[Asterisk-Users] Unable to create channel of type SIP-please help

[asterisk]

Hi there,

 

I'm having a hard time getting outbound calling to my SIP-->PSTN
gateway. I continuasly get the following result in my log files:

 

Jun 5 10:07:50 WARNING[1568]: No such host: t2y
Jun 5 10:07:50 NOTICE[1568]: Unable to create channel of type 'SIP'
Jun 5 10:07:50 VERBOSE[1568]: == Everyone is busy/congested at this time

 

I make the following context in my extensions.conf file

 

exten => s,1,SetCallerID(${T2Y1})

exten => s,2,SetCIDName(${MYNAME})

exten => s,3,Dial(SIP/${ARG1}@t2y,${ARG2})

exten => s,4,Playback(new/acnt-or-cir-busy-now)

exten => s,5,Hangup

 

I do have a [t2y] reference in my SIP.conf file

 

[t2y]

type=friend

secret=PASSWORD

username=USERNAME

host=budgetphone.nl

dtmfmode=rfc2833

fromuser= USERNAME

fromdomain=budgetphone.nl 

username= USERNAME 

insecure=very 

nat=yes 

qualify=no

register=yes

 

I tried changing to change the reference t2y in the context in
extensions.conf to @budgetphone.nl and @sip.budgetphone.nl, but then I
get a authorization error.

 

Jun 5 10:16:24 WARNING[1568]: Forbidden - wrong password on
authentication for INVITE to '"Guy Soudant" ;tag=as27b52309'
Jun 5 10:16:24 VERBOSE[1568]: -- SIP/sip.budgetphone.nl-adeb is
circuit-busy

 

I'm lost here. Could someone help me out?

 

Thanks

Guy

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