On Fri, Sep 01, 2006 at 08:11:50AM -0500, Rich Adamson wrote:
> > I have a question on configuration of SPA3000 with asterisk.
> > 1. I want all incoming calls are redirected from SPA3000 to my
> > asterisk server.
> > 2. Asterisk then should direct this call to my SIP phones (including
On Sat, Aug 26, 2006 at 01:35:36AM +0800, Sam Tam wrote:
> WE can provide you with budget GSM Gateway if you are interested?
which is commercial nope? wrong list again? could have been private
Email?
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715
On Sun, Aug 20, 2006 at 04:06:06PM -0400, John Novack wrote:
> Be aware that Cellsocket is a dead end.
> WHP wireless is gone
> When these are gone, they are gone
> They do work, however.
> GSM versions do not forward CLID
> GSM requires a "#" to end the dial string
> They do work however
Anythin
On Mon, Aug 07, 2006 at 04:26:28PM -0700, Elpidio Ramos wrote:
>I am tryin to make the asterisk work on my linux box but when I launch
>asterisk I get several warnings.
>I am using Fedora Core 3 (just installed and not yet updated with the
>latest)
>Asterisk Business Edition
>
Further UK prompts have been added to the www.tel.net site.
There's now a complete list of UK (England, Scotland, Wales) Counties,
Towns and London Boroughs, as well as the standard 1.2 base and
additional Asterisk sounds.
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 /
On Thu, Jul 27, 2006 at 10:06:35AM +1000, Edwin Groothuis wrote:
> Hello,
> Does anybody have experience with the Quad T1/E1 PRI cards in an
> HP DL380? Just a "yes it works fine" or a "never again" is enough :-)
I've had a couple of Digium cards in a DL360 working fine, no problems
at all.
Ste
The telco I used had a fault with the switch and the PRI (E1) went down.
This seems to have caused Asterisk (1.2.10) to crash. Latest zaptel and
libpri.
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo
On Wed, Jul 19, 2006 at 07:05:55PM +0800, Sam Tam wrote:
> WE have found this type of phone work better than E61
> http://cyber-telecom.net/shop/product_info.php/cPath/21/products_id/31
This is not the .biz list, Sam works for Cyber Telecom !!!
So it probably does work better, hm
Steve
--
On Tue, Jul 11, 2006 at 06:53:14AM +1000, Eric Bishop wrote:
>What us meant by "blended rate"?
It generally means a rate that the provider sets that is fixed even if
the provider is charged different rates
i.e. BT International provide blended fixed and mobile termination
rates, even though
On Mon, Jul 10, 2006 at 03:14:27PM +0800, Sam Tam wrote:
>Have a look at cyber-telecom.net. CT-GSM-1000 seems to be one of the
>cheapest GSM Gateway that you can buy right now.
Which is biz, and Sam works for Cyber-Telecom ...
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)2
On Sat, Jul 08, 2006 at 01:55:00PM +0100, Thomas Kenyon wrote:
[snip]
> I thought that the line would now go through talktalk (It is an LLU
> service after all).
> FWIW, the same thing happened to me with a line that moved to bulldog.
In the UK BT still own 85% of all copper into premises. Ofcom
On Sat, Jul 08, 2006 at 10:59:49AM +0100, Angus Comber wrote:
> I had an Asterisk installation working fine for CallerID on BT analog lines
> using a Digium analog 4 port card. However, user switched to TalkTalk
> without telling me and CallerID no longer works. However, if you connect a
> UK Ca
On Tue, Jun 27, 2006 at 03:10:38PM -0700, Steven Ringwald wrote:
> Mike Fedyk wrote:
> >Is anyone else getting messages from the lists.digium.com mail server
> >with errors about a mail loop?
> >I've been getting this for the last few weeks, but I don't have any
> >list software on my server. A
New versions of the Male UK English sound files are now available.
We believe these are complete for v1.2.x of Asterisk and v1.2.1 of
Asterisk-sounds.
The LouisLouis song is missing (which is US anyway) and 7 seconds of
silence, but everything else should be there. The vm voice prompts are
now co
On Thu, Jun 22, 2006 at 05:47:47PM +0100, Steve Kennedy wrote:
> Is it possible to get Joburg DIDs (probably need 4 at the moment), to be
> delivered via SIP preferrably to UK.
> If it's legal, please send pricing.
And that should have gone to the biz list, sorry.
Steve
--
NetT
Is it possible to get Joburg DIDs (probably need 4 at the moment), to be
delivered via SIP preferrably to UK.
If it's legal, please send pricing.
Thanks
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gi
On Thu, Jun 08, 2006 at 01:42:25PM -0600, Colin Anderson wrote:
> Just hit Slashdot:
> http://slashdot.org/article.pl?sid=06/06/08/1725215&threshold=1
> http://money.cnn.com/2006/06/07/news/companies/pluggedin_fortune/index.htm
> >From TFA:
> "But what's really cool about what will happen in Cardi
On Tue, Jun 06, 2006 at 03:01:17PM -0700, trixter aka Bret McDanel wrote:
[snip]
> personally given the genre of the game a brittish female with only a
> slight accent might do better than allison. Of course if you are clever
> (set(language) or whatever) and have the user select the voice that
>
I'd like to announce that the UK Male English Voices are now up on
http://www.tel.net/
There's a complete set of base sounds and additional sounds (it should
be complete compared to current Asterisk and Asterisk-Sounds-1.2.1).
There's also a set with the word 'pound' replaced by 'hash' for both t
I'd like to announce that the UK Male English Voices are now up on
http://www.tel.net/
There's a complete set of base sounds and additional sounds (it should
be complete compared to current Asterisk and Asterisk-Sounds-1.2.1).
There's also a set with the word 'pound' replaced by 'hash' for both t
On Fri, May 26, 2006 at 04:58:34PM +0100, Paul Redstone wrote:
[snip]
> I found a wiki which said that the DDI numbers we want as caller IDs need to
> be
> flagged as allowed CallerID number - this is done by BT - but BT do not seem
> to
> understand this.
> Also our old local exchange was a S
On Tue, May 23, 2006 at 10:37:07AM +0100, Joao Pereira wrote:
> Hello
> Just 2 ideas:
> How cares about GSM WiFi handovers? I just want to make free VoIP calls.
Lots of people do, as they want to use GSM which has ubiquotous coverage
and then utilise a more cost effective solution like WiFi when
On Tue, May 23, 2006 at 02:50:33AM +0800, Sam Tam wrote:
> Well it is incorrect to say that.
> In places like USA or London, a lot of areas are covered by local wifi
> providers, some are free, some aren't.
> You then can use them to drop some of your local or international calls
> cheaply by usi
On Mon, May 22, 2006 at 08:14:19AM -0500, Eric ManxPower Wieling wrote:
> If you want to roam between GSM and WiFi while on a call, the GSM
> carrier is going to have to support it.
There is a protocol for this (UMA), however few operators as yet support
it.
T-Mobile offer a webnwalk tarrif (un
On Thu, May 18, 2006 at 05:42:05PM -0600, Kris Cote wrote:
> Can I trouble you for a copy as well?
Why don't you put it up somewhere, if you need space I can put it on
tel.net ?
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
On Tue, May 02, 2006 at 05:39:53PM +0100, Chris Bagnall wrote:
[snip]
> I think disabling asterisk from getting caller ID off an analogue line
> improves its answering speed considerably. Of course, if you want CLID info
> off your analogue line (and are paying BT for the privilege), you may not
>
Is there a list of sounds (base - as with Asterisk itself, and
additional) for the 1.2 release. As in a list with what the content of
each file is.
There's a list for 1.0.7 on the wiki, but that seems woefully out of
date.
Any help appreciated.
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
U
On Fri, Apr 21, 2006 at 11:23:16AM -0400, Andrew Latham wrote:
> D-link has a nice one, optional 5 year warranty on some of the
> commercial stuff
Though beware, some of the D-Link ones only have half the ports with
PoE.
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 7993261
In sox terms is SLIN .ul (as in unsigned linear).
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com
___
On Tue, Apr 18, 2006 at 06:17:22PM -0400, Tzafrir Cohen wrote:
> Any chance of a better license?
> http://people.debian.org/~evan/ccsummary.html describes why Debian
> considrs version 2.0 of the same license problematic. The reasons
> mentioned are pragmatic reasons (of the sort of: the license m
On Tue, Apr 18, 2006 at 08:51:02PM +, Tony Mountifield wrote:
> In article <[EMAIL PROTECTED]>,
> Steve Kennedy <[EMAIL PROTECTED]> wrote:
> > I have made available the base Asterisk sounds (i.e. as included with
> > Asterisk) in male UK English in gsm forma
I have made available the base Asterisk sounds (i.e. as included with
Asterisk) in male UK English in gsm format. They are released under the
Creative Commons Attribution 2.5 License.
They are available as a single compressed tar file via: -
http://www.tel.net/
The Asterisk sounds (as in the sepe
On Mon, Apr 17, 2006 at 03:15:06PM -0700, Maxx Lobo wrote:
> Any recommendations for a VoIP provider in the UK?
> I have a few guys in a field office in the UK with SIP phones and a VPN
> tunnel back to a working Asterisk setup in the US. The Asterisk setup
> has an IAX trunk with TelaSIP/VoipXp
Is there a way to differentiate between a SIP address which hasn't
registered (but is within sip.conf) and one that's not there at all
(i.e. not in sip.conf) using a straight dialplan.
I'd like to differentiate actions depending the state of a SIP device
and whether it's in my config or not (if th
Trying to get an Asterisk system v1.0.10 as per Junghans site, using
Bristuff/qozap with a quadBri card and 2 x Digium TDM400P (with phone
ports) working.
Used zaptel1.10 and the libpri that Junghans recommend.
Everything seemed to patch ok and build, though the qozap did come out
with some error
On Sun, Apr 02, 2006 at 12:55:13PM -0500, Kevin P. Fleming wrote:
> Steve Kennedy wrote:
> > Each channel needs TWO licenses, one for each way (I think).
> Nope. The encoder/decoder licenses are counted separately, and each
> license you purchase entitles you to one encoder and on
On Sun, Apr 02, 2006 at 09:32:09AM +1000, RumaTech wrote:
> Sorry, I was "out of action" for some time.
> I am using Voipstnt to plae calls to USA/Canada and I bouhj 1 copy og G.729.
> This was mainly to get one of the local Australians VoIP providers working.
Each channel needs TWO licenses, one
On Tue, Mar 28, 2006 at 01:20:06AM +0300, Tofik Suleymanov wrote:
> How to reproduce this bug (?) :
> 1. register sipura spa2 with 2 lines on asterisk.
> 2. use first line to call somewhere.
> 3. while using first line try to call from second line somewhere else
> in 3 step i hear fast busy tones
I'm using a Digium TE411P connected to a UK switch (EuroISDN).
Everything is working, but if I dial a busy number (from SIP) is seems
to stay busy until I hang up, even though the dial-plan drops through
some other stuff using CALLSTATUS variable (i.e. S-BUSY), none of the
timeouts come into play.
On Thu, Mar 23, 2006 at 01:48:16PM +0100, Tomislav Parina wrote:
> In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
> > I believe the OP wants to use GSM handsets as extensions, like running
> > your own localized GSM network. That's not the same as using a GSM
> > terminal to connect A
On Tue, Mar 21, 2006 at 10:57:06PM +0100, Francesco Peeters (Asterisk) wrote:
> > Why I have 'Format for call is ulaw'? I'd like to have alaw but keep ulaw
> > to
> > accomodate errors in various configurations (if any, not here!).
> EuroISDN uses uLaw, so Asterisk does as well, because it doesn't
On Sat, Mar 18, 2006 at 10:16:27PM +1100, James Harper wrote:
[snip]
> Ah. More complicated than I'd hoped but not more than I suspected :)
> So the product that can accept gsm phone registrations and calls and
> trunk them to asterisk via E1/TDMoE/TDMoIP/SIP/IAX is still wishware? Oh
> well. I gu
On Sat, Mar 18, 2006 at 04:49:53PM +1100, James Harper wrote:
> > I believe the OP wants to use GSM handsets as extensions, like running
> > your own localized GSM network. That's not the same as using a GSM
> > terminal to connect Asterisk to the cellular network.
> Correct!
> > IP Access makes s
On Thu, Mar 16, 2006 at 11:28:34PM -, Magnus Kelly wrote:
> Has any one found any issues (bug?) with ver 1.2.5 as CallerID generated on
> an outgoing FXS port (to the handset) fails when UK tones are used, with a
> message 'Didn't finish Caller-ID spill. Cancelling.'
> Any tips on getting this
Are there any step by step instrunctions on how to install drivers and I
guess bristuff for this card?
Just need to use it to handle voice on 2 BRI circuits (UK) then utilise
with Asterisk and some Digium cards handling POTS phones (and some VoIP
out the back).
It's the EICON card stuff and how t
On Mon, Mar 13, 2006 at 07:38:01PM -0500, Watkins, Bradley wrote:
> That depends on what you mean by default. The supplied sample
> extensions.conf contains the priorityjumping=no by default, but if this
> parameter is absent then the default is to jump n+101.
OK, that explains it, just wonderin
On Mon, Mar 13, 2006 at 04:25:20PM -0800, Gabriel Afana wrote:
> I think in 1.2.x, this jumping feature was disabled by default.
So should priorities still increase when the Dial returns busy (i.e.
jumping to priority + 101)?
Or should something else be done?
Steve
--
NetTek Ltd UK mob +44-
I've been trying to use a set-up whereby I have several TA's connected
to an Asterisk server (1.2.4) and they act like they're in a hunt-group
i.e. try the first, if busy jump to the next etc.
in my extensions.conf I had something like
[inbound-trunk]
exten => 441234123456,1,Dial(SIP/s1a,20,r)
ext
On Wed, Mar 01, 2006 at 12:45:45AM +0800, Sam Tam wrote:
> I think I have seen a post about that before. But can't find it
> again
> Can some people light me up with the detail
GSM extenders I don't think are legal in the UK, except if
installed/operated by a GSM network operator (
On Fri, Feb 24, 2006 at 07:17:52AM -0600, Rich Adamson wrote:
> Its my understanding the cell phone coordinates are sent to the cell phone
> provider and their equipment reads (and holds) that data. Its not part
> of any data available to you in any form unless you talk to the cell
> provider and
On Tue, Feb 21, 2006 at 06:16:06AM -0500, Alexander Burke wrote:
> Hello, Steve!
> At 03:55 AM 02/21/2006, you wrote:
> >ztdummy was only used for timing. Linux 2.6 provides this function in
> >the kernel and I assume Solaris already has timing functions there.
> Page 36 of Asterisk: The Future Of
On Mon, Feb 20, 2006 at 06:24:16PM -0500, Alexander Burke wrote:
> I really appreciate the replies I've gotten about this so far
> (especially the support for wanting to run it on Solaris!).
> The core issue seems to have been missed, though -- is there any way
> to run a complete Asterisk solut
On Mon, Feb 20, 2006 at 03:15:52PM +, bails wrote:
> Anyone know if asterisk supports q931 85 in the uk?
Nope, it only supports Q.931 110 (which is EuroISDN). 85 is UK ISDN,
most providers can set the line to 110, but you may have to ask for it.
Marconi System X switches (as used by BT, THUS
On Tue, Feb 21, 2006 at 12:17:43AM +1100, Mark Edwards wrote:
> At 06:33 AM 02/20/2006, you wrote:
> >Please forgive the question, but what is the rationale behind using
> Solaris
> >over Linux as an asterisk hosting platform?
Solaris is also a "supported" OS (well if you pay for it). It's also
On Fri, Feb 17, 2006 at 01:58:26PM -0500, Rusty Dekema wrote:
> I don't think it takes a great leap of the imagination to infer that
> Mr. Kennedy is in fact having the problem he describes and that,
> although it may not be 100% standard and correct usage, the question
> mark at the end of his se
I have some Digium licensed Digium codecs, but when making a call and
transcoding the call is only heard in one direction?
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PR
I'm having trouble with Asterisk-1.2.4 negotiating codecs with a Sipura
3000 which is running the latest v3 firmware.
The SPA-3K seems to use the "preferred" codec only and doesn't
negotiate? The SPA is set to no in "use only preferred codec".
Does anyone know if Sipura will support gsm at some p
On Wed, Feb 15, 2006 at 01:29:10PM +, bails wrote:
> We have 3 existing switches interconnected via dpnss, we need to
> integrate asterisk with these switches via a dpnss link.
> Any suggestions?
> also does anyone have a link to the differences between isdn30 and dpnss.
Get a DPNSS to somet
On Tue, Jan 24, 2006 at 09:13:45AM +, scott wrote:
> Does anyone know a UK Voip Proivder that will give me more than 1 telephone
> number and point it to my sip account.
> www.SipGate.co.uk are great but they only allow 1 telephone number per user,
> you can register another telephone numbe
On Tue, Jan 10, 2006 at 05:17:02PM +0400, Jean-Michel Hiver wrote:
[snippage]
> I dunno... it looks like a cell phone, except it's not one. It would be
> nice if it was a dual GSM / wifi phones which transparently switch to
> VoIP when you have a strong enough signal. This way, it would provide
On Thu, Jan 05, 2006 at 11:20:24PM +0100, Hans Witvliet wrote:
> On Thu, 2006-01-05 at 15:57 -0500, Cory Andrews wrote:
> > SICPE has a new product called the GSM Call Director that may be of
> > interest
> > to GSM enthusiasts.
> > http://www.sipcpe.com/fx300GSM.html
> Looks nice, doing triple
On Fri, Jan 06, 2006 at 06:48:27PM +0400, Jean-Michel Hiver wrote:
> >However there are some disadvantages, the main being you cant set CLI of
> >the outgoing call as it will always be tied to the SIM of the mobile
> >terminal.
> That's true. You can however choose to mask the caller ID.
Yup, for
On Fri, Jan 06, 2006 at 01:23:26PM -, Chris Bagnall wrote:
> > I don't get it. What is the advantage of using a GSM gateway?
> > VOIP calls are pretty inexpensive as they are now.
> It largely depends on the country you're calling. Here in the UK, calls to
> mobiles are maintained at an artif
On Fri, Dec 30, 2005 at 08:22:27PM +, Ron Wellsted wrote:
> Within the UK, Number Portability between providers of the same type of
> service is a legal requirement. Since we charge differently for calls
> on landlines and mobiles, you cannot port mobile numbers to landlines or
> landlines t
On Tue, Dec 20, 2005 at 03:50:27AM -0800, Luigi Rizzo wrote:
> On Tue, Dec 20, 2005 at 01:41:30PM +0200, Tzafrir Cohen wrote:
> ...
> > And this is bad for us. With Gizmo we can talk. With google talk we have
> > stand a chance of talking. But we're blocked from Skype.
> since you cite it, what co
On Wed, Nov 02, 2005 at 12:31:48PM -0500, Adam Moffett wrote:
> I have no experience in the matter whatsoever ;)
> But, I can say that long distance phone calls (non-voip) are sometimes
> carried over sattelite when fiber is not available.
> It must be possible for voip, but the latency and jitt
On Fri, Oct 28, 2005 at 05:43:03PM +0100, David Cook wrote:
> You might want to investigate a Nokia 22
> (http://europe.nokia.com/nokia/0,8764,56024,00.html). This provides a single
> GSM line which is interfaced to the PBX by an anlogue trunk/extension. From
> memory they cost around £100-150.
On Fri, Oct 28, 2005 at 05:40:05PM +0100, Charles Trevor wrote:
> > Well, the major incumbent is BT.
> > Are you sitting down ?
> > Installation :
> > Per channel 1 year contract 3/5y contract 3/5y+commitment
> > First 15 channels (min 8)GBP 125 GBP 80GBP 0
> > 1
On Fri, Oct 28, 2005 at 04:40:07PM +0200, Dave Cotton wrote:
> On Fri, 2005-10-28 at 16:22 +0200, Tomasz Chmielewski wrote:
> > looks interesting.
> > do you know by chance how much such a single-cell box cost (more or less)?
> I found it here http://www.thehightechstore.com/plugcell.html
> at 295
On Wed, Oct 26, 2005 at 03:33:16PM +0100, Mark Ackroyd wrote:
> > You should also ensure the PRI is really configured for EuroISDN, many
> > BT PRI's are actually UK ISDN which Asterisk doesn't support (it's an
> > older version).
> I had a problem along these lines, when I first started with aste
On Wed, Oct 26, 2005 at 02:17:11PM +0100, Chris Shucksmith wrote:
> I have a shed load of UK (BT) analogue equipment at our office (18
> phones, 6 faxes) hooked up via structured cabling to an old Avaya
> ArgentOffice phone system. The on-its-way-out phone system has an
> IDSN30e PRI interface
On Tue, Oct 11, 2005 at 02:21:03AM -0700, trixter http://www.0xdecafbad.com
wrote:
[snip]
> > Also you have to know who's terminating it. You can make an assumption
> > re BT termination, but directly connected businesses may use another
> > telco with different termination rates etc.
> > A lot o
On Tue, Oct 11, 2005 at 02:29:04AM -0700, trixter http://www.0xdecafbad.com
wrote:
> Yeah but that still doesnt answer the fundamental question. While they
> do have who owns stuff they do it based on prefix (ie 44 871 59 is
> pipemedia) but it doesnt tell you how many digits are past that (5 mo
On Tue, Oct 11, 2005 at 07:53:09AM +0100, Chris Bagnall wrote:
> For the UK, your most accurate source of info is probably the first few
> pages of the BT Phone Book (delivered free to all UK homes/businesses).
> There's quite a comprehensive list of what each number range is for, how it
> breaks
On Mon, Oct 10, 2005 at 10:12:58PM -0700, trixter http://www.0xdecafbad.com
wrote:
> I was wondering if anyone has put together a comprehensive list (that is
> reasonably maintained) that lists country codes, landline numbers,
> mobile numbers, etc. The particular requirement is for a dialplan t
On Sat, Oct 08, 2005 at 08:43:07PM +0300, Tzafrir Cohen wrote:
> On Sat, Oct 08, 2005 at 11:59:04AM -0400, Mike M wrote:
> > On Sat, Oct 08, 2005 at 09:20:07AM -0400, Paul wrote:
> > > Closed source might delay the cracker but it also delays pre-crack and
> > > post-crack countermeasures.
> > Wha
On Fri, Sep 30, 2005 at 10:19:59AM +0200, Amaury BOSSE wrote:
>Are there patents rights applicable to France?
Yes, most of the world.
>Which licence could I use and how many ones are required (only one per
>phone or also for voicemail and MOH)?
One per translating service (concurren
On Mon, Sep 12, 2005 at 06:00:25PM -0700, Matt wrote:
> anyone knows how skype provide world wide call service to regular phones by
> voip at such low rate?
> is this by partnerships with various * isps?
Skype do deals with local telcos, they use G.729/SIP for SkypeIn/Out
between them and the loc
On Mon, Sep 12, 2005 at 05:38:31PM -0400, Race Vanderdecken wrote:
> Hmmm,
> Let me see.
> "Skype has 54 million registered users that the online retailer
> eBay can add to its marketing arsenal. The initial payment of $2.6
> billion values those registered users at just more than $48
On Mon, Sep 12, 2005 at 07:30:56PM +0100, Iqbal wrote:
> 70 million users, now how many of these are ALREADY ebay customers.
> Google never made a succes out of any other thing other than search and
> that will remain the case, companies never do, they are usually good at
> what they started at
On Wed, Jul 27, 2005 at 09:03:19PM -0700, Michael D Schelin wrote:
> Hello everybody, for all of you that have searched for a real fax
> solution, look no further. We now have T38 faxing. Please contact me for
> more information.
OK, I'll bite ... please send more info
Steve
--
NetTek Ltd Fax
On Sun, Jul 24, 2005 at 11:54:17PM -0700, Storm D. J. Petersen wrote:
>I don’t know if I have the same experiences. Usually my Skype
>calls are very garbled at first. I find that my G729 Asterisk calls
>are better quality. You can try using ULAW if you have the bandwidth.
>It. mig
On Thu, Jul 07, 2005 at 03:42:54PM -0400, Mark Phillips wrote:
> My take on this is that they are protecting themselves against fraud.
> Discounting the freefone numbers for a while, the "national rate"
> numbers are charged at variying rates and so how is a company to know
> just what they are
On Sun, Jul 03, 2005 at 11:31:50PM +0100, Tony Hoyle wrote:
> Mike Dent wrote:
> >I guess one big question is which type of circuit to use, ADSL in the
> >UK is only 256kbs upstream,
> >some providers do bonding but I'm not sure its supported fully by BT :(
> >The other option is SDSL which is not
On Wed, Jun 29, 2005 at 02:05:11PM -0500, Eric Wieling aka ManxPower wrote:
> Yes, but if you know that, then I'm sure you know that the terminating
> carrier almost always looks up the name themselves and set the name to
> whatever is in the name/number database.
In te SS7 world, CLI will alwa
On Wed, Jun 29, 2005 at 09:27:34AM -0700, Bryce Chidester wrote:
> The CallerID that is seen by others on calls originating from your
> PRI is set by your PRI provider; you have no control from Asterisk
> about this as it gets overridden by the provider. You must contact
> your carrier and a
On Fri, Jun 10, 2005 at 07:06:39PM -0400, Iassen Hristov wrote:
> > IMAP vs. Exchange
> > I would be very weary of using IMAP against an Exchange Server. I have
> > not touch it for years but IMAP and Exchange did not play together
> > really well back then.
> > Has anyone actually used real IMAP
On Fri, Jun 03, 2005 at 10:14:58AM -0500, Jay Milk wrote:
> Ahhh... Sneaky. Because of the special billing agreements on NCFA
> numbers, there's bound to be a lower limit to how these calls are
> priced. I doubt BT gives sipgate (or any other VOIP provider) a
> signigicant discount on these call
On Tue, May 17, 2005 at 10:45:52PM +0800, Ronald Wiplinger wrote:
> Skype is very succesfsfull and get more and more users, ... we can
> ignore them, accept them or do something,...
> My suggestion is that we try to do something, ...
> If we would peer to each other, than we get soon also a great
On Mon, May 16, 2005 at 03:51:28PM +0100, Mike Dent wrote:
> Hi,
> I'd be interested in comments from any users of the vonage service in the UK?
> http://www.vonage.co.uk is the website.
> Where are the servers located, traceroute would be useful.
> What is the general reliability like?
No idea r
On Wed, May 11, 2005 at 05:40:57PM +0200, Dave Cotton wrote:
> On Wed, 2005-05-11 at 10:09 -0500, Andrew Latham wrote:
> > Naming Conventions for Asterisk Hostnames, .
> For an internal historical reason all ours come from the legends of
> Robin Hood. I used to work with a bunch of Lord of t
On Fri, May 06, 2005 at 12:46:19PM -0700, trixter http://www.0xdecafbad.com
wrote:
> On Fri, 2005-05-06 at 15:38 -0400, list wrote:
> > Excuse my ignorance but whats an NCFA number?
> its a national rate number, in the UK it doesnt matter where you call
> from the fee is the same.
> +44 870 340 4
On Thu, Apr 07, 2005 at 10:23:50PM +0100, Gavin Hamill wrote:
> On Thursday 07 April 2005 22:17, Alex Vishnev wrote:
> > Magnus,
> > Also, compression gives voice recognition quite a challenge,
> > as the speech samples arriving at the voip voice recognition engine is not
> > the same as it was sp
On Wed, Jan 12, 2005 at 05:30:31PM -, Ben Merrills wrote:
> Most UK phone companies (i.e. BT or the smaller regional carriers) all
> use SS7, everywhere! For the most part they don't accept VoIP
> termination (although I think BT might have some facilities for this).
> So they very much try an
On Wed, Jan 12, 2005 at 05:30:31PM -, Ben Merrills wrote:
> We have the problem that our telecoms provider deals mainly in SS7 (C7,
> and it seems most in the UK do). For us to take EuroISDN off them, with
> the same features as SS7, we have to be put through a protocol
> converter, now this i
On Tue, Jan 04, 2005 at 10:13:27PM +1100, Eric Bishop wrote:
> I really am at my wits end about this one. Some people report this
> card and server working fine while others (like myself) can't get it
> going no matter what. I have been told by the Digium distributor in
> our country that this car
On Thu, Dec 09, 2004 at 07:36:41AM -0500, Tony Nichols wrote:
> On Wed, 08 Dec 2004 22:32:16 -0600, Andrew Aken <[EMAIL PROTECTED]> wrote:
> > Does anyone have any experience with running asterisk on multi-processor
> > computers (dual or quad)? Does asterisk on the latest Linux distros take
> > a
On Wed, Dec 08, 2004 at 03:54:10PM -0600, Steven Critchfield wrote:
> On Wed, 2004-12-08 at 19:44 -0200, Andre Remor wrote:
> > Hi,
> > I'm looking for a SIP client for Symbian OS...
> > Someone known one? (free or not)
> Unless Symbian has branched off of cell phones, I doubt it. SIP on a
> cell
On Tue, Dec 07, 2004 at 09:44:59AM -0500, Steve Totaro wrote:
> there was a website on the list recently that allowed you to enter text (up to
> 50 words) and it would create a wav file with various voice options. does
> anyone remember what it was? rapsody something or another.
I think it was
On Sun, Dec 05, 2004 at 02:30:17PM -0500, Greg Boehnlein wrote:
> > > Tell me which one can get me access to the LinkSys Linux using SSH?
> > > Does Satori has this feature? I am not so concerned with Voice Shaping
> > > and QOS at this time, but more interested in converting this into a
> > > L
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