• Programming VoIP phones for Asterisk using FreePBX

    This is a how to video demonstrating how to configure a SIP compliant phone ( Cisco SPA504 ) to work with a VoIP install. In this video, I am using PBX in a Flash with IncrediblePBX installed. This uses FreePBX as its web interface and makes configuration easy. Anyone using Elastix will have a similar configuration just by clicking on the PBX tab from the main screen.

    published: 27 Apr 2015
  • How to setup a PBX (step by step guide)

    Full series: http://www.ringroost.com/how-to-setup-a-pbx.php Learn how to setup your office PBX. Taylor walks through how to use RingRoost to setup your own PBX - for really cheap.

    published: 18 Jul 2014
  • How to get started on VoIP development in .NET languages (including C#, C, C++, VB, etc.)

    Dig deeper: http://www.voip-sip-sdk.com Hello Everyone! Let's get VoIP development started within 5 minutes if you want to add some VoIP funcationalities into your existing phone system instead of replacing your whole PBX. This video explains you briefly how to get started on VoIP programming in .NET languages (including C#, C, C++, VB, etc.). After watching this tutorial, you will find out all the necessary tasks you need to do to be able to create your own VoIP application as well as a softphone, call recorder, an IVR, or even a PBX. If you are looking for more: goo.gl/VV1TAY About the software: Ozeki VoIP SIP SDK is a high-performance VoIP SDK for .NET developers that can be used to create such a powerful VoIP applications as a webphone, a softphone, an IVR menu system, a call cente...

    published: 18 Jul 2014
  • 5 asterisk basics - asterisk tutorial

    Entire playlist: http://www.youtube.com/view_play_list?p=9E241EDD7EDF8E67 A few basics of getting setup. Asterisk can be activated by typing "asterisk" From there you'll have to reload the asterisk program each time you make a configuration change, the sequence is as follows. type "asterisk -r" type "reload" lastly type "exit" to exit the asterisk console Also keep in mind the two asterisk configuration files you'll be dealing with. They are located in the "/etc/asterisk" directory. The first configuration file is "sip.conf", which defines general settings for asterisk and it defines the different users that will register to your server. The second configuration file is "extensions.conf", which defines the different numbers that the users can dial, and what to do when they dial that numb...

    published: 09 Mar 2011
  • Controlling an Arduino from Asterisk PBX

    This is an example of controlling an Arduino with an ethernet shield using Asterisk PBX. To check out the code and find out more, visit my site: www.nerdybynature.com

    published: 13 Mar 2009
  • Asterisk PBX on Kingston 32GB SD Card - quick setup - step by step

    Asterisk on Kingston 32GB SD Card - quick setup - step by step guide for Raspberry Pi Micro Server. 1) Format your SD card - I use SD Formater. Get it here: https://www.sdcard.org/downloads/formatter_4/ 2) Burn Asterisk image onto your SD. Use Win32Disk Imager: http://sourceforge.net/projects/win32diskimager/ 3) Code getting you down? Use Webmin instead of typing in code. Webmin is a web-based interface for system administration for Unix/Linux. Latest versions: http://www.webmin.com/download.html I use it because it's point and click. Super simple.

    published: 30 May 2015
  • Asterisk and the Social Internet of Things

    Asterisk support for WebRTC and web socket based SIP clients combined with cloud connectivity enablement has allowed us to build out a distributed voice-enabled social network, with social connections between organizations and colleagues, and Asterisk-based PBXs. Using an IoT inspired model, our Asterisk-based PBXs participate in a pure web-tech based social network, wanderweb.net, by signing into the same https service as ordinary human users. Javascript is served up to a node based shell to enable web-driven code-on-demand applications to run on-site. The wanderweb.net IoT client is a code-on-demand application delivered to the Node.js shell on the Asterisk PBX. It authenticates with, then opens a secure web socket to wanderweb.net. It also opens web sockets to Asterisk as required for S...

    published: 03 Jan 2017
  • FreePBX 101 - Part 9 - How to Forward Calls to a Cell Phone

    This is a short video, but a very important feature! How to use Misc. Destinations to forward calls to external phone numbers (mostly used for cell phones). FreePBX 101 - Part 1: https://www.youtube.com/watch?v=LsfqSnGZ3dI FreePBX 101 - Part 2: https://www.youtube.com/watch?v=xBny4hKCM3A FreePBX 101 - Part 3: https://www.youtube.com/watch?v=WgWovGKz5v4 FreePBX 101 - Part 4: https://youtu.be/uNlygMYvNlk FreePBX 101 - Part 5: https://youtu.be/aFrMecTpoyk FreePBX 101 - Part 6: https://youtu.be/EH5XrhtUiSo FreePBX 101 - Part 7: https://youtu.be/YXNTlA3kPWI FreePBX 101 - Part 8: https://youtu.be/8ht-26pBOko Visit http://CrosstalkSolutions.com for FreePBX support, installation, and consulting. Legal mumbo jumbo: FreePBX® is a Registered Trademark of Sangoma Technologies.

    published: 24 Jun 2015
  • DEF CON 24 - Weaponize Your Feature Codes

    Nicholas Rosario (MasterChen), VoIP Administrator Almost everyone is familiar with feature codes, also known as star codes, such as *67 to block caller ID or *69 to find out who called you last. What if the feature codes could be used as a weapon? Caller ID spoofing, tDOSing (Call flooding), and SMS flooding are known attacks on phone networks, but what happens when they become as easy to launch as dialing *40? Weaponize Your Feature Codes will first take the audience through a brief history of feature codes and common usage, and then demonstrate the more nefarious applications. The presentation will share the Asterisk code used to implement these "rogue" features, and mention possible ways of mitigation. While this talk builds upon previous work from the author, referenced in past DEF ...

    published: 19 Jan 2017
  • Raspberry PI with CISCO 7940 running Asterisk on 32GB SD card 5 trunks

    Raspberry PI running Asterisk on 32GB SD card. Here is what you need. Shopping List: 1x Raspberry PI 1x 32 GB SD Card With RasPBX and Webmin just because 1x 2.1A 5V USB Power adapter 1x USB to micro-USB cable 1x CISCO 7940 or 7960 or both and more 1X CISCO SF302-08P 8 PoE Port Switch Loads of hours of studding networking, VoIP and a lot of patience to configure it to work good. I did not cover dynamic dns in this video. Too much to cram in just over 10 min but my experience with dynamic extensions is that not all pre-paid phones allow VoIP or even VPN. The only one that worked so far was O2 with Text & Web bolt on. In a nut shell. RasPBX acts as DHCP and tftp server. On many levels of setting up DHCP I'm not even going to go into. If you can live with a Linksys phone adapter and an...

    published: 29 Aug 2014
  • Audiocodes MP112 Gateway Setup | VoIP Supply

    Join Senior VoIP Engineer Marc Spehalski as he uses the AudioCodes MP112 FXS Gateway to make an analog phone ring with the RenegadePBX. http://www.voipsupply.com/audiocodes-mp-112-fxs Hi, I’m Marc Spehalski, Senior VoIP Engineer at VoIP Supply. Today, we are going to take a look at the AudioCodes MP112 FXS Gateway and we are going to be doing a basic configuration using an analog phone, and communicating with the RenegadePBX. First, lets take a look at what’s in the back. We have our power port, Ethernet port, two FXS ports for analog devices, and a reset button, for setting to factory defaults. Next, we will plug in the MP112 FSX Gateway to the power on our network so we can get started. Also, plug in the analog device you’re going to use. I’m using this analog telephone. Our...

    published: 15 Dec 2015
  • Simple Asterisk SIP Peering Trunking HowTo

    A video illustrating how-to set up an Asterisk PBX using SIP Peering Trunking. --------- From the extension.conf file ------------------------------------------ Extension names may be numbers, letters, or combinations ; thereof. If an extension name is prefixed by a '_' ; character, it is interpreted as a pattern rather than a ; literal. In patterns, some characters have special meanings: ; ; X - any digit from 0-9 ; Z - any digit from 1-9 ; N - any digit from 2-9 ; [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9) ; . - wildcard, matches anything remaining (e.g. _9011. matches ; anything starting with 9011 excluding 9011 itself) ; ! - wildcard, causes the matching process to complete as soon as ; it can unambiguously determine that no ot...

    published: 06 Sep 2013
  • Installation and Configuration of Avaya 1600/9600 Series IP Telephones (NON-DHCP) - Avaya PBX - HD

    Video Topic: Avaya 1600/9600 Series IP Telephones (NON - DHCP) System: Avaya Communication Manager I show you how to install, configure and update 1600/9600 series IP Phones using various tools and methods. This can be translated to other Avaya PBX's as well. Just refer to the Avaya Aura Communication Manager Feature Description and Implementation manual for more information. http://support.avaya.com, Click on Documentation and select the system and version for your PBX. Links to tools I use or talk about in this video: Avaya Firmware for the 1600/9600 IP Phones: http://support.avaya.com EditPad Lite: http://www.editpadlite.com/ Aprelium Web server: http://www.aprelium.com This video and my other videos cover the basic functions of the Avaya PBX. I am a former Tier 3 Engineer ...

    published: 15 Nov 2011
  • Visual Dialplan 2 overview (for Asterisk PBX)

    Visual dialplan is modern rapid application development platform for Asterisk dialplan development. Simply drag, drop and connect dialplan blocks to make company IVR, use ASR and TTS, create call center queues, inbound and outbound call flows, and much more. Model, don't code! And then deploy Asterisk Dialplan with a single press of a button using integrated SSH client. Visual Dialplan is the only Asterisk GUI that makes Asterisk dialplan development easier, without limiting use of any native Asterisk dialplan functionality.

    published: 06 Oct 2008
  • Raspberry PI to PBX Basic Install (Asterisk) Phone Server

    I do have some more vids on my channel www.youtube.com/whackstar . This video is a remake where I turn a Raspberry PI into a FreePBX phone server! It worked and was relatively easy for tech head with little Linux experience. I used the following sources to help build the server. I want to say thank you to the guys at Raspberry PI it is truly a great little machine and your site was very helpful. I also want to say thanks to the guys at Raspberry Pi to Asterisk. Your image worked amazingly well! I will be sending you a thank you in the form of a paypal donation sometime soon. Parts Amazon Links: Raspberry PI Cases: http://www.amazon.com/s/ref=bl_sr_pc?... Raspberry PI: http://www.amazon.com/gp/product/B009... SD Card 64Gb (I mean it!): http://www.amazon.com/gp/product/B008... Charger: I...

    published: 19 Jan 2014
  • How to Set-up an Enterprise Asterisk-based PBX in 10 Minutes (including coffee break)

    http://www.xorcom.com - In this video we demonstrate how fast and easy it is to set up an enterprise level Asterisk-based PBX system. In just ten minutes a technician connects a system supporting four PRI lines and 96 FXS analog extensions, complete with analog and IP video phones! The secret to the configuration speed and simplicity is the fact that Xorcom's XPP USB2 drivers are embedded in all standard Asterisk distributions. These drivers enable the PBX to automatically detect the telephony ports. See for yourself!

    published: 28 Jun 2011
Programming VoIP phones for Asterisk using FreePBX

Programming VoIP phones for Asterisk using FreePBX

  • Order:
  • Duration: 8:27
  • Updated: 27 Apr 2015
  • views: 5461
videos
This is a how to video demonstrating how to configure a SIP compliant phone ( Cisco SPA504 ) to work with a VoIP install. In this video, I am using PBX in a Flash with IncrediblePBX installed. This uses FreePBX as its web interface and makes configuration easy. Anyone using Elastix will have a similar configuration just by clicking on the PBX tab from the main screen.
https://wn.com/Programming_Voip_Phones_For_Asterisk_Using_Freepbx
How to setup a PBX (step by step guide)

How to setup a PBX (step by step guide)

  • Order:
  • Duration: 21:09
  • Updated: 18 Jul 2014
  • views: 157916
videos
Full series: http://www.ringroost.com/how-to-setup-a-pbx.php Learn how to setup your office PBX. Taylor walks through how to use RingRoost to setup your own PBX - for really cheap.
https://wn.com/How_To_Setup_A_Pbx_(Step_By_Step_Guide)
How to get started on VoIP development in .NET languages (including C#, C, C++, VB, etc.)

How to get started on VoIP development in .NET languages (including C#, C, C++, VB, etc.)

  • Order:
  • Duration: 4:37
  • Updated: 18 Jul 2014
  • views: 7391
videos
Dig deeper: http://www.voip-sip-sdk.com Hello Everyone! Let's get VoIP development started within 5 minutes if you want to add some VoIP funcationalities into your existing phone system instead of replacing your whole PBX. This video explains you briefly how to get started on VoIP programming in .NET languages (including C#, C, C++, VB, etc.). After watching this tutorial, you will find out all the necessary tasks you need to do to be able to create your own VoIP application as well as a softphone, call recorder, an IVR, or even a PBX. If you are looking for more: goo.gl/VV1TAY About the software: Ozeki VoIP SIP SDK is a high-performance VoIP SDK for .NET developers that can be used to create such a powerful VoIP applications as a webphone, a softphone, an IVR menu system, a call center, a PBX or different kind of mobile technologies. For more information please visit: http://www.voip-sip-sdk.com Or send an e-mail at: info@voip-sip-sdk.com
https://wn.com/How_To_Get_Started_On_Voip_Development_In_.Net_Languages_(Including_C_,_C,_C_,_Vb,_Etc.)
5 asterisk basics - asterisk tutorial

5 asterisk basics - asterisk tutorial

  • Order:
  • Duration: 5:17
  • Updated: 09 Mar 2011
  • views: 27908
videos
Entire playlist: http://www.youtube.com/view_play_list?p=9E241EDD7EDF8E67 A few basics of getting setup. Asterisk can be activated by typing "asterisk" From there you'll have to reload the asterisk program each time you make a configuration change, the sequence is as follows. type "asterisk -r" type "reload" lastly type "exit" to exit the asterisk console Also keep in mind the two asterisk configuration files you'll be dealing with. They are located in the "/etc/asterisk" directory. The first configuration file is "sip.conf", which defines general settings for asterisk and it defines the different users that will register to your server. The second configuration file is "extensions.conf", which defines the different numbers that the users can dial, and what to do when they dial that number. Please note this is a simplification of the above files, they can get complicated.
https://wn.com/5_Asterisk_Basics_Asterisk_Tutorial
Controlling an Arduino from Asterisk PBX

Controlling an Arduino from Asterisk PBX

  • Order:
  • Duration: 1:03
  • Updated: 13 Mar 2009
  • views: 12446
videos
This is an example of controlling an Arduino with an ethernet shield using Asterisk PBX. To check out the code and find out more, visit my site: www.nerdybynature.com
https://wn.com/Controlling_An_Arduino_From_Asterisk_Pbx
Asterisk PBX on Kingston 32GB SD Card - quick setup - step by step

Asterisk PBX on Kingston 32GB SD Card - quick setup - step by step

  • Order:
  • Duration: 11:34
  • Updated: 30 May 2015
  • views: 2415
videos
Asterisk on Kingston 32GB SD Card - quick setup - step by step guide for Raspberry Pi Micro Server. 1) Format your SD card - I use SD Formater. Get it here: https://www.sdcard.org/downloads/formatter_4/ 2) Burn Asterisk image onto your SD. Use Win32Disk Imager: http://sourceforge.net/projects/win32diskimager/ 3) Code getting you down? Use Webmin instead of typing in code. Webmin is a web-based interface for system administration for Unix/Linux. Latest versions: http://www.webmin.com/download.html I use it because it's point and click. Super simple.
https://wn.com/Asterisk_Pbx_On_Kingston_32Gb_Sd_Card_Quick_Setup_Step_By_Step
Asterisk and the Social Internet of Things

Asterisk and the Social Internet of Things

  • Order:
  • Duration: 42:45
  • Updated: 03 Jan 2017
  • views: 24
videos
Asterisk support for WebRTC and web socket based SIP clients combined with cloud connectivity enablement has allowed us to build out a distributed voice-enabled social network, with social connections between organizations and colleagues, and Asterisk-based PBXs. Using an IoT inspired model, our Asterisk-based PBXs participate in a pure web-tech based social network, wanderweb.net, by signing into the same https service as ordinary human users. Javascript is served up to a node based shell to enable web-driven code-on-demand applications to run on-site. The wanderweb.net IoT client is a code-on-demand application delivered to the Node.js shell on the Asterisk PBX. It authenticates with, then opens a secure web socket to wanderweb.net. It also opens web sockets to Asterisk as required for SIP registrations. The webs sockets are bridged in order to deliver bi-directional SIP communications between the Asterisk instance and a SIP proxy in the cloud. The same web socket is also used for OAM and HTTP connection to the platform. The rapid advances of ARM SoC based platforms, driven by mobile applications, has resulted in the availability of many powerful, low cost, energy efficient platforms that are ideal for Asterisk applications. At the same time, Linux distributions such as Ubuntu now have standardized ARM architecture software ecosystems. We have released a quad core ARM based PBX/Gateway product that runs Asterisk and the rest of our software stack, as used on our more traditional Intel architecture products. This includes the WebRTC and IoT client features. With this, we are able to offer an extremely low-cost on-site solution that supports WebRTC based UC mobility through its interaction with wanderweb.net.
https://wn.com/Asterisk_And_The_Social_Internet_Of_Things
FreePBX 101 - Part 9 - How to Forward Calls to a Cell Phone

FreePBX 101 - Part 9 - How to Forward Calls to a Cell Phone

  • Order:
  • Duration: 3:12
  • Updated: 24 Jun 2015
  • views: 14314
videos
This is a short video, but a very important feature! How to use Misc. Destinations to forward calls to external phone numbers (mostly used for cell phones). FreePBX 101 - Part 1: https://www.youtube.com/watch?v=LsfqSnGZ3dI FreePBX 101 - Part 2: https://www.youtube.com/watch?v=xBny4hKCM3A FreePBX 101 - Part 3: https://www.youtube.com/watch?v=WgWovGKz5v4 FreePBX 101 - Part 4: https://youtu.be/uNlygMYvNlk FreePBX 101 - Part 5: https://youtu.be/aFrMecTpoyk FreePBX 101 - Part 6: https://youtu.be/EH5XrhtUiSo FreePBX 101 - Part 7: https://youtu.be/YXNTlA3kPWI FreePBX 101 - Part 8: https://youtu.be/8ht-26pBOko Visit http://CrosstalkSolutions.com for FreePBX support, installation, and consulting. Legal mumbo jumbo: FreePBX® is a Registered Trademark of Sangoma Technologies.
https://wn.com/Freepbx_101_Part_9_How_To_Forward_Calls_To_A_Cell_Phone
DEF CON 24 - Weaponize Your Feature Codes

DEF CON 24 - Weaponize Your Feature Codes

  • Order:
  • Duration: 42:10
  • Updated: 19 Jan 2017
  • views: 87
videos
Nicholas Rosario (MasterChen), VoIP Administrator Almost everyone is familiar with feature codes, also known as star codes, such as *67 to block caller ID or *69 to find out who called you last. What if the feature codes could be used as a weapon? Caller ID spoofing, tDOSing (Call flooding), and SMS flooding are known attacks on phone networks, but what happens when they become as easy to launch as dialing *40? Weaponize Your Feature Codes will first take the audience through a brief history of feature codes and common usage, and then demonstrate the more nefarious applications. The presentation will share the Asterisk code used to implement these "rogue" features, and mention possible ways of mitigation. While this talk builds upon previous work from the author, referenced in past DEF CON presentations, the new code written makes carrying out such attacks ridiculously easy Nicholas RosarioMasterChen, is currently a VoIP Administrator. He has been published in 2600: The Hacker Quarterly twice for his research on the Asterisk PBX system and has given presentations at BSides Las Vegas and the DEF CON 303 Skytalks. His most recent research blends technology with psychological principles. MasterChen is an active member of the SYNShop hacker space in Las Vegas, NV and a co-founder and host of the weekly GREYNOISE infosec podcast. Twitter: @chenb0x Instagram: @chenb0x website https://greynoi.se/
https://wn.com/Def_Con_24_Weaponize_Your_Feature_Codes
Raspberry PI with CISCO 7940 running Asterisk on 32GB SD card 5 trunks

Raspberry PI with CISCO 7940 running Asterisk on 32GB SD card 5 trunks

  • Order:
  • Duration: 11:27
  • Updated: 29 Aug 2014
  • views: 29792
videos
Raspberry PI running Asterisk on 32GB SD card. Here is what you need. Shopping List: 1x Raspberry PI 1x 32 GB SD Card With RasPBX and Webmin just because 1x 2.1A 5V USB Power adapter 1x USB to micro-USB cable 1x CISCO 7940 or 7960 or both and more 1X CISCO SF302-08P 8 PoE Port Switch Loads of hours of studding networking, VoIP and a lot of patience to configure it to work good. I did not cover dynamic dns in this video. Too much to cram in just over 10 min but my experience with dynamic extensions is that not all pre-paid phones allow VoIP or even VPN. The only one that worked so far was O2 with Text & Web bolt on. In a nut shell. RasPBX acts as DHCP and tftp server. On many levels of setting up DHCP I'm not even going to go into. If you can live with a Linksys phone adapter and an old fashion phone, none of that nonsense is required and you can mess up your entire network. But if you want the call quality of CISCO then it is the optimal choice. On tftp you will have your SIP and phone config files so once blank phone boots up it will find those files and load them on so the phone starts as SIP and begins logging the extensions and graphics file. If on shared dsl or just home broadband binding DNS is a good way to go using Bind DNS. Webmin helps a lot here. No codes. But again, overwhelming setup menus may as well be in Greek if you new to all that. Incorrect DNS setup is the most common problem of connectivity. Your system may work for a bit but then you notice your trunks are down and nobody can get through. Calls cut off after 3-5 or 20 seconds. Yeah. :-) But not only Incorrect DNS. Blocked ports on router or firewall... So if you have ample time enjoy the project. It is very rewarding and once you replace your old phone setup to Asterisk, you will wonder what took you so long to do it. Not only you will never get nuisance sales calls, you can block anonymous calls, those that are usually pre-recorded but you will no longer have to set call redirect from your landline to your mobile, your mobile or tablet or PC or laptop can become your landline to suite your lifestyle and save tons of moneys. On the other hand, if you just cant find few hundred spare hours to research and do it yourself, drop me a line and I might just come up with something for the need.
https://wn.com/Raspberry_Pi_With_Cisco_7940_Running_Asterisk_On_32Gb_Sd_Card_5_Trunks
Audiocodes MP112 Gateway Setup | VoIP Supply

Audiocodes MP112 Gateway Setup | VoIP Supply

  • Order:
  • Duration: 5:59
  • Updated: 15 Dec 2015
  • views: 4717
videos
Join Senior VoIP Engineer Marc Spehalski as he uses the AudioCodes MP112 FXS Gateway to make an analog phone ring with the RenegadePBX. http://www.voipsupply.com/audiocodes-mp-112-fxs Hi, I’m Marc Spehalski, Senior VoIP Engineer at VoIP Supply. Today, we are going to take a look at the AudioCodes MP112 FXS Gateway and we are going to be doing a basic configuration using an analog phone, and communicating with the RenegadePBX. First, lets take a look at what’s in the back. We have our power port, Ethernet port, two FXS ports for analog devices, and a reset button, for setting to factory defaults. Next, we will plug in the MP112 FSX Gateway to the power on our network so we can get started. Also, plug in the analog device you’re going to use. I’m using this analog telephone. Our objective, the AudioCodes MP112 FXS gateway, is to simply plug in an analog phone, into one of the FXS ports and have it communicate with our RenegadePBX. To start, we will log in with default credentials, which are admin for username, and admin for password, both using a capital A. I’ve already changed the IP address to one that will work on my network. By default, it will be 10 dot 1 dot 10 dot 10. This can be changed, by going to “VoIP”, “Network”, “IP Interfaces Table”, and entering it under “IP Address.” For the next step, go to “Coders and Profiles”, “Coders”. For this example, we use G 711 A, Packetization Time 20, Rate 64, click “Submit.” Next, go to “SIP Definitions”, click “Proxy and Registration.” You want to click “Yes” for “Use Default Proxy”, click the arrow for “Proxy Set Table.” Enter in your PBX information. We’ll be using UDP for this example. Click “Submit.” Make sure “Always Use Proxy” is enabled, as well as, “Enable Registration.” We’ll enter some information for our PBX. Make sure “Subscription Mode” is “Per Endpoint” and “Registration Mode” is also by “Per Endpoint.” Click “Submit.” Now, click on “Gateway” in “IP to IP”, click on “Hunt Group”, and “Endpoint Phone number.” We’ll use channel 1 and extension 2003, which is already programed in our RenegadePBX. Click “Submit.” Next, go to “Analog Gateway”, and “Authentication.” We will use FXS port one, extension 2003, and I’ll enter the predefined password. Last, we’ll click “Routing”, “Telephone to IP Routing”, enter the IP address if your PBX. Our RenegadePBX, is ten dot ten dot ten dot seven three. Use port fifty sixty for SIP, set “Transport Type” to “UDP.” Click “Submit.” If your AudioCodes Gateway has not yet registered the PBX, you can click on “Endpoint Phone Number”, and click the button that says, “Register.” You can then check the status by clicking on “Status and Diagnostics”, “Registration Status.” We can see that FXS Port one is now registered to the PBX. To make sure everything is working, we’ll place a test call from our IP phone, to our analog, and then from our analog to our IP phone. So first, IP to analog. I’ll dial 2003. Now for the analog I’ll dial 2002. It looks like they both work. Finally our last step to make sure everything is saved. Click on the “Burn” button in the Gateway. So we’ve set up our AudioCodes MP112 FXS Gateway to use an analog phone, with the RenegadePBX. For any other questions regarding the MP112, you can call us at 800-398-VoIP, or go to VoIP Supply dot com. Once again, I’m Marc Spehalski, Senior VoIP Engineer at VoIP Supply. Thanks for watching.
https://wn.com/Audiocodes_Mp112_Gateway_Setup_|_Voip_Supply
Simple Asterisk SIP Peering Trunking HowTo

Simple Asterisk SIP Peering Trunking HowTo

  • Order:
  • Duration: 9:59
  • Updated: 06 Sep 2013
  • views: 6377
videos
A video illustrating how-to set up an Asterisk PBX using SIP Peering Trunking. --------- From the extension.conf file ------------------------------------------ Extension names may be numbers, letters, or combinations ; thereof. If an extension name is prefixed by a '_' ; character, it is interpreted as a pattern rather than a ; literal. In patterns, some characters have special meanings: ; ; X - any digit from 0-9 ; Z - any digit from 1-9 ; N - any digit from 2-9 ; [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9) ; . - wildcard, matches anything remaining (e.g. _9011. matches ; anything starting with 9011 excluding 9011 itself) ; ! - wildcard, causes the matching process to complete as soon as ; it can unambiguously determine that no other matches are possible ; ; For example, the extension _NXXXXXX would match normal 7 digit dialings, ; while _1NXXNXXXXXX would represent an area code plus phone number ; preceded by a one. ;
https://wn.com/Simple_Asterisk_Sip_Peering_Trunking_Howto
Installation and Configuration of Avaya 1600/9600 Series IP Telephones (NON-DHCP) - Avaya PBX - HD

Installation and Configuration of Avaya 1600/9600 Series IP Telephones (NON-DHCP) - Avaya PBX - HD

  • Order:
  • Duration: 24:33
  • Updated: 15 Nov 2011
  • views: 187485
videos
Video Topic: Avaya 1600/9600 Series IP Telephones (NON - DHCP) System: Avaya Communication Manager I show you how to install, configure and update 1600/9600 series IP Phones using various tools and methods. This can be translated to other Avaya PBX's as well. Just refer to the Avaya Aura Communication Manager Feature Description and Implementation manual for more information. http://support.avaya.com, Click on Documentation and select the system and version for your PBX. Links to tools I use or talk about in this video: Avaya Firmware for the 1600/9600 IP Phones: http://support.avaya.com EditPad Lite: http://www.editpadlite.com/ Aprelium Web server: http://www.aprelium.com This video and my other videos cover the basic functions of the Avaya PBX. I am a former Tier 3 Engineer that worked in Denver and I find helping others is very rewarding. This is a way I can share my training with others both new and experienced with the Avaya PBX telephone systems. Any views and expressions in this or any other of my videos are of my own. I am in no way employed or associated with Avaya. I just want to help people with their PBX's. Please subscribe, like and comment! It is always appreciated!
https://wn.com/Installation_And_Configuration_Of_Avaya_1600_9600_Series_Ip_Telephones_(Non_Dhcp)_Avaya_Pbx_Hd
Visual Dialplan 2 overview (for Asterisk PBX)

Visual Dialplan 2 overview (for Asterisk PBX)

  • Order:
  • Duration: 7:15
  • Updated: 06 Oct 2008
  • views: 23125
videos
Visual dialplan is modern rapid application development platform for Asterisk dialplan development. Simply drag, drop and connect dialplan blocks to make company IVR, use ASR and TTS, create call center queues, inbound and outbound call flows, and much more. Model, don't code! And then deploy Asterisk Dialplan with a single press of a button using integrated SSH client. Visual Dialplan is the only Asterisk GUI that makes Asterisk dialplan development easier, without limiting use of any native Asterisk dialplan functionality.
https://wn.com/Visual_Dialplan_2_Overview_(For_Asterisk_Pbx)
Raspberry PI to PBX Basic Install (Asterisk) Phone Server

Raspberry PI to PBX Basic Install (Asterisk) Phone Server

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  • Duration: 7:35
  • Updated: 19 Jan 2014
  • views: 78120
videos
I do have some more vids on my channel www.youtube.com/whackstar . This video is a remake where I turn a Raspberry PI into a FreePBX phone server! It worked and was relatively easy for tech head with little Linux experience. I used the following sources to help build the server. I want to say thank you to the guys at Raspberry PI it is truly a great little machine and your site was very helpful. I also want to say thanks to the guys at Raspberry Pi to Asterisk. Your image worked amazingly well! I will be sending you a thank you in the form of a paypal donation sometime soon. Parts Amazon Links: Raspberry PI Cases: http://www.amazon.com/s/ref=bl_sr_pc?... Raspberry PI: http://www.amazon.com/gp/product/B009... SD Card 64Gb (I mean it!): http://www.amazon.com/gp/product/B008... Charger: I used and old Blackberry Charger! Web Resources: Information about the Raspberry PI and recommended starter images: www.raspberrypi.org Information about Raspberry PI to FreePBX: http://www.raspberry-asterisk.org/ Information about FreePBX: http://www.freepbx.org/ Guide how to install webmin: http://www.webmin.com/deb.html My own personal site: www.waynehackman.com I would be very happy to answer any questions you can tweet me at whackman or email me at wayne@waynehackman.com Thanks so much for watching please rate and subscribe and if you are really nice I can do another additional video talking about the gear and trunks the PI-PBX connects to.
https://wn.com/Raspberry_Pi_To_Pbx_Basic_Install_(Asterisk)_Phone_Server
How to Set-up an Enterprise Asterisk-based PBX in 10 Minutes (including coffee break)

How to Set-up an Enterprise Asterisk-based PBX in 10 Minutes (including coffee break)

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  • Duration: 7:23
  • Updated: 28 Jun 2011
  • views: 148828
videos
http://www.xorcom.com - In this video we demonstrate how fast and easy it is to set up an enterprise level Asterisk-based PBX system. In just ten minutes a technician connects a system supporting four PRI lines and 96 FXS analog extensions, complete with analog and IP video phones! The secret to the configuration speed and simplicity is the fact that Xorcom's XPP USB2 drivers are embedded in all standard Asterisk distributions. These drivers enable the PBX to automatically detect the telephony ports. See for yourself!
https://wn.com/How_To_Set_Up_An_Enterprise_Asterisk_Based_Pbx_In_10_Minutes_(Including_Coffee_Break)
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