dslreports logo
 
    All Forums Hot Topics Gallery
spc
Search similar:


uniqs
1684
TheGiaKz
join:2014-08-21
United State

TheGiaKz

Member

[Asterisk] PBX... Custom Conference Bridge... FreePBX, Asterisk

Hello,

I am new here. Please forgive me if this has been asked before or answered; I just couldn't find anything related to this anywhere. Basically, what I need to accomplish is the following:

I am trying to figure out how I can create a custom conference bridge. Basically, what I need, is to be able to have a conference that when people dial in, it plays a message and then it asks them for one of three PIN access codes: Administrator, User, and Attendant (used to check the number of listeners connected to the conference). The sequence would follow like this, after they put each PIN:

Administrator: Plays a message and connects to the conference as the main speaker and administrator for the conference.

User: Plays a message asking to provide the number of listeners at the location. Once the user presses the number of listeners, a message would play and then they would get connected to the conference.

Attendant: Plays a message stating that they have entered the Attendant PIN and proceeds to say the sum of all in attendance; number is gathered from adding the "numbers of listeners at each location". Then, it plays the conference for about 10 seconds and disconnects the call.

Please, any help will be greatly appreciated. I am completely stuck and I just need some guidance and assistance. Thank you in advance for your cooperation!

Oh yeah, I am running FreePBX 2.11.0.38 with Asterisk 11.8.1.

- TheGiaKz
Stewart
join:2005-07-13

2 recommendations

Stewart

Member

Is this basically a broadcast / lecture, where all users are muted and only listen to the administrator's speech? Or is it a real conference, where users can ask questions, vote on issues and make comments?

For the latter, Asterisk is IMO unsuitable for meetings with more than half a dozen participants. There are many commercial services, ranging from free to pricey, that allow you to conduct an orderly meeting. The administrator has a web-based interface that allows him to see who is talking, identify and mute unwanted noise sources, see who has silently "raised their hand", etc. The system maintains a Q&A queue and can automatically unmute questioners in turn. Participants who are unruly, or have technical problems with echo or noise can be dropped from the conference. Speaker information can automatically annotate the conference recording and transcription, so you can be sure of who said what.

Though there are many others, you might start by looking at
»www.freeconferencecallhd.com/
»www.turbobridge.com/
»www.zipdx.info/

How will multiple participants at a given location be handled? A $20 speakerphone won't cut it. A Polycom SoundStation or similar is good for up to about eight people. With extension mics, maybe 15. Beyond that, you'll need a professionally installed system.

The enhancements you are requesting would entail quite a bit of scripting. If you want to do this yourself, you might start by looking at the existing source code for conferences, IVR, speaking clock, etc. If you'd rather find a pro to do it, post at »lists.digium.com/mailman ··· risk-biz . You can view the archives before signing up.

I don't know whether this fits your model, but a simpler alternative is to create a web page where users could register and state their phone number and the number of participants. When the form is submitted, they would be given a PIN to access the FreePBX or commercial conference as a normal user. The administrator would just dial in and give the admin PIN.
TheGiaKz
join:2014-08-21
United State

TheGiaKz

Member

Hi Stewart!

Thank you for taking the time to look into this. This is basically a broadcast/lecture, where all users are muted and only listen to the administrator's speech.

battleop
join:2005-09-28
00000

battleop to Stewart

Member

to Stewart
"For the latter, Asterisk is IMO unsuitable for meetings with more than half a dozen participants."

The conference portion of our soft switch is a modified version of the Asterisk module. The problem isn't the number of users it's the lack of crowd control. For every 10 users there is always that one idiot that can't seem to find the mute button on their phone so when you get upwards of 10-20 callers you get several of these guys and it quickly becomes unusable.
TheGiaKz
join:2014-08-21
United State

TheGiaKz

Member

Hi battleop.. Thanks for your input! However, the mic would be muted for all participants, except for the administrator, of course. Would you be so kind in telling me how you modified the Asterisk module for the conferences? Thanks!

battleop
join:2005-09-28
00000

battleop

Member

You would have to rely on the callers to self mute them selves because Asterisk won't do that for you. This seems to be a very difficult task for many people. I've been on calls with some really smart IT people and many fail to find the mute button on their phones.
Stewart
join:2005-07-13

Stewart to TheGiaKz

Member

to TheGiaKz
Unless your broadcast is very time-sensitive, why are you using the phone at all? If you made a YouTube video, podcast, etc., users could listen at a time that suits your schedule, and it is easy to keep track of your audience.
TheGiaKz
join:2014-08-21
United State

TheGiaKz

Member

Yeah, the broadcast is time-sensitive. It lasts for two hours, twice a week. Plus, the audience is elderly and ill people who can't make it to the live meeting, so not many know how to handle a computer.
kaila
join:2000-10-11
Lincolnshire, IL

kaila

Member

The freepbx gui has 'mute on join' as a conference option.

mgraves1
Premium Member
join:2004-04-05
Houston, TX

mgraves1 to TheGiaKz

Premium Member

to TheGiaKz
said by TheGiaKz:

However, the mic would be muted for all participants, except for the administrator, of course.

This is what ZipDX calls "Lecture Mode" it's a parameter of the conference template that ensures that only the formal presenter can speak.

FWIW, if it truly is a one way presentation than perhaps you could take the audio stream out of the telephony domain and put it online. That would overcome the scaling issues of Asterisk conferences.

arpawocky
Premium Member
join:2014-04-13
Columbus, OH

arpawocky

Premium Member

said by mgraves1:

FWIW, if it truly is a one way presentation than perhaps you could take the audio stream out of the telephony domain and put it online. That would overcome the scaling issues of Asterisk conferences.

A hybrid approach would be to set the recording as Music on Hold. Dial in, hear the recording as hold music.