17 projects tagged "Telephony"
Updated 11 Nov 2011
Asterisk Flite
| Pop | 80.56 |
|---|---|
| Vit | 3.08 |
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.
Updated 10 Jul 2010
Asterisk Presence Panel
| Pop | 44.54 |
|---|---|
| Vit | 1.00 |
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.
Updated 11 Nov 2011
Asterisk eSpeak
| Pop | 77.18 |
|---|---|
| Vit | 3.08 |
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.
Updated 08 Feb 2013
Elastix
| Pop | 351.87 |
|---|---|
| Vit | 30.84 |
Elastix is a software appliance that integrates the best tools available for Asterisk-based PBXs into a single, easy-to-use interface. It also adds its own set of utilities and allows for the creation of third party modules. The goals of Elastix are reliability, modularity, and ease-of-use. It also features strong reporting capabilities.
Updated 19 Mar 2011
OrderlyQ
| Pop | 19.85 |
|---|---|
| Vit | 28.68 |
OrderlyQ is a call center queue solution. Instead of forcing callers to wait for an available agent, OrderlyQ tells them when there will be an agent available and reserves a place at the head of the queue when they call back.
Updated 29 Jan 2010
PlugPBX
| Pop | 32.31 |
|---|---|
| Vit | 35.17 |
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.
Updated 01 Dec 2009
SafiServer
| Pop | 64.55 |
|---|---|
| Vit | 1.47 |
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.
Updated 20 Oct 2009
SafiWorkshop
| Pop | 53.81 |
|---|---|
| Vit | 36.62 |
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.
Updated 18 Mar 2012
Speech synthesis for asterisk
| Pop | 87.65 |
|---|---|
| Vit | 2.49 |
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.
Updated 03 Feb 2011
Taridium ipbx
| Pop | 29.63 |
|---|---|
| Vit | 29.45 |
The Taridium ipbx is a complete software-based VoIP PBX system that replaces a traditional proprietary hardware PBX. It runs on Linux without the need for extra software licenses, and is based on the SIP standard. It supports many phones and is available for free in a 5 user version.
