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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: 4502 Ambiorix Rodriguez
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: 4124 pascom GmbH & Co. KG
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: 55041 sollostech
[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: 38271 Louis Rossmann
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.
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: 4373 MikroTik
Asterisk 1.8 SIP Trunking
 
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On this video we cover the setup for a SIP Trunk between 2 Asterisk Servers. The sip.conf and dialplan configuration. We use Ekiga to test calls between both servers.
Views: 36999 Robert Thomas Zamora
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: 52364 Crosstalk Solutions
The New SIP Stack in Asterisk
 
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Learn more at http://asterisk.org Session Initiation Protocol (SIP) is almost always a critical part of any Asterisk deployment. This role has traditionally been filled with the chan_sip Asterisk module but with Asterisk 12 a new player is in the game, chan_pjsip. This new module is built upon the widely deployed PJSIP SIP stack and brings with it a new avenue for expansion and rapid development. This talk will focus on the over all design concepts, how it works, and the configuration involved. Joshua Colp & Mark Michelson, Digium Software Engineering
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: 31190 Crosstalk Solutions
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: 31211 pascom GmbH & Co. KG
Asterisk Tutorial 33 - Asterisk IVR Menu Looping [english]
 
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It's time to enhance our IVRs to account for "timeouts" by looping our IVR menus. Menu loops allow the IVR menu to be repeated should an option not be selected within a specific time frame, giving the caller the opportunity to make their selection. 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: 3981 pascom GmbH & Co. KG
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: 5890 pascom GmbH & Co. KG
Demo NAT on FreePBX (Part 2 - Configure)
 
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NAT on FreePBX
Views: 528 Minh Hoàng Lê
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: 7176 pascom GmbH & Co. KG
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: 5745 pascom GmbH & Co. KG
Config Asterisk
 
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Compañeros aquí esta el scrip *****SIP***** [general] context=default dtmfmode=rfc2833 canreinvite=yes port=5060 udpbindaddr=0.0.0.0 allowguest=no srvlookup=yes [anexo] context=[contexto] type=friend user=[anexo] secret=[Password] callerid=[ID] nat=yes qualify=yes host=dynamic mailbox=[contexto buzon] *****VOICEMAI***** [general] #include vm_general.inc #include vm_email.inc [contexto buzon] [anexo]=[pass],[id],[correo] *****DIALPLAN*****(Importante despues del = agregar signo mayor [contexto] exten = _910X,1,DIAL(SIP/${EXTEN:1},6) same = n,Voicemail(${EXTEN:1}@[contexto buzon]) same = n,Hangup() include = msgvoz [msgvoz] exten = *99,1,answer() same = n,VoiceMailMain(${CALLERID(num)}@[contexto buzon])
Views: 121 Gerald Arriola
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: 60392 nirvgorilla
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: 8815 nirvgorilla
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: 2595 n2v Solutions LLC
NAT Traversal & RTSP
 
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Views: 17297 william wong
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: 11375 YayDotCom
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: 4913 pascom GmbH & Co. KG
Demo NAT on FreePBX (Part 1 - Prepare)
 
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NAT on FreePBX
Views: 728 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: 11938 marinko113
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: 35058 NCH Software
M0n0wall NAT Forwarding and firewall rules
 
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How to creat a NAT rule and firewall rule
Views: 18512 squirrel8472
How to configure CUBE with SIP Trunk with free ITSP for Home Lab use
 
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Timeline: 00:00 – Intro 01:30 – SIP-UA.com flyby 02:06 – Quick review where CUBE fits 05:30 – Configuring CUCM 11.5 Dialplan 08:56 – “Call Legs” Review 12:06 – Review of current dial peer configs on CME router 14:03 – Configuring dial peer from PRI to SIP-T router 15:25 – Reivew PSTN facing router configs 15:48 – Configuring PSTN facing router CUBE, SIP Authentication, and SIP Registrar 23:26 – Review of dial peer preference order 24:51 – How to remember the dial peer order preference 25:24 – Configuring PSTN facing router inbound dial peers 29:33 – Derailed; ESXi losing it’s marbles 31:00 – Continuing with ESXi post RAID drive rebuild 32:41 – Post checks since ESXi HD failure 35:33 – Configuring PSTN facing router voice translation rule 37:06 – Configuring PSTN facing router outbound dial peer toward CUCM 39:33 – Configuring CME router dial peer toward CUCM 41:17 – Placing inbound test calls from PSTN to CUCM and Troubleshooting 48:22 – Configuring PSTN facing router for public ITSP calls (US: 10 and 11 digit dialing) 51:53 – Placing outbound test calls from CUCM to PSTN and Troubleshooting 54:38 - Closing
Views: 7335 wmx99
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
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: 35573 nurango
Mikrotik VoIP SIP Server Port Redirect rules setup
 
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Mikrotik VoIP SIP Server Port Redirect rules setup
Views: 27288 Tania Sultana
Install Asterisk on CentOS 6 Linux
 
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How to install Asterisk on CentOS 6 Linux 64Bit, this video guide is designed to backup my blog post over at http://techspotting.org/install-asterisk-on-centos-6 all other links & related info can be found over at TechSpotting. This guide shows you step by step how to install Asterisk 1.8 on CentOS 6 Linux, this tutorial should also work on RHEL (Redhat) and most likely Fedora Linux. Asterisk is an enterprise level telephone system that uses real desk phones capable of seamlessly calling normal phones, your users probably wont even realise they are using a VoIP phone system. Asterisk VoIP PBX can connect to existing ISDN30e / POTS telephone lines or over the Internet via a ITSP. Any problems or questions drop me a comment below, and don't forget to follow TechSpotting http://twitter.com/TechSpotting and subscribe to our RSS http://feeds.feedburner.com/tech-spotting & YouTube channel.
Views: 25500 HowToGuideSites
NAT Tranversal
 
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Views: 3544 Vinson Hsieh
Installazione e Configurazione Iptables Asterisk
 
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Guida per installare e configurare iptables su sistemi asterisk, il presente video è il primo di una serie di video per mettere in sicurezza asterisk.
Views: 997 Ecommerciando
Configuring NAT Port Forwarding in pfSense 2.0
 
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Configuring NAT port forwarding in pfSense 2.0. If you found this video helpful, check out my pfSense blog at: http://pfsensesetup.com
Views: 69544 David Zientara
What is SIP?
 
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Introduces SIP - the Session Initiation Protocol. The first lesson from http://sipsense.com, the smarter way to learn SIP. Check out http://youtu.be/FBNB-EhfHPI for an overview.
Views: 177985 sipsense.com
SIP Gateway Test
 
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SIP Gateway Test
Views: 480 MISCHMI4
How to remotely register an extension?
 
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MyPBX supports remote registration, so users can register remote extensions to MyPBX from other places to realize remote working.
Views: 8892 Yeastar
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: 7223 Rocket City Tech
Tpad and Xlite Softphone - How to bypass ISP VoIP Blocks
 
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If your ISP blocks VoIP or you get error 408 in xlite softphone, simply add these extra SIP settings to your xlite softphone to make free / low cost VoIP calls over your internet connection. They include STUN settings (stun.fwdnet.net:3478) to bypass NAT problems and also changing the SIP port from 5060 to 8891 More information can be found here: http://www.tpad.com/tpad-free-softphone-download/
Views: 10551 Steven Johns
pfSense 2.3 - Curso Grátis - Aula 12 - Como realizar NAT com Port Forward
 
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Nesta vídeo aula você irá aprender como criar regras de nat de entrada, mais especificamente o encaminhamento de portas (port forward). Este recurso é muito útil em situações onde você deseja publicar serviços da sua DMZ ou rede interna, como por exemplo, acessar via remote desktop o servidor interno. Esta aula faz parte do curso completo gratuito sobre pfSense 2.3. Faça parte da nossa lista VIP e receba dicas e tutoriais gratuitamente. http://www.cavalcantetreinamentos.com.br/site/assinar-youtube Compartilhe essa ideia com seus amigos e ajude o canal crescer em conteúdo.
05  NAT and Private IP's
 
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Views: 109 telco ITcom
IP04 QuickStart
 
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http://alsacecom.fr/go/ip04 Le IP04 est une appliance Asterisk avec quatre ports FXO ou FXS. Il comprend un système d'exploitation Open Source pré-installé Linux avec les fonctions de proxy SIP/IAX2 et NAT. Il fournit ainsi une plateforme solide et compacte pour des communications traditionnelles sur le réseau téléphonique comme pour des communications Voix sur IP. L'IP04 s'adresse principalement aux TPE/PME avec une interface graphique facile d'utilisation, fournissant une solution à bas coût pour leurs besoins en communication. Avec l'IP04, une société avec des agences dans différents pays ou régions peut très facilement s'organiser pour rationaliser les communications à travers un système commun virtuel et à travers Internet limitant ainsi les dépenses téléphoniques.
Views: 1061 alsacecom
Nat inbound outbound
 
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Views: 1617 Le Minh
freepbx setup part 1
 
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Views: 93 jamal ahmed
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: 16712 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: 746 Yeastar