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.
Yes it will roll over for outbound calls.
exten => _X.,1,Dial(Zap/g0/${EXTEN})
exten => _X.,2,Dial(Sip/${EXTEN}@provider)
Asterisk will know if g0 channels are filled. If so it moves on to the next priority being the sip trunk.
What about inbound call, if the POTS line is being used, will it roll over to any open SIP trunks? Thanks for the rply.
No. That would have to be setup on the telco side.