16 projects tagged "Telephony"
Updated 27 Aug 2010
Amatix Office
| Pop | 48.83 |
|---|---|
| Vit | 1.02 |
Amatix Office is a complete communication platform for small and medium-sized businesses. It provides email, calendar, contacts, conventional telephony like ISDN, new generation VoIP telephony like SIP, instant messaging, presence, and other features needed to power a business. Amatix Office is very easy to install and use and does not require any special IT skills. It is a live software appliance which boots a computer directly from a CD or USB flash drive without any software installation.
Updated 11 Nov 2011
Asterisk Flite
| Pop | 78.49 |
|---|---|
| Vit | 3.13 |
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 | 74.32 |
|---|---|
| Vit | 3.13 |
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 21 Feb 2013
Ekiga
| Pop | 846.75 |
|---|---|
| Vit | 46.55 |
Ekiga (formely known as GnomeMeeting) is a soft phone, video conferencing, and instant messenger application for use over the Internet. It supports HD sound quality and video up to DVD size and quality. It is interoperable with many other standards compliant software, hardware, and service providers as it uses both of the major telephony standards, SIP and H.323.
Updated 16 Jun 2010
FreeSentral
| Pop | 28.81 |
|---|---|
| Vit | 1.00 |
FreeSentral is an easy-to-use IP PBX based on the telephony engine Yate. Some of its features include call forward, extension groups, call logs, call hunt, call hold, Auto Attendant, call pick up, call transfer, conference, and voicemail. The demo offers a glance into how FreeSentral works. A wizard assists users on their first configuration of the IP PBX.
Updated 14 Jan 2013
HOMER SIP Capture
| Pop | 34.73 |
|---|---|
| Vit | 11.20 |
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.
Updated 10 Apr 2009
LOUD_LISTENER
| Pop | 36.50 |
|---|---|
| Vit | 38.75 |
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 29 Jan 2010
PlugPBX
| Pop | 37.47 |
|---|---|
| Vit | 34.71 |
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 21 Sep 2011
SIP Inspector
| Pop | 72.66 |
|---|---|
| Vit | 2.35 |
SIP Inspector is a tool to simulate different SIP messages and scenarios. You can create your own SIP signaling scenarios, customize SIP messages, and monitor incoming and outgoing messages. The tool can play RTP streams from a pcap file.
Updated 18 Mar 2012
Speech synthesis for asterisk
| Pop | 88.24 |
|---|---|
| Vit | 2.50 |
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.
