Search Results for "call"
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Showing 47 open source projects for "call"
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VoIP monitor
VoIP SIP and SKINNY quality analyzer and packet / audio recording tool
VoIPmonitor is open source network packet sniffer with commercial frontend for SIP SKINNY MGCP RTP and RTCP VoIP protocols running on linux. VoIPmonitor is designed to analyze quality of VoIP call based on network parameters - delay variation and packet loss according to ITU-T G.107 E-model which predicts quality on MOS scale. Calls with all relevant statistics are saved to MySQL or ODBC database. Optionally each call can be saved to pcap file with either only SIP / SKINNY protocol or SIP/RTP/RTCP/T.38/udptl protocols. VoIPmonitor can also decode audio. -
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BlackBelt Privacy- Tor i2p WASTE VidVoIP
Browse on Tor / i2p - Anon p2p Chat / FileTx, Conf / Video VoIP
Open Source - GPLv3 inc images. *** 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... -
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Settimino
Arduino Ethernet library for communicating with S7 Siemens PLC
Small but efficient communication library that allows your sketches the interfacing with Siemens PLC. Works starting from Arduino UNO with the W5100 ethernet shield. -
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SnapModbus
32/64 bit multi-platform Modbus communication library
SnapModbus is an Open-Source multi-platform suite (library and tools) to manage Modbus communication in strict adherence to the modbus.org specifications, both master and slave side. It supports all the functions described in the documents: * MODBUS APPLICATION PROTOCOL SPECIFICATION V1.1b3 * MODBUS MESSAGING ON TCP/IP IMPLEMENTATION GUIDE V1.0b * MODBUS over Serial Line - Specification and Implementation Guide V1.02 through the standard transport protocols and the most widespread... -
Fully Managed MySQL, PostgreSQL, and SQL ServerCloud SQL handles your database ops end to end, so you can focus on your app.
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Vodia PBX
A Better PBX For Your Business
Vodia PBX has included a complete suite of business telephony features for on-premise and cloud-based telephony deployments. Sign up at https://vodia.com/en/register and receive your free Vodia license. -
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oreka
Enterprise telephony recording and retrieval system
...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 Centers and Contact Centers for Quality monitoring (QM) purposes. -
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Arduino Si4735 radio library
Arduino and Silicon Laboratories Si4735 radio library
...Major features include: • I2C and SPI bus • Supports "original" (Uno, Duemilanove, and their clones), Mega 2560, Leonardo, and Due Arduinos • FM/AM receive modes; including short, medium, and long wave AM • Analog and/or digital audio output • FM RDS and RBDS • Localization of receive bands, channel spacing, FM de-emphasis, and RDS/RBDS • Correctly decodes RBDS PI codes into station's call letters • Can handle a mix of RDS and RBDS stations as found near the US boarders with Canada and Mexico • Example programs I2C and Due support is alpha because I do not have hardware to test these features. -
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Activa brings the Asterisk IP PBX to the call center. Built on top of Asterisk, Activa components enable successful call center implementations adding value in areas such as computer telephony, screenpop&click2dial, agent control, automatic dialing...
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Komendant ERP
ERP, IP PBX, Accounting, CRM, Time tracking, and etc.
Comprehensive business management system The full-cycle automation ERP system - [ Full Freeware ] IP PBX - Complete synchronization with CRM system - Call accounting for PBX (PBX billing) - Phone call control and monitoring - Phone recording systems Accounting, Financial management - Trade management, warehouse - Management and tax accounting Corporate information system - Client relations management system (CRM) - Project, tasks, orders management - On-line conferences (forums) - Instant messenger, chat, sending SMS Staff management system [HRM] - Personnel records, documents circulation - Time and Attendance tracking system - Computer usage monitoring (time tracking) - Work time accounting (five levels) - Accrual and deduction system - Fines system (two levels) - Salary calculation Access control systems Management of access rights to the premises Video surveillance system (IP,Analog) Photofixation, Videofixation Technological process automation -
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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.
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GNU Gatekeeper (GnuGk)
H.323 Gatekeeper for VoIP and videconferencing
...Please find current versions at https://www.gnugk.org/ The GNU Gatekeeper (GnuGk) is a full featured H.323 gatekeeper under GPL license. It supports VoIP and videoconferencing and includes Radius and database support and many call routing functions. The project has moved! Please find current versions at https://www.gnugk.org/ -
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Screen Pop for Avaya PBX
Implement Screen Pop for Avaya PBX
The ScreenPop implements screen pop function for Avaya PBX. Only web application is supported and telephony data such as AGENTID, CLI, DNIS, UCID, SPLIT, CUSTDATA and UED can be passed to the web application. -
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RetroShare
[WARNING] Project has moved to http://github.com/Retroshare/Retroshare
[IMPORTANT] Retroshare is now distributed on Github. Please go to http://retroshare.net to find the latest releases and sources. RetroShare is a cross-platform, secured and decentralized communication system. It lets you to securely chat, share photos, videos, and more with family and friends, using OpenPGP to authenticate peers and OpenSSL to encrypt all communication. -
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csmake
Cloud Service Make.
To make the native application to support cloud service, csmake will genarate a client stub and the service application, it's similar with the webservice, COBAR, omniORB, SOAP, gsoap etc. -
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Sipp is a performance testing tool for the SIP protocol. Its main features are basic SIPStone scenarios, TCP/UDP transport, customizable (XML-based) scenarios, dynamic adjustment of call-rate and a comprehensive set of real-time statistics.
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ss7mon
ss7mon an SS7/SIGTRAN signalling monitor
ss7mon will implement the following features for SIGTRAN and SS7 family of protocols: - Message sniffer. - Call flow tracing. - Traces DB. - Distributed tracing. - QoS monitoring. - KPI monitoring. - Web interface. - CLI command line interface. - ...etc. -
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Call Center Manager Gui Element
Voicent Call Center Manager statistics recording gui
This Gui Element is used by Voicent Call Center Manager to track and update statistics. -
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KEumexWatcher gets information (caller ID, callee, ...) about an incomming call from an Eumex 704 PC LAN or similar OpenCom device. The caller name is taken from the global KDE addressbook.
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CapiSuite is an ISDN CAPI telecommunication suite providing easy to use telecommunication functions which can be controlled from Python scripts. CapiSuite is also distributed with example scripts for call incoming handling and fax sending.
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spss
endnote
This project does NOT incorporate, access, call upon, or otherwise use encryption of any kind, including, but not limited to, open source algorithms and/or calls to encryption in the operating system or underlying platform -
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ktalkwake is a simple kde4 program released under the terms of the GNU/GPL. It helps the user when an incoming talk call arrives.
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SEMS is a GPLv2+ media and application server for SIP based VoIP services. Basic applications like announcement, pre-call announcement, RBT, conference, voicemail, mailbox etc. and lots of example applications available.
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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.
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Open Unified Recording (OUR) is a full featured Linux based Open Source VoIP/SIP Call Recording engine, indexing and retrieval system. The system resides on the network and passively captures SIP sessions. Project is sponsored by UnifiedRecording.com
