20210314 : Augmented Reality Gets Pandemic Boost - WSJ by the pandemic found immediate uses for the distanced expertise AR technology can provide Mercedes-Benz USA trained more than 1,200 automotive technicians at all 383 dealerships last summer on how to use AR HoloLens 2 headsets for remote assistance. PHOTO: MERCEDES-BENZ USA By Sara Castellanos Jan. 28, 2021 3:46 pm ET
Before starting, you will need to install some dependencies required to build the Asterisk server. You can install all of them
using the following command:
Once all the packages are installed, you will need to install DAHDI to communicate Asterisk with analog and digital telephones.
First, download the latest version of DAHDI to the /opt directory:
cd /opt
git clone -b next git://git.asterisk.org/dahdi/linux dahdi-linux
Next, change the directory to the downloaded directory and compile it using the following command:
cd dahdi-linux
make
make install
Next, download the DAHDI tools with the following command:
cd /opt
git clone -b next git://git.asterisk.org/dahdi/tools dahdi-tools
Once the download is completed, configure and install it with the following command:
cd dahdi-tools
autoreconf -i
./configure
make install
make install-config
dahdi_genconf modules
Next, download the LibPRI to communicate Asterisk with ISDN connections.
git clone https://gerrit.asterisk.org/libpri libpri
cd libpri
Next, install it using the following command:
make
make install
Once all the necessary tools are installed, you can proceed to install Asterisk.
Install Asterisk
By default, the Asterisk package is not included in the Ubuntu 20.04 default repository. So you will need to download and compile
it from the source. You can download it from the Git Hub using the following command:
Next, you will need to select the modules that you want to install with Asterisk. You can select it using the following command:
make menuselect
You can use the Arrow key to navigate and Enter key to select the modules.
Select and enables the Addons as shown below:
Next, enable the Core sound modules:
Next,
enable the additional MOH packages:
Next,
enable the Extra Sound Packages:
Now, click on the Save and Exit button.
Next, build the Asterisk using the following command:
make -j2
Next, Asterisk and its modules using the following command:
make install
You should get the following output:
+---- Asterisk Installation Complete -------+
+ +
+ YOU MUST READ THE SECURITY DOCUMENT +
+ +
+ Asterisk has successfully been installed. +
+ If you would like to install the sample +
+ configuration files (overwriting any +
+ existing config files), run: +
+ +
+ For generic reference documentation: +
+ make samples +
+ +
+ For a sample basic PBX: +
+ make basic-pbx +
+ +
+ +
+----------------- or ---------------------+
+ +
+ You can go ahead and install the asterisk +
+ program documentation now or later run: +
+ +
+ make progdocs +
+ +
+ **Note** This requires that you have +
+ doxygen installed on your local system +
+-------------------------------------------+
You can also install the documentation and basic PBX config files with the following command:
make samples
make basic-pbx
Next, install the Asterisk init script with the following command:
make config
Next, update the shared libraries using the following command:
ldconfig
Create Asterisk User
It is always recommended to run Asterisk as a standalone user for security reasons.
First, create a new Asterisk user with the following command:
At this point, Asterisk is installed and configured. Now, you can start the Asterisk service using the following command:
systemctl start asterisk
You can also enable the Asterisk service to start at system reboot with the following command:
systemctl enable asterisk
To check the status of the Asterisk service, run the following command:
systemctl status asterisk
You should get the following output:
â -- asterisk.service - LSB: Asterisk PBX
Loaded: loaded (/etc/init.d/asterisk; generated)
Active: active (running) since Sun 2021-05-16 12:24:29 UTC; 13s ago
Docs: man:systemd-sysv-generator(8)
Process: 60668 ExecStart=/etc/init.d/asterisk start (code=exited, status=0/SUCCESS)
Tasks: 46 (limit: 4691)
Memory: 34.7M
CGroup: /system.slice/asterisk.service
â""â"€60685 /usr/sbin/asterisk -U asterisk -G asterisk
May 16 12:24:29 ubuntu2004 systemd[1]: Starting LSB: Asterisk PBX...
May 16 12:24:29 ubuntu2004 asterisk[60668]: * Starting Asterisk PBX: asterisk
May 16 12:24:29 ubuntu2004 asterisk[60668]: ...done.
May 16 12:24:29 ubuntu2004 systemd[1]: Started LSB: Asterisk PBX.
Now, connect to the Asterisk command line utility with the following command:
asterisk -vvvr
Once connected, you should get the following output:
Asterisk GIT-18-78d7862463, Copyright (C) 1999 - 2021, Sangoma Technologies Corporation and others.
Created by Mark Spencer
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk GIT-18-78d7862463 currently running on ubuntu2004 (pid = 60685)
ubuntu2004*CLI>
Congratulations! You have successfully installed and configured Asterisk server on Ubuntu 20.04 VPS.
Of course, you don't have to install and set up Asterisk if you use one of our
Managed VPS Hosting services, in which case you can
simply ask our expert Linux admins to install Asterisk on Ubuntu 20.04, for you. They are available 24Ã -- 7 and will take care of
your request immediately. If you're looking to find something else, such as
how to install Magento
or Elasticsearch on
Ubuntu 20.04, check out our blog.
If you liked this post on how to install Asterisk on Ubuntu 20.04, please share it with your friends on the social networks using
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cat playlist.txt | while read line; do youtube-dl --extract-audio --audio-format mp3 -o
"%(title)s.%(ext)s" ytsearch:"$line" ;done
[Mar 14, 2021] Augmented Reality Gets Pandemic Boost - WSJ by the pandemic found immediate uses for the distanced expertise AR technology can provide Mercedes-Benz USA trained more than 1,200 automotive technicians at all 383 dealerships last summer on how to use AR HoloLens 2 headsets for remote assistance. PHOTO: MERCEDES-BENZ USA By Sara Castellanos Jan. 28, 2021 3:46 pm ET
The Morning Download delivers daily insights and news on business technology from the CIO
Journal team.
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Last summer, L'Oréal also began using
Microsoft
Corp.'s
HoloLens
2 headset to help employees install and troubleshoot manufacturing equipment with assistance from experts in
different parts of the world.
While wearing the HoloLens 2 headset, users can see data, instructions and 3-D visual images in their
real-world view. They can manipulate digital objects by using their fingers to grab the corners of the object
and drag it over to one side, among other gestures. With remote-assistance software, a user wearing a headset
can share their real-time view with others who are using a desktop or mobile device.
The world-wide total market value for augmented reality is expected to grow to $140 billion by 2025, up from
about $10 billion last year, according to a report this month from tech market advisory firm Allied Business
Intelligence Inc. Those figures include hardware, software and content, AR advertising, platforms and
licensing, connectivity and much more.
The hardware includes headsets such as Microsoft's HoloLens, and Glass, made by
Alphabet
Inc.'s
Google.
For smart glasses alone, ABI said world-wide shipments last year totalled around 1.8 million units; it
forecasts that will rise to 27 million in 2025.
The expected growth is attributed partly to the lasting impacts of the pandemic over the next few years, said
Eric Abbruzzese, a research director at ABI Research.
https://tpc.googlesyndication.com/safeframe/1-0-37/html/container.html
MORE FROM CIO JOURNAL
The jolt higher would also be due to new products and advances in the technology over the next few years, he
said.
Increased demand for augmented reality in some sectors is also part of a wider digital transformation in
businesses, along with investments in cloud-computing and videoconferencing, triggered by the pandemic.
"We don't see it slowing down," said Paul Travers, chief executive of Vuzix Corp., which makes
augmented-reality glasses. In the fourth quarter of last year, Vuzix's sales doubled to over $4 million
compared with the same period in 2019, he said.
Mr. Travers said customers will continue to use the products even after the pandemic. "This is the beginnings
of an inflection point for this industry," he said. "There's no doubt about it."
Microsoft saw a 44-fold rise in remote-assistance usage of HoloLens 2 between January and December of last
year, largely because of social-distancing and lockdown requirements amid the pandemic, the company said.
It's not going to be a uniform rocket to the moon, but in some areas we're seeing extremely fast growth.
-- Charlie Han, Microsoft HoloLens
Demand has increased in industries such as auto and semiconductor manufacturing, where it is being used for
remote guidance on complex assembly tasks and new installations, said Charlie Han, principal program manager of
Microsoft HoloLens.
"It's not going to be a uniform rocket to the moon, but in some areas we're seeing extremely fast growth," he
said.
Mercedes-Benz USA, a subsidiary of Daimler AG, trained more than 1,200 automotive technicians at all 383 U.S.
dealerships last summer on how to use HoloLens 2 headsets for remote assistance.
In the past, a technician might have to wait days for a field service engineer to travel to the dealership to
help with a complex problem, such as fixing a car's transmission or software, said Christian Treiber, vice
president of customer services at Mercedes-Benz USA.
Now, 60% of complex problems can be solved within 24 hours with the HoloLens 2 headset, Mr. Treiber said. A
technician with a headset at any dealership can connect right away with one of several specialists around the
country. Through remote-assist software, the specialist can see on a desktop or tablet what the technician is
seeing using the headset.
"It's a guided repair, which helps the dealership technician be more efficient and effective," said Mr.
Treiber.
The plan to distribute HoloLens 2 headsets to dealerships had been in effect since 2019, but the technology
became more valuable during the pandemic, he said. Still, he said it would be around three years for AR
headsets to become as widely used as other tools.
"The HoloLens is not yet accepted as a tool like maybe a wrench or screwdriver or diagnostic laptop," Mr.
Treiber said. "We're not there yet."
By the way, if you want to download a Twitter video, you can use this online tool: Twitter Video Downloader
All it does is break out the video into its own tab in the browser, and you can
right-click and save the video. Facebook video is the same, using this online tool: Facebook Video
Downloader
And there are other tools of course, and there always will be. Information wants to be
free, and it will be.
Storage is so cheap that it makes sense to download everything to watch at leisure and to
keep for review.
~~
What made the social media platforms so successful is that people love to communicate and
share. And the platforms behaved like jealous gods, grabbing images and videos into their own
caches, not caring about copyright or royalties, not caring about destroying the quality of
the original, just grabbing it all and then creating walled gardens - so that Pinterest, for
example, can't save an image from Facebook - an image that was uploaded from a private
individual - but to hell with ownership rights or credits, for these gods.
First they gave it all away free to get a lot of people using their platform, then they
monetized it with ads or subscriptions. We're dealing with merchants here, greedheads - petty
tyrants, not real tyrants. Fuck 'em. Let them kill themselves from their own gluttony.
We'll just move on. And yes, there are places to go. There is still more of this world and
this human experience to explore and inhabit.
Leave these losers to their own hell. And that goes for the politicians, spooks, fixers,
players, grifters, bankers and secret societies as well. They will all destroy
themselves.
The University of California, Santa Barbara's David Tadres and Matthieu Louis designed a
virtual reality (VR) system for presenting environments to small, freely moving animals like
flies and fish larvae during optogenetic experiments.
The customizable Raspberry Pi VR system (PiVR) combines a behavioral environment, a camera,
a Raspberry Pi microcomputer, a light-emitting diode (LED) controller, and a touchscreen
display.
The researchers employed the system to explore sensory navigation in response to gradients
of chemicals and light in various animals.
Said Tadres and Louis, "
Our goal has been to make virtual reality paradigms accessible to
everyone, from professional scientists to high-school students. PiVR should help democratize
cutting-edge technology to study behavior and brain functions." Full
Article
How to configure an Asterisk dialplan for intra-office calling
Learn how to configure Asterisk to let two softphones call each other.
Posted January 21, 2020
|
by
Anthony Critelli
Image
"Telephone - Amalgamated Wireless of Australasia, 300 CBT, circa 1940"
is licensed under
CC
BY 4.0
In the
previous article
, you learned how to
configure the PJSIP channel driver to connect a simple softphone client with your Asterisk
installation. However, your phones still can't call each other, and you haven't given them numerical
"extensions" yet. Connecting channels together in Asterisk is the work of the dialplan. In this
article, you'll learn the basics of the dialplan: What it is, how it's configured, and how to use it
to connect phones together.
As a reminder, this is the setup we're configuring:
The Asterisk
dialplan
is responsible for routing calls, so it is often referred to as the
heart of an Asterisk system. The dialplan is written in a special scripting language, and it is
extremely powerful. You might think of phone systems as simply accepting and connecting calls, but
Asterisk is capable of much more. With the dialplan, you can design rich, voice-driven applications.
For example, you could create the following call flow for a small business:
An external call comes into Asterisk from a standard telephone number.
An
Interactive Voice Response (IVR)
system might ask the user to enter basic
information, such as their account number.
Asterisk accepts the user's input. The IVR looks up their account and presents them with
information (e.g., information about outstanding invoices).
Optionally, the user can be routed to a
queue
of available customer service
representatives. When the employee receives the customer's call, the system provides them with all
of the customer's details and saves the caller some time.
While there are other programming interfaces for interacting with Asterisk, the dialplan is the
most basic, and understanding it is fundamental to understanding how Asterisk handles calls. According
to
Asterisk the
Definitive Guide
, there are four fundamental components to the Asterisk dialplan:
Contexts:
A
context
is a logical section in the dialplan. Contexts
contain one or more
extensions
.
Extensions:
An
extension
is simply a grouping of steps used to handle
a particular call. Unlike many traditional phone systems, extensions don't have to be numerical and
they aren't tied to a single device. You could have an extension called "mainIVR" if you wanted.
Priorities:
A
priority
is a step in an extension. Priorities handle
ordering, and they can also have labels attached to them so that a call can jump between priorities
as needed.
Applications:
Applications are a lot like functions in traditional programming
languages. They tell Asterisk
what
to do with a call. For example, you might have an
internal extension of *86. When *86 is dialed, you might have Asterisk play a message of the day
using the
Playback application
.
If you're new to Asterisk, this breakdown probably sounds complicated. While Asterisk dialplans
certainly can be complex, a simple phone system only requires a simple dialplan. Let's take a look at
the dialplan needed to support your intra-office calling scenario. The dialplan is configured in
/etc/asterisk/extensions.conf
:
The snippet above is all that is necessary to allow your two phones to call each other. Let's step
through each part of this dialplan:
office-phones
is the context. This context contains two extensions.
1001
and
1002
are the extensions. Notice how this setup decouples the
numbers from the phones themselves. While these numbers are currently hardcoded to dial Alice's and
Bob's phones, they could just as easily be used for more complex routing (e.g., automatically roll
over to an overnight line during certain hours).
The
1
after the extension is the priority. Remember, a priority is just a step in
extension handling. The first priority is always
1
. In this example, each extension
only has a single priority.
Dial
is the application. The
Dial
application
is used to ring a remote device. Dialing occurs via SIP or other signaling
protocols (if you need a refresher on VoIP protocols, head over to our [first article]).
[Note: Don't forget to add the link.]
To recap:
When a call comes into the
office-phones
context, Asterisk
tries matching that call to an extension. When extension 1001 is dialed, the first step (priority)
tells Asterisk to dial the PJSIP endpoint for Alice's phone. When extension 1002 is dialed, the same
thing happens for Bob's phone.
This is great so far, but how exactly does a call make its way
into
the dialplan? The
answer lies in the PJSIP endpoint configuration from the previous article:
Notice that the context for each phone is set to
office-phones
. This setting tells
Asterisk that any calls coming from the
alice-softphone
or
bob-softphone
endpoints should enter the dialplan in the
office-phones
context. When Bob dials a number
(say, 9000) from his softphone, Asterisk looks in the
office-phones
context for the
matching extension 9000. In the sample dialplan above, this call will fail because there is no
matching extension.
You don't have to configure all of your phones to enter the dialplan in the same context. In fact,
you'll likely find good reasons to specifically put phones in other contexts. Consider a business that
wants to only allow certain people to make international calls, while everyone else is restricted to
local calls. You might have two extensions: One to allow unrestricted calling, and one that only
allows calls to numbers that start with the local area code. Those with international calling
privileges would be placed in the
international
context, while everyone else would be
placed in the
local-only
context.
That was a lot of theory. Let's get back to the command line and test out the changes that we made
to the dialplan. First, you must non-disruptively
reload
the dialplan to enact the
changes you made in the config file:
Next, you can inspect the dialplan directly from the Asterisk CLI to ensure that your changes are
present:
asterisk-1*CLI> dialplan show office-phones
[ Context 'office-phones' created by 'pbx_config' ]
'1001' => 1. Dial(PJSIP/alice-softphone) [extensions.conf:3]
'1002' => 1. Dial(PJSIP/bob-softphone) [extensions.conf:5]
-= 2 extensions (2 priorities) in 1 context. =-
Notice that Asterisk includes the exact file name and line number where an extension and its
priority can be found. This information is useful when troubleshooting behavior in your phone system.
With the dialplan reloaded and your changes clearly in place, you should be able to place a test
call from Linphone (or whatever SIP endpoint you're using). First, launch the Asterisk CLI with extra
verbosity using
asterisk -rvvv
:
[root@asterisk-1 asterisk]# asterisk -rvvv
Asterisk 16.6.1, Copyright (C) 1999 - 2018, Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 16.6.1 currently running on asterisk-1 (pid = 4138)
Next, place a call from Alice's phone to extension 1002. Assuming that you registered an additional
softphone (or physical phone) for Bob, the extension should show as
ringing
:
Image
The Asterisk CLI also prints informational messages about the call's progression since it was set
to verbose mode. You can see the inbound call being handled by the dialplan and handed off to the
PJSIP channel driver to dial Bob's softphone. Eventually, once Bob answers, Asterisk bridges the audio
for the call together so that both parties can hear each other:
== Setting global variable 'SIPDOMAIN' to 'asterisk-1.acritelli.com'
-- Executing [1002@office-phones:1] Dial("PJSIP/alice-softphone-00000008", "PJSIP/bob-softphone") in new stack
-- Called PJSIP/bob-softphone
-- PJSIP/bob-softphone-00000009 is ringing
-- PJSIP/bob-softphone-00000009 is ringing
-- PJSIP/bob-softphone-00000009 answered PJSIP/alice-softphone-00000008
-- Channel PJSIP/bob-softphone-00000009 joined 'simple_bridge' basic-bridge <edd9402c-6df0-4fff-a81e-57826dadc652>
-- Channel PJSIP/alice-softphone-00000008 joined 'simple_bridge' basic-bridge <edd9402c-6df0-4fff-a81e-57826dadc652>
-- Channel PJSIP/bob-softphone-00000009 left 'native_rtp' basic-bridge <edd9402c-6df0-4fff-a81e-57826dadc652>
-- Channel PJSIP/alice-softphone-00000008 left 'native_rtp' basic-bridge <edd9402c-6df0-4fff-a81e-57826dadc652>
== Spawn extension (office-phones, 1002, 1) exited non-zero on 'PJSIP/alice-softphone-00000008'
asterisk-1*CLI>
You have now created enough Asterisk configuration to allow both of your phones to call each other.
Congratulations!
Adding another extension
You've now seen basic dialplan configuration that allows two phones to call each other. I also
mentioned a few times that Asterisk decouples the concept of a physical phone from an extension
because an extension is simply a set of instructions in the dialplan. Let's add another simple
extension to the dialplan to see exactly what I mean:
[office-phones]
exten => 1001,1,Dial(PJSIP/alice-softphone)
exten => 1002,1,Dial(PJSIP/bob-softphone)
exten => 9000,1,Answer()
same => n,Playback(hello-world)
same => n,Hangup()
The above configuration adds an additional extension (9000) to the dialplan. When this extension is
dialed, Asterisk:
Answers the call.
Plays a
hello-world
file. This is a sound file included with Asterisk. By default,
Asterisk searches for sounds in
/usr/lib/asterisk/sounds/
.
Hangs up the call.
Notice the use of the
same => n
syntax. This is a common and helpful bit of syntactic
sugar in the dialplan. Remember that each extension has one or more
priorities
, or steps,
associated with it. The
same => n
syntax saves you some typing and tells Asterisk that
this step is just the next priority for the same extension. The above configuration could also be
written as:
With your new configuration in place, reload the dialplan and try dialing extension 9000 to see
what happens. Again, the key concept to understand is that you have created an extension that has no
physical device associated with it. Asterisk fully decouples the concept of devices and extensions.
Wrapping up
In this article, you learned about the Asterisk dialplan and wrote enough dialplan configuration to
enable two phones to call each other. The Asterisk dialplan is extremely powerful, allowing you to
build rich communications applications. I strongly recommend that you check out the official
Asterisk dialplan
documentation
and the fifth edition of
Asterisk: The Definitive Guide
to help you better
understand everything that the dialplan has to offer.
In the [next article], you'll work on connecting your phone system to an external provider to
enable inbound and outbound calling.
[Note: Don't forget to add the link.]
Topics:
Networking
VOIP
Anthony Critelli
Anthony Critelli is a Linux systems engineer with interests in automation, containerization,
tracing, and performance. He started his professional career as a network engineer and
eventually made the switch to the Linux systems side of IT. He holds a B.S. and an M.S.
More about me
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Which tools are used for monitoring the
network traffic and issue in an enterprise network?
You can use any
number of
tools to monitor
your company's network. Some of
these tools specialize in just one thing. Wireshark, for example,
records and analyzes data traffic.
Co...
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Terry Lambert
,
Did kernel and Bluetooth stack work on the iPhone, cell
modems on Chromebooks
Answered Mar 28, 2019
·
Author has
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answers and
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Yes.
A landline
is an analog telephony line, and is usually called a POTS (
Plain
Old Telephone Service
) line.
You can put
an analog telephony card into a PC that has slots for cards.
This is
pretty much how you build your own PBX using
Asterisk
.
You can also
use digital telephony cards to provide connectivity for VOIP
handsets within your office.
Typically
these handsets are what are known as "IP Phones".
If you want
to do this with a laptop, you can do it with a USB dongle.
You can
still find USB interfaces "voice modems", such as the US
Robotics USR5637 56K USB Controller Dial-Up External Fax Mo...
A
landline is an analog telephony line, and is usually called
a POTS (
Plain Old Telephone Service
) line.
You can
put an analog telephony card into a PC that has slots for
cards.
This is
pretty much how you build your own PBX using
Asterisk
.
You can
also use digital telephony cards to provide connectivity for
VOIP handsets within your office.
Typically
these handsets are what are known as "IP Phones".
If you
want to do this with a laptop, you can do it with a USB
dongle.
You can
still find USB interfaces "voice modems", such as the US
Robotics USR5637 56K USB Controller Dial-Up External Fax
Modem with Voice available for sale at various locations.
There are
also cheaper versions, but they are a crap shoot as to
whether they're going to work very well or not. The Conexant
ones, which are used in the Imported520 products are about
1/6th the price of the US Robotics, and tend to have good
reviews. I've never personally use one.
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Dean Rubine
Former Faculty at Carnegie
Mellon School Of Computer
Science
1991-1994
Studied at Carnegie Mellon
University
Graduated
1991
Lives in New Hampshire
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Dean Rubine
Former Faculty at Carnegie
Mellon School Of Computer
Science
1991-1994
Studied at Carnegie Mellon
University
Graduated
1991
Lives in New Hampshire
4.5m content views
224.9k
this month
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Writer
2018
Active
in 4 Spaces
Dean Rubine
,
former Faculty at Carnegie Mellon School Of Computer Science
(1991-1994)
Answered Mar 27, 2019
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You
certainly used to be able to back in the nineties. Back then
computers usually came with RJ11 jacks to plug them into the
phone network so you could access dialup services, even dialup
internet. Some of the modems evolved into general phone
devices, supporting two way voice audio as well as data.
I actually
won a computer in 1993; it was an ASUS with Windows 3.11 if I
recall, which was subtitled "Windows for Multimedia" or some
such. It had a telephone app (we called them programs back
then) that let you use the computer as a speakerphone and
answering machine. I think it used the then n...
You
certainly used to be able to back in the nineties. Back then
computers usually came with RJ11 jacks to plug them into the
phone network so you could access dialup services, even
dialup internet. Some of the modems evolved into general
phone devices, supporting two way voice audio as well as
data.
I
actually won a computer in 1993; it was an ASUS with Windows
3.11 if I recall, which was subtitled "Windows for
Multimedia" or some such. It had a telephone app (we called
them programs back then) that let you use the computer as a
speakerphone and answering machine. I think it used the then
new TAPI API to control it.
The prize
computer has this very goofy feature where it would actually
boot up when the phone rang, presumably to fire up the
answering machine and take a message. But it took a good
fraction of a minute to boot; any caller had long hung up.
These
days if you want to use your computer as a phone it's
generally a VOIP thing like Skype, going out over the
internet to some server which can access the regular phone
network. But some offices have VOIP with a local PBX
connected to shared local POTS (Plain Old Telephone Service)
lines right in the office.
You need
to find the right device to let you do it these days. You
can probably find a TAPI modem with USB on one end and RJ11
on the other and some software to make it work. I haven't
tried it for 25 years so I won't try to be more specific.
2.6k views
You've
described what a Reverse ATA is and does, approximately. I've
seen this question here on Quora before. Such devices used to
exist. You could put together a solution that does this today,
at a little cost and some effort.
You're
better off ditching your landline and using Skype or a VoIP
service.
If it's for
an elderly person or someone with hearing loss ot a disability,
get a seniors phone or a seniors video phone.
1.2k views
·
Answer requested by
Alberto Bucciante
Chris Summers
,
Worked in cellphone sales and service for 5 years, computer
also for 15 years.
Answered May 24, 2019
·
Author has
7.9k
answers
and
5m
answer views
PCs use to
come with a 56k modem that did just that. You could still do
that either by getting a card for a tower or by a USB adapter.
There are several programs out there that will also work with
this set to use the line as your telephone and you could use a
headset with a mic to use the computer as your telephone too.
You would need to look on a site like Newegg or tigerdirect for
the cards or adapters.
1.3k views
·
Answer requested by
Alberto Bucciante
Studied at Universidad De León
Lives in Bristol, UK
Alberto Bucciante
Related Questions
More
Answers Below
It used to
be called CTI (Computer Telephony Integration) and you could do
things like get your PC to dial numbers from a database, or
automatically recognise incoming calls and display the contact
record for the caller.
There was a
software standard called TAPI (Telephony Application
Programmers Interface) that standardised the software/hardware
interaction, although it suffered from quite bad fragmentation
in different manufacturers.
This kind of
technology has largely been replaced or superseded by VOIP, so
it may be tricky to find hardware or software that still talks
...
It used
to be called CTI (Computer Telephony Integration) and you
could do things like get your PC to dial numbers from a
database, or automatically recognise incoming calls and
display the contact record for the caller.
There was
a software standard called TAPI (Telephony Application
Programmers Interface) that standardised the
software/hardware interaction, although it suffered from
quite bad fragmentation in different manufacturers.
This kind
of technology has largely been replaced or superseded by
VOIP, so it may be tricky to find hardware or software that
still talks to good old fashioned landlines now.
However, if
you're open to suggestions, I'd recommend small landline
extension devices that connect directly to phone. They come
with a simple but functional headset with mic. You just need to
connect t...
However,
if you're open to suggestions, I'd recommend small landline
extension devices that connect directly to phone. They come
with a simple but functional headset with mic. You just need
to connect the small dialpad unit to the handset slot of
your phone, plug the headset into the dialpad unit and
you're done. This setup is not prone to software crashes,
incompatibilities or performance issues yet they allow you
to make and answer phone calls easily while leaving your
hands free.
Been more
than 20 years since I've used one so I'm not sure about the
availability.
Learn how to set up Asterisk so your softphones can receive incoming calls from outside and make outgoing
calls outside your organization as well.
Posted January 23, 2020
|
by
Anthony Critelli
Image
"Telephone Pole"
by
Fried
Dough
is licensed under
CC
PDM 1.0
In two previous articles, you learned how to configure
two SIP phones
and the
Asterisk dialplan
to enable the phones to
call each other. Having two phones that can call each other is great, but most organizations want to
connect their phone system to the
public switched
telephone network (PSTN)
to allow for inbound and outbound calling to others outside of the
organization. In this article, you will learn some background about how to connect to the PSTN and you
will see the Asterisk configuration needed for this connectivity.
Note:
I've mentioned
it several times throughout this series, but it is especially important for this article to understand
how to secure your phone system. Connecting to a telephony provider can open you up to all sorts of
issues, such as toll fraud, if you aren't careful about controlling who can connect to your phone system
(e.g., via firewall rules). Be sure to read the
README-SERIOUSLY.bestpractices.md
documentation that comes with the Asterisk source code and understand every piece of configuration that
you add to Asterisk.
Connecting to the PSTN
Obtaining a telephone number and connecting your phone system to the PSTN might sound like a difficult
task, but it's actually pretty easy. Larger organizations may choose to use a local telecommunications
provider or their local Internet company, but there are also a variety of online providers available
(e.g.,
Twilio
and
VoIP.ms
, among many others). I will be using VoIP.ms for
the phone numbers in this article.
No matter who you choose, the process is fairly straightforward. You will buy a phone number (often
called a direct inward dial, or DID), and you'll configure your phone system to connect with the
provider's phone system via SIP. This setup is often referred to as a
SIP trunk
. The process of
configuring your phone system to work with your chosen provider can vary, so it's always best to consult
the provider's documentation or work with their voice engineers if you run into trouble.
Let's take a look at the general best practices for configuring your phone service. These are not
comprehensive, but they should provide a base of sound advice when working with your provider:
Obtain the provider's endpoint IP (or IPs) and heavily restrict SIP and RTP communication to only
those IP addresses. Some providers also support only allowing your account to connect from a specified
list of IPs.
Ensure that your SIP connection minimally uses a username and password for security. If your
provider supports mutual TLS authentication, investigate that option as well.
Don't sign up for features that you do not need. If you won't be placing international calls, then
don't sign up for international calling. Doing this lessens your attack surface if your phone system
is ever breached. Toll fraud can be costly. You can avoid the risk of expensive international calls by
opting out of international calling.
Investigate the fraud detection and prevention options that your provider has available. Some
providers can immediately alert you if they detect any suspicious phone calls being placed from your
account.
PJSIP configuration
The first step in configuring PSTN connectivity is to define the SIP configuration necessary for
Asterisk to communicate with the IP telephony provider. This information will vary a bit by provider, but
many of them provide information about the parameters that you need (VoIP.ms actually provides
Asterisk-specific instructions
):
Image
In my case, the configuration in
/etc/asterisk/pjsip.conf
looks like this:
Most of this configuration probably
looks
familiar
, but this configuration does introduce two new PJSIP section types: a
registration
and an
identify
. The
registration
section tells Asterisk to explicitly register
with the upstream voice provider's server. The
identify
section tells Asterisk that SIP
traffic coming from
newyork1.voip.ms
should match the
voipms
endpoint.
After reloading PJSIP, I can see that my local Asterisk server successfully registered with the
provider's SIP infrastructure. Note that issues during this stage of the process are fairly common, and
you may need to work with your provider to understand what "they see" so that you can effectively
troubleshoot:
Next, Asterisk has to be told what to do with incoming and outgoing calls. Incoming and outgoing calls
in Asterisk aren't fancy, they are just extensions in the dialplan like any other extension. I will
discuss incoming calls first. Like any programming language, it's important to understand what your goals
are before you start writing code. In my case, I wanted a dialplan that would:
Answer an inbound call from the DID that I purchased from my provider.
Allow the user to enter an extension (one for Alice, two for Bob).
Call the appropriate user once the extension was dialed.
Notice that in the above PJSIP configuration, I am using the
inbound-calls
context for
calls from my SIP provider. Here's what that context looks like in
/etc/asterisk/extensions.conf
:
[inbound-calls]
exten => 1234567890,1,Answer()
same => n,Playback(hello)
same => n,WaitExten(30)
same => n,Hangup()
exten => 1,1,Answer()
same => n,Dial(PJSIP/alice-softphone)
exten => 2,1,Answer()
same => n,Dial(PJSIP/bob-softphone)
Notice that I replaced my real DID with the fake 1234567890 for privacy reasons. You will want to use
your real DID if you are following along. Also, notice that my phone number follows a United States
numbering convention (a three-digit area code followed by a seven-digit number).
Let's step through this contents of this configuration together:
When my DID is called, Asterisk matches the 1234567890 extension. It answers the call.
Asterisk then plays a built-in "hello" message. In a real environment, you might want to record
your own prompts for Asterisk to use.
Asterisk then calls the
WaitExten
application
with a value of 30. This value tells Asterisk to wait up to 30 seconds for the user to
enter an extension.
Assuming the user enters an extension of "1" or "2", the dialplan will jump to that extension.
Notice that the "1" extension will call Alice's softphone, and the "2" extension will call Bob's
softphone.
Once the above configuration is in place, you can reload the dialplan and place a test call to your
PSTN phone number. You should hear Asterisk say "hello," and you should be able to dial one of the
extensions and have the phone on the other end ring.
Outbound dialplan configuration
Being able to place inbound calls is great, but most businesses expect outbound calling functionality
as well. Luckily, this is fairly easy to configure.
First, recall from the previous article about intra-office calling that outbound calls for Alice and
Bob's phones enter the dialplan in the
office-phones
context. Therefore, we need a way to
match all outbound calls in this context and send them to the upstream provider.
Until now, you've seen extensions configured as exact matches. Extension 1001 matches and dials Alice,
extension 1002 matches and dials Bob, and the extension for your inbound DID matches and allows the
caller to input a selection. It would obviously be impractical to match every single possible outbound
number that a user might dial. Thankfully, Asterisk provides a way to perform
pattern matching.
To provide outbound calling for American numbers -- e.g., (123) 456-7890 -- my
office-phones
context looks like this:
[office-phones]
exten => 1001,1,Dial(PJSIP/alice-softphone)
exten => 1002,1,Dial(PJSIP/bob-softphone)
exten => 9000,1,Answer()
same => n,Playback(hello-world)
same => n,Hangup()
exten => _XXXXXXXXXX,1,Set(CALLERID(all)="Anthony Critelli <1234567890>")
same => n,Dial(PJSIP/${EXTEN}@voipms)
If you already took a look at the Asterisk wiki's linked pattern matching documentation, this syntax
will look familiar. The outbound extension matches
_XXXXXXXXXX
, which is a fancy way of
saying: "Match a sequence of 10 digits." The underscore indicates that this character is the beginning of
a pattern match, similar to the use of the forward-slash (/) in many programming languages to indicate
the use of a regular expression. Therefore, this extension will match any 10-digit extension that I send
to it, such as standard 10-digit US phone numbers. A more complete example would also account for country
codes, local calling, and other considerations. That is an exercise left up to the reader.
The first priority in this extension sets the
CALLERID
to a string of my choosing. You
should replace the name and number with your own name and DID. This practice also introduces another
concept in the Asterisk dialplan: The use of variables. In the configuration above, both
CALLERID
and
EXTEN
are variables that you have available to manipulate.
The second, and final, priority in the extension simply sends the call to the upstream provider via
the PJSIP channel driver. This syntax is pretty straightforward, and it looks a lot like the dialplan
that you already wrote to allow two phones to call each other.
With this configuration in place, you can reload your dialplan and try placing a test outbound call
from Alice or Bob's phone. Assuming that everything went well, your call should succeed.
Wrapping up
If you've stuck with me from
the
beginning
, then you have successfully implemented a basic phone system using the open source Asterisk
PBX. You've come a long way. From understanding VoIP and Asterisk basics to installing and configuring
Asterisk from scratch, you should now have an idea of how to configure a simple phone system. Minimally,
I hope this series has sparked your interest in learning more about telephony and Asterisk.
If you want to continue your VoIP and Asterisk journey, then I recommend that you check out these
resources:
Videos from
AstriCon
, the Asterisk Users Conference, may help to spark some ideas about telephony projects.
The
Wireshark
packet analysis tool has a
trove of utilities
for analyzing
phone calls. These include call ladders, traffic analysis, and even media playback.
Finally, if you are interested in understanding the fundamental protocols that support VoIP
networks, you should review the
IETF RFCs
for a
technical deep-dive. There are tons of VoIP-related RFCs, but the SIP, SDP, and RTP/RTCP RFCs are good
starting points.
The improvements will allow users to take better advantage of the phone's voice
recording functionality, as it will be able to turn the recordings into text even when
there's no internet connectivity. This presents a new competitor to others in voice
transcriptions that are leveraging similar AI advances, like Otter.ai, Reason8, Trint and
others, for example.
As Google explained, all the recorder functionality happens directly on the device --
meaning you can use the phone while in airplane mode and still have accurate
recordings.
"This means you can transcribe meetings, lectures, interviews, or anything you want to
save," said Sabrina Ellis, VP of Product Management at Google.
The Recorder app was demonstrated onstage during the event, live, and was offering --
from what was shown -- an error-free transcription.
There is a Linux command line application called youtube-dl, which stands for Youtube downloader.
This can be used to
download youtube
videos .
It may be the case that you want to download just the audio from the video file to be played on
your portable music devices such as an MP3 player or smartphone.
This article will show you how to download just the audio from a Youtube video.
Copyright should be adhered to when downloading and extracting the audio from videos but that
doesn't make the practice illegal.
For instance, there are lots of how-to guides that are created in video format and certain video
bloggers create content that is just as accessible in pure audio format. An example would be something
like the Linux
Action Show podcast although they do actually provide an MP3 download link from their site.
The best tool for converting Youtube videos to MP3 is youtube-dl.
How To Install The Youtube Downloader
You can download and install youtube-dl using the following commands:
Debian, Ubuntu, Linux Mint, Zorin and Other Derivatives
There are other ways to download and convert videos from Youtube to MP3.
For instance, there are a number of
Chrome plugins
which can be used. The upside to using an add-on is that it is generally a one-click process
with no command line knowledge required.
One such add-on is called Youtube To MP3. This will place a little icon in the address bar when
you can convert a video to MP3.
Simply find the video you wish to convert and then press the icon. This takes you to
http://www.theyoump3.com/ which performs
the conversion. When the conversion has completed you can download the MP3 file.
Personally, I think the youtube-dl tool is the cleanest solution. Not all of the add-ons in the
Chrome store and within Firefox are safe and many of the add-ons are supported by adverts.
If you like the youtube-dl command, here are another
15 Linux terminal commands that will rock your world . Learn how to run commands in the background,
run commands at a certain time and how to download files from the web from the command line.
(fortune.com)
61
Posted by msmash
on Friday December 23, 2016 @01:00PM
from the
movie-time
dept.
Holidays are the time when many of us get some extra time to catch up on all
the good movies and TV shows that our friends wanted us to watch, but we never
did. To make things enticing for people, Amazon and Google are
prepping for lowering the rent
this holiday season. From a report on
Fortune:
Google and Amazon -- which have been jostling to compete with
Netflix in video streaming -- have announced a new online promotion for the
holidays. Google Play will allow viewers to select any movie available on the
streaming service to rent for $0.99. Amazon Video will allow viewers the same
using the promo code "MOVIE99." The catch: Users can only get one movie per
account. Both services offer an extensive list of titles, including this year's
Suicide Squad, Bridge Jones's Baby, and Finding Dory among others.
(bbc.com)
124
Posted by
BeauHD
on Wednesday November 30, 2016 @08:40PM
from
the
quick-before-it's-too-late
dept.
An anonymous reader quotes a report from BBC:
Thousands of British TV
programs
are to
be digitized before they are lost forever
, the British Film Institute says.
Anarchic children's show Tiswas and The Basil Brush Show are among the programs
in line for preservation. The initiative was announced as part of the BFI's
five-year strategy
for 2017-2022. "Material from the 70s and early 80s is
at risk," said Heather Stewart, the BFI's creative director. "It has a five or
six-year shelf life and if we don't do something about it will just go, no
matter how great the environment is we keep it in. "Our job is make sure that
things are there in 200 years' time." The BFI has budgeted $14.3 million of
Lottery funding towards its goal of making the UK's entire screen heritage
digitally accessible. This includes an estimated 100,000 of the "most at-risk"
British TV episodes and clips held on obsolete video formats. The list includes
"early children's programming, little-seen dramas, regional programs and the
beginnings of breakfast television." The issue for the BFI, Ms Stewart added,
was also to do with freeing up storage space. "We have a whole vault which is
wall-to-wall video. If we digitized it, it would be in a robot about the size
of a wardrobe," she said.
People across the world are turning amateur photos into elaborate works of art with a new
viral app that relies on AI technology to let users instantly transform mundane images into
Picasso paintings.
Prisma, an app that has attracted
1 million daily users as of Thursday, is reinventing the concept of filtering photos with
technology. While the concept of adding filters to photos has been around for years, the Prisma
iOS app is unique in the way that it relies on a "combination of neural networks and
artificial intelligence" to remake the image.
What that means is the Prisma tools aren't the kind of art filters that Instagram uses where
the filters overlay the original photo. Instead, Prisma goes through different
layers and
recreates the photo from scratch, according to the app makers, who are based in Moscow.
"We do the image fresh,"
Prisma
co-founder Alexey Moiseenkov said in an interview Thursday. "It's not similar to the Instagram
filter where you just layer over … We draw something like a real artist would."
Moiseenkov, 25, is part of a team of four founders who built the app. It was first released in
June, but has skyrocketed in popularity over the last week, with Prisma-altered photos spreading
on Facebook, Twitter and Instagram.
The app is easy to use and functions similarly to Instagram, the Facebook-owned photo-sharing
app that has more than
400 million users.
Users can take photos through the app or pick one from their camera roll. After cropping your
image, you then choose one of 33 filters, such as impression, mosaic and gothic, along with
filters modeled after specific iconic paintings, like the
Great Wave or
The Scream. Prisma will continue to add new filters in the coming weeks, Moiseenkov added.
An artistic
take on the
now famous photo of a demonstrator protesting the shooting death of Alton Sterling in Baton
Rouge. Photograph: Reuters
After the app adds the filter, you can adjust the intensity and then post to Instagram or
Facebook.
Since Prisma has spread, some have
complained that the app could devalue the work of real artists and
take away work from painters who make art by hand – not within seconds on a smartphone.
But for now, the app remains hugely popular, and Moiseenkov said he expects its user base to
continue its rapid growth.
Moiseenkov's background is computer science and he's not an artist himself. But he said he
grew up loving painting and that his favorite artist is Camille Pissarro, the Danish-French
impressionist.
"People want to create something, and we allow them to experiment," he said.
A still from
Kanye West and Kim Kardashian in West's music video for Famous. Photograph: Tidal
The developers are also working on expanding its filter technology to video, with an
innovation that hasn't been done before in any sophisticated manner.
Moiseenkov published a 360-degree
image on Facebook, which offers a glimpse of how Prisma video filters may work in the future.
While there have long been apps that allow users to add filters to footage, such as basic
color changes through
iMovie, the Prisma technology could dramatically expand this concept through videos that
create an entire world that appears intricately painted in every shot.
"Video is … an easier way to express yourself," Moiseenkov said. "It's dynamic. It's not just
a photo or static picture … It's really cool that you can create something in motion."
The co-founder said he wasn't ready yet to offer details on when the video feature would be
released or how exactly it would function, but he said he expects it to be very popular.
Moiseenkov said he also hoped to eventually expand the technology to virtual reality.
SUSE & openSUSE: "ClipGrab is a free GUI Video downloader from the famous online video
sharing websites like youtube, veoh, Dailymotion, MyVideo and many other. ClipGrab is very simple
(basically just one window to work on)"
(Jun 30, 2008, 19:45 UTC) (2343 reads) (0
talkbacks) (feedback) Echoes: "SMPlayer is a fully featured video player built using the Qt 4 libraries. It basically
plays anything, including DVDs and ISO images, but it skipped the menus when reading DVD ISOs. It's
complete and it offers plenty configuration options, including for subtitles and the interface itself,
allowing to choose the icon set and the style used. You can also configure the language SMPlayer
uses for its interface, including English, Romanian, Polish, German and many other."
The Last but not LeastTechnology is dominated by
two types of people: those who understand what they do not manage and those who manage what they do not understand ~Archibald Putt.
Ph.D
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Next, enable the additional MOH packages:
Next, enable the Extra Sound Packages:
Dean Rubine
John Nakulski , I worked for Australia's largest phone company. Answered Mar 30, 2019 · Author has 1.1k answers and 1.5m answer views
Chris Summers , Worked in cellphone sales and service for 5 years, computer also for 15 years. Answered May 24, 2019 · Author has 7.9k answers and 5m answer views
Alberto Bucciante
Charles Verrier , Worked in IT for 30 years Answered Mar 27, 2019 · Author has 1.6k answers and 413.2k answer views
