Search Results for "c call tree"
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Showing 50 open source projects for "c call tree"
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Amazon Connect connect-rtc-js
Provide softphone support to AmazonConnect customers
connect-rtc.js provides softphone support to AmazonConnect customers when they choose to directly integrate with AmazonConnect API and not use the AmazonConnect web application. It implements Amazon Connect WebRTC signaling protocol and integrates with browser WebRTC APIs to provide a simple contact session interface that can seamlessly integrate with Amazon Connect StreamJS. In a typical amazon-connect-streams integration, connect-rtc-js is not required on parent page. Softphone 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... -
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pcapsipdump is libpcap-based SIP sniffer with per-call sorting capabilities. It writes SIP/RTP sessions to disk in a same format, as "tcpdump -w", but one file per SIP session (even if there is thousands of concurrent SIP sessions). Getting started: http://pcapsipdump.sf.net/
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Telemarketing (Junk) Call Blocker
Blocks unwanted calls using caller ID and a modem
Program to block junk calls. Connects to a modem and listens for a caller ID. It then reads strings from a blacklist file and scans them against the caller ID. On a match, it sends commands to the modem that terminate the call. RECENT ACTIVITY: (11/2014):Added Raspberry Pi support for two USB modem brands. (01/2015): Added Raspberry Pi support for USR5686G modem. (06/2015): Added full-function program for ATian voice/FAX modem. (01/2016): Added jcEdit.py file edit program. (05/2016):... -
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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
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... -
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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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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
The project has moved! 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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The Gigaset 3070 isdn is the ancestor of a family of ISDN DECT base stations allowing the connection of a PC via USB or DECT. This project is building the Linux support (drivers and frontend utilities) which Siemens neglected to provide.
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11
atinout
AT commands as input are sent to modem and responses given as output.
This program will read a file (or stdin) containing a list of AT commands. Each command will be send to the modem, and all the response for the command will be output to file (or stdout). Example, to hang up any ongoing call: $ echo ATH | atinout - /dev/ttyACM0 - ATH OK $ -
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AVAYA CMS API Server
CMS API Server with web site
Free version of AVREMO.CMS.API exposing REST Services. With this software you can browse for all CMS reports. Design a report with AVAYA CMS Supervisor designer, application automatically expose data as XML or JSON. You can also browse reports with your preferred browser (desktop or mobile) and execute CMS operations like modify dictionaries, modify agent skills, modify VDN/Vector associations and more. Application package include : - A responsive WEB server interface (compatible... -
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BI4sftp enables any Asterisk PBX to execute a scheduled secure file transfer of the Call Detailed Records related to the telephone traffic. These data can be collected and analyzed by the BI4Data Service available through web interface (www.bi4data.com)
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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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tui is a user interface for mobile phones with keypads. It runs on text console or under nano-X, and communicates with modem part with AT commands. It runs well on Siemens SX1 running linux, and allows you to call from it and answer calls.
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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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tcliaxc is a Tcl extension which provides you access to iax voip protocol via iaxclient library. tcliaxc lets you use a Tcl package to do basic iax operations such as registering to a server, make a call, handle iax event and so on.
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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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Asterisk based SIP b2bua realization. It's patched asterisk and some AGI script. Features: full vovida's b2bua radius emulation, extended radius attributes, radius failover, LCR, Call failover, Codec based routing and other useful things.
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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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DAHDI for HFC-S PCI A with OSLEC
(Latest DAHDI + HFC-S PCI A support + latest OSLEC) in one git.
This project is for people who want to access one or more HFC-S PCI A based ISDN BRI cards over the DAHDI interface and maybe want to use other DAHDI hardware in parallel, too. DAHDI driven hardware can be used by Asterisk, FreeSWITCH, Yate and other telephony software. The HFC-S PCI A based cards are very cheap and thus ideal for private BRI users. The OSLEC is being developed since a long time ago and said to be very good. The superiority of "hardware echo cancellers" is a myth. This... -
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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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Quintum Tenor VoIP Gateway Call Detail Record (CDR) Advanced Server. This CDR Server can collect Call Detail Records (CDRs) from multiple Tenors. The data is collected simultaneously and continuously. Supported MySQL and file storages.
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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
