How to setup call Id

Right now, my ATA’s extension is what shows up as the caller id (CID) on my internal phones for incoming calls from my PSTN line. On my ATA (spa-3000) I have “PSTN CID For VoIP CID:” set to “yes”. How do I tell Asterisk not to pass along the extension number of the FXO port on my ATA, but instead pass along the CID being supplied by the ATA.

Also, I’m assuming that the CID (service) from my provider (AT&T) doesn’t need to be changed on the provider’s end before Asterisk can work with it.

Any input would be appreciated.

EDIT: on the subject – I was curious if I actually have my trunk setup the way it is supposed to be. I have the FXO port on my ATA setup as “just another extension” in sip.conf and extensions.conf. Is that how the trunk is supposed to be treated?

Command> sip show peers

5 sip peers [Monitored: 0 online, 0 offline Unmonitored: 5 online, 0 offline]

Thanks again.

Per this thread the trunk CAN be setup as an extension, so I’m good there.

My first question still stands though - how can I get Asterisk to pass the incoming call’s call ID with a call, instead of the extension number of my ATA’s FXO port.