On 10/05/12 09:49, Bart Coninckx wrote:
Hi all,
for smaller (or maybe even bigger) sites I'm looking for a smaller,
appliance-type like PC, preferably solid state and fanless PC.
Since it's only going to run Asterisk for a couple of extensions I don't
think CPU and RAM need to be maxed out.
On 26/08/11 12:28, linux guy wrote:
Great discussion, all of it. Thanks, people.
How much power does the home asterisk box need ?
Not much :-)
I've been running our phone system and home media/storage network on a
VIA C7 cpu based home build that I *downclocked* to 1Ghz from 1.2Ghz for
On 26/08/11 19:02, linux guy wrote:
I'm looking for 4 to 6 good, inexpensive VOIP handsets for my home
asterisk system.
We've been using the Siemens Gigaset 685IP range for over three years
and I'm (still) very pleased with them:
On 19/08/10 18:20, equis software wrote:
I want to know about asterisk and openBTS
If anybody made any test and experience...
I saw a presentation a few months ago where one of the openBTS project
founders talked about one early system they set up on a very small and
remote Pacific island
On 19/08/10 18:20, equis software wrote:
I want to know about asterisk and openBTS
If anybody made any test and experience...
This island runs it's GSM network on OpenBTS: http://www.niueisland.com/
This was the place he presented about.
Read the blog here:
On 13/08/10 14:08, Albert Bonomo wrote:
This time, the server was up with Fedora 13. No problem.
Well not so fast. Yes problem !!! I can not install it !!!
Fedora is *not* a server operating system and not one I would choose to
run asterisk on.
I would recommend using either CentOS or a
http://gigaom.com/2010/08/03/2600hz-project/
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http://www.theopenlearningcentre.com
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On 02/08/10 17:35, Ronaldo Zacarias Afonso wrote:
Hi all,
Can some one suggest me an IAX client for Linux and Windows?
I used KIAX once, but know it seems complicated to have it working on Ubuntu.
This one is great on Ubuntu/Linux. http://www.sflphone.org/
Unfortunately I know not about
On 22/02/10 16:18, --[ UxBoD ]-- wrote:
Hi,
looking for your valued input on suitable suggestions for high quality VoIP
DECT phones. I am having real issues with my Snom M3s and Asterisk 1.6 and
looking to a new manufacturer.
Another vote for the Siemens Gigaset range. Been using the
On 23/02/10 08:38, Randy R wrote:
On Tue, Feb 23, 2010 at 9:23 AM, Alan Lord (News)alansli...@gmail.com
wrote:
Another vote for the Siemens Gigaset range. Been using the S685IP almost
since the day it was released here in the UK. Nice handsets, great voice
quality, but as others have said
When you configure the Siemens gigaset handsets (I have S685IP), there
is a single option for all handsets to use either the POTS interface or
VOIP as the default outbound destination - you then need to add a dial
suffix if you want to use an alternate outbound route.
Does anyone have any
On 10/11/09 18:19, Christina Casey wrote:
Hi Klaus,
Yes all the below is possible/easy with the OrderlyStats call centre
management and reporting tool.
It's a free download - please see http://www.orderlyq.com/orderlystats.html
Kind regards,
Christina Casey
Accounts Manager
Orderly
On 11/11/09 10:00, Steve Howes wrote:
On 11 Nov 2009, at 09:06, Alan Lord (News) wrote:
Warning: Your browser may not be able to handle this site! Please
upgrade your browser to the latest version of Internet Explorer,
Firefox, Mozilla or Netscape.
Sorry for any inconvenience
On 24/10/09 11:05, Steve Howes wrote:
On 24 Oct 2009, at 10:52, giancarlo lombardo wrote:
at the moment machine is standalone, so i need to start GUI from
console.
There is no Asterisk GUI that runs like that. You could install X and
Firefox but thats just a bit retarded. Plug in a network
On 23/10/09 08:34, --[ UxBoD ]-- wrote:
snip /
S685 set turned up the other day and have had a good chance to try it out ...
Voice quality is definitely superior to the M3 though I guess that will be
addressed in the M9 with G722 support.
Impressions of the Siemens phone :-
Pros
On 17/10/09 15:02, --[ UxBoD ]-- wrote:
Hi,
I have three Snom M3s at the moment but getting pretty fed up with the issues
:( I am UK based and would be interested to hear of other peoples
recommendations. Key features :-
* VM Notification
* Good Range
* G729 codec support
*
Just FYI Really, nothing to do with me...
http://www.thevarguy.com/2009/10/01/systems-integrator-dials-skype-for-asterisk/
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On 01/10/09 00:57, Kirill 'Big K' Katsnelson wrote:
Asterisk 1.6 behind a firewall and NAT. We are terminating multiple DID
calls, originating and transferring.
A provider offers both SIP and IAX trunking. Cateris paribus, what is
the preferred solution to choose? What points to consider?
We
On 26/09/09 19:42, Hans Witvliet wrote:
snip /
What you can do (perhaps not the best solution...) is having one
asterisk server behind your firewall, serving all your local
sip-clients. And another at the other side of the firewall, only for
serving remote clients. And have both systems
On 12/09/09 08:03, DHAVAL INDRODIYA wrote:
hello
while i try to compile zaptel
it gives following error to me
you do not appear to have the sources for the 2.6.27-7-server kernel
installed
You will need to install the source tree (or at least the headers) for
the running kernel.
On
On 18/08/09 08:08, Olivier wrote:
Hi,
I need to replace digital handsets in offices where there cabling is
appareantly not Ethernet-compliant.
Today's usage is to press a key to toggle between private ou public line
before issuing an outgoing call.
Are you aware of a DECT handset (to
On 04/08/09 23:57, Miguel Molina wrote:
Edwin Quijada escribió:
It depends about your traffic.
But myabe and I guess Core 2Duo 4Gb ram , Sata 160Gb
+_
It's pretty well known that asterisk is CPU intensive, not RAM
intensive. It think 4GB is much more than enough. BTW, if your asterisk
On 25/07/09 00:08, John A. Sullivan III wrote:
Hello, all. After many pages of googling and testing in the lab, I'm
still a bit perplexed about how to implement tls protection for the
asterisk manager. manager.conf allows one to specify the cert file but
one normally must also specify the
On 19/07/09 04:50, Maxi Belino wrote:
snip /
It get's really hard to to try and deal with all the possibilities
reliably.
IMHO, the Destination field *should* contain simply the number of the
destination ext. of the call; as it rightly does when digits are
actually
On 18/07/09 00:35, Gavin Henry wrote:
This has to be an Asterisk based appliance no?
http://www.truecall.co.uk/acatalog/trueCall_Features.html
I saw this on the TV the other night. Couldn't believe how the dragons
all thought it was such a cool idea.
I was shouting at the telly saying You
Hi all,
I am trying to understand how I can get a simple IVR scenario to work
properly (having already removed most of my hair...).
The basic requirement is as follows:
* Caller arrives at our main number
* Caller is greeted and then told they can enter an extension number, if
known, or wait
-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alan Lord
(News)
Sent: Friday, July 17, 2009 3:23 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How do I create an IVR/Dial Group that
worksproperly?
Hi all,
I am trying to understand how
the call gets connected.
Cheers
Alan
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alan Lord (News)
Sent: Friday, July 17, 2009 11:12 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users
On 17/07/09 16:30, David Backeberg wrote:
On Fri, Jul 17, 2009 at 4:22 AM, Alan Lord (News)alansli...@gmail.com
wrote:
* Caller arrives at our main number
* Caller is greeted and then told they can enter an extension number, if
known, or wait and their call will be connected to an available
are
actually dialled by the caller. Why it doesn't when the call is
generated by the dialplan IVR is just plain inconsistent.
Alan
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alan Lord
(News)
Sent: Friday, July
Hi all,
I'm sure this has been done before but I just can't figure it out.
On my * box I have a simple IVR:
[tolc_menu] ; Welcome and information to callers
exten = s,1,Answer()
exten = s,n,Wait(2)
exten = s,n,Background(welcome-to-tolc) ; Say Hello
exten = s,n,Wait(1)
exten =
Hi All,
I've just upgraded our CRM and it has an Asterisk Integration Module
that I would like to test out.
The CRM is running on one of our hosted servers in the cloud. The
Asterisk server is running in my office.
I am running Asterisk 1.4.21.2~dfsg-1ubuntu3.
Reading the page
On 09/07/09 14:40, Steve Howes wrote:
On 9 Jul 2009, at 13:05, Alan Lord (News) wrote:
Reading the page
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20manager.conf
got me a little concerned regarding having an open channel between the
two machines and there is scant
On 29/06/09 18:26, Gordon Henderson wrote:
Looking for a (windows) app. that will listen to the manager interface
then pop-up a web browser pointing to a page on an incoming phone call..
Not looking for outlook integration, or outbound dialling, just to
recognise an incoming call and poke a
On 01/07/09 16:29, Gordon Henderson wrote:
On Wed, 1 Jul 2009, Alan Lord (News) wrote:
On 29/06/09 18:26, Gordon Henderson wrote:
Looking for a (windows) app. that will listen to the manager interface
then pop-up a web browser pointing to a page on an incoming phone call..
Not looking
On 22/06/09 18:20, David @ULC wrote:
What the best website and book to start learning asterisk ?
Website: Google, http://www.voip-info.org
Book: TFOT (The future of Telephony) Google for it , it is
freely/legally downloadable.
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On 03/06/09 08:37, Rilawich Ango wrote:
Hi all,
Any good recommendation of IP phone in term of sound quality and
price (reasonable) using with asterisk?
ango
Not sure where you are in the world, or what you really need but I like
the Siemens Gigaset IP DECT phones.
The S685IP is really
On 01/06/09 10:27, Vincent wrote:
Hello
I'm thinking of selling an Asterisk server based on Atcom's IP02
solid-state unit with one FXO and one FXS ports:
http://atcom.cn/En_products_IP02.htm
By default, this unit based on a 400MHz Blackfin 532 chip only has
64MB RAM and 256MB of
On 24/05/09 19:21, Gordon Henderson wrote:
snip /
Here's an example:
http://www.wppltd.demon.co.uk/WPP/Wiring/UK_telephone/uk_telephone.html
In these enlightened days, it's normal to not use the 3rd wire internally
on extension as it has been known to degrade an ADSL signal.
I think this
On 11/05/09 04:21, John F. Ervin wrote:
snip /
Are there (??) instructions for people who are experienced at the
Trixbox level but wish to move on?
Sure, the TFOT book is a great start. If you want to use Ubuntu or
Debian rather than Centos then Asterisk is in the Debian and Ubuntu
Server
On 06/05/09 08:28, Gordon Henderson wrote:
snip /
One little tip: You need to compile Asterisk for an i586 processor as
the VIA processor is missing a few (mmx, etc.) instructions that a full
blown i686 has.
Hi Gordon,
I'm using a VIA C7 on a Jetway board
On 06/05/09 13:43, Vincent wrote:
Hello,
I'm looking for a dirt cheap solution for SOHO use to handle at most
a couple of POTS lines, and I notice that X10?P cards go for $15 on
eBay as opposed to $90 for an OpenVox card or over $200 for a Sangoma.
I have a couple of questions about
On 04/05/09 21:17, James A. Shigley wrote:
I’m just trying to make a real simple Survey via php. Just want it to
play the Question Files, wait for a response, save the response into the
correct variable and then email it all.
Packt sent me a book to review recently: Asterisk AGI Programming.
Hi all,
This isn't meant to be spam I thought some of you might find it interesting.
Packt Publishing approached me a few weeks ago and asked if I would like
to review a book or two for them on my blog.
The first one they sent me is called Asterisk Gateway Interface
Programming and has only
Hi, I know this is a little OT but there are many Asterisk users of the
excellent Siemens DECT/VOIP phones like the S685IP and 475IP and this is
probably newsworthy for them.
One of the biggest bug bears has been no mute function on the handset.
When I woke up this morning, the handset told me
Manolet Gmail wrote:
I have an ubuntu 8.10 machine. Linux 2.6.27-14-generic
i want to configure a x100p card an use it with asterisk, so i download,
compile and install:
asterisk-1.4.24
dahdi-linux-2.1.0.4
dahdi-tools-2.1.0.2
libpri-1.4.9
i try almost everything i found on the net
Fred wrote:
Hello
Considering how cheap PCI modems are compared to even the cheapest
PCI hardware from Digium, OpenVox, Sangoma, etc I was wondering
why Zaptel can't be used with those to connect an Asterisk server to
a POTS line for low-level use? It just seems overkill for SOHO
Fred wrote:
Hello
Considering how cheap PCI modems are compared to even the cheapest
PCI hardware from Digium, OpenVox, Sangoma, etc I was wondering
why Zaptel can't be used with those to connect an Asterisk server to
a POTS line for low-level use? It just seems overkill for SOHO
Anthony Plack wrote:
Hey all,
I have a potential project which calls for a very small form-factor computer
like this:
http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp
However, I am needing an FXS port integrated into a small footprint computer.
Paul Hales wrote:
Noojeeclick?
http://www.noojee.com.au/Page/NoojeeClick
Thanks for that. Not heard of NoojeeClick before. Their site is not
responding right now but the Firefox add-on page is up. when I get
chance I will try it out.
https://addons.mozilla.org/en-US/firefox/addon/8510
I
Dean Collins wrote:
ADA Forums: http://forums.digium.com/index.php?c=8
ADA Download: http://dl1.digium.com/ADA/ADAInstall.exe
ADA Administrators Guide: http://dl1.digium.com/ADA1.1/ADA_Admin_Manual.pdf
Thanks for the links. I hadn't seen that before. The product is kind
of interesting, but
Tzafrir Cohen wrote:
snip /
But when the wcfxo module is loaded, it is not loading the oslec module.
There is an oslec.ko in /lib/modules/my-running-kernel-version/misc/oslec/
According to launchpad, oslec should be the default ec now for zaptel.
Anyone got any ideas please?
I wonder if anyone has any ideas on this.
I have recently migrated my server from a custom built Linux with
Asterisk, Zaptel and Oslec running fine, to Ubuntu Server 8.10.
I have Asterisk installed via synaptic at it works fine.
I have built and installed the zaptel package by doing the
Alan Lord (News) wrote:
I wonder if anyone has any ideas on this.
I have recently migrated my server from a custom built Linux with
Asterisk, Zaptel and Oslec running fine, to Ubuntu Server 8.10.
I have Asterisk installed via synaptic at it works fine.
I have built and installed
That was quite an interesting set of responses. I didn't get any
impression that there is a strong preference either way.
Thanks for all the replies.
Al
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asterisk-users
Hi all,
I built my first asterisk using the traditional (?) .conf files and
constructs.
I recall reading books at the time about AEL but it seemed new and
untested so I left it alone. Now, I'm interested to poll the audience
here to see if I should look into using AEL instead (or in addition
Ronald Wiplinger (Lists) wrote:
I know I can setup asterisk without Internet at all and it works as
local pbx.
Would an asterisk box work with a dynamic IP, with a dyndns name?
What must I take care if I try that?
I had my * server behind my adsl router that was getting a dynamic Ip
Pezhman Lali wrote:
Dear,
the sip phones that registered, in to the asterisk 1.4.x have the echo
in their callings to pstn.
how this echo can be canceled?
H - you don't give much to go on...
What is the connection to the PSTN (i.e. what kind of card, interface
etc...)
The echo is
Cory Andrews wrote:
http://blog.voipsupply.com/new-products/free-sip-softphone-roundup
Good roundup, thanks.
Alan
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asterisk-users mailing list
To UNSUBSCRIBE or update
]'. Giving up.
which I believe might have something to do with it?
10.0.0.2 is the Asterisk server.
My sip.conf is like so:
[general]
srvlookup=yes
disallow=all
allow=alaw
allow=g722
allow=gsm
dtmfmode=auto
subscribemwi=yes
[101]
type=friend
callerid=Alan Lord 101
secret=bigsecret
Olivier wrote:
snip /
I'll reply to the correct thread
[featuremap]
blindxfer = ## ; Blind transfer
;disconnect = *0 ; Disconnect
;automon = *1 ; One Touch Record
atxfer = A ; Attended transfer
so set
Gordon Henderson wrote:
snip /
Damn - I've just found it in the UK too:
http://www.lambda-tek.com/componentshop/index.pl?prodID=1606310
Must resisst .
I just wish there was a fanless version - one feature which I like in the
VIA boards I use.
Wow, that's an amazing
Gordon Henderson wrote:
On Fri, 24 Oct 2008, Alan Lord wrote:
snip /
I used to have an ISDN-2 line into my home office. BT wrote to me about
2 years ago and said they were discontinuing the service. They converted
my dual channel BRI back into a single POTS.
Sure it was ISDN2e and not Home
Phil Knighton wrote:
Hello all
What I'm looking for is some plain speaking advice on ISDN.
Currently using 4 analog lines connecting via a four port TDM400P FXO
card. We need to physically move our installations, and on requesting
the analog lines be moved - our telco (BT) is
Satish Patel wrote:
snip /
I am using cross compile so i can't update GCC other wise it will effect on
my other packages anyway... tell me one thing i have host system kernel
version is 2.6.18 and i am compiling ARM embedded rootbuild with other
kernel version 2.6.22 so i need to compile my
satish patel wrote:
snip /
I have set env on shell
KVERS=2.6.22.5
KSRC=/path/to/kernel-2.6.22.5/source
Maybe I misunderstood you then. I thought you said that your ARM system
was using a 2.6.18 kernel? If that is the case, then surely you need to
build your zaptel module against that
Steven Howes wrote:
Just copy the src folder and do `make install` on each machine?
Then tar and copy the /etc/asterisk folder if config is important too.
On 29 Sep 2008, at 08:41, Jim Boykin wrote:
Is there a script to create an Asterisk binary package after it is
compiled on one system.
Gordon Henderson wrote:
On Tue, 23 Sep 2008, Steve Totaro wrote:
FYI
It looks like FWD is looking for value added service ideas for free as
a volunteer.
I got this too - looks like a bit of a mass mailling!
And me!
And I haven't visited their site, or connected to their servers as IAX2
Jaap Winius wrote:
Hi list,
Are there any reliable wireless SIP phones available on the market yet?
snip /
Since the firmware seems to be the same, there's no way I'm going to
upgrade to the 685IP. I was thinking of trying out the Snom M3, but
according to voip-info.org, that model
--[ UxBoD ]-- wrote:
From what people have said Asterisk does not require a huge amount of memory
or CPU then ? I only have a couple of extensions. Running the G729 codec and
will look at the Octava software for the PSTN to reduce echo.
Regards,
It doesn't seem so to me. For a home
Joseph L. Casale wrote:
snip /
Al,
What did you finally settle on as a firewall for this project?
jlc
Ahhh - good question; I was waiting for someone to ask... I haven't.
I only really needed a content filter for the younger members of my
family. When I upgraded the boys' computer to
[EMAIL PROTECTED] wrote:
That's the purely technological answer, which is completely correct.
There's a business side to it as well. Siemens is simply not in the
consumer electronics business in North America. They make this decision
consciously.
That's a shame. It's a cracking phone...
Shaun Wingrin wrote:
Hi,
I've followed instructions of the book AsteriskFutureOf
TelephonySecEdit on page 295 onwards ) Link to the Asterisk book:
http://downloads.oreilly.com/books/9780596510480.pdf) and
http://downloads.oreilly.com/books/9780596510480.pdf) and get an error
when
Ronald Wiplinger wrote:
I had installed in the office an Asterisk server, but the company is
gone and I could keep the server.
However, for my family with three members and two phone lines this
server is overkill. I am looking for a compact solution, which is more
suitable for me.
I want
Felippe Silvestre wrote:
Hi all,
Our users are complaining about beeps that happen in the middle of some
calls. They are similar to the sound heard you are in a call and press
any button in your phone. Please find bellow some examples of these
beeps(the recordings are in Portuguese, but
Steve Totaro wrote:
On Mon, Jul 7, 2008 at 6:58 AM, FaberK [EMAIL PROTECTED] wrote:
Hi folks,
we use meetme application with pin so when a customer joins he's
prompted for his name.
Then the voice say:press one to accept the recording...
My question is, is it possible to cut off that request
Steve Finkelstein wrote:
Hi all,
I was curious if anyone can recommend a company that would work with
small businesses, and capable of using a fallback number (mobile
phone, home number etc) in the event SIP or IAX2 peering was to
terminate because of some outage. This could be useful when
James Sneeringer wrote:
snip /
Also note that asterisk.conf options override command-line options (and
not the other way around, as you might have learned to expect from most
other applications).
Some asterisk.conf options, such as runuser and rungroup, don't appear
to work at all. I can
Stefan Guenther wrote:
Alan Lord wrote:
When I connect to various asterisk services such as VoicemailMain(),
MeetMe() as examples, I do not get to hear the first greeting messages.
I've tried adding a Wait(1) before or after the application but this
seems to have no effect
Sherwood McGowan wrote:
snip /
Hrm...I have encountered this before and sometimes doing an explicit
Answer() then a Wait(2), then calling the service can help.
Hope this is helpful
Sherwood McGowan
Bingo!
Thanks a bunch. That sorted it.
Al
--
The way out is open!
Lee, John (Sydney) wrote:
I was following the instruction on
http://www.voip-info.org/wiki-Asterisk+non-root to re-install my
Asterisk as non-root when I had the following questions/issues:
1) Use your system's preferred method of adding a new user. Examples:
Red Hat: adduser -c Asterisk
Steve Repo wrote:
Hello,
Please forgive me for i'm not an asterisk user yet. I've done as much
research as I can .. and have the following questions.
I'm setting up a new office and a home office and i'm shopping for
hardware.
Office: 2 analog lines
Hardware: TDM412B (2 FXO, 1FXO)
Marco wrote:
Respectfully, I don't agree. I've purchased an original clone :-P of
the X100P card, on the long period they almost always have some
drawbacks... Faxing have been troubling for me. Don't know if it was for
the line or else, but with a Digium card I had no problem at all.
No
Christian wrote:
Hi all,
I have seen discussions on this earlier on, but just want to hear some quick
thoughts.
I am running v1.6 of Asterisk on my Ubuntu installation, I did make config to
make it run at boot. Since I've got a firewall and don't have any other
servers running I am not
Hi all,
I'm seeing a lot of these messages:
[Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response:
Remote host can't match request NOTIFY to call
'[EMAIL PROTECTED]'. Giving up.
[Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response:
Remote host can't match request NOTIFY
Olivier wrote:
Do we agree on the fact you can't change a S68 handset display name (S68
should be the model name of the handset included in a S685IP package)
from a computer ?
If my memory serves me right, you can change S685IP base station
settings but not handset settings (display
Mik Cheez wrote:
Hmph...and it appears no kernel-smp-source exists. You should be able
to compile going to a non-SMP kernel, but there must be a better
solution. I can't believe this hasn't come up before.
Sorry.
You only need the kernel headers in reality I believe. Why not just mail
Grey Man wrote:
On Thu, May 1, 2008 at 7:54 AM, Alan Lord [EMAIL PROTECTED] wrote:
Hi all,
I'm seeing a lot of these messages:
[Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response:
Remote host can't match request NOTIFY to call
'[EMAIL PROTECTED]'. Giving up.
[Apr 30 20
Marco wrote:
Hi Alan,
yeah, latest Siemens DECT phones with VoIP support are quite the new
Chuck Norris of cordless phones. Personally I use the C470IP on a
business context and a C475IP at home (for the integrated answering
machine). The audio quality is amazing, and the extra services
Jaap Winius wrote:
snip / However, if you
want to use the unit's integrated answering machine -- in English -- I
would think that would just be a question of finding and installing a
UK firmware version... unless maybe the pre-recorded messages are
separate files.
Cheers,
Jaap
I
Hi there,
in case anyone is interested, I've just taken ownership of a small home
network (3 handsets) of the brand new Siemens DECT PSTN/VOIP phone.
It works great with Asterisk. Here's my overview and review so far...
http://www.theopensourcerer.com/2008/04/27/siemens-gigaset-685ip-phones/
Kashif Naeem wrote:
Hello All,
A company has two requirements:
1) They are looking to develop its own CRM
2) Second thing is that they want to develop enhancements / new features
in Asterisk like Thirdlane.
What are your comments about technology to be used. Which one would be
most
Tzafrir Cohen wrote:
On Fri, Apr 18, 2008 at 04:15:32PM +0200, giuliano curti wrote:
On Thu, 17 Apr 2008 13:25:19 +0100
Alan Lord [EMAIL PROTECTED] wrote:
[cut]
I bought an X100p card from ..
I have a similar card (X101P Tiger Jet) but seems does not
recognize dmtf: external pstn
Rizwan Hisham wrote:
Hi all,
i want to buy a pci or whatever card for asterisk to plug in my
telephone line into it and use asterisk as a pbx. i have only one
telephone line at home. can you recommend me a simple cheap card which
i can buy in pakistan.
I live in pakistan, and i dont
Tony Mountifield wrote:
snip /
Does anyone know what it would take to make the GUI compatible with IE
as well as FF?
That's a big question.
There are *so many* inconsistencies with IE. See
http://www.positioniseverything.net/ for a good accumulated list of many
of the bugs and hacks to try
Darrick Hartman (lists) wrote:
snip /
I didn't find it too much trouble in a Via C700N system. But I wouldn't
use one of the mainstream distros for the OS. They chew up system
resources just trying to accommodate any hardware.
The solution is to roll-your-own. See this series of articles
Tzafrir Cohen wrote:
snip /
You can easily take a standard distro and remove all the services you
don't really need.
Yes, but you can't easily change the way the apps are built or setup,
e.g. compiler optimisations, use of initrd when not necessary, kernel
bloat just to accommodate any
Gordon Henderson wrote:
On Mon, 31 Mar 2008, Alan Lord wrote:
Also, can you
find 300Gb of solid state storage for about £30. ;-)
Where??
Gordon
Sorry my bad. It was a question...
Al
--
The way out is open!
http://www.theopensourcerer.com
Lenz wrote:
Hello list,
after spending the best part of an afternoon trying to build Asterisk on
an old EPIA VIA C3, I thought that writing a tutorial would make life
easier for future compilers:
http://astrecipes.net/index.php?n=356
I had never compiled Asterisk for a different
Hi all,
I am close to purchasing some new DECT phones for our home office here
in the UK.
We use Asterisk and I am sorely tempted by the Siemens C475IP or the
soon-to-become-available-in-the-uk S685IP.
Both systems have great feature sets and, on-paper at least, look to be
the bee's knees.
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