[Asterisk-Dev] Lost in Translation (DTMF Relay)

[sales at minixel.com]

Now that Jeremy has started, but not finished, the huge
endeavor of fixing H323, perhaps somebody else may
attempt to fix the other real issue that keeps many
people from using Asterisk in a more business-oriented
model: The DTMF Relay is useless, using either protocol
or mode. I did this experiment which failed and I had
to bypass Asterisk: The calls come from A Cisco 5300
using SIP, where the dialpeer has the usual lines

 “dial-peer voice 400001 voip
 destination-pattern 5619531599
 translate-outgoing called 1
 session protocol sipv2
 session target ipv4:xx.xxx.xxx.x
 dtmf-relay cisco-rtp rtp-nte h245-signal
h245-alphanumeric
 codec g729br8 bytes 40
 fax rate 14400
 fax protocol t38 ls-redundancy 0 hs-redundancy 0
 no vad
!

The middle man is Asterisk, and the receiving end is
also another Cisco 5300, configured similarly. Whatever
I do, the dtmf gets lost using any codec except g711
and Inband, which is not exactly DTMF relay at all. It
fails with rfc2833, info, etc. The bridge kills the
dtmf. There are many business models that require this
functionality. In this case, I had to send the call
directly from Cisco to Cisco, because I could not
receive a single DTMF tone on the other end. My client
really would benefit from having Asterisk in the
middle, but to no avail. I use only G729.

Any ideas?