Re: [Asterisk-Users] Re: Re: T1/DS1/ISDN PRI
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
[Asterisk-Users] Re: Re: T1/DS1/ISDN PRI
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 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? 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. 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. ___ 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
Re: [Asterisk-Users] Re: Re: T1/DS1/ISDN PRI
Max TNT's are pretty cheap they'll need to price it accordingly. On Thu, 28 Apr 2005, David Josephson wrote: 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. ___ 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
Re: [Asterisk-Users] Re: Re: T1/DS1/ISDN PRI
On April 28, 2005 08:02 pm, Matt Klein wrote: Max TNT's are pretty cheap they'll need to price it accordingly. Be careful. MaxTNTs are fine, but you need a fully unlocked shelf controller to do SIP/h323. You also need their fancy ethernet card; for whatever reason their standard ones don't work. You also need enough modem/dsp cards (newer ones only, the old ones don't work) to run the number of channels you want. Yes they're pretty inexpensive but not just any one will do. There are lots of little caveats. -A. ___ 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
