Aterisk RealTime unable to listen voice messages

Hello,

I’m really new in this so please bare with me. I have found one tutorial how to implement real time with mysql and dundi.

Everything work fine and I’m able to register sip’s and ring to each other. Also I’m able to leave voice message but when I call to voicemail from the sip it telling me that there is no voice messages.

This is what I have in extension.conf

[code][general]
static=yes
writeprotect=no
autofallthrough=no
clearglobalvars=no
priorityjumping=no

[lookupdundi]
switch => DUNDi/priv

[internal]
exten => 101,1,VoicemailMain(${CALLERID(num)}@VoiceMail)
exten => 101,2,Playback(Goodbye)
exten => 101,3,Hangup

include => dundiextens

exten => _XXXXXXXXXXX,1,Voicemail(su${EXTEN})
exten => _XXXXXXXXXXX,2,Playback(Goodbye)
exten => _XXXXXXXXXXX,3,Hangup

include => lookupdundi

switch => Realtime

[incomingdundi]
exten => _XXXXXXXXXXX,1,Voicemail(su${EXTEN})
exten => _XXXXXXXXXXX,2,Playback(Goodbye)
exten => _XXXXXXXXXXX,3,Hangup
[/code]
If I change this line exten => _XXXXXXXXXXX,1,Voicemail(1001@VoiceMail) I’m able to hear the message.

This is what I see in CLI

== Using SIP RTP CoS mark 5 -- Executing [101@internal:1] VoiceMailMain("SIP/1001-00000002", "1001@VoiceMail") in new stack > 0x7f7528000980 -- Probation passed - setting RTP source address to MY_IP:8042 -- <SIP/1001-00000002> Playing 'vm-password.gsm' (language 'en') > 0x7f7528000980 -- Probation passed - setting RTP source address to MY_IP:8042 -- <SIP/1001-00000002> Playing 'vm-youhave.gsm' (language 'en') -- <SIP/1001-00000002> Playing 'vm-no.gsm' (language 'en') -- <SIP/1001-00000002> Playing 'vm-messages.gsm' (language 'en') -- <SIP/1001-00000002> Playing 'vm-opts.gsm' (language 'en') == Spawn extension (internal, 101, 1) exited non-zero on 'SIP/1001-00000002'

So I just guessing that the problem may be in extensions.conf and dialplan or no? If you need some other info please let me know.

Thank you