MoIP: How do we make modem calls over IP? (and general chat about phone networks and modems, T1, E1, BRI 56k etc)

[MiaM]

If you can easily identify any low pass filters (and/or high pass filters, but that's less likely to cause issues), it might be worth just taking a chance and halving the values of the capacitors for one full signal path.
I get that you probably don't want to invest time in this, but on the other hand it would be cool to solve this. As a bonus if you ever use it for voice you would likely improve the sound quality. My impression is that analogue phones tend to have a frequency response above the 4kHz limit that digital 8kHz sample rate telephony imposes.

 

[maxtherabbit]

If it was as simple as change Cxx sure I'd give it a try, but that schematic is more complex than my patience will allow for at this time

 

[mR_Slug]

I made a very interesting but somewhat disappointing discovery today. My analog PBX, the Panasonic KX-TA824, is causing some type of degradation when using v.90 (PCM) modulation.

To be honest, that is to be expected as a PBX, is essentially an exchange. As I understand it 56k is modem>ONE line>AtoD. To call anywhere at 56K, the local exchange must be digital. I guess 80s-90s tech is going to do some kinda amplification, maybe a Strowger or crossbar would work but then you only have pulse dial. I've had a look about and I'm going to go the same route of separate PBXs for different tasks. Also while two PBX's may use more power than one big one, you don't need to power both. And you don't need to compromise, each is suited to the task.

I'm not sure if Panasonic made any ISDN PRI/BRI PBXs that had that specific in-call pulse-to-tone function. Would suit all my needs then.

I am kinda surprised that you can get 56k on SIP at all. I thought it would only handle fax and slower speeds.

Please do keep us posted of any experiments. Do appreciate you posting your findings.

 

[MiaM]

If I had either the IP telephony version of the CPU card for my Ericsson Businessphone 250 (ASB 150/02), or something that can provide an E1 link to it, I could test if 56k works through it or not.
At the time it was a new product, they proudly stated that fax calls would pass through without difficulty even if both internal and external lines were analogue.

It for sure can do more or less all versions of pulse dialing and whatnot, and it also has a bunch of settings for different impedance and whatnot for various countries, and weird R button things like "R earth" where R would connect one of the phone line wires to ground (apparently to a ground rod (!!)), rather than the regular "R break" that would just be the same as dialing the first digit on a rotary phone. Don't know how much these settings would affect modems.

 

[maxtherabbit]

To be honest, that is to be expected as a PBX, is essentially an exchange. As I understand it 56k is modem>ONE line>AtoD. To call anywhere at 56K, the local exchange must be digital. I guess 80s-90s tech is going to do some kinda amplification, maybe a Strowger or crossbar would work but then you only have pulse dial. I've had a look about and I'm going to go the same route of separate PBXs for different tasks. Also while two PBX's may use more power than one big one, you don't need to power both. And you don't need to compromise, each is suited to the task.

I'm not sure if Panasonic made any ISDN PRI/BRI PBXs that had that specific in-call pulse-to-tone function. Would suit all my needs then.

I am kinda surprised that you can get 56k on SIP at all. I thought it would only handle fax and slower speeds.

Please do keep us posted of any experiments. Do appreciate you posting your findings.

With the modem connected directly to the smartnode and using Anveo direct as the provider, I'm getting completely reliable connects at 50066 which will sustain a file transfer. Sometimes even 52000 when the stars align.

I have the 8-FXS port SN4141 now, so my SN4112 (2-FXS) is available for sale if anyone following along wants to give it a try. I got it pretty cheap and would sell it for what I paid. I'd also consider a trade for any other cool retro hardware

 

[maxtherabbit]

I wonder if the Panasonic is doing some kind of compression, echo cancellation, or VAD or something that kills it. Does it make any strange sounds during the training sequence? If you have a line tap, record it and maybe a snippet where it gets errors, and I might be able to pinpoint the issue more specifically.

I have been using an Adtran TA924e as my """PBX""" for this

When you make a local FXS-FXS V.34 call through the adtran, what round trip latency figure do the modems report (ATI11)

 

[maxtherabbit]

I ask about the adtran because the Patton SN4141 was a fail. There was a quality regression between it and the previous generation with respect to the FXS hardware. I'm getting way more near end echo with the 4141 vs the 4112 and lower SNR. Measured with the same modems calling FXS->FXS (no PBX inline)

 

[MrImprovement]

I ask about the adtran because the Patton SN4141 was a fail. There was a quality regression between it and the previous generation with respect to the FXS hardware. I'm getting way more near end echo with the 4141 vs the 4112 and lower SNR. Measured with the same modems calling FXS->FXS (no PBX inline)

I don't have experience with the Patton but have a lot of experience with the Adtrans; there are many ways to tweak the config file which is similar to classic Cisco style config; you can set gain, etc. there are many tunables that would influence modem operation.

 

[maxtherabbit]

I don't have experience with the Patton but have a lot of experience with the Adtrans; there are many ways to tweak the config file which is similar to classic Cisco style config; you can set gain, etc. there are many tunables that would influence modem operation.

The patron does indeed have gain adjustments. I have experimented with a wide range of adjustments on gain and nothing outperforms 0/0. There are no other "electrical" parameters which can be adjusted.

Are you able to answer my question about the round trip latency reported by the modem when making an FXS-FXS call on the adtran?

 

[MrImprovement]

The patron does indeed have gain adjustments. I have experimented with a wide range of adjustments on gain and nothing outperforms 0/0. There are no other "electrical" parameters which can be adjusted.

Are you able to answer my question about the round trip latency reported by the modem when making an FXS-FXS call on the adtran?

I am not the OP but I am testing another piece of equipment later this week, will see what I can find...

 

[maxtherabbit]

I am not the OP but I am testing another piece of equipment later this week, will see what I can find...

Thanks, I know the question wasn't originally directed to you, but since @famicomaster2 hasn't been back to the thread in a while I thought you may know - since you mentioned familiarity with the Adtran Total Access.

And to both of you: can you recommend any publicly available resources on using the CLI command set for the Adtran? Trying to google manuals/programming guides etc. keeps hitting paywalls.

 

[MrImprovement]

The Adtran software is called AOS ; last I checked the typical sites like manualslib had them.

If you login, run "sh conf" (show configuration), "sh ver" for version, this will tell you what you are running. A02, A04 are very old versions, R10.x or higher, preferably R12,x will have a lot more features. Basic help is available by using the question mark. show ? will show you sub-commands, then doing it in a sub-command will prompt you if there are further options. For instance "sh arp ?" will show you if there are any more options for the arp command.

I can help more, if you get stuck...

 

[maxtherabbit]

I found quite a few knowledge base articles on adtran's support site, but tracking down actual PDF manuals is still proving challenging. It doesn't help that they seem to have "upgraded" their website software somewhere along the way and broke all their internal links.


Is there any reason to avoid the 1st/2nd gen models? They are far more accessible on the secondary market than the 3rds

 

[MrImprovement]

I found quite a few knowledge base articles on adtran's support site, but tracking down actual PDF manuals is still proving challenging. It doesn't help that they seem to have "upgraded" their website software somewhere along the way and broke all their internal links.


Is there any reason to avoid the 1st/2nd gen models? They are far more accessible on the secondary market than the 3rds

The 3rd Gen is not that much different; probably the 2nd gen is the sweet spot on price. They all have dedicated chips for handling the T1 and FXS ports, and then use Motorola ColdFire CPUs running VXWorks (an embedded operating system). The 3rd Gen is still supported if you are paying Adtran for a support contract; but since you probably won't need that, no reason to pay extra.

 

[maxtherabbit]

Alright I've got a first gen 912 on the way. Got it for $65 shipped and I'm returning the patton SN4141

 

[davidrg]

Most of my actual phones are pulse dial, the Panasonic KX-TA824 and the like of analog PBXs are sought after as they do pulse to tone conversion & in call pulse to tone conversion. My Cisco routers support pulse to tone conversion for dialing, but not in call. So you can't use a menu system. Do you know if the Panasonic KX-TD1232 support in call pulse to dtmf?

Picked up a rotary phone recently (an old GEC 332 type) to test out, and at least out of the box in its default configuration when calling from one extension to another the KX-TD1232 doesn't do in call pulse to tone conversion, and its not obvious from the manual if there is a way to turn this on. For a call to an outside line, maybe it does it? At least I could sometimes here a very brief tone on the other end which varied based on the number dialed on the rotary phone. Perhaps a different rotary phone would produce better results, or maybe there are some settings that need to be changed for it to work better? I'm not able to dial out through my ATA for cost reasons, so I don't know if selecting an outside line and then dialing works or not.

Additionally (though this is hardly surprising) by default it doesn't support the reverse (0,1,2,...,9) numbering of the dial used in NZ (and apparently Oslo, Norway). Given its the NZ version of this PBX I'd think if it could handle this at all it would be on by default, but perhaps its buried somewhere in the programming manual under some non-obvious name.

 

[maxtherabbit]

Picked up a rotary phone recently (an old GEC 332 type) to test out, and at least out of the box in its default configuration when calling from one extension to another the KX-TD1232 doesn't do in call pulse to tone conversion, and its not obvious from the manual if there is a way to turn this on. For a call to an outside line, maybe it does it? At least I could sometimes here a very brief tone on the other end which varied based on the number dialed on the rotary phone. Perhaps a different rotary phone would produce better results, or maybe there are some settings that need to be changed for it to work better? I'm not able to dial out through my ATA for cost reasons, so I don't know if selecting an outside line and then dialing works or not.

Additionally (though this is hardly surprising) by default it doesn't support the reverse (0,1,2,...,9) numbering of the dial used in NZ (and apparently Oslo, Norway). Given its the NZ version of this PBX I'd think if it could handle this at all it would be on by default, but perhaps its buried somewhere in the programming manual under some non-obvious name.

It definitely does work on a call to an outside line

 

[MiaM]

Picked up a rotary phone recently (an old GEC 332 type) to test out, and at least out of the box in its default configuration when calling from one extension to another the KX-TD1232 doesn't do in call pulse to tone conversion, and its not obvious from the manual if there is a way to turn this on. For a call to an outside line, maybe it does it?

Are there any case where you want to receive pulses to something that connects like a telephone?

For outgoing calls it makes more sense to do this conversion.

 

[davidrg]

Are there any case where you want to receive pulses to something that connects like a telephone?

For outgoing calls it makes more sense to do this conversion.

No, I don't have any use for in-call pulse-to-tone conversion - I was just curious if it would do it. The PBX is mostly just for in messing around with modems, and receiving the occasional inbound call via the ATA. Placing outbound calls via VoIP is unfortunately much more expensive than just using a cellphone so I've never tried.

 

[MiaM]

Sorry if this has already been mentioned, but for outside calls (both directions) you can use bluetooth with Asterisk to connect to your cellphone to use it as an external line.