Using SIP RTP CoS mark 5
    -- Executing [2222@daffyTest:1] Goto("SIP/0004f211f05c-00000000", "ALSAMEET,s,1") in new stack
    -- Goto (ALSAMEET,s,1)
    -- Executing [s@ALSAMEET:1] MeetMeCount("SIP/0004f211f05c-00000000", "9999,MEETCOUNT") in new stack
  == Parsing '/etc/asterisk/meetme.conf':   == Found
    -- Executing [s@ALSAMEET:2] NoOp("SIP/0004f211f05c-00000000", "COUNT IS 0") in new stack
    -- Executing [s@ALSAMEET:3] GotoIf("SIP/0004f211f05c-00000000", "0?addConf") in new stack
    -- Executing [s@ALSAMEET:4] System("SIP/0004f211f05c-00000000", "/usr/sbin/asterisk -rx "originate CONSOLE/dsp extension alsaMeet@ALSAMEET"") in new stack
    -- Remote UNIX connection
 << Call placed to 'dsp' on console >>
 << Auto-answered >>
    -- Remote UNIX connection disconnected
    -- Executing [alsaMeet@ALSAMEET:1] MeetMe("ALSA/hw:0,0", "9999,qt") in new stack
  == Parsing '/etc/asterisk/meetme.conf':   == Found
    -- Executing [s@ALSAMEET:5] MeetMe("SIP/0004f211f05c-00000000", "9999,ql") in new stack
    -- Created MeetMe conference 1023 for conference '9999'


daffy*CLI> core show channels
Channel Location State Application(Data)
DAHDI/pseudo-1644126 s@default:1 Rsrvd (None)
ALSA/hw:0,0 alsaMeet@ALSAMEET:1 Up MeetMe(9999,qt)
SIP/xlitetest-000000 s@ALSAMEET:5 Up MeetMe(9999,ql)
3 active channels
2 active calls
20 calls processed


This is CLI when the audio drops out:
daffy*CLI> core show channels
Channel Location State Application(Data)
DAHDI/pseudo-1644126 s@default:1 Rsrvd (None)

And here is the output when I hang up the call to the conference:

== Spawn extension (ALSAMEET, s, 5) exited non-zero on 'SIP/xlitetest-0000000b'
-- Executing [h@ALSAMEET:1] MeetMeCount("SIP/xlitetest-0000000b", "9999,MEETCOUNT") in new stack
-- Executing [h@ALSAMEET:2] NoOp("SIP/xlitetest-0000000b", "COUNT IS 1") in new stack
-- Executing [h@ALSAMEET:3] GotoIf("SIP/xlitetest-0000000b", "0?exitConf") in new stack
-- Executing [h@ALSAMEET:4] MeetMeAdmin("SIP/xlitetest-0000000b", "9999,K,") in new stack
-- Executing [h@ALSAMEET:5] NoOp("SIP/xlitetest-0000000b", "end") in new stack
-- Hungup 'DAHDI/pseudo-1644126467'
-- Auto fallthrough, channel 'ALSA/hw:0,0' status is 'UNKNOWN'
-- Executing [h@ALSAMEET:1] MeetMeCount("ALSA/hw:0,0", "9999,MEETCOUNT") in new stack
== Parsing '/etc/asterisk/meetme.conf': == Found
-- Executing [h@ALSAMEET:2] NoOp("ALSA/hw:0,0", "COUNT IS 0") in new stack
-- Executing [h@ALSAMEET:3] GotoIf("ALSA/hw:0,0", "1?exitConf") in new stack
-- Goto (ALSAMEET,h,5)
-- Executing [h@ALSAMEET:5] NoOp("ALSA/hw:0,0", "end") in new stack
<< Hangup on console >>