Dstchannel has no value in Asterisk 14.7.6

Asterisk Asterisk Support
1

Hello!
I used a streaming/SIP project on an older server with Asterisk 11.20.0. Although I copied the configuration 1:1 (with necessary adjustments of course) the dstchannel values for processing via my extensions.conf settings are always empty.

[separator]
exten => s,1,Goto(fun1,s,1)
exten => s,2,Goto(fun2,s,1)


[fun1]
exten => s,1,NoOp(${CONF})
exten => s,n,NoOp(INIT_CHANNEL: ${CDR(channel)})
exten => s,n,Set(SHARED(INIT_CHANNEL,${CDR(dstchannel)})=${CDR(channel)})
exten => s,n,Set(SHARED(CONF,${CDR(dstchannel)})=${CONF})
exten => s,n,MeetMe(${CONF},Akmqd)
exten => s,n,Hangup

[fun2]
exten => s,1,NoOp(${SHARED(CONF)})
exten => s,n,Wait(0.5)
exten => s,n,NoOp(${SHARED(INIT_CHANNEL)})
exten => s,n,MeetMe(${SHARED(CONF)},kxqd)
exten => s,n,AGI(kill_channel.sh,${SHARED(INIT_CHANNEL)})
exten => s,n,NoOp("NOT_IN_THE_MEETME")
exten => s,n,Hangup

This is from my log file:

  -- Executing [00491777000001@sipout_app_sound_voippro:1] NoOp("SIP/mpapp2-00000000", "Starting Dialout Procedure app_sound_voippro INT 00491777000001") in new stack
    -- Executing [00491777000001@sipout_app_sound_voippro:2] SIPAddHeader("SIP/mpapp2-00000000", "P-Preferred-Identity: <sip:+4917683869864@locophono.com>") in new stack
    -- Executing [00491777000001@sipout_app_sound_voippro:3] Set("SIP/mpapp2-00000000", "CONF=00491777000001") in new stack
    -- Executing [00491777000001@sipout_app_sound_voippro:4] Set("SIP/mpapp2-00000000", "FILE(var/lib/asterisk/conferences/c0b9c5d0-7805-11e8-a7ed-c9062511716c)=00491777000001") in new stack
    -- Executing [00491777000001@sipout_app_sound_voippro:5] Dial("SIP/mpapp2-00000000", "SIP/00491777000001@sipout_voippro,30,rG(separator,s,1)") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/00491777000001@sipout_voippro
    -- SIP/sipout_voippro-00000001 is making progress passing it to SIP/mpapp2-00000000
    -- SIP/sipout_voippro-00000001 answered SIP/mpapp2-00000000
    -- Executing [s@separator:1] Goto("SIP/mpapp2-00000000", "fun1,s,1") in new stack
    -- Executing [s@separator:2] Goto("SIP/sipout_voippro-00000001", "fun2,s,1") in new stack
    -- Goto (fun1,s,1)
    -- Goto (fun2,s,1)
    -- Executing [s@fun1:1] NoOp("SIP/mpapp2-00000000", "00491777000001") in new stack
    -- Executing [s@fun2:1] NoOp("SIP/sipout_voippro-00000001", "") in new stack
    -- Executing [s@fun2:2] Wait("SIP/sipout_voippro-00000001", "0.5") in new stack
    -- Executing [s@fun1:2] NoOp("SIP/mpapp2-00000000", "INIT_CHANNEL: SIP/mpapp2-00000000") in new stack
    -- Executing [s@fun1:3] Set("SIP/mpapp2-00000000", "SHARED(INIT_CHANNEL,)=SIP/mpapp2-00000000") in new stack
    -- Executing [s@fun1:4] Set("SIP/mpapp2-00000000", "SHARED(CONF,)=00491777000001") in new stack

A comparable log file from my old server looks like this:

    -- Executing [00491777000001@sipout_app_sound_voippro:1] NoOp("SIP/mpapp2-0000bd87", "Starting Dialout Procedure app_sound_voippro INT 00491777000001") in new stack
    -- Executing [00491777000001@sipout_app_sound_voippro:2] SIPAddHeader("SIP/mpapp2-0000bd87", "P-Preferred-Identity: <sip:+4917683869864@marcophono.com>") in new stack
    -- Executing [00491777000001@sipout_app_sound_voippro:3] Set("SIP/mpapp2-0000bd87", "CONF=00491777000001") in new stack
    -- Executing [00491777000001@sipout_app_sound_voippro:4] Set("SIP/mpapp2-0000bd87", "FILE(/usr/local/asterisk/var/lib/asterisk/conferences/4f14788a-8911-440f)=00491777000001") in new stack
    -- Executing [00491777000001@sipout_app_sound_voippro:5] Set("SIP/mpapp2-0000bd87", "MONITOR_FILENAME=/var/www/bestofwerbung/html/recs/0491777000001") in new stack
    -- Executing [00491777000001@sipout_app_sound_voippro:6] Monitor("SIP/mpapp2-0000bd87", "wav,/var/www/bestofwerbung/html/recs/0491777000001,i") in new stack
    -- Executing [00491777000001@sipout_app_sound_voippro:7] Dial("SIP/mpapp2-0000bd87", "SIP/00491777000001@sipout_voippro,30,rG(separator,s,1)") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/00491777000001@sipout_voippro
    -- SIP/sipout_voippro-0000bd88 is making progress passing it to SIP/mpapp2-0000bd87
    -- SIP/sipout_voippro-0000bd88 answered SIP/mpapp2-0000bd87
    -- Executing [s@separator:1] Goto("SIP/mpapp2-0000bd87", "fun1,s,1") in new stack
    -- Goto (fun1,s,1)
    -- Executing [s@fun1:1] NoOp("SIP/mpapp2-0000bd87", "00491777000001") in new stack
    -- Executing [s@separator:2] Goto("SIP/sipout_voippro-0000bd88", "fun2,s,1") in new stack
    -- Goto (fun2,s,1)
    -- Executing [s@fun1:2] NoOp("SIP/mpapp2-0000bd87", "INIT_CHANNEL: SIP/mpapp2-0000bd87") in new stack
    -- Executing [s@fun2:1] NoOp("SIP/sipout_voippro-0000bd88", "") in new stack
    -- Executing [s@fun2:2] Wait("SIP/sipout_voippro-0000bd88", "0.5") in new stack
    -- Executing [s@fun1:3] Set("SIP/mpapp2-0000bd87", "SHARED(INIT_CHANNEL,SIP/sipout_voippro-0000bd88)=SIP/mpapp2-0000bd87") in new stack
    -- Executing [s@fun1:4] Set("SIP/mpapp2-0000bd87", "SHARED(CONF,SIP/sipout_voippro-0000bd88)=00491777000001") in new stack

It seems that the processing order is a bit different. But can that be the reason? And if so, why is the order so different?

Other differences between old and new server:
CentOS 6.7 vs Centos 7.4
Apache vs nginx

Best regards
Marc

2

Solved! It sounds stupid but it seems that the macros from the extensions.conf in Asterisk 11 only worked due to a bug. The dest channel normally is not overgivable if setting the G option.
Anyway, if someone should find this thread anywhen in the future here is my solution. Please compare it with the code I entered above:

[separator]
exten => s,1,Goto(fun1,s,1)
exten => s,2,Goto(fun2,s,1)


[fun1]
exten => s,1,NoOp(Data 1: CONF=${CONF} INIT_CHANNEL=${INIT_CHANNEL})
exten => s,n,MeetMe(${CONF},Akmqd)
exten => s,n,Hangup

[fun2]
exten => s,1,NoOp(Data 2: CONF=${CONF} INIT_CHANNEL=${INIT_CHANNEL})
exten => s,n,Wait(0.5)
exten => s,n,MeetMe(${CONF},kxqd)
exten => s,n,AGI(kill_channel.sh,${INIT_CHANNEL})
exten => s,n,NoOp("NOT_IN_THE_MEETME")
exten => s,n,Hangup

Also an adjustment in another part was necessary:

exten => _0049ZX.,1,NoOp(Starting Dialout Procedure app_sound_voippro INT ${EXTEN})
same => n,SIPAddHeader(P-Preferred-Identity: <sip:+4917683869864@lochophono.com>)
same => n,Set(__CONF=${EXTEN})
same => n,Set(__INIT_CHANNEL=${CDR(channel)})
same => n,Set(FILE(/var/lib/asterisk/conferences/${SIPCALLID})=${CONF})
same => n,Dial(SIP/${EXTEN}@sipout_voippro,30,rG(separator,s,1))
same => n,NoOp("Call should not end here")
same => n,Hangup()

The underline char made also a difference.

Best regards
Marc