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... _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users