[asterisk-dev] chan_sip SIP Authentication

[Klaus Darilion]

Johansson Olle E schrieb:
>> 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...

true

> 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...


This one I do not understand. If the device does not accept the contact 
the registered then the device is buggy. So what do these devices accept?

klaus