.call file to pipe audio to conference room at scheduled time

Hi There -

I’m running latest freepbx at the moment although I dare say what I’m asking for isnt version specific.

I have a number of users in a conference meeting at as soon as a mp3 stream goes live I need to pipe it into the conference room (using MeetMe for the time being, but no particular reason why I cant using ConfBridge if necessary)

I’ve been reading the asterisk 1.4-1.6 training book from http://the-asterisk-book.com/ but I’m missing something

Heres my context at the moment:

exten => s,1,Answer()
exten => s,n,Wait(2)
exten => s,n,Playback(/var/lib/asterisk/sounds/en/what-time-it-is)
exten => s,n,MeetMe(8000,x,secretpassword)

And heres my test .call file I’m pumping into the spool output area:

Callerid: 101010
MaxRetries: 5
RetryTime: 300
WaitTime: 45
Context: custom_playstream
Extension: s
Priority: 1

Extension 101010 is just a virtual extension and I think this might be one of my problems (dont think you can answer a virtual extension??)

I think my main problem is it seems to assume the call is originating from 101010 whereas I don’t need it to originate from anywhere. Sorry if this is already covered, but I’ve been looking everywhere and google searched for items like auto dial out , connect call to conference etc

I will add obviously the playback command there is just a placeholder for either a Playback, Background or MP3Player command for the stream (which would be generated based on a variable passed in on the .call file)

I’m thinking I need to pass the Channel to be 8000 in the call file instead - Would that make sense?