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[part 10] Setting up SIP trunk on your FreePBX system so it can talk to the phone company
 
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👉 DISCORD chat server: https://discord.gg/NWuBUxC 👉 Rossmann Repair Group Inc is a participant in the Amazon Services LLC Associates Program, an affiliate advertising program designed to provide a means for sites to earn advertising fees by advertising and linking to amazon.com ✖ Buying on eBay? Support us while you shop! https://www.rossmanngroup.com/ebay Here we go over setting up your PBX box with your "phone company." We decided to use Voicepulse as our "phone company", aka SIP trunk services provider. We will be configuring the PBX to use the Voicepulse trunk we configured in an earlier video. For information on having us set up your phone system, check out our website: this is something we can do for you! http://www.rossmanngroup.com/business-phone-system-solutions/
Views: 41920 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: 1436 Ambiorix Rodriguez
Comparing Performance of Chan SIP and pjsip
 
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Since the Asterisk project launched the latest sip channel “chan_pjsip”, there were very few publications showing the performance gains or even losses of the new channel. In this presentation, we are going to use SIPP to measure the SIP performance of both channels for the latest versions of Asterisk. Below a list of topics to be presented. • Main differences between channels from practical perspective • Testing methodology and failure criteria • SIP registration performance • SIP performance for calls to the server (echo) • SIP performance for calls between UAC and UAS • Tips on how to increase SIP performance and registration performance
Testing with SIPP - AstriCon 2014
 
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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.
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: 6868 pascom GmbH & Co. KG
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
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: 2299 L Bergey
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.
11-Connect mobile to IPX via SIP Asterisk | كيفية ربط جوالك مع السنترال الاي بي
 
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how to connect your mobile to IPX via sip كيفية ربط جوالك مع السنترال الاي بي *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
Asterisk on Highly Available Scalable Middleware
 
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How to achieve high-availablity and/or scalablity is one key point when introducing asterisk into relatively large enterprise environment. There already exist several options to achieve this goal. In this session, we introduce a "new" option which uses "NFV compliant" high available/scalable middle war
Bitrix24  Total Telephony (August 2018)
 
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Bitrix24 Telephony includes the very latest phone system technology but should you rent a new number from Bitrix or integrate your existing number and PBX. We show you the options and how to set it up. We will cover: 1. Setting up a new line and connecting a number or picking a new on through Bitrix24 2. Routing, recording and permission settings 3. Using USB phones, SIP connectors and API integration
Views: 1403 intreface
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: 212701 Alta3 Research, Inc.
Mikrotik VoIP SIP Server Port Redirect rules setup
 
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Mikrotik VoIP SIP Server Port Redirect rules setup
Views: 28770 Tania Sultana
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.
Config Edit Installation For Raspberry-Asterisk PBX GUI - AREDN N2MH MeshPhone
 
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Short video on how to install config edit on a Raspberry-Asterisk PBX system that is version 14.X.X.XX. This options was dropped as a standard item between version 13 to version 14.
Views: 298 Commsprepper
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: 8780 Fakhar Ali
Outbound calling in FreePBX
 
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http://sysadminman.net - rent a virtual hosted FreePBX server for only £20/month
Views: 46708 SysAdminMan
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: 25389 Crosstalk Solutions
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
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: 8855 pascom GmbH & Co. KG
voip protocol :: Session Initiation Protocol SIP Overview, rfc 3261, wireshark sip tutorial New 2014
 
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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: 22513 Zariga Tongy
Developing an IVR Payment System - AstriCon 2014
 
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Presentation will be based on an IVR payment architecture i just completed for one of the leading Payment Switching Company in Nigeria. Session will cover: 1. Introduction to Payment Systems. Various payment solutions will be discussed. 2. Why IVR Payment Channel. Discussing why people will prefer IVR payment as opposed to other payment channels. 3. Challenges an IVR payment systems might face, and how to resolve such challenges (From Business and Security Angle). 4. Requirements to develop & deploy an IVR payment with Asterisk for production (Number of Concurrent users and how to scale will be points of discussion). 5. Architectural overview (diagram) and explanation with one of the recently completed IVR payment solution by CODEC SYSTEM. (The class will dial a toll free line, prepared for this presentation, to have a feel of a running DEMO IVR payment system.) 6. Walking through the IVR payment MENU, and explaining some deep security concerns, and how it was handled with the MENU options. 7. The IVR Payment Solution architecture itself. Will discuss what and what Hardware was used, Linux Distro (with some implementation considerations, E1/T1 lines, E1 gateways deployed, SIP trunk considerations and challenges, etc 8. Some sample code snippet with the DEMO system showing how we utilize Asterisk functions, applications, AGI, func-odbc especially, and with API (majorly in JAVA) to achieve the IVR payment system. 9. What will the future looks like for this project. i.e Where do we go from here. Will discuss what we plan to do in the nearest future with the IVR payment system. How we intend to do? Add TTS, Add Voice recognition engines, custom SIP mobile app, moving IVR to the cloud completely, etc. 10. Will also briefly discuss similar ongoing project (with IVR system) as a third level users Registration, Verifications, Authentication, and Authorisations with Bank online payments and transactions..
Flowroute SIP Trunk Setup on FreePBX
 
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This is a how-to video for setting up a Flowroute SIP trunk on FreePBX. FreePBX version 2.11 running Asterisk 11. To contact Chris, please visit http://CrosstalkSolutions.com. Legal mumbo jumbo: FreePBX® is a Registered Trademark of Sangoma Technologies.
Views: 32303 Crosstalk Solutions
PJSIP: Tuning for Performance
 
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Learn how to tune the Asterisk PJSIP channel driver for a high volume environment. Includes discussions about, and examples of configuring real-time database access, the use of caches and other configure options and distribution of workload
FreeSwitch from SIP to WebRTC
 
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Implementing the technological changes from images to audio and video and beyond from a FreeSWITCH perspective. When FreeSWITCH started, 12 years ago, everyone was excited to get 8 kilohertz ulaw and G.729 from point A to point B. Over the last decade, expectations have grown to include 1080p Video, High-Definition audio, texting and more. Two founding members of the FreeSWITCH team will explain the communication platform and how it can be used in a variety of ways in combination with Asterisk and other open source multimedia applications to form a complete solution.
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: 2745 W00DY1848
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
FreePBX Account Configuration, and Test in ZOIPER
 
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A quick walkthrough on what fields must be filled out.
Views: 17913 Aaron Hurst
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: 26176 Crosstalk Solutions
Asterisk BLF y Cisco SCCP
 
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Fuente: https://www.spreaker.com/user/xavi74/asterisk-blf-y-cisco-sccp
Views: 132 xavifermas
Asterisk Tutorial 18 - Voicemail Greetings [english]
 
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"Sorry we're not at our desks right now, leave us a message and we'll get back to you" Having setup our dialplans to send an incoming call to our voicemail boxes which we set up a few episodes ago, it's now time to take a look at how to setup new greetings for when you are unavailable, busy etc, why you should differentiate between them & how to configure our dialplans to playback our new voicemail greetings. In order to do this we will show you how to use the Asterisk Variable DIALSTATUS in combination with the Application VoiceMail options within your Dialplans. 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: 7537 pascom GmbH & Co. KG
LEARN | New product Elastic SIP Trunking - Annie Benitez Pelaez & Jonas Borjesson (Twilio)
 
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Register to attend SIGNAL 2016: http://bit.ly/1Rr3C70
Views: 2383 Twilio
cisco phone spa 303 with Asterisk Trixbox
 
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this video show basic steps needed to setup that sip phone with asterisk trixbox
Views: 1341 Seldom Tutorials
[part 15] Configuring Cisco SPA525G VoIP phones as extensions with FreePBX
 
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👉 DISCORD chat server: https://discord.gg/NWuBUxC 👉 Rossmann Repair Group Inc is a participant in the Amazon Services LLC Associates Program, an affiliate advertising program designed to provide a means for sites to earn advertising fees by advertising and linking to amazon.com ✖ Buying on eBay? Support us while you shop! https://www.rossmanngroup.com/ebay Here we will configure our phones as extensions within FreePBX so they ring when they are called. 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: 17968 Louis Rossmann
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: 26170 Bob Langys
ASTERISK PBX
 
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ASTERISK PBX TUTORIAL GIVING THE BASICS OF OPERATION
Views: 4525 PhoneComPBX
Is your telephone company a DIDX member? If yes, your number can ring to Skype or SIP.
 
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DIDX makes available Ring to Options of SIP IAX2 (Asterisk) and Skype. Yes, DIDX members can route your wholesale DIDs to your customer's Skype ID at no extra charge. Please remember that DIDXchange at www.didx.net is a wholesale DID marketplace so there is a minimum number of purchased DIDs required in your account at all times of 50. Meeting this requirement will mean you avoid the monthly $50 minimum quantity fee.
Views: 768 didexchange
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: 10406 Bob Langys
VoIP with Obihai OBi200 and Grandstream!
 
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What is VoIP? Voice over Internet Protocol takes your voice calls off the POTS (plain old telephone service) and puts the calls on the Internet. VoIP calls can withstand about 150ms of latency before call problems start occuring. Want to dip your toe into VoIP? Buy one of the Obihai OBi200 devices and follow along! We'll have a full integration with our Grandstream UCM6202! More Info: Contact us for network consulting and best practices deployment today! We support all Obihai, Polycom, Plantronics, Ubiquiti Networks, Grandstream, MikroTik, Extreme, Palo Alto, and more! My Amazon Link: https://www.amazon.com/shop/williehowe H5 Mailing List: http://h5llc.com H5 Discord: https://discord.gg/3xyT8NX Netool: https://netool.io use code WILLIEHOWE to save at least 10%! Digital Ocean Referral Link: https://m.do.co/c/39aaf717223f SIP.US: http://h5.sip.us Consulting Contact: https://h5technology.com Support Agreement: https://h5technology.com/support Support my channel and keep the lab growing! Come back for the next video! Twitter - @WillieHowe Instagram - @howex5 SUBSCRIBE! THUMBS-UP! Comment and Share!
Views: 13146 Willie Howe
SIP Trunking Configuration in Avaya Communication Manager
 
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The video will detail the SIP Trunking configuration in Avaya Communication Manager. Produced by Chandrakandan, Chandragiri.
Views: 34041 Avaya Mentor
SIP Trunks on ShoreTel using the Ingate Siparator
 
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Not a SIP trunk Tutorial, but a quick look at the Ingate Siparator -quick start- tool for brining Sip Trunks up on a ShoreTel iPBX
Views: 6515 DrVoIP
How to register SIP in fresh folio on NSE NMF?
 
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NSE Mutual Fund Platform envisions an inclusive growth with wider reach and deeper penetration by leveraging the exchange infrastructure in conjunction with the best practices of mutual funds industry. It has been designed for the mutual fund distributors keeping their business and operational challenges and opportunities in perspective and offering compelling value proposition by adopting the best industry features and beyond. It is its endeavour to deliver a comprehensive solution with unique user benefits that will address the evolving needs of the MF industry and empanel the MF Distributors to use the stock exchange infrastructure albeit with lesser financial and compliance requirements.
Kamailio
 
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Working with a small, medium or large VoIP deployment? In any of the cases, this talk is for you. The subscriber base is no longer relevant, any VoIP system out there must be able to cope with DoS and DDoS attacks. Realtime communications went way beyond just voice communications, integration with other services is the key to retain as well as attract customers. Kamailio has a very good reputation for its scalability to do SIP routing and the excellent interworking with Asterisk or other VoIP systems. This talk is presenting the latest options offered by Kamailio to secure and scale your deployment, from REST API-based solutions, to scripting the routing as well as integration with third party services using various embedded languages, such as Lua, Python or JavaScript, allowing everyone to choose the preferred tool to build the best solutions that meet the nowadays market demands, such as SIP load balancers, least cost routing engines, SIP-layer firewalls, VoIP residential or carrier platforms.
VoIP Features With Marc Spehalski | Call Parking on FreePBX
 
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Everyone remembers the old days before VoIP when PSTN lines ruled the business telecommunication world. PSTN lines were, let's just say, a lot less advantageous than what we have today. One thing we can all remember is putting people on hold. We can still do this today with VoIP, but now there is something more advanced called Call Parking. This has the same concept of placing a call on hold but gives you more options. In this video, Marc Spehalski will not only talk about the differences between Call parking and putting someone on hold, but he will also demonstrate Call Parking in action using FreePBX! VoIP Supply has Everything You Need For VoIP! https://www.voipsupply.com/ Sangoma s400 https://www.voipsupply.com/sangoma-s400?utm_source=YouTube&utm_medium=Video&utm_campaign=Sangoma%20S400 Sangoma s700 https://www.voipsupply.com/sangoma-s700-sip-phone?utm_source=YouTube&utm_medium=Video&utm_campaign=Sangoma%20S700 FreePBX https://www.voipsupply.com/manufacturer/freepbx?utm_source=YouTube&utm_medium=Video&utm_campaign=FreePBX Get the latest news, information, and tips on VoIP daily from The VoIP Insider Blog! https://www.voipsupply.com/blog/voip-insider/ Video By: Stephen Lopian
Views: 656 VoIP Supply
VoIP Supply Review | Digium A-Series IP Phones
 
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Digium has released a brand new series of SIP phones to be used with any Asterisk-based phone system! Join Marc Spehalski as he goes under the hood and reveals the features and functionality of all four phones! VoIP Supply has Everything You Need For VoIP! https://www.voipsupply.com/?utm_source=YouTube&utm_medium=Video&utm_campaign=VoIP%20Supply Digium A-Series Phones https://www.voipsupply.com/manufacturer/digium/digium-phones/digium-a-series-asterisk-phones?utm_source=YouTube&utm_medium=Video&utm_campaign=Digium%20A%20Series Digium Switchvox https://www.voipsupply.com/manufacturer/switchvox/smb?utm_source=YouTube&utm_medium=Video&utm_campaign=Digium%20Switchvox%20Virtualization ALL Digium Products https://www.voipsupply.com/manufacturer/switchvox?utm_source=YouTube&utm_medium=Video&utm_campaign=Digium%20Switchvox%20Virtualization Follow VoIP Supply On Social Media! VoIP Supply Twitter https://twitter.com/VoIPSupply VoIP Supply Facebook https://www.facebook.com/voipsupply/ VoIP Supply LinkedIn https://www.linkedin.com/company/108762/ VoIP Supply Google+ https://plus.google.com/u/0/+voipsupply-phones Hi this is Mark back at the lab here at VOIP Supply and today we're here to talk about the A-Series phones by Digium which is Digium's latest affordable option for those who have asterisk-based phone systems. There's four of them and they all have a lot of similarities but there's a couple of interesting differences that we're going to talk about. Let's look at the first two phones in the offering, the A20 and the A22, let's take a look. So right off the bat, you'll notice that the Digium A20 and A22 look identical, there is one main difference between the two in that the A22 has a gigabit port where the A20 has just a 10-100. If you're going to be connecting your PC to a phone you're probably going to want the gigabit if the phone is going to be standalone, the 10-100 is pretty much all you need for any kind of voice communication. With most VOIP phones you have three-way conferencing and with the A20 and A22, you have two SIP registrations, which are marked here by line one and line two which really means SIP registration one and SIP registration two. We have a 2.8-inch color screen with soft keys on both and a directional pad which you'll find on most phones for navigating the menu, other than that all your basic buttons, mute, voicemail button, headset, it does have electronic hook switch, Mute, redial and speakerphone. Both phones have identical features. Taking a look at the back of the A20 it is identical to the A22 so we start with our power port, this is going to be a five volt DC, we have two switches 10-100 ports on the A20, it is gigabit on the A22. POE powered, we have our handset port and our headset port. The middle range of the Digium A series is the A25 which also gives you gigabit ports just like the A22 but there's a couple of extra features here. We see prominently the BLF screen, there are six keys here and you can have up to 30 contacts and you can also have four SIP registrations. Couple other important remarks as far as the A-Series goes, it is provisioned with the HP Options 66, TFTP and uses XML configuration. The phone does, of course, have a web interface so you can make configuration changes and make manual provisioning if you needed to. Moving on to the top of the line of the Digium A series phone, the A30 has a 4.3 inch color screen which is quite a bit larger than the 2.8 inch on the other models, also you can have up to six SIP registrations and does sport a couple of extra feature keys, just makes it a bit more convenient when you want a specific function, other than that it's pretty much similar in function to all of its other predecessors in the series. It's also worth mentioning just like the A22 and A25 the A30 does also have Gigabit speed ports on the back. That's the Digium A series phones, they're very attractive, professional looking phones that are built again specifically for Asterisk. If you're in the market for phones just like these and you have an Asterisk phone system you can pick these up at VOIPSupply.Com, once again I'm Mark here in the lab at VOIP Supply, and remember, all we are is VOIP in the wind. Video by: Stephen Lopian
Views: 555 VoIP Supply
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.
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On today's how, we take a closer look at the new Skype for Business services as part of today’s Office 365 announcements. Cloud PBX, PSTN Calling and PSTN Conferencing bring phone calling and meeting dial-in capability to Skype for Business. We demonstrate how easy it is to get numbers and configure calling capabilities for your users with Cloud PBX. Plus, we set up dial-in access for Skype meetings and present the options you have to integrate your current infrastructure with Cloud PBX.
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Digium IP Phones with Asterisk
 
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Learn more at http://asterisk.org Digium IP Phones are the only phones specifically designed to work with Asterisk and Asterisk-based phones systems. Come and learn about the unique custom integration options that are available with Digium phones that no other phones allow. Malcolm Davenport,
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In this video, we take a quick look into the most obvious VoIP options in the market and why we believe in CompanyTone that 3CX is currently the best one to go with.
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