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

[Yasuro]

Could someone please help me with this? I am completely stuck and need 
your help desperately...

Yasuro wrote (11/19/2010 8:02 PM):
> 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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