Asterisk PBX Integration

We have an app dev/integration project: Interested? Introduction Adobe Connect, a web-based collaboration, and interactive meeting software, provides the facility to integrate telephone-based audio conferences with the live meeting room. You can control the audio conference using a Connect meeting room through this integration. Controlling the audio conference includes actions like starting/stopping an audio conference, recording, muting/unmuting audio conference participants etc. Getting Started Adobe Connect’s telephony-related functionalities are handled through a separate service, namely, Adobe Connect Telephony Service. Within the telephony service are the various telephony adaptors, which act as the glue code between their respective conferencing service interfaces and Adobe Connect. It is here that you need plug-in the adaptor as well to integrate our conferencing service. Writing the adaptor In order to write a telephony adaptor, you have to implement a java based interface. The adaptor implementation more or less revolves around the implementation of this interface. The interface has methods to facilitate interaction between the Connect meeting room with the conferencing service. More specifically, it has methods that can be called by the meeting room to do actions on the conferencing bridge end. For e.g. start/stop an audio conference, record the audio conference etc. Additionally, it provides a callback facility so that the conferencing bridge can make a call in the meeting room. For e.g. calling back to indicate that a new participant has joined the audio conference. The Adobe Connect Adaptor (we're seeking you to build) will integrate with our Audio Reservationless Bridge/Platform and allow the following features: 1. Allow an organizer or participant to initiate a conference call through Adobe Connect App. 2. Enable to make dial-out calls to a participant, and merge the call into a conference room handled by Adobe Connect. 3. Display caller ID information and real-time status of all connected participants on the Adobe Connect interface (whether they joined using Connect app, Connect Dial-out or separate DID call). 4. Allow Organizer to mute/unmute individual participants from the Adobe Connect interface. 5. Allow Organizer to split calls into Adobe Breakout rooms using the Adobe Connect interface. 6. Allow Organizer to hang up individual participant calls using the Adobe Connect interface. Once the participants are connected to the Audio Reservationless Bridge/Platform, they will also be allowed to use the available DTMF commands already available for routine conference calls. We have the technical documentation and source file samples available, these code samples are based on Asterisk, and they give us an overall view of how to develop the adaptor and allow to interconnect the Audio Reservationless Bridge/Platform to Adobe Connect. The Audio Reservationless Bridge/Platform already working and has been tested. This API/Interface/Adapter will be considered IP and owned by us. See the attached drawing for additional details. We are seeking an experienced developer intimately familiar with Asterisk. You must beable to start on the project ASAP. Thanks Tom!