asterisk users - May 2006 - asterisk <-> SIP provider, two way connection

Please give me some heads up.

I'm having troube setting up my asterisk connecting to my SIP provider (SP).
Here's the setup.

in sip.conf
I register asterisk to SP using

register => 15551234567:pwd123@sip.abc.com

everything was ok for incoming call until I want to dial out using the same
line. since in order to dial out, i added in sip.conf

[sip_out_to_SP]
type=peer
username=15551234567
fromuser=15551234567
secret=pwd123
host=sip.abc.com
port=5160
usereqphone=yes
call-limit=5

Then I can make phone call out when routed to this channel
SIP/sip_out_to_SP. But I no longer able to receive calls. The sniffer told
me when my SP send asterisk an invite, asterisk returns a 407 to
authenticate SP, which kills the connection.

I tried to change "type =friend" but doesn't work. I have to
either remove
"secret= " or "host=" to make my incoming call back.

So currently, I can only either dial out or receive calls on this sip line.
not both. (I'm not talking about at the same time). Can anyone give me a
hint. REALLY APPRECIATE IT.

Michael
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