Filip Drągowski wrote:
Does phones are registered in Asterisk ? (CLI>sip show peers)
CLI log showing such connections will be usefull (no debug for now).
Thomas Jacobsen wrote:
Hello list,
I'm a asterisk-beginner and could use some assistance with my
configfiles(sip.conf & extensions.conf). I'll attach them to this mail,
and I hope some of you prof's can give me some advice or point me in the
right direction. At the moment by using this configuration, and call
somebody internal i get instant voicemail, and sometimes a call isn't
even possible. Please help! :)
BTW:
[internal]
exten => _ZX[0-8]X,1,DBget(temp=CFIM/${EXTEN})
exten => _ZX[0-8]X,2,Dial(SIP/${temp})
exten => _ZX[0-8]X,3,Dial(SIP/${EXTEN},20)
exten => _ZX[0-8]X,102,Goto(${EXTEN},3)
^^^^
Shouldn't this be "103"?
if 103 is there so failing on 2nd priority will go to 3rd priority...
failing on DBget will go to 3rd and this looks ok for me.
Failing on DBget and user not registered will always go to voicemail
So in this config, it will be the same if he leaves away priority 102.
exten => _ZX[0-8]X,4,VoiceMail(u${EXTEN})
exten => _ZX[0-8]X,104,VoiceMail(b${EXTEN})
exten => _ZX[0-8]X,5,Hangup
And if then VoiceMail fails for some reason, it will just hangup, right?
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