[Asterisk-Dev] SIP INVITE vs TO URI

[William Lloyd]

That sounds like it would work for me.

Ok, I just did a

find . | xargs grep -i sipheader on the asterisk source and nothing  
comes up....

My source in only a couple days old.  where is this function hiding?

-bill

On 14-Sep-05, at 3:26 PM, Kevin P. Fleming wrote:

> 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.
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