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<DIV><FONT face=Arial size=2>We need to configure asterisk to authenticate two
sip ATAs, but the stream must go directly from one to another ata without
tuching asterisk.</FONT></DIV>
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<DIV><FONT face=Arial size=2>Is this possible adding canreinvite=yes into
sip.conf?</FONT></DIV>
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<DIV><FONT face=Arial size=2>is it true laso if asterisk doesn't recognize the
spd (t38)?</FONT></DIV>
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<DIV><FONT face=Arial size=2>thanks</FONT></DIV>
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<DIV><FONT face=Arial size=2>Rosario</FONT></DIV>
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