Search results “Asterisk peer options”
Asterisk basic configuration: SIP Extensions
This video features a SIP extensions setup procedure for the IP PBX Asterisk on Linux environment. » TUTORIAL: • http://techexpert.tips/asterisk/asterisk-sip-extension-on-ubuntu-linux/ » EQUIPMENT LIST: • Power supply 500w - http://amzn.to/2zwjbf0 • Power cord - http://amzn.to/2ze41bp • Mother Board - http://amzn.to/2zwvJDn • Processor - http://amzn.to/2y0cXj9 • Hard disk - http://amzn.to/2rlDB7p
Views: 3809 FKIT
FreePBX simple config Простая настройка FreePBX asterisk 13
На видео показана самая простая типовая настройка freepbx. Простая настройка 1 транка для примера Создание 1 extension для примера Создание 1 входящего маршрута на extension Создание 1 входящего маршрута на ring group Создание 1 Исходящиего маршрута с диалпланом на мобильные Включение записей разговоров на 1 extension Отключение anonymous sip calls в advanced settings
Views: 4844 rstayalive
Getting the Best out of WebRTC - AstriCon 2014
This talk is aimed at giving you the tools to help you scale your web webRTC deployments while still meeting your user's needs. Session will cover: User expectation - what kind of connectivity do your users have and what quality are they expecting. Do they need video or wide band audio? Topology - where are your users, will they communicate Peer to Peer or is this a star topology? Monitoring - what tools are there to monitor quality? Selecting your codec - which codec is best for your situation? Tuning your codec - can you optimise the codec settings for your environment? Auxiliary servers STUN/TURN/Proxy - What other servers will you need? Crypto choices - what options do you have for cryptography and identity management? Transcoding - how much CPU time will this cost?
Flowroute SIP Trunk Setup on FreePBX
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: 29826 Crosstalk Solutions
Asterisk Tutorial 15 - Asterisk Subroutines [english]
Ever wanted to know how to get rid of all those lines of code that repeat themselves over and over again? Today we get yet even more real world like by reducing our business hours dialplan settings to just 2 lines of subroutine coding. In our example, we demonstrate how to use a subroutine to remove the unnecessary lines of dialplan coding when setting up your business hours - although subroutines are by no means limited to solely this function. Important information here is, if you can avoid using the "macro" function, you should, as this option will only provide a depth of seven levels, after which Asterisk will probably crash - use the "GoSub" application instead. For more information on our turnkey business phones systems and UC solutions, visit our website: https://www.pascom.net/en/
Views: 8251 pascom GmbH & Co. KG
Asterisk Jabber Notifications - http://geekhut.org
An additional module added to Asterisk Stickies that immediately sends Jabber instant messages with option to take call notes or view caller info. Built on asterisk stickies 1.0 (now version1.2).
Views: 1365 Gregory
Asterisk Tutorial 33 - Asterisk IVR Menu Looping [english]
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 on our turnkey business phones systems and UC solutions, visit our website: https://www.pascom.net/en/
Views: 3749 pascom GmbH & Co. KG
Configuring a SIP Peer
We support a couple of Peering options which you manage yourself via the CloudPBX portal
ASTERIkast Episode 1 - Installing Asterisk 1.2.1
This is our first release. We know that the video is a bit crapy, however we may reshoot this episode. We mainly show you how to get asterisk up and running on Slackware Linux. We also show how to setup a sip phone.
Views: 1843 Tony Virelli
How To Setup SIP Calling On Android
Setup SIP account on android phones and tablets with getonsip. It allows you to make free Wi-Fi calls without using google voice and skype. Full tutorial for Motorola, Nexus, Samsung and Sony Xperia devices. http://www.pcnexus.net/2014/06/how-to-setup-sip-account-for-voip-calling-android-without-google-skype.html
Views: 42039 Pcnexus
CallXML script development - SIP REGISTER and INVITE
example video recording how we write CallXML script to test SIP PBX (or SBC) in StarTrinity SIP Tester
Views: 227 Sergey A
Asterisk Tutorial 21 - Introduction to Call Queueing [english]
Welcome back to our Introducing Asterisk Series Building on from our last tutorial on Automatic Call Distribution (ACD) in Asterisk, today's tutorial focuses on Call Queueing as we take a look at the queues.conf and what you need to do to configure your queues. For more information on our turnkey business phones systems and UC solutions, visit our website: https://www.pascom.net/en/
Views: 7810 pascom GmbH & Co. KG
DTMF Issue - voice-class sip dtmf-relay force rtp-nte
The issue was that when a Sprint customer called in to our UCCX IVR main auto attendant they have 2 options (1 for ENG or 2 for SP) those options work fine, no problems. But once the customer was in the sub menu the first DTMF digit pressed would not register and would only register on the 2nd press of the DTMF digit. It only happened when using a Sprint cell phone and only happened with our sub menus but not the main AA. The solution was to enter the command 'voice-class sip dtmf-relay force rtp-nte' under dial-peer voice 200 voip which is our dial-peer for CUCM. That is a hidden command and will not show up as an option in the IOS when using the help function '?' . But it will work if entered as is under a dial-peer. This Command ensures that the CUBE will always uses RFC2833 for DTMF even if it was not offered by the provider in the initial invite. Your SIP provider must support RFC2833, and lucky for us, most providers will because RFC2833 is pretty common. ***INFO*** voice-class sip dtmf-relay force rtp-nte ---------------------------------------------------------------------------------------- https://anetworkerblog.com/2011/02/06/dtmf-on-voip/ https://supportforums.cisco.com/discussion/10709181/dtmf-relay-unrecognized-command-cli Understanding DTMF --------------------------------------------------------------------------------- DTMF Relay - http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/dtmf-relay.html DTMF AND RFC 2833 / 4733 - https://andrewjprokop.wordpress.com/2013/09/27/dtmf-and-rfc-2833-4733/ Understanding DTMF negotiation and troubleshooting on SIP Trunks - https://supportforums.cisco.com/document/144711/understanding-dtmf-negotiation-and-troubleshooting-sip-trunks Configuring and debugging DTMF (RFC 2833) - https://blogs.msdn.microsoft.com/rita_z/2005/10/10/configuring-and-debugging-dtmf-rfc-2833/ ==================================================== Multiple DTMF Methods ----------------------------------------------------------------------- Multiple DTMF methods may be configured on CUBE simultaneously in order to minimize MTP requirements. If you configure more than one out-of-band DTMF method, preference goes from highest to lowest in the order of configuration. If an endpoint does not support any of the DTMF relay mechanism configured on CUBE, an MTP or transcoder is required. Cisco UCCX jtapi ports only support out of band DTMF, you can configure your cube dial-peer pointing to CUCM to use both rtp-nte and sip-kpml. SIP-KPML will be out of band and hopefully you will not need MTP. Example: Router(config)# dial-peer voice xx voip Router(config-dial-peer)# dtmf-relay rtp-nte sip-kpml Source - https://supportforums.cisco.com/discussion/12394051/dtmf-incoming-over-sip-trunk-not-working
Views: 2281 W00DY1848
voip protocol :: Session Initiation Protocol SIP Overview, rfc 3261, wireshark sip tutorial New 2014
Session Initiation Protocol (SIP) is an application-layer control protocol that can establish, modify, and terminate multimedia sessions such as Internet telephony calls. This video will demonstrate SIP Packets flow in Wireshark, SIP Register,INVITE Wireshark SIP protocol Tutorial
Views: 21875 Zariga Tongy
X-lite Softphone setup with Asterisk
How to use a software based phone or Softphone to make calls using Asterisk Thanks to www.HotButteredIT.com for sharing this video.
Views: 41199 VoIPalicious
Asterisk Tutorial 22 - Queue Call Strategies [english]
Hey Guys, Welcome back to the Introducing Asterisk Series. Following on from last week, where we introduced the concept of Call Queues, this time we take a more advanced look at the Queue Application & explain in more detail the Call Strategies available to you & the different timeout options, what they are, how they differ and why they are important. For more information on our turnkey business phones systems and UC solutions, visit our website: https://www.pascom.net/en/
Views: 6192 pascom GmbH & Co. KG
Asterisk MeetMe
Views: 1134 Lukáš Palacký
Testing with SIPP - AstriCon 2014
This session will introduce how to use SIPP. Many integrators or developers are troubleshooting their SIP problems on their network and this software is a perfect tool to replicate some call flows. The session will explain the vocabulary, the exchange of SIP messages and how to create different scenarios for your different needs when troubleshooting your SIP Networks. Clod will also cover a few other useful options when running SIPP that will make your troubleshooting easier.
FreePBX 101 v14 Part 5 - Endpoint Manager
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: 2334 Crosstalk Solutions
Configure your Twilio Elastic SIP Trunking with FreePBX - Part I: Placing outbound calls
There are three steps to connecting Twilio Elastic SIP Trunk to your FreePBX. In this video, we are going to go over the Trunking Termination - which is the first step to start placing calls from your communications infrastructure to the PSTN.
Views: 10951 Twilio
SIP Training Class
A quick hands on overview of the REGISTER method. We discuss the basics of the SIP header fields and how the MD5 Authentication process is used with SIP.
Views: 1838 TrainingCity
How to install and configure Asterisk in CentOS
This video explains the Installation and configuration of Asterisk . It is a free server for creating communication applications. For more explanation on this video: https://www.linuxhelp.com/how-to-install-and-configure-asterisk-in-centos/
Views: 2060 Linux Help
Making Your Computer Accessible to the Public Internet: STUN (4 of 4)
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: 34347 NCH Software
Troubleshooting Choppy Audio
This is a tutorial video on how to troubleshoot choppy VoIP audio using wireshark. Supporting files are located here: https://drive.google.com/folderview?id=0B_9grX6Kpvq1VmFlSGcwbzlKVGs&usp=sharing
Views: 17185 Terrell Boyer
Converting AVAYA phone Software from H.323 to SIP
to convert your AVAYA H.323 phone to work on SIP FW : 1- upload SIP FW on your FTP server 2-configure your phone SIG to be SIP 3-reset phone to take new FW 4-configure phone to new SIP parameters
Views: 21278 shika joe
SIP Trunking Configuration in Avaya Communication Manager
The video will detail the SIP Trunking configuration in Avaya Communication Manager. Produced by Chandrakandan, Chandragiri.
Views: 31449 Avaya Mentor
V14 Configuring ShoreTel SIP Trunks - P2  (Using SonicWall as SBC)
Can you configure SIP trunks without a Session Boarder Controller? The simple answer is yes, the better answer is no. Clearly you can configure your firewall to manage SIP trunks, but is it really worth the effort? Bringing a SIP trunk through the same Firewall as your normal Internet traffic has problems, adds latency and lacks the feature set of a dedicated SBC. This video takes a look at the Firewall SIP configuration options and the illustrates how to configure a SonicWall to support ShoreTel SIP. The first part covers some theory, the second half cover the actual configuration.
Views: 2876 DrVoIP
Asterisk installation and configuration on CentOS7
Here is a simple and easy way to install Asterisk over CentOS 7 [[[[[[[[[[[[[[[[[[[[[[[[[[[[[[[[[[[[ Asterisk is an open source telephony switching and private branch exchange service for Linux and is completely free framework for creating programs and is subsidized by Digium. Asterisk can change a common pc into a emails server. Asterisk abilities IP PBX methods, VoIP gateways, conference web servers,customer solutions and is used by companies such as telecommunication, suppliers and for countries worldwide For this install I am using Asterisk 11.0.0 and will be compiling from source on CentOS 7. This tutorial should also work on Fedora and RHEL (Red Hat Enterprise Linux) systems with little or no modification. First type command su Then, you will want to be sure that your server OS is up to date. yum update -y Disable SELinux by changing “enforcing” to “disabled” in /etc/selinux/config. Use a text editor or copy and paste this command. sed -i s/SELINUX=enforcing/SELINUX=disabled/g /etc/selinux/config After you update and disable SELinux, you’ll need to reboot. reboot Next, you will want to resolve basic dependencies. (More information on Asterisk dependencies.) yum install -y make wget openssl-devel ncurses-devel newt-devel libxml2-devel kernel-devel gcc gcc-c++ sqlite-devel libuuid-devel Change into the /usr/src/ directory to store your source code. cd /usr/src/ Download the source tarballs. These commands will get the current release of DAHDI 2.6, libpri 1.8 and Asterisk 11. wget http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-current.tar.gz wget downloads.asterisk.org/pub/telephony/libpri/libpri-current.tar.gz wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11-current.tar.gz Extract the files from the tarballs. tar zxvf dahdi-linux-complete* tar zxvf libpri* tar zxvf asterisk* For the next set of commands it is important to follow the proper order: DAHDI first, then libpri, then Asterisk. Install DAHDI. cd /usr/src/dahdi-linux-complete* make && make install && make config Install libpri. cd /usr/src/libpri* make && make install Change to the Asterisk directory. cd /usr/src/asterisk* In the next step, running the “configure” script will vary depending on whether your system is 32-bit or 64-bit. (Watch the video for more details.) When the menu select command runs, select your options, then choose “Save and Exit” and the install will continue. Use this command if you are installing Asterisk on 32bit CentOS. ./configure && make menuselect && make && make install Use this command if you are installing Asterisk on 64bit CentOS. ./configure --libdir=/usr/lib64 && make menuselect && make && make install Optional: If you ran into errors you will want to clean the install directory before recompiling. make clean && make distclean Once you have an error-free install, copy the sample files from the configs subdirectory into /etc/asterisk. make samples Then add the Asterisk start script to the /etc/init.d/ directory make config Start DAHDI. service dahdi start Enable the Asterisk services at system boot time. chkconfig asterisk on Start Asterisk. service asterisk start Connect to the Asterisk CLI. asterisk -rvvv And now you have Asterisk 11 running on CentOS 7! If you’d like to continue configuring Asterisk you can check out this guide to setting up basic pbx functionality or leave a comment to share your thoughts below! You can also check out some of our training and certification options.
Views: 7887 Fakhar Ali
Grandstream IP PBX UCM6100  setup in 10 minutes (UCM6102, UCM6104, etc)
Quick setup and configuration of Grandstream UCM6100 IP PBX. Creating Extensions Creating Trunks Creating Inbound routes Creating Outbound routes
Views: 121979 LucidPhone
How to View SIP Message Logs for Radvision iView B2BUA
This video will demonstrate how to view SIP message logs for the Radvision iView SIP back-to-back user agent (B2BUA). Produced by Russell Singer.
Views: 821 Avaya Mentor
SIP Internet Calling and VoIP on Galaxy S4
http://www.androidmobilevoip.com/2013/04/native-voip-sip-client-in-samsung.html Samsung Galaxy S4 has inbuilt Internet calling facility to use your VoIP to place Internet calls. VoIP is a great way to save money on your call when you are in roaming. You just need an internet to place a call without roaming charges. This video helps in how to set up sip voip account with galaxy S4
Views: 53341 dialandroid
how to create aws peering connections tutorial
AWS peering connection tutorial aws vpc peering
Views: 8762 Zariga Tongy
justforfunc #11: code review of an IRC package's API
Let's review and improve the API of a IRC client! We want the API to be easier to use, but even more important, harder to misuse. The original code is on https://github.com/davidjpeacock/shelbot The PR is https://github.com/davidjpeacock/shelbot/pull/36 More references on Functional Options: - Original blog post by Rob Pike https://commandcenter.blogspot.com/2014/01/self-referential-functions-and-design.html - Dave Cheney's talk: https://www.youtube.com/watch?v=24lFtGHWxAQ
Configuring IP Phones and DHCP Server in Packet Tracer
In this video you will learn how to configure IP Phones in a network of voice and data. This topology uses one DHCP server to assign IP addresses to both phones and PCs which are in different VLANs.
Views: 15955 CBTVid
IP Multimedia Subsystem - IMS Test Suite
Internet Protocol (IP) Multimedia Subsystem, popularly known as “IMS”, is built on Session Initiation Protocol (SIP) as the base to further support packaging of voice, video, data, fixed, and mobile services on a single platform to end users. It provides a unique convergence platform for different types of networks – whether it is mobile, satellite, broadband, cable, and fixed networks, with a goal of building an efficient interoperating networks. GL’s MAPS™ SIP IMS test suite provide an advanced full-fledged network environment that enables user to test their applications, devices, and services prior to deployment on a real-time network. It can be used to simulate all or specific elements within IMS network infrastructure using simple ready-to-use test bed setups. http://www.gl.com/maps-ims-network-simulator.html
FreePBX VoIP Tutorial Part 12 - Now what? Get a free phone number with IPKALL + Hold Music
Want an extra phone number that you can give to people you don't trust such as someone on Craigslist? I show you how with http://www.ipkall.com/ I also mention how you can set up hold music and use it to test call quality with either WiFi or 3g/4g mobile data. These are both completely optional, but are some of the cool things you can do with your own server. Enter the following in PEER Details in your IPKALL Trunk for Outgoing Settings: insecure=port,invite host=voiper.ipkall.com directmedia=no disallow=all allow=ulaw allow=g729 type=peer Jump to 5:56 if you just want to learn how to set up Hold Music. Screenshot for Goldwave settings: http://i.imgur.com/LOB3QOB.png Relevant links: Free Washington area code incoming phone number: http://www.ipkall.com/ Audio editor for resampling Music On Hold files to 16bit 8kHz mono: http://www.goldwave.com/ 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: 14790 nirvgorilla
How to Set Up Trunks and Outbound Routes in CompletePBX
https://www.xorcom.com - In order to place external calls from your CompletePBX telephony system you'll need to set up trunks (the connection between the CompletePBX and your service provider) and outbound routes (the connection between your extensions and the trunk). This short video will demonstrate how to create both of those entities. More detailed information can be found in our CompletePBX Reference Guide, which can be viewed or downloaded from this page: http://www.xorcom.com/completepbx-technical-documentation
The Ultimate SIP Tutorial
This video is a review of a SIP trace using wireshark. The actual call scenario is a call transfer from a phone inside the session border controller to a phone on the outside of a session border controller. This video is intended for people wanting to learn more about SIP.
Views: 51235 Terrell Boyer
SIP Trunking: What Is It?
Continuing our series, we answer FAQs regarding business communication systems including VoIP, Cellular & Cloud-based vs On-Site PBX solutions. SIP Trunks are an extremely effective way of reducing your business communication costs as well as improving efficiency. A SIP Trunk provides your company with a dial tone. They are extremely popular because they consolidate an organization’s connections and are easier and less expensive to operate and maintain. A traditional premise-based telephone systems will connect to the service provider using a Primary Rate Interface (PRI) circuit and their connections are physical, meaning each circuit requires costly hardware. A SIP Trunk’s connections are virtual and the number of Trunks available is down to bandwidth and not the physical termination or hardware. This enables organizations to save money on maintenance and hardware expenses, paying only for what they need. A traditional PRI circuit provides 23 voice channels and so when an organization decides to scale up they would be required to do so in increments of 23. A SIP trunk can offer automatic or on-demand burst capabilities of any number of paths. Another significant advantage of deploying a SIP Trunk is in its ability to provide backup and redundancy capabilities. If necessary, re-routing capabilities can quickly and easily enable the distribution of voice services to alternative sites. Even if you have an investment in a traditional premise-based telephone system a SIP Trunk can still provide significant savings by streamlining costs and resources. Long distance calls cost less for those using a SIP Trunk and once you’re connected to a SIP Trunk then you have the ability to benefit from a number of rich communication methods like video conferencing and Instant Messaging. In essence a SIP Trunk can enable your company to save money and grow as your company grows! We offer a selection of SIP Trunk services with our partners, including Metered and Unlimited options.Our solutions include SIP Trunks that work with Plain Old Telephone Systems (POTS) or even in conjunction with a PRI as well as SIP to SIP (PBX).
Views: 6576 Connect Everywhere
SIP Training
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: 206231 Alta3 Research, Inc.
How to setup a basic outgoing  trunk in Elastix.
How to setup a basic outgoing trunk in Elastix.
Views: 103395 synapseglobal
LEARN | New product Elastic SIP Trunking - Annie Benitez Pelaez & Jonas Borjesson (Twilio)
Register to attend SIGNAL 2016: http://bit.ly/1Rr3C70
Views: 2277 Twilio
Genesys webRTC Demo 3-2013.mp4
Short Demo of Genesys webRTC capabilities
Views: 1814 Genesyslabs
[Hindi] Make call with Private Number ( without Application ), 100% working !
[Hindi] Make call with Private Number ( without Application ), 100% working ►Subscribe Our Channel : http://bit.ly/1OiOnes Dosto aaj ki is video me maine apko btaya hai ki aap kisi ko bhi bina kisi sim ke or bina kisi number ke cll kese lga skte hai or apka number ki jgha dusre ke phone me '' Private Number '' dikha hua jayega, Dosto aaj ke ye video apke liye bahut upyogi hai, main umeed krta hu apko ye video bahut pasand ayegi to is video ko jada se jada LIKE & SHARE kijiye or aise hi usefull video ke liye humare DK Tech Hindi Channel ko abhi hi SUBSCRIBE kr lijiye... video dekhne ke liye Dhaniyabad... Server : sip.call2india.com Site : www.call2india.com You Can Follow Me On Following Platforms-- ►Follow us on Google+ http://bit.ly/28Zjksr ►Follow us on Twitter http://bit.ly/2hJSGbN ►Follow Me on InstaGram https://goo.gl/cQYzXl ►Like us on Facebook https://goo.gl/FD18KX ►Thanks for watching... Please Like, Subscribe and Share This Video
Views: 1478 Mr.Junior Tech
Call Routes - Basics
Call routing allows you to direct incoming calls anywhere and to any phone. Forward calls to your mobile, give your customer a press 1or press 2 option, divert calls during office closing hours or even setup a conference bridge. Call routes can be as simple or complex as you like and offer incredible flexibility.
Views: 168 YayDotCom
FreePBX 101 - Part 9 - How to Forward Calls to a Cell Phone
This is a short video, but a very important feature! How to use Misc. Destinations to forward calls to external phone numbers (mostly used for cell phones). FreePBX 101 - Part 1: https://www.youtube.com/watch?v=LsfqSnGZ3dI FreePBX 101 - Part 2: https://www.youtube.com/watch?v=xBny4hKCM3A FreePBX 101 - Part 3: https://www.youtube.com/watch?v=WgWovGKz5v4 FreePBX 101 - Part 4: https://youtu.be/uNlygMYvNlk FreePBX 101 - Part 5: https://youtu.be/aFrMecTpoyk FreePBX 101 - Part 6: https://youtu.be/EH5XrhtUiSo FreePBX 101 - Part 7: https://youtu.be/YXNTlA3kPWI FreePBX 101 - Part 8: https://youtu.be/8ht-26pBOko Visit http://CrosstalkSolutions.com for FreePBX support, installation, and consulting. Legal mumbo jumbo: FreePBX® is a Registered Trademark of Sangoma Technologies.
Views: 29806 Crosstalk Solutions
Understand SIP in UNDER 2 minutes
CHECK OUT OUR WEBSITE: https://jive.com/ SIP (Session Initiation Protocol) allows people to communicate over the internet with their different devices, but SIP can be a confusing term. Let's break it down to better understand what SIP is and how we use it. WHAT IS JIVE? Jive Hosted VoIP is the easiest and most affordable option for your business phone system. Jive’s cloud VoIP service helps thousands of organizations simplify how they manage their phone setup. FOLLOW US: Facebook: https://www.facebook.com/jive.communications.inc/?fref=ts Twitter: https://twitter.com/GetJive LinkedIn: https://www.linkedin.com/company/66738/ Google+: https://plus.google.com/112881549445108116618 Instagram: https://www.instagram.com/getjive/
Views: 1147 Jive Communications

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