Search Results for "sofia-sip" - Page 5

Core integration to track inbound/outbound Calls from Freepbx/Elastix/Trixbox in SugarCRM with no user interaction (only server to server).Identify the caller for inbound calls and display it in any SIP phone like X-lite.Upgrade safe module loaders

SIP Inspector is a tool written in JAVA to simulate different SIP messages and scenarios. You can create your own SIP signaling scenarios, customize SIP messages and monitor incoming and outgoing messages. The tool can play RTP streams from a pcap fi

SIP Express Media Server (SEMS) is an extensible media server for SIP based on VoIP services.It has been supporting applications like voicecall ,conference etc but has no support for video. The goal is to add MPEG4 to the list of supported apps.

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.

This is an university project for the Software Technologies course in the Faculty of Mathematics and Informatics department of the Sofia University. It will present a repository for agile projects that will allow creation and search for projects.

Open source implementation of the Open Mobile Alliance (OMA) Location in SIP (LOCSIP) requirements, providing extensions to the Session Initiation Protocol (SIP) for conveying locations requests and responses to enable location based services.

This is a open source media gateway controller. The protocol used is MEGACO/H.248. In a later version also SIP will be supported.

sIP Config is small tool that do anything related to IP Address. sIP Config can find your internal/external ip address, mac address and host name. It can also find ip address of websites.

Máquina virtual com instalação básica do asterisk no centos para fins de testes. Inicie a máquina e configure um sip-phone com os logins e senhas contidos em /etc/asterisk/sip.conf.

Txt2XMLParser is a versatile and generic text to xml converter library. Based on Regular Expression it can be configured using a single xml file allowing to parse any txt file format (csv , generic txt files, protocol formats such as SIP)

This projects is a simple Media Gateway, build under Sofia-Sip and Gstreamer libraries. It was build for a project that uses a MediaGateway and Open-IMS to distributed audio publicity.

OpenBTS is an implementation of the GSM air interface (Um) that allows cellular handsets to be used directly as SIP endpoints. It uses a software-defined radio to generate its air interface and uses Asterisk or yate as its network interface.

Fluent is a suite of tools for the management of diverse VoIP networks. Fluent RTE, a reporting and templating engine, is at the core. Add-on modules are available to manage provisioning, CDR, feature sets, routing and more.

AuDA (Availability under DoS Attacks) security testing environment. Doing automated & reuseable tests. First target it is written for is SIP system. But it is easily extensable and may adopted to other internet services.

Simu server project is aimed for IP telephony (VoIP) between classical SIP phones and Flash streaming applications.

Dubtel One is a Cloud API for quickly building advanced voice and chat applications. The API was developed using industry standards including XML, SIP and XMPP.

SIP.NET is SIP stack .NET library written in C#. SIP message parser realized on Deterministic Finite-state Machine (DFA parse all SIP message for one pass). DFA generates automatically from Augmented Backus–Naur Form from RFC3261.

Live-cd com Servidor SIP baseado em opensips. O Servidor de Telefone Livre transformará um simples PC numa central SIP multidominio. Estamos construindo a documentacao do projeto na wiki do sistema trac.

jSipgateConnect library allows Java-based access to all functions specified in the 'Sip Advanced Messaging, Usability, Routing and Accounting Interface' specification.

Rabboni VOIP is an Open Source Cross Platform Softphone developed with trolltech QT LIBRARY it uses PJSIP - Open Source SIP Stack

PJSGUA is an Open Source SIP UAC based on PJSUA reference UAC from PJSIP project

Telepathy-SofiaSIP is a SIP connection manager (protocol plugin) for the Telepathy framework (http://telepathy.freedesktop.org). It's based on the Sofia-SIP stack.

... 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.

The program captures SIP traffic (from port-mirroring on a switch) and provides statistics about the SIP message latency. The latency is the delta between incoming and outgoing messages to a SBC / proxy that process SIP messages.

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