Connecting an existing conference via PJSIP?


#1

Let’s say I have a conference room of 8 users. At some point in the
evening, we need to hook up with a web-based Zoom conference via PSTN.

That means hooking up that existing pool of users phone to a new PJSIP
channel. An admin would dial in, enter a pin, and initiate that
connection.

Sounds really simple, but I’ve read:

https://wiki.asterisk.org/wiki/display/AST/ConfBridge
https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_ConfBridge
https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Configuration_app_confbridge
https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_CONFBRIDGE
https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_CONFBRIDGE_INFO

Perhaps I’m not seeing the wood for the trees, but I’m not figuring it out.

I think I need to set a variable when the admin dials the pin, and
then within that variable, write the name of the new PJSIP channel
which the admin caused to happen.

And then join the existing channel into the conference. That’s the bit
I can’t figure out.

Thanks.

Oh, and finally, when dialling the Zoom channel, I think I need to use

https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_PJSIP_DTMF_MODE
to “PJSIP_DTMF_MODE(inband)”

because in PJSIP wizard, I’ve set “endpoint/dtmf_mode=rfc4733”

otherwise it doesn’t work with everything else.
And of course, Zoom needs audible touchtone digits.

(Before anyone says “Why not use Zoom Sip Connector” - well, £39 per
month for the SIP connector, or 3*10 minutes at 1p/minute (ie:30p) per
month via PSTN is why!


Is order of channels shown by Function_CHANNELS consistently newest first?
#2

What does this mean?

If you already have 8 users in your conference bridge what does the external conference do for you?


#3

Ah, OK - some background. Zoom is a web-based conference system.
PSTN means “public switched telephone network” - ie: basic copper, not VOIP.

Zoom participants can also join a conference via a normal (PSTN) phone call, followed by a fairly long sequence of digits, including pauses, a conference ID and a PIN number.

The people dialling in via the phone conference are very technophobic, older, and have no internet access.
It’s like their chat room. They certainly wouldn’t be up to remembering and entering long dial strings.

The people joining via the Zoom meeting are usually assistance, experts, sighted guides etc, may be based in other parts of the world, and would be using the web-based solution.

At various points in the month, we need to bring these 2 groups together for defined 30 minute periods.
Which is where the Asterisk “phone only” conference “taps into” the Zoom conference by dialling it, using the SendDTMF command to deal with the dial strings, pin etc.

All the seperate parts work well - the missing link is how I join the two; an existing Asterisk conference to a new PJSIP dial channel, without knowing that channel name in advance, on a system where there may be other active calls. However, I DO know that only one call would be initiated to the particular Zoom dial-in number.

In other words, a cron job kicks of at 20:00, initiates the using AMI and then bridges the 2 parties, closing the link 30 minutes later (after a 1 minute warning)

That’s the point at which I suspect there are 10 different really good ways of doing it, but I’ve not figured it out.

Thanks!


#4

I think the easiest way would be to use the Originate command to call the Zoom telephone number and then direct it to a extension in your dialplan that joins the call to your conference in progress.


Is order of channels shown by Function_CHANNELS consistently newest first?