Search Results for "c source code" - Page 4

ContactList plug-in extension for PJSGUA SIP client. Minimum required version of PJSGUA: v0.7.1

The project is a preconfigured VoIP PBX VM Image based on Asterisk. But provides more advance features, such as LDAP integration, VQ monitoring via RTCP XR reports, LDAP user login, integrated billing system and telephone directory PDF generator.

...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.

Open Unified Recording (OUR) is a full featured Linux based Open Source VoIP/SIP Call Recording engine, indexing and retrieval system. The system resides on the network and passively captures SIP sessions. Project is sponsored by UnifiedRecording.com

This is a SIP signaling layer to create a fully operative multipoint (video) conference server using SIP clients and RTP media streams in combination with strManager as a media management layer.

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

DAHDI Linux Drivers for Rhino Telephony Cards

This project provides a meter of QoS on VoIP using the protocol SIP Media Loopback

QoS Number is a project to control de quality of service ( QoS ) in a IP-Centrex Plattform .

Apreta is about sharing information in real-time meetings. It provides desktop sharing, presentation sharing, a shared whiteboard, and audio conferencing.

Farfon, was the world's first designed from the ground up IAX2 phone created to work with Asterisk PBX.

An XCAP server is used by XCAP clients to store data like buddy lists and presence policy in combination with a SIP Presence server that supports PUBLISH, SUBSCRIBE and NOTIFY methods to provide a complete SIP SIMPLE server solution.

The goal of this project is to provide a liveCD allowing the user to analyze networks for VoIP installations. This project gives you a global network state.

Quality plugin for PJSGUA SIP client.

8ix Zenith CE spells Asterisk derived IP Telephony solution.When you want robust communication, without the huge cost and recurring bill that comes with the way you do business, 8ix Zenith CE addresses your every communication need.

Vorilon is a Voice Teleconferencing software consisting of a Client and Server applications to be used for Game Voice chatting and General Voice Teleconferencing.

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.

This program monitors a given email address for a certain subject with a list of phone numbers, and then calls them - all at once. Requires Skype.

Kiax is a softphone (soft phone, VoIP client) with a simple and comfortable user interface for making VoIP calls to Asterisk PBX. It depends on the iaxclient library to use Asterisk's IAX2 protocol for easy call configuration and audio setup.

Olyo is an open source project which aims at implementing a prototype for Peer-to-Peer Session Initiation Protocol (P2PSIP) and providing real-time multimedia services over Internet based on this prototype. Olyo is promoted by MINE lab, BUPT.

Pops up a URL when a Call on an Asterisk Server is Answered, passing the Extension, Caller ID name and number. Connects to the Asterisk Server using the Manager Interface. Can be used to connect to a web based CRM.

This project aims to build a software reference implementation of the EBU standard for the transmission of high quality, low latency, audio streams over IP networks. (EBU-tech 3326)

This is a patch for Asterisk PBX that makes it usable as a call agent in PacketCable networks. It adds support of NCS messaging to Asterisk PBX. NCS, in its turn, is supported by the majority of vendors who produce EMTAs.

The OpenOSP server protocol stack is a full function, secure, highly scaleable C implementation of OSP (ETSI TS 101 321 v2.1.0) for the Sparc Solaris platform.

JSR180 is the standard API spec by jcp.org which is been implemented over Resiprocate Stack. Our purpose is to enable "Open Source Code with Open APIs". By implementing JSR180 we could easily access the SIP Stack by the API documented in the JSR180 Spe