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How to set up a dial-in node for Internet access? (Being my own ISP)

I just found interesting info - with a voice modem you can (if supported) issue a VTS command to generate a tone.

In the US the dialing tone is interestingly not the typical 425Hz tone but a mix of 350 and 440 hz tones.

To do that from the modem you could use it like this:

# resets the modem
ATZE1

# puts the modem into voice mode
AT+FCLASS=8

# opens the connection
AT+VLS=1

# produces the tone for 2550 miliseconds (Maximum time as i see it)
AT+VTS=[440,350,255]

# closes the connection so it can answer
ATH

# answers the connection
ATA
 
g4ugm said:
If you are using a pair of Modems you won't get above 28.8K Faster speeds rely on the "Answer" being ISDN rather than a normal Modem...
Click to expand...

That's fine in my case, most of my modems are internal ISA so slower than 28.8k anyway. I don't think anyone is doing this for the speed :) Interestingly though if you do connect two modems together there will presumably be no low-pass filter applied, so the modems should see perfect line quality - which means the signal should be even cleaner than with a receiver modem on a digital line. Maybe with a V.92 modem you could get at least 48k?

kyodai said:
If you just wanna hear a 425 hz tone then this youtube video was made for you...
Click to expand...

Unfortunately it wasn't made for me, because I'm in Australia and that's not our dialtone :) Of course listening to a YouTube video is a lot different from sitting at a PC and hearing it come out of the modem's speaker!

Interesting point about the voice modem though, I didn't realise they could generate tones like that.
 
glitch said:
If you want to experiment with Asterisk (either directly on a Linux distro of your choice, or one of the pre-packaged ready to go Asterisk-specific distros), you can pick up a cheap Digium Wildcard like the TDM410, plug it into a Pentium 2 or better (I'm using a VIA C3 fanless Mini-ITX machine), and have four analog lines from which you can get real dialtone. You can then set up extensions to allow for real PBX dialing between your systems. Modems do work over these links (I've tried it), the trouble seems to be when you want to connect to a remote system over a VoIP connection.

You can pick up a TDM410 with FXS modules for under $50 if you wait around. In fact, there's one for sale in Australia at the moment, at $36 USD, on eBay. The TDM410 (or larger 8-line TDM800) allow mixing FXS and FXO modules, so if you decide to connect it to a POTS line, you can add a module in as needed.
Click to expand...

Well I bought that TDM410 which came with two FXS modules, but sadly it arrived with two FXO modules instead, with the eBay seller claiming he doesn't know the difference between FXS and FXO (so of course he just picked one at random to mislead buyers I guess.)

At least those modules are interchangeable so I just bought four cheap FXS modules to use with the card. Hopefully they'll arrive soon and I can start some experimenting.
 
Just to comment on the speed issue - I doubt you will get more than 28.8K under any setup. Nearly all telecom equipment does D/A & A/D at 64 KHz @ 8-bit. With most older systems reserving a few of those bits every so often for in-band signaling or presuming they will be used. From that, Mr. Nyquest will tell you 32 KHz max bandwidth while Mr. Shannon will tell you much less with aliasing artifacts on even the cleanest line. The only way modems were able to achieve > 28.8K was to track and sync to the sampling carrier at 64 KHz - with most relying on raw digital access at the head-end - on the downstream channel only - to do that. Thus modem standards were generally limited to 28.8K in the up-link direction and near 56K in the down-link direction by design. Couple two end-consumer modems together, and you will get 28.8K or less bi-directional. You need to algorithmicly craft the modulation standard in the digital domain @ 64KHz to go faster.

VoIP may will cause modems to under-perform from these speeds due to network jitter, voice codecs and enhancement filters, redundant echo cancellation, and other factors.
 
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