Re: [Asterisk-Users] DTMF detection on SIP provider ?

[John Todd]

Hmm... haven't been able to get this to work on my Cisco ATA-186. 
Perhaps I'm trying the incorrect knobs?  I'm making outbound calls 
ATA-186->*->iconnecthere->PSTN.

I've set my ATA-186 to these various settings:

AudioMode: 0x00150015
AudioMode: 0x00250025
AudioMode: 0x00050005
(per the settings for "negotiated", "out-of-band" and "in-band" found 
on Cisco's notes 
http://www.cisco.com/univercd/cc/td/doc/product/voice/ata/ataadmn/ata88sip/sip88ape.htm#1027044

I've set my peer with iconnect to dtmfode=inband, rfc2833, and info, 
and tested each with each of the three AudioMode settings on the ATA 
listed above.

No changes across all of them, except when I switch to "inband only" 
on the ATA, my Asterisk server no longer shows the DTMF pending and 
DTMF sending messages on the console debug (obviously.)  I sometimes 
hear slight snatches of DTMF tones as I press keys, but something is 
muting or snagging them from the audio stream and not allowing them 
to go all the way through.

My CVS updated code is as of one hour ago.

Has anyone made this work with the ATA-186 systems?  Did I just miss 
the magic combination somewhere in there?

JT



try the new "dtmfmode" parameters on the user or peer.  Note they are not
currently valid in the "[general]" section. you can set dtmfmode=inband or
dtmfmode=rfc2833
Mark

On Sun, 9 Mar 2003, Mikael Andersson wrote:

 Hi..

 I just wondering why DTMF are not recognized by aterisk on incoming calls
 from my SIP provider ...
 ANy suggesteions ?`

 > /Mike
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