www.voipstunt.com
Philippe Deleye
Just read on a german newsgroup: another Voip client from the swiss Finarea
group:
www.voipstunt.com
Very similar to voipbuster, voipcheap, netappel.fr etc ...
Only difference: many more countries with FREE calls (to landlines)
And nice to know: a german user just reported that you can start making free
calls WITHOUT any credit on your account (and without disconnect after 1
minute ...)
In the FAQ you can read:
Q-- I want to configure my own IAX/SIP device for calling with Voip Stunt,
is that possible?
A-- It is possible to use your own IAX/SIP device, however we do not support
it. We advise you to use SIP-Discount instead.
This would suggest that SIP is supported (similar to voipbuster), however no
mention of the correct regstrar. Apparently sip.voipstunt.com is not working
....
Anybody found out the correct SIP entry??
Thanks and Happy New Year
Philippe (Belgium)
Jono
Try the voipbuster ones with your voipstunt credentials. (that's how I use
sipdiscount)
Jono
CiscoHeadsetAdapter.com
news:aaYsf.16621$iz3....@text.news.blueyonder.co.uk...
>
> Try the voipbuster ones with your voipstunt credentials. (that's how I use
> sipdiscount)
It's interesting. When I tried to register with the SIPDiscount using the
same credentials, I was denied. Now it works, but kind of strange - one day
I am able to authenticate properly and make a call, another day I am getting
the "Unauthorized" error... have you experienced something like this?
Mike
www.ciscoheadsetadapter.com
Jono
I do occasionally get "unavailable" when I use Sipdiscount. I put it down to
capacity.
paul123
Jono wrote:
> || www.voipstunt.com
I've just signed up, got over 3 minutes on a call from the software,
with no cut off (not like the 1 or 2 minutes limit of
voipbuster/sipdiscount BEFORE putting credit into the account etc?).
Configuring an IAX trunk in Asterisk@home with iax.voipbuster or
iax.voipstunt.com hosts shows the account as registered but when
calling I get the "all circuits are busy now" blurb.... have to
play/experiment a bit more.
Paul
Jono
Yeah, I'm having the same issue with a@h. (I'm actually getting the same on
sipdiscount as well ATM)
I also tried using the IP address that appears in a "netstat -a" when
connected using the softfone to no avail
PeterW
news:1135952882.6...@o13g2000cwo.googlegroups.com:
Charging the account is via a £1.50 a shot 0911 number. Beware that the
code the website gives you doesn't work. You are £1.50 poorer with no
credit!
Peter
paul123
Jono wrote:
>
> Yeah, I'm having the same issue with a@h. (I'm actually getting the same on
> sipdiscount as well ATM)
No issues with sipdiscount here as an iax trunk (today anyway)
>
> I also tried using the IP address that appears in a "netstat -a" when
> connected using the softfone to no avail
I seem to remember reading on a forum post somewhere (can't remeber
where) that voipcheap and netappel were encrypted (and so impossible to
configure a SIP/IAX device) and wondered if that might be part of the
problem?
.....Perhaps not, as I wasn't able to show as registered with
iax.voipcheap trunk on a@h and it is possible with iax.voipstunt.....
just thinking out loud.
As an aside, there seems to be no reference on the voipstunt website to
call time limits before adding credit, as in VB and SD et alia. That's
handy
PeterW
news:Xns973C9DCC93...@192.168.1.250:
>
> Charging the account is via a £1.50 a shot 0911 number. Beware that the
> code the website gives you doesn't work. You are £1.50 poorer with no
> credit!
>
> Peter
>
PS:
It registers to sip.voipstunt.com now with my Voipstunt credentials. The
servers are different from the Spidiscount ones and appear to be hosted by
C&W / Telia .
Peter
Jono
"PeterW" <peterw1...@f2s.com> wrote in message
news:Xns973CAFD87A1...@192.168.1.250...
Thank you, Peter.
I have now got it working also, using asterisk@home.
I can't get an iax connection, though sip is fine.
for anyone that's interested, I created a new SIP trunk with the following
details:
Outgoing:
authuser=voipstuntusername
canreinvite=no
context=from-pstn
fromdomain=voipstunt.com
fromuser=voipstuntusername
host=sip.voipstunt.com
insecure=very
secret=voipstuntpassword
type=peer
username=voipstuntusername
Register string:
voipstuntusername:voipstun...@sip.voipstunt.com/voipstuntusername
Don't forget to dial in the full international format.
NutCracker
{{{{{Welcome}}}}}
Now have it working with my PAP2 ATA. The incoming number they give you seems
a little strange, haven't been able to find out how the number is allocated.
Jono
00801....?
{{{{{Welcome}}}}}
Yeah.
glo...@gmail.com
asterisk gives the error:
-- Got SIP response 415 "Unsupported media type" back from
194.221.62.201
I've used asterisk from the beginning, never have seen -this- error!
IAX would be far preferable.....
PeterW
news:rVetf.17115$iz3....@text.news.blueyonder.co.uk:
They are now accepting Paypal and CC, for minimum of EUR5. The SIP quality
seems a lot better and more reliable than SIPDiscount.
Peter
Jono
Not half, I've even had a three-way call on the go, only slight echo on one
of the legs.
I couldn't get SipDiscount to work at all with asterisk using SIP, it worked
fine with IAX using voipbuster's servers, although since I registered with
voipstunt, my iax trunk for Sipdiscount has stopped working. I'm sure it's
co-incidence.
paul123
Nice job on the a@h sip config - works a treat - thanks!
CrouchScores
two production asterisk servers, using the SIP config given in this
discussion.... Does this work without any money deposited, or do I
have to dump 5 euros into the box?
-- Executing Dial("IAX2/pbx--pod@pod/7", "SIP/+12061234567@voipstunt")
in new stack
-- Called +12061234567@voipstunt
-- Got SIP response 415 "Unsupported media type" back from
194.221.62.211
I've also tried the 001 prefix in place of the +1, same result.
The account works fine with their (ugh) Windows client.
paul123
CrouchScores wrote:
> I'm still getting the 'media type unsupported error'. Have tried from
> two production asterisk servers, using the SIP config given in this
> discussion.... Does this work without any money deposited, or do I
> have to dump 5 euros into the box?
No CrouchScores, no money required....I'm registering and calling out
on Asterisk@home on the Jono's SIP settings - without having paid
anything.....
Jono
"paul123" <pa...@redy.net> wrote in message
news:1136046414.6...@z14g2000cwz.googlegroups.com...
Glad it worked!
CrouchScores
posting!)
Voipstunt demands the use of the G711-A (alaw) codec. I had this
(randomly) disallowed in my asterisk config. Now it's working great!
Happy New Year to the folks here!
Jono
"CrouchScores" <glo...@gmail.com> wrote in message
news:1136070145....@g47g2000cwa.googlegroups.com...
Again, glad it worked!
Can you make use of DTMF, if you receive a call using voipstunt?
I have set up a callback number - I dial a DID which hangs up on me after 2
rings, then asterisk rings me back and asks for a password, which I can't
enter, if I use voipstunt to perform the callback.
CrouchScores
Add the following to the [voipstunt] section of your sip.conf, be sure
to do
a reload, or a 'sip reload' command afterwards:
dtmfmode=inband
Jono
Cheers. Seems OK now.
Daniel
news:rVetf.17115$iz3....@text.news.blueyonder.co.uk...
>
> I have now got it working also, using asterisk@home.
>
> I can't get an iax connection, though sip is fine.
How customisable is asterisk@home ??
Can you set different SIP providers based on the telephone number dialled ??
eg. 01 and 02 numbers via voipstunt.com and a particular country via another
SIP provider, and then 08 numbers via normal PSTN line ??
Thanks in advance,
Daniel
Ivor Jones
"Daniel" <djb...@gmx.net> wrote in message
news:43b7f...@x-privat.org
I can do that on the AVM Fritz!Box Fon.. A lot easier to set up and
doesn't require a PC. Asterisk is ok if you like that sort of thing but I
want a box that I can plug in and leave alone.
Ivor
CrouchScores
number dialed is one of the most basic features. Once you realize what
you can do, you'll never let go of it.
True, some hardware devices have -some- flexibility in this regard, but
nothing compared to what you can do with asterisk.
Ivor Jones
"CrouchScores" <glo...@gmail.com> wrote in message
news:1136144305....@g44g2000cwa.googlegroups.com
> Asterisk is totally configurable. Choosing VoIP
> providers based on the number dialed is one of the most
> basic features. Once you realize what you can do, you'll
> never let go of it.
>
> True, some hardware devices have -some- flexibility in
> this regard, but nothing compared to what you can do with
> asterisk.
But Asterisk requires a computer. I don't want to use one. I want a
hardware box. The one I use does everything I want it to do, I can see
nothing so far that would persuade me to abandon it for a PC-based system
with all its inherent problems.
Ivor
Jono
Asterisk does have its uses:
Call recording;
Call Data Records;
Visualization of channels/phones in use;
Limitless number of providers;
Multiple call handling.
.....to name a few.
Ivor Jones
"Jono" <no...@none.za.uk> wrote in message
news:vVWtf.18258$iz3....@text.news.blueyonder.co.uk
> Ivor Jones wrote:
> > > "CrouchScores" <glo...@gmail.com> wrote in message
> > > news:1136144305....@g44g2000cwa.googlegroups.com
> > > > Asterisk is totally configurable. Choosing VoIP
> > > > providers based on the number dialed is one of the
> > > > most basic features. Once you realize what you can
> > > > do, you'll never let go of it.
> > > >
> > > > True, some hardware devices have -some- flexibility
> > > > in this regard, but nothing compared to what you
> > > > can do with asterisk.
> > >
> > > But Asterisk requires a computer. I don't want to use
> > > one. I want a hardware box. The one I use does
> > > everything I want it to do, I can see nothing so far
> > > that would persuade me to abandon it for a PC-based
> > > system with all its inherent problems. Ivor
>
> Asterisk does have its uses:
>
> Call recording;
No requirement.
> Call Data Records;
Depends what you want, my system records most call details.
> Visualization of channels/phones in use;
The PABX does that.
> Limitless number of providers;
I have capability for up to 10 which is more than enough.
> Multiple call handling.
I can have up to 4, again more than enough.
Ivor
Daniel
news:41qdvoF...@individual.net...
>> telephone number dialled ??
>> eg. 01 and 02 numbers via voipstunt.com and a particular
>> country via another SIP provider, and then 08 numbers via
>> normal PSTN line ??
>
> doesn't require a PC. Asterisk is ok if you like that sort of thing but I
> want a box that I can plug in and leave alone.
Hi Ivor,
Thanks - That sounds ideal for me (and probably a lot easier too).
Do you know anyone that sells them in this country ??
Thanks
Daniel
Ivor Jones
"Daniel" <djb...@gmx.net> wrote in message
[snip]
> Hi Ivor,
>
> Thanks - That sounds ideal for me (and probably a lot
> easier too).
> Do you know anyone that sells them in this country ??
Yes, Sipgate (www.sipgate.co.uk and click on "shop") sell them, but
they're not the cheapest devices around. They do work well though. I wrote
a review on it for a forum that I'm involved in setting up, it's not quite
ready for public consumption yet, but I'll send you a copy if you're
interested. The AVM website (www.avm.de/en) gives most of the details you
need though.
>
> Thanks
> Daniel
Scope
> On Sun, 1 Jan 2006 23:54:37 -0000, "Ivor Jones"
> <iv...@despammed.invalid> wrote:
>>Yes, Sipgate (www.sipgate.co.uk and click on "shop")
> "Do you know anyone that sells them in this country ??"
Dude, you say alot of crap, but this takes the cake. Give it a rest will
you?
Thomas Kenyon
> "Jono" <ho...@home.co.uk> wrote in message
> news:rVetf.17115$iz3....@text.news.blueyonder.co.uk...
>
>>I have now got it working also, using asterisk@home.
>>
>>I can't get an iax connection, though sip is fine.
>
>
> How customisable is asterisk@home ??
>
> Can you set different SIP providers based on the telephone number dialled ??
>
providers for any pattern or range of numbers and you can even set
different providers/account per-extension you use.
Pretty much anything you can think of can be done.
(including neat things like, if someone calls you, it can tell you which
company name/number they have called you on whilst the caller still
hears a ringtone, or recording all the phone calls to a particular
number/number range).
> eg. 01 and 02 numbers via voipstunt.com and a particular country via another
> SIP provider, and then 08 numbers via normal PSTN line ??
>
This is very very simple to do. (Okay I've not really used the
front-ends on a@h, but it;s bound to be simple, it is with plain asterisk).
> Thanks in advance,
> Daniel
>
Thomas Kenyon
>
> I can have up to 4, again more than enough.
>
> Ivor
>
He wasn't suggesting you should be forced to use it, just that it is
more flexible than any embedded-system based device.
It's horses for courses, if you need the extra flexibility, then it's
worth paying for. If you don't then don't.
Thomas Kenyon
And sipgate do indeed sell them in this country.
As far as I can see, they are the cheapest supplier for the UK.
Ivor Jones
"Thomas Kenyon" <ukt...@sanguinarius.co.uk> wrote in
message news:LC7uf.64317$7p5....@newsfe4-win.ntli.net
> Ivor Jones wrote:
> >
> > I can have up to 4, again more than enough.
> >
> > Ivor
> >
> Now now children, play nicely.
> He wasn't suggesting you should be forced to use it, just
> that it is more flexible than any embedded-system based
> device.
It certainly is, but I don't need it. I never said he was trying to force
me to use it, merely that the equipment I have is more than adequate for
my needs.
> It's horses for courses, if you need the extra
> flexibility, then it's worth paying for. If you don't
> then don't.
Indeed.
Ivor
Jono
|| Ivor Jones wrote:
|||
||| I can have up to 4, again more than enough.
|||
||| Ivor
|||
|| Now now children, play nicely.
We were.
We were discussing the pros & cons, that's all!
Jono
|| if someone calls you, it can tell you
|| which company name/number they have called you on whilst the caller
|| still
|| hears a ringtone,
That'd be good. Can you tell me where I can read about this feature?
Thomas Kenyon
www.voip-info.org and iirc in the remarked text in queues.conf (since it
really does it based on queing).
Of course, an alternative is asterisks rather neat way of being able to
manipulate CLID information, so if your phone supports textual caller-id
you can get it to simply display the company name based on the number
the caller is dialling instead of their phone number (can be done
entirely in extensions.conf), or even have a blacklist of people you
don't like come up as a particular word/string.
Jono
Cheers.
paul123
> Dex...@blueyonder.co.uk wrote:
Sipgate are based in Naziland aren't they ???????
Oh dear, there's always someone who has to lower the tone..... Just
'cos one has an issue with Sipgate doesn't mean one has to descend to
xenophobic comments!?!? The continual sipgate bashing does get a touch
boring.
(As an aside, I even set myself up an account to try Sipgate out. Set
up was easy and everything works well. I have absolutely no issues with
them. Furthermore, I've got another working UK incoming number!)
Ivor Jones
"paul123" <pa...@redy.net> wrote in message
news:1136204003.3...@z14g2000cwz.googlegroups.com
Try and ignore him, I have. I had endless arguments with him, but my New
Year's Resolution is not to respond to him any more. (I'm responding to
you, not him, ok..?!)
He isn't even a Sipgate customer any more, so I really don't understand
his beef.
This is the last comment I will ever make in relation to any of his
postings. Treat him as the troll he is and hopefully he will go away and
annoy someone else.
Ivor