Were having issues with Asterisk boxes registering to a trunk on our FreeSwitch box.

We are sending the DID number called down the trunk in the to: field, however Asterisk doesn’t seem to like this.

Is there anyway to manipulate the INVITE to be INVITE sip:DID@ instead of what is currently INVITE sip:user@

Im hoping there is something like sip_invite_to_uri, however google is failing me.

Regards