asterisk users - Jan 2010 - Dialing OutBound SIP trunk using Dial() command

Hello users,

i am working on directly calling the numbers from the sip provider of my
choice from asterisk using Dial command as follows.

extensions.conf

[dial-out]

exten => _XXXXXXXXXX,1,NoOp(Dialing out)
exten =>
_XXXXXXXXXX,n,Dial(SIP/1{EXTEN}:password:md5secret:authname:tarnsport at
host:port
, 20,r)
exten => _XXXXXXXXXX,n,Hangup()



//so i am trying to call the number using voip provider details i have

but i am getting the following error in asterisk CLI


SIP/408XXXXXXX:xxxxx::XXXXXXX:udp at xxxxxx
Called 140XXXXXXXX:xxxxx::XXXXXXX:udp at xxxxxx
    -- SIP/xxxxxx-0a155070 is circuit-busy

when i try with other service provider i am getting a similar error in
asterisk CLI

SIP/1408XXXXXXXXX:yyyyy::YYYYYY:udp at yyyyyyyyyyy
 Got SIP response 500 "Nice try" back from 64.xx.xx.xx
    -- SIP/yyyyyyyyyyy-0a16ac20 is circuit-busy


my idea is to allow users to enter their own voip providers for outgoing
calls
so that customer can use his own voip provider


i am NOT LOOKING FOR  A SOLUTION  in  /etc/sip.conf entries

like

register => username:password at myprovider
[myprovider]
usernamesecretfromuserfromdomainhost

any help is appreciated.

Thanks
srinvias
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