The system has a sip trunk with 8 channel, Now this works well with the older version, but not 2.0
any way I used my cell phone to call into the system via DISA and call out via the PBX
but I can't get it to work on this version.
To further elucidate Bob and Andy's input, Cell phones invariable use GSM, that is just the nature of the technology, so any attempt to use any inband signalling will allways fail miserably, 64k > 32k (or vise versa) = bad dtmf it's just fysix . if you don't use rfc2833 you are doomed to failure, If your VSP does the unthinkable and tries to transcode to inband (strangely some do) then you are SOL check with "sip debug" at the asterisk CLI to make sure that the SIP signalling is apparent to both parties
in all legs of the call, particularly the one to the cell-phone, as stated before, some VSP's don't honor that and the user will have to dig deep into his phone to extend the dtmf length, truly bogus I agree but all VSP's are not equal, try another one before you go further. (JM2CWAE)
Now the painful part, and as Dicko said...not all of them are created equal..
You may need to work with your VSP and determine the right settings. Ring and talk to them about your issue, they might have already come across it with another user and will be able to give you some answers....
As Dicko mentioned, the codec could play a part as well. The less transcoding between codecs the better. Please note CODEC's and DTMF are different things, but the codec and transcoding can have an effect (as I mentioned, so can line noise and poor transcoding).
For almost all setups, I stick the GSM codec for all external voice, so this means setting your Trunks to disallow = all and allow=GSM, so that it is the only codec negotiated. If your VSP has a way of locking the codec (as we said they are not all equal), then do that also. The most likely DTMF mode is rfc2833, but try to confirm this with your VSP....
At least you are now heading in the right direction.