On Thu, May 26, 2011 at 02:28:31PM +0100, Steven Howes wrote:
On 26 May 2011, at 14:09, Ishfaq Malik wrote:
Does anyone know if there are any free UK accented English sounds packs?
http://www.tel.net
Not perfect, but damned near :)
If anything's missing please let me know and I can get
On Mon, Sep 13, 2010 at 12:31:55PM -0400, Vince Vielhaber wrote:
change from mySQL to PostgreSQL.
I love mySQL but am getting very concerned about i'ts new owners.
Should I be able to move all my realtime stuff to PostgreSQL is it fully
[snippage and probably off topic]
Why are you worried
On Fri, Aug 13, 2010 at 12:46:51PM +0100, Faris Raouf wrote:
They mean PhonePayPlus (formerly ICSTIS). www.phonepayplus.org.uk
I am not aware of them certifying particular phone systems. Rather, they
impose certain requirements and obligations on the service provider
depending on the nature
On Wed, Aug 04, 2010 at 01:13:56PM -0400, Matt wrote:
Can you recommend any 3G femtocell to VoIP manufacturers? I'm coming
up very dry. OpenBTS sounds like it would work, but is way too
expensive to roll out to residential homes.
Pretty much all Femtocells use 3G locally and send
On Mon, Aug 02, 2010 at 03:36:59PM -0400, Matt wrote:
Is anyone aware of a GSM femtocell that will trunk back to a VoIP
softswitch such as Asterisk?
Most people seem to be concentrating on 3G femtocells (there are various
companies making designs based on picoChip soft radios).
OpenBTS
On Sun, Jul 18, 2010 at 09:56:30AM -0700, Vieri wrote:
As I said above, once you have purchased your SIP channel
you can make
free calls to your PBX using the special number but it's
only INBOUND
AFAIK.
[lots snipped]
With Skype's just released SkypeKit it should be possible to build
On Wed, Jul 14, 2010 at 10:27:13PM +0100, Wipe_Out wrote:
Might be off topic but I thought it would be a good place to ask.. I am
investigating switching to a hosted PBX and dumping my old Asterisk box
thats been running in my office for the last few years.. The few I have
found
On Mon, Feb 08, 2010 at 02:52:33PM +0200, Peter wrote:
I am looking for a gsm gateway that is SIP based i.e no need of FXO/FXS
analogue connection.
I searched the email archives and found messages from 2008 but not sure
how accurate these are.
What do you use and how well it works ? The only
On Sat, Oct 10, 2009 at 10:16:43AM +0200, Patrick wrote:
Thank to Frank and Steve for your answers
My understanding is that you need to place on operator premise an
equipment that checks first the availability of the user on VoIP. If
not registered, it's routing the call through the cellular
On Sat, Oct 10, 2009 at 03:15:20AM +0200, Patrick wrote:
Hello guys,
I'm wondering what is required and involved in order to provide a
wifi/GSM handover to customers.
After googling I haven't found any product/vendor. Do you have an idea ?
That's called UMA and you need operator cooperation.
On Tue, Jul 14, 2009 at 06:46:50PM -0500, Karl Fife wrote:
[snip]
missed the original message
- Original Message -
From: Gordon Henderson gordon+aster...@drogon.net
To: Asterisk Users Mailing List Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, July 14, 2009 9:14 AM
Just founnd a weirdy. My end is Asterisk 1.2.32 using an IAX2 link to
the US.
The IP address of the remote end changed (though in the config file it's
registered as a name i.e. asterisk.remote.end), my system didn't
recognised the IP change, it must be cached once and then the cached
value used
On Tue, Jun 09, 2009 at 02:02:50PM -0500, Danny Nicholas wrote:
Did you do an IAX2 show peer on it?
Remote end unreachable and old IP address
Steve
--
NetTek Ltd UK mob +44 7775 755503
UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455
On Sat, Mar 21, 2009 at 09:39:47AM +0100, randulo wrote:
Hi,
The OpenBTS Project is an effort to construct an open-source Unix
application that uses the Universal Software Radio Peripheral (USRP)
to present a GSM air interface (Um) to standard GSM handset and uses
the Asterisk software PBX
On Wed, Jan 14, 2009 at 02:56:44PM -0200, Leonardo Gomes Figueira wrote:
Tilghman Lesher escreveu:
On Monday 12 January 2009 01:26:02 pm Steve Kennedy wrote:
I think it happened when I upgraded an install to 1.2.31
The variable CALLERIDNUM no longer works and CallerID(num) has to be
used
I think it happened when I upgraded an install to 1.2.31
The variable CALLERIDNUM no longer works and CallerID(num) has to be
used.
I know the initial one was being depreciated, but I didn't see any
mention of it.
Steve
--
NetTek Ltd UK mob +44 7775 755503
UK +44 20 7993 2612 / US +1 310
On Wed, Oct 01, 2008 at 12:53:41PM +0100, Julian Lyndon-Smith wrote:
More than 60% of our outbound calls are now to mobiles, so the time has
come to whack in a gsm channel bank.
Does anyone have any preference of bank ? Do you use a PRI or VOIP
connection from the bank to asterisk ?
On Mon, Sep 29, 2008 at 09:17:11AM -0800, Babcock, Michael Alex wrote:
right will stay away from them, smile.
On Mon, 29 Sep 2008, Babcock, Michael Alex wrote:
what are 70 numbers?
Prefix 070 (then 8 more digits) These are so-called personal
numbers.
They're a blot and an anomaly.
On Sat, Sep 20, 2008 at 12:18:42PM -0400, Dean Collins wrote:
No I know they just bought the company and not the protocol basically
they bought engineering bums on seats.
[1]http://deancollinsblog.blogspot.com/2008/09/cisco-acquires-jabber.ht
ml
Cisco obviously didn't buy
, but Cell to PSTN and
PSTN to Cell are NOT OK.Dean Collins
Poland: Not Today but possibly in 2009 Daniel
UK: Portable if Telco has a porting agreement. Not all Telco have
agreements in place. Steve Kennedy
France: Porting from France Telcom to another
On Wed, Jun 25, 2008 at 10:49:18AM -0400, Alexander Lopez wrote:
Are phone numbers portable in other countries?
Depends what country
Are the same rules and conditions that exist here in the States
mirrored elsewhere?
How does a person in Europe go fully VoIP and still keep the
On Mon, Jun 23, 2008 at 08:24:42AM -0400, Matt Watson wrote:
On June 23, 2008 08:08:53 am OCG Technical Support wrote:
I little more digging and I confirmed that cell phone VM and FAX waiting
icons are in fact controlled by a proprietary SMS message format. Here's
what I found:
[snip]
On Mon, Jun 23, 2008 at 11:03:49AM -0400, Jay R. Ashworth wrote:
On Mon, Jun 23, 2008 at 08:24:42AM -0400, Matt Watson wrote:
Now the tough part...does anyone want to create an app to send
notification
to a cell phone to set/clear these bits?
could you provide a link to where you got
On Wed, May 07, 2008 at 07:46:59PM +0100, Tim Guy wrote:
Installing a new box onto UK NTL (Virgin Media)
During testing phase the callerid worked, now it doesn't.
Can someone confirm that my syntax is right before I start ripping the
configs to bits
exten =
On Mon, Apr 21, 2008 at 11:02:13AM +0100, Mike Dent wrote:
sorry for off topic post, struggling to find any information on UMA in
the UK. I have a Blackberry 8320 phone with wi-fi and UMA
capability, its actually an unlocked Orange branded phone.
T-Mobile don't support UMA in the
On Wed, Mar 12, 2008 at 03:03:38AM +0530, [EMAIL PROTECTED] wrote:
Thanks everyone for the reply.
Till now we had simple IVR so we recorded it ourself.
Now I have a requirement where customer needs a customized message to be
played to customer. I am basically looking for some Text to Speech
On Tue, Feb 26, 2008 at 01:38:31PM -0600, Erik Anderson wrote:
On Tue, Feb 26, 2008 at 1:19 PM, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
Greetings,
How can I call cheap to UK cell phones. I am located in Toronto, Canada, but
need to call UK cell phones both from Toronto and London.
I'd
On Thu, Jan 31, 2008 at 05:59:03AM +0800, Sam Tam wrote:
Try the CT-G1000 from cyber-telecom.net it is 39.99 GBP atm.
err biz again ...
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/Mac
On Tue, Jan 29, 2008 at 09:12:11AM +0800, Sam Tam wrote:
Try cyber-telecom.net
May be get a X100P with a CT-G1000 or G2000
a) this should be on the biz list
b) why don't you post from your cyber-telecom.net address?
c) it must be the end of the sales cycle and trying to get a bit more
revenue
On Mon, Jan 14, 2008 at 08:20:11AM -0500, Steve Totaro wrote:
On Jan 14, 2008 7:42 AM, bilal ghayyad [EMAIL PROTECTED] wrote:
Is there an Digium cards support GSM SIM cards so we
can fix an SIM card to be used for calls within
mobiles as it is less rate?
Or I have to
On Wed, Nov 21, 2007 at 11:35:42AM -0500, Dean Collins wrote:
There's an application server that sits between asterisk and the gprs network
that can switch calls real time between wifi, your office pabx extensions and
the gsm network.
I've forgotten the name of it but I remember it costs
On Thu, Nov 01, 2007 at 01:09:24PM +0100, Benny Amorsen wrote:
AM == Anselm Martin Hoffmeister [EMAIL PROTECTED] writes:
AM Maybe the GSM codec is implanted to the GSM chip and that one
AM does alaw, ulaw...
Also, modern handsets like the E90 rarely use the plain GSM codec.
They use newer
On Wed, Oct 10, 2007 at 02:10:54PM -0400, SIP wrote:
[snip]
I think that using 1.5.x as the name for a release candidate for 1.6 is
pretty close to as unintuitive as it can possibly be.
1.6.Xrc-Y is a strikingly MORE intuitive naming scheme for 1.6 release
candidates.
mutt uses the x.y
On Fri, Aug 31, 2007 at 10:03:07AM -0600, Kai-Uwe Jensen wrote:
On 8/31/07, Anthony Francis [EMAIL PROTECTED] wrote:
Mindfully wanting to use a + instead of knowing the international access
code seems like willful ignorance to me.
I beg to differ. Consider cell phones as an example. They
On Wed, Aug 22, 2007 at 01:19:14PM -0500, Asterisk Development Team wrote:
The Asterisk.org development team has announced the release of Zaptel
versions 1.2.20.1 and 1.4.5.1. These releases are to correct an error in
the install target in the Makefile of the 1.4.5 and 1.2.20 Zaptel
On Tue, Jul 31, 2007 at 03:05:32PM -0400, Jay R. Ashworth wrote:
On Tue, Jul 31, 2007 at 01:37:00PM -0500, voiplist wrote:
I have done this in the past and I don't recall ever finding any
popular music by popular artist.
For example, if I wanted to play oh I don't know an original song
On Tue, Jul 31, 2007 at 05:22:20PM -0400, Jon Pounder wrote:
Quoting John Millican [EMAIL PROTECTED]:
there are plenty of radio stations with internet feeds of their audio,
piping that in would not change any coverage area since anyone with
internet could listen anywhere already, you're
On Tue, Jul 17, 2007 at 11:56:35AM +0200, Anselm Martin Hoffmeister wrote:
Am Donnerstag, den 12.07.2007, 16:57 -0700 schrieb Russ McBride:
Newbie question(s):
From what I can determine it sounds like the SMS messaging isn't as
robust as it could be (?). I'm wondering if there's active
On Wed, Jul 04, 2007 at 08:06:49AM -0500, Lacy Moore - Aspendora wrote:
On 7/3/07, Joe acquisto [EMAIL PROTECTED] wrote:
Contrary to the opinions of Anglo-Philes, we, here in the Colonies,
speak American, not English. In some places, 'Murican.
Merkins speaking Murican ...
On Tue, Jun 26, 2007 at 09:45:48PM +0200, Olivier wrote:
Has anyone met any success, installing localized (ie non-english) menus
within SIP firmware enabled Cisco 7941 ?
Those phones seem to be trying to download localized menus from Cisco
Call Manager but as they are managed by
On Fri, Jun 22, 2007 at 08:06:39AM -0400, Dave Bour wrote:
So I'll ask the question. What's wrong with top posting. I use a
blackberry to read most of my email, and bottom posting means excessive
scrolling, often waiting to download additional content resulting in
higher usage
Is there anyway to change the flash time on a TDM400 phone port (a
user has a phone that seems to generate a short flash which isn't being
picked up).
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 Wed, Jun 06, 2007 at 08:46:20AM -0500, Eric ManxPower Wieling wrote:
Steve Kennedy wrote:
Is there anyway to change the flash time on a TDM400 phone port (a
user has a phone that seems to generate a short flash which isn't being
picked up).
I suspect the phone us going off hook every once
On Mon, May 21, 2007 at 03:10:23PM -0400, Lee Jenkins wrote:
Mike Hammett wrote:
I was looking at the ILECs? web sites to determine how their users
access voicemail.
What method should I use for my users checking their voicemail? Can
Asterisk voicemail be made to accept hitting * during
I'm trying to get zap fallback to VoIP working. I dial the zap channel
and if it fails I want to then try another route.
If the channel is busy (i.e. CHANUNAVAILABLE) it works, however if
there's no dial-tone, it doesn't seem to detect this?
Using a TDM400 with UK settings.
Steve
--
NetTek
On Fri, May 18, 2007 at 12:54:19PM -0500, Matthew Fredrickson wrote:
On May 18, 2007, at 11:50 AM, Steve Kennedy wrote:
I'm trying to get zap fallback to VoIP working. I dial the zap channel
and if it fails I want to then try another route.
If the channel is busy (i.e. CHANUNAVAILABLE
On Wed, May 16, 2007 at 09:15:49PM +0100, Matt Brown wrote:
[snip]
No, this client has a number of engineers all over the UK and they
have a large mobile contract with several handsets - their current
tariff includes free calls to other mobiles under the contract
so what they are trying
On Wed, May 16, 2007 at 02:17:11PM +0100, Matt Brown wrote:
I am currently building a 1.4.4 Asterisk box for a client and they
are interested in GSM functionality.
Does anyone have any experience with a GSM card, preferably Quad Span
(4 GSM modules or higher) for use in the UK. I have
On Sun, May 13, 2007 at 01:54:08AM +0100, Chris Bagnall wrote:
3. a list of bogus entries..so when you look at it, you know it's a
fake phone number...one that recently came in that got me thinking
this was 407 111 .
I don't know much about the legal position over the other side of the
On Thu, May 10, 2007 at 06:36:53PM +0200, nik600 wrote:
i have a Te205P connected to a PRI E1, can i force the outgoing
callerid to change for each context?
for example:
[outgoing_context_one]
;force callerid to 12345
exten = _XXX,1,Set(CALLERID(number)=12345)
exten =
On Thu, May 03, 2007 at 01:22:07PM -0300, Ronaldo wrote:
Can you suggest me any documentation about using IAX trunking?
Thank you.
There are examples in the iax.conf files I think, but basically just put
something like
[iax-toremote]
type=friend
username=whatever
secret=somesecret
On Thu, May 03, 2007 at 03:23:16PM -0300, Ronaldo wrote:
OK Steve,
Just one more question. Using this configuration can I make more than
one call at the same time?
The whole point of trunking is to support multiple calls down the same
IAX trunk (well actually down the same packets).
Steve
Does anyone know of an (E)AGI or program to develop a IVR dial-plan
which will take a list of words and then do something when a unique
branch has been found.
i.e.
Say there's 3 words
demon
deacon
bishop
On a phone they'd be represented as
33666
332266
247467
So if the user enters 2 we know
On Thu, Apr 26, 2007 at 06:46:41AM -0400, J. Oquendo wrote:
Steve Totaro wrote:
I suspect that this will happen more and more. I also suspect that many
people who have weak SIP credentials like user=100 secret=100 will be
the victim of toll fraud and worse, call to 900 and other very high
Just been getting lots of failed SIP registrations to a system here.
All coming from Taiwan.
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/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News
Has anyone got a sensible zaptel.conf and zapata.conf for 2 TDM400P's
working with UK set-up.
They're set-up with 7 analogue phones and 1 PSTN port.
Currently zaptel.conf has
fxoks=1-7
fxsks=8
loadzone=uk
defaultzone=uk
It's really zapata.conf that would be useful.
Currently using the
On Thu, Apr 19, 2007 at 08:36:12AM -0400, Steve Totaro wrote:
Just a thought, try kannal, use system in your dialplan and call lynx with a
properly formatted URL for Kannal.
Or indeed Kannel (www.kannel.org)
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US
Just a FYI to the list.
It seems that although BT only present 6 digits (as standard) for CLI
they expect the full number minus the leading 0 to set CLI.
So if a number is 01234 987654
They will present 987654
and you need to present to them 1234 987654
Hmmm
Steve
--
NetTek Ltd UK mob
On Thu, Mar 29, 2007 at 10:40:50PM +0100, Julian Lyndon-Smith wrote:
We only present the 6 digits ... and they give us 6 digits. For our
outbound calls, for the the numbers 01702 1234[00-99] we have to present
1234[00-99].
BT isdn pri line.
Weird, seems they're inconsistant or there's some
Has anyone got a working zaptel.conf and zapata.conf for a Digium
Wildcard TE110P T1/E1 Card.
It's connected to a BT ISDN PRI (EuroISDN) with 24 channels.
Inbound works fine, but outbound isn't setting CLI (it seems the line
supports 6 digit CLI). Inbound CLI works fine.
In the dial-plan using
A quick question on queues in Asterisk, if you specify a specific
resource as a queue member (i.e. member = SIP/40 say) is it
automatically a member of the queue without having to specifically log
on via AgentLogin stuff?
I under stand if you specify something like member = Agent/100 you then
On Sat, Mar 17, 2007 at 11:44:52AM -0700, Steve Edwards wrote:
Yes.
to which bit? auto-agent (as per resource)
or voicemail to an agent?
Steve
On Sat, 17 Mar 2007, Steve Kennedy wrote:
A quick question on queues in Asterisk, if you specify a specific
resource as a queue member (i.e
On Sat, Mar 03, 2007 at 12:01:58PM +, Gordon Henderson wrote:
You're missing nothing; The telcos have us by the short curlys. For
them, it's money for old rope. They probably (in the UK at least) make
many times more money through TXT messages than voice. The base rate
here is about
On Thu, Mar 01, 2007 at 12:17:49PM -, Chris Stenton wrote:
I have used www.voiptalk.org for a number of years with their IAX2
connectivity and they seem very reliable with no echo issues. They will
also change the CID to your number if you fax them proof of ownership.
There's several
On Wed, Feb 14, 2007 at 03:42:23AM +0100, Patrick wrote:
On Tue, 2007-02-13 at 18:31 -0700, Stephen Bosch wrote:
Singer Wang wrote:
by your .ca address I assume your in Canada..
both Telus and Rogers have a email-to-SMS gateway...
Well, those are notoriously unreliable. I've had
On Wed, Feb 14, 2007 at 09:41:32AM +0100, Dave Cotton wrote:
On Wed, 2007-02-14 at 15:33 +0800, Sam Tam wrote:
Hello All
This month we would like to offer our GSM Gateway range for less to
clear up some spaces.
etc
Perhaps, you could explain what is NON COMMERCIAL about your post.
I
On Wed, Feb 14, 2007 at 10:29:20AM -0700, Stephen Bosch wrote:
[snippage]
If I understand correctly, this means I'll need an extra SIM just to
send messages -- is that right? I build a Kannel server so that it can
talk to a terminal that is on the network and can send messages.
(It's an awful
On Wed, Feb 14, 2007 at 07:17:32AM +0800, Ronald Wiplinger wrote:
Where can I get a starting point for setting up sms via VoIP and via web.
I want to send SMS from VoIP or web to VoIP phones and GSM phones.
1. how to set-up?
2. which smsc should I use? (what is the price?)
3. which phones
On Thu, Feb 08, 2007 at 05:58:08PM +, younss azzayani wrote:
when i compile zaptel
make linux26
make install
i got these errors:
make[1]: Leaving directory `/usr/src/zaptel-1.4/wct4xxp'
make -C datamods clean
make[1]: Entering directory `/usr/src/zaptel-1.4/datamods'
make -C
On Sat, Jan 27, 2007 at 10:20:40AM -0500, Matthew Rubenstein wrote:
How does H.264 compare with GSM and G.729 in CPU demand (MIPS:Kbps) and
audio quality at low bitrates? GSM is $free, but G.729 is higher quality
(tho patented with at least $10 per running codec instance royalties).
On Wed, Jan 10, 2007 at 06:33:05PM -0500, M.Hockings wrote:
That is more what I was thinking of but it is still a cell provider type
of hardware. In my mind I was thinking of something very low powered
and turning off the roaming, etc on the phone so they only work with the
one base.
On Tue, Jan 09, 2007 at 05:11:55PM -0500, M.Hockings wrote:
I don't really know the name of what I want to look for but maybe
someone could tell me if it would be available.
I have a number of old analogue cell phones laying about here and I was
thinking it would be useful if I could set up
On Tue, Jan 09, 2007 at 02:40:07PM -0800, mitcheloc wrote:
Wait for the iPhone...seriously.
I assume you mean Apple iPhone not Linksys iPhone ?
It looks lovely, shame it's not available in UK until Q4.
(also not FCC approved yet, but I assume that was deliberate as most
phone leaks tend to
On Mon, Jan 08, 2007 at 10:51:03AM -0500, Al Bochter wrote:
What about the free open source G729
There's no such thing ... g.729 (as per the ITU specification) is patent
encumbered. Anyone USING the codec has to pay a license to the patent
holders.
Digium have negotiated a bulk-buying
On Mon, Jan 08, 2007 at 02:53:39PM -0500, Al Bochter wrote:
So tell me what this FREE open source G729 is
I am told that you can use these Codecs with your Asterisk !
[1]http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
You can do it Freely !!
No, Ready Technology
I believe there were some new prompts added for 1.4 for Directory Info.
These have now been added to http://www.tel.net
Have a good 2007.
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/Mac
On Fri, Dec 01, 2006 at 06:24:46AM +0800, Sam Tam wrote:
We do have @cough VoIP GSM Gateway for sell as well @ cough
Try to search on ebay for gsm voip gateway and you will see some in there
As far as I am concern it is cheaper than 2n.
And if you are looking for multi ports then it will come
On Tue, Nov 28, 2006 at 08:30:55AM -0500, Frank Tarczynski wrote:
I'm looking to build the zaptel drivers on a Solaris 10 X86 box. I've
found the driver source code on
https://svn.sunlabs.com/svn/solaris-asterisk but this source is posted
along with Asterisk 1.2.7.1 Does anyone know of
On Fri, Nov 24, 2006 at 11:51:41PM -, Neil Tancock wrote:
Anyone know if Asterisk will work with ISDN 30 and what sort of device I'll
need to connect it?
It will work with UK ISDN, but ensure it's EuroISDN and NOT UK ISDN
(it's set in the telco switch and can generally be changed).
UK
On Thu, Nov 23, 2006 at 12:40:03AM -0800, Brad Templeton wrote:
[snip]
The USA uses 120v for house current. That's enough to hurt you and can
kill you if you touch it wrong, though I've touched it a few times.
A lot of the world uses 220. This causes enough of a spark that they
require all
On Thu, Nov 23, 2006 at 08:55:41PM +, Tim Panton wrote:
I've asked gradwell about my second point (still waiting...), but your
thoughts are the same as mine. In theory it should be ok, because I
have to authenticate the IAX connection with a username/password,
which
in turn they own
On Wed, Nov 22, 2006 at 06:58:13PM +0100, Huib van Wees wrote:
On 11/22/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
Why Aastra phones use more electricity, i.e. 48VDC whereas other
phones use much less, e.g. Grandstream and Linksys both use only
5VDC. I first thought it was
On Wed, Nov 15, 2006 at 11:15:16AM +, Gordon Henderson wrote:
(I'm in the UK if that makes a difference)
There seems to be a plethora of different ISDN cards available in both the
BRI and PRI range - all with varying prices too - from ?25 to nearly ?1000
from some popular reseller
On Thu, Nov 02, 2006 at 02:47:42PM -0500, Erick Perez wrote:
This one will surely heat up.
Usually the telcos have to calculate the subscribers vs telco capacity.
I use simple figures, so extrapolate this to millions of customers,
millions of lines, peak amount of calls at any given time of
On Sun, Oct 29, 2006 at 03:09:42PM +, Conrad Wood wrote:
On 29 Oct 2006, at 11:02, Matthew Thompson wrote:
On 26 Oct 2006, at 11:59, Conrad Wood wrote:
A client used to use BT isdn30 and ported the numbers to telewest
several years ago.
Now, the client moved to adept telecom. I *think*
On Wed, Oct 04, 2006 at 06:01:25PM -0400, Jay R. Ashworth wrote:
On Wed, Oct 04, 2006 at 02:18:49PM -0600, Natambu Obleton wrote:
Anyone have happen know how to reset the password on a TNT Max? Thanks.
Does your asking here suggest that the the MAX's can do, say, voice
gateway service?
On Mon, Sep 25, 2006 at 06:09:02PM -0400, Alex Robar wrote:
This is a non-commercial discussion list, hence the name Asterisk
Users Mailing List - Non-Commercial Discussion. Post this to the -biz
group.
He does this every month or so
Steve
--
NetTek Ltd UK mob
On Thu, Sep 21, 2006 at 10:11:43PM +0100, Brian Candler wrote:
Does anyone here know of an ADSL router with integrated SIP proxy?
Netscreen 5GT ADSL, it has what's called an ALG (application layer
gateway) and it does indeed support SIP. Full featured firewall etc too.
Steve
p.s Hi Brian :)
On Fri, Sep 22, 2006 at 02:56:39PM +0100, Will Tatam wrote:
Steve Kennedy wrote:
I'd like to announce that the UK Male English Voices are now up on
http://www.tel.net/
[snip]
The website appears to be down
Yup, did an upgrade on Fri and something went wrong - will be fixed
tomorrow.
Steve
I managed to work around my Dialplan.
The ChanIsAvail application is great, except it only returns the 1st
available channel.
Could there be a ChansAreAvail which returns all the channels available
instead of just the first. I'm sure it could be implemented as a macro
or I guess a rewrite of the
I'm trying to dial multiple SIP channels and check availability before I
dial them.
i.e. say I have an internal group that I define (extension 50) which
actually dials SIP extensions 51 and 53
I'd use Dial(SIP/51SIP/53), but if a phone isn't registered (i.e.
someone's unplugged 53) it does weird
On Wed, Sep 20, 2006 at 04:12:34PM +0100, Faris Raouf wrote:
magnus wrote:
Hi all, could anyone share how to perform attended transfers with Asterisk
and Grandstream SX2000's - we are able to perform blind transfers with no
problem, but attended transfers fail - is it necessary to set two
On Sun, Sep 17, 2006 at 03:04:35PM -0700, Net Nut wrote:
Well this would not be for comercial use.. I just want it for my own
cell phone to talk on my own asterisk system.
is that ok?
Voiceage are quite agressive in terms of licensing. However as an
individual it's probably not worth their
I'm not anything to do with them, but sounds a nice design.
CSR have introduced a VoWiFi reference design that costs around $20.
The interesting thing is that it supports both SIP and IAX2.
Maybe Digium should make a WiFi handset ...
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK
On Fri, Sep 15, 2006 at 07:28:29PM -0700, Net Nut wrote:
I have been searching, but I have not found the answer.. How might I add
the amr codec to my asterisk server?
I believe I found the amr source from
http://www.3gpp.org/ftp/Specs/latest/Rel-6/26_series/26073-600.zip
I compiled it but
On Wed, Sep 13, 2006 at 12:33:01PM -0400, Steven Totaro wrote:
Use SVN and not the tarball.
Digium updated to 1.2.9.1 earlier this week.
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/Mac
On Wed, Sep 13, 2006 at 01:34:30PM -0400, Mark Hulber wrote:
Yes, it worked. I didn't get the announcement of 1.2.9.1.
Seems it wasn't announced, nor Asterisk 1.2.12.1
Nor their new Asterisk Appliance that seems to run off Flash (with a GUI
that configures it all). ALso the new 4 port BRI
When are Digium going to upload a corrected 1.2.9 zaptel tarball that
compiles?
I know it's correct in svn, but the public ftp servers still hold the
incorrect version.
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
I have Asterisk 1.2.11 running and a Cisco 7960 (SIP v7.3). I cant seem
to get the message waiting indicator working.
I did try changing the MIME type as suggest, but then the phone kept
continuously ringing.
Any pointers?
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612
On Wed, Sep 06, 2006 at 03:36:53PM -0400, Doug Lytle wrote:
Steve Kennedy wrote:
Phone itself.
[S-5200]
This is incorrect. It should be:
[5200]
mailbox=5200
That bit seems to work, phones registers ok and can receive and make
calls.
You're missing the @context on your mailbox line. i.e
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