[asterisk-users] DTMF Corruption Problem in 1.4.2 for SIP RFC2833 plz halp

[Justin Tunney]

Hello mailing list,

I have been porting one of my Asterisk boxes to 1.4 and I have
encountered a nasty DTMF problem.  What happens is someone might come
in to my IVR and enter "12345" and what will actually come through
could be along the lines of "12234445".  Sometimes it works, sometimes
it doesn't.

I had this problem with 1.2 back in November but was able to solve it
with the following bug and patch:

http://bugs.digium.com/view.php?id=5970
http://bugs.digium.com/file_download.php?file_id=11337&type=bug

Because DTMF was reimplemented in 1.4, I can not apply the above patch.

Does anyone have any ideas on how I could fix this problem?  Help
would be greatly appreciated.  And by the way, my Asterisk box is
talking to a Level 3 SIP gateway with the following configuration:

[bandwidth]
type=peer
host=x.x.x.x
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=ulaw
context=incoming
reinvite=no
canreinvite=no
nat=no
directrtpsetup=yes
rfc2833compensate=yes
rtpkeepalive=60

Thanks in advance!

- Justin Tunney