[asterisk-dev] chan_sip SIP Authentication

Johansson Olle E oej at edvina.net
Wed Jan 28 10:54:48 CST 2009

> 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  
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...


More information about the asterisk-dev mailing list