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

[ Post a follow-up to this message ]

On 2007-03-24, JC <nospam@nospam.comm> 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.

[ Post a follow-up to this message ]

On Mon, 26 Mar 2007 13:11:22 +0000 (UTC), Brian <bandj@o2.co.uk>
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

[ Post a follow-up to this message ]

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.

[ Post a follow-up to this message ]

JC wrote:
> On Mon, 26 Mar 2007 13:11:22 +0000 (UTC), Brian <bandj@o2.co.uk>
> 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).

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

Do you expect gradwell to give stuff away?

Tim

[ Post a follow-up to this message ]

Desk Rabbit wrote:
> JC wrote: 
>
> 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.
 
>
> 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

[ Post a follow-up to this message ]

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

[ Post a follow-up to this message ]

Peter Gradwell wrote:
> Desk Rabbit wrote: 
>
> 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

> 
>
> 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...h/prices/02.htm

[ Post a follow-up to this message ]

Desk Rabbit wrote:
> Peter Gradwell wrote: 
>
> 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

[ Post a follow-up to this message ]

Peter Gradwell wrote:
> Desk Rabbit 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.
>
> 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>

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