Is there a way to implement some type of user number or call back system to 
integrate the users in the web portal with the users in the audio bridges.

Scenario #1:
User calls in to audio bridge in asterisk, says name, etc. User is fully 
participating in audio bridge.
User then logs in as a participant or any other level of user to the web 
session and is given a notice to enter a certain unique passcode into the audio 
bridge.
Upon entering the unique passcode, the user is then recognized as having audio 
over the phone bridge in the web conference user list.

Scenario #2:
User logs into web conference, is displayed a pop-up stating that to use phone 
audio to type in their direct number.
Upon submitting their direct number, a call is initiated from the server and 
joins the user to the audio bridge.
The system also marks a phone/mic next to users name in the web conference.

This helps to merge the users in the audio bridge and the users in the web 
conference so that you don't have to take two roll-calls and it minimizes any 
other attendee confusion.

I'm pretty good with Asterisk and can configure the call-back contexts, and how 
to pass the call into the conference bridge once the user answers. I'm not good 
at java or web programming.
I would love to help out making this happen and other Asterisk/SIP 
improvements, I just don't know how to do it all.

Thanks


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