RSS 303 projects tagged "Telephony"

Download Website Updated 10 Jul 2010 Asterisk Presence Panel

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Pop 35.89
Vit 1.00

Asterisk Presence Panel is a simple application that allows the user to monitor the status of extensions on multiple Asterisk based PBX systems. The application connects to the Asterisk server using the manager interface. When it starts, it uses a manager command to probe the status of the extension, after which it relies on the extension status messages in the manager session to update its status. The application features contact groups, the ability to click to dial a contact, computer driven dialling for any number, the ability to connect to multiple Asterisk servers, and integration with the system tray on supported platforms.

Download Website Updated 02 Jul 2010 GnuDialer

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Pop 55.27
Vit 1.92

GnuDialer is a predictive dialer for contact centers. GnuDialer currently supports inbound, outbound, open/closer, and auto campaigns. It has a multi-process object oriented design and uses the Asterisk PBX. Gnudialer is separate from any agent interface or CRM, but does include a capable (Java-based) CRM application that uses Firefox, Internet Explorer, and Mozilla.

Download Website Updated 17 Jun 2010 ObexTool

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Pop 76.99
Vit 4.02

ObexTool is a graphical frontend for ObexFTP, which is able to communicate with mobiles and other devices using the Obex Protocol. The Siemens S45, S45i, S25, S35, SL45i, SL45, M50, C55, S55, C65, C65V, Ericsson R320, T68i, Sony/Ericsson T300, Ki700, and Nokia 6230 and 6670 have been reported to work with obexftp, though it should also work with other phones.

Download Website Updated 17 Jun 2010 YateClient

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Pop 34.03
Vit 37.52

YateClient is an universal, skinnable VoIP client that supports H.323, SIP, Jingle, and IAX protocols. This software is part of Yate and is available with the Yate packages for Mac.

Download No website Updated 16 Jun 2010 FreeSentral

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Pop 26.53
Vit 1.00

FreeSentral is an easy-to-use IP PBX based on the telephony engine Yate. Some of its features include call forward, extension groups, call logs, call hunt, call hold, Auto Attendant, call pick up, call transfer, conference, and voicemail. The demo offers a glance into how FreeSentral works. A wizard assists users on their first configuration of the IP PBX.

Download Website Updated 26 Feb 2010 PHP eFax

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Pop 60.40
Vit 2.05

PHP eFax is a PHP class that allows you to send and receive faxes anywhere in the world from your PHP code on your Web server. Processing is usually instantaneous. It interfaces with eFax Developer and works on any platform that runs PHP 4 or later. It is easy to setup and use.

Download Website Updated 02 Feb 2010 Asterisk

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Pop 675.08
Vit 8.04

Asterisk is a hybrid TDM and packet voice PBX (Private Branch eXchange) and IVR platform with ACD functionality. It acts as middleware between the Internet (IAX, SIP, MGCP, Skinny, H.323), telephony channels (like Zaptel, T1, PRI, E1, FXO, FXS, VoIP, VoFR, ISDN, modems, Internet Phone Jack, etc.), and applications (like voice-mail, conferencing, directories, MP3 players, intercoms, etc.). It has many advanced features such as a codec translation API. The base distribution includes several channel backends, as well as applications. However, the beauty of Asterisk is its ability to be extended using its APIs, dynamic module loader, and AGI scripting interface. End users can even write their own applications that run on the system in C or any scripting language of their choice.

Download Website Updated 29 Jan 2010 PlugPBX

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Pop 33.20
Vit 39.33

PlugPBX is a prebuilt, ARM-based Debian system for end users to run Asterisk and FreePBX on the Marvell SheevaPlug low power platform. It includes Asterisk 1.6.1 with compiled DAHDI kernel mods, FreePBX 2.5, Apache2, MySQL, Samba, Munin, Webmin, Avahai, and OpenSSH. It is built on top of Debian Squeeze.

Download Website Updated 13 Jan 2010 MCS MyVoIP

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Pop 86.67
Vit 4.89

MCS MyVoIP very accurately measures the quality and performance of Internet connections for Voice over IP (VoIP) usage by simulating UDP voice data traffic between a server and browser clients. Connections are tested for jitter and packet loss and rated for the supported level of sound quality. The VoIP test can be set to various codecs or customized by packet size, packet rate, and test length. The test can further be combined with a bandwidth speed test or network route diagnostics for more in-depth connection analysis.

Download Website Updated 09 Jan 2010 MMS Decoder

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Pop 86.01
Vit 3.01

MMS Decoder can receive MMS messages, decode them, and display them on a Web page. This is done by acting as an MMSC, which is a server to which MMS messages are sent.

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