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On Tue, 3 Jul 2007 11:05:30 +0000 (UTC), Brian <> wrote:


>You have to have a provider who will accept a SIP address. That is, one
>who will route user@domain. Apparently voipuser does. Sipgate is one
>provider who doesn't.

So it seems there is some subtly in the terminology here .. like the
difference between a SIP 'number' (which I understand, and a
telephony term) and SIP 'address' which has no relevance to me under
my very limited exposure to SIP 'std' VoIP.

ie, I've been using Skype for ages but was therefore shielded from all
the complicated stuff by the fact that it's a closed network etc.

By the looks of it there isn't a 'simple' way to explain all the
various functions in less than a full FAQ (and that would have to be
aimed at a non regular VoIP user to be of any use to me).

But, I've got incoming VoIP calls ringing the DECT phone that is
normally *just* plugged into Ext2 of our PABX and have also tested
that I can still dial out (although that seems quite messy now ...
(*111#9 etc) is there a way where outgoing calls default to analogue?)
so I am getting somewhere.

I suppose it's like many things for many of us, as long we can get
something to work even if only in it's very basic form then that's
often good enough ... (video's, mobile phones, computers ... ) but we
may pick up some extra bits as / when we are shown or have a specific
need etc?

For the moment and certainly for the purpose of replacing our second
telephone line I'm happy to have a basic Sipgate number and pay for
the odd outgoing PSTN call.

All the best ..

T i m

> So it seems there is some subtly in the terminology here .. like the
> difference between a SIP 'number' (which I understand, and a
> telephony term) and SIP 'address' which has no relevance to me under
> my very limited exposure to SIP 'std' VoIP.

Your ATA is registered with a VOIP provider by using the SIP protocol.

Now, there are several ways how people can reach you:

- via a regular phonenumber (012/3456789), if your VOIP provider has linked
one to your account
- via a SIP address (), if your VOIP provider has linked
one to your account

If you use that VOIP provider only for outgoing calls (and your fixed line
for incoming calls), you don't need any of the above!

> ie, I've been using Skype for ages but was therefore shielded from all
> the complicated stuff by the fact that it's a closed network etc.
> By the looks of it there isn't a 'simple' way to explain all the
> various functions in less than a full FAQ (and that would have to be
> aimed at a non regular VoIP user to be of any use to me).

The advantage of SIP is that it is open, and very flexible and configurable.
So I guess the "disadvantage" can be that it is a bit more complex maybe.

I was also struggeling in the beginning with this new technology; but once
you understand the details it is really exciting how everything works and
can be configured. The funny thing is that I've spend more money on my VOIP
equipment lately, than I could ever save because of using SIP compared to
normal telephony

I am certainly open to help writing a FAQ concerning this matter!

> But, I've got incoming VoIP calls ringing the DECT phone that is
> normally *just* plugged into Ext2 of our PABX and have also tested
> that I can still dial out (although that seems quite messy now ...
> (*111#9 etc) is there a way where outgoing calls default to analogue?)
> so I am getting somewhere.

In the properties of Settings/Telephony/Extensions/FON1 (or whatever
connector you've connected your DECT phone to) you should select "fixed
line" as the default for all calls. It will then always dial out over the
fixed line, if you don't have any dial rules setup. There is no need to
enter any prefixes at all; it should just work like a "normale" phone
system.

>>So if you enable ENUM and you dial a number, it will check if it can call
>>that number for free and directly (without any conversion from one PSTN
>>provider to another).

> Assuming all called parties also have enabled ENUM I assume though
> John?

Yes. That is why we should encourage all SIP callers to setup ENUM for
their system! Voipuser.org is free and works very well.

> Yes. That is why we should encourage all SIP callers to setup ENUM for
> their system! Voipuser.org is free and works very well.

I meant e164.org instead of voipuser.org.

But voipuser.org also works very well

On Tue, 3 Jul 2007 15:13:37 +0200, "John Miller"
<> wrote:

>> So it seems there is some subtly in the terminology here .. like the
>> difference between a SIP 'number' (which I understand, and a
>> telephony term) and SIP 'address' which has no relevance to me under
>> my very limited exposure to SIP 'std' VoIP.
>
>Your ATA is registered with a VOIP provider by using the SIP protocol.
>
>Now, there are several ways how people can reach you:
>
>- via a regular phonenumber (012/3456789), if your VOIP provider has linked
>one to your account

Ok, that one I understand and make sense John ..

>- via a SIP address (), if your VOIP provider has linked
>one to your account

Ok, that one doesn't. I mean. I hear what you say but I'm not sure I
see the point of doing so?
>
>If you use that VOIP provider only for outgoing calls (and your fixed line
>for incoming calls), you don't need any of the above!

Understood.
>
>> ie, I've been using Skype for ages but was therefore shielded from all
>> the complicated stuff by the fact that it's a closed network etc.
>> By the looks of it there isn't a 'simple' way to explain all the
>> various functions in less than a full FAQ (and that would have to be
>> aimed at a non regular VoIP user to be of any use to me).
>
>The advantage of SIP is that it is open, and very flexible and configurable.
>So I guess the "disadvantage" can be that it is a bit more complex maybe.

Maybe!?! I never had any issues with Skype but then didn't take it any
further than just PC to PC comms.
>
>I was also struggeling in the beginning with this new technology; but once
>you understand the details it is really exciting how everything works and
>can be configured.

Oh I can see that John and why I was (am) interested in setting
something up, even if only as a test and to have in the background (to
save money on line rental) or in case I can ever help someone else.

> The funny thing is that I've spend more money on my VOIP
>equipment lately, than I could ever save because of using SIP compared to
>normal telephony

Isn't that often the way! In fact though, 1) I did need a more
reliable router, 2) If I *can* loose the rental on the second line
that will save me ~£10/month and 3) if we just end up with just the
BB on Virgin (no phones) I may well pay that via DD (rather than
paying on demand over the net) saving a further £5/month.
>
>I am certainly open to help writing a FAQ concerning this matter!

That could be very handy (for me anyway).
>
>> But, I've got incoming VoIP calls ringing the DECT phone that is
>> normally *just* plugged into Ext2 of our PABX and have also tested
>> that I can still dial out (although that seems quite messy now ...
>> (*111#9 etc) is there a way where outgoing calls default to analogue?)
>> so I am getting somewhere.
>
>In the properties of Settings/Telephony/Extensions/FON1 (or whatever
>connector you've connected your DECT phone to) you should select "fixed
>line" as the default for all calls. It will then always dial out over the
>fixed line, if you don't have any dial rules setup. There is no need to
>enter any prefixes at all; it should just work like a "normale" phone
>system.

<Tim tries new settings> Nice, thanks (again) John!

Now, say I wanted to dial out via the SIP(gate) service ...would that
be the "*121#" in the Internet Telephony ? Internet Numbers field,
Sipgate ?

All the best ..

T i m.

p.s. Is there a particular DECT phone that works well with these boxes
/ SIP in general please (I'm thinking something that has a good
display / flexible phonebook etc)?

p.p.s. Can you have more than one number per Sipgate account or would
I have to create a new complete account for my daughters line?

On Tue, 3 Jul 2007 15:16:51 +0200, "John Miller"
<> wrote:

>>>So if you enable ENUM and you dial a number, it will check if it can call
>>>that number for free and directly (without any conversion from one PSTN
>>>provider to another).
>
>> Assuming all called parties also have enabled ENUM I assume though
>> John?
>
>Yes. That is why we should encourage all SIP callers to setup ENUM for
>their system! Voipuser.org is free and works very well.
>
Makes sense.

Like I said I have registered with them but gave up on the number
allocation stage as I really didn't know what the implications of all
the extra boxes were? ;-(

All the best ..

T i m

On Tue, 3 Jul 2007 15:19:44 +0200, "John Miller"
<> wrote:

>> Yes. That is why we should encourage all SIP callers to setup ENUM for
>> their system! Voipuser.org is free and works very well.
>
>I meant e164.org instead of voipuser.org.
>
>But voipuser.org also works very well

Ah, yes, I've actually registered with both but still trying to put
the right numbers in the right boxes. ;-(

All the best ..

T i m

T i m wrote on 03/07/2007 :
>> - via a SIP address (), if your VOIP provider has linked
>> one to your account
>
> Ok, that one doesn't. I mean. I hear what you say but I'm not sure I
> see the point of doing so?

Think of it like an email address.

If you were with the same ISP as me, in theory, I could email you
simply with the address "tim" and the mail would reach you.

However, if we were on different ISP, I would have to email you using
"tim@yourISP"

The SIP number is specific to your VoIP provider, your SIP address
allows calls to reach you from (any) other network. (SIPGATE have
chosen to block most calls originating off their network) Your PSTN
number allows calls to reach you from the normal phone network.

On Tue, 03 Jul 2007 19:19:48 +0100, Jono <>
wrote:

>T i m wrote on 03/07/2007 :
>>> - via a SIP address (), if your VOIP provider has linked
>>> one to your account
>>
>> Ok, that one doesn't. I mean. I hear what you say but I'm not sure I
>> see the point of doing so?
>
>Think of it like an email address.

Ok ..
>
>If you were with the same ISP as me, in theory, I could email you
>simply with the address "tim" and the mail would reach you.

Ok, understood, like not using the STD code on the land line etc.
>
>However, if we were on different ISP, I would have to email you using
>"tim@yourISP"

Ok again .. routing outside your own network etc ..
>
>The SIP number is specific to your VoIP provider,

But transparent to anyone calling it.

> your SIP address
>allows calls to reach you from (any) other network.

And I don't have that from Sipgate I assume.

> (SIPGATE have
>chosen to block most calls originating off their network)

And is that not typical of other similar spec (free local number /
free incoming PSTN / SIUP calls) SIP providers Jono?


>Your PSTN
>number allows calls to reach you from the normal phone network.

Understood.

Ok, I would like to use what is the most flexible and I'd rather do
that from the beginning. However I have registered with Sipgate and
also have 333 free mins with them to use up and at the moment the only
system simple enough for me to fully comprehend (because they only use
telephone numbers in a 'telephone environment' (as I see it)).

So, if I went with 'another' SIP provider I guess I would get two
'things' from my registration, a STD number for ordinary PSTN users to
call me on and a number linked 'address' for some (all) SIP clients to
use?

So ...

I've used X-Lite to take a PSTN call via Sipgate (using my sipgate SIP
'number').

I've used a DECT phone connected to my Fritz!Box to take a PSTN call
via Sipgate (using my sipgate SIP 'number').

I've used Skype to make and receive calls (using the Skype username,
similar to MSN or X-Talk for in game chat).

How would someone with a stand alone SIP phone call me over VoIP, I
mean, what would they enter on their analogue phone if I only had an
address?

(sorry to be slow) ;-(

All the best ..

T i m

On 03-07-2007, T i m <> wrote:

> On Tue, 03 Jul 2007 19:19:48 +0100, Jono <>
> wrote:
>
>> (SIPGATE have
>> chosen to block most calls originating off their network)
>
> And is that not typical of other similar spec (free local number /
> free incoming PSTN / SIUP calls) SIP providers Jono?

Not paticularily. Voiptalk, Gradwell and Voipfone don't block SIP-to-SIP
calls. Vonage and Sipgate do. It does not correlate with freeness.

> So, if I went with 'another' SIP provider I guess I would get two
> 'things' from my registration, a STD number for ordinary PSTN users to
> call me on and a number linked 'address' for some (all) SIP clients to
> use?

The important item is the SIP address, user@domain. Without it you are
not contactable. At some point a PSTN number has to point to it. The
phone call would not be delivered otherwise.

> How would someone with a stand alone SIP phone call me over VoIP, I
> mean, what would they enter on their analogue phone if I only had an
> address?

This depends on the ATA used and is not straightforward. If your Sipgate
ID was 1112223 I would need to dial , and the
letters present a problem with an analogue phone. It can be done if the
ATA has a quick dial or address book facility. Or, rather tediously,
dial 111222333*217*10*79*23 from the handset. * replaces @ and . on my
ATA.

--
Brian