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@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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