Re: voipcheap.co.uk rates

Discussion in 'UK VOIP' started by Graham., Jun 15, 2011.

  1. Graham.

    Graham. Guest

    "www.GymRatZ.co.uk" <> wrote in message news:itamca$bg4$...
    > Just checking the ever fluctuating betamax prices and I see
    > voipcheap.co.uk is right down at the cheapest UK mobile rates (at the
    > moment).
    > I can't see on the site anywhere but is there a connection charge which
    > might skew things back?
    >
    > Cheers
    > Pete
    > --
    > http://www.GymRatZ.co.uk - Fitness+Gym Equipment.
    > http://www.bodysolid-gym-equipment.co.uk
    > http://www.trade-price-supplements.co.uk
    > http://www.water-rower.co.uk


    Yes I noticed that last week and have already decided to revive my old voipcheap.co.uk
    account when my smartvoip credit runs out. I have about 7 euros left, and with my usage
    it may take some weeks.
    Just in case anyome does not know there are two betamax clones with the name voipcheap,
    and it's the .co.uk one we are talking about, not the .com one.

    Also, if you attempt to buy credit using the same Paypal account that you have associated with
    an existing clone, the payment will bounce back into your account
    Short of setting up a fresh PP account, and you won't be able to link the same card to it,
    you can delete the existing Betamax account, the feature is well hidden on the website:

    other options / customer service / account / delete my account

    (BTDTGTTS)

    --
    Graham.

    %Profound_observation%
    Graham., Jun 15, 2011
    #1
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  2. I want to send a string of digits after establishing a VoipCheap connection.
    Is there any alternative to pecking out each of the digits on-by-one into the
    screen keypad of the application, such as say chain-dialing multiple contacts
    or adding a pause plus the following group into just the one contact, etc?

    Thanks.
    Anthony R. Gold, Jun 15, 2011
    #2
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  3. Graham.

    Graham. Guest

    "Anthony R. Gold" <> wrote in message news:...
    >I want to send a string of digits after establishing a VoipCheap connection.
    > Is there any alternative to pecking out each of the digits on-by-one into the
    > screen keypad of the application, such as say chain-dialing multiple contacts
    > or adding a pause plus the following group into just the one contact, etc?
    >
    > Thanks.


    Like most people here I use a hardware phone to make calls via Betamax so can use
    the phone's memory.
    I tried to find pause functionality in one of the Betamax softphones but could not find any.
    However...
    There is a generic free softphone available called X-Lite that will process pauses and post-dial strings
    http://www.counterpath.com/x-lite.html

    Put your account details in and store the number followed by one or more "P"s for each half second pause,
    followed by the post-dial string with a semicolon at the end.
    https://support.counterpath.com/default.asp?W366



    --
    Graham.

    %Profound_observation%
    Graham., Jun 16, 2011
    #3
  4. Graham.

    Graham. Guest

    >> Put your account details in and store the number followed by one or more "P"s for each half second pause,
    >> followed by the post-dial string with a semicolon at the end.
    >> https://support.counterpath.com/default.asp?W366

    >
    > That works a treat. Setting up a remote call diversion under BT's Smart
    > Divert requires the accurate entry of 33 digits after the intermediate dial
    > tone and I was finding that tricky using point and click. Many thanks.


    Now all you need is a mouse macro recorder and a timed windows task to automate the entire
    affair and the world's your chopped+fried :)

    --
    Graham.

    %Profound_observation%
    Graham., Jun 16, 2011
    #4
  5. On Thu, 16 Jun 2011 01:59:30 +0100, "Graham." <> wrote:

    >>> Put your account details in and store the number followed by one or more "P"s for each half second pause,
    >>> followed by the post-dial string with a semicolon at the end.
    >>> https://support.counterpath.com/default.asp?W366

    >>
    >> That works a treat. Setting up a remote call diversion under BT's Smart
    >> Divert requires the accurate entry of 33 digits after the intermediate dial
    >> tone and I was finding that tricky using point and click. Many thanks.

    >
    > Now all you need is a mouse macro recorder and a timed windows task to automate the entire
    > affair and the world's your chopped+fried :)


    BTW how did you discover pause dialing with the semicolon? I see in the
    support forum where the company explicitly denies such a capability exists:
    http://forums.counterpath.com/viewtopic.php?f=26&t=18265&p=66943&hilit=pause#p66943
    Anthony R. Gold, Jun 16, 2011
    #5
  6. Graham.

    Graham. Guest

    "Anthony R. Gold" <> wrote in message news:...
    > On Thu, 16 Jun 2011 01:59:30 +0100, "Graham." <> wrote:
    >
    >>>> Put your account details in and store the number followed by one or more "P"s for each half second pause,
    >>>> followed by the post-dial string with a semicolon at the end.
    >>>> https://support.counterpath.com/default.asp?W366
    >>>
    >>> That works a treat. Setting up a remote call diversion under BT's Smart
    >>> Divert requires the accurate entry of 33 digits after the intermediate dial
    >>> tone and I was finding that tricky using point and click. Many thanks.

    >>
    >> Now all you need is a mouse macro recorder and a timed windows task to automate the entire
    >> affair and the world's your chopped+fried :)

    >
    > BTW how did you discover pause dialing with the semicolon? I see in the
    > support forum where the company explicitly denies such a capability exists:
    > http://forums.counterpath.com/viewtopic.php?f=26&t=18265&p=66943&hilit=pause#p66943



    It was in the FAQs rather than the forum.
    Doing a little forensics with my browser history I appear to have been here
    https://support.counterpath.com/

    clicked FAQ

    https://support.counterpath.com/default.asp?W12

    then put "pause" in the search box

    https://support.counterpath.com/default.asp?ixWiki=3&pg=pgSearchWiki&qWiki=pause

    Then I selected the top item

    https://support.counterpath.com/default.asp?W366

    Looks like the forum Admin's primary function is to encourage take-up of the chargeable product, so
    no surprises there.

    --
    Graham.

    %Profound_observation%
    Graham., Jun 16, 2011
    #6
  7. On Thu, 16 Jun 2011 00:26:57 +0100, "Graham." <> wrote:

    > Like most people here I use a hardware phone to make calls via Betamax so can use
    > the phone's memory.


    A hardware solution may be attractive but I've browsed through adverts and
    find a couple of potential gotcha's for the pig-ignorant, like me. One is
    devices that mention they only handle usernames constructed from a character
    set of Numbers 0-9 plus Letters A-D, so I guess they can not cope with most
    VOIPCheap logins. The other is devices that claim to be unlocked and/or
    jailbroken, the mere mention of which makes one sensitive to the risk that
    other devices out there in the Ebay marketplace are service-provider locked.

    Are both of those risks real or does "username" mean something different?

    A dream hardware solution would be a cordless WiFi phone with built-in SIP
    capabilities, so one could bring one's phone and its number from hotspot to
    hotspot. Does such a device exist? Does such a device exist with the added
    capability of simultaneously being logged-in to multiple SIP services?

    As a starter I may toy with a Linksys SPA3102, which is both cabled and needs
    an analogue phone, but VOIPCheap does list it as being compatible.

    Thanks.
    Anthony R. Gold, Jun 19, 2011
    #7
  8. Anthony R. Gold wrote:

    >A dream hardware solution would be a cordless WiFi phone with built-in SIP
    >capabilities, so one could bring one's phone and its number from hotspot to
    >hotspot. Does such a device exist? Does such a device exist with the added
    >capability of simultaneously being logged-in to multiple SIP services?


    Yes and sort-of-no, if you're a cheaparse like me and buy a UTStarCom
    GF-210 from sipgate for 40 quid.
    (http://www.sipgate.co.uk/voipshop/utstarcom/gf-210) On its own, it'll
    hop on any unsecured 802.11 network (or up to four with credentials set
    up in the phone) and log in to a single SIP service. If you have a
    proper computer live 24/7, you can run a SIP gateway on it and mux/demux
    from there, which IMO is much easier than doing it at the device end. (I
    use Asterisk, but if that's all you want I'm sure I've seen a package
    that just does SIP multiplexing.)

    So from anywhere I can get network access, I can talk via my home phone
    line, external VoIP accounts, and my private VoIP network. (And my home
    phone rings the mobile when I'm in the pub, at no extra cost. Everybody
    wins.)

    Roger
    Roger Burton West, Jun 19, 2011
    #8
  9. In article <>,
    Anthony R. Gold <> wrote:

    >A dream hardware solution would be a cordless WiFi phone with built-in SIP
    >capabilities, so one could bring one's phone and its number from hotspot to
    >hotspot. Does such a device exist? Does such a device exist with the added
    >capability of simultaneously being logged-in to multiple SIP services?


    There are many Wi-Fi capable SIP devices. E.g. most moden "smartphones"
    can have soft-phone applications running on them. Battery life is terrible
    though (at least that's my expeirence with N900 and HTC desire when using
    Wi-Fi)

    Have a look here for some pure Wi-Fi handsets:

    http://www.voipon.co.uk/unidata-wireless-ip-phones-c-148_213.html

    and there are many others.

    However, I have to say, SIP over Wi-Fi is rubbish when you don't have
    good control over the Wi-Fi. A public hotspot might well block VoIP too
    - and even if it doesn't, you're battling with everyone else using that
    hotspot and just one PC uploading/downloading will interfere with VoIP.

    (Then there's the issue of logging on to the hotspot in the first place -
    sometimes it's just not possible without a web browser!)

    However if it's for cordless use in the home, then there are several
    DECT based phones which are SIP compliant. E.g. the Gigaset IP range
    are all very good. (I use them myself)

    To use with a betamax service, you'll need to extract the SIP credentials
    from the system to program into your cordless device.

    >As a starter I may toy with a Linksys SPA3102, which is both cabled and needs
    >an analogue phone, but VOIPCheap does list it as being compatible.


    As long as you get the SIP details then you have the choice of 1000's of
    devices, wired and wireless.

    Gordon
    Gordon Henderson, Jun 19, 2011
    #9
  10. Graham.

    Graham. Guest


    > To use with a betamax service, you'll need to extract the SIP credentials
    > from the system to program into your cordless device.


    I wouldn't want anyone to interpret the above as if Betamax discourage the BYOD
    model, they don't
    http://www.voipcheap.com/en/sipp.html


    --
    Graham.

    %Profound_observation%
    Graham., Jun 19, 2011
    #10
  11. Graham.

    alexd Guest

    Meanwhile, at the uk.telecom.voip Job Justification Hearings, Anthony R.
    Gold chose the tried and tested strategy of:

    > A dream hardware solution would be a cordless WiFi phone with built-in SIP
    > capabilities,


    Wifi isn't great for mobile devices because of power consumption [although
    I'm sure this will have got better over the years], and it's not
    particularly good for voice either, because of latency and QoS. It's a nice
    idea in theory to be able to drift from hotspot to hotspot using one's SIP
    client, but unless you manage them yourself [home, work and mum's place, for
    example] you may not get SIP working at all.

    --
    <http://ale.cx/> (AIM:troffasky) ()
    20:41:22 up 4 days, 3:15, 5 users, load average: 0.02, 0.06, 0.13
    "People believe any quote they read on the internet
    if it fits their preconceived notions." - Martin Luther King
    alexd, Jun 23, 2011
    #11
  12. On Thu, 23 Jun 2011 21:04:55 +0100, alexd <> wrote:

    > Meanwhile, at the uk.telecom.voip Job Justification Hearings, Anthony R.
    > Gold chose the tried and tested strategy of:
    >
    >> A dream hardware solution would be a cordless WiFi phone with built-in SIP
    >> capabilities,

    >
    > Wifi isn't great for mobile devices because of power consumption [although
    > I'm sure this will have got better over the years], and it's not
    > particularly good for voice either, because of latency and QoS. It's a nice
    > idea in theory to be able to drift from hotspot to hotspot using one's SIP
    > client, but unless you manage them yourself [home, work and mum's place, for
    > example] you may not get SIP working at all.


    Thanks for that. I discovered that I already owned my "dream hardware" which
    was my Nokia E72 cell phone onto which I installed a softphone ap. And the
    call quality via WiFi was indeed very bad - far worse that when using the
    same ap over the Orange G3 packet network. I was surprised, based only on
    difference between the underlying bandwidths of those two transports.

    I've now bought a Linksys/Cisco SPA-3102 as a SIP ATA because VOIPCheap
    listed it as compatible. The choice may be a mistake because the device
    contains a router which I did not need. Apart from the router the gadget has
    a phone socket for a POTS PSTN line, a socket for a handset and an Ethernet
    socket for the Internet. The way it works is that incoming SIP calls come to
    the handset and incoming PSTN calls come to the handset. The handset calls
    out via VOIP and I believe it will fail-over to the PSTN line. Does anyone
    know if there is a way to force an outgoing call out over the PSTN? If that
    is in the manual I did not find it but the management screen has a display
    for "Last Called PSTN Number:" which is now blank.

    [I ask after waiting 33 minutes in an 0800 music-on-hold queue which waiting
    in a Tesco Bank CS. I presume I was being charged by VOIPCheap for the 0800
    call by the minute when that same call over the PSTN would have been free.]

    Thanks.
    Anthony R. Gold, Jun 23, 2011
    #12
  13. Graham.

    Bob Eager Guest

    On Thu, 23 Jun 2011 22:41:26 +0100, Anthony R. Gold wrote:


    > I've now bought a Linksys/Cisco SPA-3102 as a SIP ATA because VOIPCheap
    > listed it as compatible. The choice may be a mistake because the device
    > contains a router which I did not need. Apart from the router the gadget
    > has a phone socket for a POTS PSTN line, a socket for a handset and an
    > Ethernet socket for the Internet. The way it works is that incoming SIP
    > calls come to the handset and incoming PSTN calls come to the handset.
    > The handset calls out via VOIP and I believe it will fail-over to the
    > PSTN line. Does anyone know if there is a way to force an outgoing call
    > out over the PSTN? If that is in the manual I did not find it but the
    > management screen has a display for "Last Called PSTN Number:" which is
    > now blank.


    It's in the Provisioning Guide, and may be in the user manual. You do it
    in the dial plan. Essentially, the FXO port is gw0, so you add that as
    part of the dial string:

    <9,:>xx.<:gw0>

    which means that if a number is preceded by 9, force it out the FXO port.



    --
    Use the BIG mirror service in the UK:
    http://www.mirrorservice.org

    *lightning protection* - a w_tom conductor
    Bob Eager, Jun 23, 2011
    #13
  14. On 23 Jun 2011 22:42:51 GMT, Bob Eager <> wrote:

    > On Thu, 23 Jun 2011 22:41:26 +0100, Anthony R. Gold wrote:
    >
    >
    >> I've now bought a Linksys/Cisco SPA-3102 as a SIP ATA because VOIPCheap
    >> listed it as compatible. The choice may be a mistake because the device
    >> contains a router which I did not need. Apart from the router the gadget
    >> has a phone socket for a POTS PSTN line, a socket for a handset and an
    >> Ethernet socket for the Internet. The way it works is that incoming SIP
    >> calls come to the handset and incoming PSTN calls come to the handset.
    >> The handset calls out via VOIP and I believe it will fail-over to the
    >> PSTN line. Does anyone know if there is a way to force an outgoing call
    >> out over the PSTN? If that is in the manual I did not find it but the
    >> management screen has a display for "Last Called PSTN Number:" which is
    >> now blank.

    >
    > It's in the Provisioning Guide, and may be in the user manual. You do it
    > in the dial plan. Essentially, the FXO port is gw0, so you add that as
    > part of the dial string:
    >
    > <9,:>xx.<:gw0>
    >
    > which means that if a number is preceded by 9, force it out the FXO port.


    Thanks, a leading 9 could work fine for me if I could figure where and how to
    install that dial plan. I am impressed by the quantity, size and
    incomprehensibility of the vast array of tabbed menus.

    In the Line 1 menu I found this existing Dial Plan:
    (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)

    And in the PSTN Line menu I see a list of 8 numbered Dial Plans which each
    consists of just this: (xx.)

    Is one of those places appropriate to insert the <9,:>xx.<:gw0> string? And
    if so, what would be the entire resulting string, including anything that
    must remain from before?
    Anthony R. Gold, Jun 24, 2011
    #14
  15. On Fri, 24 Jun 2011 08:18:41 +0100, "Anthony R. Gold"
    <> wrote:

    > In the Line 1 menu I found this existing Dial Plan:
    > (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)
    >
    > And in the PSTN Line menu I see a list of 8 numbered Dial Plans which each
    > consists of just this: (xx.)
    >
    > Is one of those places appropriate to insert the <9,:>xx.<:gw0> string? And
    > if so, what would be the entire resulting string, including anything that
    > must remain from before?


    I fiddled blindly and finally found this works (using #9 for as access code):
    (<#9:>xx.<:mad:gw0>|*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)

    Can anyone can improve on that to reduce the access code to a mere "9"?

    My "blind" fiddling was helped by this offline Dial Plan simulator that I
    stumbled upon:
    http://www.softpedia.com/get/Internet/Telephony-SMS-GSM/Sipura-3000-Dial-Plan-Manager.shtml
    Anthony R. Gold, Jun 24, 2011
    #15
  16. Graham.

    Bob Eager Guest

    On Fri, 24 Jun 2011 13:06:22 +0100, Anthony R. Gold wrote:

    > On Fri, 24 Jun 2011 08:18:41 +0100, "Anthony R. Gold"
    > <> wrote:
    >
    >> In the Line 1 menu I found this existing Dial Plan:
    >> (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)
    >>
    >> And in the PSTN Line menu I see a list of 8 numbered Dial Plans which
    >> each consists of just this: (xx.)
    >>
    >> Is one of those places appropriate to insert the <9,:>xx.<:gw0> string?
    >> And if so, what would be the entire resulting string, including
    >> anything that must remain from before?

    >
    > I fiddled blindly and finally found this works (using #9 for as access
    > code):
    > (<#9:>xx.<:mad:gw0>|*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|

    xxxxxxxxxxxx.)
    >
    > Can anyone can improve on that to reduce the access code to a mere "9"?
    >
    > My "blind" fiddling was helped by this offline Dial Plan simulator that
    > I stumbled upon:
    > http://www.softpedia.com/get/Internet/Telephony-SMS-GSM/Sipura-3000-

    Dial-Plan-Manager.shtml

    Do you not have the user manual? I can let you have acopy.

    Anyway, it's pretty simple, but the default setting is geared to North
    American numbers. As you may have gathered, the vertical bars separate
    acceptable number patterns. X means any digit. Digits (and * or #) can
    stand for themselves, or a set of options can be in square brackets (0-7,
    for example, or 3456). Dot means 0 or more repetitions of the preceding
    character. <a:b> means that if the user dials a sequence a, it's replaced
    by b. Comma plays an 'outside' dial tone. ! bars anything matching the
    preceding sequence (e.g. you might have 09X.!). @gwn means use gateway n.
    You needed #9 because there is a later sequence that clashes, also
    starting with 9.

    A simplified version might be:

    (<9:>xx.<:mad:gw0>|0[123578]X.|00X.|09X.!)

    You could try adding your local code (assuming 6 digit numbers, let's say
    it's 01227):

    <:01227>XXXXXX

    to save dialling that code for local numbers. I haven't tried this
    dialplan, as mine is built to go through an Asterisk box; all mine does
    is ensure that valid strings get through without timeouts.

    If you get dialling delays, tweak the inter-digit timeout. There's a long
    one and a short one. In essence, it hangs on to digits until a timeout
    expires, and the long one is 10 seconds by default.

    --
    Use the BIG mirror service in the UK:
    http://www.mirrorservice.org

    *lightning protection* - a w_tom conductor
    Bob Eager, Jun 24, 2011
    #16
  17. On 24 Jun 2011 13:16:08 GMT, Bob Eager <> wrote:

    > On Fri, 24 Jun 2011 13:06:22 +0100, Anthony R. Gold wrote:
    >
    >> On Fri, 24 Jun 2011 08:18:41 +0100, "Anthony R. Gold"
    >> <> wrote:
    >>
    >>> In the Line 1 menu I found this existing Dial Plan:
    >>> (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)
    >>>
    >>> And in the PSTN Line menu I see a list of 8 numbered Dial Plans which
    >>> each consists of just this: (xx.)
    >>>
    >>> Is one of those places appropriate to insert the <9,:>xx.<:gw0> string?
    >>> And if so, what would be the entire resulting string, including
    >>> anything that must remain from before?

    >>
    >> I fiddled blindly and finally found this works (using #9 for as access
    >> code):
    >> (<#9:>xx.<:mad:gw0>|*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|

    > xxxxxxxxxxxx.)
    >>
    >> Can anyone can improve on that to reduce the access code to a mere "9"?
    >>
    >> My "blind" fiddling was helped by this offline Dial Plan simulator that
    >> I stumbled upon:
    >> http://www.softpedia.com/get/Internet/Telephony-SMS-GSM/Sipura-3000-

    > Dial-Plan-Manager.shtml
    >
    > Do you not have the user manual? I can let you have acopy.
    >
    > Anyway, it's pretty simple, but the default setting is geared to North
    > American numbers. As you may have gathered, the vertical bars separate
    > acceptable number patterns. X means any digit. Digits (and * or #) can
    > stand for themselves, or a set of options can be in square brackets (0-7,
    > for example, or 3456). Dot means 0 or more repetitions of the preceding
    > character. <a:b> means that if the user dials a sequence a, it's replaced
    > by b. Comma plays an 'outside' dial tone. ! bars anything matching the
    > preceding sequence (e.g. you might have 09X.!). @gwn means use gateway n.
    > You needed #9 because there is a later sequence that clashes, also
    > starting with 9.
    >
    > A simplified version might be:
    >
    > (<9:>xx.<:mad:gw0>|0[123578]X.|00X.|09X.!)
    >
    > You could try adding your local code (assuming 6 digit numbers, let's say
    > it's 01227):
    >
    > <:01227>XXXXXX
    >
    > to save dialling that code for local numbers. I haven't tried this
    > dialplan, as mine is built to go through an Asterisk box; all mine does
    > is ensure that valid strings get through without timeouts.
    >
    > If you get dialling delays, tweak the inter-digit timeout. There's a long
    > one and a short one. In essence, it hangs on to digits until a timeout
    > expires, and the long one is 10 seconds by default.


    Thanks for all, and especially for mentioning the clash. My skimpy reading
    had misled me to believe that by placing the rule in the front, a match on 9
    would trigger the action of the rule such that later rules would be skipped.
    Anthony R. Gold, Jun 24, 2011
    #17
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