hi there,
I have just downloaded and installed asterisk a couple of days ago, it compiled
correctly and starts up and runs, on a Redhat 9 system freshly installed for
testing. I don't have any extra hardware installed so far, was attempting to
just try out connectivity. I am having some probs with the configuration, maybe
someone out there can give me some tips :
firstly on modifying the sip.conf file I got stuck at the line
register => 1234@mysipprovider.com
What exactly is a SIP provider? is this essential? Leaving the line as it was in
the
sample config file, asterisk crashes the machine after trying to read the
SIP.conf
(Crashes to the extent that the machine freezes .. )
What I would *like* the system to do is as follows :
for now, just take an input call from a softphone and route it through to an
internet
calling gateaway (I have an account with Go2Call) in such a way that I can play
around with the scripts & work out how to bill it ..
in future I'd like to route calls from a number of H323 calling gateways in
different
locations to pass through a central node & bill everything before forwarding
the
calls to a gateway in USA.
If anyone has unlimited patience and feels like helping, it's more than
appreciated.
Finally, are there any good books on the subject ? I'm ok with IP networks
and
the likes, but pretty green when it comes to telephony.
many cheers,
Dave A. Caruana
Malta
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