Could you tell me how you can change the To: SIP header from "tel:' format to "sip:"? I found a thread on the mailing list about someone who was trying to do the opposite (http://freeswitch-users.2379917.n2.nabble.com/How-to-change-SIP-To-header-td5300587.html) and it mentions sip_to_uri variable, but just as Mr. David Ponzone wrote there, setting/exporting it before transferring an incoming call through a SIP ITSP does not seem to have any effect.
What I am trying to do is to have FreeSWITCH act as IVR/AA to incoming calls through ITSPs. My setup works fine with Gizmo but not with a Japanese ITSP. I figured the ITSP's SIP server does not conform to the standards entirely, and does not understand or like something FS sends to it, to which it responds by BYE-ing the call without giving any reason.
So I have been doing a lot of trial and error, and now I am reasonably certain the header above is the culprit. Since I could not figure out how to change it by myself, I placed AsterikWin32 as a middleman, and FS responded perfectly. This worked, I believe, because AsteriskWin32 uses the sip: scheme, not the tel: scheme.
Now I have a working setup, but I'd very much like to get AsteriskWin32 out of the way. Your help will be highly appreciated!