RJ11 to ATA to PSTN cable?

Discussion in 'UK VOIP' started by jkn, Jul 2, 2007.

  1. jkn

    jkn Guest

    Hi All
    following my posting last week I've got my SPA3102 up and running
    VoIP through my Router.

    The next thing I want to do, and am having a little trouble with, is
    using the same phone for both VoIP and PSTN. I've put in a dial plan
    which I think should work (based on postings in this fine newsgroup)
    and connected the ATA 'Line' RJ-11 port to my Telephone socket usng
    the supplied RJ-11 to RJ-11 lead, and a PSTN-to-modem adapter I had
    kicking around (I did remember to buy the other adapter which was
    required, to connect my 'phone to the ATA ;-) ).

    However things don't seem to be working - keying in '*9' to get the
    PSTN line is not giving me the familiar tone. I know that American RJ-
    ll cables come in different varieties; do I need the other style' of
    adapter, or another (which?) variety of RJ-11 to PSTN cable. I see
    that Maplin, for instance, flog two types: would I need a

    "Telephone Socket to Modem Cord"
    This telephone conversion line cord is 3m (approximately) and has a BT
    type telephone plug at one end and a US type telephone plug (RJ11) at
    the other. The conductors are suitably crossed (1-2, 2-1, 3-4 & 4-3)
    and the cord finds most use in connecting fax/modems, etc.

    or a

    "BT to US Extension Cord"
    A range of BT to RJ11 pin to pin extension cables, suitable as a
    replacement line cords for most telephones.

    Thanks for your thoughts

    jon N
    jkn, Jul 2, 2007
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  2. jkn

    Jono Guest

    jkn was thinking very hard :
    Have you not got an old modem lead lying around - RJ11 to BT style? Can
    you canibalise a POTS phone...?

    With the cables as you have them, if you power off the ATA & lift the
    connected handset, do you hear the PSTN line dial tone? You should, if
    the cables/adapters are correct.
    It won't. (could do with seeing your dialplan at this point)

    The Line 1 socket it only ever connected directly to the PSTN line if
    you turn the power off. At all other times, an internal voip call is
    made between Line 1 & PSTN accounts.

    Have you got any credentials set on the PSTN tab? even dummy ones?
    Jono, Jul 2, 2007
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  3. jkn

    jkn Guest

    Hi Jono

    'Yes, possibly' to both - that's why I was asking whether this would
    be necessary. Also see below.
    Since my original postings I've hacked the little adapter I referred
    to, swapping the inner & outer pair. With that done, the phone passes
    your test above. So the implication is that it wasn't right before.

    FWIW my dialplan is:

    No - and if what you say above is true then I guess this could be the
    issue. But I may have to be convinced further, that I have
    misunderstood the operation of this particular feature.

    Jon N
    jkn, Jul 2, 2007
  4. jkn

    Jono Guest

    jkn brought next idea :
    If you have dialtone with the power off, you're a significant step
    Your *9, element (the comma in particular) is what's resposible for
    giving you a "second" dial tone. The pitch of which is determined by
    settings on the regional tab.

    The comma isn't actually necessary, except to provide you with an
    audible indication of choosing a different line.

    You won't hear the BT dial tone, unless you power off the adapter.
    Yes it's true. It is. Really. (I've never found anything to the

    Yes, gw0 is the PSTN line, however, the way the call is set up is for
    an internal VoIP call between the two halves of the device.
    IIWY, I would put the same VoIP credentials on the PSTN tab as you have
    on the Line One tab.

    Have you got anywhere you could upload a saved copy of the SPA's html
    pages for us to look at? Files/Save As .mht. (do this whilst signed in
    as admin/advanced.) Although your usernames will be visible, none of
    your passwords will be.
    Jono, Jul 3, 2007
  5. jkn

    PhilT Guest

    you can use the PSTN out without any VoIP settings attributed to the
    PSTN line so I'm failing to see how this can work.

    It has the capability to receive a voice call and route it out onto
    the PSTN, but I can't see that happening to make a PSTN call from the
    attached handset - you would be charged twice for starters !

    PhilT, Jul 3, 2007
  6. jkn

    jkn Guest

    Hi Jono
    thanks for your reply:

    Yes, I realise this.

    The pitch of which is determined by
    I don't think so.
    Yes, I get that.
    OK - I tried changing the outside dial tone setting, from

    [email protected],[email protected];10(*/0/1+2)

    to something else, and the difference between the two 'tones after I
    press *9' was audible - so I'll grant you this one ;-)
    I *think* PhilT in the posting below is also questioning this. I
    haven't tried his proof yet...
    I guess by 'credentials' you mean Proxy & Registration settings etc?

    As it happens I have done that, and things seem to be getting better.
    I can only test at certain times of day for familial harmony...
    I can do that but at the moment things are looking promising. My
    question has changed a bit, now it's more like "What is the correct
    description of how making a PSTN call works with this setup?"

    jon n
    jkn, Jul 3, 2007
  7. jkn

    Jono Guest

    PhilT explained :
    We are talking about an SPA3X0X aren't we?

    I know it's not entirely necessary to have VoIP credentials on the PSTN
    page, however, some of the gateway function only work if there are some
    in there, at least in my experience. So, as the PSTN voip credentials
    won't cause a 2nd registration with your ITSP to occur, there appears
    to be no harm in inserting them.
    Eh? The VoIP call is purely /internal/ to the SPA - it goes nowhere
    near your ITSP.

    I actually use two of these devices to provide outbound & inbound PSTN
    & GSM connectivity for an Asterisk server. Originally, though, they
    were set up in the way the OP is attempting. You can, quite
    categorically, make calls on the PSTN from the handset attached to Line

    A good resource for instructions can be found on the support pages at
    www.provu.com &/or in the Linksys/Sipura forums on the voxilla.com
    Jono, Jul 3, 2007
  8. jkn

    Jono Guest

    jkn brought next idea :
    Oh yes it is ;-)
    Believe me now?

    Erm, what proof did he offer..? I couldn't spot it. I am tired though.
    Username & password too.
    Outbound calls shouldn't affect the harmoneous state of the jkn houshold.
    Perhaps you could change it further to something I can understand...? :)

    Below are some settings {{{{Welcome}}}} was kind enough to post elsewhere:

    These are not specific to the SPA3X0X range ATAs and son't deal directly
    with your PSTN functionality. Some may be useful to you, though.

    Under Regional Tab.

    Dial Tone: [email protected],[email protected];60(*/0/1+2)
    Second Dial Tone: [email protected],[email protected];10(*/0/1+2)
    Outside Dial Tone: [email protected];10(*/0/1)
    Prompt Tone: [email protected],[email protected];10(*/0/1+2)
    Busy Tone: [email protected];30(.375/.375/1)
    Reorder Tone: [email protected];20(*/0/1)
    Off Hook Warning Tone: [email protected],[email protected];10(.125/.125/1+2)
    Ring Back Tone: [email protected],[email protected];*(.4/.2/1+2,.4/2/1+2)
    Confirm Tone: [email protected];1(.25/.25/1)
    SIT1 Tone: [email protected],[email protected],[email protected];20(.330/0/1,.330/0/2,.330/0/3,0/1/0)
    SIT2 Tone: [email protected],[email protected],[email protected];20(.274/0/1,.274/0/2,.380/0/3,0/4/0)
    SIT3 Tone: [email protected],[email protected],[email protected];20(.380/0/1,.380/0/2,.380/0/3,0/4/0)
    SIT4 Tone: [email protected],[email protected],[email protected];20(.380/0/1,.274/0/2,.380/0/3,0/4/0)
    MWI Dial Tone: [email protected],[email protected];10(.75/.75/1+2)
    Cfwd Dial Tone: [email protected],[email protected];2(.2/.2/1+2);10(*/0/1+2)
    DND Dial Tone: [email protected],[email protected];2(.2/.2/2);10(*/0/1+2)
    Holding Tone: [email protected];*(.1/.1/1,.1/.1/1,.1/9.5/1)
    Conference Tone: [email protected];20(.1/.1/1,.1/9.7/1)
    Secure Call Indication Tone: [email protected],[email protected];15(0/2/0,.2/.1/1,.1/2.1/2)
    Feature Invocation Tone: [email protected];*(.1/.1/1)

    Ring1 Cadence: 60(.4/.2,.4/2)
    Ring2 Cadence: 60(1/2)
    Ring3 Cadence: 60(.25/.25,.25/.25,.25/1.75)


    CWT1 Cadence: 30(.2/.2,.2/4.4)

    Ring Waveform: Sinusoid (though Trapezoid may help problematic phones to ring).
    Ring Frequency: 25
    Ring Voltage: 80 (You can use 70, if problems try 75, 80, 85, 90)
    CWT Frequency: [email protected]
    Synchronized Ring: Yes

    Hook Flash Timer Min: .06
    Hook Flash Timer Max: .2
    Callee On Hook Delay: 0
    Reorder Delay: 5
    Call Back Expires: 1800
    Call Back Retry Intvl: 30
    Call Back Delay: .5
    VMWI Refresh Intvl: 0
    Interdigit Long Timer: 10
    Interdigit Short Timer: 3
    CPC Delay: .5
    CPC Duration: .1


    Time Zone: GMT
    FXS Port Impedance: 370+620||310nF
    Daylight Saving Time Rule: start=3/-1/7/2:0:0;end=10/-1/7/2:0:0;save=1:0:0
    DTMF Playback Level: -16
    DTMF Playback Length: .25
    Caller ID Method: ETSI FSK With PR(UK)
    FXS Port Power Limit: 3
    Caller ID FSK Standard: v.23
    Feature Invocation Method: Default
    More Echo Suppression: yes

    In Line 1 (or Line 2)

    Line Enable: yes
    NAT Mapping Enable: yes
    NAT Keep Alive Enable: yes
    Network Jitter Level: low (Depends on how good your broadband is / route to VoIP server)
    Jitter Buffer Adjustment: Up and Down
    Jono, Jul 3, 2007
  9. jkn

    jkn Guest

    Hi Jono

    My apologies - I meant to delete that line after my little test...

    I meant his comment that

    * you can use the PSTN out without any VoIP settings attributed to the
    * PSTN line so I'm failing to see how this can work.

    (IIUC, I would phrase this as 'so i'm failing to see that it works
    this way')

    Well, I'm still puzzled (as is PhilT I think) about your assertion
    that a PSTN call is made by means of an internal (to the SPA3102) VoIP
    call. Can't such a unit 'just' make the equivalent of a passthrough
    connection from the phone in to the line jack?

    Can anyone else comment on this? It does seem strange to me.

    Thanks. I think i've got most of these from prior Googling, but I'll
    go through and check.


    Jon N
    jkn, Jul 3, 2007
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