<br><font size=2 face="sans-serif">Let's say you were wanted to terminate
calls onto your Asterisk system but your only available codec was G.729
and you had no control over the remote SIP proxy sending you the traffic.
What would you do?</font>
<br>
<br><font size=2 face="sans-serif">Does anyone have an update on Asterisk
supporting T.38 with SIP?</font>
<br>
<br><font size=2 face="sans-serif">Thanks!</font>
<br>
<br><font size=2 face="sans-serif">chris</font>