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Asterisk Tutorial 15 - Asterisk Subroutines [english]
 
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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 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: 8711 pascom GmbH & Co. KG
FreePBX 101 - Part 8 - Queues - Crosstalk Solutions
 
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Welcome to FreePBX101 Part 8 - Queues! This is a BIG video where I cover a lot of detail and options for call queuing on FreePBX. Call queuing is an art, and your settings should be continually tweaked and optimized for efficiency, but this video should give you a great head start. 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 Visit http://CrosstalkSolutions.com for FreePBX support, installation, and consulting. Legal mumbo jumbo: FreePBX® is a Registered Trademark of Sangoma Technologies.
Views: 25417 Crosstalk Solutions
Asterisk Call Center Cisco Phones Vicidial.wmv
 
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Short demo of Vicidial or GoAutodial from an agent perspective. This is with manual calls, not with predictive dialing enabled. The call center application, Vicidial, supports predictive dialing but that is only recommended when you have more than a couple of agents. Call Centers often have many agents and this outbound or inbound call center solution supports many options. Show with the Asterisk system, but also viable with Cisco, Avaya, Toshiba, Shoretel and many other through SIP or other trunks.
Views: 26109 Bob Langys
Bitcally: the click to dial Chrome Extension for your Asterisk Call Center
 
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Download Bitcally: https://chrome.google.com/webstore/detail/bitcally/pckcnakclkpcbbmkogfkcilplchlabhb?hl=en Bitcally is the click to dial Chrome extension for your xCALLY Asterisk Call Center. Bitcally is simple and fast to use: With a simple click, it automatically starts the call for your Agents. The phone numbers can be the Contacts inside your CRM, Ticketing system or any kind of Web page your call center agents manage. The installation is really easy: just search Bitcally in the Chrome Web Store and add it as a Chrome Extension. Done? Well, now you are ready to use Bitcally with Zendesk, SugarCRM, Salesforce or your custom CRM application. Let's see how! Just select the phone number that you want to call, click with the right mouse button on Dial and Bitcally will show you the number on its phone keypad. Finally click on the CALL button and the call will immediately start through the xCALLY Windows Phone bar. If you prefer, you can use other SIP accounts: just edit the Bitcally settings, defining the xCALLY server URL and the xCALLY Agent Username. You can also insert a prefix that will be automatically added to each dialed extension. Fast, Easy, Bitcally. Enjoy!
Views: 1612 xenialab xcally
Home Automation with Asterisk
 
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There are several options to integrate Home Automation with Asterisk. AGI and AMI is there and could be used. And why not in future to have a chan_homeautomation. I would present some ideas and demo (dangerous) to show how to integrate easy Asterisk in our home to control some feature and make affordable for people with physical limitations for example.
High Availabilty / HA Asterisk in 5 minutes
 
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Detailed demo of installation of the 5 minute High Availability (HA) PBX. The scripts have been updated to work with Cloud options as well!
Views: 1930 L Bergey
Asterisk Tutorial 21 - Introduction to Call Queueing [english]
 
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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 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: 8327 pascom GmbH & Co. KG
FreePBX 101 - Part 10 - Conferencing, Parking, and Paging
 
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In this video, I discuss some of the important ancillary services that you will want to take advantage of to make full use of your FreePBX server. These are: Conferencing - How to set up and use conference bridges. Parking - An explanation of the call parking system and how to use it. Paging and Intercom - the differences between paging and intercom as well as options and usage for both. 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 FreePBX 101 - Part 9: https://youtu.be/WxMrNuNtrGY Visit http://CrosstalkSolutions.com for FreePBX support, installation, and consulting. Legal mumbo jumbo: FreePBX® is a Registered Trademark of Sangoma Technologies.
Views: 24432 Crosstalk Solutions
Asterisk 123: Configuring Endpoints
 
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Learn more at http://asterisk.org Asterisk 123 is a technical introduction to the Asterisk Open Source project. The day-long lecture covers the basics of installing and configuring Asterisk in 4 separate session. This session covers SIP and IP Phone configuration. Using the DPMA (Digium Phone Module for Asterisk) along with Digium IP Phones Asterisk can auto-configure phones without an external provision mechanism.
Receptionist Asterisk Freepbx Elastix training Cisco SPA 504g, 508g 303, linksys 942, 962 Sip 500s
 
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The video was made to train a receptionist or a main person that would take a lot of calls coming in. visit our new page www.alldigitalphones.com or call for paid support 844-937-8647 toll free ok to text on this number. We can set up your onsite or hosted pbx server anywhere in the world. (remotely)
Views: 1271 theciscoguys
Asterisk Tutorial 22 - Queue Call Strategies [english]
 
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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 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: 6541 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
FreePBX simple config Простая настройка FreePBX asterisk 13
 
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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: 4934 rstayalive
How to use Asterisk Wildcard Character in Excel in Hindi | * Asterisk Wildcard in Excel
 
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Teachtoeach presents How to use Asterisk Wildcard Character in Excel in Hindi . In this video am trying to teachyou about * Asterisk wildcard character in excel. A wildcard is a special character that lets you "fuzzy" matching in your Excel formulas. A limited number of Excel worksheet functions can use wildcard characters to filter results. Here a guide to using them. An asterisk (*) - zero or more characters You can use Asterisk wildcard with below formula list :- AVERAGEIF, AVERAGEIFS COUNTIF, COUNTIFS SUMIF, SUMIFS VLOOKUP, HLOOKUP MATCH SEARCH It is a very difficult task to anyone. If you are really interested to learn this trick then this video never lets you down. On Youtube, only just few videos are available on this How to use Asterisk Wildcard Character in Excel in Hindi topic Thanks for watching. It is humble request to you. Please subscribe my YouTube channel by click on this link. www.youtube.com/teachtoeach Website : www.Teachtoeach.in #Asterisk #Wildcardcharacter #Asteriskwildcardinexcel #teachtoeach Some Popular Videos are :- 1) How to Count Words in a Cell in Excel in Hindi - https://www.youtube.com/watch?v=TJaPwUfbMvU 2) How to use Vlookup Hlookup Formula Together in Excel in Hindi - https://www.youtube.com/watch?v=KyundTAJOFU 3) How to use ISBlank Formula in Excel in Hindi - https://www.youtube.com/watch?v=cI5b7_cZr-k 4) How to use Indirect Custom Validation in Excel in Hindi - https://www.youtube.com/watch?v=uUndruZkccc 5) How to Enter Large Digit in Excel in Hindi - https://www.youtube.com/watch?v=rdAWbOtBLVQ 6) Magic Trick - How to Find any Position of Space or Character in Excel - https://www.youtube.com/watch?v=L27XO0SfY8Q 7) Motivational Video for Students | अगर CBSE UP Borad Exam में फेल हो गए तो इस वीडियो को देखें - https://www.youtube.com/watch?v=sCZNYSrQE20 8) What is Objects Methods Properties in Excel VBA in Hindi | Part 4 - https://www.youtube.com/watch?v=T5wnUUe3PyU 9) How to Create Class Intervals Auto Fill List in Excel in Hindi - https://www.youtube.com/watch?v=xtUHXo6mXZE 10) Excel VBA Editor Tutorial in Hindi - https://www.youtube.com/watch?v=9WjWRUSKwrk 11) Nested IF Formula in Excel in Hindi - https://www.youtube.com/watch?v=j-PJTLgRXNY 12) How to Record Macro in Excel in Hindi - https://www.youtube.com/watch?v=VJq3ejJMmmw 13) How to make Paytm QR code in Excel in Hindi - https://www.youtube.com/watch?v=LQ1srotDpQE 14) Excel VBA in Hindi | Excel VBA Tutorial in Hindi - https://www.youtube.com/watch?v=qYfvzzo2ZJg 15) IF Formula in Excel Hindi IF Function in excel - https://www.youtube.com/watch?v=OH0SLZUzBv4 16) How to Find Duplicates Using Vlookup Formula in Excel Hindi - https://www.youtube.com/watch?v=kcYyzIy4rF4 17) Reverse Name in Excel Hindi - https://www.youtube.com/watch?v=ujs29_MbmQw 18) How to split middle name in excel in hindi - https://www.youtube.com/watch?v=KFX2FQbcWFw 19) Fill series option in excel - https://www.youtube.com/watch?v=WQ_JBsdHP7o 20) How to split first name in excel - https://www.youtube.com/watch?v=r_saghnF6yI
Views: 2122 Teach To Each
Asterisk Freepbx Elastix training Cisco SPA 504g, 508g 303, linksys 942, 962 Sip 500S Side car
 
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www.theciscoguys.com 844-YES-VOIP This video will help you with your Cisco/ linksys phone.
Views: 9118 theciscoguys
Asterisk: Extension Mobility
 
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Пример реализации Extension Mobility в Asterisk Сайт автора: http://snakeproject.ru/
Views: 721 Mihail Kozlov
MS Outlook 2016 Integrated with Asterisk - Click to Dial
 
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MS Outlook 2016 Integrated with Asterisk - Click to Dial from Contacts. Demonstration.
Views: 1983 DVCOM Technology
Asterisk installation and configuration on CentOS7
 
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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: 8308 Fakhar Ali
How to log debug messages in Asterisk dial plan
 
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Asterisk dial plan development - short movie no. 10 - How to log debug messages in Asterisk dial plan This is the tenth video in series of videos that will show you the basic elements of Asterisk dial plan. In this video we will show you how to log debug messages in Asterisk dial plan . We used Visual Dialplan for Asterisk in this demo. Visual Dialplan is rapid application development platform for Asterisk dial plan development. Full functional, free of charge trial version of Visual Dialplan can be downloaded from www.apstel.com.
Views: 1956 Apstel LLC
Asterisk  PHP click to call
 
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Making calls using Asterisk and PHP web form, this call can be Internally or to external numbers, you can set any custom caller ID on each call leg
Views: 4119 Ambiorix Rodriguez
9-Planet IPX-330 IPX-2100  Advanced Options Asterisk | الخيارات المتقدمة
 
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Global SIP , Default Configuration الاعدادات الافتراضية واعدادات ربط التلفونات SIP من الانترنت *Subscribe To My Channel and Get More Great Tips * اشتركوا معنا ليصلكم كل جديد ومميز فقط اضغط على الرابط التالي: http://www.youtube.com/channel/UCc7Xi94x7aLsripC0bnRxww?sub_confirmation=1 تابعوني ايضا على : https://www.facebook.com/HamadAbsi https://www.twitter.com/hamadabsi https://www.instagram.com/hamad.alabsi
Aster CTI User Portal | Asterisk PBX
 
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For More Details Website : https://astercti.techextension.com Contact us for Free Installation help Skype: tech.extension Email: [email protected] Try for 15 days demo Free. Aster CTI Portal as a call center software improves efficiency of your phone communication by giving you more information and more options for each call you make or receive. Gives you an utterly new experience of effective phone communication right in your crm like sugarcrm , SuiteCRM, VtigerCRM, Odoo, Sales Force, Zoho, Pipedrive etc and drives your business processes to advanced standards.
Asterisk basic configuration: SIP Extensions
 
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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: 4343 FKIT
Asterisk AGI/AMI to ARI
 
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Learn more at http://asterisk.org Getting started with AGI,(Asterisk Gateway Interface), AMI (Asterisk Manager Interface) and ARI (Asterisk REST Interface) Matt Riddell, Author of Daily Asterisk News and CEO of VentureVoIP
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
[part 18] How to program FreePBX call routing from your phone(call forwarding, etc)
 
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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 You know how you can dial *49 or something on your phone at work and it will automatically start forwarding calls to everyone else when you leave? Or maybe your boss dials a number when he leaves for the day so that his calls go to your extension instead? Here is how that is done. You can set up FreePBX so that, when you dial a "feature code" into your phone, it changes call routing within FreePBX! 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: 6679 Louis Rossmann
Twilio  Elastic SIP trunk  and Asterisk
 
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Setting up Twilio SIP Elastic trunk and Asterisk for outbound calls. This is only for outbound calls and calls are authenticated based on the source IP address
Views: 1164 Ambiorix Rodriguez
FreePBX 13 Made Easy - Part 3 - Extensions, Phones, and the FreePBX Endpoint Manager
 
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In Part 3, we are going to go over how to set up extensions and phones using the FreePBX Endpoint Manager. 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: 54224 Crosstalk Solutions
XCALLY Shuttle advanced updates II
 
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Administration new modules: – SMS channel and Cally Square app – Zendesk Plus – Tiger Dial Web disposal + Planning peer – CM Customer journey history – Call Back delay – SIP trunk Fallback strategy – Analytics: Transfer Call Log – GUI security – Queue application: answer control – Dashboards: paginations and chart control – Salesforce call activity Phone bar new features: – TCP support – Login on pause – CTI phone bar pop-up control – Attended and Warm Transfer – Call when busy option
Views: 227 xenialab xcally
Phonebook XML App for Snom 760 - Asterisk Phonebook
 
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Hi, today I am going to demonstrate FonB XML App on Snom IP Phones. Our FonB XML App works on all Snom Phones. Allow me to show you its features: You can start FonB XML App using the Application Key or Applications Menu, depending upon your Snom Model. The FonB XML app provides Login Screen as a security checkpoint before the user contacts are accessed. It requires an Extension Number followed by the Pin Code generated by your FonB Administrator. The FonB XML App has 3 main Options which are List Phonebook, Call History, and Logout. List Phonebook enlists all of your contacts in your IP Phone. The App will show you first 25 contacts as a preview. You can search any contact here by pressing the search key. Let's search Max. The App will provide all contacts including Max letters. You can further filter this listing by pressing Filter Key and selecting particular type of contact, let's say, Google Contact. It will show you Google Contacts including letter Max. FonB XML App features Unified Call History that brings your call logs right in your IP Phone no matter from which device you made that call. It becomes very useful to use the official call logs you made being at home or travelling. All these calls will be available in your office IP Phone using FonB XML App. Call History can also be filtered to enlist calls of special direction. Pressing Filter key in Call History menu will provide users with an option to display Inbound, Outbound, Missed or All Calls at a time. FonB XML App ensures user's privacy by providing time based login sessions. It provides a balance between ease of use and security. You will be prompted to enter your credentials once. After that, the session will remain active for 8 hours meaning that you won't have to put your extension number and pin every time you need to use FonB XML App. After 8 hours, the previous session will expire and you will have to enter your credentials again to start a new 8 hours session. To further take care of your privacy, we have made Logout option available in case you're leaving your office early someday and don't want anyone to access your contacts. In that case, a default 8 hours session is ended the moment you logout. We differentiate and handle your different contact lists exactly as we do in FonB Web Portal. The FonB XML App can clearly differentiate between your Internal extensions, Highrise Contacts, Google Contacts, and My Contacts. It will also handle your Mobile Contacts once FonB Mobile license is available and installed on your FonB Server as an addon. The FonB XML App is developed to enhance your productivity by simplifying your telephony related tasks. The best thing is that it's free for use and Open Source. We highly recommend to download and use it now to have the endless benefits.
Views: 1595 AptusTel
Asterisk Ultra Cell 2.0 Motocross Knee Braces
 
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Introducing the new Ultra Cell 2.0. With continued ultimate support as in the current Ultra Cell, Asterisk has trimmed down the profile of the liners and made getting in and out of the brace very easy with 4 snap-in strap liner attachments, also allowing independent adjustment for the best comfort and support. Although many of Asterisk customers have been extremely satisfied with the Ultra Cell with the BOA system, Asterisk felt it was necessary to have a lower profile model with a quick release system available as an option." Shop Now | Asterisk Ultra Cell 2.0 Knee Braces https://www.rockymountainatvmc.com/Sales/923/Asterisk-Knee-Braces?utm_source=youtube&utm_medium=asteriskultracell2.0kneebraces&utm_campaign=youtubevideos&utm_term=description&utm_content=youtubevideos Click Here To Subscribe To Our Channel! http://www.youtube.com/channel/UCQSXoYEPBXNPSx1hWbO2WRQ?sub_confirmation=1 Follow Us On Social Media! FACEBOOK: https://www.facebook.com/rmatvmc?utm_source=Youtube&utm_medium=default&utm_content=DescriptionFB&utm_campaign=Youtubevideos INSTAGRAM: https://www.instagram.com/rmatvmc/?utm_source=Youtube&utm_medium=default&utm_content=DescriptionIG&utm_campaign=Youtubevideos TWITTER: https://twitter.com/RMatvmc?utm_source=Youtube&utm_medium=default&utm_content=DescriptionTW&utm_campaign=Youtubevideos
Cisco SPA502G con Asterisk
 
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Ejemplo de configuración de teléfonos IP Cisco SPA502G, funcionando con una centralita Elastix 2.4
Views: 2154 Turman Dreams
Installing Asterisk on Ubuntu 10.04 Server
 
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Installing Asterisk on Ubuntu 10.04 Server Link to Script http://randystech.com/index.php?option=com_content&view=article&id=134:ubuntu-1004-asterisk-install-script&catid=4:linux-server&Itemid=8 Check out RandysTech.com
Views: 13250 randystech
FreePBX VoIP Tutorial Part 7 - Configuring Google Voice
 
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Update: If you'd like to not have to put the + or the 1 when dialing out, change your dialing patterns to look like this: http://i.imgur.com/yW3mWoE.png Make sure you're using the exact same dialing patterns. This includes adding your 10-digit Google Voice number to the CallerID field on all 3 patterns. I will explain why this is important when adding more than one Google Voice account to your Asterisk server in a later video. Commands used in the video: amportal restart amportal stop amportal start It could take several minutes to restart the server. Just be patient. After making this video, I went into FreePBX, Admin, Module Admin, pressed Check Online, and then clicked the "Show only upgradeable." There was a Google Voice motif update I applied and I haven't had trouble with amportal restart since. I talk a little more about module updating in Part 10. Relevant links: PuTTY (SSH): http://www.chiark.greenend.org.uk/~sgtatham/putty/download.html 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: 44535 nirvgorilla
Alt codes ◄ ƒöΓ symbols öñ Laptop keyboards ♥
 
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Guys, there's a better way to write symbols, really. Just use http://tell.wtf/ and paint a symbol to find it. https://chrome.google.com/webstore/detail/tellwtf/ageiikgnhpnpcnaifeaikpphhgealfia - Chrome plugin for Tell.WTF Also, to those whom this video guide doesn't help ⇨ http://fsymbols.com/keyboard/windows/layouts/ is even better than Alt codes. http://fsymbols.com/keyboard/windows/alt-codes/laptop/ In this video I will show you how to type symbols from PC-type laptop keyboards by using alt codes. First thing you have to do is to find a numeric keypad on your laptop's keyboard. It consists of "7", "8", "9", "u", "i", "o", "j", "k", "l" and "m" buttons. You can see some tiny numbers on those keys. I will be calling them Ten-Key later. Now let's enable Num Lock. On most laptops, Num Lock can be enabled, or disabled by NumLk key , or by a combination of FN (the function key) and ScrLk (screen lock key). When you'll enable Num Lock, some corresponding indicator will light up. Choose input field where you want to write symbols. Now when you'll hit those ten-key keys - you'll be entering numbers. That's good. Just for example let's take per mile ‰ sign. It's alt code is 0137. I remind you that Num Lock has to be ON. To input per mile press and hold down Alt key. Now use the Ten-Key to input per mile symbol's alt code. Type 0 1 3 7. Let go of all the keys. If everything went okay, you should have inputted the per mile symbol. If not then try typing alt code while simultaneously holding Alt AND FN keys. That may work. So thanks, for watching if you want to find out more, or if you have any questions, suggestions then go to my website: http://fsymbols.com/. By the way, please rate this video.
Views: 736671 Ihor Menshykov
Asterisk Live - Allison Smith
 
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Learn more at http://www.asterisk.org and http://www.theivrvoice.com/ Allison Smith, "The Voice of Asterisk" joins this week's Asterisk Live show for a live interview from ITEXPO in Miami. Allison is a professional voice actor who provides voice audio prompts for business IVRs and menus. Because Allison recorded the default prompts that come with Asterisk she is able to provide custom prompts that match, giving a professional sound when customers call in to a business. Allison speaks about her experience with Astricon having been a member of the community for many years along with what it was like to attend the very first Astricon. In 2014, Asterisk is a professional enterprise-grade solution for businesses around the world and Allison shares how she has seen the community change over the years. With the release of Asterisk 12 and the Asterisk REST Interface a new excitement has infused the community.
FreePBX 101 - Part 9 - How to Forward Calls to a Cell Phone
 
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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: 31257 Crosstalk Solutions
Configure your Twilio Elastic SIP Trunking with FreePBX - Part I: Placing outbound calls
 
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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: 11996 Twilio
How to do a conference call?
 
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Video Transcription How to Do a Conference Call? I'll demonstrate how to make a conference call using Asterisk 'Meet me' application. You can have several party conference up to 6 different parties or even more. I will Call Henry first at his extension 7575, and I answer the call here, and then I try to call Max. I answer the IP Phone CallBack and then I answer on behalf of Max. Now, I have two active calls:- The first one is between me and Henry. and the second one is between me and Max. I start the conference, by clicking on the conference button for all active calls that I would like to engage in the conference. so I click Henry's Conference Button and Max's Conference Button and here we go, I have established a conference call between Max and Henry. As you can see, the previously two active call panels for Henry and Max has joined to one active call that has one panel showing one admin button of the three calling options: Mobile phone IP Phone and WebPhone. Notice that the IP Phone Button is appearing in green since it is the device I am using to connect to this Conference - Meetme Room. I can also do a Call Switch to a mobile phone or a Webphone....
Views: 3143 AptusTel
Asterisk BLF y Cisco SCCP
 
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Fuente: https://www.spreaker.com/user/xavi74/asterisk-blf-y-cisco-sccp
Views: 123 xavifermas
Asterisk BLF with Cisco IP Phone
 
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Video demonstrating busy lamp field functionality with Asterisk PBX and Cisco SPA504G IP phones. Including phone state indication and call pickup. By popular demand I have created a rough video tutorial on how to get this working. http://www.youtube.com/watch?v=mQlaxu1sS8E
Views: 13743 agoodm
How to Set Up Trunks and Outbound Routes in CompletePBX
 
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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 Click Dialer - Click to Call or Dial Any Number on your PC
 
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http://clickdialing.com/ Click Dialer - Click to Call For quick calling of phone numbers on your PC screen, including CRM's, email and the web! Agents can simply click a button to record conversations. Pre-typed scripts will pop up to assist during the phonecall. You receive Simultaneous Unlimited Calls on Unlimited Lines, simply Click a button to leave your pre-recorded message & instantly move to another call. Supervisory Features include silently monitoring, recording, and whisper coaching your agents. Once you click to call, you can transfer or conference calls to any number in the United States or Canada. Finally, display any 10 digit Caller ID you like. Sign up for a free trial today, no credit card required!
Cisco on Asterisk Training.wmv
 
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User training on how to operate the Cisco 509g and the Cisco 525g telephone with Elastix / Asterisk. Provided by Bob Langys and Medlin Communications in the Chicago area.
Views: 10389 Bob Langys
Sip Phone Extension ( Web dialer )
 
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It is cloud based Voip Phone from TechExtension have features like click to call, Call pop up, Call tracking, Call analytic Reports and have integration with CRM, ERP,, Support Portal and other web portals.
Dial Plan Editor Tutorial | Jive Communications
 
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This video shows you how to use Jive's dial plan editor to setup your auto-attendant and call flows. The Jive Communications' administration portal makes managing your hosted PBX quick and simple. Dial Plan Editor: Jive Communications Jive Core: Video Tutorial Site: http://www.getjive.com
Views: 37594 Jive Communications
Asterisk Jabber Notifications - http://geekhut.org
 
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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: 1374 Gregory
DTMF Issue - voice-class sip dtmf-relay force rtp-nte
 
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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: 2501 W00DY1848
[part 17] How to set up an IVR in FreePBX, press 1 to learn how!
 
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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 IVRs are those annoying automated pieces of shit we curse at when we call customer support. Here you will learn how to set one up that other people will likely curse at. 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: 17908 Louis Rossmann

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