[Freeswitch-users] Where to go from here?

[T Fred Farmington]

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.