Dial in/out problem with asterisk 1.2-beta1, someone help?


#1

Hello,

I have this slight delemma with asterisk 1.2-beta1.
This is how i configured my in/out dialplan:

— sip.conf :: [i]
register => +3589XXXXX:******:user@provider.fi/1000

[provider.fi]
type=peer
secret=*******
username=user
dtmfmode=inband
fromuser=+3589XXXXX
fromdomain=provider.fi
host=provider.fi[/i]

— extensions.conf ::
exten => _0XXXX.,1,Dial(SIP/${EXTEN:1}@provider.fi)

The server register OK and i can call out via provider.fi normally, but when somebody calls in the server matches the user in sip.conf and ignores my request to dial extension ‘1000’.

And naturally if remove the user from sip.conf. Im not able to dial out, but when people dial in it goes to the extension ‘1000’. What am i doing wrong? this same configuration worked on asterisk 1.0.9.
Any pointers how to get this to work?