RSS 26 projects tagged "Telephony"

Download Website Updated 15 May 2012 Text translation for Asterisk using Google...

Screenshot
Pop 19.95
Vit 1.00

Text translation for Asterisk using Google Translate uses the Google Translate API to translate text strings or detect their language and return them as channel variables.

Download Website Updated 22 Jul 2012 Text translation for Asterisk using MS Tra...

Screenshot
Pop 38.81
Vit 1.49

Text translation for Asterisk using MS Translator uses the Microsoft Translator API to translate text strings or detect their language and return them as Asterisk channel variables.

Download Website Updated 18 Mar 2012 Speech synthesis for asterisk

Screenshot
Pop 73.80
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 11 Nov 2011 Asterisk eSpeak

Screenshot
Pop 75.31
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 Website Updated 11 Nov 2011 Asterisk Flite

Screenshot
Pop 77.64
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 24 Mar 2014 OpenSIPS

Screenshot
Pop 244.13
Vit 33.73

OpenSIPS is a mature implementation of a SIP server/proxy. It is more than a SIP proxy/router, as it includes application-level functionalities. OpenSIPS, as a SIP server, can server as the core component of any SIP-based VoIP solution.

Download Website Updated 19 May 2009 Callweaver

Screenshot
Pop 102.86
Vit 2.74

Callweaver is a community driven software PBX project. The most important differences between Callweaver and Asterisk are built-in STUN support, the use of SpanDSP for better codecs and full T.38 fax over IP support, Sqlite instead of Berkeley DB, universal jitterbuffer, POSIX timers to avoid Zaptel timing dependencies, greater speed, more efficient dialplan execution, and greater stability.

No download Website Updated 30 Jan 2006 FreeSWITCH

Screenshot
Pop 34.39
Vit 54.76

FreeSwitch is a telephony application built from the ground up and designed to take advantage of as many existing software libraries as possible.

Download Website Updated 06 Mar 2014 Kamailio

Screenshot
Pop 411.59
Vit 54.52

Kamailio (formerly OpenSER) is a high-performance SIP (RFC3261) server with a flexible architecture and many extensions. The server implements proxy, registrar, redirect, and location SIP/VoIP services. It has support for UDP, TCP, TLS, and SCTP transport layers, DNSsec, ENUM, AAA via database, RADIUS, DIAMETER, gateways to SMS and XMPP, least cost routing, load balancing, NAT traversal, and call processing language. Kamailio implements SIMPLE presence and instant messaging extensions, and includes an embedded XCAP server and MSRP relay. It can be also used as a routing SIP sever for WebRTC via WebSocket.

Download Website Updated 24 Jun 2005 Mobicents

Screenshot
Pop 29.34
Vit 1.00

Mobicents is a professional VoIP middleware platform. Mobicents is a JAIN SLEE 1.0 Certified product which brings to telecom application developers what J2EE brings to Web and enterprise application developers. In the scope of telecom Next Generation Intelligent Networks (NGIN), Mobicents fits in as a high-performance core engine for Service Delivery Platforms (SDP) and IP Multimedia SubSystems (IMS). Mobicents enables the composition of Service Building Blocks (SBB) such as call control, billing, user provisioning, administration, and presence sensitive features.

Screenshot

Project Spotlight

openDCIM

Data center infrastructure management.

Screenshot

Project Spotlight

Podget

A simple podcast aggregator.