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Spice up your Mumble experience using this recreation of 4 different skins merged and adjusted into a new one. It is tested and running stable using Mumble 1.2.x.
To install: Download, extract and follow the readme.txt contained in the archive.
All credits go to the devs of the individual pieces including Mumble itself.
A simple interface to work with SIP numbers of Asterisk.
Add, delete, and edit SIP numbers in the MySQL database
As well as viewing CDR records
Простой интерфейс для работы с SIP номерами сервера Asterisk.
Добавление, удаление и редактирование SIP номеров в базе MySQL
А так же просмотр CDR записей
OfficeSIP Softphone and Messenger are two enterprise VoIP SIP clients written in C# in .NET Framework. The SIP clients make use of Microsoft UCC API SDK, ensuring the highest quality of audio and video. Compatible with Office Communications Server.
See also open source, cross-platform:
1) simple messenger Brief Msg at http://briefmsg.org
2) MUVConf is a multi-user video conferencing, see demo video http://youtu.be/YrBU-Aqtvrk, download https://code.google.com/p/muvconf/downloads/list
OpenCallshop for VoIP Switch was started to be an alternitive to VoipSwitch Inc. closed source callshop systems, You can extend development of this project and customize it to fit your needs. if you would like to get a custom developed callshop desktop or web software do not hesitate to contact me. if you're reseller this product consider a donation.
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VOIP client/server in python >= 2.6. Audio in/out: ossaudiodev (UNIX like) or SoX. Network: bzip2 compression, speex or ogg audio compression, you can configure all, minimum bytes per second: 350-400 in speex mode: U8, 6 kHz, quality 0, bzip2, buf 4K
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.
SkypeXMPP is a Karaka™ fork aimed to provide Skype to XMPP gateway It supports presence, instant messaging, multi-user chat and voice (thanks to Freeswitch) This product uses the Skype™ API but is not endorsed or certified by Skype.
OpenBTS is an implementation of the GSM air interface (Um) that allows cellular handsets to be used directly as SIP endpoints. It uses a software-defined radio to generate its air interface and uses Asterisk or yate as its network interface.
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Lwazi is a robust telephony platform aiming to facilitate speedy development of experimental applications without sacrificing power by combining Asterisk with the MobilIVR Pythoninterface bundled into a single build with a unified control interface.
It is an appliance software that integrates the best tools available for Asterisk-based PBXs into a single, easy-to-use interface. include astercrm astercc voicemail,fax,et
A web services front-end to the Asterisk Management Interface, written in C# / WCF. Need help to maintain the C# codebase, help to get it working on Mono on Linux, and help to port it to Java / CXF / Axis 2.
jSipgateConnect library allows Java-based access to all functions specified in the 'Sip Advanced Messaging, Usability, Routing and Accounting Interface' specification.
Plug and play provisioning server for snom VoIP phones written in Python.
The server supports the plug and play provisioning feature embedded in snom phones of the snom 3xx, snom8xx and snom MeetingPoint series.
I'm sad to say that, unfortunately, the serMyAdmin project is being discontinued starting in September 1st, 2010. After three years of development, the project no longer makes sense. The main reason to create the serMyAdmin was the lack of a GUI for OpenSER/OpenSIPS. The OpenSIPS control panel fulfill this need and it is not wise to maintain two GUI projects. Our team will support the opensips-cp initiative to avoid splitting the community between two open source projects for OpenSIPS GUI. I...
MurMan, the Murmur Manager, is a user friendly, robust, and attractive web management interface for Murmur servers based on PHP, using Murmur's (server component for the Open-Source voice communication Mumble) Ice RCP.
Sipflanker helps you find IP phones and IP PBXs in your network that may have Web GUIs available for browsing. It also shows their default usernames and passwords so you can test them and make sure they are not vulnerable to attack.
Astmontray is a systray applet written in PyQt so users can monitor whether their PABX servers running Asterisk.org, the open source telephony platform, are up or not. It supports AMI authentication and runs fine on Linux, OSX and Maemo tablets.
CHAROZT is an Asterisk MOH (Music on Hold) Uploader. It is a Linux-based application installed in Linux desktops that converts MP3 to WAV for Asterisk supported MOH.
It can also upload converted files to the Asterisk Server.
Kiax is a softphone (soft phone, VoIP client) with a simple and comfortable userinterface for making VoIP calls to Asterisk PBX. It depends on the iaxclient library to use Asterisk's IAX2 protocol for easy call configuration and audio setup.
Nhabla-Queue: Is a Asterisk report system for inbounds call center campaigns, provide many reports with useful information and graphics for analysis, which can be printed on PDF.
Nhabla-Bill: Is a report system a billing application inspired on Asterisk-Stat by Areski, offering many useful reports for the analysis, add to the billing function.
This is a daemon that runs side by side with asterisk. And translates Emails with an XML body into an Asterisk Call with all variables passed from XML as well. Has functionality like an autodialer
Pops up a URL when a Call on an Asterisk Server is Answered, passing the Extension, Caller ID name and number. Connects to the Asterisk Server using the Manager Interface. Can be used to connect to a web based CRM.