asterisk users - Oct 2018 - How can I connect an existing Confbridge to a new SIP channel when DIALEDPEERNAME is empty?

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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Would really appreciate some help here - into day 4 of trying to bridge a
PJSIP call to an existing confbridge.

There's a fair amount of dialplan and log to show which doesn't really
work
well via plain text email, so I've taken it over to the forum at
https://community.asterisk.org/t/bridging-an-existing-conference-to-a-new-call/76806/7

Many thanks in advance.

On Wed, 24 Oct 2018 at 17:17, Jonathan H <lardconcepts at gmail.com>
wrote:
> 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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