[asterisk-dev] chan_sip SIP Authentication

[Johansson Olle E]

>
> A Workaround is to put the originally called number in the To header -
> but this is ugly as To based routing it is against all RFCs.
´
...and not using the contact URI bind to the AOR in the REGISTER as a  
reqeust-URI
on the INVITE also breaks the RFCs...

I've found devices who DO NOT accept calls to the URI they register  
with Asterisk...
Which is why you can find both IP and port registred with us in the  
SIP_PEER function...

/O