Search Results for "service"

.... *** PLEASE NOTE: There are now 2 seperate versions here. *** One is Pre Firefox 57. The other is Post Firefox 57. *** For those providing mirrors, please enable your users to realize this. Vidalia Based, Tor as a Service Solution. MicroSip: enables FREE PC to PC video calling with no account sign-up and no middleman server. WASTE: enables FREE Conference VoIP, chat, file transfer and support. *** AI Powered *** Tor/i2p: enables safer browsing. Tor/i2p Profile: Browse over Tor/i2p on Firefox (Vanilla, ESR), Waterfox (Classic, New), Palemoon , LibreWolf and legacy CyberFox. ...

Enterprise telephony recording and retrieval system with web based user interface. The project currently supports recording voice from VoIP SIP, Cisco Skinny (aka SCCP), raw RTP and audio sound device and runs on multiple operating systems and database systems. It can record audio from most PBX and telephony systems such as BroadWorks, Metaswitch, Asterisk, FreeSwitch, OpenSIPS, Avaya, Nortel, Mitel, Siemens, Cisco Call Manager, Cosmocom, NEC, etc... It is amongst others being used in Call...

SIPSL (SIP Service Layer) is a Programmable Session Border Controller (SBC) and SIP B2BUA, as defined in RFC3261. Written in c++ and multithreaded. The Application logic can be implemented by extending the application class and implementing the call backs.

...https://github.com/medooze/media-server Video Multiconference Media Server with WebRTC support. Provide Multiconference and video broadcasting services to any SIP service. Supports VP8, H264, MP4V-ES, H263 and H263P, continuous presence, RTMP flash broadcasting, adhoc conferences, load balancing and administrative WEB interface. JSR309 driver implementation under development. .

SIGTRAN based Short Message Center (SMSC) - proof of concept This is home of my thesis project which had goal of doing open source based Short Message Service Center to be connected into real mobile network (2G, 3G, ...). As being developed in very limited schedule (bachelor thesis work) it was limited to have very basic functionality: receiving message and deliver it immediately as-is to the receiver. There is no message caching nor different message charset support. It uses following protocol stack: MAP (moForwardSM and mtForwardSM) TCAP (Begin with Invoke + End with ReturnResultLast) SUA (SCCP User Adaptation): ASP UP/ASP ACTIVE + ACK's and CLDT messages only SCTP (Linux Kernel - SCTP implementation) IP Current implementation is tested in SAMK NGN [http://www.samk.fi/ngn] laboratory's 3G mobile network (real Nokia MSC/VLR network hardware with UTRAN RLC and node-B). ...

This Gui Element is used by Voicent Call Center Manager to track and update statistics.

This project is currently under development for three pizza delivery services in my home city. It supports point-of-sale-like recording of in-house and order-by-phone orders located in gastronomy/catering sector (order desk).

The open-source C++ implementation of the Open Mobile Alliance (OMA) specification - Mobile Location Protocol (MLP).

This project is developed as a database application using Qt 4.0 and MySQL 5.0. Main aim of the project is store call details of each customer. At any time (usually end of month) admin can send call details to the selected customer.

ContactList plug-in extension for PJSGUA SIP client. Minimum required version of PJSGUA: v0.7.1

PduLib is a C++ library for converting SMS messages into PDU format. It's a detailed implementation of GSM 3.40 (Technical realization of the Short Message Service) specification for six PDUs of which Short Message Transfer Layer is comprised.

SMSDecoder is a command line tool for decoding binary SMS (Short Message Service) messages, described in standard GSM 3.40 and 3.38.

OpenSMSC is ETSI GSM based realtime, distributed, highly available Short Message Service Center. It transfers short messages between mobile phones(MO and MT) over SS7 network and to applications(ESME) using open standard protocols like SMPP, HTTP & SOAP

HAOpenSS7 is a High-Available Open SS7 Gateway that consists of multinodes providing the intended functionalities needed by the most of the Inteligent-Network(IN) and Value-Added-Service(VAS) applications.

Wavelet EMP is a web-based J2EE ERP system running on PostgreSQL DB. It is in production status and used by more than 100 shops in South Eash Asia. The software team is in Malaysia. Powered by WAVELET.BIZ. For demo/help, email vincent@wavelet.biz

AstTapi, an opensource Asterisk Tapi driver for windows. This allows users of TAPI compliant applications such as Outlook and Act to dial contacts directly from the application using an Asterisk PBX Server.

Time of day service over telephone using Voicent Gateway, a VoiceXML gateway that specially designed for voice modems. A Free version is available for download at http://www.voicent.com/download. Sample code for interactive telephony applications.

PublicVoiceXML is a VoiceXML 2.0 browser, designed to work also on low cost telephony hardware using DTMF navigation with hooks to 3rd party TTS Modules. VoIP support is under development. Tech support and sample applications available.

It's new technology for Development Personal Creation/Customization Services , based on XML standard and DAFSM (Finite State Machine Theory) , using multiple call control protocols.

CaminoTalks is a open source, cross platform videoconferencing system. It is designed to work behind a firewall and can have mutiple connections. It works with all popular Messengers like MSN, AIM, ICQ, GAIM. It overcome Netmeeting/H323.Videoconference.

java telephony API, Dialogic's hardware support, telephony application IDE, mutlichannel telephony applications Server

This project adds database connectivity to the built in IAS (RADIUS) Windows 2000 service. This adds a great deal of flexibility for returning custom attributes, implementing custom authorization methods, etc. It includes PAP and CHAP support.

OpenSVMS is an open source Short Voice Messaging System. SVMS (also known as VSMS or VoiceSMS) is a telecom service which enables users to send voice clips as messages rather than plain text messages.

COMONet is a novel concept in which a voice call is routed via an ad hoc mobile network avoiding the conventional Mobile Service Provider. COMONet is consisting of community of mobile users who stay between two mobile users who are making the call.