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<DIV><SPAN class=673485018-06042004><FONT face=Arial color=#0000ff size=2>Just
to follow up, it does not matter what codec I use, and when I listen to the call
on the far end, I can hear a very quick blip that sounds like the correct tone,
but it is not long enough for an IVR to recognize. Is there a way to boost the
length of this tone in Asterisk? Any help would be greatly
appreciated.</FONT></SPAN></DIV>
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<DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma
size=2>-----Original Message-----<BR><B>From:</B> Brian J. Rathman
[mailto:brian@ilk.com]<BR><B>Sent:</B> Tuesday, April 06, 2004 1:29
PM<BR><B>To:</B> asterisk-users@lists.digium.com<BR><B>Subject:</B>
[Asterisk-Users] Passing DTMF<BR><BR></FONT></DIV><!-- Converted from text/plain format -->
<P><FONT size=2>Does anyone know how I can pass dtmf digits from a SNOM 200 to
a cisco</FONT> <BR><FONT size=2>AS5300 with * in the media stream.
Unfortunately, the only way I can get the</FONT> <BR><FONT size=2>calls to
connect is with t or T at the end of the Dial() statement and then</FONT>
<BR><FONT size=2>that picks off the dtmf digits. I have tried the
canreinvite=yes on both the</FONT> <BR><FONT size=2>phone peer and the gateway
peer and I still have to add the T to the Dial</FONT> <BR><FONT
size=2>statement to make the call complete. Any suggestions???</FONT> </P>
<P><FONT size=2>Thanks,</FONT> <BR><FONT size=2>Brian</FONT> </P>
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