<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>Re: [Asterisk-Users] Passing DTMF</TITLE>
<META http-equiv=Content-Type content="text/html; charset=iso-8859-1">
<META content="MSHTML 6.00.2800.1400" name=GENERATOR></HEAD>
<BODY>
<DIV><SPAN class=006224720-06042004><FONT face=Arial color=#0000ff size=2>I have
tried every combination of codec and dtmfmode. I can hear the dtmf tone on the
far end phone, it just appears to be to short. Is there a way to increase the
duration of the tone?</FONT></SPAN></DIV>
<BLOCKQUOTE dir=ltr style="MARGIN-RIGHT: 0px">
<DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma
size=2>-----Original Message-----<BR><B>From:</B> Eric Wieling
[mailto:eric@fnords.org]<BR><B>Sent:</B> Tuesday, April 06, 2004 3:10
PM<BR><B>To:</B> asterisk-users@lists.digium.com<BR><B>Subject:</B> Re:
[Asterisk-Users] Passing DTMF<BR><BR></FONT></DIV><!-- Converted from text/plain format -->
<P><FONT size=2>On Tue, 2004-04-06 at 12:29, Brian Rathman wrote:</FONT>
<BR><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>cantrinvite=yes tells asterisk to, if it can, remove itself
from the</FONT> <BR><FONT size=2>media stream. T and t and r and many
other Dial options tells Asterisk</FONT> <BR><FONT size=2>to stay in the media
stream so it can listen to the DTMF. None of this</FONT> <BR><FONT
size=2>has ANYTHING to do with passing DTMF between the two endpoints
(except</FONT> <BR><FONT size=2>of course passing # for t or T). If you
cannot pass DTMF between the</FONT> <BR><FONT size=2>two endpoints then
something ELSE is wrong. Maybe you are trying to use</FONT> <BR><FONT
size=2>inband DTMF with a compressed codec. Inband DTMF will only work
with</FONT> <BR><FONT size=2>ulaw or alaw codecs.</FONT> </P>
<P><FONT size=2>--Eric</FONT> <BR><FONT size=2>-- </FONT><BR><FONT
size=2>Useful Asterisk Docs (BOOKMARK THEM!):</FONT> <BR><FONT size=2><A
href="http://www.digium.com/index.php?menu=documentation">http://www.digium.com/index.php?menu=documentation</A>
(look at the</FONT> <BR><FONT size=2>"Unofficial Links") and <A
href="http://www.voip-info.org/wiki-Asterisk">http://www.voip-info.org/wiki-Asterisk</A>
and</FONT> <BR><FONT size=2><A
href="http://www.fnords.org/~eric/asterisk/">http://www.fnords.org/~eric/asterisk/</A>
(my site) and</FONT> <BR><FONT size=2><A
href="http://asteriskdocs.org/">http://asteriskdocs.org/</A></FONT> </P>
<P><FONT size=2>_______________________________________________</FONT>
<BR><FONT size=2>Asterisk-Users mailing list</FONT> <BR><FONT
size=2>Asterisk-Users@lists.digium.com</FONT> <BR><FONT size=2><A
href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</A></FONT>
<BR><FONT size=2>To UNSUBSCRIBE or update options visit:</FONT> <BR><FONT
size=2> <A
href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</A></FONT>
</P></BLOCKQUOTE></BODY></HTML>