303 projects tagged "Telephony"

Download Website Updated 06 Jul 2013 chan-sccp-b

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Pop 126.17
Vit 9.36

chan-sccp-b is an extension of the original chan-sccp implementation for the Asterisk soft-PBX. It lets you hook up a Cisco/SCCP Phone to your Asterisk server using the SCCP protocol, which works a lot better than the SIP firmware on the same phone. It provides full phone functionality instead of just a simple SIP channel provider. It offers functionality like shared lines, hotline functionality, guest login, dynamic speeddials, private line automatic ring-down (PLAR), personal softkey configurations, Dundi support, SCCP extended dialplan functions, manager support, and custom device state buttons.

No download No website Updated 21 Sep 2011 SIP Inspector

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Pop 65.75
Vit 2.25

SIP Inspector is a tool to simulate different SIP messages and scenarios. You can create your own SIP signaling scenarios, customize SIP messages, and monitor incoming and outgoing messages. The tool can play RTP streams from a pcap file.

Download Website Updated 29 Jan 2010 PlugPBX

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Pop 30.00
Vit 39.78

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 20 Oct 2009 SafiWorkshop

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Pop 43.42
Vit 41.06

SafiWorkshop allows you to design, test, debug, and deploy advanced process flow applications (like IVRs) from a single, unified development environment. For database work, it includes a fully featured SQL query editor. SafiWorkshop also includes some Asterisk PBX specific features like an audio prompt editor.

Download Website Updated 11 Nov 2011 Asterisk eSpeak

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Pop 69.63
Vit 2.73

Asterisk eSpeak provides the "eSpeak" dialplan application, which allows you to use the eSpeak speech synthesizer with Asterisk. This module invokes the eSpeak TTS engine locally, and uses it to render text to speech. It supports the following languages: Afrikaans, Albanian, Armenian, Cantonese, Catalan, Croatian, Czech, Danish, Dutch, English, Esperanto, Estonian, Finnish, French, Georgian, German, Greek, Hindi, Hungarian, Icelandic, Indonesian, Italian, Kannada, Kurdish, Latvian, Lojban, Macedonian, Malayalam, Mandarin, Norwegian, Polish, Portuguese, Romanian, Russian, Serbian, Slovak, Spanish, Swahili, Swedish, Tamil, Turkish, Vietnamese, and Welsh. It supports 8kHz and 16kHz sample rates to provide the best possible sound quality along with the use of wideband codecs. It works with Asterisk 1.6, 1.8, and 10.

Download Website Updated 11 Nov 2011 Asterisk Flite

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Pop 72.97
Vit 2.73

Flite For Asterisk provides the "Flite" dialplan application, which allows you to use the Flite TTS Engine with Asterisk. This module invokes the Flite TTS engine locally, and uses it to render text to speech. It supports voice selection and 8kHz or 16kHz sample rates to provide the best possible sound quality along with the use of wideband codecs. It works with Asterisk 1.6.x, 1.8, and 10.

Download Website Updated 01 Dec 2009 SafiServer

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Pop 54.33
Vit 1.46

SafiServer is an engine that powers the applications called Saflets that control one or more Asterisk PBXs. These can be used for IVR applications such as information, account management, nearest dealer routing, and more. It also acts as an application repository for multi-developer environments.

Download Website Updated 18 Apr 2011 Appkonference

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Pop 58.60
Vit 2.84

Appkonference is a high performance voice/video conferencing system for Asterisk. It is a fork of appconference, and it focuses on reliability and scalability. Appkonference has been tested on both Asterisk 1.4 and 1.6.X. Both Asterisk installations supported more than 1200 participants at a time.

No download Website Updated 14 Apr 2012 chan_nms

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Pop 31.18
Vit 1.00

CHAN_NMS is an Asterisk PBX channel driver that supports the NMS Communications (acquired by Dialogic) Open Access line of products, which includes E1/T1 boards for the PCI(-X), PCIe, and cPCI(e) buses.

No download Website Updated 27 Apr 2009 MeetmeAutoMute

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Pop 24.58
Vit 1.00

MeetmeAutoMute is a utility for Asterisk that can automatically mute participants in a Meetme conference. This can be handy for when you are using the conference to broadcast some audio (like during a speech or presentation, and do not want to have people interfere with the speaker. It is written in Python and uses the AMI through the pyst library. Everything is configurable through a config file, and can be controlled at run time through the use of "signals". An "init.d" style bash script is included that will start the app as a background daemon.

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