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BT-101 - dodgy dc connector - corrupted flash - cure found?

 
 
Ivor Jones
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      07-25-2005
Alan Foster wrote:

[snip]

> For the time being I've put the Fritz on the back burner so to
> speak. It's a pity really, as I would dearly like to get my
> household DECT phones bridged over to VoIP as well as the PSTN. However,
> apart from the echo problem - still unresolved - as far as
> I can see there is no way to make dialling out default to the PSTN.
> I would rather have to dial a code to get a VoIP line with default
> to PSTN than the other way round.


Hmm, that's quite an unusual request, most people get VoIP to make calls
on it as it's usually cheaper than the PSTN..!

Actually, it's quite easy. You could use dialling rules to do what you
want, I have my Fritz!Box set to dial all non-geographic (0845, 0870,
mobile etc.) numbers via the PSTN as it's cheaper for those than Sipgate,
but Sipgate is cheaper for normal geographic numbers so they go that way.

If you put 0 in the dialling rules for PSTN access then all numbers
beginning with 0 will go that way, then all Sipgate numbers dialled on
their 7 digit SIP ID's will go that way. You can then still dial the
access code to force a call whichever way you want.

I still don't understand your audio problems though, mine is perfect in
that respect. I would suspect you have a faulty unit, I suggest you return
it for an exchange one and see if that is any better, if not you should
have no problems in getting a refund.

Ivor


 
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JC
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      07-25-2005
On Mon, 25 Jul 2005 08:57:10 +0100, "Sparks" <(E-Mail Removed)> wrote:

>I have two, 'connected' to my Asterisk Box, both working fine for the 6 or
>so months I have had them!
>
>Sparks...


Maybe I'm just bitter because of the pain mine caused but every
firmware revision seemed to break a different feature. The original
firmware was unstable, the latest stable (at the time) revision broke
the message waiting notification, early dial never worked, caller ID
display would be a garbled version of the text and not the number,
then the thing just hung with flashing lights, etc etc.

There also still doesn't (and I'd like to be corrected on this) appear
to be any way to easily localise (to UK) the call status tones and the
built in switch on the 102 is only 10 Mbps.

Rgds
Jonathan

 
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Alan Foster
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      07-25-2005
On Mon, 25 Jul 2005 20:47:51 +0100, Ivor Jones <(E-Mail Removed)>
wrote:


>
> Hmm, that's quite an unusual request, most people get VoIP to make calls
> on it as it's usually cheaper than the PSTN..!


Yeah, I know, but I do have my reasons!!

>
> Actually, it's quite easy. You could use dialling rules to do what you
> want,If you put 0 in the dialling rules for PSTN access then all numbers
> beginning with 0 will go that way, then all Sipgate numbers dialled on
> their 7 digit SIP ID's will go that way. You can then still dial the
> access code to force a call whichever way you want.
>


Yes, I understand all that, but I have two requirements:

1. I need to be able to route <any> number by <either> route, so dialling
rules won't work!
2. Others in the household need access to the PSTN by just picking up a
phone. It is undesirable to have their calls going through my VoIP
account, or for them to have any impedement to accessing the PSTN account.


> I still don't understand your audio problems though, mine is perfect in
> that respect. I would suspect you have a faulty unit, I suggest you
> return
> it for an exchange one and see if that is any better, if not you should
> have no problems in getting a refund.



In fairness, as I said, I haven't had time to check it out thoroughly yet
- it may just be a wrong codec - so I won't feel justified in returning it
unless and until I can prove it is faulty. Unfortunately I have too many
calls on my time at the moment to do much about it... one day...

Regards
Alan.

--
Using Opera's revolutionary e-mail client: http://www.opera.com/mail/
 
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Ivor Jones
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      07-25-2005
Alan Foster wrote:
> On Mon, 25 Jul 2005 20:47:51 +0100, Ivor Jones
> <(E-Mail Removed)> wrote:
>
>
>>
>> Hmm, that's quite an unusual request, most people get VoIP to make
>> calls on it as it's usually cheaper than the PSTN..!

>
> Yeah, I know, but I do have my reasons!!


Fair enough..!

>> Actually, it's quite easy. You could use dialling rules to do what
>> you want,If you put 0 in the dialling rules for PSTN access then
>> all numbers beginning with 0 will go that way, then all Sipgate
>> numbers dialled on their 7 digit SIP ID's will go that way. You
>> can then still dial the access code to force a call whichever way
>> you want.

>
> Yes, I understand all that, but I have two requirements:
>
> 1. I need to be able to route <any> number by <either> route, so
> dialling rules won't work!
> 2. Others in the household need access to the PSTN by just picking
> up a phone. It is undesirable to have their calls going through my
> VoIP account, or for them to have any impedement to accessing the
> PSTN account.


I don't see an easy way to do this without dialling the access code for
each line (PSTN or VoIP) first. Actually, I don't need to do this as I
have a PABX, the lines from the Fritz!Box are just two of the incoming
lines, so if I want to force a call via a particular route I just dial the
appropriate line's direct access code (81, 82, etc) rather than the usual
9, which picks a line at random. It works for me, but probably won't suit
you..!

>> I still don't understand your audio problems though, mine is
>> perfect in that respect. I would suspect you have a faulty unit, I
>> suggest you return
>> it for an exchange one and see if that is any better, if not you
>> should have no problems in getting a refund.

>
> In fairness, as I said, I haven't had time to check it out
> thoroughly yet - it may just be a wrong codec - so I won't feel
> justified in returning it unless and until I can prove it is
> faulty. Unfortunately I have too many calls on my time at the
> moment to do much about it... one day...


Could be. I don't see a way to select a specific codec in the setup
though, but I will check it out with AVM. Do get in touch if you need to
though.

Ivor


 
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Ian
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      07-26-2005
<snip>

> Hmm, that's quite an unusual request, most people get VoIP to make calls
> on it as it's usually cheaper than the PSTN..!
>

<snip>

Sorry but it had to be done !

BT to mobile
O2 numbers (fm1) 12.60 11.81 3.61
Orange numbers (fm4) 13.60 11.40 6.00
Vodafone numbers (fm5) 14.88 7.68 4.78

gradwell.com
Mobile 12.00 10.00 7.00

Voiptalk Silver
Mobile (02)
10.00 8.00 5.00

all the above also support per second billing

And finally

Sipgate 14.9p at all times.........

and they even bill by the minute.

So that 1 min 10 sec evening moble call will either cost 6p with one
supplier and the most expensive other supplier 30p !!

Now maybe people will see why sipgate are "free" they are paying by the
second but billing by the minute..

So what im saying is VOIP should not be seen as cheaper its an altenative,
but watch the costs.

Ian








 
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Ivor Jones
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      07-26-2005

"Ian" <spam"AT"bathfordhill.co.uk> wrote in message
news:(E-Mail Removed)...

[snip]

> So that 1 min 10 sec evening moble call will either cost 6p with one
> supplier and the most expensive other supplier 30p !!
>
> Now maybe people will see why sipgate are "free" they are paying by the
> second but billing by the minute..
>
> So what im saying is VOIP should not be seen as cheaper its an
> altenative,
> but watch the costs.
>
> Ian


You're talking exclusively about calls to mobiles. I never use VoIP for
those, I use my own mobile to call other mobiles. I only use Sipgate for
calls to other Sipgate users (free) or landlines (1.19p/min).

Ivor


 
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Paul D.Smith
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Posts: n/a
 
      07-27-2005
> All owners of BT-100's will know that the power connector is decidedly
> dodgy, and I have now experienced a real heart-stopping moment as a

result!

Not just me then! Get a sharp knife and carefully remove about 5mm of the
moulding plastic around the end of the power connector, increasing the
length of the visible silver end. If your connector is like mine, it will
still be in one piece and solid and you will now get a good contact.

Paul DS.


 
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Ralf Janssen
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Posts: n/a
 
      07-28-2005
Jonathan,
what provider are you using, and what's your home base ?

Kind Greetings

Ralf

"JC" <(E-Mail Removed)> wrote in message
news:(E-Mail Removed)...
> On Mon, 25 Jul 2005 08:57:10 +0100, "Sparks" <(E-Mail Removed)> wrote:
>
> >I have two, 'connected' to my Asterisk Box, both working fine for the 6

or
> >so months I have had them!
> >
> >Sparks...

>
> Maybe I'm just bitter because of the pain mine caused but every
> firmware revision seemed to break a different feature. The original
> firmware was unstable, the latest stable (at the time) revision broke
> the message waiting notification, early dial never worked, caller ID
> display would be a garbled version of the text and not the number,
> then the thing just hung with flashing lights, etc etc.
>
> There also still doesn't (and I'd like to be corrected on this) appear
> to be any way to easily localise (to UK) the call status tones and the
> built in switch on the 102 is only 10 Mbps.
>
> Rgds
> Jonathan
>



 
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JC
Guest
Posts: n/a
 
      07-28-2005
On Thu, 28 Jul 2005 11:07:21 +0200, "Ralf Janssen"
<(E-Mail Removed)> wrote:

>Jonathan,
>what provider are you using, and what's your home base ?


This was using my own platform based on Asterisk and SER with the
Budgetone registering directly with Asterisk.

Rgds
Jonathan

 
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