Search Results for "source code claude code" - Page 3
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Showing 65 open source projects for "source code claude code"
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DAHDI for HFC-S PCI A with OSLEC
(Latest DAHDI + HFC-S PCI A support + latest OSLEC) in one git.
This project is for people who want to access one or more HFC-S PCI A based ISDN BRI cards over the DAHDI interface and maybe want to use other DAHDI hardware in parallel, too. DAHDI driven hardware can be used by Asterisk, FreeSWITCH, Yate and other telephony software. The HFC-S PCI A based cards are very cheap and thus ideal for private BRI users. The OSLEC is being developed since a long time ago and said to be very good. The superiority of "hardware echo cancellers" is a myth. This... -
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The OpenSIPStack Library is an implementation of the Session Initiation Protocol as described in RFC 3261. Applications: * OpenSBC (B2BUA) * OSSPhone (Softphone) Source code is available at http://www.opensipstack.org.
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SerMyAdmin
Web interface for OpenSIPS
...The OpenSIPS control panel fulfill this need and it is not wise to maintain two GUI projects. Our team will support the opensips-cp initiative to avoid splitting the community between two open source projects for OpenSIPS GUI. I would like to say thanks to all the people that in some way contributed with the project. The svn code is available to anyone who wants to maintain the project. Most of our development efforts are now focused in a new commercial product based on OpenSIPS called SIP Pulse, completely rewritten in Java and Glassfish www.sippulse.com supporting pre and post paid users with billing and reseller portal. -
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VoIP's project with Speex codec, speech detecting and crypto coding with traffic counting. All code is a pure Java, so completely cross-platforms. Traffic is about 9 MB per hour in both ends, sources: http://www.open-source-soft.narod.ru/arrow.7z
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RTP text/t140 Library is a reference implementation for RTP Payload Type for Text Conversation (RFC 4103). The library has source code for encoding and decoding RFC 4103 data, and may be used either as a plug-in to JMF or in a separate RTP sender/receive
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Java version of Dialogic's SwitchKit API. This API is actually a more "Java Centric" API, with listeners, filters, and a very extensible and modular framework that finally uses JNI to access the native libraries provided by Dialogic.
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BladeWareVXML is a portable VoiceXML 2.1 interpreter that is an enhanced version (performance, usability and integration) of OpenVXI. A commercial version, with documentation, sample code, and support options, is available from the Commetrex Website.
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An extention to Macromedia Dreamweaver to work with vxml in php, jsp, asp. Fetaures - Code Complition, Shows possible attributes of a vxml tag, faster vxml development
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Implementation of Media Resource Control Protocol Client (MRCP). Supports ASR and TTS functionality. Design pattern implementation. Documentation, sample application and library source code.
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October 2009 - this month the code will be re-released under the LGPL. Open Source CSTA is an implementation of the ECMA-269 standard (Services for Computer Supported Telecommunications Applications (CSTA) Phase III). Java.
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OSG ( OpenH323 Study Group ) has many sample programs (very basic, code base) which would help you to understand how OpenH323 works. The mailing list of OSG would be a good place to ask about basic information OpenH323.
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SMTP2SMS is a SMS client for sending SMS to the mobile for which EMAIL2SMS service is provided. It uses SMTP for sending SMS. Currently the code comes with a configurable Operators list (pre-configured for several Indian Mobile Operators)
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Chopped LIVR
Generate Asterisk IVR dialplans
Chopped LIVR takes YAML input and generates Asterisk dialplans. It is primarily useful for generating IVR (Interactive Voice Response) applications; however, it may be used as a tool to generate any Asterisk dialplan. Chooped LIVR also incorporates control structures and functionality enhancements to the base Asterisk dialplan language. -
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sippet
C++ SIP stack based on Chrome source code
This C++ library has been designed as a Chrome SIP stack. Sippet is an open-source SIP User-Agent library, compliant with the IETF RFC 3261 specification. It can be used as a building block for SIP client software for uses such as VoIP, IM, and many other real-time and P2P communication services. The main target was to enable Javascript applications to use UDP, TCP and TLS transports along WebSocket. Existing SIP solutions for the browser are forced to use the WebSockets API to...
