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RTP text/t140 Library is a reference implementation for RTP Payload Type for Text Conversation (RFC 4103). The library has source code for encoding and decoding RFC 4103 data, and may be used either as a plug-in to JMF or in a separate RTP sender/receive
Olyo is an open source project which aims at implementing a prototype for Peer-to-Peer Session Initiation Protocol (P2PSIP) and providing real-time multimedia services over Internet based on this prototype.
Olyo is promoted by MINE lab, BUPT.
Java version of Dialogic's SwitchKit API. This API is actually a more "Java Centric" API, with listeners, filters, and a very extensible and modular framework that finally uses JNI to access the native libraries provided by Dialogic.
SipManager is a full-feature IP-Centrex PBX web app for hosted/multi-client PBX based on Asterisk and OpenSer. VoIP/ToIP platform (PBX, Fax, workflows, etc.). User-friendly web interface. Realtime architecture with SQL database. ASP. http://www.ovvoe.com
This is a patch for Asterisk PBX that makes it usable as a call agent in PacketCable networks. It adds support of NCS messaging to Asterisk PBX. NCS, in its turn, is supported by the majority of vendors who produce EMTAs.
The Jive API is a lightweight Java framework for writing IVR applications.
The premise for the Jive API is to allow powerful IVR applications to be written without the user having to worry (or know) about the underlying IVR platforms.
RAGUI is a Ruby on Rails based package for managing an Asterisk(tm) PBX in real time mode. Support for extension provisioning, queues, customized scripting and reporting are provided with the system.
MOR is Billing System for Asterisk PBX. It supports postpaid/prepaid billing with LCR and Fail-Over. Realtime Asterisk management using web interface. Reporting and advanced rating. Based on fast C application for Asterisk and Ruby On Rails GUI.
SIP to Skype Gateway/Bridge/Converter/Adapter. Make and receive SIP to Skype Calls and Skype to SIP Calls. Call Skype users using speed dial or use SkypeOut. Make SIP calls from Skype using a SIP provider or SIP PBX. Use as a Skype Trunk with a PBX.
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Web based Asterisk Open Source PBX management suite that adds a lot of features such as night service toggle, operator, music on hold management... Currently only in Brazilian portuguese.
Sipana is a distributed SIP monitoring tool to monitor the SIP signaling behavior using sequence diagrams and providing SIP end-to-end performance metrics through a centralized Web interface. For more information please visit http://sipana.org/
Astercard is a asterisk addon that will enable an asterisk PBX to accept credit card payments over the telephone.The AGI script written in PHP that will process credit card payments with the EWAY payment gateway 30 downloads and no donations *DONATE NOW*
FATS - FATS is a Twisted and Fast Asterisk's Telephony Services. Project contains implementation of FastAGI, AMI protocols for the Twisted framework. Using it you can develop fast and pretty services for the Asterisk IP-PBX.
Asterisk Realtime Administration. Administer realtime extensions, SIP users, voicemail all in realtime via a web interface. Works with the mysql asterisk realtime extension.
Development of software and alternative firmware for Asterisk-appliance D-Link HorstBox (DVA-G3342SB), free kubuntu live-dvd for firmware development, sourcecode (GPL) for original D-Link firmware, modifications/patches.
VoIP Toolkit / Call Control with Integrated Media. High-level Java API for creating SIP enabled VoIP applications. Suitable for either the desktop (softphone, phone applet, incoming call gatekeeper) or server-side (auto attendant, ACD, voicemail).
eSpeak text-to-speech module for Asterisk. This provides the "espeak" dialplan application, which allows you to use the eSpeak TTS Engine as a speech synthesizer in Asterisk.
An tray icon tool to allow automatic Dial using an XMLRPC framework.
The application is based on two separate tools:
- a light desktop application
- a XMLRPC server witch make an PBX abstraction. (first driver is for Asterisk)
Flite text-to-speech module for Asterisk. This provides the "Flite" dialplan application, which allows you to use the Flite TTS Engine as a speech synthesizer in Asterisk.
AGI Audio File Transfer Addons for Asterisk - this is a module for Asterisk that extends Asterisk's AGI interface by adding commands that allow the transfer of audio files to and from Asterisk via the AGI session. Very useful for FastAGI connections.
This library can allow you to connect to the Asterisk Manager Interface (AMI).
The AMI allows a client program to connect to an Asterisk instance and issue commands or read PBX events over a TCP/IP stream.