[Asterisk-Users] sip.conf and SIP client host= not recognized in some cases

[Ing Isianto Istiadi]

Dear All, I have the configurations like this:
Newest cvs asterisk, newest cvs zaptel, 2 budgetone firmware 1.0.4.55, 1 fxo
card, slackware 9.1, kernel 2.6.5
I have strange problem with budgetone and *.
1. When the call from pstn arrive at *, it rings the budgetone, I pick the
call up just fine. The problem arises when I try to transfer the call to
another extension (123), after I check at *, * only sees digit 1, * doesn't
see 23, so * return there's no extenstion to budgetone. So I try to transfer
the call to my parked calles (ext 7), it works, I can picked it up using the
same budgetone or the other one. Then I try to transfer the same call again
to another extension (123), strangely it works. It seems that the budgetone
can only do transfer with more that 1 digit if I have transferred the same
call to 1 digit extension first.
2. in my sip. I've tried the two dtmf setting (rfc and info) with the same
setting in my budgetone (rfc and info).
3. In my extention there's tT.
4. In my sip show peers, the time is 5ms.
5. in my * debug, there's the message saying destroying call.

How do I fix that problem?
I need every help guys.
Thanks