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 _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
