El Viernes, 25 de Enero de 2008, Jon Farmer escribió:
I now want to provide inbound service from other
telephone numbers to these registed users and want to
be able to put the number dialled into the SIP header.
However I am unsure how to represent this in the SIP
message. Can anyone help me?
The original dialed number should appear in "To" header always (except if a
not SIP compliant node modifies it, but it shouldn't occur).
This is: If your proxy receives a call from a PSTN gateways to 12345678 number
the INVITE is something like:
INVITE sip:12345678@IP_proxy SIP/2.0
From: <sip:00112233@IP_gateway>
To: <sip:12345678@IP_proxy>
Later, your proxy will do a ENUM query or any other mechanism in order to map
that PSTN number to a local SIP AoR, so the INVITE will become:
INVITE sip:client001@IP_proxy SIP/2.0
From: <sip:00112233@IP_gateway>
To: <sip:12345678@IP_proxy>
After that, probably the proxy will look for the location of that AoR so it
will do a "lookup()". If that AoR is registered the INVITE will become:
INVITE sip:client001@IP_client001 SIP/2.0
From: <sip:00112233@IP_gateway>
To: <sip:12345678@IP_proxy>
As you see, the "To" hasn't been modified so it will arrive to the user
(client001). For example some UACs as Twinkle show the "To" header.
Of course, most of the deskphones just show the "From" header in the screen.
--
Iñaki Baz Castillo