asterisk users - Dec 2005 - incoming dtmf handling by ATA devices ?

sorry if the answer is well known but i couldn't find
a relevant pointer.

I am trying to figure out if/how it is possible to
connect a dtmf-controlled device (e.g. answering machine)
to an ATA, and how to configure asterisk to achieve this.

A bit of expermients with a HandyTone 286 shows that
my ATA only produces audible tones on the phone when
using inband dtmf and ulaw codec.  Other options
(rfc2833, info) do not produce any audible sound,
though the SIP or RTP message do get delivered.

Am i missing something ?

cheers
luigi

> sorry if the answer is well known but i couldn't find
> a relevant pointer.
> 
> I am trying to figure out if/how it is possible to
> connect a dtmf-controlled device (e.g. answering machine)
> to an ATA, and how to configure asterisk to achieve this.
> 
> A bit of expermients with a HandyTone 286 shows that
> my ATA only produces audible tones on the phone when
> using inband dtmf and ulaw codec.  Other options
> (rfc2833, info) do not produce any audible sound,
> though the SIP or RTP message do get delivered.
> 
> Am i missing something ?
Not sure this will help much, but just tested the following:
 C7960 -> asterisk(a) -> iax2/gsm -> asterisk(b) -> spa3k

The sip definition for asterisk(b) to the spa3k is rfc2833 and g711u.

When a call is completed between the C7960 and the spa3k, pressing
any key on the C7960 results in dtmf being heard on the analog phone
attached to the spa3k.

An ethereal inspection of the sip packets flowing into the spa3k does
not indicate the presence of rfc2833-formated packets. Therefore it
would appear that either asterisk(a) or asterisk(b) is actually generating
the dtmf tones inband. The dtmf tones are always approx 100 ms in duration.

You might take a look at an ethereal trace of the sip packets delivered
to the ata to see what might be happening.