[Asterisk-Dev] SIP INVITE vs TO URI

[Kevin P. Fleming]

William Lloyd wrote:

> The provider sends the destination as part of the TO field.  In the  
> example the asterisk host is behind NAT.
> 
> For example
> INVITE sip:s at 192.168.100.20
> TO: <sip:15145551213 at 69.23.45.200>
> 
> or
> 
> INVITE sip:s at 192.168.100.20
> TO: <sip:15145551212 at 69.23.45.200>
> 
> Both get handled by Asterisk s destination.

The target URI of the call is what is presented in the INVITE request 
line of the SIP packet. The To: header is _not_ used for call routing.

If you are running CVS HEAD or a 1.2 beta release, you can use the 
SIPHEADER() function to read the To: header in the dialplan and decide 
where to send the call from there.