[Asterisk-Users] X-lite not hanging up / DTMF not present through voipuser.org

[Mark Charlton]

Hi

I have been lurking for a while, but now have a small problem or 3.

1) I have my inbound line via sip from VOIPUSER.ORG and have a simple
extension selection menu on my * box.  Internally the DTMF tones are
present, (for xlite and * on same LAN), however calling in via the sip line
from a pstn doesn't register any tones in asterisk.  I have tried all the
different DTMFMODE settings in the context for the sip.conf file, but
nothing is helping.  Any suggestons?

2) When I dial in from the pstn to the * box via the SIP line I can get into
the * and on to the hard wired externsion. (I can't select manually see
#1!).  When I hand up the pstn * maintains the line, and keeps the line
open.  If answered the SIP xlite stays connected forever, and is not
answered, it goes through the voice mail sequence, and then hangs up after
the voice mail has timed out.  I also have an inbound line for IAX which
works, but I haven't tested the DTMF or hangup yet.

sip.conf http://pastebin.ca/7765
Extensions.conf  http://pastebin.ca/7766
Iax.conf http://pastebin.ca/7767

3) When x-lite is ringing from an inbound call I don't have any hangup
button to reject the call?  Is it just me?

Thanks everyone
Mark 
-----
A NewB hoping to move onto intermediate 
or maybe even competant sometime soon!