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<DIV><SPAN class=826512808-21052007><FONT face=Arial
size=2>Hi,</FONT></SPAN></DIV>
<DIV><SPAN class=826512808-21052007><FONT face=Arial size=2>I am trying to
configure asterisk to translate between rfc2833 and inband
DTMF.</FONT></SPAN></DIV>
<DIV><SPAN class=826512808-21052007><FONT face=Arial size=2>I have a cisco
gateway which is configured as a trunk, and a cisco IP phone which is registered
to asterisk. The gateway does not support rfc2833 and the IP phone does.
</FONT></SPAN></DIV>
<DIV><SPAN class=826512808-21052007><FONT face=Arial size=2>I tried changing
directrtpsetup to "no", and that didn't help. I tried changing "canreinvite" to
"no", but that didn't help either.</FONT></SPAN></DIV>
<DIV><SPAN class=826512808-21052007><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=826512808-21052007><FONT face=Arial size=2>I tried adding some
device-specific configuaration to sip.conf, and now my calls are rejected with a
status code of "404 not found".</FONT></SPAN></DIV>
<DIV><SPAN class=826512808-21052007><FONT face=Arial size=2>This is what I added
in sip.conf:</FONT></SPAN></DIV>
<DIV><SPAN class=826512808-21052007><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=826512808-21052007>
<P><FONT size=2>[6102]</FONT></P>
<P><FONT size=2>type=friend</FONT></P>
<P><FONT size=2>canreinvite=no</FONT></P>
<P><FONT size=2>host=dynamic</FONT></P>
<P><FONT size=2>dtmfmode=rfc2833</FONT></P>
<P><FONT size=2>[trunk_1]</FONT></P>
<P><FONT size=2>type=peer</FONT></P>
<P><FONT size=2>host=192.168.20.58</FONT></P>
<P><FONT size=2>canreinvite=no</FONT></P>
<P><FONT size=2>dtmfmode=inband</FONT></P>
<P><SPAN class=826512808-21052007><FONT face=Arial size=2>What am I doing
wrong?</FONT></SPAN></P>
<P><SPAN class=826512808-21052007><FONT face=Arial
size=2>Hagai.</FONT></SPAN></P></SPAN></DIV>
<DIV><SPAN class=826512808-21052007><FONT face=Arial
size=2></FONT></SPAN> </DIV>
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