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<DIV><FONT face=Arial size=2>I apologize. When you said that you were looking in
the SIP debug, I thought that you were expecting the rfc2833 to be a SIP
message.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>You might could do something with Ethereal that
would show you what is going on.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Sean</FONT></DIV>
<BLOCKQUOTE dir=ltr
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=scott@homelan.com href="mailto:scott@homelan.com">Scott England</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asterisk-users@lists.digium.com
href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Monday, November 17, 2003 1:10
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Asterisk-Users] DTMF</DIV>
<DIV><BR></DIV>I dont expect to see an ascii code or such since the tones are
in a rtp stream. But when I place the dtmf type to "info" in the sip.conf and
make a call I see this under asterisk with sip debug on.<BR><BR>DEBUG[122896]:
File rtp.c, Line 942 (ast_rtp_raw_write): Difference is 960, ms i<BR>s
140<BR><BR>I assume this is asterisk sending the dtmf tone, but if I switch to
rfc2833 I dont see anything.<BR><BR>What I am looking for is a way to verify *
is sending the dtmf to the vocal server, since it does not see the dtmf even
though the audio portion is in operation and I know dtmf works between the
vocal server and a cisco AS5300.<BR><BR>Scott England<BR><BR>Sean P. Robertson
wrote:<BR>
<BLOCKQUOTE cite=mid005201c3ad16$8c7aa700$23c9a8c0@spr type="cite"><PRE wrap="">I think that you are thinking of SIP INFO messages if you are expecting to
see something in the SIP messaging. RFC2833 is sent as part of the RTP
packets so you are not going to see a plain text 1,2,3,4,etc in a trace when
using it.
<A class=moz-txt-link-freetext href="http://www.faqs.org/rfcs/rfc2833.html">http://www.faqs.org/rfcs/rfc2833.html</A>
Sean
----- Original Message -----
From: "Scott England" <A class=moz-txt-link-rfc2396E href="mailto:scott@homelan.com"><scott@homelan.com></A>
To: <A class=moz-txt-link-rfc2396E href="mailto:asterisk-users@lists.digium.com"><asterisk-users@lists.digium.com></A>
Sent: Monday, November 17, 2003 5:58 AM
Subject: [Asterisk-Users] DTMF
</PRE>
<BLOCKQUOTE type="cite"><PRE wrap="">I am trying to connect to a vocal server from an asterisk server. A call
is received via iax2 to my asterisk server. I then initiate a SIP
connection to the vocal server. everything works great except dtmf
doesnt work. A cisco 5300 can connect to this vocal server and do dtmf
without a problem. I have my dtmf set to rfc2833 in the general section
of the sip.conf . I can confirm that the channel is in rfc2833 during
the call via show channel. With SIP debug though I dont see any event
for dtmf. I do see dtmf in IAX though.
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