Search Results for "drcom-for-linux" - Page 4
Sort By:
Showing 268 open source projects for "drcom-for-linux"
View related business solutions-
Over two-thirds of people reuse passwords across sites, resulting in an increasingly insecure e-commerce ecosystem. Learn how passwordless can not only mitigate these issues but make the authentication experience delightful. Implement Auth0 in any application in just five minutes
-
With clubs, you can understand and nurture the culture that exists in your business. Staff join together through shared interests, bringing the human to HR.
-
1
Experimental open source project, which aims to test possibility of VoIP calls "from browser to browser". Main part is Java application implementing UDP hole punching, which is sent to both clients using JWS. See it online in action at project homepage
-
2
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
-
3
The system is used to control elements of a smart home remotely, with a web or a telephony interface. Automation can also be created to automatically execute actions based on events.
-
4
Eclipse plugin providing form editing and validations on JAIN SLEE 1.1 deployment descriptors. This plugin requires Eclipse 3.5 (Galileo) or later.
-
Thousands of IT Managers and Sysadmins use EasyDMARC every day to manage, monitor, and secure email infrastructure.
-
5
-
6
LibDreamSpeak is an example implementation of the TeamSpeak2 protocol (client-side) in Java. Some day it may be used as base for alternative TeamSpeak2 (mobile?) clients.
-
7
AsterFax provides an Email to Fax gateway for Asterisk. AsterFax lets you send an email by Fax. Enter the phone no. in the 'To' address, compose your email message and click send. You can also fax a MS-Word document or other attachment.
-
8
miTester for SIP is an automated SIP testing tool designed and developed to take care of the complex pre-deployment testing of SIP applications easily. This SIP testing tool can be used to simulate SIP call-flows & automate functional, regression tests.
-
9
Asterisk based callcenter system. Administration interface to manage queues, agents, accounts, skills - track calls, create statistics. Realtime queue monitor. Agent web interface with build in IAX softphone.
-
Cloud based, mobile-ready job boards for associations, recruiters, corporations, and entrepreneurs. Features include job maps, custom fill-in forms, job alerts, resume search, email marketing, job feeds, job scraping, category filters, and management tools. E-Commerce with flexible pricing packages, custom products, and automatic recurring billing. Replace your current web site or run your board as a sub-domain. Built and supported in the USA. No term commitment and support is free.
-
10
For developers who are seeking to integrate SMS functionality into their software or applications, this project offers a range of simple solutions in the form of our SOAP, HTTP and SMTP API's that will allow seamless integration with our systems.
-
11
The Genalyze log analyzer, a simple yet highly extensible framework for parsing log files, was designed for parsing T-Server log files generated by Genesys contact center software platform, yet is generic and easy enough to adapt for other log formats.
-
12
Prototype testbed implementation of the IETF Media Server Control (MEDIACTRL) SIP Control Framework, comprehensive of both control and processing functionality (as in IMS MRF, Media Resource Function).
-
13
Java library for encoding/decoding MMS messages. Also provides a simlpe client for sending MMS through a WAP gateway.
-
14
Java Media Library for wiring together Sinks and Sources. Can have rtp Sinks and Sources, FileRecorders/FilePlayers, Microphone, etc. The framework has default sink/sources already. It does not provide Text-to-Speech Source or Speech Recognizer Sink.
-
15
A Sip live audio feeding agent. The agent captures audio from sound card and sends live audio stream(uLaw) to caller(sip phone) using RTP. It is based on Peers 0.3(http://peers.sourceforge.net/). Can be used in IP telephony to broadcast live audio.
-
16
This is a repository for QueueMetrics add-ons that are contribuited by the community and are open-source.
-
17
J8051 is an API for communication between a 8051 microcontroller and a cell phone using Java ME, Bluetooth and "C" language callback functions, providing a easy way for embed systems developers build a communication between.
-
18
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
-
19
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.
-
20
Xen-Agi is an open source java library that allow you to build java applications that interact with the Asterisk Server for VOIP and PBX functionality. This project supports the FastAGI protocol exclusively and is inspired by the Asterisk-Java project.
-
21
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.
-
22
The Jive API is a lightweight Java framework for writing IVR applications. The premise for the Jive API is to allow powerful IVR applications to be written without the user having to worry (or know) about the underlying IVR platforms.
-
23
SipManager is a full-feature IP-Centrex PBX web app for hosted/multi-client PBX based on Asterisk and OpenSer. VoIP/ToIP platform (PBX, Fax, workflows, etc.). User-friendly web interface. Realtime architecture with SQL database. ASP. http://www.ovvoe.com
-
24
SIP to Skype Gateway/Bridge/Converter/Adapter. Make and receive SIP to Skype Calls and Skype to SIP Calls. Call Skype users using speed dial or use SkypeOut. Make SIP calls from Skype using a SIP provider or SIP PBX. Use as a Skype Trunk with a PBX.
-
25
Sipana is a distributed SIP monitoring tool to monitor the SIP signaling behavior using sequence diagrams and providing SIP end-to-end performance metrics through a centralized Web interface. For more information please visit http://sipana.org/