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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: 3296 FKIT
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: 1430 L Bergey
Asterisk SIP Server Settings
 
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For Faran....
Views: 22480 Maaz Bin Mahmood
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
SIP Troubleshooting for Beginners - Outgoing Call Trace Review
 
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This video is a breakdown of a wireshark trace that captured an outgoing call between a PBX and an ITSP (SIP Provider). It will be one part of a series of videos designed to give a better understanding of the SIP protocol.
Views: 119379 Terrell Boyer
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: 733 Ambiorix Rodriguez
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: 28741 Crosstalk Solutions
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: 1420 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.
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: 7473 Fakhar Ali
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
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: 2077 W00DY1848
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: 21530 Zariga Tongy
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
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
[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: 14789 Louis Rossmann
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: 10160 Twilio
Converting Avaya 9608 h323 to SIP
 
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How to convert 9608 to SIP
Views: 6390 Eric S
The Ultimate SIP Tutorial
 
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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: 48053 Terrell Boyer
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: 8455 theciscoguys
Cisco SPA502G con Asterisk
 
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Ejemplo de configuración de teléfonos IP Cisco SPA502G, funcionando con una centralita Elastix 2.4
Views: 1903 Turman Dreams
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
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: 1155 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 on our turnkey business phones systems and UC solutions, visit our website: https://www.pascom.net/en/
Views: 5954 pascom GmbH & Co. KG
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: 931 Seldom Tutorials
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.
How to setup DHCP for IP Phones to receive phone configuration
 
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Author, teacher, and talk show host Robert McMillen shows you how to how to setup DHCP for IP, or VOIP, Phones to receive phone configuration from the phone switch. This is done by using predefined options in DHCP manager.
Views: 9567 Robert McMillen
Google Voice to SIP Gateway | GVsip.com - Setup Tutorial
 
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GVsip allows you to use your Google Voice account with any VoIP device. Register up to 10 VoIP devices to your Google Voice account to make and receive unlimited free calls to the United States and Canada. Signup today to get your unlimited 30 day risk free trial. If you decide you like the service there is only a one time upfront fee to keep your acount active forever.
Views: 9317 Vestalink VoIP
FreePBX 14 - Setup & Configuration Part 2 - EP-220
 
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Part Two of the FreePBX14 VOIP Server Setup Thanks to "Music: Little Idea - Bensound.com" Thanks for watching! I hope you all enjoy... Facebook: https://www.facebook.com/UnkyjoesPlayhouse/ Twitter :https://twitter.com/unkyjosplyhouse Email:[email protected] For PayPal or Patreon donations to Unkyjoe's Playhouse, please visit the "About" section on my channel. All cash donations are directly put back into Unkyjoe's Playhouse channel projects. I cannot respond to all emails, but give it a go! *PLEASE NOTE* I do not respond to YouTube or Google+ private messages. Please contact me via the official Facebook page or via my email address to get in touch.
Views: 357 Unkyjoe's Playhouse
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
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.
FreePBX VoIP Tutorial Part 2 - Gmail and Google Voice Setup
 
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Note: If you have SipDroid or any other VoIP SIP app installed on your Android phone, it would be wise to uninstall it will likely get in the way of what we're doing. In this section, I talk about disabling Google Chat (GTalk) in Gmail and the Talk app in Android. These are both usually on by default. Note: With these settings, you will still be able to text people using either the Google Voice Android App or the Google Voice client simultaneously. If you want to record your phone calls, enable it in CSipSimple under Settings, Call Options, Auto Record Calls. Files will be in .wav format in your phone's internal memory under CSipSimple. Another option is to record directly on your server, but it's a little trickier to set up. This method might be ideal for those with phones that aren't capable of handing that kind of computation. Note: You don't HAVE to disable Two-factor Google Authentication. You can go here https://accounts.google.com/b/0/IssuedAuthSubTokens#accesscodes and set up a new password for the server. If I knew the exact instructions I'd give them but I don't, sorry. Relevant links: https://accounts.google.com/DisplayUnlockCaptcha 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: 38516 nirvgorilla
Twilio SIP Trunk - IPBRICK
 
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Know more about our solutions ➡ https://www.ipbrick.com Follow our social networks ➡ Facebook: https://www.facebook.com/IPBRICKSA | LinkedIn: https://www.linkedin.com/company/ipbrick-international | Twitter: https://twitter.com/IPBrick
Views: 1730 IPBRICK
SIP Training
 
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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: 202945 Alta3 Research, Inc.
Cisco SPA 504g 500S Asterisk Freepbx Tutorial Side Car Receptionist
 
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www,theciscoguys.com sip trunks $110 for 10 concurrent calls. 844-YES-VOIP
Views: 3909 theciscoguys
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: 25989 Bob Langys
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..
✅ Ultimate pfsense Router - Part 4 of 6 (Voip Setup)
 
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6 port intel nics router - passively cooled, AES-NI support for pfsense 2.5. Options for 1U rackmount or wallmount. SATA 2.5 support or Compact flash. Part 5 - will be released at 350 subs Part 6 - will be released at 400 subs ==================================================== Part 1 - Unboxing, Introduction to router hardware, look inside the actual router. Initial pfsense login, enabling cpu temperature sensors and crypto options. Part 2 - BIOS access, Console Cable, Serial to USB cable, Putty terminal setup. Checking power button/reset. Part 3 - configuring pfsense into a 5 port switch just like the Cisco Small Business routers. Fully configure pfsense bridge settings and firewall rules to make this happen. Part 4 - Voip Setup - Configure network, gateway, firewall, and traffic shaping wizard for a reliable voip setup. Part 5 - Backup and Restore to same hardware and different hardware. Part 6 - Installation. Prep a USB stick, boot options, and initial web ui setup. =================CREDITS==================== Tech Live Kevin MacLeod (incompetech.com) Licensed under Creative Commons: By Attribution 3.0 License http://creativecommons.org/licenses/by/3.0/
Views: 5102 Nick's Hardware
SIP Cluster (Engineering Day 2013)
 
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An overview of the architecture for the Genesys SIP Cluster.
Views: 448 GenesysDocs
AddPac IP Phone interworking Asterisk PBX
 
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VoIP, SIP, H.323, Asterisk
Views: 390 AddPacMarketing
asterisk with callerid
 
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asterisk enviando tag de callerid
Views: 491 manzurek
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,
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: 10344 Bob Langys
Elastix-SIPTRUNK
 
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En el siguiente video vemos de que forma configurar un SIPTRUNK en un servidor Asterisk de forma de cursar llamadas a la red de telefonía pública.
Views: 57641 netlineip
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: 288 VoIP Supply
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