Search results “Freepbx asterisk dial command options”
FreePBX 101 - Part 9 - How to Forward Calls to a Cell Phone
This is a short video, but a very important feature! How to use Misc. Destinations to forward calls to external phone numbers (mostly used for cell phones). FreePBX 101 - Part 1: https://www.youtube.com/watch?v=LsfqSnGZ3dI FreePBX 101 - Part 2: https://www.youtube.com/watch?v=xBny4hKCM3A FreePBX 101 - Part 3: https://www.youtube.com/watch?v=WgWovGKz5v4 FreePBX 101 - Part 4: https://youtu.be/uNlygMYvNlk FreePBX 101 - Part 5: https://youtu.be/aFrMecTpoyk FreePBX 101 - Part 6: https://youtu.be/EH5XrhtUiSo FreePBX 101 - Part 7: https://youtu.be/YXNTlA3kPWI FreePBX 101 - Part 8: https://youtu.be/8ht-26pBOko Visit http://CrosstalkSolutions.com for FreePBX support, installation, and consulting. Legal mumbo jumbo: FreePBX® is a Registered Trademark of Sangoma Technologies.
Views: 31726 Crosstalk Solutions
Asterisk Part 4  - Using "Contexts" and Dial Plans -- Part 1
In the fourth video of this 10 part series on Asterisk, I introduce your to the concept of contexts and dial plans. Due to Youtube's small video size, some of command line stuff is hard to see. You can view a 640X480 video on my website: www.HotButteredIT.com. They are currently free, but I'll have to keep an eye out on the bandwidth usage. Enjoy!
Views: 38512 Simons Tech
Outbound calling in FreePBX
http://sysadminman.net - rent a virtual hosted FreePBX server for only £20/month
Views: 46568 SysAdminMan
FreePBX VoIP Tutorial Part 7 - Configuring Google Voice
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: 44750 nirvgorilla
[part 17] How to set up an IVR in FreePBX, press 1 to learn how!
👉 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: 18354 Louis Rossmann
Asterisk 123: Configuring Endpoints
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.
FreePBX 101 v14 Part 19 - IVRs
FreePBX 101 v14 Part 19 - IVRs BEST PRICING on Sangoma and Yealink phones and equipment! Contact Crosstalk @ https://crosstalksolutions.com. Download IVR design PDF here: https://crosstalksolutions.com/freepbx/freepbx101-v14-ivr-diagram.pdf Crosstalk Store on Amazon - RECOMMENDED PRODUCTS: https://www.amazon.com/shop/crosstalksolutions Crosstalk Discord: https://discord.io/crosstalk Amazon Wish List: http://a.co/7dRXc67 Crosstalk Solutions offers best practice phone systems, network design and deployment, and UniFi Video camera systems. Visit https://CrosstalkSolutions.com for details. Crosstalk Solutions is an authorized Sangoma partner and reseller. Connect with Chris: Twitter: @CrosstalkSol LinkedIn: https://goo.gl/j2Ucgg YouTube: https://goo.gl/g4G58M
Views: 1080 Crosstalk Solutions
Asterisk basic configuration: SIP Trunk
This video features a SIP Trunk setup procedure for the IP PBX Asterisk on Linux environment.
Views: 4415 FKIT
Asterisk Tutorial 52 - Asterisk Call Detail Records Part 2 [english]
In part 2 of our Introducing Asterisk Call Detail Records, we take a more detailed look the CDR records. That means having a look at CDR fields as well as how to label a CDR record for billing or reference purposes. 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 free download • https://www.pascom.net/en/downloads/ ► Our Blog • https://www.pascom.net/en/blog/
Views: 3177 pascom GmbH & Co. KG
Comparing Performance of Chan SIP and pjsip
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
How to Capture and Debug SIP Packets from asterisk using tcpdump and Wireshark
Troubleshooting VoIP can be a daunting task. Luckily we can easily capture SIP packets in asterisk using tcpdump and analyze the call data results within Wireshark. Here are the tools we will be using in this tutorial: Putty / SSH Client -http://www.chiark.greenend.org.uk/~sgtatham/putty/download.html tcpdump - yum install tcpdump (depending on your OS) WinSCP - http://winscp.net/eng/download.php Wireshark - http://www.wireshark.org/download.html More on using tcpdump: http://www.jonathanmanning.com/2009/10/26/how-to-voip-sip-capture-with-tcpdump-on-linux/ -------- For Encrypted SIP Trunking, Global DID's and Hosted Phone System, check us out! nurango https://www.nurango.ca https://twitter.com/nurangotel
Views: 35807 nurango
CompletePBX 5.0 Software PBX Preview,  VoIP PBX Software, Asterisk PBX Software
Xorcom CompletePBX 5.0 is a a full software IP PBX system. Unlike previous CPBX version, which were based on FreePBX Asterisk GUI, version 5.0 and up is powered by Ombutel, which provides a fast and clean interface which is fast, secure and intuitive to use, and fully responsive to support administration in any screen size, including tablets and mobile phones. CompletePBX 5.0 software will be offered in two options as a software PBX and integrated with Xorcom hardware as stand alone phone systems. Coming September. CompletePBX Software will be available for download on our website: https://www.xorcom.com VoIP PBX Software by Xorcom.
How to setup Follow Me in Elastix
This tutorial will show you how to setup "Follow Me" using Elastix.
Views: 23712 synapseglobal
Configure a Softphone for your PBX or VoIP account
Using X-Lite and Bria as an example, we show the basic settings needed to connect your softphone with an Elastix Asterisk-based PBX, as well as an individual VoIP service.
Views: 5753 VoicePulse
IP04 QuickStart
http://alsacecom.fr/go/ip04 Le IP04 est une appliance Asterisk avec quatre ports FXO ou FXS. Il comprend un système d'exploitation Open Source pré-installé Linux avec les fonctions de proxy SIP/IAX2 et NAT. Il fournit ainsi une plateforme solide et compacte pour des communications traditionnelles sur le réseau téléphonique comme pour des communications Voix sur IP. L'IP04 s'adresse principalement aux TPE/PME avec une interface graphique facile d'utilisation, fournissant une solution à bas coût pour leurs besoins en communication. Avec l'IP04, une société avec des agences dans différents pays ou régions peut très facilement s'organiser pour rationaliser les communications à travers un système commun virtuel et à travers Internet limitant ainsi les dépenses téléphoniques.
Views: 1064 alsacecom
Установка Asterisk 14 на CentOS 7
В данном видео расскажем как установить последнюю версию Asterisk 14.6.0 на операционную систему CentOS 7. Следуя нашей инструкции, Вы без труда сможете собрать Asterisk из источников. Итак, поехали! Установка Asterisk 14 на CentOS 7: http://wiki.merionet.ru/ip-telephoniya/32/ustanovka-asterisk-14-na-centos-7/ Калькулятор инсталляции IP - АТС Asterisk: http://wiki.merionet.ru/asterisk-calculator/ Установка Centos 7 в Hyper-V: https://youtu.be/ZUr2TdUqT2s Документ по установке (команды): http://wiki.merionet.ru/ip-telephoniya/32/ustanovka-asterisk-14-na-centos-7/Asterisk_Installation_Script.txt
How to setup a basic outgoing  trunk in Elastix.
How to setup a basic outgoing trunk in Elastix.
Views: 103902 synapseglobal
Asterisk 10 on Ubuntu 10.04
How to install Asterisk on Ubuntu, specifically Asterisk 10 on Ubuntu LTS (10.04) - I recommend copying and pasting the commands directly from: http://linuxmoz.com/ubuntu-asterisk-10-install-guide/ Out tutorial will walk you through a source install of Asterisk 10 (the latest version) on Ubuntu LTS.
Views: 2386 linuxmoz
PJSIP: Tuning for Performance
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
Asterisk installation and configuration on CentOS7
Here is a simple and easy way to install Asterisk over CentOS 7 [[[[[[[[[[[[[[[[[[[[[[[[[[[[[[[[[[[[ Asterisk is an open source telephony switching and private branch exchange service for Linux and is completely free framework for creating programs and is subsidized by Digium. Asterisk can change a common pc into a emails server. Asterisk abilities IP PBX methods, VoIP gateways, conference web servers,customer solutions and is used by companies such as telecommunication, suppliers and for countries worldwide For this install I am using Asterisk 11.0.0 and will be compiling from source on CentOS 7. This tutorial should also work on Fedora and RHEL (Red Hat Enterprise Linux) systems with little or no modification. First type command su Then, you will want to be sure that your server OS is up to date. yum update -y Disable SELinux by changing “enforcing” to “disabled” in /etc/selinux/config. Use a text editor or copy and paste this command. sed -i s/SELINUX=enforcing/SELINUX=disabled/g /etc/selinux/config After you update and disable SELinux, you’ll need to reboot. reboot Next, you will want to resolve basic dependencies. (More information on Asterisk dependencies.) yum install -y make wget openssl-devel ncurses-devel newt-devel libxml2-devel kernel-devel gcc gcc-c++ sqlite-devel libuuid-devel Change into the /usr/src/ directory to store your source code. cd /usr/src/ Download the source tarballs. These commands will get the current release of DAHDI 2.6, libpri 1.8 and Asterisk 11. wget http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-current.tar.gz wget downloads.asterisk.org/pub/telephony/libpri/libpri-current.tar.gz wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11-current.tar.gz Extract the files from the tarballs. tar zxvf dahdi-linux-complete* tar zxvf libpri* tar zxvf asterisk* For the next set of commands it is important to follow the proper order: DAHDI first, then libpri, then Asterisk. Install DAHDI. cd /usr/src/dahdi-linux-complete* make && make install && make config Install libpri. cd /usr/src/libpri* make && make install Change to the Asterisk directory. cd /usr/src/asterisk* In the next step, running the “configure” script will vary depending on whether your system is 32-bit or 64-bit. (Watch the video for more details.) When the menu select command runs, select your options, then choose “Save and Exit” and the install will continue. Use this command if you are installing Asterisk on 32bit CentOS. ./configure && make menuselect && make && make install Use this command if you are installing Asterisk on 64bit CentOS. ./configure --libdir=/usr/lib64 && make menuselect && make && make install Optional: If you ran into errors you will want to clean the install directory before recompiling. make clean && make distclean Once you have an error-free install, copy the sample files from the configs subdirectory into /etc/asterisk. make samples Then add the Asterisk start script to the /etc/init.d/ directory make config Start DAHDI. service dahdi start Enable the Asterisk services at system boot time. chkconfig asterisk on Start Asterisk. service asterisk start Connect to the Asterisk CLI. asterisk -rvvv And now you have Asterisk 11 running on CentOS 7! If you’d like to continue configuring Asterisk you can check out this guide to setting up basic pbx functionality or leave a comment to share your thoughts below! You can also check out some of our training and certification options.
Views: 8455 Fakhar Ali
How to Configuring VoIP Phones in Cisco Packet Tracer
Please read the below command . DHCP server on RouterA (2811 router) : ip dhcp pool VOICE network default-router option 150 ip Configure the Call Manager Express telephony service on RouterA . telephony-service max-ephones 5 max-dn 5 ip source-address port 2000 auto assign 1 to 6 Configure the phone directory for IP Phone 1 ephone-dn 1 number 123 ephone 1 device-security-mode none type 7960 button 1:1
Views: 152 DvDnet
Configuring VoIP Phones in Cisco Packet Tracer
Do you want to know how to configure VoIP (Voice over IP) phone systems? Have you always wondered how to configure Telephony Services on a router to run VoIP on your networks? Have you wondered how to configure DHCP for your VoIP network? Take my Cisco CCNA iOS Administration Labs Course for $10.00 https://www.udemy.com/cisco-ccna-rs-200-125-ios-administration-labs/?couponCode=YOUTUBEPROMOTION1 Take my Subnetting - Become a subnet master Course for $10.00 (Best selling online subnetting course) https://www.udemy.com/subnetting-become-a-master/?couponCode=YOUTUBEPROMOTION01 DONT FORGET TO SUBSCRIBE AND LIKE THIS VIDEO! LIKE US ON FACEBOOK: https://www.facebook.com/learntechtraining/ For more Courses visit www.learntechtraining.com Buy the Cisco CCNA RS Complete Study Guide Here: https://goo.gl/8SpGcN Buy the Cisco CCNA Voice Study Guide Here: https://goo.gl/uy6qoU
Views: 87146 LearnTech Training
Setup Polycom VVX for VOIP service
Simple, step-by-step, video on how to setup the SIP (Voice Over IP) service on a Polycom VVX phone using CallCentric as the service provider. This video is geared for people who are new to the VOIP field. SIP is the dominate VOIP protocol in use today. A SIP Account is required in order to be able to make calls with your Polycom phone. This video demonstrates setup using the Call Centric SIP serivce. You can find them here: http://www.callcentric.com/?aid=233840 Things you will need: ------------------------ - Polycom VVX phone (201,300,310,311,401,400,411,500,511,600) - Callcentric account - wired Network connection - Polycom power supply adpater (unless your network provide Power Over Ehternet) You must begin with the Polycom phone either being new or factory defaulted. For example if you purchased the phone in a used condition, you may need to factory default it before you can enable the SIP service. There is a link in the video to another video which has these instructions. Follow the steps in the video to create a new extension in your CallCentric account Then, take the Authentication ID number (usually 1777xxxxxxx) and the password you created to setup the phone's SIP information. From the SETTINGS screen Go to ADVANCED Password is 456 (if this does not work, you phone is probably not factory defaulted) Go to Administration Settings Go to LINE CONFIGURATION Choose LINE 1 Enter the ADDRESS as the 1777xxxxx number Then go to LINE KEYS, set that to 2 or 3 (your prefrence) Then go to AUTHENTICATION - use login Creds = NO - DOMAIN = Callcentric.com - USER ID = (your 1777xxxxx number) - password = (you createdd in extension setup on Callcentric) Go To SIP PROTOCOL choose SERVER 1 Address = callcentric.com port: 5060 Register = yes Transport = UDP Only Then hit the back key until you get to a screen that says: Exit w/o save save config Resume setup choose SAVE CONFIG Phone should now attempt to register to the registrar for Callcentric. Try to make a call. If you do not have a CallCentric account, you can setup one up here http://www.callcentric.com/?aid=233840
Views: 8709 VoIP Tech
Enable Custom Web Access with Cisco Web Security Appliance (WSA)
This Security Chalk Talks video focuses on a new feature in our Web Security Appliance (WSA) AsyncOS 10.0 release - Referer Header. Aniket Arondekar, Manager, Technical Marketing, talks about how Cisco WSA can enable custom web access based on the value of the HTTP referer header. For example, customers can allow certain YouTube content while blocking YouTube as a whole. To learn more visit http://cs.co/6055BCmHS.
Views: 1069 Cisco
2U VoIP / GSM Gateway with PC on board unpacking
To purchase Mini Rack click here http://en.antrax.mobi/gsm-termination-business/ It already has 2 on-board GSM modules - Smart, small and well-built sub-rack - PC on board (with Linux Cent OS installed) - 3 universal slots for your Sim-boards or GSM-boards - Secure VPN connection Description Our ANTRAX 2U solution was specially created for those, who values the simplicity and compactness. It already has on-board PC, which helps the customer to avoid the problems connected with obtaining and placing the separate one. Having 2 on-board GSM modules, it still provides the users with three additional universal slots, which can be filled by 2-6 GSM modules or 1-60 SIM cards. Main components - Sub Rack with 2 on-board GSM modules and 3 universal slots (could be taken by SIM boards or GSM boards) - x1 on-board PC - up to 3 additional SIM boards (20 sim cards on each) - up to 3 additional GSM boards (2 GSM channels on each) Physical Parameters - Height: 8,9 cm - Width: 48,3 cm - Depth: 28,5 cm - Weight: 2,75 kg - Mounting: 19" 2U Rack - Power Supply: 90--264VAC, 50/60Hz Capabilities Scalability: 2-8 GSM channels up to 60 SIM cards inside one Sub Rack Connection specifications - Ethernet: 1 x Ethernet 10/100 Base-T RJ-45 - PC specifications - CPU: lntel Atom D2500 - RAM: 4 GB - Hard drive: 320 GB Skype: flamesgroupsia Mail: [email protected] Phone: 371-67-333-777 http://www.antrax.mobi
Views: 16828 Flames Group
How to Create Extension on Elastix
This video explains how to create an extension in Elastix. Elastix is a unified communication server software. For more explanation on this video: https://www.linuxhelp.com/how-to-create-extension-on-elastix/
Views: 655 Linux Help
IVR Example: Step 1 - Adding the VBVFrame
Sean White, Pronexus Support Manager, takes you through the process on how to build a simple IVR application using free IVR software. Step 1 demonstrates how to create a new Visual Studio doc and insert the VBVFrame. VBVoice allows you to build your own IVR using Pronexus VBVoice™ software, a drag and drop rapid application development IVR toolkit that integrates with Microsoft® Visual Studio®.
Views: 2731 Pronexus
How to Discover / Configure IP Address for Xorcom XE Series
http://www.xorcom.com - The XE series of Xorcom's Asterisk-based IP-PBX appliances features a multi-function Liquid Crystal Display, or "LCD", touch panel. This feature allows the system administrator to perform several of the most common administrative functions directly on the front panel of the PBX, without having to attach a keyboard and monitor. This video describes how to use the LCD panel to discover the PBX IP Address when using the factory-default DHCP defined option. It also describes how to manually configure the IP Address using the LCD panel. You can download the "How To" document about this functionality from this page: http://www.xorcom.com/product-manuals/xeseries-technical-documentation.html
Stereo Voice Chat test for the Raspberry Pi -  TeamTalk - QSO with voice & morse
Just a test to see how well a raspberry pi 2b CPU would perform using STEREO voice chat at 48k sample rate and 128 bit audio OPUS codec setting. TEST SETUP... voice on the RIGHT CHANNEL and MORSE CODE AUDIO TONES on the LEFT CHANNEL This allows for mixing independent channels or combing both audio and morse to both left and right... alsa_in & alsa_out was used on the PI for the INPUT and OUTPUT of a Stereo Voice Chat APP called TeamTalk http://bearware.dk/ TeamTalk has FREE servers in USA or EUROPE and a few other countries...or you can easily setup your own TeamTalk server... CPU use monitored... and averages less than 50 percent on a Raspberry Pi 2b running Raspbian Jessie 2 individual ALSA LOOPBACK cables were created and are used to pipe the audio to the INPUT and OUTPUT of TeamTalk (alsa_in & alsa_out commands in terminal) Jack Audio Connection Kit , JACKROUTER, and qjackctl help to route all the inputs and outputs of all the audio apps being used on the PI... on the laptop another instance of TeamTalk was used...in order to act as the transmitter & receiver of all audio going TO the pi's TeamTalk... then inside the PI's JackRouter(qjackctl), the audio was directed back to the Pi's TeamTalk's INPUT channel, acting like a repeater of sorts.. so that 2 way, duplex audio...was constantly going from and to each of the 2 TeamTalk clients ...to stress test the CPU of the Pi as much as possible with this setup...FULL DUPLEX, full time audio on both receive and transmit....for the duration of the test... TeamTalk is one of the few if not the only FREE Voice Chat app that has STEREO audio as an option...as well as options for SAMPLE RATE and BIT RATE... NOTE: podcast audio was used for the VOICE EXAMPLE from a podcast from here: http://www.qsotoday.com/podcasts/w6obb
Views: 334 QRQcw
Setup VICIDdial InGroup Destination for an Inbound Number- Switch2Voip
http://switch2voip.us Prices under 1 cent per minute VoIP France 0.0078 per Minute VoIP USA 0.008 per Minute VoIP United Kingdom 0.008 per Minute VoIP Canada 0.005 per Minute Switch2Voip is a leading provider of VoIP (Voice Over Internet Protocol) internet broadband telephone services often called PC to Phone service. While Switch2Voip offers VoIP calling plans to France home, residential and business customers it focuses their marketing efforts servicing Call Center and other companies worldwide that uses auto dialer or predictive dialer with a business model of pay as you go or prepaid. Asterisk VoIP Business VoIP Provider Call Center VoIP SIP Trunking 1-800 Toll Free Numbers Free Online Chat Support on Vicidial, Goautodial, etc. http://switch2voip.us voip provider voip provider vergleich voip provider kostenlos voip provider free voip provider österreich voip provider uk voip provider usa voip provider deutschland vergleich voip provider schweiz voip provider deutschland kostenlos voip provider vergleich voip provider kostenlos voip provider free voip provider deutschland voip provider österreich voip provider uk voip provider usa voip provider deutschland vergleich voip provider schweiz voip provider deutschland kostenlos voip provider australia voip provider asterisk voip provider api voip provider austria voip provider anmeldung nicht erfolgreich voip provider africa voip provider asterix voip provider albania voip provider android voip provider australia comparison voip provider best voip provider business voip provider bangladesh voip provider belgie provider voip belgium voip bad company 2 voip providers belgium voip providers betamax voip service providers business voip service providers byod voip provider canada voip provider cheap voip provider.com voip provider comparison voip providers compare best voip provider canada voip service providers compare voip service providers cheap voip providers list.com voip providers chicago voip provider deutschland voip provider deutschland vergleich voip provider deutschland kostenlos voip provider did rynga voip discount provider discount voip provider voip providers delhi voip service providers delhi voip provider europe voip provider elastix voip provider en france voip providers e911 voip providers egypt voip providers enterprise voip equipment providers voip provider free voip provider for asterisk voip provider for india voip provider for australia voip providers for international voip providers for mobile voip providers for 3cx voip providers for canada voip providers for uk voip free providers usa voip provider germany voip provider gateway voip provider gratis voip providers global voip providers gigaset voip providers greece good voip provider voip carrier grade goedkoopste voip provider goedkope voip provider voip provider hong kong voip provider how to voip provider how to become voip hosted providers voip service providers hyderabad voip providers houston voip providers home voip providers hawaii voip providers hyderabad voip hosted provider voip provider in pakistan voip provider india voip provider in bangladesh voip provider ireland voip provider in kolkata voip provider in canada voip provider in singapore voip provider in malaysia voip provider in mumbai voip provider in pune voip provider jakarta voip provider japan voip providers jordan voip providers jamaica voip providers johannesburg voip providers magic jack voip providers san jose voip provider kostenlos voip provider kolkata voip provider hong kong voip providers kenya voip providers keep phone number voip providers kerala voip providers kuwait voip providers ksa voip service providers kolkata voip minutes provider in kolkata voip provider list voip provider lingo voip provider list in bangladesh voip providers list usa voip providers lync voip company list voip carrier list voip providers sri lanka largest voip provider voip providers los angeles voip provider malaysia voip provider montreal voip provider melbourne voip providers mobile voip minutes provider
AudioCodes Mediant 1000 Video Datasheet
The Mediant 1000 is AudioCodes converged wireline VoIP media gateway, designed to interface between TDM & IP networks in enterprises or small-scale carrier locations.
Views: 13690 AudioCodes Media
Telappliant VoIPOffice - Setting up an IVR (virtual receptionist) tutorial
Learn how to set up your own virtual receptionist in VoIPOffice , ensuring calls are answered as effectively as possible and directed the the right department. VoIPOffice is a feature-rich, flexible and cost-effective business telephone system from Telappliant. It offers crystal-clear call quality, low call tariffs and a multitude of features that enable your business to communicate more effectively. It's a fully hosted solution, eliminating the need to purchase, install and maintain expensive telephony hardware. For more information, call 0203 384 8648 or visit http://www.telappliant.com/ip-pbx. VIDEO TRANSCRIPT: 'IVR' stands for 'Interactive Virtual Receptionist' - also known as an auto-attendant. When you call a company and hear a message saying, 'Welcome to Company A, press '1' for sales, '2' for support or '3' for accounts', this is an IVR. The vocal prompts direct your caller to the relevant department. To set up an IVR in VoipOffice, select the IVR on the left of the Admin Panel screen and click on 'Add IVR' at the top. A new box will appear on your screen. 'Name' is the name of your IVR; we will call it 'Main'. In 'Number', enter a unique number for the IVR; we will label our IVR as 5001. We will leave the 'Greeting' as a default greeting for now. The 'IVR Type' should be left as a 'Standard IVR'. When you construct an IVR, the options should reflect the departments in your company. The numbers each correspond to a vocal prompt on your IVR, for example 1 for sales, 2 for accounts. In VoipOffice, you can have up to 11 options. The destination and extension are where callers are directed when they enter the option number on their telephone keypads. A destination can be set from the drop-down menu as an extension, IVR and so on. In our example we will set up a sales department and an accounts department. We will set option 1 to a sales Ring Group labelled as 3001. We will set option 2 to an accounts Ring Group labelled 3002. Please note that, to set up a Ring Group, we actually select extensions. When completed, press 'Save'. You will now see that the new IVR has been created. Next you will need to record your IVR greeting. To do this, dial '*301' from your phone and you will hear a beep. After the beep, record your IVR message and then hang up. This has now created a greeting file, which can be found in the IVR you just created by pressing 'Edit'. In the greeting drop-down menu, you will find the greeting file that you just recorded. It will be labelled as 'greeting-hyphen' and then today's date and time'. Select it from the drop-down menu and press 'Save'. The final stage is to link your IVR message to a number. This is the number that a caller dials and is then presented your IVR message and directed to the relevant department. To do this, click on 'DIDs' on the left of your screen. Choose the number you want to link to the IVR and press 'Edit'. This will bring up a new box on your screen; set the 'Destination as 'IVR' and set the value as the IVR number you created; in our example it was '5001'. Once complete, press 'Save'. The IVR is now fully functional.
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