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Search results “Asterisk sip options ping”
Panasonic KX-HDV130 / KX-HDV100 firmware upgrade
 
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Firmware upgrade takes a bit more than 10 minutes, your SIP settings and phonebook are preserved after the upgrade but I recommend backing them up anyway. To update the firmware you basically have to check the connectivity to the firmware server from the phone (Menu-System Settings-Network Settings-Link Speed), then turn on the web interface (Menu-Basic Settings-Other Option-Embedded Web), login to the web interface from your PC (login: admin, password: adminpass), go to Maintenance-Upgrade Firmware and enter the URL to the first part of firmware. To upgrade to version 8.101 you can enter http://50.3.81.167/HDV130-08.101.fw.part1 The firmware is on my server, you should check if you can download it beforehand and if your phone pings the IP, this server is provided AS IS.
Views: 855 Yury Grigoryev
SIP Training
 
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https://alta3.com/courses/sip Want more information? Our next instructor-led SIP Course will be running the week of July 24-28, 2017 from 10:00 a.m. to 6:00 p.m. EDT. Find more information here: https://alta3.com/courses/sip Enroll in Alta3 Research's 5 Day instructor-led Public Virtual SIP Course OR take a look at the self-paced course here: https://goo.gl/D2L5zi This is an overview of Alta3 Research's SIP Training Course. This presentation reviews the outline of our SIP Essentials class and provides introductory training on what students will learn in class. We hope you enjoy this 30 minute tutorial.
Views: 214579 Alta3 Research, Inc.
Mikrotik VoIP SIP Server Port Redirect rules setup
 
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Mikrotik VoIP SIP Server Port Redirect rules setup
Views: 30415 Tania Sultana
How to setup DHCP for IP Phones to receive phone configuration
 
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Author, teacher, and talk show host Robert McMillen shows you how to how to setup DHCP for IP, or VOIP, Phones to receive phone configuration from the phone switch. This is done by using predefined options in DHCP manager.
Views: 16093 Robert McMillen
What You Need to Know About Delay and Jitter in IP Telephony
 
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http://www.xorcom.com - Technical video discussing causes and treatment of voice quality issues in remote phones, such as latency (delay) and jitter. Covers G.729 license application, fees and policy, including option to run this protocol on a separate board, which relieves the CPU from handling the processing.
Wireshark's tshark duration option
 
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Learn how to use the tshark command line utility with the autostop duration parameter
Views: 45083 The Technology Firm
SIP Record Route Header
 
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This video explains the concept of sip (session initiation protocol) record route header. It explains the concept using two phone & two intermediate proxy.
Views: 10663 Vishal Patel
Cisco IP Phone System - 7960 Configuration For Voip.MS
 
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http://www.anetcomputers.com/cisco-ip-phone-system-7960-configuration-for-voip-ms/ Patreon: https://www.patreon.com/anetcomputers Website: http://AnetComputers.com Facebook: http://facebook.com/anetcomputers Twitter: https://twitter.com/anetcomputers Paypal: http://www.anetcomputers.com/contribute Streamlabs: http://youtube.streamlabs.com/anetcomputers
Views: 11987 Anet Computers
Avaya one-X Communicator SIP Configuration Settings
 
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How to view and change the SIP configuration settings for the Avaya one-X Communicator client. Produced by Bob Kuberski.
Views: 14642 Avaya Mentor
Config Cisco SX20 H323 SIP For Video Conference
 
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Config Cisco SX20 H323 SIP for E-Classroom
Configuring Cisco Unified Customer Voice Portal SIP Server Groups
 
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In this presentation our Cisco Contact Center instructor, Mike Keutzer, gives several examples of configuring Cisco Unified Customer Voice Portal SIP Server Groups. Mike demonstrates on SIP Server Groups in relation to Cisco CVP, shows differences between server groups and static routes, and illustrates some key differences in versions of CVP.
Telephone System Call Routing
 
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Info Level: Beginner Presenter: Eli the Computer Guy Date Created: August 8, 2010 Length of Class: 60 Minutes Tracks Telephone Systems Prerequisites Introduction to Telephone Systems Purpose of Class This class discusses how calls get routed within a telephone system PBX. Topics Covered Extensions Call Paths Out Calling Incoming Trunk Groups Auto Attendants Hunt Groups Call Groups Out Going Call Routing Class Notes Introduction All telephone systems use the same basic concepts to route calls Extensions PBX's relate to everything as an extension. A station is an extension. An Auto Attendant is an extension, etc. You should create a range of extensions for use for stations and subscribers, and a different range of extensions to be used for administrative purposes (Auto Attendants, Hunt Groups, etc) You can determine how many numbers make up an extension (2,3,4) Call Paths Call paths determine how an incoming call is routed. A standard call path states that a station is rung 3 times, and then the call is routed to voicemail. You can have call paths with 20+ steps. Out Calling Out calling allows the PBX to route calls from the outside to outside lines. A call from the outside can be routed to a cell phone. Out calling requires 2 trunk lines (1 for the incoming call, and one for the outgoing call) Out calling can be a HUGE security problem if not administered properly. Incoming Trunks Incoming trunk lines are programmed into Trunk Groups. Individual Trunk Groups are pointed at a specific extension for incoming calls (Usually an Auto Attendant) Multiple businesses in the same building can use the same PBX by putting their phone lines into Trunk Groups and then pointing the Trunk Group to their Auto Attendant. Auto Attendants "If you would like Sales press 1" The message for the Auto Attendant resides on the Voicemail System. You create an Extension, make it an Auto Attendant, point the message to a Voicemail box, determine what will happen when users press number keys, determine what happens if the user does nothing. Hunt Groups Are Extensions that when called ring a series of other extensions in order. If the first extension in the hunt Group is busy, the next extension in the Hunt Group is rung. Weighted or Smart Hunt Groups can route calls to extensions based on programmed parameters. Call Groups Call Groups are Extensions where numerous Extensions are rung at the same time when the Extension is dialed. Outgoing Call Routing You can create Outgoing Trunk Groups based on whether the trunk lines have local, long distance, or international calling privileges. You can create codes to allow managers to be able to access any Outgoing Trunk Group. Outgoing Call Routing is based on the number of digits dialed, and whether those digits match a pattern that allows the call to be routed to a specific Outgoing Trunk Group.
Views: 156624 Eli the Computer Guy
Alcatel-Lucent 8028 / 8029 Premium Deskphone on OXE - Demo and User Guide
 
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Overview for the 8028 and 8029 desk phones used on the Alcatel-Lucent OmniPCX Enterprise system. 1) External headset - 0:30 3) Handset/Electronic hook flash - 0:40 3) Fixed feature keys - 1:09 4) Mute key - 1:27 5) Volume/Contrast keys - 2:19 6) Speakerphone key - 2:30 7) Hold key - 3:05 8) Transfer key - 3:15 9) Redial key - 3:27 10) Info key - 3:58 11) C key - 4:25 12) Navigation array - 4:35 13) Message key - 5:14 14) Dynamic feature keys - 5:33 15) Optional QWERTY keyboard - 5:57 16) Dial-by-name - 6:21 17) Smart display - 6:52 18) Program a key - 7:15 19) Erase a key - 8:35 20) Make a call - 9:22 21) Place a call on hold - 10:45 22) Make a second call with one call on hold - 11:01 23) Toggle between two calls - 11:16 24) Callback request - 11:59 24) Forward a phone - 13:14 25) Manage Messages - 14:50 26) Menu tab - 16:17 27) Adjust screen contrast - 17:02 28) Adjust ring tone - 17:18
Views: 40363 ICONVoiceNetworks
Configure a Softphone for your PBX or VoIP account
 
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Using X-Lite and Bria as an example, we show the basic settings needed to connect your softphone with an Elastix Asterisk-based PBX, as well as an individual VoIP service.
Views: 6688 VoicePulse
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: 12218 Tania Sultana
60# Kali Linux - DHCP Spoofing
 
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60 - Kali Linux - DHCP Spoofing
Views: 15 Noob Engineer
Cisco Voice Gateway Configuration with CUCM | H.323 Gateway | PSTN FXO | Easy Steps
 
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This lab demonstrates the Cisco Voice Gateway configuration and integration with CUCM for incoming/outgoing calls. The VG has two FXO ports and configured as H.323 Gateway with CUCM. After setup, test calls were made to verify the expected output. Lab Environment ============== 1. CUCM 2. CIPC on Windows Workstation 3. Voice Gateway with 2FXO 4. VMWare Workstation IOU stands for IOS in Unix; is a simulation tools. This lab demonstrates the installation of IOU in VMWare Workstation 10 and integration with GNS3. Integration of GNS3 with IOU is benefited in a sense that you can simulate special type of Router and L2/L3 Switch IOS. Lab Environment ============== 1. Router 2. Windows 7 PC with GNS3 3. Windows 8 PC with IOU 4. VMWare Workstation 10 Please subscribe the channel and give comments. Your opinion is highly appreciated
Views: 49279 Lab Video Solutions
How to setup Caller ID routing on Avaya IP Office
 
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Best viewed in 1080p. Denwa explain how to setup caller ID routing on the Avaya IP500 phone system, programming this feature using the Avaya Manager software is quick and easy. For more information visit our website: http://www.denwa.uk.com/avaya-phone-s...
Views: 27367 Denwa
How to Configure PABX KX NS300 for Panasonic with Algorithms
 
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This video will show you how to configure Phone System PABX KX-NS300 Series with example. After watched this videos hope that you can do it by yourself. Please click on Subscribe button in order to get more videos
Views: 56264 Narith Heang
Capturing with Multiple Interfaces Using Wireshark
 
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Multitrace analysis can be the most interesting, rewarding and unfortunately, most frustrating exercise an analyst will face. Before we get to the packet analysis, setting up your tools for simultaneous capturing can be a feat in itself. The time issue is the most critical when using 2 devices since the time is used to calculate the delay, jitter or latency. Some people are fine with syncing both devices to a common ntp server. Then there's the "how the #[email protected]#!!" do I physically capture . This is where you have to be familiar with the problem, the network you are working on and what equipment is available to you. If you are lucky enough to be able to change the speed and duplex to 100 half duplex a good old hub fits the bill. Other than the mirror/span command, a tap is also very helpful. Trust me every one of these suggestions comes with their own caveats. You may have to try different tools for different scenarios. For example, if I am doing a simple pc bootup/login baseline, I am interested in things like total data transferred, which IP's I am talking to, protocols used, errors, etc. In this case speed and duplex is not important and I can go with a hub. But if I was troubleshooting why something is taking too long, like a backup or replication, changing the speed and duplex would not be a good idea. If you are lucky enough and can capture from one device, the time accuracy issue goes away and life does get a bit easier. But now you have 2 different captures in the same trace, Yikes!!!! Not to mention that different network interfaces have different latency or behaviors. I remember trying a usb to 10/100 ethernet adapter to capture packets and quickly realized that this adapter added 30 ms to every packet. Again, if I was troubleshooting latency, this won't do. Lastly, if you're fortunate enough, you might even have an application that takes multiple trace files and calculates all sorts of stuff out for you (hmm.. next article?). In this example I use Wireshark, my laptops WiFi and Ethernet ports to capture my packet traversing a residential home router. I show some tips and tricks along the way and hope this will help you out. Linkedin Profile http://ca.linkedin.com/in/fortunat Lovemytool Blog: http://www.lovemytool.com/blog/tony-fortunato/ Youtube Channel: http://www.youtube.com/user/thetechfirm ________________________________________
Views: 9790 The Technology Firm
How to Configure an Avaya B179 Conference Phone to Register to SIP Enablement Services
 
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Configure the B179 in both CM and the B179. Configure the B179 to register to the SES server. Produced by Mike Cannon.
Views: 28903 Avaya Mentor
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: 16794 TEK411.com
Grandstream IP PBX UCM6100  setup in 10 minutes (UCM6102, UCM6104, etc)
 
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Quick setup and configuration of Grandstream UCM6100 IP PBX. Creating Extensions Creating Trunks Creating Inbound routes Creating Outbound routes
Views: 129560 LucidPhone
How to stop calls from unwanted numbers 100, 200, etc to your VoIP device
 
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How to stop calls from numbers like 100, 200, 300 to your VoIP device.
Views: 3903 LucidPhone
Cisco packet tracer: How to, Basic IPphone Configuration
 
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In this video i show you how to configure a Cisco router to allow IP phone to get a line number and dial over the LAN. i use the built in call manager and telephony services in the router to give ip and line numbers to 2 different ip phones I am putting a more advanced tutorial up soon Please Subscribe, Like, and Comment
Views: 120066 deadatzero
What is INVITE OF DEATH? What does INVITE OF DEATH mean? INVITE OF DEATH meaning & explanation
 
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What is INVITE OF DEATH? What does INVITE OF DEATH mean? INVITE OF DEATH meaning - INVITE OF DEATH definition - INVITE OF DEATH explanation. Source: Wikipedia.org article, adapted under https://creativecommons.org/licenses/by-sa/3.0/ license. An INVITE of Death is a type of attack on a VoIP-system that involves sending a malformed or otherwise malicious SIP INVITE request to a telephony server, resulting in a crash of that server. Because telephony is usually a critical application, this damage causes significant disruption to the users and poses tremendous acceptance problems with VoIP. These kinds of attacks do not necessarily affect only SIP-based systems; all implementations with vulnerabilities in the VoIP area are affected. The DoS attack can also be transported in other messages than INVITE. For example, in December 2007 there was a report about a vulnerability in the BYE message ("BYE BYE") by using an obsolete header with the name "Also". However, sending INVITE packets is the most popular way of attacking telephony systems. The name is a reference to the ping of death attack that caused serious trouble in 1995-1997. The INVITE of Death vulnerability was found on February 16, 2009. The vulnerability allows the attacker to crash the server causing remote Denial of Service (DoS) by sending a single malformed packet. An impersonator can, using a malformed packet, overflow the specific string buffers, add a large number of token characters, and modify fields in an illegal fashion. As a result, a server is tricked into an undefined state, which can lead to call processing delays, unauthorized access, and a complete denial of service. The problem specifically exists in OpenSBC version 1.1.5-25 in the handling of the “Via” field from a maliciously crafted SIP packet. The INVITE of Death packet was also used to find a new vulnerability in the patched OpenSBC server through network dialog minimization. For the popular, open source-based Asterisk PBX there are security advisories that cover not only signaling-related problems, but also problems with other protocols and their resolution. Problems may be malformed SDP attachments where codex numbers are out of the valid range or obsolete headers such as “Also”. The INVITE of Death is specifically a problem for operators that run their servers on the public internet. Because SIP allows the usage of UDP packets, it is easy for an attacker to spoof any source address in the internet and send the INVITE of death from untraceable locations. By sending these kinds of requests periodically, attackers can completely interrupt the telephony service. The only choice for the service provider is to upgrade their systems until the attack does not crash the system anymore. A large number of vulnerabilities exist for VoIP phones. DoS attacks on VoIP phones are less critical than attacks on central devices like IP-PBX, as, usually, only the endpoint is affected.
Views: 146 The Audiopedia
Configuring and capturing Debug Recording on an AudioCodes MediaPack or Mediant Device - Version 7.0
 
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This video demonstrates how to configure and capture Debug Recording traces utilizing the Logging Filter rules from the web browser and capturing the output in Wireshark. Version 7.00
Views: 4067 AudioCodes Media
Demo 2 nmap and Wireshark
 
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made with ezvid, free download at http://ezvid.com
Views: 34 Sandi Samuel
Voip Service - One of the best
 
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Click for a 30 day risk free trial of VOIPo at http://bit.do/voipservice Voice Over IP (VOIP) -- The best way to make unlimited calls to the US & Canada. You can call from anywhere via the internet with your laptop, cordless phone and smart phone. It is very flexible. It has over 40 incredible features including, 3 way calling, advanced voicemail, call forwarding and lots more. Take advantage of the unbeatable price of $6.21/month & free equipment which saves you at least 75% on your phone bill. Click link below for special pricing. Get the free 30-day trial today. http://bit.do/voipservice
Views: 2830 voipservice
Wireshark Tutorial 1BITC S1G1 2016/2017 UTeM - STEP BY STEP 100% WORKING  NOT SCAM
 
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https://www.wireshark.org/download.html
Views: 21 Amirul Faiz
Packet Capturing with TCPDUMP command in linux
 
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Packet Analyzer TCPDUMP Command 1. tcpdump command is also called as packet analyzer. 2. tcpdump command will work on most flavors of unix operating system. 3. Save the packets that are captured. 4. So that we can use it for future analysis. 5. open source software like wireshark to read the tcpdump pcap files. 6. Wireshark works only in graphical interface, tcpdump on CLI.
tcpdump: filters: logical operators NOT, AND, OR
 
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tcpdump: logical operators NOT (!), AND (&&), OR (||) and more options for filtering TCP flags
Views: 83 AskFrank15
Phone Love
 
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What would it be like if someone fell in love with their Cisco IP Phone? (Video to promote the new phone system to employees)
Views: 232 Bryan Parsons
Spa303 dns fail test 2
 
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Firmware 7.4.5(01261122) Invalid primary dns server
How to multi public WAN IP's with PFSense
 
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How to setup and configure your pfsense box to work with multiple public IP addresses from your ISP. I offer no guarantee that this will work for you 100%, this may also not be the standards way of configuing it but it seems to work for me and I have my various servers configured to work with my public ip's in pfsense. Based on my UK isp.
Views: 124533 0oD4nK1rbo0
Secret Codes in HUAWEI P9 - HUAWEI Hidden Menu
 
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These publicly available backchannels allow users to directly communicate with their service provider's computers and/or access back-end features in their device. They are accessed by inputting them into the phone's dialer (the screen you use to start a phone call) and usually begin and end with the * or # keys with a sequence of numbers in between (there's close-to-zero chance that anyone would accidentally access them). More info: http://www.hardreset.info/devices/huawei/huawei-p9/
Views: 150580 HardReset.Info
Zyxel USG Series - How to Set Up Bandwidth Management (BWM, QoS)
 
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One of the main reasons users enable bandwidth management (QoS) is to prioritize VoIP traffic. This guide will provide instructions on creating a QoS rule to prioritize VoIP as well as prioritizing traffic for specific device(s). http://onesecurity.zyxel.com/img/uploads/ZyWALL_BWM_Setup.pdf More guidelines and tutorials can be found here: http://onesecurity.zyxel.com/tutorials/ If you need support, please contact us via http://www.zyxel.com/form/contact_support.shtml?
Views: 14942 Zyxel
How to reset network settings on 7841 series phone
 
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Suyash Pal Singh is a customer support engineer in Cisco TAC team for Unified Communications technology based in Bangalore. His areas of expertise include Cisco Unified Communications Manager and UC applications which integrates with it. He has over 7 years of industry experience working with large enterprises and Cisco Partners. He holds a Bachelor of Engineering degree in Electronics and Telecommunication from Rajasthan University. He also holds CCIE certification (#50865) in Collaboration technology.
Views: 4736 Cisco Community
How to perform factory reset on 7841 phone using keypad
 
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Suyash Pal Singh is a customer support engineer in Cisco TAC team for Unified Communications technology based in Bangalore. His areas of expertise include Cisco Unified Communications Manager and UC applications which integrates with it. He has over 7 years of industry experience working with large enterprises and Cisco Partners. He holds a Bachelor of Engineering degree in Electronics and Telecommunication from Rajasthan University. He also holds CCIE certification (#50865) in Collaboration technology.
Views: 52233 Cisco Community
Configure Polycom RealPresence Group 300 as Lync 2013 endpoint
 
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This video demonstrates the steps required to make a Polycom Group 300 VC unit work with Microsft Lync 2013
Views: 21003 1plex Ltd
VoIP training with certification - Queues - Virtual PBX training videos
 
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Complete the whole course of free VoIP online training to receive free certification. What are Queues, and how to implement them in CompletePBX 5, virtual software PBX, free training video. Download and install free virtual PBX software and start the training here https://xorcom.com/ip-pbx-online-training/ Visit our website for free PBX online training and courses. Complete online VoIP certification for free. Xorcom virtual IP PBX for hotels, enterprise, schools, call centers, free for download. Asterisk-based IP PBX appliances and hardware, phone systems for call centers.
Setting up your home phone on the nbn™ network. FTTN or FTTB
 
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Learn how to setup your home phone service on the nbn network, Fibre to the Node (FTTN) or Fibre to Building (FTTB). For more info on connection to the nbn network, visit https://www.telstra.com.au/broadband/nbn/how-to-connect
Views: 65638 Telstra
Basic Setup for Fanvil X4 IP Phone
 
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This Video shows Basic IP Phone Setup for Fanvil X4
Views: 9720 DiscoverBD
An Introduction to WebRTC
 
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WebRTC is an open project that enables web browsers with real-time communications capabilities via simple Javascript APIs.
Views: 14459 Google Developers
Bulk Destination Set Route Managment
 
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This tutorial demonstrates how you can create, update and delete Routes, Rates & Accounts in your Sippy softswicth using the bulk upload facility. http://www.sippysoft.com/
Views: 4503 Sippy Software, Inc.