Telephony module (real phone line) for Pi?

[David.WE.Roberts]

I had another 'silent' call this morning, which reminded me that I'd
really like to screen incoming calls and handle known offenders (perhaps
in an offensive way). :-)

I Googled for telephony on the Pi but all I found were references to PABXs
for Internet Telephony.

Anyone know of a telephony module for the Pi?

I guess that a really old modem with voice capability might form a basis,
but modems are so last century.

So - anything in the pipeline or already there?

Cheers

Dave R

[Theo Markettos]

David.WE.Roberts <nos...@nospam.net> wrote:
> Anyone know of a telephony module for the Pi?
>
> I guess that a really old modem with voice capability might form a basis,
> but modems are so last century.

A USB softmodem might do the trick - see:
https://groups.google.com/d/msg/uk.telecom.voip/AcJJLLUgA4w/_B4xZbpfIAYJ
aka <oHv*4r...@news.chiark.greenend.org.uk>

(looks like my idea to modify it to make an FXS port won't fly, though)

The rest of that thread has some other ideas too.

Theo

[Rob]

David.WE.Roberts <nos...@nospam.net> wrote:
> I had another 'silent' call this morning, which reminded me that I'd
> really like to screen incoming calls and handle known offenders (perhaps
> in an offensive way). :-)
>
> I Googled for telephony on the Pi but all I found were references to PABXs
> for Internet Telephony.
>
> Anyone know of a telephony module for the Pi?

Why don't you switch to Internet Telephony?
It is a lot more convenient to handle things like that in a PABX like
Asterisk than to twiddle with phonelines.

> I guess that a really old modem with voice capability might form a basis,
> but modems are so last century.

So are analog phone lines...

I wrote software to do what you want for ZyXEL modems of the time,
and it should work on a Pi when you have a serial port (via USB or
via TTL->RS232 conversion) but as you wrote, it is all so last century...

[Jasen Betts]

there's linksys voip adaptors you could connect to the network socket.

OTOH a USB-POTS adaptor is basically a modem

--
⚂⚃ 100% natural

[Paul]

In article <b20d60...@mid.individual.net>, nos...@nospam.net says...

>
> I had another 'silent' call this morning, which reminded me that I'd
> really like to screen incoming calls and handle known offenders (perhaps
> in an offensive way). :-)

I would love to, but I use a BT service for that "Choose to refuse" as I
get customers using Withheld or Unavailable numbers, that way the BT
exchange blocks the call even if Withheld.

This is something you cannot do easily if anybody you need to speak
with ever uses International, Withheld or Unavailable numbers.

> I Googled for telephony on the Pi but all I found were references to PABXs
> for Internet Telephony.
>
> Anyone know of a telephony module for the Pi?
>
> I guess that a really old modem with voice capability might form a basis,
> but modems are so last century.

To be pedantic even broadband routers have modems in them but for higher
speeds than older dial up modems.

--
Paul Carpenter | pa...@pcserviceselectronics.co.uk
<http://www.pcserviceselectronics.co.uk/> PC Services
<http://www.pcserviceselectronics.co.uk/pi/> Raspberry Pi Add-ons
<http://www.pcserviceselectronics.co.uk/fonts/> Timing Diagram Font
<http://www.gnuh8.org.uk/> GNU H8 - compiler & Renesas H8/H8S/H8 Tiny
<http://www.badweb.org.uk/> For those web sites you hate

[Gordon Henderson]

In article <b20d60...@mid.individual.net>,

David.WE.Roberts <nos...@nospam.net> wrote:
>I had another 'silent' call this morning, which reminded me that I'd
>really like to screen incoming calls and handle known offenders (perhaps
>in an offensive way). :-)
>
>I Googled for telephony on the Pi but all I found were references to PABXs
>for Internet Telephony.
>
>Anyone know of a telephony module for the Pi?
>
>I guess that a really old modem with voice capability might form a basis,
>but modems are so last century.

The Pi is just another Linux box - admittedly without PCI, etc. slots. So
in that respect it's no different from running Asterisk (or Freeswitch)
with an external ATA to do what you want.

>So - anything in the pipeline or already there?

It's been done - it's Linux. There are even some specialist versions
of Raspbian for the Pi with asterisk which will do what you want. Just
serch the raspberypi.org forums. Someone did it over a year ago. It's
nothing special under Linux anymore.

Gordon

[LP]

On 2013-06-14, Theo Markettos <theom...@chiark.greenend.org.uk> wrote:
>
> A USB softmodem might do the trick - see:
> https://groups.google.com/d/msg/uk.telecom.voip/AcJJLLUgA4w/_B4xZbpfIAYJ
> aka <oHv*4r...@news.chiark.greenend.org.uk>

If anyone manages to make that work I'd love to hear about it. I've tried
and failed to get a couple of USB Winmodems to play with asterisk. Best
I've managed so far is to get linux to see it, load the kernel modules etc
but I can't get asterisk to talk to it at all.

(Admittedly not on a Pi, but it's all the same stuff really)

I'd love to get this working as I've got a 1957 strowger exchange in my
front room which is currently hooked up to the internet via a PC with a
4 port FXO/FXS PCI card in it and I'd *love* to be able to replace that
PC with a Pi and a bunch of USB modems all running off the exchange
batteries.

I find it immensely irksome that the strowger kit consumes no power at
all when idle (it only consumes power during a call) but the PC is sat
there all the time humming away running up my leccy bill.

> (looks like my idea to modify it to make an FXS port won't fly, though)

Rumour has it, you can make FXO/FXS ports out of sound cards, spit and
string: http://www.ckts.info/max-hax.php

I've never actually tried it though, as it looks a bit erm... nasty.

-Paul
--
http://paulseward.com

[David.WE.Roberts]

On Fri, 14 Jun 2013 11:44:48 +0000, Roger Bell_West wrote:

> On 2013-06-14, David.WE.Roberts wrote:
>>Anyone know of a telephony module for the Pi?
>>I guess that a really old modem with voice capability might form a
>>basis,
>>but modems are so last century.
>

> I don't know of anything you could plug directly into a phone line. The
> only FXO hardware for computers these days tends to be on a PCI card (I
> use an OpenVox A400); something like a PAP2T wants to do everything
> itself.
>
> It's not too hard to get the CLID presentation off a modem, but actually
> handling a voice channel is a bit more of a challenge.

Thanks :-)

A sensible answer to the original question (some more further down as
well).

It is quite a long time since I've seen an external modem which offers
voice as well as data externally - could be as much as 30 years :-(

Just found this article

<http://www.raspberrypi.org/phpBB3/viewtopic.php?t=40326&p=330143>

which points to

<http://www.broadbandbuyer.co.uk/Shop/ShopDetail.asp?ProductID=3473>

which is over £50.

As usual, double standards.

If I was connecting it to a £500 PC it might look reasonably priced.

More than the cost of a Pi though :-)

Cheers

Dave R

[Martin Gregorie]

On Fri, 14 Jun 2013 12:57:19 +0000, Gordon Henderson wrote:

> In article <b20d60...@mid.individual.net>,
> David.WE.Roberts <nos...@nospam.net> wrote:
>>I had another 'silent' call this morning, which reminded me that I'd
>>really like to screen incoming calls and handle known offenders (perhaps
>>in an offensive way). :-)
>>
>

I just leave an answering machine in circuit and turned on. At least the
bastards will get billed for making the call even if they drop it without
leaving a message.

However, a real solution needs legislation. Something along these lines
would be good: if you're registered with the TPS and the call comes from
a known offender, i.e. a number reported to the TPS or the ICO or a
premium number then your telco should refuse to connect the number and
should be entitled to recover the service's running costs by charging the
caller. Note that its only the victim who is affected by withheld
numbers. The telco certainly knows who made the call because they'll be
billing them for it.


--
martin@ | Martin Gregorie
gregorie. | Essex, UK
org |

[Theo Markettos]

LP <use...@lpbk.net> wrote:
> On 2013-06-14, Theo Markettos <theom...@chiark.greenend.org.uk> wrote:
> >
> > A USB softmodem might do the trick - see:
> > https://groups.google.com/d/msg/uk.telecom.voip/AcJJLLUgA4w/_B4xZbpfIAYJ
> > aka <oHv*4r...@news.chiark.greenend.org.uk>
>
> If anyone manages to make that work I'd love to hear about it. I've tried
> and failed to get a couple of USB Winmodems to play with asterisk. Best
> I've managed so far is to get linux to see it, load the kernel modules etc
> but I can't get asterisk to talk to it at all.

What sort of USB winmodems were you using? AFAICT getting winmodems to work
at all in Linux is tricky (mainly because you need the whole DSP stack to do
data and it's easier to just buy a hardmodem, so nobody wrote any drivers).
And nobody uses modems any more so things have bitrotted (but then the
hardware hasn't moved on either). I didn't manage to find (as in for sale)
any winmodems that appeared as sound devices which Asterisk supports
natively.

I do wonder how hard it can be to get audio in and out of a winmodem though
- it's just a DAC/ADC on the end of a USB port. From what I saw in the
slusb driver, most of the effort was on setting the sample rate than
anything more complex. Possibly 5 minutes on a USB analyser would enable
working out the protocol.

As another thought: are there any USB soundcards that use chipsets that
provide the AMR riser normally found on motherboards? I suspect not, but
an idea.

A fallback might be an RS232 voice modem - but I suspect this is going to be
trickier to put into 'dumb DAC' mode. There might be more chance of it
being documented, though.

> Rumour has it, you can make FXO/FXS ports out of sound cards, spit and
> string: http://www.ckts.info/max-hax.php
>
> I've never actually tried it though, as it looks a bit erm... nasty.

Interesting. "Doesn't support ringing" is a bit of a showstopper. I imagine
you could fake something up with a little transformer or a charge pump - it
just might not ring a 1950s phone.

My objective was to find a winmodem that works as FXO without modification
(so you can tell people to buy model X from ebay/Alibaba/etc) which I think
the above will do (if I've successfully navigated the winmodem minefield),
but providing ring current is going to mean something custom for FXS in any
case.

Theo

[Jasen Betts]

On 2013-06-14, Theo Markettos <theom...@chiark.greenend.org.uk> wrote:

> David.WE.Roberts <nos...@nospam.net> wrote:
>> Anyone know of a telephony module for the Pi?
>>
>> I guess that a really old modem with voice capability might form a basis,
>> but modems are so last century.
>
> A USB softmodem might do the trick - see:
> https://groups.google.com/d/msg/uk.telecom.voip/AcJJLLUgA4w/_B4xZbpfIAYJ
> aka <oHv*4r...@news.chiark.greenend.org.uk>
>
> (looks like my idea to modify it to make an FXS port won't fly, though)

yeah, no ring or line voltage generator.

--
⚂⚃ 100% natural

[Graham.]

On 14 Jun 2013 11:30:08 GMT, "David.WE.Roberts" <nos...@nospam.net>
wrote:

I got my Pi specifically to run an Asterisk based package called
RASPBX and it is performing very well indeed.

Previously I ran a similar package called Trixbox on various desktop
PCs and although that was also good it's very satisfying to do it on
hardware powered by a spare Blackberry charger.

Incoming trunks and entire DDI ranges can be had for free, for most UK
geographic STD codes, and outgoing calls can be made very cheaply on a
PAYG basis eg less than 3p/min to call a UK mobile

Marketing calls can be discreetly transferred to a recording with
strategic pauses that will keep the marketing agent in conversation
for as long as it takes for them to "twig".
Moreover, the system can record the result for my amusement.

RASPBX will do practically anything you could wish a PABX to do, eg.
Interactive Voice Response menus of any degree of complexity you
desire.

My favourite feature actually combines two features, Callback and
DISA.

I dial a DDI on my mobile phone, the Pi received the call but does not
answer the call, it effectively hangs up on me (so I incur no charge)
The Pi now waits 5 seconds then calls back the number that has just
called, I answer it and I now hear a PSTN style dial tone. I can now
dial any number I like via the Pi either internally or externally.

This can be augmented so "family" mobile phones work as described
above, but other CLIs will be asked to enter a PIN before they are
played the dial tone.

The limiting factor of a system with comparatively limited processing
power is the number of simultaneous calls that it can handle, as can
limited upload broadband throughput, however, the Pi can cope with
domestic usage and probably a lot of small business demands as well.





--
Graham.

%Profound_observation%

[Dave Liquorice]

On Sat, 15 Jun 2013 01:05:44 +0100, Graham. wrote:

> Incoming trunks and entire DDI ranges can be had for free, for most UK
> geographic STD codes, and outgoing calls can be made very cheaply on a
> PAYG basis eg less than 3p/min to call a UK mobile

True enough as it stands but the OP needs to interface a POTS line to
a Pi running Asterix (or WHY) not VoIP provisioned "lines".

It's finding an "FXO Gateway" that doesn't cost more than a Pi that
is the stumbling block. It might be simpler for the OP port his
existing POTS number to a VoIP provider, whilst retaining the POTS
line (with a new number) for delivery of internet and VoIP.

The problems that can come from this are BT ceasing the POTS line
when the number is ported, the number change "breaking" the ADSL on
the line. Running VoIP behind a NAT router (or on a dynamic IP) can
be a "challenge" though I think Asterix is quite good at handling
those situations. If it doesn't work you'll also need an ISP whose
"customer support" have half a clue about IP address's, ports,
filters, blocking, QOS etc...

--
Cheers
Dave.

[The Natural Philosopher]

Its not totally relevant, but a VOIP equipped router will give you
usually two VOIP phone plus one 'the raw POTS LINE' sockets on the back
iand is totally happy to route and prioritise calls using SIP, I know
cos I has such (till the router died).

New CISCO 527 on the christmas list..


--
Ineptocracy

(in-ep-toc’-ra-cy) – a system of government where the least capable to lead are elected by the least capable of producing, and where the members of society least likely to sustain themselves or succeed, are rewarded with goods and services paid for by the confiscated wealth of a diminishing number of producers.

[David.WE.Roberts]

Just for context I am a Virgin (ooo err Missus) Media cable customer.
So I don't have ADSL on a POTS line.

I am reluctant to port the POTS number to a VOIP provider (although that
is an interesting thought) because we don't use the land line much at all,
apart from providing a non-Mobile number to show that we are genuine house
owners and decent people and, of course, to receive Mumbai marketing calls.

We have had the same phone number for decades so it is also a port of call
for long lost relatives etc.

Mainly I was looking at luxuries I could try out with the Pi - itself
being a luxury.

One obvious thing was call screening - it would be nice to avoid
interruptions during something important.
The last silent call I was in the middle of cracking an egg which is
something you can't easily just leave.
[And no, that isn't code....]

We also have a '3' mobile with unlimited calls so that is used for
virtually all outgoing calls apart from 0800 numbers (which I don't think
VOIP makes free, although I could be wrong).

So - a nice project to play with incoming call handling on the Pi.

<http://www.amazon.co.uk/Cisco-Business-SPA3102-Gateway-Router/dp/
B000TSJ5JK> for £39.99 looks potentially the best option so far and about
the same cost as a Pi.

Also I can play with it using some of the other computers.

However, I somehow expected there to be a board for the Pi to do telephony
- but then again 'real' telephony using POTS is so last century so
probably no interest these days. :-)

Anyway, the challenge is now to beat £39.99 for an FXO.

Cheers

Dave R

P.S. if we ported our current land line number to a VOIP provider does
this mean we could re-direct calls to handsets/computers abroad, make
calls from this number from abroad, and have a network based answering
service?

It would be nice sometimes to appear at home when you are on holiday!

[Dave Liquorice]

On 15 Jun 2013 16:38:55 GMT, David.WE.Roberts wrote:

> Just for context I am a Virgin (ooo err Missus) Media cable customer.
> So I don't have ADSL on a POTS line.

I'd check that VM don't do silly things with ports, QOS, bandwidth
control etc. But that's me I don't trust any of the big ISPs not to
mess about with things.

> I am reluctant to port the POTS number to a VOIP provider (although that
> is an interesting thought) because we don't use the land line much at
> all, apart from providing a non-Mobile number to show that we are

> genuine house owners and decent people ...

I've just ported a couple of numbers to Sipgate, the outgoing CLI can
be set to either the Sipgate number or the ported BT one via the web
interface. Other providers are available and once the number is under
their control can't see why it shouldn't be available for outgoing
CLI.

> ... and, of course, to receive Mumbai marketing calls.

B-)

> We have had the same phone number for decades so it is also a port of
> call for long lost relatives etc.

Keeping the numbers is why I've ported.

> The last silent call I was in the middle of cracking an egg which is
> something you can't easily just leave.
> [And no, that isn't code....]

So just ignore the phone if you are too busy to answer it or let an
answering machine take it. They'll either leave a message or call
back if it's important. Why do people have this compulsion to answer
the phone

> We also have a '3' mobile with unlimited calls so that is used for

> virtually all outgoing calls ...

Not an option for us, mobile signals (all notworks) are too weak for
reliable calls and the delay/quality of mobiles is awful.

> ... apart from 0800 numbers (which I don't think VOIP makes free,

> although I could be wrong).

Sipgate are free, you can sign up for free and make test calls to
0800 numbers without crediting the account. You can also call it.
Again I think 0800 are free on most VoIP services, may even be a
requirement as it could be very hard to tell if a "domestic phone" is
a VoIP one or landline one.

> However, I somehow expected there to be a board for the Pi to do
> telephony - but then again 'real' telephony using POTS is so last
> century so probably no interest these days. :-)

Interest but getting approval for it without shoving the end price
through the roof maybe a problem...

> P.S. if we ported our current land line number to a VOIP provider does
> this mean we could re-direct calls to handsets/computers abroad, make
> calls from this number from abroad, and have a network based answering
> service?

Probably I've not looked at that side very hard. But I know I can
connect to my Sipgate account from anywhere on the internet (with
enough bandwidth, a call to BT the other day ran at 100kbps each way)
and make/receive calls. The outgoing CLI would be that set in the
account. Think of the VoIP provider <> VoIP "instrument" link over
the internet as a very long telephone extension cord. B-)

--
Cheers
Dave.

[Dave Liquorice]

On Sat, 15 Jun 2013 13:56:54 +0100, The Natural Philosopher wrote:

>> It's finding an "FXO Gateway" that doesn't cost more than a Pi
that
>> is the stumbling block.
>

> Its not totally relevant, but a VOIP equipped router will give you
> usually two VOIP phone plus one 'the raw POTS LINE' sockets on the back
> iand is totally happy to route and prioritise calls using SIP, I know
> cos I has such (till the router died).

Some do some don't, FXO ports are not that common. Are you sure the
Cisco 527 has an active FXO port? Looking
at:

http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/gatecont/ps1
0500/data_sheet_c78-550705.html

There is a distinction on the FXO ports between "relay" (520 models)
and "active" (540 models). Haven't checked but I'd assume that the
"relay" ones just connect the PSTN to the FXS ports under (power?)
failure conditions. The "active" ones meaning that you can
selectively route calls to/from the PSTN/VoIP.

--
Cheers
Dave.

[David.WE.Roberts]

Looking at the manual for the 3102 (and several others) it says

"The SPA3102 and the SPA8800 act as SIP-PSTN gateways"

So it looks (subject to further studies) that with luck a software PABX
might be able to manage the PSTN line and after that there's a whole load
of innocent fun to be had.

Including getting hacked and finding 10,000 calls to Mumbai on your POTS
phone bill :-(

The whole thing looks absurdly tempting - although divorce might be an
unexpected down side.

Cheers

Dave R

[David.WE.Roberts]

Now into hairpinning and shuffling - which is age appropriate if not
gender appropriate.

Every day something new :-)

Cheers

Dave R

[The Natural Philosopher]

sipgate thumbs up from me and look at thios

http://www.sjlabs.com/sjp.html

I THINK it means a linux machine equipped with cans and a mic is then a
SIP phone..

in fact looking at the ubuntu repo there's loads of VOIP clients

If you are prepared NOT to use a bog standard POTS interface

[The Natural Philosopher]

On 15/06/13 19:47, Dave Liquorice wrote:
> On Sat, 15 Jun 2013 13:56:54 +0100, The Natural Philosopher wrote:
>
>>> It's finding an "FXO Gateway" that doesn't cost more than a Pi
> that
>>> is the stumbling block.
>> Its not totally relevant, but a VOIP equipped router will give you
>> usually two VOIP phone plus one 'the raw POTS LINE' sockets on the back
>> iand is totally happy to route and prioritise calls using SIP, I know
>> cos I has such (till the router died).
> Some do some don't, FXO ports are not that common. Are you sure the
> Cisco 527 has an active FXO port? Looking
> at:
>
> http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/gatecont/ps1
> 0500/data_sheet_c78-550705.html
>
> There is a distinction on the FXO ports between "relay" (520 models)
> and "active" (540 models). Haven't checked but I'd assume that the
> "relay" ones just connect the PSTN to the FXS ports under (power?)
> failure conditions.

No. IIRC you plug the router into the wall and the phone into the POTS
port on the router. That is esentially the same as having a microfilter
in te way, and probably that is in fact all it is.

The other ports are SIP protocol POTS ports and will only work when the
router us up. To make calls via SIP to otherVOIP phiones around the
world, or PSTN pones via a paid for relay.

> The "active" ones meaning that you can
> selectively route calls to/from the PSTN/VoIP.
>


--

[LP]

On 2013-06-14, Theo Markettos <theom...@chiark.greenend.org.uk> wrote:
>
> What sort of USB winmodems were you using? AFAICT getting winmodems to work
> at all in Linux is tricky (mainly because you need the whole DSP stack to do
> data and it's easier to just buy a hardmodem, so nobody wrote any drivers).
> And nobody uses modems any more so things have bitrotted (but then the
> hardware hasn't moved on either). I didn't manage to find (as in for sale)
> any winmodems that appeared as sound devices which Asterisk supports
> natively.

It was about a year ago that I was trying, and I think I got furthest with
one of these: http://www.ebay.co.uk/itm/320617076822 (although I only paid
a fiver for it, inc postage)

As I said, "furthest" isn't all that far in this case. I'll see if I can
dig it out of the junk box tonight and work out what the chipset is.

These: http://www.ebay.co.uk/itm/230874075741 apparently come with linux
drivers and an asterisk module which makes them work - but they're a bit
expensive for my tastes (and I've yet to find a second hand one!)

For the cost of a Sangoma and a pi - I could buy a second hand grandstream
ATA which would do the job perfectly well.

> I do wonder how hard it can be to get audio in and out of a winmodem though
> - it's just a DAC/ADC on the end of a USB port. From what I saw in the
> slusb driver, most of the effort was on setting the sample rate than
> anything more complex. Possibly 5 minutes on a USB analyser would enable
> working out the protocol.

It's probably not that hard at all, but is beyond my programming skills
(I'm a sysadmin, not a programmer. C is effectively read-only as far as
I'm concerned ;)

>> Rumour has it, you can make FXO/FXS ports out of sound cards, spit and
>> string: http://www.ckts.info/max-hax.php
>>
>> I've never actually tried it though, as it looks a bit erm... nasty.
>
> Interesting. "Doesn't support ringing" is a bit of a showstopper. I imagine
> you could fake something up with a little transformer or a charge pump - it
> just might not ring a 1950s phone.

In my case it doesn't actually need to provide ringing current as it wouldn't
be ringing a phone anyway, that's what the phone exchange is there for.

Just incase anyone is interested, this is the beast I want to connect to:
http://www.paulseward.com/blog/20110926/pax-update/

> My objective was to find a winmodem that works as FXO without modification
> (so you can tell people to buy model X from ebay/Alibaba/etc) which I think
> the above will do (if I've successfully navigated the winmodem minefield),
> but providing ring current is going to mean something custom for FXS in any
> case.

The one I linked to on ebay should do all the right things wrt ringing current
etc (as it manages to work as advertised as a skype gateway)

I might start looking at this a bit more seriously as I appear to have blown
an FXO port on the PCI card I'm using at the moment - and replacing just the
FXO module seems to be approximately 50% of the cost of replacing the whole
card.

-Paul
--
http://paulseward.com

[Theo Markettos]

LP <use...@lpbk.net> wrote:
> On 2013-06-14, Theo Markettos <theom...@chiark.greenend.org.uk> wrote:
> >
> > What sort of USB winmodems were you using? AFAICT getting winmodems to work
> > at all in Linux is tricky (mainly because you need the whole DSP stack to do
> > data and it's easier to just buy a hardmodem, so nobody wrote any drivers).
> > And nobody uses modems any more so things have bitrotted (but then the
> > hardware hasn't moved on either). I didn't manage to find (as in for sale)
> > any winmodems that appeared as sound devices which Asterisk supports
> > natively.
>
> It was about a year ago that I was trying, and I think I got furthest with
> one of these: http://www.ebay.co.uk/itm/320617076822 (although I only paid
> a fiver for it, inc postage)

That's very interesting - I hadn't thought of Skype-FXO/FXS adaptors. That
listing says it's 'Smartlink 3800/3801' which I can't find much info about.
It would be handy to know what kind of USB device it presents itself as.

Supply of them seems a little odd - lots on ebay at the same price, which
suggests single supplier. Maybe Alibaba has more, I haven't looked.

> It's probably not that hard at all, but is beyond my programming skills
> (I'm a sysadmin, not a programmer. C is effectively read-only as far as
> I'm concerned ;)

You don't even need to write C - a bit of perl or python would do...
(/dev/slusb0 is a device you can poke from userland - you just treat it as a
file, with the only 'special' being an ioctl call to set data rate and off
hook)

Anyway, this is the Raspberry Pi group - we don't take 'I'm not a
programmer' for an answer ;-)

> The one I linked to on ebay should do all the right things wrt ringing current
> etc (as it manages to work as advertised as a skype gateway)
>
> I might start looking at this a bit more seriously as I appear to have blown
> an FXO port on the PCI card I'm using at the moment - and replacing just the
> FXO module seems to be approximately 50% of the cost of replacing the whole
> card.

Would be interesting to know if you make any progress...

Theo

[Rob Morley]

On Mon, 17 Jun 2013 00:44:47 +0100
"Brian Gregory [UK]" <n...@bgdsv.co.uk> wrote:

> Thanks for that pointless comment.
>
> It was just what was needed.
>
How much do you think your snarky comments are contributing to the
group?

[Dave Liquorice]

On 17 Jun 2013 12:15:20 +0100 (BST), Theo Markettos wrote:

> That's very interesting - I hadn't thought of Skype-FXO/FXS adaptors.

Doesn't Skype use a proprietary protocol rather than standards
compliant VoIP (SIP/RTP)? The only mention of VoIP I see on that eBay
add is in the image. The "... configurable greeting massage ..." has
me intrigued. B-)

--
Cheers
Dave.

[David.WE.Roberts]

Works for me :-)

[LP]

On 2013-06-17, Theo Markettos <theom...@chiark.greenend.org.uk> wrote:
>
>> It's probably not that hard at all, but is beyond my programming skills
>> (I'm a sysadmin, not a programmer. C is effectively read-only as far as
>> I'm concerned ;)
>
> You don't even need to write C - a bit of perl or python would do...
> (/dev/slusb0 is a device you can poke from userland - you just treat it as a
> file, with the only 'special' being an ioctl call to set data rate and off
> hook)

Interesting. Thanks!

> Anyway, this is the Raspberry Pi group - we don't take 'I'm not a
> programmer' for an answer ;-)

Fair point, well made.

>> The one I linked to on ebay should do all the right things wrt ringing current
>> etc (as it manages to work as advertised as a skype gateway)
>>
>> I might start looking at this a bit more seriously as I appear to have blown
>> an FXO port on the PCI card I'm using at the moment - and replacing just the
>> FXO module seems to be approximately 50% of the cost of replacing the whole
>> card.
>
> Would be interesting to know if you make any progress...

Any progress I do make will get written up and bandied about various relevant
places, don't worry about that :)

-Paul
--
http://paulseward.com

[Brian Gregory [UK]]

"Rob Morley" <nos...@ntlworld.com> wrote in message
news:20130617122607.3a53b2a9@hyperion...

You think it was acceptable to put down somebodys suggestion by saying their
ideas were based on stuff so old you would reject them off hand and ancient
history?

--

Brian Gregory. (In the UK)
n...@bgdsv.co.uk
To email me remove the letter vee.

[Theo Markettos]

Theo Markettos <theom...@chiark.greenend.org.uk> wrote:
> As another thought: are there any USB soundcards that use chipsets that
> provide the AMR riser normally found on motherboards? I suspect not, but
> an idea.

Yet another evil thought: AMR risers have all the modem functions wrapped up
in an AC'97 stream. You just push out audio at a specific clock rate to
meet the AC-Link standard, with additional pins being hook, ring detect,
etc. It turns out that AC-Link looks a lot like SPI - I wonder if the Pi's
SPI controller could be sufficiently abused into driving it. Then you'd
just wire up the SPI lines to an AMR riser and you're done.

You could try to bitbang AC-Link as well, but it's 12MHz so might be a bit
tricky.

Theo

[Jasen Betts]

On 2013-06-17, Windmill <spam-n...@Onetel.net.uk.invalid> wrote:

>>Most of this might already exist in something like Asterisk.
>
> Is there such a thing as an off-the-shelf Class D amplifier? (I know
> there are inexpensive ICs). Maybe one could be used together with one
> of the Pi's output pins to synthesize a ring voltage and pattern. (Step
> up the voltage with a small transformer if required.)

ring voltage is tricky, the phone ringer is a capacitive load the the
required voltage is higher than is convenient and bipolar.

only a small current is needed but the low frequency means makes
finding a suitable transformer harder, because as reducing the frequency
also reduces the saturation voltage by the same factor.

you want about 100V out

perhaps a 50Hz 240V to 12V transformer run in reverse with
5VAC at 20Hz on the 12V winding.

for 3ma out and with a transformer ratio of 20:1 you want a
transformer with a 60mA secondary.

--
⚂⚃ 100% natural

[LP]

On 2013-06-18, Jasen Betts <ja...@xnet.co.nz> wrote:
> On 2013-06-17, Windmill <spam-n...@Onetel.net.uk.invalid> wrote:
>
> ring voltage is tricky, the phone ringer is a capacitive load the the
> required voltage is higher than is convenient and bipolar.

The ringer is an inductive load. Ring voltage is a remarkably flexible
beast (in the UK at least) and most phones will ring on anything between
about 40V and 150V AC, frequency between 15Hz and 60 Hz.

I think the US is *strictly* 20Hz and some of the older american phones
won't ring at any other frequency because they use mechanically resonant
armatures for the clapper.

UK phones aren't anywhere near as fussy.

I have heard it said that on the UK network the 50V DC feed to the phone
is maintained during ringing, although none of the exchanges I've ever
worked on[1] do this so I don't think it's strictly speaking necessary.

While 50V DC is the standard for the speech path, in practice most
phones are happy with anything above 5V (older phones with carbon mics
are happy below 5V)

> only a small current is needed but the low frequency means makes
> finding a suitable transformer harder, because as reducing the frequency
> also reduces the saturation voltage by the same factor.
>
> you want about 100V out
>
> perhaps a 50Hz 240V to 12V transformer run in reverse with
> 5VAC at 20Hz on the 12V winding.
>
> for 3ma out and with a transformer ratio of 20:1 you want a
> transformer with a 60mA secondary.

This approach does work, as does just using an audio amplifier.

I'll leave protecting the pi from all these nasty volts an exercise
for the reader...

-Paul
[1] All strowger kit, moden phone exchanges are boring ;)
--
http://paulseward.com

[Rob]

LP <use...@lpbk.net> wrote:
> I have heard it said that on the UK network the 50V DC feed to the phone
> is maintained during ringing, although none of the exchanges I've ever
> worked on[1] do this so I don't think it's strictly speaking necessary.

Details like that vary by country. In the old days, the local state
telecoms defined their line protocol and several versions are in use.

Over here, when the phone rings first the DC feed polarity is reversed,
then the calling number is sent as a DTMF string, then the AC voltage
is superimposed on the DC. When the calling party hangs up, the DC
is reversed back. So you can always determine if you still have a
connection.

However, standard modem and answering machine equipment has no logic to
detect this and relies on detecting carrier, busy signal, etc.

Also, in most other countries the calling number is sent in 1200 bps FSK
and not before the first ring but between the first and second ring.
This invariably causes problems when people buy equipment supporting
calling number ID and this equipment not being designed for use in
the Netherlands.

Complicating it further, there now are competing telecom providers and
they (for practical reasons) all use the FSK calling number ID.

[The Natural Philosopher]

On 18/06/13 08:22, Jasen Betts wrote:
> On 2013-06-17, Windmill <spam-n...@Onetel.net.uk.invalid> wrote:
>
>>> Most of this might already exist in something like Asterisk.
>> Is there such a thing as an off-the-shelf Class D amplifier? (I know
>> there are inexpensive ICs). Maybe one could be used together with one
>> of the Pi's output pins to synthesize a ring voltage and pattern. (Step
>> up the voltage with a small transformer if required.)
> ring voltage is tricky, the phone ringer is a capacitive load the the
> required voltage is higher than is convenient and bipolar.
>
> only a small current is needed but the low frequency means makes
> finding a suitable transformer harder, because as reducing the frequency
> also reduces the saturation voltage by the same factor.

nah. You dont do it that way. You drive a HF inverter using a small
ferrite core, rectify and smooth te outpout, but not too much, and
simply chop the input to it.

> you want about 100V out

IIRC is 70V DC that goes up and down when ringing..

But actually driving a phone is a complext task.

Beter to buy a box that does it all.

> perhaps a 50Hz 240V to 12V transformer run in reverse with
> 5VAC at 20Hz on the 12V winding.
>
> for 3ma out and with a transformer ratio of 20:1 you want a
> transformer with a 60mA secondary.
>


--

[druck]

On 14/06/2013 13:50, Paul wrote:
> To be pedantic even broadband routers have modems in them but for higher
> speeds than older dial up modems.

If you are really pedantic, yes they have a modulator and demodulator
for converting digital data in to DSL frequencies. But the important
thing for this discussion, is that they have no capability of
interacting with the voice network, such as connecting/disconnecting
calls, extracting caller ID, etc.

---druck

[Johny B Good]

On Tue, 18 Jun 2013 15:22:56 +0100, The Natural Philosopher
<t...@invalid.invalid> wrote:

>On 18/06/13 08:22, Jasen Betts wrote:
>> On 2013-06-17, Windmill <spam-n...@Onetel.net.uk.invalid> wrote:
>>
>>>> Most of this might already exist in something like Asterisk.
>>> Is there such a thing as an off-the-shelf Class D amplifier? (I know
>>> there are inexpensive ICs). Maybe one could be used together with one
>>> of the Pi's output pins to synthesize a ring voltage and pattern. (Step
>>> up the voltage with a small transformer if required.)
>> ring voltage is tricky, the phone ringer is a capacitive load the the
>> required voltage is higher than is convenient and bipolar.
>>
>> only a small current is needed but the low frequency means makes
>> finding a suitable transformer harder, because as reducing the frequency
>> also reduces the saturation voltage by the same factor.
>
>nah. You dont do it that way. You drive a HF inverter using a small
>ferrite core, rectify and smooth te outpout, but not too much, and
>simply chop the input to it.
>
>> you want about 100V out

It's been a good 30 years since I last worked in a strowger exchange
but I do clearly recall that the line polarity on an incoming call is
reversed with respect to that when the line is idle or engaged by an
outgoing call, along with the reason why.

The two legs of the balanced phone line are referred to as the 'A'
and 'B' legs of the line. When idle, galvanically speaking, the B leg
is connected to the negative of the exchange battery via one of the
two windings of the A relay in the calling cct (typically a linefinder
for residential customers and uniselector for business customers). The
A leg is connected to the exchange earth (positive connection of the
exch battery) via the other winding on the A relay .

For outgoing calls, the polarity remains the same as at idle. The 50v
supply allows for plenty of volt drop due to line resistance since the
phone only needs about 10% of this (it's current that's important,
circa 30mA or so for a carbon mic telephone handset) and, afaicr,
about 12 to 15 mA for a modern electronic phone.

On an incoming call, the Final Selector (FS) supplies the operating
DC current and the AC ringing current to ring the bell or tone sounder
of the called telephone. The ring current generator supply in the
exchange is a 75v RMS at 17.67Hz single phase supply[1] interrupted at
the appropriate ringing cadence (normally, the supply was generated by
a DC to AC motor/generator machine which also drove the cams which
generated the ring cadence, providing 3 different feeds which balanced
the generator loading in a timeshare fashion).

The reason for the polarity reversal was on account the ringing
current generator was an unbalanced source of AC which used a common
ground return. The ringing current would be applied to the B leg[2]
and the necessary -50v bias supply to detect when the phone had gone
'Off Hook' to answer the call and power the mic cct would be fed over
the A leg.

This meant that the relatively high voltage ac (peaks of 100v) didn't
have to be superimposed on top of a further 50v, neatly serving two
requirements, namely health and safety issues and minimising
complexity.

>IIRC is 70V DC that goes up and down when ringing..

If by that you mean a 70v DC chopped at 17Hz or so, that's one
possibility. Another one is to rapidly reverse the 50v at 15 to 17Hz
as ISTR doing with a home made setup in the early 70s using a fast
reversing relay.

>
>But actually driving a phone is a complext task.

It isn't, really.
>
>Better to buy a box that does it all.

In most cases that's tue, simply to avoid spending time that could
otherwise be used to earn the money to buy several such 'boxes'

>> perhaps a 50Hz 240V to 12V transformer run in reverse with
>> 5VAC at 20Hz on the 12V winding.

You could generate 25Hz from a cct locked to the 50Hz mains and a 5v
peak to peak should generate an 80v peak to peak output without
saturation ofthe transformer core.

>>
>> for 3ma out and with a transformer ratio of 20:1 you want a
>> transformer with a 60mA secondary.

You seem to be out by about a factor of 10. I'd be looking at a 25mA
output, hence a half amp 'secondary' rating.


[1] 16.67Hz (17Hz) was standard for public telephone exchanges but
other frequencies were used in PABXes, typically 25Hz which I believe
could be readily synthesised from the UK PSU's 50Hz mains supply -
it's quite likely the yanks and countries using the 60Hz PSU standard
might well have used 30Hz for the same reason. It's also possible that
a lower AC voltage may have been used in PABX equipment for the ringer
supply since they'd normally only be expected to service much shorter
line lengths to the extension phones within the site they were
designed to serve.

[2] On normal phone lines, the A leg would provide the return path for
the ringing current (the 250v DC rated 1.8 microfarad capacitor within
the phone (main if extension phones were involved or else the master
socket on the later and current plug in phone system) provided the
isolation from the -50v battery connection via the FS at the exchange
during the ringing phase prior to the called subscriber taking the
phone offhook to answer the call.

Of course, it was possible to arrange for the ringing current return
to be via the subscriber's local earth connection. This permitted a
means of providing shared service in parts of the exchange's coverage
area that experienced demand for customer service that had exceeded
the telephone planner's expectations.

A single phone line could provide automatic shared service to two
customers on a time shared basis. Shared service customers had to
agree to an etiquate with regard to the use of the line. Each customer
being expected to listen for an existing call by their shared party
before pressing the call button[3], apologising, if need be for
interrupting an existing call, optionally ascertaining how long they
might have to wait or else explain the need to make an emergency call
right then and there.

Incoming calls would automatically ring the appropriate customer's
telephone by virtue of the use of the local earth return cct at each
customers' premises and the fact that the Y customer's line would have
the A and B legs reversed with respect to that of the X customer such
that the 'hot' leg for the ringer connection would be on the B leg for
the X subscriber whilst the Y subscriber would use the A leg.

The "P" (Private) wires used by the exchange equipment associated
with each customer's number and calling equipment are jumpered
together on the IDF to allow the exchange equipment to correctly
return a busy signal whenever the line is engaged regardless of which
of the shared numbers is being called.

An unshared number would just simply have the single P wire jumpered
to link the calling equipment to the associated FS outlet's P wire
connection so that a 'line engaged tone' could be returned to the
caller in the event the line was busy due to an established incoming
or outgoing call.

Statistically speaking, this usually worked ok most of the time
(shared service was only offered to the low usage rate residential
customer) but it was possible for an incoming call to be accidentally
intercepted by the other customer trying to make an outgoing call.

When this happened, the caller would be advised of the shared line
situation and either asked to attempt another call or try later if the
calling customer who had answered by accident had an urgent need to
make their call. However, there was nothing to stop an obliging shared
partner from asking the caller to hang on whilst he alerted his
sharing neighbour of the incoming call.

The GPO (as it was when I joined the civil service those many years
ago) only offered shared service if both parties were aware of and
willing to accept the limitations and pitfalls of a shared service
party line.

The usual difficulty lay with finding an existing customer willing to
accept the reduction in their quarterly bill as sufficient inducement
to downgrade to a shared service line. In some cases, iirc, some
existing customers were chosen on the basis of minimum call usage and
offered the downgrade to shared service when the additional subscriber
was also an existing subscriber who had moved from another part of the
exchange's catchment area.

In these cases, the customer being asked to downgrade to a shared
service line could make strenuous objections which resulted in the GPO
testing the waters with other likely candidates fed from the same DP
until they found one that had the least objections or a better rental
reduction offer resolved the issue.

Shared service was deprecated a while before I transferred from
exchange maintainance to district maintainance duty so I very rarely
found myself embroiled in any shared service acrimony.

I believe another form of shared service, or party line working, was
practiced in the US which took the form of different ringer burst
counts to identify which of anywhere from 2 to 4 sharing customers the
call was intended for. In this case, the issue of who was responsible
for picking up the tab on outgoing calls could only have been resolved
by an operator so must have been a very early form of shared service
offered in remote rural areas that still relied on manual switchboard
working. Whether this form of party line operation was ever practiced
here in the UK, I couldn't say.

[3] The Call Button was required to allow the exchange equipment to
detect which of the two parties would be footing the bill for the
outgoing call. The mechanism relied upon the exch equipment detecting
which of the two legs of the line the earth calling signal appeared
on. As you can imagine, this system was open to abuse by a
knowledgable and crafty customer, as well as due to errors of
maintainance and repairwork which could introduce a line reversal.

These days, there are modern technology solutions for delivering
'shared service' transparently to high demand 'hot spots' not yet
served by sufficient line capacity - essentially, technology related
to that used by ADSL broadband.

It's been over two decades since I quit BT plus another five years on
top of that since I last dealt with any residential shared service
lines. I don't imagine that, after all this time, there'd still be any
'legacy' shared service lines still in service today.
--
Regards, J B Good

[Michael J. Mahon]

Actually, I think he meant using a high-frequency PWM signal (many
kilohertz) to synthesize a low-frequency sine wave after the transformer
and filtering.

In the US, three wires are used for service: tip, ring, and sleeve (named
for the switchboard plug connections). Four parties on a service connection
were separately ringed by ringing on either tip or ring wires, with either
positive or negative bias. Gas tube (valve) rectifiers in the handsets
separated the ringing polarities.

The four parties on a line were referred to as tip or ring, positive or
negative.

The low-frequency ring current (about 20Hz) was generated at the exchange
using a motor-generator set with cadence switching as you have described.

This is all great fun, but I suspect that it belongs in a retro-telephony
forum... ;-)

> [3] The Call Button was required to allow the exchange equipment to
> detect which of the two parties would be footing the bill for the
> outgoing call. The mechanism relied upon the exch equipment detecting
> which of the two legs of the line the earth calling signal appeared
> on. As you can imagine, this system was open to abuse by a
> knowledgable and crafty customer, as well as due to errors of
> maintainance and repairwork which could introduce a line reversal.
>
> These days, there are modern technology solutions for delivering
> 'shared service' transparently to high demand 'hot spots' not yet
> served by sufficient line capacity - essentially, technology related
> to that used by ADSL broadband.
>
> It's been over two decades since I quit BT plus another five years on
> top of that since I last dealt with any residential shared service
> lines. I don't imagine that, after all this time, there'd still be any
> 'legacy' shared service lines still in service today.

--

-michael - NadaNet 3.1 and AppleCrate II: http://home.comcast.net/~mjmahon

[Dave Liquorice]

On Wed, 19 Jun 2013 22:24:04 +0100, Johny B Good wrote:

> It's been over two decades since I quit BT plus another five years on
> top of that since I last dealt with any residential shared service
> lines. I don't imagine that, after all this time, there'd still be any
> 'legacy' shared service lines still in service today.

Probably not of the type you describe but wasn't that replaced with a
carrier system of some sort? One customer got the base band circuit
and another a circuit carried to a small battery powered box at the
DP or where the single line "split". Mostly worked, unless the person
on the carrier circuit talked a lot and the battery went flat...

DACS, I know if I waited long enough I'd remember the acronym.

http://en.wikipedia.org/wiki/Digital_Access_Carrier_System

Rather more complex than I thought. There are some subs end DACS
boxes attached to poles around here, don't know if they are still in
service though.

--
Cheers
Dave.

[LP]

On 2013-06-17, LP <use...@lpbk.net> wrote:
>
> It was about a year ago that I was trying, and I think I got furthest with
> one of these: http://www.ebay.co.uk/itm/320617076822 (although I only paid
> a fiver for it, inc postage)
>
> As I said, "furthest" isn't all that far in this case. I'll see if I can
> dig it out of the junk box tonight and work out what the chipset is.

I've managed to find the thing and get it plugged into a linux box. My
RPi isn't accessible at the moment (being on loan to someone else) so
mileage will almost certainly vary depending on what bits of sl-modem are
available in the repos.

paul@stathand:~$ lsusb
...
Bus 001 Device 003: ID 0483:7554 SGS Thomson Microelectronics 56k SoftModem
...
paul@stathand:~$

Taking the thing apart[1] reveals it's a SmartLink chipset (based on the
relevant chips being marked SL3800 and SL3801) and that matches up with
what the internet says about the device ID. I've not (yet) managed to
find datasheets for the SL3800/SL38001 but I suspect one is for the FXO
and one for the FXS ports (given that each one appears to have a pair
of OpAmps associated with it - presumably TX/RX)

When I took it apart, a perusal of the board layout makes it look very
much as though all the gubbins is there for a true FXO/FXS port pair (at
least from the line interface side of things) so it should "do the right
thing" if tickled appropriately.

sl-modem claims to work with this chipset, but during the brief time I
had to play with it last night I couldn't get sl-modem-daemon to start:

paul@stathand:~$ sudo /etc/init.d/sl-modem-daemon start
Only access through ALSA is available on amd64 but slamr driver was chosen!
Make sure that an ALSA driver for your chipset is available and is loaded
and that access to SmartLink modem components is supported by it.
paul@stathand:~$

So it looks like I'll need to fix ALSA a bit before I can get any further
but it very much looks as though it's recognisable.

Apart from fixing the ALSA stuff, I'll also need to work out how to get
Asterisk to talk to it, but yeah - it's not looking impossible.

-Paul
[1] I'm sure this is standard behaviour for everyone right?
--
http://paulseward.com

[Theo Markettos]

LP <use...@lpbk.net> wrote:
> paul@stathand:~$ lsusb
> ...
> Bus 001 Device 003: ID 0483:7554 SGS Thomson Microelectronics 56k SoftModem

Looks promising.

> sl-modem claims to work with this chipset, but during the brief time I
> had to play with it last night I couldn't get sl-modem-daemon to start:
>
> paul@stathand:~$ sudo /etc/init.d/sl-modem-daemon start
> Only access through ALSA is available on amd64 but slamr driver was chosen!
> Make sure that an ALSA driver for your chipset is available and is loaded
> and that access to SmartLink modem components is supported by it.
> paul@stathand:~$

IIRC to drive it as a modem requires a proprietary DSP binary library to do
the modem stuff, which I don't think is shipped with distros. If you get
/dev/slusb0 then that's good going, however.
http://wiki.debian.org/slmodem
That might be the reason for amd64 errors, if the DSP library is only built
for i386.

slamr is for the AMR header on your motherboard. AMR is part of AC'97, so
your motherboard chipset will probably have some audio channels for it even
if they don't actually go anywhere. So you can start ALSA on it, but it's
a red herring. Most of the notes out there are for AMR or PCI modems, which
are completely different.

> So it looks like I'll need to fix ALSA a bit before I can get any further
> but it very much looks as though it's recognisable.

Does look like it might work. Be interesting to find out how they do the
FXS port.

> [1] I'm sure this is standard behaviour for everyone right?

1. Open mailing envelope
2. Tear blister pack
3. Throw away instructions
4. Unscrew lid
5. Look inside
6. Plug into computer (without lid)
7. Google
8. Recover instructions from bin, find they're worthless, throw away again
9. Make bird scarer from driver CD

Theo

[Johny B Good]

On Thu, 20 Jun 2013 10:27:17 +0100 (BST), "Dave Liquorice"
<allsortsn...@howhill.com> wrote:

>On Wed, 19 Jun 2013 22:24:04 +0100, Johny B Good wrote:
>
>> It's been over two decades since I quit BT plus another five years on
>> top of that since I last dealt with any residential shared service
>> lines. I don't imagine that, after all this time, there'd still be any
>> 'legacy' shared service lines still in service today.
>
>Probably not of the type you describe but wasn't that replaced with a
>carrier system of some sort? One customer got the base band circuit
>and another a circuit carried to a small battery powered box at the
>DP or where the single line "split". Mostly worked, unless the person
>on the carrier circuit talked a lot and the battery went flat...

WB900, probably. It's mentioned in the wiki article you linked below.

>
>DACS, I know if I waited long enough I'd remember the acronym.
>
>http://en.wikipedia.org/wiki/Digital_Access_Carrier_System
>
>Rather more complex than I thought. There are some subs end DACS
>boxes attached to poles around here, don't know if they are still in
>service though.

Quite probably. The extra electronics is far far cheaper than
upgrading the cable capacity that feeds the DPs. The most likely
motive for upgrading capacity will be more to do with broadband
provisioning and even here, fibre to the DP (or even FTTH) might
provide the better solution.

[David.WE.Roberts]

<snip>

>
> In the US, three wires are used for service: tip, ring, and sleeve
> (named for the switchboard plug connections). Four parties on a service
> connection were separately ringed by ringing on either tip or ring
> wires, with either positive or negative bias. Gas tube (valve)
> rectifiers in the handsets separated the ringing polarities.
>
> The four parties on a line were referred to as tip or ring, positive or
> negative.
>
> The low-frequency ring current (about 20Hz) was generated at the
> exchange using a motor-generator set with cadence switching as you have
> described.
>
> This is all great fun, but I suspect that it belongs in a
> retro-telephony forum... ;-)
>

<snip>

Unless, of course, the hardware is readily available for the Pi along with
the appropriate drivers :-)

[Jeff Jonas]

>> I had another 'silent' call this morning, which reminded me
>> that I'd really like to screen incoming calls
>> and handle known offenders (perhaps in an offensive way). :-)

In the USA, I think the FCC had a contest
for methods of handling nusance calls
because the "do not call" list is being ignored
and the perpretrators are (or pretend to be)
out of USA jurisdiction.

Anything less than bodily hard to the OWNERS of the systems
is useless. The main culprit are the "predictive dialers":
computers that call numbers in advance, wait for a reply,
determine if the reply is a human or an answering maching/voice mail,
and only THEN connect the call to sales-slime.
The really rude ones play a pre-recorded message,
in total violation of USA FCC phone regulations.
But without enforcement, the rules are useless.
So we're all on our own.

>Why don't you switch to Internet Telephony?
>It is a lot more convenient to handle things like that in a PABX like
>Asterisk than to twiddle with phonelines.

That's a totally logical solution
but I'm reluctant to cut over so fast.
Only 3 months ago, I finally converted from copper-pair
POTS (plain phone service with dialtone)
to Verizon FiOS (fiber optic) which is probably VoIP
(or whatever the cable companies use).

Happily, I never invested in ISDN (it still does nothing)
which was the digital telephony of the 80s.
It was a great technology but the USA phone companies
priced it too high for common adoption.

>> I guess that a really old modem with voice capability might form a basis,
>> but modems are so last century.

If it works, use it!
At least it provides the DAA (phone line isolation)
CLID (caller ID handling) and some had voice mail and FAX
built in for total off-line operations.

>So are analog phone lines...

Until superstorm Sandy (which flooded many phone exchanges
and destroyed many cables), copper pair was really the most reliable
phone service, so long as the CO (central office)
was running and provided the battery power.
It's hard to jam like wireless/cellphones
(except for the occasional backhoe breaking a cable).

>I wrote software to do what you want for ZyXEL modems of the time,
>and it should work on a Pi when you have a serial port (via USB or
>via TTL->RS232 conversion) but as you wrote, it is all so last century...

That sounds clever!
I always wanted a voice mail system that was caller-id triggered,
so I could deliver a message specific to the caller,
and prioritize calls.
Or pop up the related notes on a screen,
so when a sales-person calls,
I can see what they're calling about
of if it's just another cold-call.

The only "retaliation" I can think of
is playing a message starting with the
SIT (tri-tone that starts error messages)
and your own message such as
"if you are a human, dial 5 now,
otherwise PUT ME ON YOUR DO NOT CALL LIST and stop annoying me!"
(and drop the call if '5' is not pressed withing a few seconds)

(and excalate to more vulgar messages if the caller continues)