I am considering signing up with Gradwell for VoIP, for home/small
business use.
By far the most important criterion for me is RELIABILITY.
1. When you pick up the phone, do you always get a dial tone?
2. When you dial a number, do you get connected almost immediately, or
do you have to wait 30 seconds to connect?
3. Do you get cut off in mid-call?
4. Does 1471 tell you the date/time that someone called you, as well as
the calling number?
In other words, I'm looking for a basic telephone service, as reliable
as BT or NTL. A service that has got the fundamentals right. A service
that understands what single-point-of-failure means. The frills can come
later.
news wrote:
> I am considering signing up with Gradwell for VoIP, for home/small
> business use.
>
> By far the most important criterion for me is RELIABILITY.
>
> 1. When you pick up the phone, do you always get a dial tone?
>
> 2. When you dial a number, do you get connected almost immediately, or
> do you have to wait 30 seconds to connect?
>
> 3. Do you get cut off in mid-call?
>
> 4. Does 1471 tell you the date/time that someone called you, as well as
> the calling number?
>
> In other words, I'm looking for a basic telephone service, as reliable
> as BT or NTL. A service that has got the fundamentals right. A service
> that understands what single-point-of-failure means. The frills can come
> later.
>
> Experiences of Gradwell anyone?
>
1. I have never not had a dial tone
2. There is no noticeable difference with the PSTN
3. Only if you've messed up setting up your phone
4. ? (Peter may be along to answer) Though my SPA841 gives me that
information in my missed calls list
> I am considering signing up with Gradwell for VoIP, for home/small
> business use.
>
> By far the most important criterion for me is RELIABILITY.
So just use a PSTN line then.
> In other words, I'm looking for a basic telephone service, as reliable
> as BT or NTL. A service that has got the fundamentals right. A service
> that understands what single-point-of-failure means. The frills can come
> later.
I reckon your internet connection is far more likely to be the single point
of failure than Gradwell.
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"news" <news@care4free.net> wrote in message
news:raw1k8FnQVaFFwrj@care4free.net
> I am considering signing up with Gradwell for VoIP, for
> home/small business use.
>
> By far the most important criterion for me is RELIABILITY.
>
> 1. When you pick up the phone, do you always get a dial
> tone?
Depends on your ATA - dial tone is generated by the ATA not by the
provider.
> 2. When you dial a number, do you get connected almost
> immediately, or do you have to wait 30 seconds to connect?
Usually, yes.
> 3. Do you get cut off in mid-call?
Not in my experience.
> 4. Does 1471 tell you the date/time that someone called
> you, as well as the calling number?
No idea, never used 1471 on Gradwell, but the website status page gives
time/date of made/received calls (although with a short delay, not
immediately).
> In other words, I'm looking for a basic telephone
> service, as reliable as BT or NTL. A service that has got
> the fundamentals right. A service that understands what
> single-point-of-failure means. The frills can come later.
>
> Experiences of Gradwell anyone?
I can only speak as a single line customer, in my experience they are
excellent. For the single line service, they offer a 3 month free trial:
In message <4sttn1F11fqnjU1@mid.individual.net>, Ivor Jones
<ivor@despammed.invalid> writes
>"news" <news@care4free.net> wrote in message
>news:raw1k8FnQVaFFwrj@care4free.net
>>
>> 1. When you pick up the phone, do you always get a dial
>> tone?
>
>Depends on your ATA - dial tone is generated by the ATA not by the
>provider.
>
You're right, of course, but that isn't quite what I was getting at. You
will only get the dial tone if the phone is registered. If the VoIP
provider's system has crashed or is otherwise unavailable, the ATA or
VoIP phone will fall back to a non-registered condition and you will not
get a dial tone. The availability of the provider's system was what I
had in mind in my question.
So, to re-phrase the question: How often have you experienced the NR
condition with Gradwell? Never? Almost never? A couple of times a year?
At least once a month? More frequently? Has the NR condition ever
extended to several minutes (or even hours), rather than just a
transient blip of a few seconds?
"news" <news@care4free.net> wrote in message
news:vx0EuhD1zfaFFwpa@care4free.net
[snip]
> So, to re-phrase the question: How often have you
> experienced the NR condition with Gradwell? Never? Almost
> never? A couple of times a year? At least once a month?
> More frequently? Has the NR condition ever extended to
> several minutes (or even hours), rather than just a
> transient blip of a few seconds?
Never. Since I've been using Gradwell (about a year) the only problem I
had was incoming calls going straight to voicemail without ringing the
phone. This occurred only on PSTN calls, not on SIP calls from other
Gradwell users. It was corrected by a slight parameter change in my
account at Gradwell's end and has been fine ever since.
They have an excellent reputation for business use and I can recommend
them wholeheartedly. Additionally, the MD Peter Gradwell posts here
sometimes, and may well answer a question if you ask..!
> In message <4sttn1F11fqnjU1@mid.individual.net>, Ivor Jones
><ivor@despammed.invalid> writes
>>"news" <news@care4free.net> wrote in message
>>news:raw1k8FnQVaFFwrj@care4free.net
>>>
>>> 1. When you pick up the phone, do you always get a dial
>>> tone?
>>
>>Depends on your ATA - dial tone is generated by the ATA not by the
>>provider.
>>
>
> You're right, of course, but that isn't quite what I was getting at. You
> will only get the dial tone if the phone is registered. If the VoIP
> provider's system has crashed or is otherwise unavailable, the ATA or
> VoIP phone will fall back to a non-registered condition and you will not
> get a dial tone. The availability of the provider's system was what I
> had in mind in my question.
To eliminate the dial tone on a SPA2000 it is necessary to have a line
not registered and the parameter 'Make Call Without Reg:' set to 'no'.
With 'Make Call Without Reg:' as 'yes' there is a dial tone. Other ATAs
or softphones may behave differently I suppose but I cannot see why they
should.
In message <ekde7i$dja$1$8300dec7@news.demon.co.uk>, Brian
<bandj@o2.co.uk> writes
>
>To eliminate the dial tone on a SPA2000 it is necessary to have a line
>not registered and the parameter 'Make Call Without Reg:' set to 'no'.
>With 'Make Call Without Reg:' as 'yes' there is a dial tone. Other ATAs
>or softphones may behave differently I suppose but I cannot see why they
>should.
>
>Brian.
All understood. But as I explained in my previous reply to Ivor (and
didn't make clear enough in my original post), my concern was really
about the provider end, not the subscriber end.
The same comment applies to another of my questions, about being cut off
in mid call. Certainly this could happen if the subscriber end is not
set up properly, but what I wanted to know was if Gradwell had a habit
of terminating calls prematurely.
"news" <news@care4free.net> wrote in message
news:vx0EuhD1zfaFFwpa@care4free.net...
>
> You're right, of course, but that isn't quite what I was getting at. You
> will only get the dial tone if the phone is registered. If the VoIP
> provider's system has crashed or is otherwise unavailable, the ATA or
> VoIP phone will fall back to a non-registered condition and you will not
> get a dial tone. The availability of the provider's system was what I
> had in mind in my question.
The DrayTek Vigor2800VG has an option "Play dial tone only when account
registered", which depending on whether it is selected or not, gives a dial
tone in either state of the VoIP account (ie registered or non-registered).
In message <456acb79$0$8735$ed2619ec@ptn-nntp-reader02.plus.net>, Rob
<nobody@this.place.invalid> writes
>
>
>The DrayTek Vigor2800VG has an option "Play dial tone only when account
>registered", which depending on whether it is selected or not, gives a dial
>tone in either state of the VoIP account (ie registered or non-registered).
>
OK, but is there any point in presenting a dial tone to the user if the
phone/ATA is not registered?
"news" <news@care4free.net> wrote in message
news:raw1k8FnQVaFFwrj@care4free.net...
>I am considering signing up with Gradwell for VoIP, for home/small business
>use.
>
> By far the most important criterion for me is RELIABILITY.
>
> 1. When you pick up the phone, do you always get a dial tone?
>
> 2. When you dial a number, do you get connected almost immediately, or do
> you have to wait 30 seconds to connect?
>
> 3. Do you get cut off in mid-call?
>
> 4. Does 1471 tell you the date/time that someone called you, as well as
> the calling number?
>
> In other words, I'm looking for a basic telephone service, as reliable as
> BT or NTL. A service that has got the fundamentals right. A service that
> understands what single-point-of-failure means. The frills can come later.
>
> Experiences of Gradwell anyone?
My experience of Gradwell:
Since I signed up about a year ago [and once I had sorted out all firewall
issues at this end] I have had only one problem with the basic telephony
part of the service.
This occurred when Telewest had screwed up a datafill and "lost" the linkage
between the DDI number and Gradwell's numbering provider - not actually
Gradwell's fault (and confused Telewest greatly as it was local to a
specific exchange or exchange group - ie completely a screw-up at their
end).
I am also aware of a couple of brief outages (because they were listed on
Gradwell's site) that would have removed telephony briefly (less than an
hour IIRC) during this period - but I didn't notice them directly myself.
I've also encountered a couple of quirks with the web setup for their more
sophisticated "Virtual PBX" product - namely that stuff you do on the
website is not (quite) instantly reflected in the telephony side, and in
particular if you point a PSTN number at something (conference, voice menu,
etc) you have only just setup there may be a brief period where the PSTN
number does something odd (rings out, or responds with silence) before the
telephony system catches up with the web setup.
Finally I've had a problem with using the conferencing facility with larger
numbers of participants, particularly when some of them were US based - on
one occasion we had a ~10 second echo (at more or less full volume) which
made the conference almost impossible to use! [However I think here we (a)
are probably pushing the limits of the technology and (b) need to look as
much at reducing far end (phone/softphone) echo].
Basically I'm very satisfied - I thought a "warts and all" description was
more useful than "it's fab/works for me". I'd concur with the other
responder who pointed out that your net connectivity is likely to be an
order of magnitude less reliable. [Interestingly my greatest period of no
connectivity during the time I've been with Gradwell was when BT replaced
the telegraph pole my voice/ADSL line is provided from].
Of course an advantage of Gradwell is that they can provide a divert to a
PSTN number, so if you are without (primary) connectivity for a period of
time you could arrange for incoming calls to be routed to a landline (or
even mobile), albeit at your cost...
"news" <news@care4free.net> wrote in message
news:sCE0ixE1EtaFFwrj@care4free.net...
>
> OK, but is there any point in presenting a dial tone to the user if the
> phone/ATA is not registered?
Under certain circumstances, yes, after you realise the account may not be
registered because a call has failed.
Even though you could "dial out" using that particular unregistered account,
it would obviously fail, but as multiple accounts can be associated with
each FXS port, then if dial tone is present you could go ahead and make a
call using another account either by selecting it by dialling a prefix, or
perhaps letting the dialplan route a different call via another
pre-determined account according to the type of call, eg international or
mobile.
"news" <news@care4free.net> wrote in message
news:raw1k8FnQVaFFwrj@care4free.net...
>I am considering signing up with Gradwell for VoIP, for home/small business
>use.
>
> By far the most important criterion for me is RELIABILITY.
>
> 1. When you pick up the phone, do you always get a dial tone?
>
> 2. When you dial a number, do you get connected almost immediately, or do
> you have to wait 30 seconds to connect?
>
> 3. Do you get cut off in mid-call?
>
> 4. Does 1471 tell you the date/time that someone called you, as well as
> the calling number?
>
> In other words, I'm looking for a basic telephone service, as reliable as
> BT or NTL. A service that has got the fundamentals right. A service that
> understands what single-point-of-failure means. The frills can come later.
I have been using gradwell at a small office (7 phones) using the centrex
pbx facilities.
On Mon, 27 Nov 2006 11:46:00 GMT, news <news@care4free.net> wrote:
>In message <456acb79$0$8735$ed2619ec@ptn-nntp-reader02.plus.net>, Rob
><nobody@this.place.invalid> writes
>>
>>
>>The DrayTek Vigor2800VG has an option "Play dial tone only when account
>>registered", which depending on whether it is selected or not, gives a dial
>>tone in either state of the VoIP account (ie registered or non-registered).
>>
>
>OK, but is there any point in presenting a dial tone to the user if the
>phone/ATA is not registered?
Its also possible to make peer-to peer calls to a phone connected to
another router by direct IP addressing without the need for a SIP
registrar. They may have included the option to avoid confusion if
that function is used.
>
> In other words, I'm looking for a basic telephone service, as reliable
> as BT or NTL. A service that has got the fundamentals right. A service
> that understands what single-point-of-failure means. The frills can come
> later.
Lots of people have written nice things about us - but, before I read
those, I was going to say that, if you want a BT phone line, you should
get one from BT, because VoIP services in general, are not BT phone lines.
There are loads of points of failure in VoIP - we route calls using
Linux servers instead of Marconi switches and we run it over ADSL lines.
Plus, SIP isn't as good as the PSTN routing protocol (ss7). It only has
6 error modes, rather than the 40 odd error codes SS7 gives you.
That's not to say voip doesn't work well for a good number of people
(many thousand on our system) and it offers some great features - but
it's not a BT line.
>
> In other words, I'm looking for a basic telephone service, as reliable
> as BT or NTL. A service that has got the fundamentals right. A service
> that understands what single-point-of-failure means. The frills can come
> later.
>
> Experiences of Gradwell anyone?
Gradwell seem to be about as good as VoIP gets in terms of reliability.
Which is to say nearly as good as BT and for many people functionally
no less good than BT - but when you get right down to it they are not
and can not be as reliable. Not to put too fine a point on it they
simply have more transport layers for your call to be carried over and
hence more things that could go wrong. Some of those transport layers
are not built on the same reliability-at-all-costs basis as the PSTN.
VoIP simply cannot deliver the sort of single-point-of-failure system
you want unless you happen to be on 21CN in Wick - and I would not want
to bet real money on normal PSTN levels of reliability for Wick over
the next few months.
POTS is still the most reliable if reliability is your overriding
criteria. If you want the features and benefits of VoIP and as much
reliability as you can get then from my experience I would say that
Gradwell is a good choice. Personally I find the Gradwell service
sufficiently reliable that I have not yet had a problem and so for me
there has been no real-world difference in reliability.
NicHughes wrote:
> Gradwell seem to be about as good as VoIP gets in terms of reliability.
> Which is to say nearly as good as BT and for many people functionally
> no less good than BT - but when you get right down to it they are not
> and can not be as reliable. Not to put too fine a point on it they
> simply have more transport layers for your call to be carried over and
> hence more things that could go wrong. Some of those transport layers
> are not built on the same reliability-at-all-costs basis as the PSTN.
> VoIP simply cannot deliver the sort of single-point-of-failure system
> you want unless you happen to be on 21CN in Wick - and I would not want
> to bet real money on normal PSTN levels of reliability for Wick over
> the next few months.
Yes.
But it depends what you want.
For instance, you can get an easy 8 channels of G.711 voip down an ADSL
max line, if you are near the exchange.
Compared to paying BT for 8 ISDN lines, the costs quickly roll in favour
of VoIP.
5 minutes of downtime a month when the ADSL decides to resync may well
be preferable to paying BT $$$ every month.
Its a tradeoff.
There are other advantages too - like BT won't usually port numbers
between areas. But you can port the numbers to a VoIP provider and take
them anywhere you like.
Or maybe, buying extra services online in real time may be preferable to
talking to BT's call centre.
> But would anyone want to be associated with anything that Richard (R)
> Ashton was connected with?
> It put me off straight away
Explanation?
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<http://ale.cx/> (AIM:troffasky) (UnSoEsNpEaTm@ale.cx)
15:17:02 up 9 days, 18:59, 2 users, load average: 0.00, 0.00, 0.00
This is my BOOOOOOOOOOOOOOOOOOOOOMSTICK
news wrote:
> In message <456acb79$0$8735$ed2619ec@ptn-nntp-reader02.plus.net>, Rob
> <nobody@this.place.invalid> writes
>>
>>
>> The DrayTek Vigor2800VG has an option "Play dial tone only when account
>> registered", which depending on whether it is selected or not, gives a
>> dial
>> tone in either state of the VoIP account (ie registered or
>> non-registered).
>>
>
> OK, but is there any point in presenting a dial tone to the user if the
> phone/ATA is not registered?
>
Yes, if you run the service without registrations. (only really need
registrations if the IP of the client is dynamic). Most ITSPs require
this, but not all. The Sipura range of ATAs support this.
Thomas Sandford wrote:
> Of course an advantage of Gradwell is that they can provide a divert to a
> PSTN number, so if you are without (primary) connectivity for a period of
> time you could arrange for incoming calls to be routed to a landline (or
> even mobile), albeit at your cost...
>
You can? I've asked about this in the past and have been told that it
couldn't be done.
Thomas Kenyon formulated on Saturday :
> Thomas Sandford wrote:
>> Of course an advantage of Gradwell is that they can provide a divert to a
>> PSTN number, so if you are without (primary) connectivity for a period of
>> time you could arrange for incoming calls to be routed to a landline (or
>> even mobile), albeit at your cost...
>>
>
> You can? I've asked about this in the past and have been told that it
> couldn't be done.
<http://www.gradwell.com/voip/PSTN_forwarding.php>.......who have you
asked?
Jono wrote:
> Thomas Kenyon formulated on Saturday :
>> Thomas Sandford wrote:
>>> Of course an advantage of Gradwell is that they can provide a divert
>>> to a PSTN number, so if you are without (primary) connectivity for a
>>> period of time you could arrange for incoming calls to be routed to a
>>> landline (or even mobile), albeit at your cost...
>>>
>>
>> You can? I've asked about this in the past and have been told that it
>> couldn't be done.
>
> <http://www.gradwell.com/voip/PSTN_forwarding.php>.......who have you
> asked?
>
Sorry, I was thinking of it as a failover services. (Like adding a
second Dial statement to an asterisk dialplan). Didn't read tyhe OP
carefully enough.
Thomas Kenyon formulated the question :
> Jono wrote:
>> Thomas Kenyon formulated on Saturday :
>>> Thomas Sandford wrote:
>>>> Of course an advantage of Gradwell is that they can provide a divert to a
>>>> PSTN number, so if you are without (primary) connectivity for a period of
>>>> time you could arrange for incoming calls to be routed to a landline (or
>>>> even mobile), albeit at your cost...
>>>>
>>>
>>> You can? I've asked about this in the past and have been told that it
>>> couldn't be done.
>>
>> <http://www.gradwell.com/voip/PSTN_forwarding.php>.......who have you
>> asked?
>>
> Sorry, I was thinking of it as a failover services. (Like adding a second
> Dial statement to an asterisk dialplan). Didn't read tyhe OP carefully
> enough.
I was surprised.
Even so though, can Gradwell not provide failover to PSTN? If that's
the case, I'm even more surprised.
On 2006-12-02, Thomas Kenyon <tom@art-it-services.co.uk> wrote:
> news wrote:
>>
>> OK, but is there any point in presenting a dial tone to the user if the
>> phone/ATA is not registered?
>>
> Yes, if you run the service without registrations. (only really need
> registrations if the IP of the client is dynamic). Most ITSPs require
> this, but not all. The Sipura range of ATAs support this.
Is having a static or dynamic IP really an issue for registration? A
phone which moves from one static IP to another would still have to be
registered with an ITSP to be located.
Thomas Kenyon wrote:
> Sorry, I was thinking of it as a failover services. (Like adding a
> second Dial statement to an asterisk dialplan). Didn't read tyhe OP
> carefully enough.
Can't you use a hunt group with a very long timeout?
I haven't tried this myself, but on most of the SIP systems I've played
with, a hunt group will fail over to the next choice if the first choice
doesn't ring.
In article <eksibu$a16$1$8300dec7@news.demon.co.uk>,
Brian <bandj@o2.co.uk> writes:
> On 2006-12-02, Thomas Kenyon <tom@art-it-services.co.uk> wrote:
>
>> news wrote:
>>>
>>> OK, but is there any point in presenting a dial tone to the user if the
>>> phone/ATA is not registered?
>>>
>> Yes, if you run the service without registrations. (only really need
>> registrations if the IP of the client is dynamic). Most ITSPs require
>> this, but not all. The Sipura range of ATAs support this.
>
> Is having a static or dynamic IP really an issue for registration? A
> phone which moves from one static IP to another would still have to be
> registered with an ITSP to be located.
On 2006-12-02, Andrew Gabriel <andrew@cucumber.demon.co.uk> wrote:
> In article <eksibu$a16$1$8300dec7@news.demon.co.uk>,
> Brian <bandj@o2.co.uk> writes:
>> On 2006-12-02, Thomas Kenyon <tom@art-it-services.co.uk> wrote:
>>> Yes, if you run the service without registrations. (only really need
>>> registrations if the IP of the client is dynamic). Most ITSPs require
>>> this, but not all. The Sipura range of ATAs support this.
>>
>> Is having a static or dynamic IP really an issue for registration? A
>> phone which moves from one static IP to another would still have to be
>> registered with an ITSP to be located.
>
> Well, that would make it a dynamic IP then.
My thinking was in terms of a different permanent or temporary address
being assigned on each network the phone is used. The relocations have
the same effect as having a dynamic IP but, static or dynamic,
registration of the IP has to take place for the phone to be found.
Brian wrote:
> On 2006-12-02, Andrew Gabriel <andrew@cucumber.demon.co.uk> wrote:
>
>> In article <eksibu$a16$1$8300dec7@news.demon.co.uk>,
>> Brian <bandj@o2.co.uk> writes:
>>> On 2006-12-02, Thomas Kenyon <tom@art-it-services.co.uk> wrote:
>>>> Yes, if you run the service without registrations. (only really need
>>>> registrations if the IP of the client is dynamic). Most ITSPs require
>>>> this, but not all. The Sipura range of ATAs support this.
>>> Is having a static or dynamic IP really an issue for registration? A
>>> phone which moves from one static IP to another would still have to be
>>> registered with an ITSP to be located.
>> Well, that would make it a dynamic IP then.
>
> My thinking was in terms of a different permanent or temporary address
> being assigned on each network the phone is used. The relocations have
> the same effect as having a dynamic IP but, static or dynamic,
> registration of the IP has to take place for the phone to be found.
>
> Brian.
That isn't neccesary, some ITSPs (gradwell for one, voiptalk for
another) will allow you to specify a SIP URL for calls to be terminated
upon which does not rely upon you to register to their registration
serviers. As I stated before the Sipura range of ATAs support this.
I apologise if my english is a bit poor, I'm more than a bit slarnyeyed
as I write this.
Gradwell -- reliable? 
Linear Mode
