Is order of channels shown by Function_CHANNELS consistently newest first?

[lardconcepts]

@david551

Also, if you told us your overall goal, there may be easier ways of achieving it. Using Originate is just a way you are thinking of approaching the goal. Hanging up a channel created by originate is not your real goal.

Sure; OK well this explains the overall goal most clearly, which is:

First person that dials in triggers an outgoing call which connects to a web-based ZOOM conference.
First person and Zoom conference are bridged.
Subsequent callers go straight to the conference.
Last caller to hang up their call to Asterisk (meaning 1 channel (the zoom conference) is left in the conference) causes the outgoing call to terminate.

So I asked this question:

Bridging an existing conference to a new call

For the first caller to a conference, I want to dial out and bridge that conference to a new PJSIP external call. For the next callers, I just want them to join the local Asterisk conference. After the last caller leaves the conference, I want to hangup the call it initiated. Most of this works, but there are two problems - after the dial string is done sending, no further audio flows between the Confbridge conference and the external call. Secondly, I understand that I need the name of the “…

Along the way I encountered this issue with incorrect documentation

Struggling to make sense of sending DTMF and why DIAL is trying to make multiple calls?

Hi there; I’m trying to dial into a Zoom conference, send some digits, wait, send a name, and be “in the room”, as it were. I thought this would work: same => n,Dial(PJSIP/02036950088@voipfone-205,12,r(callWaiting)D(WWW12345W#WW::)) But it didn’t, so I tried all of these: same => n,Dial(PJSIP/02036950088@voipfone-205,12,r(callWaiting)D(:WWW12345W#WW:)) same => n,Dial(PJSIP/02036950088@voipfone-205,12,r(callWaiting)D(::WWW12345W#WW)) same => n,Dial(PJSIP/02036950088@voipfone-205,12,r(callWai…

I then tried phrasing it a different way

Connecting an existing conference via PJSIP?

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/displa…

After trying every single possible combination of options, it seems that “DIAL” followed by confbridge ended any possible further interaction on that channel.

In fact, the solution became clear from this answer:

@johnkiniston

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.

As soon as I used Originate rather than DIAL, everything fell into place. Except one thing - DIAL sets:

• DIALEDPEERNAME - The name of the outbound channel that answered the call.

but Originate doesn’t have an equivalent. But I know I can get a list of CHANNELS - all I need to know is, reliably, which is the channel created by Originate.

If there’s a better way of doing it, I’m very much open to it, but I’ve spent over 40 hours since Aug 3rd, along with help from a former Asterisk dev via email as a second pair of eyes, and no-one over the course of 4 threads and three months has come up with anything closer than “Originate”.

So I’m very definitely receptive to new ideas here! But like I said, the only missing piece for me at this point is getting the correct channel that was Originated.

Thanks