It appears to me that the vanilla configuration files that come with FreeSWITCH allow you to run in a very limited mode. As I indicated before I can get my in-house softphones to successfully call into the FreeSWITCH and reach one of the default extensions. When I do this I hear the associated FreeSWITCH default voice recordings But when I try to call in via my outside SIP line, I cannot get the connection. I see the call arrive in the FS_CLI and in the log files, but I only get a Busy signal and no connection is made. Editing my sip_profiles - internal & external xml files did not change that behavior. Either: 1. I made the wrong edits in those files 2. I did not edit enough files 3. or I have no idea what I am doing. I would rather understand what should be done than just 'groping in the dark' making edits here and there based on a partial understanding (or lack of understanding) of some web reference. I thought that the FreeSWITCH 1.2. book would help, but nowhere does it say: If you want to receive 'outside' calls, do this...... My current SIP line is configured very simply. * No SIP authentication required * A single IP address on Port 5060 upd * DTMF type is set to RFC2833 * Rtp will be from 208.85.184.41 udp ports 10,000 to 59,000 (although we are doing voice only at this time) I have a number of things that I would like the FreeSWITCH phone switch to do, but I feel that I need to take 'baby steps'. So my First goal is to just successfully receive an 'outside' call to one of the existing default extensions - similar to how the softphones work now. My second 'baby step' goal would be to to use one of my softphones to place a call OUT on the same SIP line. After that I can look at beginning to understand how to configure the various internal extensions to do various tasks. Thanks for your continued patience and support.