Addresses the needs of small businesses and large global organizations with thousands of users in multiple locations.
Choose from a complete set of software solutions across EHSQ that address all aspects of top performing Environmental, Health and Safety, and Quality management programs.
Build a scalable voice experience with the API that's connecting millions around the world.
With Twilio Voice, you can build unique phone call experiences with one API, to create, receive, control and monitor calls with just a few lines of code. Create an engaging voice experience that you can quickly scale and modify with a wide array of customization options and resources.
Open source Call Center software, based on H.323/SIP protocols.
Contains Nauphone(Software multi-channel IP Phone, with conferences and ability
to integrate with other systems(CRM)) and NauLib(VoIP Library based on OpenH323 and Vovida SIP)
The CRM, sales reporting, and commission tracking tool uniquely tailored to the needs of manufacturers, sales reps, and distributors.
Repfabric is a customer relationship management (CRM) software designed specifically for multi-line sales teams (i.e. reps, distributors, wholesalers, dealers, and manufacturers). It streamlines and simplifies the sales process by providing deep integration with email, contacts, calendars, and deal tracking. The platform enables users to track commissions from CRM to sale, make updates directly from mobile devices, and document sales calls using voice-to-text features.
PNX system develops the middle-ware source code to glue Asterisk with a number of powerful telephony products such as: 1) OpenH323 H.323 stack 2) Vovida SIP stack 3) Bayonne Voice Automation Platform The advantages? An advanced IP PBX supporting the wide
Currently, if someone should decide to develop a Voice over IP application, they must decide on whether to use H.323 or SIP as the underlying protocol.
This stack will allow the user control over which protocol the application uses, i.e. the user puts
Siphon will ultimately be a Software Voice-Over-IP phone using the SIP protocol. It will support quicknet cards as well as traditionnal soundcards. It is developped for Linux (gtk) and the win32 platform.
Your unified business intelligence platform. Self-service. Governed. Embedded.
Chat with your business data with Looker. More than just a modern business intelligence platform, you can turn to Looker for self-service or governed BI, build your own custom applications with trusted metrics, or even bring Looker modeling to your existing BI environment.
Very simple and easy to use self-hosted blog system like wordpress, but sip is most interactive and easy to use whit many new feautures. It's written in PHP and using MySQL database.
SEMS is a free (GPL 2) SIP media and application server featuring announcement, voicemail, conference, ivr, B2BUA and more. Please visit our homepage for more.
SipSpy is a distributed monitoring tool for SIP networks. SpyAgents run on each of the nodes to be monitored, and a SipSpy connects to each of these nodes, receiving information and displaying it in real-time for all the SIP packets monitored.
SIP Video Conferencing server for Video Voice-over-IP clients
OpenVCX can be used to bridge codec level, media format, transmission format, and resolution mismatches to allow multiple clients to interoperate. The server can be used to allow two SIP video endpoints to communicate together as well as host up to eight video endpoints in a unified conference.
WebRTC is supported as a video chat client. Standard SIP video phones are supported, the likes of X-Lite, Bria, Vippie, Linphone, etc.
OpenVCX is a Java based SIP service based on the Mobicents...
The FCP deamon controls the netfilter (linux 2.4) firewall of the localhost by parsing its own protocol. The main purpose of this deamon is to support application level gateways (especially SIP server) to bypass firewall and NAT.
The G.O.N.E. is a softphone (or soft phone) running over the web, fully multi-plataform, it implements the SIP protocol, and is built to work on any SIP server, like Asterisk, and others. GONE will work on a complex sistem, but this will be showed a bit
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...