David Josephson,
Not off-base, but you haven't made it all the way home yet. This is
another layer of the puzzle, and again we are not talking about apples
and apples here. Circuit switched means that there is a (real or
virtual) circuit that takes data on an input port and delivers it to
an output port somewhere. Packet switched means that each packet of
data is examined by each port it passes, to see where it should be
sent. Normally this layer of VoIP traffic is handled not in Asterisk,
but in a router. You could run the router on the same Linux box that's
running Asterisk (and send packets to different Ethernet ports
depending on their destination address) but normally this task is
handled by a separate router. There is a small computational overhead
associated with adding and decoding Ethernet packets but the main
routing work is done outside Asterisk, and isn't too intensive. You
could read up on TCP/IP routing and understand how this works in more
detail.
We plan on using a Gb switch with 100 Mbps ports to handle the routing.
It's not something you can take a look at in my experience. Some of
the Bell System training material that comes up on eBay is good. You
need to follow the progress from circuit-switched voice telephony
circa 1930 through modern TDM, and then look at the development of
TCP/IP switching separately.
75 years of telephony and network technology to cover, eh? Looks like
it's going to be a long weekend. ; )
No sound card, no monitor. Recording to the various file formats is
possible, as Herman mentioned.
This seems like an odd limitation to me. Any idea why it's designed so
that you must have a sound card to digitally record calls? They could
always be moved to another box in order to listen to them.
Your reference picture is fine ... but note that Asterisk can be the
TDM/VoIP gateway, particularly when Digium releases their DS3 card
(644 voice channels!) working, a lot more cheaply than a standalone
box from some hardware vendor.
I'm not sure that the DS3000P is in our timeframe. I am interested in
knowing how it will perform, considering more than two Digium quad-span
cards currently overload the CPU with interrupts. It seems that Monitor
cannot handle digitally recording more than ~50 concurrent calls,
either. Maybe these limitations are being addressed as we speak.
Thank you for sharing your knowledge with me,
Matthew Roth
http://voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian
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