Search Results for "java voip" - Page 3
Sort By:
Showing 66 open source projects for "java voip"
View related business solutions-
Custom VMs From 1 to 96 vCPUs With 99.95% UptimeLive migration and automatic failover keep workloads online through maintenance. One free e2-micro VM every month.
-
MongoDB Atlas runs apps anywhereMongoDB Atlas gives you the freedom to build and run modern applications anywhere—across AWS, Azure, and Google Cloud. With global availability in over 115 regions, Atlas lets you deploy close to your users, meet compliance needs, and scale with confidence across any geography.
-
1
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
-
2
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.
-
3
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/
-
4
Spark plugin for integratión with CentricCRM. This plugin for Spark XMPP client complements the integration between Asterisk PBX, Centric CRM and XMPP Server. Automatically open session on Centric CRM and redirect to contact the page
-
Try Google Cloud Risk-Free With $300 in CreditUse your credit across every product. Compute, storage, AI, analytics. When it runs out, 20+ products stay free. You only pay when you choose to.
-
5
With SIP Proxy you will have the opportunity to eavesdrop and manipulate SIP traffic. Furthermore, predefined security test cases can be executed to find weak spots in VoIP devices. Security analysts can add and execute custom test cases.
-
6
-
7
RAMS is a java based server useful for managing inter-domain VoIP routing, called number translation and Call Detail Record collection. RAMS supports the European Telecommunications Standard Institute's OSP Peering protocol (ETSI TS 101 321).
-
8
The purpose of this project is to write a plugin for the Openexchange (OX) groupware server (http://www.openexchange.com) that supports various VoIP (SIP) functionality, like direct dialing a SIP telephone, broadcasting, messaging, etc.
-
9
Java CTI application server for the open-source Asterisk (TM) PBX (includes GUI). OrderlyCalls supports both the AGI / FastAGI & Manager protocols, and includes a WebDeployer that allows you to create your own integrated VOIP and HTML applications.
-
Enterprise-grade ITSM, for every businessFreshservice is an intuitive, AI-powered platform that helps IT, operations, and business teams deliver exceptional service without the usual complexity. Automate repetitive tasks, resolve issues faster, and provide seamless support across the organization. From managing incidents and assets to driving smarter decisions, Freshservice makes it easy to stay efficient and scale with confidence.
-
10
JOpenPhone is an opensource UA (User Agent) JAVA application that allows users to connect to a voip servers using various voip protocols such as SIP, H323.
-
11
Present is an attendance dialer for educational organizations. It currently will integrate directly into Powerschool SIS, with the potential to connect to other SIS systems as well. It can be used to make automated attendance and tardy calls
-
12
-
13
-
14
-
15
A complete contact center solution for the Asterisk PBX, capable of handling inbound and outbound campaigns. Provides an agent toolbar for CTI. For managers an admin panel controls every aspect of the call center. Stunning historical and realtime stats.
-
16
sippet
C++ SIP stack based on Chrome source code
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. Existing SIP solutions for the browser are forced to use the WebSockets API to...
