Could you please explain what you want to do, why you want asterisk to register but not take the calls?
You could take the calls into the dialplan (extensions.conf) and dial out from there with an agi script that performed the same thing. If you have canreinvite=yes, asterisk will leave the media back to the clients. It's not a redirect, but it's more Asterisk.
Asterisk SIP channel is not designed as a SIP proxy so I believe this is hard to do. From the code it looks like a good attempt, but since you didn't include any debug output I don't know what did not work for you. Please send SIP DEBUG output.
/O
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