RSS 12 projects tagged "voip"

Download Website Updated 08 Jan 2012 SIPFwd

Screenshot
Pop 69.35
Vit 3.99

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.

Download Website Updated 11 Nov 2011 Asterisk Flite

Screenshot
Pop 77.57
Vit 2.76

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.

Download Website Updated 11 Nov 2011 Asterisk eSpeak

Screenshot
Pop 76.00
Vit 2.76

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.

Download No website Updated 16 Jun 2010 FreeSentral

Screenshot
Pop 26.42
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.

Download Website Updated 17 Jun 2010 YateClient

Screenshot
Pop 34.29
Vit 37.47

YateClient is an universal, skinnable VoIP client that supports H.323, SIP, Jingle, and IAX protocols. This software is part of Yate and is available with the Yate packages for Mac.

Download No website Updated 27 Aug 2010 Amatix Office

Screenshot
Pop 47.87
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.

Download Website Updated 13 Dec 2010 SEMS

Screenshot
Pop 52.54
Vit 1.45

SEMS is a media and application server for SIP based VoIP services. It shows good performance doing basic services like announcements and conference for combination with external application servers. Thanks to its easy-to-use and flexible application development framework and back-to-back user agent support, application logic and media serving can be combined in the same process. Basic applications like announcement, pre-call announcement, RBT, conference, voicemail, mailbox, and lots of example applications are available. Scripting can be done in Python and a simple state machine description language. Support All commonly used free codecs (including g711, gsm, iLBC, speex, adpcm, and l16) are supported. Other features include wideband, ZRTP encryption, a SIP registrar client, an XMLRPC server/client, and a DIAMETER client.

Download Website Updated 18 Mar 2012 Speech synthesis for asterisk

Screenshot
Pop 74.59
Vit 2.39

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.

Download Website Updated 01 Feb 2012 Asterisk speech recognition

Screenshot
Pop 75.54
Vit 1.43

Asterisk speech recognition is an AGI script that makes use of the Google voice recognition engine in order to render speech to text and return it back to the dialplan as an asterisk channel variable.

Download Website Updated 22 Jul 2012 Speech synthesis for Asterisk using MS Tra...

Screenshot
Pop 37.71
Vit 1.89

Speech synthesis for Asterisk using MS Translator text-to-speech service to synthesize speech and play it back to the user. It supports a variety of languages, local caching of voice data, and 8kHz or 16kHz sample rates to provide the best possible sound quality along with the use of wideband codecs.

Screenshot

Project Spotlight

Kwatee Agile Deployment

Lightweight and powerful automated software deployment.

Screenshot

Project Spotlight

psensor

A graphical temperature monitor for Linux.