This is a newbie question : With a simple Asterisk server on a private
LAN, an FXO port to handle the PSTN, and an ADSL connection to the
Net, ie. with no VOSP in the mix… how should I configure Asterisk so
that SIP clients can dial SIP numbers on the Net, such as those below
to perform an echo test?
I tried pasting numbers in XLite, but nothing happens. Do I need to
add something to extensions.conf for magic to happen?