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

[mm_202]

Yasuro,

Try the IRC channel (#freeswitch on irc.freenode.net) for real-time help,
you'll most likely get a faster answer there.

-- MM.

On Sun, Nov 21, 2010 at 10:00, Yasuro <yasuro at yasuro.com> wrote:

>  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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