AFAIU currently "hard" phone users are not listed in the room their sound is being transferred by "SIP Transport"
On Tue, Jan 29, 2013 at 9:43 AM, Jeff Clay < jeff.c...@infotech-enterprises.com> wrote: > 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 [mailto:solomax...@gmail.com] > *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 < > jeff.c...@infotech-enterprises.com> 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 **** > -- WBR Maxim aka solomax
