[Asterisk-Users] * to * sip authentication failure

[Chris Stenton]

I have setup my * box to allow unregistered sip users to be allowed access to the sip context sip-in. If someone calls from an unknown standalone sip phone then I get the calls routed through ok. 

If someone adds Dial(SIP/202 at foo.bar)  to their extensions config in * then when they try and dial me my * server generates a "failed to authenticate user" message. How can I let them through to the default context?

Chris