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

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@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()