[asterisk-users] rcf2833 DTMF broken in asterisk SIP channel?

[tracinet]

I posted this bug yesterday:

http://bugs.digium.com/view.php?id=10058

but really was hoping that one of you would be willing to try something
simple for me and reply back with your results.

Basically - I have run into a problem where Asterisk RFC2833 DTMF does not
seem to be compatible with large SIP providers such as Level 3 and Global
Crossing.  Can someone who is using rfc2833 DTMF with a non-asterisk SIP
provider try inserting this in their dial plan to see if it works (trying to
see if the DTMF tones are heard on the PSTN side of the equation).  What
happens to me is the first digit gets heard but then silence as the next 8
digits are sent.

extensions.conf:
exten => 5555555555,1,Dial(SIP/sip_provider/5555555555,20,D(123456789))

For your info - here is what I have in my sip.conf:

[general]
disallow = all
allow=ulaw
port = 5060
context = incoming
maxexpirey=180
defaultexpirey=160
canreinvite=no
srvlookup=yes
videosupport=no
nat=no
tos=reliability
dtmfmode=rfc2833

[sip_provider]
type=friend
username=123456789
secret=password
host=10.0.0.1
disallow=all
allow=ulaw
maxexpirey=15
relaxdtmf=yes
dtmfmode=rfc2833
nat=no
insecure=very
canreinvite=no
promiscredir=yes

Thanks in advance for your help!
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