We changed several customers from older Asterisk systems (version 1.8). All of the systems are version 13. Right from the start we got complaints about dropped calls. I am now wondering if the “Audio” the user hears is the difference. I think that now the system will drop a call that is rejected by the SIP trunk provider. Can I change what the system provides to the phones? In a nutshell if it isn’t ringing to be answered can’t I set it so that they get a busy? I don’t want any of the verbal “Can’t be completed as dialed”, etc…