers). Configuration information is stored in an SQLite database, and low resource consumption is a main priority for the project.
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.
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.
Billion Softswitch works as a switchboard in SIP/H.323 VoIP networks, proxying both signalling and media streams while performing cross-protocol conversion and even media stream transcoding. It integrates advanced call routing features as well as embedded compact billing system. It supports subscriber registration by SIP Registrar and RAS, call authorization, call data record (CDR) list maintenance, external billing system interfaces, NAT/firewall support, and report generation, and is easy to install and maintain.
Billion PBX is IP phone system that uses the SIP protocol. In addition to voice calls it supports call forwarding, voicemail, callback, and much more, and features simple installation and configuration, reliability, low hardware requirements, and a flexible pricing policy. Its main features are subscriber registration by SIP Registrar and RAS protocols, call routing by SIP and H.323, call authorization, subscriber self-service, Call Data Records (CDR), embedded rating and billing, integration with external billing systems, NAT traversal, IVR, report generation, and a convenient administrative Web interface allowing remote management.