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From Jeff Clay <>
Subject RE: red5sip and OM integration
Date Tue, 29 Jan 2013 02:43:21 GMT
Ok, thanks for forwarding it.

I’m not really referring to a PIN for the entire room, but a PIN that a user enters after
already part of the conference to identify/mark them in the web conference as being called
in and on phone audio.

From: Maxim Solodovnik []
Sent: Monday, January 28, 2013 8:28 PM
To: user
Subject: Re: red5sip and OM integration

Hello Jeff,

I'll forward your email to the guy in our team working on SIP.

below are my answers:

currently we have sort of 2 level integration:

1) calls from "hard" phone to the room:
   a) red5sip.enable should be "yes" (enabled by default since 2.1)
   b) "Enable SIP transport in the room" should be CHECKED for the room
   c) red5sip should be configured on the machine
After that SIP extention number will be  created for the room and user can call to the room
from the hard phone using that extention and optional PIN (can be set for the room)

2) calls from "soft" phone to the room: (we using Linphone for the testing since in is available
for Win/Mac/Linux/iOS/Android)
   a) red5sip.enable should be "yes" (enabled by default since 2.1)
   b) "Enable SIP transport in the room" should be CHECKED for the room
   c) red5sip should be configured on the machine
After that any OM user can register himself on Asterist using: <om_username>@<asterisk_address>
with his/her OM password and make call to the room using: <om_room_id>@<asterisk_address>

If you see how all this can be simplified/improved please share your thoughts :)

On Tue, Jan 29, 2013 at 9:09 AM, Jeff Clay <<>>
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.




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Maxim aka solomax
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