Hi all,
I am replying to myself to update this topic with the last thing I have reached maybe someone can help me further.
Actually this is what I have succeeded to do :
Xlite(6000) -> Asterisk -> H323 Gateway -> Analog phone (550)
this is what I have set in my *Now extensions.conf :
[defaults]
exten => 6050,1,VoiceMailMain
exten = 7000,1,Goto(voicemenu-custom-1|s|1)
exten => 550,1,Dial(H323/550@192.168.39.5,10)
actually this makes the analog phone ring and then the call is finished and CLI shows me :
– Executing [550@my-dialplan:1] Dial(“SIP/6001-081e5008”, “H323/550@192.168.39.5|10”) in new stack
– Requested transfer capability: 0x00 - SPEECH
– Called 550@192.168.39.5
– H323/192.168.39.5-3 answered SIP/6001-081e5008
== Spawn extension (my-dialplan, 550, 1) exited non-zero on ‘SIP/6001-081e5008’
Have someone an idea why does the call get disconnected after the first ring ?
I will update the status
Thanks to all