[Asterisk-Users] Issue with DTMF Tones - Codec Issues

[Aaron W]

Topology:
PSTN<-T1 PRI->NEAX2400<-T1 PRI->Cisco 3825<-Ethernet-> Asterisk VoIP server

When I make a call to a VoIP user from the PSTN, the call gets routed
through the PBX, and Cisco.  Because of that the DTMF tones are passed
inband, which I can hear on the VoIP end of the call. However, I have
one extension on asterisk set up so that I can check voice mail when
away from my phone.  When I call that number again via the PSTN, and I
am prompted to enter my extension number Asterisk never "hears" the
dtmf tones.  I have done some digging around, and my guess is that the
issue relates to the codec being used messing up the tones.

Am I on the right track? Is there a ideal way to handle this?  what do
others do?

I have posted my sip.conf below.

Thanks,
Aaron

[general]
port = 5060                 ; Port to bind to
bindaddr = 0.0.0.0          ; Address to bind to
context = default           ; Default for incoming calls (default
context has no routing for security purposes)
;dtmfmode=rfc2833
dtmfmode=inband
srvlookup = yes
disallow=all                ; Disallow all codecs
;allow=g729                  ; Codecs that we allow (in order of preference)
allow=ulaw
;allow=alaw
allow=g729
;allow=ulaw
;allow=all


[3120]
callerid=Aaron Walsh <3120>
type=friend
host=dynamic
canreinvite=no
qualify=yes
nat=yes
setvar=LDPREFIX=1999999
context=XXXXXXX
secret=XXXXX
mailbox=3120 at XXXXX