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.