>When we call in on the analog line, I can see the call begin in the cli, and >after 15 >seconds I see the call switch over to my sip provider, and after about 30 >seconds I get >the 3 raising tone signals and the call is hungup.
Sorry guys, been a long day staring at the tube:) Answer() followed by a Dial() with an "r" worked. Still curious on why the call was dropped in the first setup when I wasn't answering the call. Is this normal behavior of the telco? Thanks, jlc _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
