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
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 natur
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 t
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
>foun
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
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 cellul
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 cooperati
On Tue, Jul 14, 2009 at 06:46:50PM -0500, Karl Fife wrote:
[snip]
missed the original message
> - Original Message -
> From: "Gordon Henderson"
> To: "Asterisk Users Mailing List Discussion"
>
> Sent: Tuesday, July 14, 2009 9:14 AM
> Subject: [asterisk-users] Is Enum safe from spamme
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
Skype/GoogleTalk/AIM/Gizmo/.Mac/Twit
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 fo
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 softwar
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 wor
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 85
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 ? Real
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 a
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
PSTN and Cell to Cell are OK , 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
>
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 th
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 wh
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:
[sni
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 => _9.,1,Set(CALLERID(number)=01
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 t
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 Spee
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.
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 steve
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 hav
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.
>
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
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
> rel
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
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 origina
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
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 b
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
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).
>
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
Skype/GoogleTalk/AIM/Gizmo
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 *
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
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 Ltd
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 tryin
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
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 => _XXX
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).
St
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
auth=plain
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 t
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 h
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 B
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 zaptel/as
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 +1-(31
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 s
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 +44-
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 S
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
> >resou
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
ha
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 a
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 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 aw
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
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
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 phon
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 /l
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 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 co
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
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
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 agreemen
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 stevekenn
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
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
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 IS
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 t
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 a
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 i
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 site
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 o
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.
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
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NetTek Ltd UK mob +
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 -
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 :)
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
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 t
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/51&SIP/53), but if a phone isn't registered (i.e.
someone's unplugged 53) it does weird
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
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NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)2
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
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 bu
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 car
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
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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 stevek
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
Skype/Goo
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 m
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
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UK +44-(0)20 79932612 /
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