[asterisk-users] How can I connect an existing Confbridge to a new SIP channel when DIALEDPEERNAME is empty?

[Jonathan H]

Asterisk 16.0, PJSIP

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 and
username 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 "dialling out" channel:

https://wiki.asterisk.org/wiki/display/AST/Pre-Bridge+Handlers
> This application sets the following channel variables:
> DIALEDPEERNAME - The name of the outbound channel that answered the call.

But  DIALEDPEERNAME is  empty. Can anyone please suggest where I might be
going wrong here, and how to complete this? Thank you!

[bcab-dial-zoom]
exten => s,1,Answer()
same => n,Dial(PJSIP/0203456789 at voipfone-201,,U(bcab-send-dtmf))

[bcab-send-dtmf]
    exten => s,1,Wait(1)
    same => n,Verbose(1,***Dialled channel is ${DIALEDPEERNAME});  just
gives :**Dialled channel is
    same => n,Set(dialedname=${DIALEDPEERNAME})
    same => n,SendDTMF(WW123456#WWWWW#WWWWW)
    same => n,Playback(technical-support)
    same => n,SendDTMF(#)

    same => n,SET(GOSUB_RESULT=GOTO:bcab-bridge-conference^s^1)
    same => n,Return()

[bcab-bridge-conference]
exten => s,1,Verbose(1,*** Entered bcab-bridge-conference)
    same => n,Answer()
    same => n,ConfBridge(1234)
    same => n,Wait(55)
    same => n,Hangup()
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