My set up with asterisk is the following:
I have one ZAP/g0 trunk (digium card TDM400 with one FXS/FXO), two license g729 codecs and three SIP trunks. Would asterisk roll over the call to the next available SIP trunk if the ZAP/g0 trunk is in use? If it will, how to configure it? Are there any configuration that I need to do with the PSTN line that is being assigned to ZAP/g0 trunk? Thank you.