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Asterisk Tutorial 48 - Introducing NAT Part 1 [english]
 
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By popular demand, here we are on the subject of Network Address Translation (NAT). In the first of a couple of tutorials on NAT, Mathias sets about explaining what NAT is, what it does and why we need it. For more information regarding our Business Communications and VoIP telephony solutions, please check out our website: ► We upgrade business communications • https://www.pascom.net/en/ ► Free pascom cloud business phone system • https://www.pascom.net/en/voip-installation/ ► pascom free download • https://www.pascom.net/en/downloads/ ► Our Blog • https://www.pascom.net/en/blog/
Views: 4466 pascom GmbH & Co. KG
FreePBX Disabling PJSIP and Changing SIP Default port
 
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Disabling PJSIP and Changing default FreePBX SIP port and enabling NAT support
Views: 5416 Ambiorix Rodriguez
Port Forwards on pfSense Firewall for Asterisk SIP traffic
 
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Tutorial video for port forwarding SIP and RTP traffic to an Asterisk server behind a pfSense Firewall.
Views: 55652 sollostech
Flowroute SIP Trunk Setup on FreePBX
 
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This is a how-to video for setting up a Flowroute SIP trunk on FreePBX. FreePBX version 2.11 running Asterisk 11. To contact Chris, please visit http://CrosstalkSolutions.com. Legal mumbo jumbo: FreePBX® is a Registered Trademark of Sangoma Technologies.
Views: 32300 Crosstalk Solutions
What You Need to Know about NAT (Network Address Translation) for Asterisk Internet Telephony
 
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http://www.xorcom.com - This technical lecture discusses NAT (Network Address Translation), including configuration tips, specifically for an Asterisk based PBX. It contains advice about using the smart SIP settings in routers.
Asterisk Tutorial 04 - Asterisk PBX Network Setup [english]
 
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Welcome to back to our Asterisk Video Tutorial series. Due to popular demand, we have made a slight change to our planned episode - we have added an episode covering Asterisk Network Settings. That all means that in this episode, I challenged Mathias to configure our demo network within 8 minutes - which impressively he pretty much managed to do. For more information regarding our Business Communications and VoIP telephony solutions, please check out our website: ► We upgrade business communications • https://www.pascom.net/en/ ► Free pascom cloud business phone system • https://www.pascom.net/en/voip-installation/ ► pascom phone system free download • https://www.pascom.net/en/downloads/ ► Our Blog • https://www.pascom.net/en/blog/
Views: 32909 pascom GmbH & Co. KG
[part 10] Setting up SIP trunk on your FreePBX system so it can talk to the phone company
 
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👉 DISCORD chat server: https://discord.gg/NWuBUxC 👉 Rossmann Repair Group Inc is a participant in the Amazon Services LLC Associates Program, an affiliate advertising program designed to provide a means for sites to earn advertising fees by advertising and linking to amazon.com ✖ Buying on eBay? Support us while you shop! https://www.rossmanngroup.com/ebay Here we go over setting up your PBX box with your "phone company." We decided to use Voicepulse as our "phone company", aka SIP trunk services provider. We will be configuring the PBX to use the Voicepulse trunk we configured in an earlier video. For information on having us set up your phone system, check out our website: this is something we can do for you! http://www.rossmanngroup.com/business-phone-system-solutions/
Views: 41906 Louis Rossmann
Asterisk Tutorial 03 - Asterisk PBX Start Stop Scripts [english]
 
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Welcome to episode 3 of our Introducing Asterisk series. In today's episode we cover Asterisk Start scripts, explaining why we do not want asterisk to run as a root user as well as why it is important to have a good clean and professional setup for your asterisk PBX. For more information regarding our Business Communications and VoIP telephony solutions, please check out our website: ► We upgrade business communications • https://www.pascom.net/en/ ► Free pascom cloud business phone system • https://www.pascom.net/en/voip-installation/ ► pascom phone system free download • https://www.pascom.net/en/downloads/ ► Our Blog • https://www.pascom.net/en/blog/
Views: 35183 pascom GmbH & Co. KG
Port Forwarding to SIP
 
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Port Forwarding to SIP, Ajibola Olayemi (Biztech Infrastructure Systems Limited, Nigeria). I will like to show how port forwarding and destination NAT can be used in Mikrotik with the IP/PBX application for 3CX and the ports to open.. PDF: https:.
Views: 5105 MikroTik
Astercti : Real Time  Live Agent  / Extension Panel Asterisk PBX
 
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Website : https://astercti.techextension.com Contact us for free installation help Skype: tech.extension Email: [email protected] Try for 15 days demo Free. a. Shows the Live Status of Agents Which Includes : Agent Registered/Unregistered. Agent Ringing/Connected Agent Extensions Type (SIP/IAX/PJSIP etc.) Include Features : Transfer calls, Voice mail, Spy Calls, Hangup Call b. Live Channel Monitoring of Asterisk PBX with ease. c. Live Queue Monitoring and get CLI monitor with Single Click
FreePBX VoIP Tutorial Part 5 - Router Settings
 
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Static DNS (optional. You can use your ISP's, Google's, or OpenDNS.) Google: 8.8.8.8 8.8.4.4 OpenDNS: 208.67.222.222 208.67.220.220 Set a Static DHCP for your server's emulated network card. Find the MAC address either in your router's interface or in VirtualBox, settings, network Port forwarding is as follows UDP - 10001-20000 (RTP Calling) TCP - 5222 (Google Voice) UDP - 5060-5082 (SIP) Point these to your VirtualBox machine's internal IP address. In the example, it's 192.168.1.125. So your router will accept incoming connections on the ports we specify, but IP Tables (Linux Firewall) will reject unwanted traffic connecting to those ports. IP Tables is more difficult to configure than router ports, so we'll skip them for now. My router: Linksys/Cisco WRT320N / E2000 with Tomato-E2000-1.28.9054MIPSR2-beta-vpn3.6 firmware Relevant links: Tomato Firmware Wiki: http://en.wikipedia.org/wiki/Tomato_%28firmware%29 fail2ban Wiki: http://en.wikipedia.org/wiki/Fail2ban UDP Timeout thread: UDP Timeout thread: http://www.linksysinfo.org/index.php?threads/default-tomato-timeouts-break-voip.33471/ Part 1 - Introduction - http://www.youtube.com/watch?v=u9DzN1Pu6-Q&hd=1 Part 2 - Gmail and Google Voice Setup - http://www.youtube.com/watch?v=TJ_mZ_3t3r0&hd=1 Part 3 - VirtualBox and PBX in a Flash - http://www.youtube.com/watch?v=tl-knKvixzs&hd=1 Part 4 - DDNS - Free Static IP Setup - http://www.youtube.com/watch?v=gVeh1zdtJsE&hd=1 Part 5 - Router Settings - http://www.youtube.com/watch?v=bGq1_G6rKHY&hd=1 Part 6 - Configuring FreePBX - http://www.youtube.com/watch?v=G1dXb85Bzts&hd=1 Part 7 - Configuring Google Voice - http://www.youtube.com/watch?v=qA8Qw7GmPug&hd=1 Part 8 - Configuring CSipSimple for your first call - http://www.youtube.com/watch?v=3A-yar-JoOQ&hd=1 Part 9 - Installing codecs - G.729 - http://www.youtube.com/watch?v=xI91X6pcvF4&hd=1 Part 10 - Module updates + Backup & Restore - http://www.youtube.com/watch?v=R8RDTZBsUdc&hd=1 Part 11 - Setting up multiple Google Voice phones - http://www.youtube.com/watch?v=t1mFdRFNRsY&hd=1 Part 12 - Now what? Get a free phone number with IPKALL + Hold Music - http://www.youtube.com/watch?v=fR7O_DbUZI0&hd=1
Views: 25300 nirvgorilla
Asterisk Tutorial 45 - SIP Provider Inbound Calls [english]
 
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Welcome back to Introducing Asterisk from the VoIP Guys. In today's tutorial. Mathias walks us through how to configure our Asterisk dialplan to allow inbound calls from our SIP provider as well as demonstrating how to force Asterisk to ignore the user name and only authenticate the host name and port. For more information regarding our Business Communications and VoIP telephony solutions, please check out our website: ► We upgrade business communications • https://www.pascom.net/en/ ► Free pascom cloud business phone system • https://www.pascom.net/en/voip-installation/ ► pascom free download • https://www.pascom.net/en/downloads/ ► Our Blog • https://www.pascom.net/en/blog/
Views: 6351 pascom GmbH & Co. KG
NAT Traversal & RTSP
 
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Views: 18048 william wong
Webinar Networking Routing, NAT e QoS in ambito VoIP SIP
 
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Corso tenuto da Matteo Sala di Eurylink.
Views: 181 VoipVoice srl
Demo NAT on FreePBX (Part 2 - Configure)
 
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NAT on FreePBX
Views: 599 Minh Hoàng Lê
pfSense v2.1.3 -   Disable source port rewriting (Help for VOIP no audio w/ Remote SIP Extension)
 
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I had issue with factory and custom firmware on my Asus RT-AC68U where when remotely connected to my Elastix v2.4.0 (FreePBX v2.8.1) via extension I was not getting any audio when calling to check say voicemail *97 using either CSipSimple or Zoiper on my Android phone. I could see in CLI call coming in but I was not hearing anything. Installed fresh copy of pfSense and only after disabling source port rewriting I was able to hear and DTMF tones worked also and was not a hit and miss as before. Instructions were taken from the official pfSense website for VoIP Configuration - https://doc.pfsense.org/index.php/VoIP_Configuration Hope it helps someone as I was not able to find any video tutorials how to overcome this common issue. Also, here's the screenshot of my port forward page http://i.imgur.com/lzKjwvb.jpg Enjoy!
Views: 12332 marinko113
Demo NAT on FreePBX (Part 1 - Prepare)
 
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NAT on FreePBX
Views: 796 Minh Hoàng Lê
What is SIP ALG?
 
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SIP ALG is a feature on most routers that is intended to assist users on private IP addresses, but in many cases it is implemented poorly and can actually cause more problems than it solves! Find out how you can stop strange behaviour on your VoIP calls today. When SIP ALG is active it can modify the SIP packets of calls in unexpected ways, corrupting them and making them unreadable. This can lead to various results from calls being dropped or rejected to loss of audio and failed registrations. SIP ALG can be a tricky character to pin down as it can cause problems intermittently and may seem to go away, but is usually just waiting to strike again. That is why we recommend turning SIP ALG off on your router and firewall. Once it is disabled you will find that all your issues disappear and you can return to enjoying your crystal clear VoIP calls.
Views: 12384 YayDotCom
Asterisk Tutorial 42 - SIP Provider Registration [english]
 
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Today we do a thing with a SIP provider and Mathias does some product placement - so welcome back to the VoIP Guys and the complex world of SIP provider registration. All things being equal, the process would be the same for every VoIP provider. Sadly, however this is just not the case which is where it gets complicated with two most common authentication approaches being IP based and Registration based. In order to continue with registration based authentication in your Asterisk solution you will need your SIP Registar / proxy, username and password. For more information regarding our Business Communications and VoIP telephony solutions, please check out our website: ► We upgrade business communications • https://www.pascom.net/en/ ► Free pascom cloud business phone system • https://www.pascom.net/en/voip-installation/ ► pascom free download • https://www.pascom.net/en/downloads/ ► Our Blog • https://www.pascom.net/en/blog/
Views: 7707 pascom GmbH & Co. KG
FreePBX 13 Made Easy - Part 2 - Initial Setup and Firewall
 
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In Part 2, we are going to discuss FreePBX initial setup and the FreePBX firewall. This covers best practices for FreePBX security and initial checklist of items to configure. FreePBX 13 Made Easy! playlist: https://www.youtube.com/playlist?list=PL1fn6oC5ndU8QTUpny7Gif9QeuN1fP2F9 Chris Sherwood with Crosstalk Solutions is available for best practice network, WiFi, VoIP, and PBX consulting services. Visit http://CrosstalkSolutions.com for details. Crosstalk Solutions is an authorized FreePBX and Sangoma partner and reseller. Connect with Chris: Twitter: @CrosstalkSol LinkedIn: https://goo.gl/j2Ucgg YouTube: https://goo.gl/g4G58M Amazon Wish List: https://amzn.com/w/M8KHAYD73CB4
Views: 54721 Crosstalk Solutions
FreePBX VoIP Tutorial Part 6 - Configuring FreePBX
 
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Update: Make sure to set NAT from "No - RFC3518" to "YES" in all extensions you add or else you could have trouble making calls. G.711, ulaw, and PCMU are the same. It's lossless 8kHz audio which is the standard for typical landlines. Regarding setting qualify=no in your Extension, I'm still not 100% sure why it works better than having it set to yes, but I have not missed a single call since I've set it up this way. If you have different experience or knowledge, by all means leave a comment. Part 1 - Introduction - http://www.youtube.com/watch?v=u9DzN1Pu6-Q&hd=1 Part 2 - Gmail and Google Voice Setup - http://www.youtube.com/watch?v=TJ_mZ_3t3r0&hd=1 Part 3 - VirtualBox and PBX in a Flash - http://www.youtube.com/watch?v=tl-knKvixzs&hd=1 Part 4 - DDNS - Free Static IP Setup - http://www.youtube.com/watch?v=gVeh1zdtJsE&hd=1 Part 5 - Router Settings - http://www.youtube.com/watch?v=bGq1_G6rKHY&hd=1 Part 6 - Configuring FreePBX - http://www.youtube.com/watch?v=G1dXb85Bzts&hd=1 Part 7 - Configuring Google Voice - http://www.youtube.com/watch?v=qA8Qw7GmPug&hd=1 Part 8 - Configuring CSipSimple for your first call - http://www.youtube.com/watch?v=3A-yar-JoOQ&hd=1 Part 9 - Installing codecs - G.729 - http://www.youtube.com/watch?v=xI91X6pcvF4&hd=1 Part 10 - Module updates + Backup & Restore - http://www.youtube.com/watch?v=R8RDTZBsUdc&hd=1 Part 11 - Setting up multiple Google Voice phones - http://www.youtube.com/watch?v=t1mFdRFNRsY&hd=1 Part 12 - Now what? Get a free phone number with IPKALL + Hold Music - http://www.youtube.com/watch?v=fR7O_DbUZI0&hd=1
Views: 60827 nirvgorilla
Asterisk Tutorial 56 - Asterisk AMI Configuration [english]
 
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Welcome to Introducing Asterisk and configuring the Asterisk AMI. This Asterisk tutorial focuses on enabling and configuring the Asterisk AMI, AMI access and security best practices as well as demonstrating an example configuration. For more information regarding our Business Communications and VoIP telephony solutions, please check out our website: ► We upgrade business communications • https://www.pascom.net/en/ ► Free pascom cloud business phone system • https://www.pascom.net/en/voip-installation/ ► pascom free download • https://www.pascom.net/en/downloads/ ► Our Blog • https://www.pascom.net/en/blog/
Views: 5363 pascom GmbH & Co. KG
FreeSwitch from SIP to WebRTC
 
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Implementing the technological changes from images to audio and video and beyond from a FreeSWITCH perspective. When FreeSWITCH started, 12 years ago, everyone was excited to get 8 kilohertz ulaw and G.729 from point A to point B. Over the last decade, expectations have grown to include 1080p Video, High-Definition audio, texting and more. Two founding members of the FreeSWITCH team will explain the communication platform and how it can be used in a variety of ways in combination with Asterisk and other open source multimedia applications to form a complete solution.
Mikrotik VoIP SIP Server Port Redirect rules setup
 
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Mikrotik VoIP SIP Server Port Redirect rules setup
Views: 28761 Tania Sultana
How to Capture and Debug SIP Packets from asterisk using tcpdump and Wireshark
 
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Troubleshooting VoIP can be a daunting task. Luckily we can easily capture SIP packets in asterisk using tcpdump and analyze the call data results within Wireshark. Here are the tools we will be using in this tutorial: Putty / SSH Client -http://www.chiark.greenend.org.uk/~sgtatham/putty/download.html tcpdump - yum install tcpdump (depending on your OS) WinSCP - http://winscp.net/eng/download.php Wireshark - http://www.wireshark.org/download.html More on using tcpdump: http://www.jonathanmanning.com/2009/10/26/how-to-voip-sip-capture-with-tcpdump-on-linux/ -------- For Encrypted SIP Trunking, Global DID's and Hosted Phone System, check us out! nurango https://www.nurango.ca https://twitter.com/nurangotel
Views: 36208 nurango
Tech Talk: Remote Extensions Configuration
 
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Configuring your SIP phones to a remote Yeastar IP PBX and having remote extensions are easy things, as long as you know the exact steps. Tips: remember to enable the NAT and Register Remotely settings.
Views: 435 Yeastar
Amazon AWS - Asterisk installation on the cloud  - VoIP Server (Part 01)
 
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This video will demonstrate the installation of a VoIP server Asterisk using the Amazon Cloud infrastructure (AWS EC2 ).
Views: 13642 FKIT
Public Opensips + 2 Xlite (in the same private subnet)
 
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Opensips server has a public IP address. Both clients are in the same private network. RTP flow is contained in the private network. SIP dialog reaches Opensips and is NAT-aware. RTP and SDP are NAT-Unaware in this scenario.
Making Your Computer Accessible to the Public Internet: STUN (4 of 4)
 
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Particularly with VoIP software, you may see STUN as an option for keeping your line of communication open. This video explains what STUN is, how it works, and if you can configure it. The video is also part of a series of tutorial about how to make your computer accessible from the Internet, found at http://www.nch.com.au/kb/10046.html
Views: 35831 NCH Software
SIP Operation in the Public Internet: What Makes Running it a Challenge
 
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Speakers: Jiri Kuthan, iptel.org Based on experience at iptel.org, we review current issues related to operation of SIP services on the public Internet. The issues include operational challenges such as NAT traversal, service reliability, scalability, and deployability. We show current practices and identify gaps that still need to be addressed by manufacturers. Topics to be covered include: Status update: existing deployments, available services, interoperability Problem summary: NAT traversal, reliability, scalability, miscellaneous Operational practices NAT: ALGs, manual configuration, UPnP, STUN, relay reliability: use of DNS/SRV versus other solutions, replication mechanisms deployability: scaling concerns, request routing in distributed networks, routing troubleshooting Conclusions: summary of current issues that still need to be addressed
Views: 145 TeamNANOG
FreePBX 101 v14 Part 5 - Endpoint Manager
 
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FreePBX 101 for FreePBX version 14 - Part 5 - Endpoint Manager. Purchase the Endpoint Manager from https://crosstalksolutions.com/product/endpoint-manager/ Crosstalk Store on Amazon - RECOMMENDED PRODUCTS: https://www.amazon.com/shop/crosstalksolutions Amazon Wish List: http://a.co/7dRXc67 Crosstalk Solutions offers best practice phone systems, network design and deployment, and UniFi Video camera systems. Visit https://CrosstalkSolutions.com for details. Crosstalk Solutions is an authorized Sangoma partner and reseller. Connect with Chris: Twitter: @CrosstalkSol LinkedIn: https://goo.gl/j2Ucgg YouTube: https://goo.gl/g4G58M
Views: 5809 Crosstalk Solutions
Asterisk Tutorial 47 - SIP Provider Caller ID [english]
 
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We're back again & we're nearly finished on the fundamentals of SIP Providers. As we promised at the end of our last tutorial, today we take a look at how to set / change your outbound caller ID for your specific provider in the extensions.conf. For more information regarding our Business Communications and VoIP telephony solutions, please check out our website: ► We upgrade business communications • https://www.pascom.net/en/ ► Free pascom cloud business phone system • https://www.pascom.net/en/voip-installation/ ► pascom free download • https://www.pascom.net/en/downloads/ ► Our Blog • https://www.pascom.net/en/blog/
Views: 6279 pascom GmbH & Co. KG
How to change your CallerID on any soft phone like Xlite or Zoiper
 
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Follow this instructions to setup your CallerID on any softphone using a Switch2VoIP account. Change Caller ID on Xlite/Zoiper: 1- Login with your Username and SIP Password 2- Go to the menu Item 'Phone' 3- Click the pencil icon to edit 'CallerID' 4- Edit the 'CallerID' field and click Submit to save. Note:You need to perform these steps only if you use a softphone https://switch2voip.us Prices under 1 cent per minute VoIP UK 0.008 per Minute VoIP USA 0.008 per Minute VoIP Canada 0.005 per Minute VoIP France 0.0078 per Minute Switch2VoIP is a leading provider of VoIP (Voice Over Internet Protocol) internet broadband telephone services often called PC to Phone service. While Switch2VoIP offers international VoIP service to residential and business customers it focuses their marketing efforts servicing Call Center and other companies worldwide that uses auto dialer or predictive dialer with a business model of pay as you go or prepaid. Asterisk VoIP http://www.switch2voip.us/voip-services/voip-service-for-asterisk Business VoIP Provider http://www.switch2voip.us/voip-services/voip-for-business Call Center VoIP http://www.switch2voip.us/voip-services/voip-for-call-centers-and-dialers-vicidial-goautodial-vicibox-vicidialnow-asterisk SIP Trunking http://www.switch2voip.us/voip-services/call-center-sip-trunking 1-800 Toll Free Numbers http://www.switch2voip.us/voip-services/toll-free-numbers Free Online Chat Support on Vicidial, Goautodial, etc. http://switch2voip.us voip provider voip provider vergleich voip provider kostenlos voip provider free voip provider österreich voip provider uk voip provider usa voip provider deutschland vergleich voip provider schweiz voip provider deutschland kostenlos voip provider vergleich voip provider kostenlos voip provider free voip provider deutschland voip provider österreich voip provider uk voip provider usa voip provider deutschland vergleich voip provider schweiz voip provider deutschland kostenlos voip provider australia voip provider asterisk voip provider api voip provider austria voip provider anmeldung nicht erfolgreich voip provider africa voip provider asterix voip provider albania voip provider android voip provider australia comparison voip provider best voip provider business voip provider bangladesh voip provider belgie provider voip belgium voip bad company 2 voip providers belgium voip providers betamax voip service providers business voip service providers byod voip provider canada voip provider cheap voip provider.com voip provider comparison voip providers compare best voip provider canada voip service providers compare voip service providers cheap voip providers list.com voip providers chicago voip provider deutschland voip provider deutschland vergleich voip provider deutschland kostenlos voip provider did rynga voip discount provider discount voip provider voip providers delhi voip service providers delhi voip provider europe voip provider elastix voip provider en france voip providers e911 voip providers egypt voip providers enterprise voip equipment providers voip provider free voip provider for asterisk voip provider for india voip provider for australia voip providers for international voip providers for mobile voip providers for 3cx voip providers for canada voip providers for uk voip free providers usa voip provider germany voip provider gateway voip provider gratis voip providers global voip providers gigaset voip providers greece good voip provider voip carrier grade goedkoopste voip provider goedkope voip provider voip provider hong kong voip provider how to voip provider how to become voip hosted providers voip service providers hyderabad voip providers houston voip providers home voip providers hawaii voip providers hyderabad voip hosted provider voip provider in pakistan voip provider india voip provider in bangladesh voip provider ireland voip provider in kolkata voip provider in canada voip provider in singapore voip provider in malaysia voip provider in mumbai voip provider in pune voip provider jakarta voip provider japan voip providers jordan voip providers jamaica voip providers johannesburg voip providers magic jack voip providers san jose voip provider kostenlos voip provider kolkata voip provider hong kong voip providers kenya voip providers keep phone number voip providers kerala voip providers kuwait voip providers ksa voip service providers kolkata voip minutes provider in kolkata voip provider list voip provider lingo voip provider list in bangladesh voip providers list usa voip providers lync voip company list voip carrier list voip providers sri lanka largest voip provider voip providers los angeles voip provider malaysia voip provider montreal voip provider melbourne voip providers mobile voip minutes provider
Change SIP port on Grandstream UCM PBX
 
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We go through how to change the default SIP port from 5060 to something different. You might do this to help prevent a SIP attack or to just add an extra layer of security. You will also need to change your port forwards in your firewall to accommodate the new SIP Port. Equipment used in this video: UCM6202: http://www.grandstream.com/products/ip-pbxs/ucm-series-ip-pbxs/product/ucm6200-series Want Grandstream support or products? With monthly plans starting at $19.99 and additional savings on products visit www.shopn2v.com for more information and to sign up today! For questions you can call us at 507-205-4025 or email [email protected]
Views: 2926 n2v Solutions LLC
2 installing centos - asterisk tutorial
 
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Next Video: http://www.youtube.com/watch?v=iDdjubxNRnQ In order to do asterisk labs your going to need the right tools. The biggies outside of asterisk include -virtualbox (virtual machine software) -CentOS linux (operating system)
Views: 17649 miamimanni
FreePBX VoIP Tutorial Part 11 - Setting up multiple Google Voice phones
 
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Note: Slight correction. At 1:07 in the video, you want to set NAT to YES. Command used in PuTTY: asterisk -rvvv To leave the Asterisk CLI after typing the above command, simply type exit and hit enter. It will go back to the [email protected] command prompt. Part 1 - Introduction - http://www.youtube.com/watch?v=u9DzN1Pu6-Q&hd=1 Part 2 - Gmail and Google Voice Setup - http://www.youtube.com/watch?v=TJ_mZ_3t3r0&hd=1 Part 3 - VirtualBox and PBX in a Flash - http://www.youtube.com/watch?v=tl-knKvixzs&hd=1 Part 4 - DDNS - Free Static IP Setup - http://www.youtube.com/watch?v=gVeh1zdtJsE&hd=1 Part 5 - Router Settings - http://www.youtube.com/watch?v=bGq1_G6rKHY&hd=1 Part 6 - Configuring FreePBX - http://www.youtube.com/watch?v=G1dXb85Bzts&hd=1 Part 7 - Configuring Google Voice - http://www.youtube.com/watch?v=qA8Qw7GmPug&hd=1 Part 8 - Configuring CSipSimple for your first call - http://www.youtube.com/watch?v=3A-yar-JoOQ&hd=1 Part 9 - Installing codecs - G.729 - http://www.youtube.com/watch?v=xI91X6pcvF4&hd=1 Part 10 - Module updates + Backup & Restore - http://www.youtube.com/watch?v=R8RDTZBsUdc&hd=1 Part 11 - Setting up multiple Google Voice phones - http://www.youtube.com/watch?v=t1mFdRFNRsY&hd=1 Part 12 - Now what? Get a free phone number with IPKALL + Hold Music - http://www.youtube.com/watch?v=fR7O_DbUZI0&hd=1
Views: 8874 nirvgorilla
Step-By-Step Opening NAT Fixing "Strict Firewall" Settings!!
 
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This opens your port, making outside devices more compatible (such as the Xbox 360, Wii, and Playstation 3). Want a lag-free expierence? View this vid!!
Views: 49779 ryan4588
Yahoo VoIP STUN
 
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(Yahoo talk about VoIP) - kilka slow nt. VoIP od Yahoo. Material dostarczyl serwis Aviaa.com [ http://www.aviaa.com ]
Views: 65041 aviaa1
2. Extension Settings (v15.5 SP3)
 
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This module demonstrates the available settings for the extensions and how to configure them. Join our live webinars for a more detailed demonstration: https://www.3cx.com/blog/event-trainings/
Views: 2386 3CX
Quick Configs - NAT (NVI, ALG, static, dynamic, route-map, pool)
 
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This CCIE oriented episode of quick configs goes into Network Address Translation (NAT). See http://bit.ly/1VZYkFi for all CCIE notes.
Views: 2189 Ben Pin
4 - The PFSense Firewall - Alias , NAT , Rules , Traffic Shaper , and virtual ip (12-2014)
 
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Support: https://forum.pfsense.org/ Documentation: https://doc.pfsense.org/index.php/Main_Page The pfSense project is a free, open source customized distribution of FreeBSD specifically tailored for use as a firewall and router that is entirely managed via web interface. In addition to being a powerful, flexible firewalling and routing platform, it includes a long list of related features and a package system allowing further expandability without adding bloat and potential security vulnerabilities to the base distribution. The pfSense project has become a fairly popular project with more than 1 million downloads since its inception, and proven in countless installations ranging from small home networks protecting a single computer to large corporations, universities and other organizations protecting thousands of network devices.
Views: 16755 TEK411.com
Tech Talk: How to Set up Firewall in Yeastar S-Series VoIP PBX
 
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Firewall protects S-Series VoIP PBX from malicious attacks and prevents call fraud. Learn how to protect your S-Series VoIP PBX in this video.
Views: 940 Yeastar
M0n0wall NAT Forwarding and firewall rules
 
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How to creat a NAT rule and firewall rule
Views: 18555 squirrel8472
NAT Port Forwarding in pfSense 2.4
 
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IMPORTANT NOTE: I am using a private IP for my WAN IP only because my test environment is behind another NAT device. Your WAN IP from your ISP MUST be a public/routable IP, and a static IP from your ISP is even better. How to forward traffic from your external WAN IP address to an internal server. This allows access to your internal server from offsite/remote locations.
Views: 8767 Rocket City Tech
FreePBX 13 asterisk 11 with Twilio Sip Trunking
 
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Setup Twilio Elastic Sip trunk with FreePBX http://hwdevelopment.com/blog/27-freepbx-13-asterisk-11-twilio-elastic-sip-trunk-setup
Grandstream UCM 6xxx Overview and setup
 
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If you want to learn how to do this, watch the video and make your own diagrams to assist yourself. If you don't want to learn this, call or email us and we are happy to do it for you! This is a long video that covers the following content: Intro [remember to like and subscribe ;-) ] Hardware overview-1:27 The plan-5:45 How to Upgrade(and more plan)-7:15 How to Factory reset-15:22 Overview of each section of software-25:40 UCM configuration-1:28:20 (this includes trunks, routes, Ring Groups, Queues, etc) Call or email for support 507-205-4025 [email protected]
Views: 535 n2v Solutions LLC
StarTrinity VoIP Status - 01 VoIP Readiness Tests
 
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Tutorial: http://startrinity.com/VoIP/VoipStatus/VoipStatusTutorial.aspx#voip_r_t_running SIP ALG test The test is performed using range of server-side SIP ports (default 5060..5069) Client sends SIP calls to server If SIP contact header is delivered to server without changes, test is marked as "SIP ALG is not detected" If there is SIP ALG between client and server, it changes SIP Contact header and test is marked as "SIP ALG detected" NAT test The test is performed using range of server-side SIP ports (default 5060..5069) Client sends REGISTER to server Client sends to server, as SIP phone to IP PBX Server proxies the SIP call to registered extension. SIP and RTP packets go from client to server through client-side IP router(s), NAT to server and back to client Client answers to the received incoming call and echoes RTP stream back to server Client sends DTMF digit via RTP stream from outgoing call leg to incoming call leg, the DTMF is echoed back to caller side via server. This test checks 2-way RTP audio stream Client verifies the RTP stream path, saves report to the VoIP Status web server Bandwidth+DSCP test Client sends multiple concurrent SIP calls according to pre-configured settings Server accepts the calls The calls are kept alive during pre-configured duration Both client and server send RTP packets to each other with specified DSCP flag, capture IP traffic using winpcap library, analyse RTP packet loss, jitter, G.107 MOS based on winpcap RTP packets timestamps, received (RX) DSCP fields Results are stored in CDR files on SIP server side
Views: 113 Sergey A
ShoreTel 8 Admin: SIP Trunks by DrVoIP.COM
 
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SIPP Trunks. One of a series of videos taken from the DrVoIP.com ShoreTel 8 training. View more of our segments here on YouTube. You can get immediate online viewing access to the complete 15 hours of training, or you can purchase it on DVD, at www.Dr.VoIP.com. 19 of 24
Views: 2023 DrVoIP