[Freeswitch-users] How to Change the TO: SIP Header from "tel:" to "sip:"?

[Yasuro]

Hi!

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!


Yasuro
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