MCS MyVoIP very accurately measures the quality and performance of Internet connections for Voice over IP (VoIP) usage by simulating UDP voice data traffic between a server and browser clients. Connections are tested for jitter and packet loss and rated for the supported level of sound quality. The VoIP test can be set to various codecs or customized by packet size, packet rate, and test length. The test can further be combined with a bandwidth speed test or network route diagnostics for more in-depth connection analysis.
Flash Operator Panel displays information about your Asterisk PBX activity in real time via a standard Web browser with the Flash plugin. The display and button layout is configurable, so you can have more than 100 buttons on the screen at once. It also supports contexts: you can have one server running and many different client displays (for hosted PBX, different departments, etc). It can monitor several asterisk servers at once. It can integrate with CRM software, by popping up a Web page (and passing the CLID) when a specified button is ringing. It also can be used to enable click-to-dial for Web-based applications.
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.
SipUnit provides a class library that allows software developers to create automated unit tests for SIP applications. Session Initiation Protocol (SIP) is widely used for providing telephone services over the Internet. SipUnit extends the JUnit framework to incorporate SIP-specific assertions, and it provides a high-level API for performing the SIP operations needed to interact with or invoke a test target. A test program using the SipUnit API is written in Java and acts as a network element that sends/receives SIP requests and responses.
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.