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Could someone please help me with this? I am completely stuck and
need your help desperately...<br>
<br>
Yasuro wrote (11/19/2010 8:02 PM):
<blockquote cite="mid:4CE65959.5020907@yasuro.com" type="cite">
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Hi!<br>
<br>
Could you tell me how you can change the To: SIP header from
"tel:' format to "sip:"? I found <a moz-do-not-send="true"
href="http://freeswitch-users.2379917.n2.nabble.com/How-to-change-SIP-To-header-td5300587.html">a
thread on the mailing list about someone who was trying to do
the opposite</a> and it mentions <span><span class="highlight">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.<br>
<br>
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.<br>
<br>
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.<br>
<br>
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!<br>
<br>
<br>
Yasuro</span> <br>
</span> </blockquote>
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