Telephony module (real phone line) for Pi?

that
Some do some don't, FXO ports are not that common. Are you sure the Cisco 527 has an active FXO port? Looking at:
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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.
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Cheers 
Dave.
Reply to
Dave Liquorice
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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
Reply to
David.WE.Roberts
Now into hairpinning and shuffling - which is age appropriate if not gender appropriate.
Every day something new :-)
Cheers
Dave R
Reply to
David.WE.Roberts
Yeah, the problem is it wants to do everything itself. You could probably tweak it to pass through all incoming PSTN calls to a SIP trunk to your Pi, but that's not what it's designed for; it wants to have a PSTN line, an internet connection, and a POTS handset, and to be the plumbing between the handset and the other two.
R
Reply to
Roger Bell_West
sipgate thumbs up from me and look at thios
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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
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Reply to
The Natural Philosopher
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.
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Ineptocracy 

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Reply to
The Natural Philosopher
Thanks for that pointless comment.
It was just what was needed.
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Brian Gregory. (In the UK) 
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Reply to
Brian Gregory [UK]
I find most nuisance calls come from overseas call centres and don't have any meaningful CLI. (Either "International" or nonsense like "00000000000" or "01111111111".
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Brian Gregory. (In the UK) 
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Reply to
Brian Gregory [UK]
It was about a year ago that I was trying, and I think I got furthest with one of these:
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(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:
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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.
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 ;)
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:
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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
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Reply to
LP
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.
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 ;-)
Would be interesting to know if you make any progress...
Theo
Reply to
Theo Markettos
How much do you think your snarky comments are contributing to the group?
Reply to
Rob Morley
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-)
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Cheers 
Dave.
Reply to
Dave Liquorice
Works for me :-)
Reply to
David.WE.Roberts
The skype "phones" that are USB attachments to pcs are dual devices. One part is standard USB HID ("Human Interface Device") for audio, and the other is a proprietary console for dialling and signalling on/off hook etc.
The linux drivers handle the first part really well, and the asterisk drivers interface directly to these linux drivers. For the signalling and keypad stuff you are on your own. (I am not aware of any reverse engineering drivers for these, but I haven't been too involved during the last 18 months or so).
There is some magic in the drivers, detecting on-hook status from the voltage levels in the audio. They detect most countries callerid too using the fsk library, so it really depends on the device.
-- mrr
Reply to
Morten Reistad
Interesting. Thanks!
Fair point, well made.
Any progress I do make will get written up and bandied about various relevant places, don't worry about that :)
-Paul
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Reply to
LP
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?
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Brian Gregory. (In the UK) 
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Reply to
Brian Gregory [UK]
I note there's this:
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which seems to be for a Yealink device: 6993:b001 Freshtel FT-102 VoIP USB Phone The web SVN is worth looking at.
There's loads of Skype-stuff in there, but it does seem to do the basic buttons/hookswitch/etc.
One thing that does concern me is if the PSTN is a simple cut-through - in other words there's no PSTN-VOIP adaptor, the PSTN is connected to the handset at all times except when the handset is in use. That would mean there's no way to drive the PSTN from the VOIP side.
I'd have thought it would be relatively simple to do FXO with one:
Hookswitch is a relay (1 bit output). Pulse dial by clicking the hookswitch Tone dial by generating DTMF audio samples Caller ID is one for the audio side
And FXS is only slightly more complex: Ring voltage is a relay (1 bit output) Ring by pulsing the ring voltage in an appropriate pattern Detect on-hook from voltages Detect pulse dial from pattern of on-hook Detect DTMF from the audio (need some call state so we don't parse your bank passwords) Emit caller ID by FSK encoding
Most of this might already exist in something like Asterisk.
Theo
Reply to
Theo Markettos
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.)
I've been wondering about a low distortion audio amp of this type, but while I might just manage to cobble up with my rusted C something which would make recognisable sounds, hi-fi would be far beyond my understanding.
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Reply to
Windmill
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
Reply to
Theo Markettos
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.
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Reply to
Jasen Betts

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