Integration of VoIP with Tunstall healthcare alarm

[Graham J]

My client has an internet connection and Cat5 wiring to a granny annexe.
In that granny annexe there is a Linksys adapter, Product Name: PAP2T.
Connected to that is an ordinary POTS phone.

Voipfone provides the telephone service for this adapter, and for
several SNOM300 phones in other parts of the property. The system has
worked well for several years.

We now need to equip the granny annexe with an alarm call system, and
have acquired a unit from Careline365. This unit is manufactured (which
probably means badged) by Tunstall. It looks like the one shown here:

https://uk.tunstall.com/services/our-products/lifeline-vi/

Tunstall cannot confirm that the unit will work with VoIP, but their
technical support people don't really understand the question, confusing
it with integration with a fibre connection to the home.

However I have tried testing the device connected to the Linksys
adapter. It seizes the line, sends DTMF tones, and the Careline system
appears to answer. I then hear a few more tones, and after a several
seconds the call is dropped and I can hear the number unobtainable tone.

The Voipfone outgoing call log shows the number dialled and the duration
of the call at 28 seconds.

The Careline365 administration requires many details, including the
telephone number of the line that the alarm system is connected to.
They can see the incoming test calls from this number; but their system
fails to connect properly.

Has anybody here any experience with getting this sort of system to work
properly?

TIA

--
Graham J

[Andy Burns]

Graham J wrote:

> My client has an internet connection and Cat5 wiring to a granny annexe.
> In that granny annexe there is a Linksys adapter

> We now need to equip the granny annexe with an alarm call system

Either the ATA provides a circuit that's close enough to the SIN 351
spec for the alarm to work, or it doesn't, I expect the supplier will
only claim support for an actual PSTN line.

<https://www.btplc.com/SINet/SINs/pdf/351v4p8.pdf>

I know BT didn't previously like direct links to individual SINs, if it
doesn't work, start at

<https://www.btplc.com/SINet/SINs/index.htm>

[Graham.]

one possibility although not very likley, is the PAP-2 by default uses
North American supervisory tone frequencies and cadences. If you have
the Admin password to the unit you can change them to UK tones
https://basichelp.sipgate.co.uk/hc/en-gb/articles/206289069-UK-Regional-Settings-Cisco-Linksys-Sipura-Adaptors-


I take it that Careline 365 is a pendent button that activates a
dedicated hands free speakerphone?

So are these extra tones a handshake, and is it initiated from your
side or the remote side, I suppose you could dial the service from
annother line to try.

Is the CLID sent by Voipfone the one they are expecting, assuming they
filter on that?





--

Graham.
%Profound_observation%

[Graham J]

Graham. wrote:

[snip]

> one possibility although not very likley, is the PAP-2 by default uses
> North American supervisory tone frequencies and cadences. If you have
> the Admin password to the unit you can change them to UK tones
> https://basichelp.sipgate.co.uk/hc/en-gb/articles/206289069-UK-Regional-Settings-Cisco-Linksys-Sipura-Adaptors-
>

I read that document. A BT style handset is normally used with this
adapter, and works correctly via the Voipfone system. I have admin
access to the adapter via a VPN, and although there are many
configuration settings in the adapter, very few look anything like those
shown in your document. I don't want to change any settings without
reference to Voipfone since that may break their system; but I will ask
their advice. It may take time to find somebody there that understands
the issue.

> I take it that Careline 365 is a pendent button that activates a
> dedicated hands free speakerphone?

Careline365 is the service provider, see: https://www.careline.co.uk/

As previously explained, they provide a Tunstall device, which is as you
describe a dedicated hands free speakerphone.

> So are these extra tones a handshake, and is it initiated from your
> side or the remote side, I suppose you could dial the service from

> another line to try.

I tried the service number from an ordinary phone on another line; the
call is answered, I hear some further tones, then the call is cleared at
the remote end. This is the same sequence of events that I hear when
testing the Tunstall device.

So the Tunstall device would appear not to respond to the tones received
from the monitoring centre.

> Is the CLID sent by Voipfone the one they are expecting, assuming they
> filter on that?

Yes, the monitoring centre centre recognises that there is a failed
call, and rang the nominated family member to alert us to the problem.

THanks.

--
Graham J

[Graham.]

I would like to know more about these post-connection tones.
Do they sound like DTMF or single tones, or is it a burst of modem
data?
It is not uncommon when handling data through an ATA (typically FAX)
to have to tweek the TX and RX gain parameters.

Can we hear the failed session for ourselves?
--

Graham.
%Profound_observation%

[Graham J]

Graham. wrote:
> I would like to know more about these post-connection tones.
> Do they sound like DTMF or single tones, or is it a burst of modem
> data?
> It is not uncommon when handling data through an ATA (typically FAX)
> to have to tweek the TX and RX gain parameters.
>
> Can we hear the failed session for ourselves?
>

This is at a remote site and I have nothing at present to record the
failed session. However I listened via a mobile while my friend made a
test call. The sequence went:

1. Press button.

2. Recorded message "do not worry, your call will be answered shortly ...".

3. Rapid sequence of DTMF noises, dialling a number of the form 0208522****

4. Silence for perhaps 3 seconds.

5. Three pairs of two tones, might be DTMF but at a higher pitch. I've
listened to the demo at:

https://www.youtube.com/watch?v=qg9xrP_U1Cg

... and it is not any of the numbers 0 - 9 but # and * are not
demonstrated. The same two tones are repeated three times as if I am
hearing #* #* #* - the sound level is loud, the same as for the DTMF
tones used for dialling.

There are some clicks, then after 28 seconds the line drops.

The noises are nothing like the modem data for a fax machine or dial-up
modem.


If I dial the number from here, I hear my phone make the DTMF noises,
then 3 seconds of silence during which there is a faint click. After
that I hear the three pairs of two tones, a short silence, then several
more short groups of different tones. The CLI I present is clearly not
known to the answering service, so perhaps it knows to try several
different protocols.

Hope this helps.

--
Graham J

[Andy Burns]

Graham J wrote:

> Three pairs of two tones, might be DTMF but at a higher pitch.

Does the ATA have any settings to control in-band vs out-of-band DTMF?
Or settings to disable lower bandwidth codecs that might mangle DTMF?

[Graham J]

In a word, no. I show below the parameters I can see, with some blanked
out for anonymity. There's a similar group of parameters for Line 2,
which is not currently registered.

Cheers

--
Graham J

-------------------------------------------------------------------------


DHCP: Enabled Current IP: 192.168.26.204
Host Name: LinksysPAP Domain: Home
Current Netmask: 255.255.255.0 Current Gateway: 192.168.26.254
Primary DNS: 192.168.26.254
Secondary DNS: 8.8.8.8 208.67.222.222
Product Information
Product Name: PAP2T Serial Number: FLI00L123180
Software Version: 3.1.15(LS) Hardware Version: 0.3.5
MAC Address: 687F745AE08B Client Certificate: Installed
Customization: Open
System Status
Current Time: 6/29/2018 17:10:21 Elapsed Time: 10:46:47
Broadcast Pkts Sent: 6 Broadcast Bytes Sent: 2052
Broadcast Pkts Recv: 2401 Broadcast Bytes Recv: 235216
Broadcast Pkts Dropped: 0 Broadcast Bytes Dropped: 0
RTP Packets Sent: 38976 RTP Bytes Sent: 9354240
RTP Packets Recv: 49865 RTP Bytes Recv: 7745180
SIP Messages Sent: 1191 SIP Bytes Sent: 740682
SIP Messages Recv: 686 SIP Bytes Recv: 331484
External IP:
Line 1 Status
Display Name: Ext 204 User ID: XXXXXXXX*204
Hook State: On Registration State: Online
Last Registration At: 6/29/2018 17:09:25 Next Registration In: 1 s
Message Waiting: No Call Back Active: No
Last Called Number: 0208522**** Last Caller Number: 01366******
Mapped SIP Port:
Call 1 State: Idle Call 2 State: Idle
Call 1 Tone: None Call 2 Tone: None
Call 1 Encoder: Call 2 Encoder:
Call 1 Decoder: Call 2 Decoder:
Call 1 FAX: Call 2 FAX:
Call 1 Type: Call 2 Type:
Call 1 Remote Hold: Call 2 Remote Hold:
Call 1 Callback: Call 2 Callback:
Call 1 Peer Name: Call 2 Peer Name:
Call 1 Peer Phone: Call 2 Peer Phone:
Call 1 Duration: Call 2 Duration:
Call 1 Packets Sent: Call 2 Packets Sent:
Call 1 Packets Recv: Call 2 Packets Recv:
Call 1 Bytes Sent: Call 2 Bytes Sent:
Call 1 Bytes Recv: Call 2 Bytes Recv:
Call 1 Decode Latency: Call 2 Decode Latency:
Call 1 Jitter: Call 2 Jitter:
Call 1 Round Trip Delay: Call 2 Round Trip Delay:
Call 1 Packets Lost: Call 2 Packets Lost:
Call 1 Packet Error: Call 2 Packet Error:
Call 1 Mapped RTP Port: Call 2 Mapped RTP Port:

[Andrew Gabriel]

In article <ph5m4r$s6e$1...@dont-email.me>,

Graham J <gra...@invalid.com> writes:
> Andy Burns wrote:
>> Graham J wrote:
>>
>>> Three pairs of two tones, might be DTMF but at a higher pitch.
>>
>> Does the ATA have any settings to control in-band vs out-of-band DTMF?
>> Or settings to disable lower bandwidth codecs that might mangle DTMF?
>
> In a word, no. I show below the parameters I can see, with some blanked
> out for anonymity. There's a similar group of parameters for Line 2,
> which is not currently registered.

I believe the PAP2 is similar to the SPA3000 (but with
a second phone line instead of the exchange line interface)
which I have configured.

You might want to look at the codec selection, for which
you need to select admin login, and then advanced (at least
on the SPA3000).

Then you should find a menu like this in the Line 1 and
Line 2 tabs:

Audio Configuration

Preferred Codec:G711u Silence Supp Enable:no
Use Pref Codec Only:no Silence Threshold:medium
G729a Enable:yes Echo Canc Enable:yes
G723 Enable:yes Echo Canc Adapt Enable:yes
G726-16 Enable:yes Echo Supp Enable:yes
G726-24 Enable:yes FAX CED Detect Enable:yes
G726-32 Enable:yes FAX CNG Detect Enable:yes
G726-40 Enable:yes FAX Passthru Codec:G711u
DTMF Process INFO:yes FAX Codec Symmetric:yes
DTMF Process AVT:yes FAX Passthru Method:NSE
DTMF Tx Method:Auto DTMF Tx Mode:Strict
FAX Process NSE:yes Hook Flash Tx Method:none
FAX Disable ECAN:no Release Unused Codec:yes
Symmetric RTP:yes

These are the values I happen to have set which were the
SPA3000 defaults. It might be that you have some of the
better ones disabled as many VOIP providers did in the
early days to limit internet bandwidth used.

I don't think it can do any harm to enable them, as this
is negotiated with the other end anyway, and one which
both ends support is selected. You might want to note
what they all are to start with though, in case you need
to revert.

I used to know what many of these codecs were, 15 years
ago when I set the system up ;-)
Others here may be able to comment.

--
Andrew Gabriel
[email address is not usable -- followup in the newsgroup]

[Graham.]

It looks to me that Graham's PAP-2 has had "provisioning" enabled and
is therefore locked to Voipfone, and most of the settings hidden.


This is what the "Line 1" tab of my not-locked PAP-2 looks like.
https://www.flickr.com/gp/g3zvt/g41R7k
--

Graham.
%Profound_observation%

[Graham.]

and here is the top of the "Line 1" page, not how we are in "Advanced
View"
https://www.flickr.com/gp/g3zvt/VV142y
--

Graham.
%Profound_observation%

[Graham J]

Thanks. I can see these parameters, and they are virtually the same. I
note that my DTMF mode is "Auto".

As others have suggested, Voipfone does provision this device
automatically so if I make changes they may revert later. However I
have a support call with Voipfone arranged for Monday and if a specific
change gets the system to work then I'm sure Voipfone will be able to
save a custom configuration for me.

I've opened a discussion in uk.telecom.broadband to see if there is a
network-based product that I can simply plug into the LAN; something
that is integrated with a call centre since the important issue is that
if the patient invokes the alarm the call centre will try a list of
contacts until somebody is found who can visit the patient and resolve
any problems.

--
Graham J

[Bob Eager]

Assuming that he is not just logged in as a user, rather than admin. Even
on an unlocked unit, a lot is hidden as 'user'.

[spuorg...@gowanhill.com]

On Friday, 29 June 2018 15:16:48 UTC+1, Graham J wrote:
> ... and it is not any of the numbers 0 - 9 but # and * are not
> demonstrated. The same two tones are repeated three times as if I am
> hearing #* #* #* - the sound level is loud, the same as for the DTMF
> tones used for dialling.

The fourth column ABCD tones perhaps?

The installation manual page 20 says

It is important to set the correct destination type otherwise the recipient of the alarm call will not be able to deal with it correctly. A CC call expects a particular handshake from the control centre, a PR call requires a recipient with a touch tone telephone and a POTS call is a normal telephone call (i.e. fast dial button).

There is a programming code 48 accessible from the series connected phone to change the destination type.

There is also an option on page 22 to change from DTMF to Sequential Tone Multi Frequency (STMF)

A setting of "Unit always uses STMF (for use when operating on GSM and/or NGN networks)." This setting may be worth trying for a VoIP line.

NOTE: Before using STMF, the PNC monitoring centre and back up centre must be configured to receive STMF protocol. It's possible that your Careline provider doesn't support STMF.

The above comment does suggest another option, of using a GSM connection rather than VoIP. This might also be advisable if the modem/router/PAP2 isn't UPS backed-up.

Apart from trying STMF the options are
- try a different codec and DTMF settings as advised
- try a different ATA
- try a different VoIP host. Sipgate are free to try assuming the Careline number is 0800 so you don't need to add any credit.

Owain

[Graham J]

spuorg...@gowanhill.com wrote:
> On Friday, 29 June 2018 15:16:48 UTC+1, Graham J wrote:
>> ... and it is not any of the numbers 0 - 9 but # and * are not
>> demonstrated. The same two tones are repeated three times as if I am
>> hearing #* #* #* - the sound level is loud, the same as for the DTMF
>> tones used for dialling.
>
> The fourth column ABCD tones perhaps?
>
> The installation manual page 20 says
>
> It is important to set the correct destination type otherwise the recipient of the alarm call will not be able to deal with it correctly. A CC call expects a particular handshake from the control centre, a PR call requires a recipient with a touch tone telephone and a POTS call is a normal telephone call (i.e. fast dial button).
>
> There is a programming code 48 accessible from the series connected phone to change the destination type.
>
> There is also an option on page 22 to change from DTMF to Sequential Tone Multi Frequency (STMF)
>
> A setting of "Unit always uses STMF (for use when operating on GSM and/or NGN networks)." This setting may be worth trying for a VoIP line.
>
> NOTE: Before using STMF, the PNC monitoring centre and back up centre must be configured to receive STMF protocol. It's possible that your Careline provider doesn't support STMF.

There's nobody at Careline365 who understands the issue so I can't get
them to change the settings at their monitoring centre.

> The above comment does suggest another option, of using a GSM connection rather than VoIP. This might also be advisable if the modem/router/PAP2 isn't UPS backed-up.

The Tunstall device is decribed as supporting a module to make a GSM
connection. However they haven't responded to an email request for a
quote to supply a device and registration with an appropriate monitoring
service. So I will try ringing them ...

> Apart from trying STMF the options are
> - try a different codec and DTMF settings as advised
> - try a different ATA
> - try a different VoIP host. Sipgate are free to try assuming the Careline number is 0800 so you don't need to add any credit.

No, the number tht the Tunstall device dials for Careline365 is 0208 522
**** so clearly not free.

There is a sales number for Careline365 which is free, but the people
there stick to the specification that their device/system only works
over a BT line.

Thanks anyway.

--
Graham J

[spuorg...@gowanhill.com]

On Tuesday, 10 July 2018 07:40:24 UTC+1, Graham J wrote:
> > There is a programming code 48 accessible from the series connected phone
> > to change the destination type.
> > There is also an option on page 22 to change from DTMF to Sequential Tone
> > Multi Frequency (STMF)

> There's nobody at Careline365 who understands the issue so I can't get
> them to change the settings at their monitoring centre.

They might not be able to change even if they understood the issue; it might need a software upgrade they haven't paid for.

However you could try changing the settings on the unit from a series-connected phone as described in the manual (if it hasn't been locked down by Careline).

If the Cat5 wiring isn't being used for Gig ethernet it will have 2 spare pairs, which might be used to extend the phone line.

Owain

[Graham J]

spuorg...@gowanhill.com wrote:
> On Tuesday, 10 July 2018 07:40:24 UTC+1, Graham J wrote:
>>> There is a programming code 48 accessible from the series connected phone
>>> to change the destination type.
>>> There is also an option on page 22 to change from DTMF to Sequential Tone
>>> Multi Frequency (STMF)
>> There's nobody at Careline365 who understands the issue so I can't get
>> them to change the settings at their monitoring centre.
>
> They might not be able to change even if they understood the issue; it might need a software upgrade they haven't paid for.

Fair comment ...

>
> However you could try changing the settings on the unit from a series-connected phone as described in the manual (if it hasn't been locked down by Careline).

I didn't want to do anything that would not be supported by Careline365
so apart from understanding why the Tunstall device won't work over VoIP
what I'm now looking for is an alternative product that will work over
an IP connection - a product that is fully supported by its supplier so
I'm not involved in its configuration or management.

>
> If the Cat5 wiring isn't being used for Gig ethernet it will have 2 spare pairs, which might be used to extend the phone line.

Sadly the path from the internet router to the kitchen is achieved with
a single installed Cat5 cable - don't blame me: the builder installed it
before I got involved. The extension of the network from the kitchen to
the granny annexe (and other locations) uses a network switch. There is
a spare cable along this path, but I put that in for redundancy. So
using the spare pairs isn't really an option.

Thanks for thinking about he problem for me.

--
Graham J

[Graham.]

>
>>
>> If the Cat5 wiring isn't being used for Gig ethernet it will have 2 spare pairs, which might be used to extend the phone line.
>
>Sadly the path from the internet router to the kitchen is achieved with
>a single installed Cat5 cable

As Owen said, that means you have two redundent pairs if you forego a
"gigabit" connection " assuming both the router and the switch support
1000Mb/s

A 100Mb/s only needs two pairs and is overkill for most xDSL internet
connections.
So assuming she has exclusive use of the landline, that may well be
your best and simplest option.

If it still needs to be VoIP I concur with Andy Burns, and you need an
unlocked ATA so you have full control of the parameters, particualay
DTMF method and the codec used. I cant see any reason why you cant
continue with the same voipfone account in that device, but others
like Sipgate and Dallmont clones are also availible.

--

Graham.
%Profound_observation%

[Graham J]

Graham. wrote:
>
>>
>>>
>>> If the Cat5 wiring isn't being used for Gig ethernet it will have 2 spare pairs, which might be used to extend the phone line.
>>
>> Sadly the path from the internet router to the kitchen is achieved with
>> a single installed Cat5 cable
>
> As Owen said, that means you have two redundent pairs if you forego a
> "gigabit" connection " assuming both the router and the switch support
> 1000Mb/s
>
> A 100Mb/s only needs two pairs and is overkill for most xDSL internet
> connections.
> So assuming she has exclusive use of the landline, that may well be
> your best and simplest option.

[snip]

Owen is of course correct. It would be possible to wire up the 2 spare
pairs to take the phone via the cat5 cable all the way to the granny
annexe, and back. The Tunstall device is intended to be the only device
connected to the incoming phone line (excluding the microfilter and
router, of course); and all the ordinary phone extensions can be plugged
into its output. That way when an alarm is generated the Tunstall
device can guarantee to seize the line and make the outgoing call.

But given the physical arrangements at the site I didn't want to have
any non-standard wiring, or anything that might be disconnected by mistake.

By contrast I monitor the broadband connection and have a LAN-to-LAN VPN
through the router which allows me to monitor the Linksys ATA - so I can
see within a couple of minutes if any of these fail.

--
Graham J

[David Woolley]

On 29/06/18 10:11, Graham J wrote:

> Tunstall cannot confirm that the unit will work with VoIP, but their
> technical support people don't really understand the question, confusing
> it with integration with a fibre connection to the home.

I believe these devices contain a modem (ignoring that DTMF
encoder/decoders are modems!).

They will have been tested for use on VoIP, but only to the 21CN
specification, i.e. using G.711 and with very low latency. Any
commercial VoIP network is likely to have far too much latency for any
modem that is sensitive to latency.

Also, of the parameters mentioned, I would turn off echo suppression.
That's one of the purposes of the 2100 Hz answer tone that modems
generate, but the ATA may well not understand that.

Codecs have already been discussed, but I would add that the provider
may, remotely, force the use of incompatible codec; they are selling a
voice only service.

I'd also note that this is a safety life application and I'd be
surprised if VoIP operators didn't disclaim liability.

[Graham J]

Noted.

Do you suppose Openreach and BT would accept liability ???

--
Graham J

[Graham.]

It used to be, (and maybe still is in the case of old boxes with no
Internet connection), a requirement that Sky Multi-room boxes were
permanently connected to the same telephone line, so Sky would know
they were not being used in separate premises.

I tried, quite hard as it happened, to coax a sky box to communicate
via VoIP so the CLI could be spoofed, but it stubbornly refused to
negotiate a connection.

Others that tried this all seemed to fail too.

I came to the conclusion that Sky had purposely made their end
particularly sensitive to latency.
--

Graham.
%Profound_observation%