[adam71o]

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 (SPA3000) 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

Name/username________Host_______Port____Status
112______________192.168.0.120___5060___Unmonitore d
111______________192.168.0.120___5060___Unmonitore d
101/101__________192.168.0.125___5060___Unmonitored
100______________192.168.0.124___5060___Unmonitore d
trunk_1/112_______192.168.0.120___5061___Unmonitored
5 sip peers [Monitored: 0 online, 0 offline Unmonitored: 5 online, 0 offline]


Thanks again.