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<FONT face=Tahoma size=2><B>Da:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>Per conto di
</B>antonio<BR><B>Inviato:</B> sabato 30 settembre 2006 17.27<BR><B>A:</B>
asterisk-users@lists.digium.com<BR><B>Oggetto:</B> Re: [asterisk-users] Sip
answer one side , ring other side<BR></FONT><BR></DIV>
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<P><SPAN class=578232215-30092006><FONT face=Arial size=2>when i make the call ,
on the xlite side i see the call connected but for the sip gateway the call is
ringing and even the phone (PSTN side) is ringing.</FONT></SPAN></P>
<P><SPAN class=578232215-30092006><FONT face=Arial size=2>I thing that is only
Asterisk send to xlite the signal of connect . </FONT></SPAN></P>
<P><SPAN class=578232215-30092006><FONT face=Arial size=2>Is there any
configuration to set ??</FONT></SPAN></P>
<P><SPAN class=578232215-30092006><FONT face=Arial
size=2>Thanks</FONT></SPAN></P>
<P><FONT face=Arial size=2></FONT> </P>
<P><FONT face=Arial size=2></FONT> </P>
<P><FONT face=Arial size=2></FONT> </P>
<P><FONT face=Arial size=2>Date: Sat, 30 Sep 2006 08:21:21 +0800</FONT></P>
<P><FONT face=Arial size=2>From: Leo Ann Boon <leo@datvoiz.com></FONT></P>
<P><FONT face=Arial size=2>Subject: Re: [asterisk-users] Sip answer one side ,
ring other side</FONT></P>
<P><FONT face=Arial size=2>To: Asterisk Users Mailing List - Non-Commercial
Discussion</FONT></P>
<P><FONT face=Arial size=2><asterisk-users@lists.digium.com></FONT></P>
<P><FONT face=Arial size=2>Message-ID:
<451DB881.2080308@datvoiz.com></FONT></P>
<P><FONT face=Arial size=2>Content-Type: text/plain; charset=ISO-8859-1;
format=flowed</FONT></P>
<P><FONT face=Arial size=2>antonio wrote:</FONT></P>
<P><FONT face=Arial size=2>> Hi,</FONT></P>
<P><FONT face=Arial size=2>> the scheme is this :</FONT></P>
<P><FONT face=Arial size=2>> </FONT></P>
<P><FONT face=Arial size=2>> xlite ---> Asterisk ---> SIP gateway
---> PSTN</FONT></P>
<P><FONT face=Arial size=2>> </FONT></P>
<P><FONT face=Arial size=2>> </FONT></P>
<P><FONT face=Arial size=2>> When i make a call with xlite (sip) to asterisk
on the display of </FONT></P>
<P><FONT face=Arial size=2>> xlite i see that the call is connected but the
phone is still ringing ..</FONT></P>
<P><FONT face=Arial size=2>You must configure your gateway to NOT answer the
call before making the PSTN call. Some gateway call that '1-step'
dialing.</FONT></P></DIV></BODY></HTML>