Gradwell Flexor/Camrivox 151

Discussion in 'UK VOIP' started by JC, Mar 24, 2007.

  1. JC

    JC Guest

    Ok, I know these were mentioned recently.... so here goes....

    Has anyone had any joy making one of these pieces of cr*p usable? Ours
    came free with a Gradwell account (well not free, we had to pay P+P
    had no option to refuse it) and definitely isn't worth what we paid
    for it. Luckily I'd already ordered a Linksys.

    I was going to stick it on ebay but thought I'd check it out first and
    it's lucky I did as it was supplied pre-configured on our account
    (this wasn't mentioned on the supplied setup sheet)! Nothing major,
    after Googling for non-supplied manual to find out you have to type ##
    to get it's IP address, I did a factory reset - but our account info's
    still there and there appears to be no way to change it even to
    another provider!

    So, do I need to destroy this thing to protect our account or is there
    some way of making it usable with Asterisk?

    Rgds
    Jonathan
     
    JC, Mar 24, 2007
    #1
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  2. JC

    Brian Guest

    On 2007-03-24, JC <> wrote:

    > Ok, I know these were mentioned recently.... so here goes....
    >
    > Has anyone had any joy making one of these pieces of cr*p usable? Ours
    > came free with a Gradwell account (well not free, we had to pay P+P
    > had no option to refuse it) and definitely isn't worth what we paid
    > for it. Luckily I'd already ordered a Linksys.


    What's the problem with it? Does it not work with Gradwell?

    > I was going to stick it on ebay but thought I'd check it out first and
    > it's lucky I did as it was supplied pre-configured on our account
    > (this wasn't mentioned on the supplied setup sheet)! Nothing major,
    > after Googling for non-supplied manual to find out you have to type ##
    > to get it's IP address, I did a factory reset - but our account info's
    > still there and there appears to be no way to change it even to
    > another provider!


    Supplying preconfigured ATAs and phones is mentioned in their FAQ page
    but an interesting point is whether a preconfigured device implies a
    locked device. Details on the state of the Flexor do not appear readily
    visible on the website but having one come 'free' with an account should
    raise the question of its use with other providers. On the other hand,
    it wouldn't be a bad idea to give information on whether it is tied to a
    single account.

    > So, do I need to destroy this thing to protect our account or is there
    > some way of making it usable with Asterisk?


    If the Flexor is locked to Gradwell it cannot register with Asterisk,
    surely? Have you asked them about removing your account details from it?
    They might do a deal with you.

    Brian.
     
    Brian, Mar 26, 2007
    #2
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  3. JC

    JC Guest

    On Mon, 26 Mar 2007 13:11:22 +0000 (UTC), Brian <>
    wrote:

    >What's the problem with it? Does it not work with Gradwell?


    It seems to work but it's certainly unsuitable for our purposes - no
    way to setup a dialplan, custom tones, PSTN fallback etc etc. Also the
    quality's definitely not up to the Linksys (and even these are a bit
    iffy on the FXO side).

    >Supplying preconfigured ATAs and phones is mentioned in their FAQ page
    >but an interesting point is whether a preconfigured device implies a
    >locked device. Details on the state of the Flexor do not appear readily


    Well normally I'd commend any company sending pre-configured hardware
    but you obviously want to remove your account details if it's not
    needed. There's definitely no mention of it being locked. I've now
    contacted Gradwell and they've offered to refund the postage charge if
    it's returned, reconfigure it for another Gradwell account, or unlock
    it for a fee of £15 +vat!

    Since I've already signed up for another Gradwell SIP inbound account
    with plans to setup an Asterisk box, moving it to another account
    isn't an option and it's definitely not worth the £ 20+ it'll cost
    with unlocking and postage. Since it'll cost me time an effort to
    return I guess it'll have to be destroyed.

    All in all it's left quite a bitter taste about Gradwell which is a
    shame as their VoIP service seems technically OK. If they'd been more
    open (honest?) about what they're providing things would be very
    different. As it is I doubt we'll be opening any more accounts with
    them.

    >visible on the website but having one come 'free' with an account should
    >raise the question of its use with other providers. On the other hand,
    >it wouldn't be a bad idea to give information on whether it is tied to a
    >single account.


    Their website's a bit of a mess but not the worst I've seen. Certainly
    clarity on VAT and contract length is needed.

    >If the Flexor is locked to Gradwell it cannot register with Asterisk,
    >surely? Have you asked them about removing your account details from it?
    >They might do a deal with you.


    For those interested, a bit of research shows it appears to download
    via HTTPS it's config at power on. It's not worth my time to
    investigate further as it really is a very low quality and poorly made
    device.

    Rgds
    Jonathan
     
    JC, Mar 26, 2007
    #3
  4. JC

    Tim Guest

    JC wrote:
    > On Mon, 26 Mar 2007 13:11:22 +0000 (UTC), Brian <>
    > wrote:
    >
    >> What's the problem with it? Does it not work with Gradwell?

    >
    > It seems to work but it's certainly unsuitable for our purposes - no
    > way to setup a dialplan, custom tones, PSTN fallback etc etc. Also the
    > quality's definitely not up to the Linksys (and even these are a bit
    > iffy on the FXO side).


    Yes, but you got it for free. In return for free, it was locked.

    Do you expect gradwell to give stuff away?

    Tim
     
    Tim, Mar 26, 2007
    #4
  5. JC

    Desk Rabbit Guest

    JC wrote:
    > Since I've already signed up for another Gradwell SIP inbound account
    > with plans to setup an Asterisk box,


    OOI Why SIP and not IAX?

    > Their website's a bit of a mess but not the worst I've seen. Certainly
    > clarity on VAT and contract length is needed.


    Indeed, that has put me off them in the past.
     
    Desk Rabbit, Mar 26, 2007
    #5
  6. Desk Rabbit wrote:
    > JC wrote:
    >> Since I've already signed up for another Gradwell SIP inbound account
    >> with plans to setup an Asterisk box,

    >
    > OOI Why SIP and not IAX?


    SIP is generally more reliable, as it avoids using asterisk to do a
    protocol conversion (SIP->IAX) and back again.

    >> Their website's a bit of a mess but not the worst I've seen. Certainly
    >> clarity on VAT and contract length is needed.

    >
    > Indeed, that has put me off them in the past.


    I guess it shouldn't be too hard for us to mention that we usually only
    have a one month contract term, and prices quoted usually exclude VAT.

    cheers
    peter
     
    Peter Gradwell, Mar 26, 2007
    #6
  7. hi

    JC wrote:
    > It seems to work but it's certainly unsuitable for our purposes - no
    > way to setup a dialplan, custom tones, PSTN fallback etc etc. Also the
    > quality's definitely not up to the Linksys (and even these are a bit
    > iffy on the FXO side).


    Our view is that the camrivox box is easier to use and has better
    software than the linksys. It represents a nice, easy to use, voip adaptor.

    PSTN fall back is configured automatically I think.

    > Well normally I'd commend any company sending pre-configured hardware
    > but you obviously want to remove your account details if it's not
    > needed. There's definitely no mention of it being locked. I've now
    > contacted Gradwell and they've offered to refund the postage charge if
    > it's returned, reconfigure it for another Gradwell account, or unlock
    > it for a fee of £15 +vat!


    that represents a pretty good bargain on a device which retails for
    GBP45 (i.e. cost of 8.50 + 15). You can't buy a linksys adaptor for any
    where near that.

    > Since I've already signed up for another Gradwell SIP inbound account
    > with plans to setup an Asterisk box, moving it to another account
    > isn't an option


    We'd be happy to move it to another Gradwell account of course.

    cheers
    peter
     
    Peter Gradwell, Mar 26, 2007
    #7
  8. JC

    Desk Rabbit Guest

    Peter Gradwell wrote:
    > Desk Rabbit wrote:
    >> JC wrote:
    >>> Since I've already signed up for another Gradwell SIP inbound account
    >>> with plans to setup an Asterisk box,

    >>
    >> OOI Why SIP and not IAX?

    >
    > SIP is generally more reliable, as it avoids using asterisk to do a
    > protocol conversion (SIP->IAX) and back again.


    How is that more reliable? If the box is properly specced it shouldn't
    cause any problems. From what I've read and experienced, SIP is good on
    the LAN and IAX is good on the WAN. Now if we were talking about codec
    conversion I'd be agreeing with you.

    IAX is NAT friendly and SIP is not so that makes setup easier and more
    reliable.
    IAX uses less bandwidth so thats another plus for reliablity.
    IAX requires only one port to be opened so that a plus for security and
    reliability.

    Also worth reading Mark Spencer's article
    http://www.voip-info.org/wiki/view/IAX versus SIP

    >
    >>> Their website's a bit of a mess but not the worst I've seen. Certainly
    >>> clarity on VAT and contract length is needed.

    >>
    >> Indeed, that has put me off them in the past.

    >
    > I guess it shouldn't be too hard for us to mention that we usually only
    > have a one month contract term, and prices quoted usually exclude VAT.


    The rules on quoting VAT prices depend on who you are quoting. If you
    are quoting Joe Public, you MUST quote the VAT inclusive price. As you
    have no control over who is viewing you web site you should display both.

    http://www.consumereducation.org.uk/money/english/prices/02.htm
     
    Desk Rabbit, Mar 26, 2007
    #8
  9. Desk Rabbit wrote:
    > Peter Gradwell wrote:
    >> Desk Rabbit wrote:
    >>> JC wrote:
    >>>> Since I've already signed up for another Gradwell SIP inbound account
    >>>> with plans to setup an Asterisk box,
    >>>
    >>> OOI Why SIP and not IAX?

    >>
    >> SIP is generally more reliable, as it avoids using asterisk to do a
    >> protocol conversion (SIP->IAX) and back again.

    >
    > How is that more reliable? If the box is properly specced it shouldn't
    > cause any problems. From what I've read and experienced, SIP is good on
    > the LAN and IAX is good on the WAN. Now if we were talking about codec
    > conversion I'd be agreeing with you.


    My view is that all calls come in/go out from the PSTN over SIP - you
    can't buy a pstn switch that supports IAX.

    Further, all asterisk boxes generally speak to their phone handsets
    using SIP.

    So, why would you want to route a phone call through a point of failure
    @ our end and change the signalling method, potentially causing a loss
    of information, only to change it back.

    A call that goes cisco -> asterisk -> asterisk -> phone is never going
    to be as good as one that goes cisco -> asterisk -> phone.

    cheers
    peter
     
    Peter Gradwell, Mar 27, 2007
    #9
  10. JC

    Desk Rabbit Guest

    Peter Gradwell wrote:
    > Desk Rabbit wrote:
    >> Peter Gradwell wrote:
    >>> Desk Rabbit wrote:
    >>>> JC wrote:
    >>>>> Since I've already signed up for another Gradwell SIP inbound account
    >>>>> with plans to setup an Asterisk box,
    >>>>
    >>>> OOI Why SIP and not IAX?
    >>>
    >>> SIP is generally more reliable, as it avoids using asterisk to do a
    >>> protocol conversion (SIP->IAX) and back again.

    >>
    >> How is that more reliable? If the box is properly specced it shouldn't
    >> cause any problems. From what I've read and experienced, SIP is good
    >> on the LAN and IAX is good on the WAN. Now if we were talking about
    >> codec conversion I'd be agreeing with you.

    >
    > My view is that all calls come in/go out from the PSTN over SIP - you
    > can't buy a pstn switch that supports IAX.
    >
    > Further, all asterisk boxes generally speak to their phone handsets
    > using SIP.
    >
    > So, why would you want to route a phone call through a point of failure
    > @ our end and change the signalling method, potentially causing a loss
    > of information, only to change it back.
    >
    > A call that goes cisco -> asterisk -> asterisk -> phone is never going
    > to be as good as one that goes cisco -> asterisk -> phone.


    We will have to agree to disagree as the majority of my international
    calls go sip -> asterisk -> IAX -> VOIP_provider_system -> PSTN and vice
    versa without any problems whatsoever. When I'm working from home they go:

    sip -> asterisk -> IAX(Over vpn) -> asterisk -> IAX ->
    VOIP_provider_system -> PSTN


    <shrug>
     
    Desk Rabbit, Mar 27, 2007
    #10
  11. JC

    JC Guest

    On Mon, 26 Mar 2007 20:30:16 +0100, Tim <> wrote:

    >Yes, but you got it for free. In return for free, it was locked.
    >
    >Do you expect gradwell to give stuff away?


    It's not the fact that it was locked, so much as the fact that our
    account details were hard coded in to it. If I'd given it away or left
    it lying around, anyone could have picked it up and made calls on our
    account without our knowledge. I can't even sell or pass it on to
    another Gradwell user.

    The fact is, they offered a "free" ATA in return for an additional
    £90+ of rental a year. No mention of being locked to our account and
    no option on the signup form to reject it - I wouldn't have paid the
    original postage if there were.

    It's ironic that I actually signed up for another Gradwell inbound SIP
    only account with plans to use this ATA with my own Asterisk box.
    Because it's locked to their system this isn't possible.

    Rgds
    Jonathan
     
    JC, Mar 27, 2007
    #11
  12. JC

    JC Guest

    On Mon, 26 Mar 2007 20:30:44 +0100, Desk Rabbit <>
    wrote:

    >OOI Why SIP and not IAX?


    I'm not really a fan of IAX/IAX2. Yes it gets past NAT nicely and is a
    bit more efficient with multiple calls, but it has limitations
    regarding codecs etc. Plus a Cisco PIX can't do fixup on it AFAIK.

    >> Their website's a bit of a mess but not the worst I've seen. Certainly
    >> clarity on VAT and contract length is needed.

    >
    >Indeed, that has put me off them in the past.


    FWIW they seem like a pretty decent company. I know Peter Gradwell's a
    respected member of the UK internet community. I hope this is just an
    oversight.

    Rgds
    Jonathan
     
    JC, Mar 27, 2007
    #12
  13. JC

    Ivor Jones Guest

    "JC" <> wrote in message
    news:
    > On Mon, 26 Mar 2007 20:30:44 +0100, Desk Rabbit
    > <> wrote:
    >
    > > OOI Why SIP and not IAX?

    >
    > I'm not really a fan of IAX/IAX2. Yes it gets past NAT
    > nicely and is a bit more efficient with multiple calls,
    > but it has limitations regarding codecs etc. Plus a Cisco
    > PIX can't do fixup on it AFAIK.
    >
    > > > Their website's a bit of a mess but not the worst
    > > > I've seen. Certainly clarity on VAT and contract
    > > > length is needed.

    > >
    > > Indeed, that has put me off them in the past.

    >
    > FWIW they seem like a pretty decent company. I know Peter
    > Gradwell's a respected member of the UK internet
    > community. I hope this is just an oversight.


    Indeed. I have a single line account with them as a backup in case of
    failure of my other systems and apart from one or two hiccups when first
    set up, which were very quickly dealt with, has been faultess ever since.

    Ivor
     
    Ivor Jones, Mar 27, 2007
    #13
  14. JC

    JC Guest

    On Mon, 26 Mar 2007 20:31:38 +0100, Peter Gradwell
    <> wrote:

    >I guess it shouldn't be too hard for us to mention that we usually only
    >have a one month contract term, and prices quoted usually exclude VAT.


    Peter, don't take this the wrong way as I'm sure it's an oversight,
    but on the home page of your website you have the following banner:

    "Free calls over the internet are a great incentive to use VoIP at
    home to keep in touch with friends and family here and overseas."

    To me this reads as a service for individual or non-business users.

    Clicking on the link gives a page with the following disclaimer:

    "Note: The prices quoted here do not include VAT and may change from
    time to time. Please consult our product pages and price list for
    current pricing."

    I suspect this would be regarded as a breach of the Price Marking
    Order 2004.

    Rgds
    Jonathan
     
    JC, Mar 27, 2007
    #14
  15. JC

    JC Guest

    JC, Mar 27, 2007
    #15
  16. JC

    JC Guest

    On Tue, 27 Mar 2007 11:48:36 +0100, Peter Gradwell
    <> wrote:

    >My view is that all calls come in/go out from the PSTN over SIP - you
    >can't buy a pstn switch that supports IAX.


    ....and since the PSTN as we know it is soon to be replaced by 21CN, a
    giant IP SIP network, a very sensible plan.

    >A call that goes cisco -> asterisk -> asterisk -> phone is never going
    >to be as good as one that goes cisco -> asterisk -> phone.


    I think the issue here is that if you keep everything SIP, then the
    RTP streams can be redirected between endpoints. As soon as you have
    to convert between protocols, you loose that ability and potentially
    have an additional leg in the call.

    Rgds
    Jonathan
     
    JC, Mar 27, 2007
    #16
  17. JC

    JC Guest

    On Mon, 26 Mar 2007 20:39:08 +0100, Peter Gradwell
    <> wrote:

    >Our view is that the camrivox box is easier to use and has better
    >software than the linksys. It represents a nice, easy to use, voip adaptor.


    Since none of the options appear to be configurable, other than basic
    IP address and CODEC, I'd certainly beg to differ. ;-)

    >PSTN fall back is configured automatically I think.


    Again, impossible to tell and certainly no way to set any kind of
    dialplan (for example 999 to PSTN).

    >that represents a pretty good bargain on a device which retails for
    >GBP45 (i.e. cost of 8.50 + 15). You can't buy a linksys adaptor for any
    >where near that.


    The Linksys is £46.41
    http://www.broadbandbuyer.co.uk/Shop/ShopDetail.asp?ProductID=3473

    ....and in my opinion is an infinitely better device in all respects.
    If you're paying more than £ 15 trade for those boxes then someone's
    having a laugh. ;-)

    >We'd be happy to move it to another Gradwell account of course.


    Yes, but I've also signed up for your inbound account that forward to
    sip:user@mypbx for my Asterisk box. Unless it can be configured with
    the internal address of my server, it's of no use to me. If you could
    just remove my account details, at least I could put it on ebay. ;-)

    Rgds
    Jonathan
     
    JC, Mar 27, 2007
    #17
  18. JC

    Desk Rabbit Guest

    JC wrote:
    > On Mon, 26 Mar 2007 20:30:44 +0100, Desk Rabbit <>
    > wrote:
    >
    >> OOI Why SIP and not IAX?

    >
    > I'm not really a fan of IAX/IAX2. Yes it gets past NAT nicely and is a
    > bit more efficient with multiple calls, but it has limitations
    > regarding codecs etc.


    What codec limitations? I'm not aware of any limitations.

    >Plus a Cisco PIX can't do fixup on it AFAIK.


    I don't use Cisco kit so I don't know what that is.
     
    Desk Rabbit, Mar 27, 2007
    #18
  19. JC

    JC Guest

    On Tue, 27 Mar 2007 21:15:21 +0100, Desk Rabbit <>
    wrote:

    >What codec limitations? I'm not aware of any limitations.


    This might be more an Asterisk than a protocol limitation, but I
    recall some problems getting video and G.722 working over IAX. Again,
    probably to do with transcoding to SIP though in Asterisk.

    >>Plus a Cisco PIX can't do fixup on it AFAIK.

    >
    >I don't use Cisco kit so I don't know what that is.


    A Cisco PIX firewall can rewrite the information within SIP to allow
    access through NAT or other firewall restrictions. By implementing
    "fixup", SIP devices can appear to be on the real IP of the PIX while
    it translates to the internal phones without any need for STUN or
    opening ports on the firewall.

    Rgds
    Jonathan
     
    JC, Mar 27, 2007
    #19
  20. hi

    >> We'd be happy to move it to another Gradwell account of course.

    >
    > Yes, but I've also signed up for your inbound account that forward to
    > sip:user@mypbx for my Asterisk box. Unless it can be configured with
    > the internal address of my server, it's of no use to me. If you could
    > just remove my account details, at least I could put it on ebay. ;-)


    I don't mind changing it to that, if you mail me the details.

    cheers
    peter
     
    Peter Gradwell, Mar 27, 2007
    #20
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