Search Results for "c# source code example"

Pyst consists of a set of interfaces and libraries to allow programming of Asterisk from python. The library currently supports AGI, AMI, and the parsing of Asterisk configuration files. The library also includes debugging facilities for AGI. 2014-04-17: Moved the version control to GIT. To check out see the tab "Code". Note that the whole history including ancient CVS, then some time in monotone, then subversion was united into one GIT repository thanks to ESR's...

Replacement for the SCCP channel driver in Asterisk. Extended features include Shared Lines, Presence / BLF, customizable Feature Buttons, and Custom Device State. Visit our discussion mailing list for help and join us as a developer if you like. The project moved to https://github.com/chan-sccp/chan-sccp

New version 1.2. This is the code to STUNTMAN - an open source STUN server and client code by john selbie. Compliant with the latest RFCs including 5389, 5769, and 5780. Also includes backwards compatibility for RFC 3489. ICE and WebRTC ready. Version 1.2 compiles on Linux, MacOS, BSD, and Solaris. Supports the STUN protocol on both UDP and TCP for both IPv4 and IPv6. Windows binaries are also provided. Additional features are in development. This is a mirror of the code on...

The OSP Toolkit is a client side implementation of the ETSI OSP VoIP Peering protocol (ETSI TS 101 321). The OSP Toolkit project was begun in 1998 and the code has been incorporated into many commercial and open source VoIP products.

Project moved: Find the latest code at https://www.h323plus.org/ Follows on from now depreciated OpenH323 and designed as a drop in replacement for OpenH323. Development of advanced open source H.323 including application sharing, video conferencing and incorporates new research and development work. Project moved: Find the latest code at https://www.h323plus.org/

Baresip is a portable and modular SIP User-Agent with audio and video support. the latest source code can be found here: https://github.com/alfredh/baresip

The Session Initiation Protocol (SIP) is a signaling communications protocol widely used nowadays for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks. A good way to design optimization techniques for SIP deployment would be to analyze SIP traffic from existing networks. However, publicly available analyses of SIP traffic are rare and thus not a lot of knowledge exists about typical behavior of a SIP server (as opposed to, for...

This ASN.1 compiler turns ASN.1 specifications into C code. The asn1c is shipped together with conformant BER/DER/XER/PER codecs. The X.509, GSM TAP3, MEGACO, RRC and LDAP encoding and decoding examples are part of the source code distribution. NOTE: THE asn1c PROJECT HAS LARGELY MOVED TO GITHUB: http://github.com/vlm/asn1c

KAMAILIO (OpenSER) - robust, secure and scalable Open Source (GPL) SIP (RFC3261) server implementation with large features set (over 90 extension modules). As of May 2009, source code is hosted by GIT repository at http://sip-router.org

LuaSofia is a Lua binding of Sofia-Sip library. Sofia-SIP is an open-source SIP User-Agent library, compliant with the IETF RFC3261 specification. We decided to use git, source code can be get at: git://github.com/ppizarro/luasofia.git

SEMS is a GPLv2+ media and application server for SIP based VoIP services. Basic applications like announcement, pre-call announcement, RBT, conference, voicemail, mailbox etc. and lots of example applications available.

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.

*** MOVED ON GOOGLE CODE ***

A command line SIP/H323 softphone capable of sending and receiving audio files as well as sending out of band DTMF digits. Supports a BNF format configuration language for scripting call scenarios. Useful for example system testing.

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.

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.

Implementation of Media Resource Control Protocol Client (MRCP). Supports ASR and TTS functionality. Design pattern implementation. Documentation, sample application and library source code.

Tool (C) to split the WURFL XML mobile browser database into individual device files with symlinks (stubs for NTFS) to reduce access time and server load. Includes a simple PHP library/API as an example how to access the data and an example phtml file.

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.

Welcome to the Linux-IrDA project. The overall goal of this project is to make an implementation of the IrDA (tm) standards specifications for the Linux kernel. The code is licensed under the GNU Public licence (GPL) and is now included in Linux-2.2.

Time of day service over telephone using Voicent Gateway, a VoiceXML gateway that specially designed for voice modems. A Free version is available for download at http://www.voicent.com/download. Sample code for interactive telephony applications.

BeepagePHP provides a PHP-based web client to the beepage client/server paging system. Beepage is a Unix-based, Internet-aware text paging system that implements the Telocator Alphanumeric Protocol (TAP). Source code is hosted at http://git.io/rgikgg

The purpose of this project is creating a plugin for <a href="http://www.ethereal.com/">Ethereal</a> that will enable it to decode ITU-T Recommendation H.323 traffic. The implementation is based on <a href="http://www.openh323.org">OpenH323</a> code

An Open H.323 based channel driver for the the Asterisk PBX. This code has now been intergrated into the Asterisk source tree. Please see the Asterisk source tree for any future updates of this software.

PNX system develops the middle-ware source code to glue Asterisk with a number of powerful telephony products such as: 1) OpenH323 H.323 stack 2) Vovida SIP stack 3) Bayonne Voice Automation Platform The advantages? An advanced IP PBX supporting the wide