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 < [email protected]> wrote: > 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 > > > > ------------------------------ > > DISCLAIMER: > > This email may contain confidential information and is intended only for > the use of the specific individual(s) to which it is addressed. If you are > not the intended recipient of this email, you are hereby notified that any > unauthorized use, dissemination or copying of this email or the information > contained in it or attached to it is strictly prohibited. If you received > this message in error, please immediately notify the sender at Infotech and > delete the original message. > -- WBR Maxim aka solomax
