[Asterisk-Users] ATA-186 and fake ring

[Jim Gottlieb]

I am using an ATA-186 connecting to an asterisk SIP gateway.  When I
dial out through it (via a PRI) to a real number, I notice that I hear
a fake ringback tone.  For example, if I call my voicemail, which
answers without a ring, I still hear a bit of ringback when I call via
SIP.

In fact, if I called a busy number, I never heard a busy.  Just
continuous ringback, as if it's just playing me local ringback until it
sees answer supervision, at which time it cuts the call through.

I alleviated this by adding a line:
exten=_XXXXXXXXXX,3,Busy

so now it goes to busy when the number I call is busy, but, actually, I
still hear a ringback tone first, and then it goes to busy.

Who is generating this ringback?  The ATA or asterisk?  What if I call
a non-suping number with a "the number has been changed" recording?
Will I never hear it because audio will never be cut through without
answer supervision?

The relevant lines from my extensions.conf:

; match any US, and strip leading 1 off
exten=_1XXXXXXXXXX,1,StripMSD,1
; dial outbound on trunk group 1
exten=_XXXXXXXXXX,2,Dial,Tor/g1/BYEXTENSION
; if we don't put this in, we'll hear ringback forever on a busy number
exten=_XXXXXXXXXX,3,Busy 

Thanks for putting up with this relatively green asterisk user...