Configuring Sipura SPA3102

Discussion in 'UK VOIP' started by Chris Davies, Feb 23, 2008.

  1. Chris Davies

    Chris Davies Guest

    So I've taken the plunge and, despite warnings about its dire
    documentation, purchased a (Linksys) Sipura SPA-3102.

    I've got Sipgate working on Line 1, but I can't seem to get my ideal
    configuration. I was rather hoping someone here could help...

    * Inbound PSTN from BT to ring the phone using Ring Cadence #1
    (BT Ring)

    * Inbound VoIP from Sipgate to ring phone using Ring Cadence
    #2 (Call sign)

    * Outbound phone to use VoIP via Voipcheap

    * Outbound phone to use PSTN if prefixed with, say, 121

    * CLID to be presented to phone from PSTN and VoIP

    I don't want any of this VoiP-PSTN gateway stuff where you can enter
    a PIN to get gatewayed from one to the other. Also, I don't want the
    SPA3102 to answer an incoming call unless I've picked up the phone myself.

    Can anyone help? I'll happily offer a beer or two

    Thanks,
    Chris
    Chris Davies, Feb 23, 2008
    #1
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  2. Chris Davies

    Brian A Guest

    On Sat, 23 Feb 2008 12:24:33 +0000, Chris Davies
    <> wrote:

    >So I've taken the plunge and, despite warnings about its dire
    >documentation, purchased a (Linksys) Sipura SPA-3102.
    >
    >I've got Sipgate working on Line 1, but I can't seem to get my ideal
    >configuration. I was rather hoping someone here could help...
    >
    > * Inbound PSTN from BT to ring the phone using Ring Cadence #1
    > (BT Ring)
    >
    > * Inbound VoIP from Sipgate to ring phone using Ring Cadence
    > #2 (Call sign)
    >
    > * Outbound phone to use VoIP via Voipcheap
    >
    > * Outbound phone to use PSTN if prefixed with, say, 121
    >
    > * CLID to be presented to phone from PSTN and VoIP
    >
    >I don't want any of this VoiP-PSTN gateway stuff where you can enter
    >a PIN to get gatewayed from one to the other. Also, I don't want the
    >SPA3102 to answer an incoming call unless I've picked up the phone myself.
    >
    >Can anyone help? I'll happily offer a beer or two
    >
    >Thanks,
    >Chris

    I don't know how much you know already so I will assume not so much.
    I will just provide you here with the basic tools to solve some of the
    problems you might encounter.
    It is useful to know that, in addition to the PDF on the SPA-3102,
    there is a general PDF covering such things as extra in info. on dial
    plans. It was on the Sipura.com site so I assume that it is still
    there.

    Incoming calls can only be received from PSTN and the single, default,
    in/out provider.

    You have 4 gateways, in addition to the in/out bound provider.
    To route calls to any provider you can choose, in your dial plan:-
    1. No stated gateway - calls will be routed via your default
    in/outbound provider.
    2. gw0 - calls routed via PSTN
    3. gw1,gw2,gw3,gw4 - Calls routed via the approriate gateway.
    Include this in any rule as appropriate:-
    <:mad:gwx> where 'x' is the gateway number.

    You can route outgoing calls by number type.
    For example:
    <:00441274>[2-9]xxxxx<:mad:gw1>
    This will route local (Bradford 01274) calls via gw1
    In th eabove local calls are defined as starting with any number from
    2 to 9 inclusive followed by five other numbers. 00441274 is
    automatically inserted. If the number of digits following [2-9] is
    unknown, or flexible, just write 'x.' (without the quotes) to replace
    xxxxx
    Further dial plan info. on my crappy looking web site:-
    www.leafcom.co.uk
    Access only from your home computer - I am sharing space with another
    site that has nothing to do with me.

    I don't use a PSTN so I'll leave that bit to someone else, but I hope
    that the above will help you get started.






    ---
    Remove 'no_spam_' from email address.
    ---
    Brian A, Feb 23, 2008
    #2
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  3. Chris Davies

    Jono Guest

    on 23/02/2008, Chris Davies supposed :
    > So I've taken the plunge and, despite warnings about its dire
    > documentation, purchased a (Linksys) Sipura SPA-3102.
    >
    > I've got Sipgate working on Line 1, but I can't seem to get my ideal
    > configuration. I was rather hoping someone here could help...
    >
    > * Inbound PSTN from BT to ring the phone using Ring Cadence #1
    > (BT Ring)


    Make sure you've UK-ised your SPA.

    whilst logged in as admin/advanced, go to the Voice section & choose PSTN user page. at the bottom, there should be an option to set the default ring for the PSTN - under "PSTN Ring Thru Line 1 Ring Settings"

    > * Inbound VoIP from Sipgate to ring phone using Ring Cadence
    > #2 (Call sign)


    As above, but look on the User 1 page.

    At the bottom, there should be the Default Ring setting.
    >
    > * Outbound phone to use VoIP via Voipcheap
    >
    > * Outbound phone to use PSTN if prefixed with, say, 121


    Both of the above are dealt with using the Dial Plan on the Line 1 page. What dial Plan have you set?

    >
    > * CLID to be presented to phone from PSTN and VoIP


    What have you got set as the CLI type on the Regional page? (try ETSI FSK with PR(UK))

    >
    > I don't want any of this VoiP-PSTN gateway stuff where you can enter
    > a PIN to get gatewayed from one to the other.


    OK.

    > Also, I don't want the
    > SPA3102 to answer an incoming call unless I've picked up the phone myself.


    On the PSTN Line page, set PSTN Ring Thru Line 1 to yes and Off Hook While Calling VoIP no.


    >
    > Can anyone help? I'll happily offer a beer or two


    Below, I've pasted {{{{Welcome}}}}'s regular post on regional settings for the UK

    Admin Login
    Advanced View

    Under Regional Tab.

    Dial Tone: 350@-19,440@-22;60(*/0/1+2)
    Second Dial Tone: 420@-19,520@-19;60(*/0/1+2)
    Outside Dial Tone: 420@-16;60(*/0/1)
    Prompt Tone: 520@-19,620@-19;60(*/0/1+2)
    Busy Tone: 400@-10;30(.375/.375/1)
    Reorder Tone: 400@-20;20(*/0/1)
    Off Hook Warning Tone: 480@-10,620@0;90(.125/.125/1+2)
    Ring Back Tone: 400@-20,450@-20;*(.4/.2/1+2,.4/2/1+2)
    Confirm Tone: 600@-16;1(.25/.25/1)
    SIT1 Tone: 950@-16,1400@-16,1800@-16;20(.330/0/1,.330/0/2,.330/0/3,0/1/0)
    SIT2 Tone: 914@-16,1371@-16,1777@-16;20(.274/0/1,.274/0/2,.380/0/3,0/4/0)
    SIT3 Tone: 914@-16,1371@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,0/4/0)
    SIT4 Tone: 985@-16,1371@-16,1777@-16;20(.380/0/1,.274/0/2,.380/0/3,0/4/0)
    MWI Dial Tone: 350@-19,440@-22;10(.75/.75/1+2)
    Cfwd Dial Tone: 350@-19,440@-19;2(.2/.2/1+2);10(*/0/1+2)
    DND Dial Tone: 350@-19,440@-19;2(.2/.2/2);10(*/0/1+2)
    Holding Tone: 600@-19;*(.1/.1/1,.1/.1/1,.1/9.5/1)
    Conference Tone: 350@-19;20(.1/.1/1,.1/9.7/1)
    Secure Call Indication Tone: 397@-19,507@-19;15(0/2/0,.2/.1/1,.1/2.1/2)
    Feature Invocation Tone: 350@-16;*(.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)

    Ring1 Cadence = The UK Ringing pattern.
    ....

    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: 425@-20
    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 boadband is / route to VoIP server)
    Jitter Buffer Adjustment: Up and Down


    Proxy and Registration
    Proxy: Your Voip Proxy address (for example sip1.sipdiscount.com or sip.voipcheap.com or sipgate.co.uk etc)

    Subscriber Information - (where you enter your VoIP Account details).

    Supplementary Service Subscription - This is where you enable services such as Call Waiting, 3-Way Calling, block anonymous caller etc)

    Audio Configuration - This is where you set-up which codecs you want to use or only use.
    If you have limited broadband speed (mostly if upload is poor) then you probably only use one or more from G729a, G723 or G726)
    If you want the best quality line, then you'd choose G711a (which is the standard used on a standard line)

    (Myself, I use G711a only, have that set to my preferred codec and use that one only, but you may find other codecs are fine).

    DTMF Tx Method: (This will depend on the VoIP service you use, trial and error and depends on codec) Soon know if it don't work when
    trying telephone banking for example and it doesn't recoginse what numbers you are pressing).

    Hook Flash Tx Method: None
    Jono, Feb 23, 2008
    #3
  4. Chris Davies

    Chris Guest

    Jono wrote:

    >Chris Davies wrote


    >> Also, I don't want the
    >> SPA3102 to answer an incoming call unless I've picked up the phone myself.

    >
    >On the PSTN Line page, set PSTN Ring Thru Line 1 to yes and Off Hook While Calling VoIP no.


    Can an SPA-3000 also be configured not to answer PSTN until the VOIP
    call is picked up? That would be really handy. But there seems to be
    no "Off Hook While Calling VOIP" option in the SPA-3000's menu.

    Chris
    Chris, Feb 27, 2008
    #4
  5. Chris Davies

    Jono Guest

    Chris wrote on 27/02/2008 :
    > Jono wrote:
    >
    >> Chris Davies wrote

    >
    >>> Also, I don't want the
    >>> SPA3102 to answer an incoming call unless I've picked up the phone myself.

    >>
    >> On the PSTN Line page, set PSTN Ring Thru Line 1 to yes and Off Hook While
    >> Calling VoIP no.

    >
    > Can an SPA-3000 also be configured not to answer PSTN until the VOIP
    > call is picked up? That would be really handy. But there seems to be
    > no "Off Hook While Calling VOIP" option in the SPA-3000's menu.
    >


    It appears in (presumably) a later firmware revision than yours.

    Just upgrade your firmware to the latest & greatest!

    the latest on the sipura site is
    <http://www.sipura.com/Documents/spa3k-3.1.10d.zip> (version 3.1.10d)

    I'm using the Linksys branded firmware version 3.1.20(GW) which can be
    downloaded from the linksys.com support pages.

    This
    <http://www.linksys.com/servlet/Satellite?c=L_CASupport_C2&childpagename=US%2FLayout&cid=1169083367861&pagename=Linksys%2FCommon%2FVisitorWrapper&lid=6786143006B03>
    seems to be almost the closest I can get to a link to the firmware. (or
    <http://tinyurl.com/3ynf6n> if the long one wraps)

    Even though v 3.1.20(GW) is apparently for a "Linksys" SPA3000, it
    works perfectly well on my "Sipura" SPA3000.

    HTH.
    Jono, Feb 27, 2008
    #5
  6. Chris Davies

    Sinna Guest

    Jono wrote:
    > Chris wrote on 27/02/2008 :
    >> Jono wrote:
    >>
    >>> Chris Davies wrote

    >>
    >>>> Also, I don't want the
    >>>> SPA3102 to answer an incoming call unless I've picked up the phone
    >>>> myself.
    >>>
    >>> On the PSTN Line page, set PSTN Ring Thru Line 1 to yes and Off Hook
    >>> While Calling VoIP no.

    >>
    >> Can an SPA-3000 also be configured not to answer PSTN until the VOIP
    >> call is picked up? That would be really handy. But there seems to be
    >> no "Off Hook While Calling VOIP" option in the SPA-3000's menu.
    >>

    >
    > It appears in (presumably) a later firmware revision than yours.
    >
    > Just upgrade your firmware to the latest & greatest!
    >
    > the latest on the sipura site is
    > <http://www.sipura.com/Documents/spa3k-3.1.10d.zip> (version 3.1.10d)
    >
    > I'm using the Linksys branded firmware version 3.1.20(GW) which can be
    > downloaded from the linksys.com support pages.
    >
    > This
    > <http://www.linksys.com/servlet/Satellite?c=L_CASupport_C2&childpagename=US%2FLayout&cid=1169083367861&pagename=Linksys%2FCommon%2FVisitorWrapper&lid=6786143006B03>
    > seems to be almost the closest I can get to a link to the firmware. (or
    > <http://tinyurl.com/3ynf6n> if the long one wraps)
    >
    > Even though v 3.1.20(GW) is apparently for a "Linksys" SPA3000, it works
    > perfectly well on my "Sipura" SPA3000.
    >
    > HTH.
    >
    >

    Sipura was acquired by Linksys, acquired by Cisco.

    Sinna
    Sinna, Feb 29, 2008
    #6
  7. Chris Davies

    Chris Davies Guest

    Brian A <> wrote:
    > You have 4 gateways, in addition to the in/out bound provider.
    > To route calls to any provider you can choose, in your dial plan:-
    > 1. No stated gateway - calls will be routed via your default
    > in/outbound provider.
    > 2. gw0 - calls routed via PSTN


    These now work well; thank you to you and Jono (and {{{Welcome}})


    > 3. gw1,gw2,gw3,gw4 - Calls routed via the approriate gateway.


    I've hit all sorts of problems with these additional gateways. Reading
    around it seems that what Linksys/Sipura don't tell you is that the
    SPA3102 is broken with respect to these gateway definitions.


    > <:00441274>[2-9]xxxxx<:mad:gw1>
    > This will route local (Bradford 01274) calls via gw1


    Theoretically, yes. In my case I have Sipgate as my primary provider
    and I'm trying to use Voipcheap for outbound dialing via gw1, but it
    doesn't work. (Read on for details.)

    Here's some specifics; perhaps if one of you see that I've dropped a
    clanger they can help me out:

    1. Default provider

    Proxy: sipgate.co.uk
    Outbound proxy: sipgate.co.uk
    Use outbound proxy: Yes
    Use OB Proxy in dialog: Yes
    Register: Yes
    Make call without reg: No
    Answer call without reg: No

    Display name: Chris Davies
    User ID: 12345678 (my Sipgate ID)
    Auth ID: 12345678 (my Sipgate ID)
    Use Auth ID: Yes


    2. Gateway accounts

    Gateway 1: :5060 (previously verified)
    GW1 NAT mapping enabled: Yes/No (doesn't seem to matter)
    GW Auth ID: :5060 (my Voipcheap ID)

    Gateway 2: :5060 (previously verified)
    GW2 NAT mapping enabled: Yes/No (doesn't seem to matter)
    GW Auth ID: :5060 (my Voipcheap ID)


    3. Dial plan

    ( <**1 :> xx. <:mad:gw1> | <**2 :> xx. <:mad:gw2> | <**3 :> xx.
    <::5060;usr=ABCDEF;pwd=******> | xx. )

    I've also found references to "usrid" instead of "usr", but that seems
    to make no difference.


    What seems to happen is that regardless of the authentication details I
    provide - whether in the GW1 and GW2 configuration or in the dialplan
    itself - the authentication details for the primary provider are used.

    Looking at the wireshark network trace, I can see attempts to authenticate
    to voipcheap using my sipgate userid. Obviously voipcheap doesn't want
    to play ball as it doesn't have a clue who this is.

    This stuff's hard enough to get one's head round without buggy software,
    too.

    Chris
    Chris Davies, Feb 29, 2008
    #7
  8. Chris Davies

    Nick Guest

    Chris Davies wrote:

    >
    > 2. Gateway accounts
    >
    > Gateway 1: :5060 (previously verified)
    > GW1 NAT mapping enabled: Yes/No (doesn't seem to matter)
    > GW Auth ID: :5060 (my Voipcheap ID)
    >
    > Gateway 2: :5060 (previously verified)
    > GW2 NAT mapping enabled: Yes/No (doesn't seem to matter)
    > GW Auth ID: :5060 (my Voipcheap ID)
    >


    I would use

    Gateway 1: :5060 (no idea if + or 00 matters,
    but I use +))

    NAT enabled = yes (I have no idea if it matters)

    GW Auth ID: ABCDEF
    Nick, Feb 29, 2008
    #8
  9. Chris Davies

    Chris Davies Guest

    Chris Davies, Feb 29, 2008
    #9
  10. Chris Davies

    Chris Davies Guest

    Re: Configuring Sipura SPA3102 [thanks!]

    Chris Davies <> wrote:
    > Can anyone help? I'll happily offer a beer or two


    I don't have email addresses for all of you, but if you (Brian A, Nick,
    and Jono) would like to email me direct, I'm sure we can work out how
    to get beer (money) to you. Please make sure there's no "-usenet" in
    the email address.

    Many thanks indeed,
    Chris
    Chris Davies, Feb 29, 2008
    #10
  11. Chris Davies

    Brian A Guest

    Re: Configuring Sipura SPA3102 [thanks!]

    On Fri, 29 Feb 2008 17:30:15 +0000, Chris Davies
    <> wrote:

    >Chris Davies <> wrote:
    >> Can anyone help? I'll happily offer a beer or two

    >
    >I don't have email addresses for all of you, but if you (Brian A, Nick,
    >and Jono) would like to email me direct, I'm sure we can work out how
    >to get beer (money) to you. Please make sure there's no "-usenet" in
    >the email address.
    >
    >Many thanks indeed,
    >Chris

    I can really only speak for myself, but I think others will be the
    same, just glad to have been of help and don't expect anything in
    return. I hope that you are sucesssful in getting your SPA to work
    just how you want it to. It is always good to know if the information
    supplied has been useful.
    You will always find help in this group providing you make an effort
    to find readily available information first. Though sometimes, when
    you are starting out, it isn't always obvious how to search for what
    you want.

    ---
    Remove 'no_spam_' from email address.
    ---
    Brian A, Feb 29, 2008
    #11
  12. Chris Davies

    Jono Guest

    Sinna explained on 29/02/2008 :
    > Sipura was acquired by Linksys, acquired by Cisco.
    >


    Erm, OK, thanks.....
    Jono, Feb 29, 2008
    #12
  13. Chris Davies

    Jono Guest

    Re: Configuring Sipura SPA3102 [thanks!]

    Chris Davies formulated on Friday :
    > Chris Davies <> wrote:
    >> Can anyone help? I'll happily offer a beer or two

    >
    > I don't have email addresses for all of you, but if you (Brian A, Nick,
    > and Jono) would like to email me direct, I'm sure we can work out how
    > to get beer (money) to you. Please make sure there's no "-usenet" in
    > the email address.
    >


    No probs. Glad you've got there.

    Not a drinker myself, so no need for any beer tokens. Thanks for the
    offer anyway!
    Jono, Feb 29, 2008
    #13
  14. Chris Davies

    Chris Davies Guest

    Re: Configuring Sipura SPA3102 [thanks!]

    Brian A <> wrote:
    > I can really only speak for myself, but I think others will be the
    > same, just glad to have been of help and don't expect anything in
    > return.


    No worries; that tends to be my approach, too. (A usenet poster since,
    erm, the mid-90s, I think.)

    > I hope that you are sucesssful in getting your SPA to work
    > just how you want it to.


    Yes, I think it's pretty much there, now, thanks.
    Chris
    Chris Davies, Feb 29, 2008
    #14
  15. Chris Davies

    Nick Guest

    Re: Configuring Sipura SPA3102 [thanks!]

    Chris Davies wrote:
    > Chris Davies <> wrote:
    >> Can anyone help? I'll happily offer a beer or two

    >
    > I don't have email addresses for all of you, but if you (Brian A, Nick,
    > and Jono) would like to email me direct, I'm sure we can work out how
    > to get beer (money) to you. Please make sure there's no "-usenet" in
    > the email address.
    >
    > Many thanks indeed,


    Someday, and that day may never come, I'll call upon you to do a service
    for me. But, until that day, accept my help as a gift.
    Nick, Mar 1, 2008
    #15
  16. Chris Davies

    Chris Guest

    Jono wrote:
    >Chris wrote:
    >> Jono wrote:
    >>
    >>> Chris Davies wrote

    >>
    >>>> Also, I don't want the
    >>>> SPA3102 to answer an incoming call unless I've picked up the phone myself.
    >>>
    >>> On the PSTN Line page, set PSTN Ring Thru Line 1 to yes and Off Hook While
    >>> Calling VoIP no.

    >>
    >> Can an SPA-3000 also be configured not to answer PSTN until the VOIP
    >> call is picked up? That would be really handy. But there seems to be
    >> no "Off Hook While Calling VOIP" option in the SPA-3000's menu.
    >>

    >
    >It appears in (presumably) a later firmware revision than yours.
    >
    >Just upgrade your firmware to the latest & greatest!



    Thanks for your suggestion to upgrade the firmware. I did just that,
    and lo and behold the option "Off Hook While Dialling appeared".

    Interestly, now when the SPA3000's PSTN port detects ringing, it rings
    my IP phone at home without going off-hook as I wanted, but it doesn't
    even go off-hook when I answer it! If I leave it to go off-hook whilst
    calling (as it used to), then the call is fine. It just means my PABX
    won't hunt to the next PABX extension if I don't answer the IP phone
    at home. Ho Hum!

    I've also set up IP back-to-back dialling from one to the other, which
    seems to be more reliable than going through a SIP server.

    Thanks for your help,

    Chris
    Chris, Mar 4, 2008
    #16
  17. Chris Davies

    Jono Guest

    Chris presented the following explanation :
    > Interestly, now when the SPA3000's PSTN port detects ringing, it rings
    > my IP phone at home without going off-hook as I wanted, but it doesn't
    > even go off-hook when I answer it! If I leave it to go off-hook whilst
    > calling (as it used to), then the call is fine. It just means my PABX
    > won't hunt to the next PABX extension if I don't answer the IP phone
    > at home. Ho Hum!


    What's your complete configuration? (pbx, phones, etc)
    Jono, Mar 4, 2008
    #17
  18. Chris Davies

    Chris Guest

    Jono wrote:

    >Chris presented the following explanation :
    >> Interestly, now when the SPA3000's PSTN port detects ringing, it rings
    >> my IP phone at home without going off-hook as I wanted, but it doesn't
    >> even go off-hook when I answer it! If I leave it to go off-hook whilst
    >> calling (as it used to), then the call is fine. It just means my PABX
    >> won't hunt to the next PABX extension if I don't answer the IP phone
    >> at home. Ho Hum!

    >
    >What's your complete configuration? (pbx, phones, etc)
    >


    At the office:

    The PABX is a GDK16.

    The SPA3000 is hanging off an analogue extension of the GDK16.
    Preferred Codec G711a
    Use Pref Codec only YES
    Dial Plan 8: (xx.)(S0<:IPaddress:5060>)
    PSTN To VoIP Enable YES
    PSTN Caller Auth Mehtod NONE
    PSTN Ring Thru Line 1 YES
    PSTN Caller Default DP: 8
    Off Hook While Calling VoIP: YES or NO!
    Line 1 Signal Hook Flash to PSTN: DISABLED
    PSTN Answer delay: 0
    PSTN Ring Thru Delay: 1
    PSTN Ring THru CWT Delay: 2
    PSTN Dialling Delay: 0
    PSTN Hook Flash Len: .25
    Detect CPC YES
    Detect Polarity Reversal YES
    Detect PSTN Long Silence NO
    Detect VoIP Long Silence YES
    Mim CPC Duration 0.09
    Detect Disconnect Tone YES
    Disconnect Tone 400@-30,400@-30;2(*/0/1+2)
    FXO Impedance 600 (seems to work best for in-band DTMF tones)
    Ring Validation Time 256mS


    On an incoming PSTN call, firstly Ext 102 (my office) rings for 10
    seconds, then if no answer Ext 106 rings (the SPA), then if no answer
    finally hunts to Ext 104 (the shop itself). Thats why I had hoped to
    leave the SPA on-hook unless or until I picked up my IP Phone at home.

    The SPA3000 connects into a Draytek router with all conceivable ports
    routed through to it.

    At home:

    Simply a Grandstream BT-101. Again all ports routed to it ok.


    Jono, unless its obvious to you from the above, thanks but please
    don't worry - I am able to remotely enable the PSTN to VOIP gateway of
    the SPA from home when I'm working at home, and turn it off again
    remotely when I've had enough for the day! I suspect an SPA3000 at
    each end would be far more compatible and the best solution. And I
    don't wanna take up too much bandwidth of this group.

    Cheers, Chris
    Chris, Mar 5, 2008
    #18
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