Open Source Windows VoIP Software - Page 7

ICTAgent is a advance webphone offered as chrome extesion and when installed it will convert all available contacts as click to dial. This extension will embed WebRTC based VoIP Phone in browser which will remain connected to the main server, and will be responsible for inbound and outbound calls, transfer calls, sending DTMF as well sending Fax. It can harvest contacts from the random pages. Agent can access contents, contacts from a single application. Browser can automatically fetch required URL depending on the course of call. Automatic URL will eliminate any delay which agent take while searching and finding desired data. An Embedded Web Phone. It is Operating System Independent, we just need a browser. Click 2 Call. Agent can access contents, contacts and communications from a single application(browser). It will make agent performance much better. It can harvest contacts from random pages / websites/ web applications. CRM Integration. Inbound and Outbound Calls.

ICTBilling contains a set of Drupal 7 modules that can be used with Ubercart E-commerce solution and ICTBroadcast, a unified communication auto dialer software. This Client and Billing solution for ITSP has following modules and rules: client module, ictservice module, ratelist module and rules for ubercart. Client module handles clients user interfaces, subscribed service details, service links, list of available addons and current balance. Another module, ICTService creates simple rules that adds top-up, purchase of add-on, assign/revoke permission, create and assign agent / extensions. ICTService also manages a form that allows new customers to choose a package to subscribe. Ratelist module directly connects with service portal's rate list and imports complete rate list in the Drupal table for rate list.

VoIP/SIP AutoDialer. Broadcasts a message with the option to transfer person to another number. Now includes new survey options.

Java SIP stack, originally based on mjsip.

jphonelite is a Java SIP VoIP SoftPhone for Desktops (Windows, Linux, Mac) and Android. Features 6 lines with transfer, hold, conference (up to all 6 lines), g711 u/a, g722, g729a, and video (video support in Linux or Windows only and includes H263/H264/VP8). Applet includes full JavaScript support. STUN/TURN/ICE supported. Encrypt media with SRTP. DTLS Key Exchange. Video conferencing with jPBXLite/0.12+ NOTE : THIS PROJECT HAS BEEN SUPERSEDED BY jfPhone. Please go to jfphone.sourceforge.net to get latest version.

Administration Tool for Mumble -/- Murmur Servers. Mumb1e Admin Plugin, also named "MAP" is a powerful Murmur web interface application written with PHP that can create useres, edit themself and a lot more for Mumble server. MAP is developed over m

This is my test just for testing ....

Nhabla-Bill: Is a report system a billing application inspired on Asterisk-Stat by Areski, offering many useful reports for the analysis, add to the billing function.

A JNI wrapper for pjsip. You can use this wrapper to develop Java applications using the pjsip library. At the moment only the pjsua API is implemented. If you would like to obtain a commercial license, or need customisations, please contact us.

This is a daemon that runs side by side with asterisk. And translates Emails with an XML body into an Asterisk Call with all variables passed from XML as well. Has functionality like an autodialer

RUMPI: Real Time Multi Peer Interaction. RUMPI want to be as ubiquitous as Flash-browser-plugin, but enabling browsers for real time communication between peers: VoIP (IAX2) and Real-time video Streaming. RUMPI is not a plattform, is a Browser plugin,

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 send/receive SIP messages. But for many service providers, the WebSocket protocol could not be a feasible solution due to scalability, support and other business constraints. By the way, the stack can be used as a full client-side C++ Stack (full-featured) without Javascript integration.

Git repo: https://github.com/asipto/siremis Web management interface for Kamailio (OpenSER) - handle subscriber profiles, access control lists, accounting, least cost routing and load balancing, monitoring charts, xmlrpc communication with SIP server

The web application offers to import call protocols of phone calls and lists of billed calls from different telephone providers. They can be matched to existing call protocol entries to generate cross-provider phone bills.