I've got a setup with an ISDN trunk (passive card on FreeBSD 6.2 using ISDN4BSD) and SIP phones (Snom).
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My first setting was to have the SIP phones (sip.conf) configured as dtmfmode=inband. That did work for the *-VoiceMail applications and the like but the DTMF did not go through on the trunk.
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Changed to dtmfmode=info for all the SIP phones got the signaling to the ISDN trunk working.
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But the effect was that internal * DTMF recognition in VoiceMail was gone. This I could fix by perpending a "exten => xxx,1,SIPDtmfMode(inband)" to every internal number which needs DTMF detection.
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In short - sip.conf set to dtmfmode=info, use SIPDtmfMode(inband) for internal * applications requiring DTMF detection.

I recently went through a bit of a configuration nightmare with the
Cisco 7905 phone using the g729 codec and Asterisk and I thought I share
it here for anyone who might be searching for help on this in the
future.

The setup is three, 7905 Cisco phones with the SIP firmware attached to
a Asterisk server remotely through a NAT firewall. The Asterisk is
connecting to the PSTN via a SIP gateway (a Mediatrix box) which only
uses g711.

This setup was actually fairly easy but the one nagging problem was the
DTMF tones. We tried numerous configurations with different results.

Sometimes DTMF tones would work on outbound calls but not on the
Asterisk voice mail system. Other times they would work for voicemail
but no tone would be heard on the outside call. Even more frustrating,
sometimes we could get DTMF if the call was placed outbound, but
incoming calls had no DTMF.

Anyhow, here is what I learned.

1. When using a Cisco phone with the g729 codec, your sip.conf should be
as follows (simplified):

XXXXXXX
type=friend
context=local
username=XXXXXXX
callerid=XXXXXXX
secret=XXXXXXX
host=dynamic
mailbox=XXXXXXX
nat=yes
qualify=yes
dtmfmode=rfc2833 ; * See note.
canreinvite=no
disallow=all
allow=g729

• Note: If you use g729 you can not use "inband". Documentation on the

voip-wiki seems to indicate that you should use "dtmfmode=info" with the
Cisco phone but I found this does NOT work end-to-end with outbound,
inbound, and voicemail system.

The settings on the Cisco phone are also very important. They should be:

RxCodec:3 ;g729
TxCodec:3 ;g729
AudioMode :0x00000020 ; DTMF signalling Always out-of-band

• Note: remember you have to buy g729 licenses for Asterisk from digium.

On the flip side, the gateway is set as follows:

mediatrix
type=peer
context=mediatrix
host=xxx.xxx.xxx.xxx
dtmfmode=inband ; inband works with g711 only
disallow=all
allow=ulaw
allow=alaw

I hope this helps someone.