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<DIV><FONT face=Arial size=2>Hello,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>i have set up an asterisk sip to h.323 convertor,
it is working OK. The only problem i have is this :</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>For example when my identity is <A
href="mailto:12345@some.domain">12345@some.domain</A> , and i call a sip number
from a sip phone, the called party sees my identity (caller identity) as <A
href="mailto:12345@some.domain">12345@some.domain</A>, which is the way it has
to be.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>But when i call from the same phone with the same
identity a h.323 endpoint (asterisk converts), the h.323 endpoints sees my
identyty as '12345'. So asterisk is deleting everything after the @
(included).</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>How can i make that the oh323/asterisk sends the
whole SIP URI as caller identity to the H.323 network?</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Thankk you</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Oliver</FONT></DIV></BODY></HTML>