It is my understanding that TDM is circuit switched and VoIP is packet switched. It would seem to me that at some point in a TDM-VoIP gateway, a change from circuit switching to packet switching is happening, and vice versa depending on the direction of the signals. I was just wondering if anyone could detail that process and tell me if it is resource-intensive. If I'm completely off-base, please point me in the right direction.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.
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.It's too bad. A lot of people without any telephone background try to make up stuff using pieces of the old terminology and wonder why they stay confused. They could look it up, but they don't. For instance DID's. DID has a specific meaning and inward service from the PSTN handed off on VOIP isn't it.
Do you have a good, reliable source that I could take a look at?
Is there a way to specify the format? What if there is no sound card on the Asterisk server?No sound card, no monitor. Recording to the various file formats is possible, as Herman mentioned.
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.
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