• 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
  • SipVicious, Asterisk, Vulnerado, SIP

    El video consiste en el metodo para testiar una planta que provea servicios de Voz sobre IP, buscando usuarios, y posibles contraseñas para poder registrase. Recomiendo que visiten mi blog: http://sdrlatino.wordpress.com/ http://descsecurity.wordpress.com/ y el grupo de Investigacion en VoIP: http://busy-tone.org/

    published: 03 Jan 2013
  • LAS 400 Phones Home | Linux Action Show 400

    We celebrate 400 episodes of the Linux Action Show, show you how easy it is to setup your own free phone system, never flash another USB stick again & the big Ubuntu rumors. Plus the openSSH bug you need to patch, the Steam Link SDK, Gnome 3 changes & more! Show Notes & Download: http://bit.ly/las400

    published: 16 Jan 2016
  • SIPing on some Linux | Linux Action Show 371

    Find out about the best software and hardware to make and take amazingly good sounding calls. From the best open source solutions, to turnkey solutions today’s episode has something from the beginner to the expert. Plus Google removes "always listening" code from Chromium, The Linux Foundation invests some serious cash, why Red Hat’s CEO thinks Linux has “won” the datacenter… Show Notes & Download: http://bit.ly/las-371

    published: 29 Jun 2015
  • Raspberry Pi and Node.js

    Links: http://www.raspberrypi.org/downloads http://osxfuse.github.com/ http://nodejs.org/download/ http://expressjs.com/ Commands: sudo dd if=YourRaspianImage of=/dev/diskXYZ ssh pi@IPAddress mkdir .ssh ssh-keygen scp id_rsa.pub pi@IPAddress:.ssh/authorized_keys sshfs pi@IPAddress: SomeDirectory wget http://nodejs.org/dist/v0.8.16/node-v0.8.16.tar.gz tar -zxf node-v0.8.16.tar.gz cd node-v0.8.16 ./configure make sudo make install sudo apt-get install screen screen screen -r sudo sh install-node.sh node --version npm ---version sudo node node-server.js Node.js example server: var http = require('http'); http.createServer(function(req, res) { res.end('Hello, Raspberry Pi'); }).listen(80);

    published: 24 Dec 2012
  • GRANDSTREAM GXP 1405

    GXP1400/1405 is a next generation small-to-medium business IP phone that features 2 lines with 2 SIP account, a 128x40 graphical LCD, 3 XML programmable context-sensitive soft keys, dual network ports with integrated PoE (GXP1405 only), and 3-way conference. The GXP1400/1405 delivers superior HD audio quality, rich and leading edge telephony features, personalized information and customizable application service, automated provisioning for easy deployment, advanced security protection for privacy, and broad interoperability with most 3rd party SIP devices and leading SIP/NGN/IMS platforms. It is a perfect choice for small-to-medium businesses looking for a high quality, feature rich IP phone with affordable cost. 2 line keys with dual-color LED (2 SIP account and up to 2 call appearances),...

    published: 02 Mar 2013
  • HD config webrtc

    Cấu hình Freeswitch hỗ trợ WebRTC client (sipJS)

    published: 01 Apr 2016
  • Video manual Telefonos IP

    Manual para usar los telefonos audiocodes 310HD basicos.

    published: 30 Nov 2011
  • Ken Rice, Sean Hsieh, Casey McPhee Talk New Flowroute Features at Cluecon

    Visit https://developer.flowroute.com/docs/sms-proxy and https://github.com/flowroute. Flowroute makes last memories among Rich and Real-time IP Communications companies at conferences and not just for their very cool, totally wearable t-shirt. The Seattle, Washington-based company also sponsored Cluecon 2016 all conference pizza party. They offer calling and messaging for software as a service solutions, i.e., for unified communications, call centers, and PBX & phone systems. The Makers' Challenge, also known as Coder Games, at Cluecon was partly sponsored by Flowroute. The highly charged activity pushes every individual to represent his or her company and skills to use available APIs in ways that are fun, practical, unique and most of all empowering. One of the winning teams created ...

    published: 09 Sep 2016
  • VOIP Bandwidth Optimization Service by Nicholas Ryan

    'ViBE' is an excellent 'VOIP Bandwidth Optimization Service' provided by Voice Next Pte. Ltd., Singapore. Company has a customer base in ASIA, AFRICA & MIDDLE East. Nicholas Ryan Voice Next Pte. Ltd. Singapore - 068589 Web: www.voicenext.net skype: nicholas@voicenext.net Email: nicholas@voicenext.net

    published: 23 Jun 2013
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
SipVicious, Asterisk, Vulnerado, SIP

SipVicious, Asterisk, Vulnerado, SIP

  • Order:
  • Duration: 4:05
  • Updated: 03 Jan 2013
  • views: 1370
videos
El video consiste en el metodo para testiar una planta que provea servicios de Voz sobre IP, buscando usuarios, y posibles contraseñas para poder registrase. Recomiendo que visiten mi blog: http://sdrlatino.wordpress.com/ http://descsecurity.wordpress.com/ y el grupo de Investigacion en VoIP: http://busy-tone.org/
https://wn.com/Sipvicious,_Asterisk,_Vulnerado,_Sip
LAS 400 Phones Home | Linux Action Show 400

LAS 400 Phones Home | Linux Action Show 400

  • Order:
  • Duration: 89:10
  • Updated: 16 Jan 2016
  • views: 4825
videos
We celebrate 400 episodes of the Linux Action Show, show you how easy it is to setup your own free phone system, never flash another USB stick again & the big Ubuntu rumors. Plus the openSSH bug you need to patch, the Steam Link SDK, Gnome 3 changes & more! Show Notes & Download: http://bit.ly/las400
https://wn.com/Las_400_Phones_Home_|_Linux_Action_Show_400
SIPing on some Linux | Linux Action Show 371

SIPing on some Linux | Linux Action Show 371

  • Order:
  • Duration: 94:59
  • Updated: 29 Jun 2015
  • views: 5901
videos
Find out about the best software and hardware to make and take amazingly good sounding calls. From the best open source solutions, to turnkey solutions today’s episode has something from the beginner to the expert. Plus Google removes "always listening" code from Chromium, The Linux Foundation invests some serious cash, why Red Hat’s CEO thinks Linux has “won” the datacenter… Show Notes & Download: http://bit.ly/las-371
https://wn.com/Siping_On_Some_Linux_|_Linux_Action_Show_371
Raspberry Pi and Node.js

Raspberry Pi and Node.js

  • Order:
  • Duration: 13:26
  • Updated: 24 Dec 2012
  • views: 50701
videos
Links: http://www.raspberrypi.org/downloads http://osxfuse.github.com/ http://nodejs.org/download/ http://expressjs.com/ Commands: sudo dd if=YourRaspianImage of=/dev/diskXYZ ssh pi@IPAddress mkdir .ssh ssh-keygen scp id_rsa.pub pi@IPAddress:.ssh/authorized_keys sshfs pi@IPAddress: SomeDirectory wget http://nodejs.org/dist/v0.8.16/node-v0.8.16.tar.gz tar -zxf node-v0.8.16.tar.gz cd node-v0.8.16 ./configure make sudo make install sudo apt-get install screen screen screen -r sudo sh install-node.sh node --version npm ---version sudo node node-server.js Node.js example server: var http = require('http'); http.createServer(function(req, res) { res.end('Hello, Raspberry Pi'); }).listen(80);
https://wn.com/Raspberry_Pi_And_Node.Js
GRANDSTREAM GXP 1405

GRANDSTREAM GXP 1405

  • Order:
  • Duration: 0:27
  • Updated: 02 Mar 2013
  • views: 37694
videos
GXP1400/1405 is a next generation small-to-medium business IP phone that features 2 lines with 2 SIP account, a 128x40 graphical LCD, 3 XML programmable context-sensitive soft keys, dual network ports with integrated PoE (GXP1405 only), and 3-way conference. The GXP1400/1405 delivers superior HD audio quality, rich and leading edge telephony features, personalized information and customizable application service, automated provisioning for easy deployment, advanced security protection for privacy, and broad interoperability with most 3rd party SIP devices and leading SIP/NGN/IMS platforms. It is a perfect choice for small-to-medium businesses looking for a high quality, feature rich IP phone with affordable cost. 2 line keys with dual-color LED (2 SIP account and up to 2 call appearances), 3 XML programmable context-sensitive soft keys, 3-way conference HD wideband handset, hands-free speakerphone with advanced acoustic echo cancellation Automated personal information service (e.g., local weather), personalized music ring tone/ring back tone
https://wn.com/Grandstream_Gxp_1405
HD config webrtc

HD config webrtc

  • Order:
  • Duration: 11:53
  • Updated: 01 Apr 2016
  • views: 92
videos
Cấu hình Freeswitch hỗ trợ WebRTC client (sipJS)
https://wn.com/Hd_Config_Webrtc
Video manual Telefonos IP

Video manual Telefonos IP

  • Order:
  • Duration: 2:23
  • Updated: 30 Nov 2011
  • views: 2289
videos
Manual para usar los telefonos audiocodes 310HD basicos.
https://wn.com/Video_Manual_Telefonos_Ip
Ken Rice, Sean Hsieh, Casey McPhee Talk New Flowroute Features at Cluecon

Ken Rice, Sean Hsieh, Casey McPhee Talk New Flowroute Features at Cluecon

  • Order:
  • Duration: 4:23
  • Updated: 09 Sep 2016
  • views: 34
videos
Visit https://developer.flowroute.com/docs/sms-proxy and https://github.com/flowroute. Flowroute makes last memories among Rich and Real-time IP Communications companies at conferences and not just for their very cool, totally wearable t-shirt. The Seattle, Washington-based company also sponsored Cluecon 2016 all conference pizza party. They offer calling and messaging for software as a service solutions, i.e., for unified communications, call centers, and PBX & phone systems. The Makers' Challenge, also known as Coder Games, at Cluecon was partly sponsored by Flowroute. The highly charged activity pushes every individual to represent his or her company and skills to use available APIs in ways that are fun, practical, unique and most of all empowering. One of the winning teams created an app that used features of Internet of Things, Raspberry Pi, baristas, and coffee orders and delivery. Sean Hsieh, Flowroute's Chief Product Officer, whose "marketing and design talents have served Fortune 100 companies including Apple, Linksys (Cisco), and Motorola," joined in the short discussion at Cluecon 2016 captured on video. He mentions that several use cases were shared during the 2016 real-time communications conference. Their demo by Flowroute developer Casey MacPhee is available on the Flowroute GitHub page. See https://developer.flowroute.com/docs/sms-proxy and https://github.com/flowroute. On August 22, soon after Cluecon 2016, Flowroute announced their new SMS messaging capabilities offered through a single API. The modules that make it up are written in Python and include phone number masking, two-factor authentication, and appointment reminders. Cluecon will be back at Swissotel in Chicago, Illinois from Aug 7th-11th 2017. Extremely profitable opportunities are available to sign up as sponsors, exhibitors, speakers and other types of participants now at https://cluecon.com/.
https://wn.com/Ken_Rice,_Sean_Hsieh,_Casey_Mcphee_Talk_New_Flowroute_Features_At_Cluecon
VOIP Bandwidth Optimization Service by Nicholas Ryan

VOIP Bandwidth Optimization Service by Nicholas Ryan

  • Order:
  • Duration: 3:06
  • Updated: 23 Jun 2013
  • views: 351
videos
'ViBE' is an excellent 'VOIP Bandwidth Optimization Service' provided by Voice Next Pte. Ltd., Singapore. Company has a customer base in ASIA, AFRICA & MIDDLE East. Nicholas Ryan Voice Next Pte. Ltd. Singapore - 068589 Web: www.voicenext.net skype: nicholas@voicenext.net Email: nicholas@voicenext.net
https://wn.com/Voip_Bandwidth_Optimization_Service_By_Nicholas_Ryan
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