[Asterisk-Users] DTMF

Scott England scott at homelan.com
Mon Nov 17 11:10:41 MST 2003


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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>>Asterisk-Users mailing list
>>Asterisk-Users at lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>    
>>
>
>
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