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[Jan 27, 2020] How to configure an Asterisk dialplan for intra-office calling Enable Sysadmin

Jan 27, 2020 | www.redhat.com

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:

Image
Network diagram credit: https://extensions.libreoffice.org/extensions/vrt-network-equipment
Dialplan fundamentals

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:

  1. An external call comes into Asterisk from a standard telephone number.
  2. An Interactive Voice Response (IVR) system might ask the user to enter basic information, such as their account number.
  3. Asterisk accepts the user's input. The IVR looks up their account and presents them with information (e.g., information about outstanding invoices).
  4. 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:

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 :

[office-phones]

exten => 1001,1,Dial(PJSIP/alice-softphone)

exten => 1002,1,Dial(PJSIP/bob-softphone)

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:

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:

[alice-softphone]
type=endpoint
context=office-phones
disallow=all
allow=ulaw
auth=alice-auth
aors=alice-softphone
More Linux resources

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:

asterisk-1*CLI> dialplan reload
Dialplan reloaded.

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 <markster@digium.com>
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:

  1. Answers the call.
  2. Plays a hello-world file. This is a sound file included with Asterisk. By default, Asterisk searches for sounds in /usr/lib/asterisk/sounds/ .
  3. 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:

exten => 9000,1,Answer()
exten => 9000,2,Playback(hello-world)
exten => 9000,3,Hangup()

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

[Jan 26, 2020] Can I connect my landline to the PC and receive calls and make phone calls through this (through the PC) - Quora

Jan 26, 2020 | www.quora.com

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a p d v EoA b QCFY y YNDC U P dtacP a aDju e UuZ s mhNd s IC l w e cAOm r x ss A rfap G NkAPl 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...

(Continue Reading) You dismissed this ad. The feedback you provide will help us show you more relevant content in the future. Undo Answer Wiki 6 Answers Terry Lambert

Terry Lambert , Did kernel and Bluetooth stack work on the iPhone, cell modems on Chromebooks Answered Mar 28, 2019 · Author has 11.7k answers and 33.3m answer views

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...

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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 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. 4.9k views · View 4 Upvoters · View Sharers · Answer requested by Alberto Bucciante Related Questions More Answers Below

Dean Rubine 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 Top Writer 2018 Active in 4 Spaces Dean Rubine Dean Rubine 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 Top Writer 2018 Active in 4 Spaces Dean Rubine , former Faculty at Carnegie Mellon School Of Computer Science (1991-1994) Answered Mar 27, 2019 · Author has 4k answers and 4.5m answer views

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...

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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 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

s AUY p REfs o qFqqi n giDS s VkeB o Fg r Zhug e wKr d qgAM bpyN b smpJ y bQe WPRi T vvBy e CRiYl l cY l kg o F LDhH M Gg o Ucl b fLS i F l ims e mr Switch to Tello for unbeatable prices. Build your own plan with minutes, text & data and pay only for what you use. No contract, no catch. Start Now You dismissed this ad. The feedback you provide will help us show you more relevant content in the future. Undo John Nakulski John Nakulski , I worked for Australia's largest phone company. Answered Mar 30, 2019 · Author has 1.1k answers and 1.5m answer 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 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 Alberto Bucciante Studied at Universidad De León Lives in Bristol, UK Alberto Bucciante Related Questions More Answers Below

Charles Verrier Charles Verrier , Worked in IT for 30 years Answered Mar 27, 2019 · Author has 1.6k answers and 413.2k answer views

Yes -but not easily.

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 ...

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Yes -but not easily.

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.

There are devices like this

Computer Telephony CTI Card - USB TAPI-compatible Voice Modem

[Jan 26, 2020] How to connect my telephone line to PC and answer phone calls using headsets

Jan 26, 2020 | www.quora.com

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a RrJ d sP AsNXT b WyA y xIO pek T YH e gQDS l zTzWC l Fy o lBsG osRgW M Pb o mr b F i qmM l t e Ugyn Switch to Tello for unbeatable prices. Build your own plan with minutes, text & data and pay only for what you use. No contract, no catch. Start Now You dismissed this ad. The feedback you provide will help us show you more relevant content in the future. Undo Answer Wiki 3 Answers Altuğ Gür


Altuğ Gür , Avid Bluetooth user for about two decades Answered Dec 13 2016 · Author has 841 answers and 1.9m answer views

This was a rather interesting thing around the end of 90s but quickly lost traction.

There are still solutions but mostly outdated and compatibility/support is questionable so YMMV.

Avanquest classic phone tools promises to deliver this functionality so you may give it a go.

PhoneTools

So does CallSoft.

Call Soft Advanced Voice/Fax Software for Windows

Price range for both products are about $US30.

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...

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This was a rather interesting thing around the end of 90s but quickly lost traction.

There are still solutions but mostly outdated and compatibility/support is questionable so YMMV.

Avanquest classic phone tools promises to deliver this functionality so you may give it a go.

PhoneTools

So does CallSoft.

Call Soft Advanced Voice/Fax Software for Windows

Price range for both products are about $US30.

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.

Hope this helps!

Edit:

Managed to find one!

Amazon.com: Work From Home Office Telephone Call Center Dial Key Pad Phone + Headset Headphone with Mute Volume Control: Electronics 9.2k views · View 1 Upvoter · Answer requested by Chintan Shah Chintan Shah Chintan Shah

[Jan 26, 2020] Connect an Asterisk system to the public switched telephone network

Jan 26, 2020 | www.redhat.com

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.

More Linux resources

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:

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:

[voipms]
type=endpoint
transport=transport-udp
context=inbound-calls
disallow=all
allow=ulaw
auth=voipms
outbound_auth=voipms
aors=voipms

[voipms]
type=registration
transport=transport-udp
outbound_auth=voipms
client_uri=sip:redacted VoIP.ms username@newyork1.voip.ms:5060
server_uri=sip:newyork1.voip.ms:5060

[voipms]
type=auth
auth_type=userpass
username=redacted VoIP.ms username
password=redacted VoIP.ms password

[voipms]
type=aor
contact=sip:redacted VoIP.ms username@newyork1.voip.ms

[voipms]
type=identify
endpoint=voipms
match=newyork1.voip.ms

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:

asterisk-1*CLI> pjsip show registrations

<Registration/ServerURI..............................> <Auth..........> <Status.......>
==========================================================================================

voipms/sip:newyork1.voip.ms:5060                  voipms            Registered

Objects found: 1
Inbound dialplan configuration

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:

  1. Answer an inbound call from the DID that I purchased from my provider.
  2. Allow the user to enter an extension (one for Alice, two for Bob).
  3. 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:

  1. When my DID is called, Asterisk matches the 1234567890 extension. It answers the call.
  2. Asterisk then plays a built-in "hello" message. In a real environment, you might want to record your own prompts for Asterisk to use.
  3. 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.
  4. 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:

Want more on networking and connectivity topics? Check out the Linux networking cheat sheet .

[Oct 23, 2019] Google's New Voice Recorder App Transcribes in Real Time, Even When Offline

Oct 23, 2019 | tech.slashdot.org

(techcrunch.com) 27

Posted by msmash on Tuesday October 15, 2019 @06:10PM from the pushing-the-limits dept.

At Google's hardware event this morning , the company introduced a new voice recorder app for Android devices , which will tap into advances in real-time speech processing, speech recognition and AI to automatically transcribe recordings in real time as the person is speaking. From a report:

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.

[Nov 09, 2017] How To Convert Youtube Videos To MP3 Audio Files by Gary Newell

Oct 25, 2017 | www.lifewire.com

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

sudo apt-get install youtube-dl

Note that sudo may or may not be required. Read this guide to find out more about the sudo command .

Red Hat, Fedora, CentOS and Other Derivatives

sudo yum install youtube-dl

OpenSUSE

sudo zypper install youtube-dl

Arch, Manjaro and Other Derivatives

sudo pacman -S youtube-dl

How To Download And Extract Audio From A Youtube Video

Find the video you want to download by visiting Youtube and searching for it.

Underneath the video, there will be a share link.

Click on the share link and select the link that appears with the mouse.

Open a terminal window and type the following command replacing the text paste_link_here with the link to the share link:

youtube-dl --extract-audio --audio-format mp3 -l paste_link_here

To paste into a Linux terminal right click and select paste from the menu or press shift and insert at the same time.

The text in the terminal should look something like this:

youtube-dl --extract-audio --audio-format mp3 -l https://youtu.be/ICZ3vFNpZDE

What this does is download the video from Youtube as a video and then extract just the audio part to MP3.

You are left with just the MP3 file and the video is discarded.

If you want to keep the video, as well as the MP3 file, add a -k to the command as follows:

youtube-dl --extract-audio --audio-format mp3 -k -l https://youtu.be/ICZ3vFNpZDE

You will now be left with an MP3 and an MP4 file.

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.

There are similar add-ons available for Firefox .

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.

When you have enough MP3 files you might want to read this guide to get the best out of Rhythmbox .

[Dec 26, 2016] Google and Amazon Will Let You Rent a Movie for $0.99 During the Holiday

Dec 26, 2016 | tech.slashdot.org
(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.

[Dec 26, 2016] British Film Institute To Digitize 100,000 Old TV Shows Before They Disappear

Dec 26, 2016 | news.slashdot.org
(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.

[Dec 23, 2016] http://www.vidtomp3.com/index.php

Dec 23, 2016 | www.vidtomp3.com

The video sites that are currently supported are: YouTube , MegaVideo , Dailymotion , Metacafe , Veoh , Myspace , Break , iFilm , Bolt , ClipJunkie , ClipShack , CollegeHumor , FunnyJunk , Glumbert , GoFish , Grouper , Hallpass , MilkandCookies , Putfile , SantaBanta , Sevenload , Sharkle , Shoutfile , Vimeo , vSocial , Yikers , ZippyVideos ... and loads more coming soon!

[Jul 15, 2016] Why everyone is crazy for Prisma, the app that turns photos into works of art

www.theguardian.com

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.

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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.

Donald Trump and Indiana governor Mike Pence addressing the crowd during a campaign stop in Indiana.

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Donald Trump and Indiana governor Mike Pence addressing the crowd during a campaign stop in Indiana. Photograph: Reuters

[Dec 07, 2015] How to produce a YouTube series with open source tools

A look behind the scenes of Dototot's The Hello World Program, a YouTube channel aimed at computer science education.

sharing websites

Jan 20, 2009

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)"

Audio Rippers and Encoders in Ubuntu Linux

Choose the DVD Ripper That's Right for You

11 Video Players for Linux - A Review

(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."



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Bulletin:

Vol 25, No.12 (December, 2013) Rational Fools vs. Efficient Crooks The efficient markets hypothesis : Political Skeptic Bulletin, 2013 : Unemployment Bulletin, 2010 :  Vol 23, No.10 (October, 2011) An observation about corporate security departments : Slightly Skeptical Euromaydan Chronicles, June 2014 : Greenspan legacy bulletin, 2008 : Vol 25, No.10 (October, 2013) Cryptolocker Trojan (Win32/Crilock.A) : Vol 25, No.08 (August, 2013) Cloud providers as intelligence collection hubs : Financial Humor Bulletin, 2010 : Inequality Bulletin, 2009 : Financial Humor Bulletin, 2008 : Copyleft Problems Bulletin, 2004 : Financial Humor Bulletin, 2011 : Energy Bulletin, 2010 : Malware Protection Bulletin, 2010 : Vol 26, No.1 (January, 2013) Object-Oriented Cult : Political Skeptic Bulletin, 2011 : Vol 23, No.11 (November, 2011) Softpanorama classification of sysadmin horror stories : Vol 25, No.05 (May, 2013) Corporate bullshit as a communication method  : Vol 25, No.06 (June, 2013) A Note on the Relationship of Brooks Law and Conway Law

History:

Fifty glorious years (1950-2000): the triumph of the US computer engineering : Donald Knuth : TAoCP and its Influence of Computer Science : Richard Stallman : Linus Torvalds  : Larry Wall  : John K. Ousterhout : CTSS : Multix OS Unix History : Unix shell history : VI editor : History of pipes concept : Solaris : MS DOSProgramming Languages History : PL/1 : Simula 67 : C : History of GCC developmentScripting Languages : Perl history   : OS History : Mail : DNS : SSH : CPU Instruction Sets : SPARC systems 1987-2006 : Norton Commander : Norton Utilities : Norton Ghost : Frontpage history : Malware Defense History : GNU Screen : OSS early history

Classic books:

The Peter Principle : Parkinson Law : 1984 : The Mythical Man-MonthHow to Solve It by George Polya : The Art of Computer Programming : The Elements of Programming Style : The Unix Hater’s Handbook : The Jargon file : The True Believer : Programming Pearls : The Good Soldier Svejk : The Power Elite

Most popular humor pages:

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The Last but not Least Technology 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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Last modified: October 23, 2019