Autodial VoIP adapter or similar..

Discussion in 'UK VOIP' started by tony sayer, Feb 9, 2013.

  1. tony sayer

    tony sayer Guest

    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
    tony sayer, Feb 9, 2013
    #1
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  2. tony sayer

    Bob Eager Guest

    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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    Bob Eager, Feb 10, 2013
    #2
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  3. tony sayer

    Bob Eager Guest

    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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    Bob Eager, Feb 10, 2013
    #3
  4. tony sayer

    Graham. Guest

    On Sat, 9 Feb 2013 21:22:04 +0000, tony sayer <>
    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
    Graham., Feb 10, 2013
    #4
  5. tony sayer

    tony sayer Guest

    In article <kf7o7r$cif$>, alexd <>
    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
    tony sayer, Feb 10, 2013
    #5
  6. tony sayer

    Bob Eager Guest

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

    > On Sat, 9 Feb 2013 21:22:04 +0000, tony sayer <> 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!

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    Bob Eager, Feb 10, 2013
    #6
  7. 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.
    >
    David Woolley, Feb 10, 2013
    #7
  8. tony sayer

    Woody Guest

    "David Woolley" <> wrote in message
    news:kf8cak$tkk$...
    > 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
    Woody, Feb 10, 2013
    #8
  9. tony sayer

    tony sayer Guest

    In article <kf8cak$tkk$>, David Woolley <
    ..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
    tony sayer, Feb 10, 2013
    #9
  10. tony sayer

    tony sayer Guest

    In article <kf8d30$2nk$>, Woody <>
    scribeth thus
    >"David Woolley" <> wrote in message
    >news:kf8cak$tkk$...
    >> 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
    tony sayer, Feb 10, 2013
    #10
  11. tony sayer

    Graham. Guest

    On 10 Feb 2013 11:06:33 GMT, Bob Eager <> wrote:

    >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.


    Trivial point, but I noticed on my PAP2 whatever argument you use with
    the P parameter, there is never less than a 2.5 second delay, and that
    is true even if you leave it out completely.

    Simply putting :123 without brackets seems to work just as well.
    Graham., Feb 10, 2013
    #11
  12. tony sayer

    tony sayer Guest

    In article <kf7q26$kkf$>, alexd <>
    scribeth thus
    >alexd (for it is he) wrote:
    >
    >> 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?

    >
    >To clarify, how would the phone know that the port has been answered if it
    >didn't ring? In an analogue context, it makes more sense for the thing
    >attached to the ATA to do the auto answering, as other use cases are far
    >less common.
    >
    >However, I've had poke through all the advanced options on my SPA3102 and
    >there's a curious-sounding one called "Streaming Audio Server", and from the
    >fine manual:
    >
    ># This feature allows you to attach an audio source to one of the Linksys
    ># ATA FXS ports and use it as a streaming audio source device. The
    ># corresponding Line (1 or 2) can be configured as a streaming audio server
    ># (SAS) such that when the Line is called, the Linksys ATA answers the call
    ># automatically and starts streaming audio to the calling party provided the
    ># FXS port is off-hook. If the FXS port is on-hook when the incoming call
    ># arrives, the Linksys ATA replies with a SIP 503 response code to indicate
    ># ?Service Not Available.?
    >
    >No mention of what happens to the audio in the other direction.
    >
    >


    Just out of curiosity Linksys or Cisco seem to have the PAPT2 as end of
    life since the middle of last year. I believe the SPA-3102 is the
    current unit replacement for that is that correct?, and if so any idea
    on how to do the "off hook" auto-dial on that at all please?..

    Cheers...

    --
    Tony Sayer
    tony sayer, Feb 10, 2013
    #12
  13. tony sayer

    Bob Eager Guest

    On Sun, 10 Feb 2013 18:01:43 +0000, Graham. wrote:

    > On 10 Feb 2013 11:06:33 GMT, Bob Eager <> wrote:
    >
    >>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.

    >
    > Trivial point, but I noticed on my PAP2 whatever argument you use with
    > the P parameter, there is never less than a 2.5 second delay, and that
    > is true even if you leave it out completely.
    >
    > Simply putting :123 without brackets seems to work just as well.


    Just tested it on my SPA3102 and there is no delay at all, with or
    without the P0. The documentation says P0, however, and I know there is
    variation between models...and probably between firmware versions...so
    I'd feel happier including the P0.

    (I have an SPA2000 and an SPA8000 as well, so that makes me even more
    wary)



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    Bob Eager, Feb 10, 2013
    #13
  14. tony sayer

    Bob Eager Guest

    On Sun, 10 Feb 2013 19:00:29 +0000, tony sayer wrote:

    > In article <kf7q26$kkf$>, alexd <>
    > scribeth thus
    >>alexd (for it is he) wrote:
    >>
    >>> 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?

    >>
    >>To clarify, how would the phone know that the port has been answered if
    >>it didn't ring? In an analogue context, it makes more sense for the
    >>thing attached to the ATA to do the auto answering, as other use cases
    >>are far less common.
    >>
    >>However, I've had poke through all the advanced options on my SPA3102
    >>and there's a curious-sounding one called "Streaming Audio Server", and
    >>from the fine manual:
    >>
    >># This feature allows you to attach an audio source to one of the
    >>Linksys # ATA FXS ports and use it as a streaming audio source device.
    >>The # corresponding Line (1 or 2) can be configured as a streaming audio
    >>server # (SAS) such that when the Line is called, the Linksys ATA
    >>answers the call # automatically and starts streaming audio to the
    >>calling party provided the # FXS port is off-hook. If the FXS port is
    >>on-hook when the incoming call # arrives, the Linksys ATA replies with a
    >>SIP 503 response code to indicate # ?Service Not Available.?
    >>
    >>No mention of what happens to the audio in the other direction.
    >>
    >>
    >>

    > Just out of curiosity Linksys or Cisco seem to have the PAPT2 as end of
    > life since the middle of last year. I believe the SPA-3102 is the
    > current unit replacement for that is that correct?, and if so any idea
    > on how to do the "off hook" auto-dial on that at all please?..


    Just as I described. I just tested it for Graham, as it happens!

    --
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    Bob Eager, Feb 10, 2013
    #14
  15. tony sayer

    tony sayer Guest

    >> Just out of curiosity Linksys or Cisco seem to have the PAPT2 as end of
    >> life since the middle of last year. I believe the SPA-3102 is the
    >> current unit replacement for that is that correct?, and if so any idea
    >> on how to do the "off hook" auto-dial on that at all please?..

    >
    >Just as I described. I just tested it for Graham, as it happens!
    >


    OK thanks.

    I suppose the Cisco SPA112 Analog Adapter (ATA) with 2 FXS ports is very
    much in the same vein as the above to have more than the one "line" at
    any given location will be useful rather then stacking Two single
    channel units together:!,....

    Or even the Four port Grandstream Handytone HT704 Analog Adapter...
    --
    Tony Sayer
    tony sayer, Feb 10, 2013
    #15
  16. tony sayer

    Bob Eager Guest

    On Sun, 10 Feb 2013 20:12:53 +0000, tony sayer wrote:

    >>> Just out of curiosity Linksys or Cisco seem to have the PAPT2 as end
    >>> of life since the middle of last year. I believe the SPA-3102 is the
    >>> current unit replacement for that is that correct?, and if so any idea
    >>> on how to do the "off hook" auto-dial on that at all please?..

    >>
    >>Just as I described. I just tested it for Graham, as it happens!
    >>
    >>

    > OK thanks.
    >
    > I suppose the Cisco SPA112 Analog Adapter (ATA) with 2 FXS ports is very
    > much in the same vein as the above to have more than the one "line" at
    > any given location will be useful rather then stacking Two single
    > channel units together:!,....
    >
    > Or even the Four port Grandstream Handytone HT704 Analog Adapter...


    Yes, the SPA112 is more suitable. I have the SPA2000 which must be an
    ancestor of that. I would suspect that the SPA112 uses the same dialplan
    too, so the auto dialling should still be OK.

    No need to pay for two FXO ports you don't want!



    --
    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
    Twitter wish to tweet them they can pay me £30 a post
    *lightning surge protection* - a w_tom conductor
    Bob Eager, Feb 10, 2013
    #16
  17. alexd wrote:
    >
    > # This feature allows you to attach an audio source to one of the Linksys
    > # ATA FXS ports and use it as a streaming audio source device. The
    >
    > No mention of what happens to the audio in the other direction.
    >

    To be most useful, a streaming audio source would allow multiple
    connections, so I would guess that the the reverse audio is dumped.
    >
    David Woolley, Feb 10, 2013
    #17
  18. tony sayer

    tony sayer Guest

    In article <kf92um$elc$>, David Woolley <
    ..demon.invalid> scribeth thus
    >alexd wrote:
    >>
    >> # This feature allows you to attach an audio source to one of the Linksys
    >> # ATA FXS ports and use it as a streaming audio source device. The
    >>
    >> No mention of what happens to the audio in the other direction.
    >>

    >To be most useful, a streaming audio source would allow multiple
    >connections, so I would guess that the the reverse audio is dumped.
    >>


    It seems that you can on taking the line off hook auto dial after no
    delay a number of numbers to set up a sort of call conference witch will
    suffice for what we need.

    We haven't quite cracked the auto answer yet but it seems a simple cap
    connected bridge diode system and switch relay will do what's needed...


    Also it seems the unit has no audio suppression which means that very
    few packets are transmitted over time so you could I suppose leave it
    connected and use CTCSS signalling for TX keying?..

    Also at this price now that has come from the hundreds down to the 10's
    of quid's and the two port Cisco/Linksys can be had for 30 odd quid so
    that now essentially 15 quid an end per line!, you could use a IP based
    relay system for TX keying and other functions over the same IP
    network?..

    --
    Tony Sayer
    tony sayer, Feb 10, 2013
    #18
  19. tony sayer

    tony sayer Guest

    >>
    >> I suppose the Cisco SPA112 Analog Adapter (ATA) with 2 FXS ports is very
    >> much in the same vein as the above to have more than the one "line" at
    >> any given location will be useful rather then stacking Two single
    >> channel units together:!,....
    >>
    >> Or even the Four port Grandstream Handytone HT704 Analog Adapter...

    >
    >Yes, the SPA112 is more suitable. I have the SPA2000 which must be an
    >ancestor of that. I would suspect that the SPA112 uses the same dialplan
    >too, so the auto dialling should still be OK.
    >
    >No need to pay for two FXO ports you don't want!
    >
    >

    It does seem that the Linksys unit is in all cheaper per port. I was
    looking originally at a very expensive supplier for the Linksys/Cisco
    units..!..

    --
    Tony Sayer
    tony sayer, Feb 10, 2013
    #19
  20. tony sayer

    tony sayer Guest

    In article <kf7o7r$cif$>, alexd <>
    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].
    >


    Just has a long read of the Cisco manual very well written and
    informative..

    >> 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" :)
    >


    Indeed and it now looks like its coming out a fraction of the prices we
    were paying per channel, so a good result I reckon:))...

    Thanks to all who replied, most useful:!..
    --
    Tony Sayer
    tony sayer, Feb 10, 2013
    #20
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