Search Results for "gnu-darwin"
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Showing 239 open source projects for "gnu-darwin"
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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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Siproxd is a proxy/masquerading daemon for the SIP protocol. It allows SIP clients (softphones & hardphones) to work behind an IP masquerading firewall or router.
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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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ICTFax
Open source Fax server software for inbound / outbound internet Faxing
Note: for binaries and installation instruction please visit official website. http://ictfax.org/ ICTFax is Open Source Fax Server software to send and receive Fax over Internet & Emails developed over Freeswitch, ICTCore and Angular Framework , It works with T.38 and G.711 pass through and feature email to fax, fax to email, web to fax, fax to web and also support fax machines to send and recieve Fax using ATA. Installation... -
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ictdialer
SMS, Fax, Voice Broadcasting and auto dialer Software
ICTDialer is an open-source, multi-user auto dialer software designed for voice broadcasting and fax broadcasting. It supports both inbound and outbound communications using advanced telephony protocols such as T.38, G.711 pass-through, and SIP-based VoIP communication. ICTDialer is built on top of renowned open-source technologies, including FreeSWITCH, ICTCore communications framework, and a PHP-based Angular framework. ICTDialer can be used in following faxing scenarios Voice... -
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ictcore
ICTCore: Unified Communications Framework for web developers
ICTCore is an open-source Communications Platform as a Service (CPaaS) designed to empower developers and system integrators to build, deploy, and manage communication-enabled applications with ease. With support for voice, SMS, email, and fax channels, ICTCore provides a programmable communication layer that enables rapid development of ICT-based solutions using standard development skills.Following are few projects developed over ICTCore communications framework ICTFax open source fax... -
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ViciStack - VICIdial Call Center Tools
VICIdial call center optimization tools, monitoring utilities, and per
VICIdial call center optimization tools, scripts, and monitoring utilities for system administrators and call center managers. Includes performance tuning guides, AMD (Answering Machine Detection) analysis tools, compliance automation scripts, and real-time monitoring dashboards for VICIdial-based contact centers running on Asterisk PBX. Key features: - Server performance monitoring and alerting - AMD accuracy tuning and analysis - Campaign optimization scripts - Compliance and DNC... -
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ictagent
A WebRTC-Enabled Chrome Extension for Seamless Call Center
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... -
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IssabelPBX
Issabel PBX - Unified Communications
Open Source and Unified Communications partners created a new platform based on an Elastix® fork (currently purchased by 3CX) to provide the community with continuity, peace of mind and support needed to continue with their PBX and operation developments. Contribute to the funding of Issabel on https://www.patreon.com/issabel -
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Replacement for the SCCP channel driver in Asterisk. Extended features include Shared Lines, Presence / BLF, customizable Feature Buttons, and Custom Device State. Visit our discussion mailing list for help and join us as a developer if you like. The project moved to https://github.com/chan-sccp/chan-sccp
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Elastix
Unified Communications Server
Elastix is a software-based PBX powered by 3CX and based on Debian. An open-standards solution, Elastix is an easy to install and manage UC system compatible with popular IP phones, gateways and SIP trunks. Elastix is complete with unified communications features such as integrated WebRTC video conferencing, chat, presence and softphones and smartphone clients for Windows, Mac, iOS and Android. -
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jphonelite
Java VoIP Softphone (SIP) (replaced by jfPhone)
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... -
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Asterisk Channels Live
This is a PBX dashboard showing the peer,queue,agent at real time
http://www.astchannelslive.com/sourceforge.php This is a dashboard work with Asterisk(FreePBX) any other installation: 1. Showing the peer status 2. queue,agent at real time 3. Hangup, Spy,.... -
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Free Telephony Project
Free software and hardware for telephony
Legacy SVN repositories for projects from David Rowe http://rowetel.com Mailing lists for Codec 2 and SM1000. -
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jfBroadcast
VoIP/SIP Autodialer
VoIP/SIP AutoDialer. Broadcasts a message with the option to transfer person to another number. Now includes new survey options. -
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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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TeeBX
Asterisk based VoIP pbx appliance
TeeBX wants to be a VoIP appliance with a user interface to easy manage a communication platform based on Asterisk (forked from askozia pbx). Website, support, documentation (coming soon): http://www.teebx.com/ -
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CISCO IP Phone Scanner
Scan all CISCO phones in your LAN to get their information
*** Project moved to https://majed.xyz/coding/desktop-application/cisco-ip-phone-scanner-2/ ****** Scan all CISCO phones in your LAN to get their information(Extension,SN,Model,IP,MAC,etc..) The tool is multi-threaded with the capability of Exporting the results to Excel, the down hosts will also be retrieved under the Error List. Search for an entire Class A network The ability to include Switch info (IP/Port-Name) Include name of Phone user (works with CUCM) Export results to... -
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OpenSIPS (former OpenSER) is an GPL implementation of a multi-functionality SIP Server that targets to deliver a high-level technical solution (performance, security and quality) to be used in professional SIP server platforms. IMPORTANT: this is no longer the main hosting for the project. This was moved on GITHUB - https://github.com/OpenSIPS/opensips
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mumsys
Ein Mumble-Server-Webinterface, das in PHP geschrieben wurd
Mumble Admin Pulgin optimiert für php 7.x.x und ZeroC-Ice 3.7 Ursprünglich wurde das Webinterface von Michael Koch Geschrieben der das Projekt am 23.05.2016 Stilllegte. Features create, edit, delete Server User create, edit, delete virtual Murmur/Mumble Server edit Murmur Configs show Channels, edit ACL start, stop, restart virtual Murmur/Mumble Server Channelviewer, internal and external Serverhosting Fully accessable Pluginusers with wide... -
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SipTunnel is used to tunnel SIP UDP datagrams (and also RTP datagrams) through NAT using TCP protocol. It also can be used for many other purposes.
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Accent Tool Suite
Accent interfaces and controls systems through goals and policies
Accent (Advanced Component Control Enhancing Network Technologies) is a comprehensive tool suite that interfaces a variety of communications systems and allows these systems to be controlled through goals (high-level user aims) and policies (lower-level system rules). Accent has been applied to the domains of: o Call Control: for telecommunications, particularly call control in Internet telephony o Home Care: for home automation and telecare, particularly for domestic appliances... -
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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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asterisk-kannel
Asterisk and kannel integration
This project aims to integrate Asterisk IP-PBX with Kannel SMS gateway.
