I've been quite satisfied with one of these:
http://www.pikatechnologies.com/english/View.asp?x=652
On 03/26/2009 5:28 PM, Anthony Plack wrote:
Hey all,
I have a potential project which calls for a very small form-factor computer
like this:
The wiki says it should take about 20 minutes per handset.
Joseph L. Casale wrote:
I started this at 4pm yesterday, its 10am and the handsets still say
they are in progress?
Is that normal?
Thanks!
jlc
Is there any way we can make use of the call forwarding feature on our
Telco phone line. I've seen this question asked on this list before but
looking in the archive i don't see that it has been answered.
If someone has this working or knows how please let me know.
Thanks.
I have found many neat scripts for my home asterisk on the wiki and
elsewhere - and we really like it. But there are a couple of things I'd
still like to find.
And if anyone has some favorites that think think are great for a home
with 2 adults and 3 kids (4 phones) 2 cell phones I'd like to
Do you think you'll outgrow 1 phone line any time soon. If so You'll
want something that you don't have to completely redo when you add the
next line. That digium card you linked to has 2 more expansion slots
open for more lines or phones.
The soho pbx you linked to looks like you can have
Get a sheet of paper, write down your menu structure
find the screen where you can record to asterisk
record your voice menu based on what you wrote down
then go to the ivr menu create a new one that matched what you wrote down
choose the recording that you recorded earlier as the sound that
I've been using one for several years now. It works ok but not
spectacular.
You can almost forget ever sending or receiving faxes thru it tho. They
only work about 5% of the time.
Henry Cobb wrote:
On 6/5/07, Ronaldo [EMAIL PROTECTED] wrote:
Hi all,
I'm planning to buy a X100P clone and
Greg Kennedy wrote:
I gave up on the rxfax business as it never worked for me. I use
iaxmodem and hylafax and it works perfectly, every single time i use
it. inbound or outbound doesnt matter.
I have not read about anyone using iaxmodem and hylafax having any
issues. and its fairly easy to
We have several people in our church that recently became disabled. I am
thinking of setting up an asterisk server and several phone lined so
that they can call in to church during services to listen to the service.
The phone lines at church are also used by our private school during the
David Gomillion wrote:
On 5/17/07, *Tim Litwiller* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
We have several people in our church that recently became
disabled. I am
thinking of setting up an asterisk server and several phone lined so
that they can call in to church
Drew Gibson wrote:
Tim Litwiller wrote:
David Gomillion wrote:
On 5/17/07, *Tim Litwiller* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
We have several people in our church that recently became
disabled. I am
thinking of setting up an asterisk server and several phone
lined
Rudolf Ladyzhenskii wrote:
Hi, all
Stupid question, but how do you exit asterisk console without stopping
the asterisk?
Tried quit and exit:
*CLI exit
No such command 'exit' (type 'help' for help)
*CLI quit
No such command 'quit' (type 'help' for help)
*CLI
Any other ideas?
I started
Not, on your question - but you brought up something I would really like
to do and I was told it wasn't possible.
how do you do the transfer to cell phone with the hook flash.
Martin Roy wrote:
I doubt it's possible but I'll ask just in case there's a legal way
to do that.
I have an
tom wrote:
Andrew Kohlsmith wrote:
On Wednesday 19 April 2006 12:23, Leo Burd wrote:
I'm new to Asterisk and I'm wonder what's the best way to combine multiple
VOIP lines into a single phone number...
Do you mean VOIP lines as in numbers my customers can call to get to
it dials the userid that you put in that field as an extension.
at home I have it set to 100
and then I have this in the extensions.conf
exten = 100,1,Answer
exten = 100,2,Wait(1)
exten = 100,3,VoicemailMain,s${CALLERIDNUM}
exten = 100,4,Macro(hangupcall)
so the user doesn't need to put in a
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Jim Hanlon wrote:
The points I feel are confusing are:
1. The alterations to the config files made via AMP Setup pages are archived in the
Asterisk DBMS, but changes made via the AMP Maintenance pages are not (Apparently. It's
hard to be sure what the rules are). Such differences in behavior
I haven't been having the same problems - but I have been having a
problem with one way sound and no MWI
This is in our home
I have an AAH server on 192.168.3.82
with a sip account 200 that I assigned for this phone.
the phone gets a dhcp assigned static ip of 192.168.3.79
when a call comes in
No, what was rerecorded was the sounds that come with the asterisk
package. Digium has another package called asterisk-sounds that has
many additional sounds - that package was not rerecorded.
Douglas Garstang wrote:
You know, I'm still a little confused. Kristian, the original poster,
Installation is very simple. Simply download the prompts to a
directory on your Asterisk server. Any will do. Once you have
downloaded the formats you desire, simple follow these steps:
cd /var/lib/asterisk/
mv sounds sounds.orig
tar -xvjf /path/to/sounds.tar.bz2
[repeat last step for
Tim Litwiller wrote:
Installation is very simple. Simply download the prompts to a
directory on your Asterisk server. Any will do. Once you have
downloaded the formats you desire, simple follow these steps:
cd /var/lib/asterisk/
mv sounds sounds.orig
tar -xvjf /path/to/sounds.tar.bz2
Tim Litwiller wrote:
Installation is very simple. Simply download the prompts to a
directory on your Asterisk server. Any will do. Once you have
downloaded the formats you desire, simple follow these steps:
cd /var/lib/asterisk/
mv sounds sounds.orig
tar -xvjf /path/to/sounds.tar.bz2
PS. I can call out from the phone to the pstn also.
Tim Litwiller wrote:
All the phones, the iaxy and the server are on a 192.168.3.* network
and the only outside interface currently is the x100p card. I do have
an account with a voip provider but I haven't got that setup up yet
since
I have an iaxy the is across a 802.11b link from my asterisk server.
signal strength is good and it has been working fine there for about a year.
Friday night lightning took out the x100p card in my asterisk server and
I just got it all working again last night. Lucky I had a spare.
Since I
/
iptables setting etc. on your asterisk box. The lightning might be a red
herring.
Michelle
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Litwiller
Sent: Monday, January 30, 2006 2:43 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] help
I just finished reading the Transferring Using Flash thread and I
didn't know you could have asterisk send a flash.
For my home line I have 3 way calling from the phone company and since I
installed Asterisk missed that I could flash and call a 3rd party and
flash again and join them in the
before having to upgrade again.
[EMAIL PROTECTED] wrote:
8 lines for 10 phones is overkillreally
PaulH
- Original Message -
From: Tim Litwiller [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday
from the CO. Instead of 2 mile like Sprint does in the next
county over where I live.
[EMAIL PROTECTED] wrote:
Point taken!
Then an T1/E1 is the way to go.
PaulH
- Original Message -
From: Tim Litwiller [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
works with Asterisk.
I'm thinking I'd need 2 to support 6-8 lines - Or suggest some other
equipment that will provide up to 8 fxo ports and connect to asterisk.
for future projects I'd also like something with 2 fxo ports and 4 - 5
fxs ports - I suppose a digium card would do fine for 2 fxo
asterisk-users@lists.digium.com
Sent: Saturday, January 14, 2006 6:04 PM
Subject: Re: [Asterisk-Users] I need feed back on how an Aastra
VentureIP 4FXO
On Sat, 14 Jan 2006 11:22:51 -0600, Tim Litwiller wrote
works with Asterisk.
I'm thinking I'd need 2 to support 6-8 lines - Or suggest some
Mojo with Horan Company, LLC wrote:
Hi Tim!
Wow, I didn't imagine that asterisk on different systems would use
different date codes for the monitor filenames -- but aah isn't
asterisk ;)
AAH builds Asterisk from source during the install.
I just reinstalled to the latest AAH 2.2 which
Mojo with Horan Company, LLC wrote:
Wow, example by me. I don't read the Wiki enough lately ;)
on this topic - I had to remove a few blank lines in config.php after I
renamed it or I got a header error and nothing displayed - now it is
working as designed
but shows line like
August 20,
a', $timestamp);
$n = substr($l[3], 3, 10);
echo $i. $q, at $r, while connected to
b.$n. /b - a href=monitor/$bListen/a - a
href=\dl.php?fn=$b\Download/a - a
href=\confirm_delete.php?which=$b\Delete/abr\n;
Tim Litwiller wrote:
Mojo with Horan Company
Sometime this winter we want to move our company to asterisk from a very
old comdial executech phone system.
At this point I have a system setup at home that we've been using for
several months.
I've tried the grandstream bt101 but have had problems keeping it
working - some days the message
Doug Meredith wrote:
hugolivude [EMAIL PROTECTED] wrote:
You need to be
careful when buying the Linksys because version 5.0 saw a move from
Linux, which runs Sveasoft's Talisman firmware, to VxWorks, which does
not.
Why would I care what OS an embedded device uses? Is there a
difference in
Brian Capouch wrote:
Tim Litwiller wrote:
Absolutly - you can replace the linux firmware with an opensource
firmware from sveasoft.com and others that will give you many features
that you can't get in the price range of router.
Is sveasoft Open Source?
It didn't used
Chris Bagnall wrote:
Yep - I have one in my junk box. Maybe the SPA-841 would be
a better choice for a few dollars more (haven't played with
one personally, but everything I've heard says that they are
much better than the GS BT's.)
Having used both phones, I'd go the other way. The GS BT is
Nicolás Gudiño wrote:
2) How can I customize the location of the recorded file(s)
I don't know if you can change the location, I think not.
Well, I'd like them to drop in my voicemail when done recording - maybe
in a separate recordings folder but I'd like to use the same interface
to
Bob Goddard wrote:
On Wednesday 12 Oct 2005 14:53, Tom Rymes wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Bob Goddard
Sent: Tuesday, October 11, 2005 6:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
I've been running with a generic X100P for 5 or so months and every once
in a while I have problem receiving faxes. I see that others have the
same problems and some worse than I have with these boards so I was
wondering if using a Sipura SPA-3000 would be any more reliable.
Has anyone had
+ sign to add another trunk to the route
Zeeshan Zakaria wrote:
Hi,
In asterisk at home, in Outbound Routing menu, under the trunk sequence
(e.g. IAX2/FWD), what does little red cross mean beside the selected trunk.
Thanks
Zeeshan
Guillermo Salas M wrote:
On Thu, 2005-09-01 at 09:12 +1000, [EMAIL PROTECTED] wrote:
Hi, all
Here is a something I found on the web:
http://www.voipbuster.com
And it works OK too. Now, I want to use it via asterisk, so I ccan use my
normal phones instead of PC application.
Did anyone try
John Novack wrote:
Tim Litwiller wrote:
I've been using * at home at my house for while and like it but for
work I didn't know the answers to these questions.
But now my new employer is wanting to upgrade a very old phone system
and wants to make sure our new system has some features
I've been using * at home at my house for while and like it but for work
I didn't know the answers to these questions.
But now my new employer is wanting to upgrade a very old phone system
and wants to make sure our new system has some features
I've talked to him about using asterisk and he
this works for me
exten = 100,1,Wait(1)
exten = 100,2,VoicemailMain([EMAIL PROTECTED])
change 100 to whatever you want the common voicemail extension to be. -
the wait keeps the first part of the voice from being chopped off.
Mauro Zanin wrote:
Done both reload and Linux box re-load,
Zoltan Szecsei wrote:
Hi,
Just set up AAH 1.1 using an HFC BRI line and 5 IP phones as per
http://voip-info.org/tiki-index.php?page=ACT+P104+IP+Phone
The dialplan was configured through AMP and has nothing fancy in it.
.clipped
Any help/pointers would be wellcome.
TIA,
Zoltan
some
I'm not a programmer - but it sounds to me like you are all making it to
hard by transfer the voice files around etc. unless you really have to
have the messages stored in the mail server for some reason.
here is what I would picture
a outlook plugin that creates the illusion of several
Kris Boutilier wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Tim
Litwiller
Sent: Thursday, June 09, 2005 9:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voicemail and MS Exchange
Synchronization
On mine it is http://ipofmachine then click the crm link
but I haven't found the default login and password for it yet.
Kanuri, Seshu (Company IT) wrote:
I don't see the SugarCRM being part of the install.
How do you activate this?
Seshu
-Original Message-
From: [EMAIL
*1xx where xx is the speed dial number
Ronald Wiplinger wrote:
I use DbPut and DbGet to get a speed dial number into and out of the
database.
After that I should dial the number, but how?
User dialed
*91 xx - Set system speed dial xx to digits
*91 xx 0 - Delete system speed
No, asterisk can't do that your phone line provider would have to
provide that service.
chawki hammoud wrote:
--- Tim Litwiller [EMAIL PROTECTED] wrote:
Your pstn land line can only handle 1 call at a time
To handle more at the same number you need a
rollover or busy redirect.
Then you could
I have a customer looking for an automated way to provide his customers
information.
He found some windows software called Active Call Center - but I believe
that he did the 40 day trial and it crashed his windows machine to much.
So he wants something that can do a similar task running on
Tim Litwiller wrote:
I have a customer looking for an automated way to provide his
customers information.
He found some windows software called Active Call Center - but I
believe that he did the 40 day trial and it crashed his windows
machine to much. So he wants something that can do
Jim Lists wrote:
Hi All,
Please excuse my newbie questions, but figured this would be the best
place to ask. I have been reading up on VOIP and implementing Asterisk
as a PBX. I'm still left wondering if Asterisk supports multiple lines
at once? If I had one land line, voip line, and asterisk
about the
recipe: it makes reference to some AGI perl scripts. Is the source
available? Or may be it's irrelevant.
Thanks,
Daniel
On Apr 29, 2005, at 9:10 PM, Tim Litwiller wrote:
Daniel Salama wrote:
Question: how can I block someone from calling us?
Sometimes we get crank calls into our office
Daniel Salama wrote:
Question: how can I block someone from calling us?
Sometimes we get crank calls into our office. We'd like to build a list
of callers to be blocked. When they call, they should hear busy and then
we hang up. We have about 100 DIDs routed to different contexts and I
wouldn't
Henry, I'd certainly be interested in your paging code.
Henry Devito wrote:
I am already doing this with AGI, PERL, and PHP to set up the page
groups. I will release the code as open source if people are
interested. I'm not the best PERL scripter in the world but it works.
Colin Anderson wrote:
The AMP configuration didn't work so I decided to work up from the Wiki
example. Can anyone help?
This is a good config using AMP:
http://sourceforge.net/docman/display_doc.php?docid=26418group_id=121515
HTH
I second that - It is working fine for me.
I'm working on it - I only started a week ago - and then I didn't know I
wanted to do all these other things with it. * is adictive!
Art Zemon wrote:
Tim Litwiller wrote:
so where did you put these lines?
exten = _1NXXNXX,1,SetCallerID(4153574000)
exten = _1NXXNXX,2,Dial,IAX2/[EMAIL
so where did you put these lines?
exten = _1NXXNXX,1,SetCallerID(4153574000)
exten = _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
exten = _011.,1,SetCallerID(4153574000)
exten = _011.,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
I want asterisk to use my pots line for local calls and voipjet
timezone. Also what
is your system timezone set for? the tz option sets the zone.
;4200 = 9855,Mark
Spencer,[EMAIL PROTECTED],[EMAIL PROTECTED],attach=no|[EMAIL PROTECTED]|tz=central
- Original Message - From: Tim Litwiller [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non
I tested today with and without my modifications - and all I get is comm
err messages on the sending fax machine -
So now I wonder if asterisk can recieve faxes on my X100p that the phone
line is connected to.
Tim Litwiller wrote:
[EMAIL PROTECTED] has this for it's incoming fax macro
--- start
I just got everything working the way I want except 2 things
#1 the timestamp on voicemail is not the local time zone - I am in US
Central (-6) timezone and the voice mail is timestamped 6 hours ahead of
local time.
#2 incoming faxes - I get a comm err message from several different fax
[EMAIL PROTECTED] has this for it's incoming fax macro
--- start snip ---
[ext-fax]
exten = in_fax,1,GotoIf($[${FAX_RX} = system]?2:analog_fax,1)
exten = in_fax,2,Macro(faxreceive)
exten = in_fax,3,system(tiff2ps -2eaz -w 8.5 -h 11 ${FAXFILE} | ps2pdf
${FAXFILE}.pdf)
exten =
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