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<DIV><FONT face=Arial size=2>Hi all,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>One short question.</FONT></DIV>
<DIV><FONT face=Arial size=2>When one extension (let's say ATA-186, SIP
image, G.723 codec selected) try to call an external SIP address
like:</FONT></DIV>
<DIV><FONT face=Arial size=2><A
href="mailto:SIP/user@domain.com">SIP/user@domain.com</A>, where another
identical ATA-186 is available with G.723 codec selectrd,</FONT></DIV>
<DIV><FONT face=Arial size=2>after the signaling phase, the call is established
through Asterisk or directly between the two ATAs?</FONT></DIV>
<DIV><FONT face=Arial size=2>There is no G.723 codec available on
Asterisk</FONT></DIV>
<DIV><FONT face=Arial size=2>I need to know this because of the
firewall.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Thanks,</FONT></DIV>
<DIV><FONT face=Arial size=2>Dan</FONT></DIV></BODY></HTML>