Autodial VoIP adapter or similar..
 

Does anyone know of a VoIP ATA adapter that you could configure to auto
dial a simple DTMF number which might be just one or two digits long.
This won't be used over a public IP circuit more over VPN and or LAN
connections into a private Asterix based PABX system.

We just need it somehow when the handset is raised to make that call.

Also..

At a called unit is there one which will do an auto answer function?.

Bit of an odd application I know but if anyone has any idea of the
equipment that might do just that I'd be obliged..


Its a bit of a VoIP based duplex intercom system as such.

Put that another way say at one end we can make a "loop" over the output
connections from the "A" end ATA . This will then auto-dial the "B" end.

At the B end that unit will detect being called and then answer the line
..

Audio can then pass over the link thus formed in a bi-directional
fashion. Its not a conventional phone were connecting but some audio
amps and suchlike.


Thanks in advance....


--
Tony Sayer

Re: Autodial VoIP adapter or similar..
 

On Sun, 10 Feb 2013 09:11:57 +0000, alexd wrote:

> tony sayer (for it is he) wrote:
>
>> Does anyone know of a VoIP ATA adapter that you could configure to auto
>> dial a simple DTMF number which might be just one or two digits long.
>
> PAP2T does [did], hopefully its successors do. Grandstream HT286 does
> ["Offhook auto-dial" in the web interface].
>
>> At a called unit is there one which will do an auto answer function?.
>
> The Linksys ATA Administrator Guide alludes to it, but doesn't say where
> you turn it on.

You do it with the dial plan:

(P0<:123>)

which says "time out after 0 seconds off hook, then dial 123".


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Re: Autodial VoIP adapter or similar..
 

On Sun, 10 Feb 2013 10:34:45 +0000, Bob Eager wrote:

> On Sun, 10 Feb 2013 09:11:57 +0000, alexd wrote:
>
>> tony sayer (for it is he) wrote:
>>
>>> Does anyone know of a VoIP ATA adapter that you could configure to
>>> auto dial a simple DTMF number which might be just one or two digits
>>> long.
>>
>> PAP2T does [did], hopefully its successors do. Grandstream HT286 does
>> ["Offhook auto-dial" in the web interface].
>>
>>> At a called unit is there one which will do an auto answer function?.
>>
>> The Linksys ATA Administrator Guide alludes to it, but doesn't say
>> where you turn it on.
>
> You do it with the dial plan:
>
> (P0<:123>)
>
> which says "time out after 0 seconds off hook, then dial 123".

Sorry, interpolated that in the wrong place. That's for the off-hook
autodial, of course.



--
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Re: Autodial VoIP adapter or similar..
 

On Sat, 9 Feb 2013 21:22:04 +0000, tony sayer <tony@bancom.co.uk>
wrote:

>
>Does anyone know of a VoIP ATA adapter that you could configure to auto
>dial a simple DTMF number which might be just one or two digits long.
>This won't be used over a public IP circuit more over VPN and or LAN
>connections into a private Asterix based PABX system.
>
>We just need it somehow when the handset is raised to make that call.
>
>Also..
>
>At a called unit is there one which will do an auto answer function?.
>
>Bit of an odd application I know but if anyone has any idea of the
>equipment that might do just that I'd be obliged..
>
>
>Its a bit of a VoIP based duplex intercom system as such.
>
>Put that another way say at one end we can make a "loop" over the output
>connections from the "A" end ATA . This will then auto-dial the "B" end.
>
>At the B end that unit will detect being called and then answer the line
>.
>
>Audio can then pass over the link thus formed in a bi-directional
>fashion. Its not a conventional phone were connecting but some audio
>amps and suchlike.
>
>
>Thanks in advance....


I have just demonstrated to myself on the unused "line 2" of a PAP2
that I can do the hot-dial part.

Simply put the number you want to dial, preceded by a colon in the
dialplan field without the usual brackets.

Lift the handset and the number is dialled after a couple of seconds.

I imagine this will work on any Sipura/Linksys product with a similar
dialplan

Re: Autodial VoIP adapter or similar..
 

In article <kf7o7r$cif$1@dont-email.me>, alexd <troffasky@hotmail.com>
scribeth thus
>tony sayer (for it is he) wrote:
>
>> Does anyone know of a VoIP ATA adapter that you could configure to auto
>> dial a simple DTMF number which might be just one or two digits long.
>
>PAP2T does [did], hopefully its successors do. Grandstream HT286 does
>["Offhook auto-dial" in the web interface].
>
>> At a called unit is there one which will do an auto answer function?.
>
>The Linksys ATA Administrator Guide alludes to it, but doesn't say where you
>turn it on. In a way I can understand this, because the phone is a separate
>entity to the ATA, usually with some kind of switch across the two wires, so
>how could any ATA ever auto answer? An IP phone is a different matter, of
>course, as it's all in one.
>
>> Bit of an odd application I know
>
>What you're looking for is often referred to intercom or paging
>functionality. I take it you looked at some ethernet-audio bridges, recoiled
>in horror at the price and thought "I bet I could do that with an ATA" :-)
>

Yes and err .. yes!..

The application is for the control of a Two way radio system/s. You can
do this with the Barix Annuncicom but thats a very expensive box for
what it is. We have used units by Multitech in the past but these are
now going out of production and as usual aren't that cheap either.

It does seem on the face of it a rather simple application we are
attempting to link Two or more radio base stations together and this can
be done using a phone patch interconnect via an ATA but these too aren't
that cheap either;!..

In a normal app for two way radio you need to key the TX on and off but
in this instance if its keyed on for the call duration that isn't a
problem. As alluded to we have audio for transmission in both directions
and this is at whatever level we want it to be and supplied on floating
balanced 600 ohm audio transformers. We can at the one end use a simple
command to "loop" the line thus causing the auto dial thru an Asterix
box, the problem might be in auto answering the other end but as and
when that does if we can pick up a signal that can drive a relay or
similar then that will answer that problem.

We can and do use sub audio signalling tones, these are as implied sub
the normal audio band and are in discreet frequency steps up to 250.3 Hz
this frequency does carry thru phone bandwidth lines and is normally
transmitted several dB below the audio level.

As you can understand a simple Linksys or Grandstream bit around 30 to
40 UKP is much simpler than a several hundred pound unit especially when
you need a few of them;!..

An asterisk box, most any decent PC I've even heard of them using it now
on the raspberry pi!. The whole will be over the net on a VPN
arrangement.

Anyway thanks for that info thus far:)..


--
Tony Sayer

Re: Autodial VoIP adapter or similar..
 

On Sun, 10 Feb 2013 13:31:49 +0000, Graham. wrote:

> On Sat, 9 Feb 2013 21:22:04 +0000, tony sayer <tony@bancom.co.uk> wrote:
>
>
>>Does anyone know of a VoIP ATA adapter that you could configure to auto
>>dial a simple DTMF number which might be just one or two digits long.
>>This won't be used over a public IP circuit more over VPN and or LAN
>>connections into a private Asterix based PABX system.
>>
>>We just need it somehow when the handset is raised to make that call.
>>
>>Also..
>>
>>At a called unit is there one which will do an auto answer function?.
>>
>>Bit of an odd application I know but if anyone has any idea of the
>>equipment that might do just that I'd be obliged..
>>
>>
>>Its a bit of a VoIP based duplex intercom system as such.
>>
>>Put that another way say at one end we can make a "loop" over the output
>>connections from the "A" end ATA . This will then auto-dial the "B" end.
>>
>>At the B end that unit will detect being called and then answer the line
>>.
>>
>>Audio can then pass over the link thus formed in a bi-directional
>>fashion. Its not a conventional phone were connecting but some audio
>>amps and suchlike.
>>
>>
>>Thanks in advance....
>
>
> I have just demonstrated to myself on the unused "line 2" of a PAP2 that
> I can do the hot-dial part.
>
> Simply put the number you want to dial, preceded by a colon in the
> dialplan field without the usual brackets.
>
> Lift the handset and the number is dialled after a couple of seconds.
>
> I imagine this will work on any Sipura/Linksys product with a similar
> dialplan

Indeed; as I posted three hours previously in uk.telecom.voip!

--
Use the BIG mirror service in the UK: http://www.mirrorservice.org
My posts (including this one) are my copyright and if @diy_forums on
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*lightning surge protection* - a w_tom conductor

Re: Autodial VoIP adapter or similar..
 

tony sayer wrote:

>
> In a normal app for two way radio you need to key the TX on and off but
> in this instance if its keyed on for the call duration that isn't a
> problem. As alluded to we have audio for transmission in both directions

That concerns me a bit. PMR is normally licensed on the basis that it
is low duty cycle with short transmission times. As you mention CTCSS,
you may have frequency sharing, which makes this essential.

>
> We can and do use sub audio signalling tones, these are as implied sub
> the normal audio band and are in discreet frequency steps up to 250.3 Hz
> this frequency does carry thru phone bandwidth lines and is normally
> transmitted several dB below the audio level.

Note that, whilst CTCSS should work through a 3.1KHz audio channel
(G.711 mu- or A-Law), I would not expect it to carry through a speech
channel (GSM or G.729 codecs).

>
> As you can understand a simple Linksys or Grandstream bit around 30 to
> 40 UKP is much simpler than a several hundred pound unit especially when
> you need a few of them;!..

Depends on how much your time costs, and whether they have a sensible
succession plan for when you are no longer there to maintain the system.

>
> An asterisk box, most any decent PC I've even heard of them using it now
> on the raspberry pi!. The whole will be over the net on a VPN
> arrangement.

I think the Raspberry Pi thing is an amateur thing, to show that it can
be done, although there is no reason why it shouldn't support a small,
pure VoIP system (you would need to take steps to minimise wear on the
flash memory card, if using it commercially).

Although I have not looked into it, there is community supported (i.e.
not supported by Digium) code in Asterisk for handling two way radios.
I don't know what hardware it assumes.
>

Re: Autodial VoIP adapter or similar..
 

"David Woolley" <david@ex.djwhome.demon.invalid> wrote in message
news:kf8cak$tkk$1@dont-email.me...
> tony sayer wrote:
>
>>
>> In a normal app for two way radio you need to key the TX on
>> and off but
>> in this instance if its keyed on for the call duration that
>> isn't a
>> problem. As alluded to we have audio for transmission in both
>> directions
>
> That concerns me a bit. PMR is normally licensed on the basis
> that it is low duty cycle with short transmission times. As
> you mention CTCSS, you may have frequency sharing, which makes
> this essential.
>
[snip]

CTCSS does not necessarily mean frequency sharing. It will have
been licence specified if talkthrough is also approved, or
high-band near the coast where marine RFI is prevalent.

Tony,
What was that about floating balanced 600R line? If it is
floating it isn't balanced; balanced means that the feed
transformer at the exchange has a centre-tap earth. I would guess
you have A-over-D which is only floating.


--
Woody

harrogate three at ntlworld dot com

Re: Autodial VoIP adapter or similar..
 

In article <kf8cak$tkk$1@dont-email.me>, David Woolley <david@ex.djwhome
..demon.invalid> scribeth thus
>tony sayer wrote:
>
>>
>> In a normal app for two way radio you need to key the TX on and off but
>> in this instance if its keyed on for the call duration that isn't a
>> problem. As alluded to we have audio for transmission in both directions
>
>That concerns me a bit. PMR is normally licensed on the basis that it
>is low duty cycle with short transmission times. As you mention CTCSS,
>you may have frequency sharing, which makes this essential.

In this instance these days PMR in some areas isn't quite as busy as it
once was, and out in the sticks on some frequency bands;!..

This is more a linked base station/s thats in what's called Talkthrough
and also telephone interconnect as PABX is now licenced as default. It
can be that an exchange of conversations takes a matter of seconds or
can go on for longer i.e. base speaks, then mobile then base then
another mobile etc...

>
>>
>> We can and do use sub audio signalling tones, these are as implied sub
>> the normal audio band and are in discreet frequency steps up to 250.3 Hz
>> this frequency does carry thru phone bandwidth lines and is normally
>> transmitted several dB below the audio level.
>
>Note that, whilst CTCSS should work through a 3.1KHz audio channel
>(G.711 mu- or A-Law), I would not expect it to carry through a speech
>channel (GSM or G.729 codecs).

Indeed as their ISTR vocoders, but in the instances we have tried it, it
does work!. Its a quite a robust beast and does work under very noisy
conditions as well. As alluded to earlier we did they this in a round
path with two Multitech units and it worked at normal levels down to
151.4 Hz which is much more than adequate...
>
>>
>> As you can understand a simple Linksys or Grandstream bit around 30 to
>> 40 UKP is much simpler than a several hundred pound unit especially when
>> you need a few of them;!..
>
>Depends on how much your time costs, and whether they have a sensible
>succession plan for when you are no longer there to maintain the system.

Well time is my own and when I'm not there thats no problem as it will
all close down but I haven't any ideas re retiring as yet I'm only 61
which I'm told isn't that old these days;!.

JOOI a lot of this equipment is quite old and analogue but ticks, is low
power and low voltage and still works fine. One bit if it was made in
1985 thats the aerial filter multicoupler system and some base stations
date from around 1990 odd TAIT T800 series MK 1 and 2, and are still in
excellent nick:)...

>
>>
>> An asterisk box, most any decent PC I've even heard of them using it now
>> on the raspberry pi!. The whole will be over the net on a VPN
>> arrangement.
>
>I think the Raspberry Pi thing is an amateur thing, to show that it can
>be done, although there is no reason why it shouldn't support a small,
>pure VoIP system (you would need to take steps to minimise wear on the
>flash memory card, if using it commercially).

Indeed it is but it seems to be finding a lot of uses for itself in some
simple applications. We have made up an audio streamer for a radio
station on one and its fine, and low power consumption and best of all
quiet;)..
>
>Although I have not looked into it, there is community supported (i.e.
>not supported by Digium) code in Asterisk for handling two way radios.
>I don't know what hardware it assumes.

There are some proprietary systems around commercially but there quite
expensive for what they are and do. There is some Ham kit about but
seems to be dedicated for the Ham user most suites wanting a callsign
and there isn't one!.

Anyways I think this might be coming together just a simple PSU can run
off the existing kit theres a low cost 1 U rack case ABS plastic to put
it in a few relays and transformers perhaps and I might now have the
answer to the auto answer part..

Cheers...
>>

--
Tony Sayer

Re: Autodial VoIP adapter or similar..
 

In article <kf8d30$2nk$1@dont-email.me>, Woody <harrogate3@ntlworld.com>
scribeth thus
>"David Woolley" <david@ex.djwhome.demon.invalid> wrote in message
>news:kf8cak$tkk$1@dont-email.me...
>> tony sayer wrote:
>>
>>>
>>> In a normal app for two way radio you need to key the TX on
>>> and off but
>>> in this instance if its keyed on for the call duration that
>>> isn't a
>>> problem. As alluded to we have audio for transmission in both
>>> directions
>>
>> That concerns me a bit. PMR is normally licensed on the basis
>> that it is low duty cycle with short transmission times. As
>> you mention CTCSS, you may have frequency sharing, which makes
>> this essential.
>>
>[snip]
>
>CTCSS does not necessarily mean frequency sharing. It will have
>been licence specified if talkthrough is also approved, or
>high-band near the coast where marine RFI is prevalent.

Indeed , in some areas and frequency bands nowadays there almost
exclusive users..
>
>Tony,
>What was that about floating balanced 600R line? If it is
>floating it isn't balanced; balanced means that the feed
>transformer at the exchange has a centre-tap earth. I would guess
>you have A-over-D which is only floating.
>

Ah!, now this is from the local base stations its floating and balanced
there is no connection at all to the PSTN.

Floating and balanced I interpret as a balanced source such as a
transformer winding that only has two ends as such and isn't connected
anywhere like say a centre tap to earth like we use to use for remote DC
keying which I'm sure you'll remember;)..


Course what you connect to an ATA on a VoIP system thats essentially all
internal a PABX as such and doesn't have to be approved for PSTN
connection...

>

--
Tony Sayer