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uk.telecom.voip (UK VOIP) (uk.telecom.voip) Discussion of topics relevant to packet based voice technologies including Voice over IP (VoIP), Fax over IP (FoIP), Voice over Frame Relay (VoFR), Voice over Broadband (VoB) and Voice on the Net (VoN) as well as service providers, hardware and software for use with these technologies. Advertising is not allowed.

Integration of VoIP with Tunstall healthcare alarm



 
 
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  #1  
Old June 29th 18, 11:11 AM posted to uk.telecom.voip
Graham J[_2_]
external usenet poster
 
Posts: 723
Default Integration of VoIP with Tunstall healthcare alarm

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 he

https://uk.tunstall.com/services/our...s/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


  #2  
Old June 29th 18, 01:10 PM posted to uk.telecom.voip
Andy Burns[_5_]
external usenet poster
 
Posts: 315
Default Integration of VoIP with Tunstall healthcare alarm

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
  #3  
Old June 29th 18, 01:29 PM posted to uk.telecom.voip
Graham.[_3_]
external usenet poster
 
Posts: 289
Default Integration of VoIP with Tunstall healthcare alarm

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 he

https://uk.tunstall.com/services/our...s/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


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/e...pura-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%
  #4  
Old June 29th 18, 02:01 PM posted to uk.telecom.voip
Graham J[_2_]
external usenet poster
 
Posts: 723
Default Integration of VoIP with Tunstall healthcare alarm

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/e...pura-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


  #5  
Old June 29th 18, 02:20 PM posted to uk.telecom.voip
Graham.[_3_]
external usenet poster
 
Posts: 289
Default Integration of VoIP with Tunstall healthcare alarm

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%
  #6  
Old June 29th 18, 04:16 PM posted to uk.telecom.voip
Graham J[_2_]
external usenet poster
 
Posts: 723
Default Integration of VoIP with Tunstall healthcare alarm

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


  #7  
Old June 29th 18, 06:04 PM posted to uk.telecom.voip
Andy Burns[_5_]
external usenet poster
 
Posts: 315
Default Integration of VoIP with Tunstall healthcare alarm

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?
  #8  
Old June 29th 18, 06:18 PM posted to uk.telecom.voip
Graham J[_2_]
external usenet poster
 
Posts: 723
Default Integration of VoIP with Tunstall healthcare alarm

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.

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:
  #9  
Old June 29th 18, 11:44 PM posted to uk.telecom.voip
Andrew Gabriel
external usenet poster
 
Posts: 223
Default Integration of VoIP with Tunstall healthcare alarm

In article ,
Graham J 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]
  #10  
Old June 30th 18, 03:25 AM posted to uk.telecom.voip
Graham.[_3_]
external usenet poster
 
Posts: 289
Default Integration of VoIP with Tunstall healthcare alarm

In article ,
Graham J 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.



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%
 




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