asterisk users - Dec 2003 - BOOM! Crash when trying to use SIPDtmfMode on an outgoing call!

All,

	Here's a cool one.. I was attempting to call a retarded conferencing
service, and was having problems with it picking up my DTMF.. after trying
all the settings my Sipura SPA2000 offers, I found inband actually works..
unfortunately, I can't get anything else to pick up my inband DTMF
(including asterisk's builtin voicemail! It just times out and says I never
entered a login!).  So, I did some digging around, and figured I might try
SIPDtmfMode to change my DTMF mode when I'm calling out.. that resulted in a
prompt crash, and the info included below out of gdb.  Is it me? Am I
misunderstanding the appropriate use of SIPDtmfMode?  If so, that's fine,
just bonk me on the head with a yellow pages book or something.. Also.. how
can I change the DTMF timing? I think the SIP INFO dtmf I'm sending is too
brief for the conferencing service.. is there any way I can change the
timings?  Finally, how come * voicemail won't recognize my inband digits?
I'm using ulaw from my * box to my Sipura on a local 100megabit switched
lan.

Thanks!
Pat

--> extensions.conf <--
[toll-trunks]
exten => _1NXXNXXXXXX,1,SIPDtmfMode(inband)
exten => _1NXXNXXXXXX,2,Dial,IAX2/userid@voicepulse/${EXTEN}

--> gdb crash <--
[New Thread 278546 (LWP 4192)]
    -- Executing SIPDtmfMode("SIP/1000-9732", "inband") in
new stack
    -- Executing Dial("SIP/1000-9732",
"IAX2/userid@voicepulse/18882245408")
in new stack
    -- Called userid@voicepulse/18882245408
    -- Call accepted by 66.234.228.132 (format ULAW)
    -- Format for call is ULAW
    -- IAX2[voicepulse]/3 stopped sounds

Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread 278546 (LWP 4192)]
0x0808c75d in __ast_dsp_silence (dsp=0x0, s=0xbd7fe774, len=160,
totalsilence=0x0) at dsp.c:969
969             if (accum < dsp->threshold) {
(gdb) Quit