Well I switched over from SIP to IAX2 and had to add caller id in the dial plan before it calls out.
Reference: iax-bug
Now in bound and outbound calls work just fine…
Now if any insight to the problem with the generated dial tone. I have to keep auto patch quiet turned on to do anything with it. If I take quiet=1 out of it in rpt.conf, it breaks and asterisk complained it cannot start dial tone.