[Asterisk-Users] dtmfmode: inband question

[Brian Buhrow]

	Hello folks.  I'm not sure if this is the right list for this
question, but I'll start here.
	If I'm using a SIP provider and I have an entry in sip.conf that looks
like:

[8315551212]
type => friend
...
dtmfmode => inband
...

When I pick up the phone, call someone through this provider, and press
numeric digits to generate dtmf tones, who is actually generating the tones
at the other end?
What I'm noticing is that if I call a pstn line using an entry like this
through asterisk, and then press digits on the SIP phone connected to
asterisk, I hear very short tones on the pstn line instead of the long
tones I generate on the SIP phone.  In addition, if I press digits too
quickly on the SIP phone, where "too quickly" is not very fast at all, many
digits are dropped entirely and do not make it to the pstn phone at all.
It occurred to me that this might be a fixable problem in the Asterisk
source code, but when I read the code itself, it is not clear to me who is
generating these short dtmf bursts, and perhaps it is the fault of the SIP
instrument, a Cisco 7960 running SIP image 6.2, it self.
	So, if anyone can explain to me where the DTMF tones are coming from
when the  dtmfmode is set to "inband", I'd be most appreciative.

-thanks
-Brian