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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: 4982 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: 6813 MikroTik
Mikrotik VoIP SIP Server Port Redirect rules setup
 
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Mikrotik VoIP SIP Server Port Redirect rules setup
Views: 32034 Tania Sultana
Webinar Networking Routing, NAT e QoS in ambito VoIP SIP
 
01:19:43
Corso tenuto da Matteo Sala di Eurylink.
Views: 201 VoipVoice srl
Mikrotik Router Mark VOIP traffic using ip firewall mangle rules
 
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Mikrotik Router Mark VOIP traffic using ip firewall mangle rules
Views: 12710 Tania Sultana
Setting FreePBX dan Mikrotik IP Public Extensions External
 
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di video ini memebuat tutorial setting freepbx dan mikrotik ippublic static dengan extension di mobile
Views: 331 FreePBX-Indo
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: 56754 sollostech
NAT Traversal & RTSP
 
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Views: 19338 william wong
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.
[part 10] Setting up SIP trunk on your FreePBX system so it can talk to the phone company
 
09:36
We repair Macbook logic boards: https://rossmanngroup.com/macbook-logic-board-repair 👉 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: 48804 Louis Rossmann
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/asterisk-tutorial-04-asterisk-network-configuration/
Views: 35881 pascom GmbH & Co. KG
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: 61475 nirvgorilla
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: 7208 Ambiorix Rodriguez
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/asterisk-tutorial-03-asterisk-start-stop-scripts/
Views: 38312 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: 7282 pascom GmbH & Co. KG
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: 14254 YayDotCom
Configuracion Archivo SIP.CONF - Curso de Asterisk PBX - Capacity
 
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Curso Video Asterisk PBX de Capacity IT Academy. Para mas informacion de este curso ir a http://www.capacity.com.do
Views: 15274 Capacity Academy
SIP Helper в RouterOS
 
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SIP Helper в RouterOS by Kirill Vasilev (Vasilev Kirill - MikroTik.Me, Россия) PDF: https://mum.mikrotik.com/presentations/RU_PT18/presentation_5103_1521442154.pdf
Views: 1286 MikroTik
SIP session helper / ALG
 
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SIP session helper / ALG, David Attias (Penny Tone LLC, USA). PDF: https:https://mum.mikrotik.com/presentations/US17/presentation_4321_1496084451.pdf.
Views: 8518 MikroTik
Demo NAT on FreePBX (Part 2 - Configure)
 
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NAT on FreePBX
Views: 723 Minh Hoàng Lê
Demo NAT on FreePBX (Part 1 - Prepare)
 
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NAT on FreePBX
Views: 938 Minh Hoàng Lê
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: 4577 pascom GmbH & Co. KG
GS Tutorials - UCM Peer Trunks - Interconnecting UCMs
 
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In this episode of GS Tutorial, we show you how to interconnect two UCM6xxx series using SIP Peered Trunks. You can find our step by step guide here: http://www.grandstream.com/sites/default/files/Resources/how_to_interconnect_multiple_ucm.pdf Looking for support? Open a trouble ticket here: https://helpdesk.grandstream.com Want to become a Grandstream Reseller? Sign up here: https://partnerconnect.grandstream.com/become-our-partner
Views: 1073 GrandstreamNetworks
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: 168 TeamNANOG
IPPBX Quick Start Guide - Yeastar S-Series VoIP PBX (2019)
 
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If you are new to S-Series VoIP PBX, this quick start guide video will help you find your way around. In this video, we take S20 as an example to give you quick and simple instructions for installing and configuring your S-Series IPPBX. If you are now using Yeastar S-Series VoIP PBX, review the video to see if there's any new found you never know about S-Series VoIP PBX. What you can get: 1. Device installation 2. Web GUI login 3. Localize your PBX with timezone, system prompt, and region preference 4. Change password and set up administration email 5. Configure network settings and reboot your system 6. Bulk create extensions for users 7. Configure Linkus and enable Linkus Cloud Service 8. Connect the Trunks to make calls 9. Download Linkus UC App for your PC and smartphones 10. Create Inbound and Outbound Routes *This video is suitable for S-Series VoIP PBX, including S20, S50, S100, and S300.
Views: 1844 Yeastar
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: 8979 nirvgorilla
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: 2376 Ben Pin
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: 25588 HowToGuideSites
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: 17735 miamimanni
What is a SIP User?
 
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A SIP User is a set of personal credentials that allow you to access and use VoIP. Think of these credentials as a ‘Golden Key’ that can unlock your VoIP service from any device at any location around the world.
Views: 1223 YayDotCom
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: 58585 Crosstalk Solutions
3CX Online Advanced Training: NAT and Port Forwarding
 
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Welcome to the 3CX Online Advanced Training Video Series. In this video, we will discuss port forwarding, why it is needed, and how to use the Firewall Checker to confirm that your WAN-to-LAN device is correctly configured. Skills Covered • Port Forwarding -- When and Why • Which ports to forward for 3CX Phone System • Example Router Configuration • Issues caused by incorrect Port Forwarding configuration • How the Firewall Checker can help • Example Failure Reports from the Firewall Checker Review the full course of 3CX Phone System Training Videos at http://www.3cx.com/blog/training Additional Reading • http://www.3cx.com/blog/docs/firewall-checker/ • http://www.3cx.com/blog/tag/firewall/ • http://www.3cx.com/blog/voip-howto/stun-resolution • http://www.3cx.com/blog/docs/ports-used/ • http://www.3cx.com/blog/support/3cx-multi-tenant-port-list/ Watch the previous video in this course Least Cost Routing - http://youtu.be/YlLIBDKafmQ | Watch the next video in this course 3CX Tunnel Protocol - http://youtu.be/mo8rDGtebV4
Views: 7671 3CX
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: 16826 TEK411.com
Asterisk e Cisco CUCM Integração via SIP Trunk
 
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Visite o blog http://consultoriavoip.luissale.com para maiores informações sobre telefonia VoIP e Redes.
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: 37566 nurango
M0n0wall NAT Forwarding and firewall rules
 
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How to creat a NAT rule and firewall rule
Views: 18643 squirrel8472
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: 1019 Ecommerciando
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.
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: 7238 pascom GmbH & Co. KG
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: 11071 Steven Johns
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 install FreePBX 14 on Hyper-V server and solved some challenges
 
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Original, Approved, Hands-on, Real Life Videos in IT, Network, OS, Hardware, Servers, Firewalls, Routers, Switch, Applications etc The only channel that is backed up by computer specialist experts who will answer your questions. Subscribe it and you can come back to ask question even not related with this video.
Grandstream UCM Advanced Inbound/Outbound Routing Webinar
 
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888VoIP and Grandstream delve into some of the most crucial topics of the UCM: Inbound and Outbound Routing. Master these routing concepts when you watch out latest webinar on-demand: basic UCM scenarios, PIN groups, blacklists, CallerID filtering on inbound calls, and more. For more information on Grandstream's UCM IP-PBX, call the VoIP specialists from 888VoIP at 888-864-7786.
Views: 2715 888VoIP
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: 15038 FKIT
Setting up PORT FORWARD on pfSense
 
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We go through how to setup a standard port forward on pfSense - We do not recommend using this technique only for forwarding ports for RDP purposes, host settings should be setup as well for proper security measures. This tutorial would work well for forwarding ports for something like Plex Media Server. https://www.pfsense.org/ https://forum.pfsense.org/index.php Buy them here - http://www.spaictservices.co.uk/pfsense-store/ ---- Music: Funk Your Style by @TONEZPRO (OFFICIAL) https://soundcloud.com/tonez-pro Creative Commons — Attribution 3.0 Unported— CC BY 3.0 http://creativecommons.org/licenses/b... Music provided by Audio Library https://youtu.be/XNi1pJhFJ5I Like Apollo by jimmysquare https://soundcloud.com/jimmysquare Creative Commons — Attribution 3.0 Unported— CC BY 3.0 http://creativecommons.org/licenses/b... Music provided by Audio Library https://youtu.be/oIpjGVBY9AM ---- Credits: Host/Narration - Nathan Hutchinson Editing/Animation - Harry Jewers
Views: 27494 Chunky Tech
24/7 support and installation  asterisk goautodial vicidial freepbx elastix : call :7599967999
 
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Views: 20 Deepak RG Sharma
Asterisk Part 3  - Using Extensions with Asterisk
 
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In the third video of this 10 part series on Asterisk, I explain how to use "extensions" in Asterisk. We configure extensions to allow users to place and receive VOIP calls. Due to Youtube's small video size, some of command line stuff is hard to see. You can view a 640X480 video on my website: www.HotButteredIT.com. They are currently free, but I'll have to keep an eye out on the bandwidth usage. Enjoy!
Views: 63609 Simons Tech
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: 34125 Crosstalk Solutions
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: 2029 DrVoIP