• 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
  • Code Java Asterisk

    I made this video cause my previous video is like not very clear........

    published: 05 Dec 2011
  • 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
  • VOIP Python Client

    Demo for our EE284 project at SJSU. This is a python client used for making a voip call. Source Code : https://github.com/soumilk91/Python-SIP-Client/blob/master/EE284_SIP_Client_version2.py

    published: 09 Dec 2015
  • AstriCon 2016 Keynote - Bill Ledingham, Black Duck Software

    Bill Ledingham Black Duck Software, CTO/EVP of Engineering Bill is the CTO and EVP of Engineering at Black Duck Software, the industry leader in products and solutions which help companies secure and manage their use of open source software. He’s also a partner at Converge Venture Partners, where he actively works with a number of early-stage companies. A veteran in the voice industry, he served as an executive and member of the founding team at SpeechWorks, a leader in speech-based solutions that IPO’d in 2000 and is now part of Nuance. He also has been on the founding team at other companies in the virtualization and information security spaces. Bill holds a B.S. in Electrical Engineering and a M.S. in Industrial Engineering from Stanford, as well as an MBA from Harvard.

    published: 11 Nov 2016
  • What Is A GSM Codec?

    Voip think codecs g711, g729, ilbc, gsm. Cx software based voip ip pbx pabx. By xray tue aug 25, 2009 3 43 pm. Googleusercontent searchthe 'global system for mobile communications' (gsm) is a digital radio which extensively used throughout europe, and also in many other parts of the world audio codecs or vocoders are universally within gsm. They reduce the bit rate of speech that has been converted from its analogue for into a 14 feb 2017 gsm (global system mobile communications) is cellular phone standard popular outside usa. 625 bits audio in other words speech codec represents speech with as few bits as codec used in gsm systems is presented based on the original 13 kbits s full rate rpe vocal's range of gsm speech coders are suitable for mobile telephony and voip applications and optim...

    published: 21 Jun 2017
  • 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
  • django RC car for Raspberrypi

    django RC car for Raspberrypi https://github.com/rasplay/RasplayWeb

    published: 30 Apr 2015
  • #DailDojo 2013 Raspberry PI, router & some web skills

    Brilliant presentation using Raspberry PI , router and some web skills.

    published: 08 Jul 2013
  • 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
  • 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
  • ► SendHub.com Review - VoIP, Text, SMS Startup - AngelKings.com

    http://goo.gl/q4k48E | A full review of Sendhub's VoIP, text and SMS mobile product for businesses looking to replace Ringcentral, Grasshopper or use options like Google Voice. Expert angel investor and venture capitalist, Ross Blankenship (http://rossblankenship.com) reviews Sendhub.com and the features they use including integration of Twilio's messaging system. We also look at CEO Ash Rust and founders Ryan Pfeffer, and Garrett Johnson. Sendhub has raised capital from Bullpen capital, Kapor capital, Eric Ries, Menlo Ventures and Angel Kings/Angel List. The primary competitors in the VoIP, messaging and SMS industry include: Google voice, Grasshopper, Ringcentral, Twilio. To learn about SendHub, and get information on their product, or invest in companies go to http://angelkings.c...

    published: 03 Jul 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
  • Using the MOH Player Unit

    Introduction to our MOH Player, what's included and how to load a hold message onto the unit

    published: 22 Nov 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
SipVicious, Asterisk, Vulnerado, SIP

SipVicious, Asterisk, Vulnerado, SIP

  • Order:
  • Duration: 4:05
  • Updated: 03 Jan 2013
  • views: 1698
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
Code Java Asterisk

Code Java Asterisk

  • Order:
  • Duration: 1:37
  • Updated: 05 Dec 2011
  • views: 2150
videos
I made this video cause my previous video is like not very clear........
https://wn.com/Code_Java_Asterisk
Audiocodes MP112 Gateway Setup | VoIP Supply

Audiocodes MP112 Gateway Setup | VoIP Supply

  • Order:
  • Duration: 5:59
  • Updated: 15 Dec 2015
  • views: 10128
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
VOIP Python Client

VOIP Python Client

  • Order:
  • Duration: 4:53
  • Updated: 09 Dec 2015
  • views: 3325
videos
Demo for our EE284 project at SJSU. This is a python client used for making a voip call. Source Code : https://github.com/soumilk91/Python-SIP-Client/blob/master/EE284_SIP_Client_version2.py
https://wn.com/Voip_Python_Client
AstriCon 2016 Keynote - Bill Ledingham, Black Duck Software

AstriCon 2016 Keynote - Bill Ledingham, Black Duck Software

  • Order:
  • Duration: 48:13
  • Updated: 11 Nov 2016
  • views: 1038
videos
Bill Ledingham Black Duck Software, CTO/EVP of Engineering Bill is the CTO and EVP of Engineering at Black Duck Software, the industry leader in products and solutions which help companies secure and manage their use of open source software. He’s also a partner at Converge Venture Partners, where he actively works with a number of early-stage companies. A veteran in the voice industry, he served as an executive and member of the founding team at SpeechWorks, a leader in speech-based solutions that IPO’d in 2000 and is now part of Nuance. He also has been on the founding team at other companies in the virtualization and information security spaces. Bill holds a B.S. in Electrical Engineering and a M.S. in Industrial Engineering from Stanford, as well as an MBA from Harvard.
https://wn.com/Astricon_2016_Keynote_Bill_Ledingham,_Black_Duck_Software
What Is A GSM Codec?

What Is A GSM Codec?

  • Order:
  • Duration: 1:01
  • Updated: 21 Jun 2017
  • views: 17
videos
Voip think codecs g711, g729, ilbc, gsm. Cx software based voip ip pbx pabx. By xray tue aug 25, 2009 3 43 pm. Googleusercontent searchthe 'global system for mobile communications' (gsm) is a digital radio which extensively used throughout europe, and also in many other parts of the world audio codecs or vocoders are universally within gsm. They reduce the bit rate of speech that has been converted from its analogue for into a 14 feb 2017 gsm (global system mobile communications) is cellular phone standard popular outside usa. 625 bits audio in other words speech codec represents speech with as few bits as codec used in gsm systems is presented based on the original 13 kbits s full rate rpe vocal's range of gsm speech coders are suitable for mobile telephony and voip applications and optimized for leading dsps main codecs used in voip. Greetings, i was wondering if anyone has run into any issues trying to select the gsm codec when. Codec gsm the gsm codec mobile. Codec gsmthe gsm codecaudio voice vocoder codec voip infogsm codecgsm codecs vocal technologies. 103 speech codec list for gsm and umts tech invite. The codec operates on audio frames 20 milliseconds this paper describes the gsm enhanced full rate (efr) speech that has been standardised for mobile communication system. Here's a look at the current codecs this voice codec was first standard used in gsm. 0 (2001 03)universal mobile telecommunications system (umts);. Gsm enhanced full rate speech codec ieee xplore document. Html url? Q webcache. Its average bit rate is 13 kbit s. The gsm efr the 'global system for mobile communications' (gsm) is a digital radio full rate speech codec operates at 13 kbits s and uses regular (coder decoder) device or software program that used typically to convert analogue information (such as speech) into stream etsi ts 126 103 v4. Sorting through gsm codecs a tutorial codec codecpro. The bit rate of the codec is 13 kbit s, or 1. The original 'full rate' gsm speech codec is named rpe ltp (regular pulse excitation long term prediction) full rate was the first digital coding standard used in mobile phone system. The gsm efr this page describes amr(adaptive multi rate) basics in technology. It covers various speech codec gsm amr rates,frame formats,bit rates,amr 16 apr 2011 hi, i was wondering if anyone could help me by letting know which is used on the 3cx free softphone? I have it recently selecting. G711, g722, g723, g726, g728, g729, dvi, gsm, l16, lpc, speex, ilbc showing the bit rate, sampling rate and frame size 11 jul 2003 good codec design is an essential element in providing toll quality voice transmissions over gsm links. Ozeki c# sip stack gsm codec(umts); Speech codec list for and umts etsi. Gsm enhanced full rate speech codec (pdf download available)adaptive multi which gsm codec? . Uk speech_codecs standards gsm. The quality of speech is less by modern standards, but was a good compromise this paper describes the gsm enhanced full rate (efr) c
https://wn.com/What_Is_A_Gsm_Codec
HD config webrtc

HD config webrtc

  • Order:
  • Duration: 11:53
  • Updated: 01 Apr 2016
  • views: 230
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: 2606
videos
Manual para usar los telefonos audiocodes 310HD basicos.
https://wn.com/Video_Manual_Telefonos_Ip
django RC car for Raspberrypi

django RC car for Raspberrypi

  • Order:
  • Duration: 1:11
  • Updated: 30 Apr 2015
  • views: 59
videos
django RC car for Raspberrypi https://github.com/rasplay/RasplayWeb
https://wn.com/Django_Rc_Car_For_Raspberrypi
#DailDojo 2013 Raspberry PI, router & some web skills

#DailDojo 2013 Raspberry PI, router & some web skills

  • Order:
  • Duration: 3:28
  • Updated: 08 Jul 2013
  • views: 67
videos
Brilliant presentation using Raspberry PI , router and some web skills.
https://wn.com/Daildojo_2013_Raspberry_Pi,_Router_Some_Web_Skills
VOIP Bandwidth Optimization Service by Nicholas Ryan

VOIP Bandwidth Optimization Service by Nicholas Ryan

  • Order:
  • Duration: 3:06
  • Updated: 23 Jun 2013
  • views: 375
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
LAS 400 Phones Home | Linux Action Show 400

LAS 400 Phones Home | Linux Action Show 400

  • Order:
  • Duration: 1:29:10
  • Updated: 16 Jan 2016
  • views: 4880
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
► SendHub.com Review - VoIP, Text, SMS Startup - AngelKings.com

► SendHub.com Review - VoIP, Text, SMS Startup - AngelKings.com

  • Order:
  • Duration: 6:01
  • Updated: 03 Jul 2015
  • views: 150
videos
http://goo.gl/q4k48E | A full review of Sendhub's VoIP, text and SMS mobile product for businesses looking to replace Ringcentral, Grasshopper or use options like Google Voice. Expert angel investor and venture capitalist, Ross Blankenship (http://rossblankenship.com) reviews Sendhub.com and the features they use including integration of Twilio's messaging system. We also look at CEO Ash Rust and founders Ryan Pfeffer, and Garrett Johnson. Sendhub has raised capital from Bullpen capital, Kapor capital, Eric Ries, Menlo Ventures and Angel Kings/Angel List. The primary competitors in the VoIP, messaging and SMS industry include: Google voice, Grasshopper, Ringcentral, Twilio. To learn about SendHub, and get information on their product, or invest in companies go to http://angelkings.com.
https://wn.com/►_Sendhub.Com_Review_Voip,_Text,_Sms_Startup_Angelkings.Com
Raspberry Pi and Node.js

Raspberry Pi and Node.js

  • Order:
  • Duration: 13:26
  • Updated: 24 Dec 2012
  • views: 62664
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
Using the MOH Player Unit

Using the MOH Player Unit

  • Order:
  • Duration: 3:25
  • Updated: 22 Nov 2012
  • views: 2275
videos
Introduction to our MOH Player, what's included and how to load a hold message onto the unit
https://wn.com/Using_The_Moh_Player_Unit
GRANDSTREAM GXP 1405

GRANDSTREAM GXP 1405

  • Order:
  • Duration: 0:27
  • Updated: 02 Mar 2013
  • views: 45401
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
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