[asterisk-users] Sip 1.4.x DTMF detection not working

[John Millican]

Hello 
I have a setup where i have 2 asterisk servers connected over the public 
internet with plenty of bandwidth, NAT on one side only.  If i use IAX 
between the two *'s dtmf is flawless.  If I use SIP, DTMF detection is around 
30% or less.  I have an exten to dial into and check DTMF: 

exten => NPANXXxxxx,1,Answer(); (actual number blanked for privacy)
exten => NPANXXxxxx,n,Read(userChoice|ogm/intro|4||1|4);
exten => NPANXXxxxx,n, SayDigits(${userChoice});
exten => NPANXXxxxx,n,Hangup();

When i dial in and use IAX between the servers i always get all 4 digits, If I 
dial in using SIP between the two servers with dtmfmode=rfc2833 or 
dtmfmode=inband I MIGHT get 1 or 2 digits.  If i use dtmf=info and I dial 
slowly I usually get 4 correct digits, but not consistently enough to call it 
good, maybe 85%. If I dial 1 2 3 4 quickly I get 1122 or 1223 or the like.

I would like to use SIP as the voice quality "seems" to be better, matter of 
opinion I am sure but...

Both Asterisk's are 1.4.x on SUSE 10.2 x86_64 kernel 2.6.18.2-34
AMD opteron Dual-Core AMD Opteron(tm) Processor 2212 
and
Dual Core AMD Opteron(tm) Processor 180
2GIG memory

I have searched voip-info and google and didn't find anything that looked 
relevant, maybe just my search words.  I do seem to remember something on the 
list about this a couple months ago but I can not find it or I am remembering 
incorrectly.  
Any suggestions will be greatly appreciated.
Thank You,
JohnM