No subject

<br>
"Note: Asterisk is <em>still</em> really not a SIP
proxy in this case. The two legs have different Call-Ids, and so are
different SIP calls. However the SDP descriptors for the audio of the
two calls point directly at each other.
<br>
<div class="bitbox"><br>
This short-circuit way of setting up calls was introduced in 1.4;
for earlier versions, I believe the call is set up initially as two
legs, and then Asterisk reconnects the endpoints directly using a
re-INVITE
"<br>
</div>
<br>
I will retest on 1.6.x becouse on 1.4.x durring switch-over all cals
were lost.<br>
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