300 projects tagged "Telephony"
Updated 10 Nov 2009
Gemeinschaft
| Pop | 63.37 |
|---|---|
| Vit | 1.42 |
Gemeinschaft is a PBX based on Asterisk, MySQL, Apache, and PHP, and designed for high availability and clustering. It provides automatic provisioning for mass deployment, and can handle over 10,000 users. Administration is done via shell scripts or a Web GUI. Hot-desking and mobility are supported. German voice prompts are included. There is a standards-compliant Web GUI with internationalization (and German and English translations). Outbound and inbound routing with full PCRE support is included.
Updated 01 Mar 2009
OpenCellID
| Pop | 15.36 |
|---|---|
| Vit | 39.31 |
The OpenCellID project aims to create a complete database of CellIDs worldwide, with their locations. It provides access to tools to create the database and retrieve location information. The source code is available under the GPL, while data is available under the CC license.
Updated 10 Apr 2009
LOUD_LISTENER
| Pop | 35.03 |
|---|---|
| Vit | 38.84 |
LOUD_LISTENER is a light-weight, high-speed voice packet recorder that can target a particular call based on a SIP URL or TG, designed to satisfy regulatory lawful Intercept requirements.
Updated 27 Apr 2009
MeetmeAutoMute
| Pop | 28.84 |
|---|---|
| Vit | 1.00 |
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.
Updated 14 Apr 2012
chan_nms
| Pop | 34.99 |
|---|---|
| Vit | 1.00 |
CHAN_NMS is an Asterisk PBX channel driver that supports the NMS Communications (acquired by Dialogic) Open Access line of products, which includes E1/T1 boards for the PCI(-X), PCIe, and cPCI(e) buses.
Updated 18 Apr 2011
Appkonference
| Pop | 86.49 |
|---|---|
| Vit | 3.09 |
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.
Updated 01 Dec 2009
SafiServer
| Pop | 61.85 |
|---|---|
| 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 11 Nov 2011
Asterisk Flite
| Pop | 77.59 |
|---|---|
| Vit | 3.12 |
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 11 Nov 2011
Asterisk eSpeak
| Pop | 74.08 |
|---|---|
| Vit | 3.12 |
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 20 Oct 2009
SafiWorkshop
| Pop | 51.91 |
|---|---|
| Vit | 36.28 |
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.
