[Asterisk-Users] DTMF

Sean P. Robertson spr at netxusa.com
Mon Nov 17 12:28:12 MST 2003


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.

You might could do something with Ethereal that would show you what is going on.

Sean
  ----- Original Message ----- 
  From: Scott England 
  To: asterisk-users at lists.digium.com 
  Sent: Monday, November 17, 2003 1:10 PM
  Subject: Re: [Asterisk-Users] DTMF


  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.

  DEBUG[122896]: File rtp.c, Line 942 (ast_rtp_raw_write): Difference is 960, ms i
  s 140

  I assume this is asterisk sending the dtmf tone, but if I switch to rfc2833 I dont see anything.

  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.

  Scott England

  Sean P. Robertson wrote:

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.

http://www.faqs.org/rfcs/rfc2833.html


Sean
----- Original Message ----- 
From: "Scott England" <scott at homelan.com>
To: <asterisk-users at lists.digium.com>
Sent: Monday, November 17, 2003 5:58 AM
Subject: [Asterisk-Users] DTMF


  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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