Search Results for "c source code" - Page 12

This is a Win32, VC++ MFC application to manage Nokia mobile phones melodies. Includes VC++ source code and Rational Rose UML model.

CAPI based Voice Phone communication DLL (Win32) based on the AVM CAPI ADK. All CAPI operations are encapsulated in an own Thread. Functions: Multiple connections, voice file transmit/receive, line crossconnect

An open implementation of the SS7 core protocols, MTP, SCCP, ISUP, and TCAP.

A transport that enables the Jabber IM platform to send Short Messages (SMS) .

A system for logging/presenting information about incoming phonecalls. Currently only working for ISDN routers with the ability to send syslog messages.

A windows program that manages scheduled PA announcements and "bells" using a voice modem

Siphon will ultimately be a Software Voice-Over-IP phone using the SIP protocol. It will support quicknet cards as well as traditionnal soundcards. It is developped for Linux (gtk) and the win32 platform.

Open Telephony Server or OtServer for short it's a set of tools for developing telephony applications. A CallFlow Graphical Editor. Dialogic support.

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.

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.

Connecting NMS boards to the asterisk PBX via the Woomera Protocol

The CUAE SDK is an open source software development kit with sample applications and documentation for the Cisco Unified Application Environment.

A client-side library for secure voice conference.

OpenSVMS is an open source Short Voice Messaging System. SVMS (also known as VSMS or VoiceSMS) is a telecom service which enables users to send voice clips as messages rather than plain text messages.

openbaudot is a TIA_825A compliant fsk modulator/demodulator for 45.5 and 50 baud baudot used by TTY/TDD machines.

SpITAssassin will be an enhancement to the successful SpamAssassin to detect Spam over Internet Telephony. This projects extends the SpamAssassin-functionality to insert SIP-compatible headers marking possible Spit-calls.

This project is aimed primarily at users of the Icom R20 Communications Receiver and it's predecessors (R3, R5, R10). The intent is to provide ability to extend the capabilities of the radio, and provide an API to programmers interested in it.

openssl is well-known allover the world,but it is huge,complex.our target is develop a tinytls,it is small,simple,good understanding,and usually for embedded system.But it can support most openssl features.You can add some other features easily.

Tools and libraries for dialing URI addresses directly from telephone's numeric keypad w/o phone directories. Encoded uri (for instance, SIP VoIP phone number) consists of decimal numbers only and can be entered from any telephone or other device.

Open Source SIP Back to Back User Agent) and Billing system. Has an Embeded TCL interpreter making session control and session mediation limitless in flexibility. Useful for: Routing Engine, Calling Cards, SBC, Peering, Wholesale and more!

SipSpy is a distributed monitoring tool for SIP networks. SpyAgents run on each of the nodes to be monitored, and a SipSpy connects to each of these nodes, receiving information and displaying it in real-time for all the SIP packets monitored.

The G.O.N.E. is a softphone (or soft phone) running over the web, fully multi-plataform, it implements the SIP protocol, and is built to work on any SIP server, like Asterisk, and others. GONE will work on a complex sistem, but this will be showed a bit