ok, almost there, just 1 last question
This is my dialplan:
exten => 910,1,NoOp()
same => n,Progress()
;same => n,Originate(PJSIP/6001,exten,default,999,1,,aC(ulaw,alaw,h264)c(999)n(Deurbel))
same => n,Set(i=1)
same => n,While($[${i} < 60])
same => n,NoOp(Confbridge number of participants : ${CONFBRIDGE_INFO(parties,1)})
same => n,GotoIf($["${CONFBRIDGE_INFO(parties,1)}" >= "1"]?startconf)
same => n,Wait(1)
same => n,Set(i=$[${i} + 1]
same => n,EndWhile()
same => n,Hangup()
same => n(startconf),ConfBridge(1,myconferenceroom,default_user)
When my intercom calls 910 , before i did an originate to invite user 6001 , but early media was not possible, so now where the line is commented, i do below AMI command
Action: Originate
Channel: Local/600@default
Context: default
Exten: 999
Priority: 1
Callerid: 999
EarlyMedia: true
Codecs: ulaw,alaw,h264
Async: yes
So i get al call coming on on PJSIP/6001… when i pickup, the Confbridge is stared, in above dialplan , i do a loop and i check if there are parties in conference, so when 6001 answers, the intercom joins the conf too
So 6001 gets the call from extension 999 , thats actually an RTSP video feed, also as early media!!
exten = 999,1,Answer()
same => n,RTSP-SIP(rtsp://admin:Password01@192.168.0.70:554/Streaming/Channels/102,0,asterisk,5060)
exten => 600,1,NoOp()
same => n,Progress()
same => n,Dial(PJSIP/6001,,G(join_caller))
same => n(join_caller),ConfBridge(1,myconferenceroom,marked_user)
same => n(join_callee),ConfBridge(1,myconferenceroom,admin_user)
1 small issue remaining, how can i make sure extension 999 is with audio disabled? in my confbridge i have the setting for marked_user, that its muted, and also that only the marked_user needs to make video
[marked_user]
…
startmuted=yes
[myconferenceroom]
…
video_mode=first_marked