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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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We repair Macbook logic boards: https://rossmanngroup.com/macbook-logic-board-repair 👉 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: 48758 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: 1948 Ambiorix Rodriguez
9-Planet IPX-330 IPX-2100  Advanced Options Asterisk | الخيارات المتقدمة
 
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Global SIP , Default Configuration الاعدادات الافتراضية واعدادات ربط التلفونات SIP من الانترنت *Subscribe To My Channel and Get More Great Tips * اشتركوا معنا ليصلكم كل جديد ومميز فقط اضغط على الرابط التالي: http://www.youtube.com/channel/UCc7Xi94x7aLsripC0bnRxww?sub_confirmation=1 تابعوني ايضا على : https://www.facebook.com/HamadAbsi https://www.twitter.com/hamadabsi https://www.instagram.com/hamad.alabsi
FreePBX VoIP Tutorial Part 7 - Configuring Google Voice
 
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Update: If you'd like to not have to put the + or the 1 when dialing out, change your dialing patterns to look like this: http://i.imgur.com/yW3mWoE.png Make sure you're using the exact same dialing patterns. This includes adding your 10-digit Google Voice number to the CallerID field on all 3 patterns. I will explain why this is important when adding more than one Google Voice account to your Asterisk server in a later video. Commands used in the video: amportal restart amportal stop amportal start It could take several minutes to restart the server. Just be patient. After making this video, I went into FreePBX, Admin, Module Admin, pressed Check Online, and then clicked the "Show only upgradeable." There was a Google Voice motif update I applied and I haven't had trouble with amportal restart since. I talk a little more about module updating in Part 10. Relevant links: PuTTY (SSH): http://www.chiark.greenend.org.uk/~sgtatham/putty/download.html Part 1 - Introduction - http://www.youtube.com/watch?v=u9DzN1Pu6-Q&hd=1 Part 2 - Gmail and Google Voice Setup - http://www.youtube.com/watch?v=TJ_mZ_3t3r0&hd=1 Part 3 - VirtualBox and PBX in a Flash - http://www.youtube.com/watch?v=tl-knKvixzs&hd=1 Part 4 - DDNS - Free Static IP Setup - http://www.youtube.com/watch?v=gVeh1zdtJsE&hd=1 Part 5 - Router Settings - http://www.youtube.com/watch?v=bGq1_G6rKHY&hd=1 Part 6 - Configuring FreePBX - http://www.youtube.com/watch?v=G1dXb85Bzts&hd=1 Part 7 - Configuring Google Voice - http://www.youtube.com/watch?v=qA8Qw7GmPug&hd=1 Part 8 - Configuring CSipSimple for your first call - http://www.youtube.com/watch?v=3A-yar-JoOQ&hd=1 Part 9 - Installing codecs - G.729 - http://www.youtube.com/watch?v=xI91X6pcvF4&hd=1 Part 10 - Module updates + Backup & Restore - http://www.youtube.com/watch?v=R8RDTZBsUdc&hd=1 Part 11 - Setting up multiple Google Voice phones - http://www.youtube.com/watch?v=t1mFdRFNRsY&hd=1 Part 12 - Now what? Get a free phone number with IPKALL + Hold Music - http://www.youtube.com/watch?v=fR7O_DbUZI0&hd=1
Views: 45846 nirvgorilla
Asterisk Tutorial 15 - Asterisk Subroutines [english]
 
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Ever wanted to know how to get rid of all those lines of code that repeat themselves over and over again? Today we get yet even more real world like by reducing our business hours dialplan settings to just 2 lines of subroutine coding. In our example, we demonstrate how to use a subroutine to remove the unnecessary lines of dialplan coding when setting up your business hours - although subroutines are by no means limited to solely this function. Important information here is, if you can avoid using the "macro" function, you should, as this option will only provide a depth of seven levels, after which Asterisk will probably crash - use the "GoSub" application instead. For more information regarding our Business Communications and VoIP telephony solutions, please check out our website: ► We upgrade business communications • https://www.pascom.net/en/ ► Free pascom cloud business phone system • https://www.pascom.net/en/voip-installation/ ► pascom phone system free download • https://www.pascom.net/en/downloads/ ► Our Blog • https://www.pascom.net/en/blog/asterisk-tutorial-15-asterisk-dialplan-subroutines/
Views: 9954 pascom GmbH & Co. KG
Configure cisco 7940 7960 reset setup tftp for asterisk freepbx elastix pbx in a flash
 
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This video will show you have to reset, and set up tftp server info on a cisco 7940 7960 with sip firmware
Views: 37912 Alldigital phones
[part 15] Configuring Cisco SPA525G VoIP phones as extensions with FreePBX
 
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We repair Macbook logic boards: https://rossmanngroup.com/macbook-logic-board-repair 👉 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: 20456 Louis Rossmann
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: 2499 Twilio
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
FreePBX simple config Простая настройка FreePBX asterisk 13
 
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На видео показана самая простая типовая настройка freepbx. Простая настройка 1 транка для примера Создание 1 extension для примера Создание 1 входящего маршрута на extension Создание 1 входящего маршрута на ring group Создание 1 Исходящиего маршрута с диалпланом на мобильные Включение записей разговоров на 1 extension Отключение anonymous sip calls в advanced settings
Views: 5114 rstayalive
Vtiger CRM  Vici DIal Integration ( Asterisk Predictive  dial CRM integration )
 
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Vici Dial Predictive CRM Dialer Integration or CRM Auto Dialler improves efficiency of your phone communication by giving you more information and more options for each call you make or receive. Gives you an utterly new experience of effective phone communication right in your VtigerCRM, and drives your business processes to advanced standards. Features Vici Dial and crm Integration for campaign 1. Auto create lead in crm if lead is not present in crm on call dispo of dilaer. 2. Auto create Call logs in crm and realte to particular lead in crm with all relevant call details. 3. Can add multiple leads to multiple vici list from lead list view from crm lead , contact and account list view. 4. all data related to leads and call logs available in crm so user can create desire reports in crm. 5. One login for CRM and vici Dial 6. Can Synchronise CRM and Vici Dial User
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
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/asterisk-tutorial-22-asterisk-call-strategies/
Views: 7350 pascom GmbH & Co. KG
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.
Mikrotik VoIP SIP Server Port Redirect rules setup
 
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Mikrotik VoIP SIP Server Port Redirect rules setup
Views: 31999 Tania Sultana
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: 2848 L Bergey
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 Account Configuration, and Test in ZOIPER
 
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A quick walkthrough on what fields must be filled out.
Views: 19214 Aaron Hurst
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.
Mikrotik Router Mark VOIP traffic using ip firewall mangle rules
 
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Mikrotik Router Mark VOIP traffic using ip firewall mangle rules
Views: 12708 Tania Sultana
Asterisk Tutorial 10 - Incoming Calls Simulation [english]
 
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We're back! - so after a short break, we are back with the latest in our Introducing Asterisk Tutorial series. In today's episode, we take a look at back at what we have done so far as well as a more in depth look at configuring your dialplans to be able to receive incoming external calls by making a simulation using a softphone. 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/asterisk-tutorial-10-asterisk-incoming-external-calls/
Views: 22672 pascom GmbH & Co. KG
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: 470 Commsprepper
10-Planet IPX 330 ,IPX-2100  VoiceMail and SMTP settings Asterisk |البريد الصوتي
 
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Lecture 10:Planet IPX 330 ,IPX-2100 VoiceMail and SMTP settings البريد الصوتي *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
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: 34096 Crosstalk Solutions
Kamailio Surfing Big Waves of SIP
 
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Kamailio is an open source SIP server known for its performances and stability, working flawlessly together with Asterisk. It deals with SIP traffic at low level- which allows high flexibility in routing and management of the content for each packet. Having its own special routing language enables a wide range of optimizations, but could make it harder to learn it from scratch. However, Kamailio offers other variants that can be used to decide the routing of SIP packets, which can be easier to adopt in various scenarios. This presentation shows some typical use cases (such as load balancing, LCR and anti-fraud) implemented using different mechanisms: own routing language, embedded interpreters and external APIs, highlighting the benefits and the drawbacks for each option.
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/asterisk-tutorial-18-asterisk-voicemail-greetings/
Views: 8168 pascom GmbH & Co. KG
5-Planet IPX-330 Outbound Routes Asterisk | كيفية اعداد الاتصالات الخارجية
 
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كيفية اعداد الاتصالات الخارجية Outbound Routes *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
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: 10424 Bob Langys
LinPhone Android SIP Settings Stey by Step Video.
 
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This video we have shown, how to SIP settings Linphone with http://cheapestcall2india.com on android OS.
Views: 16683 Call to India Native
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: 3279 W00DY1848
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: 769 didexchange
Bitrix24  Total Telephony (Webinar 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: 2227 intreface
Configure a Softphone for your PBX or VoIP account
 
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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: 7268 VoicePulse
2-Planet IPX-330 Asterisk Tour | جولة سريعة على اعدادات السنترال
 
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جولة سريعة على اعدادات السنترال IP *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
Mitel, ShoreTel, FreePBX SIP Trunk Failover
 
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Mitel/ShoreTel automated phone number failover to #FreePBX using #SIP trunking with redundant configuration. Overview and demonstration of a phone number dynamically failing over from a #ShoreTel phone system to a FreePBX using SIP trunking. If you're concerned about your business phone service going down or your phone system being a single point of failure for your business then this video is for you. SIP trunking from Accent can help your business stay up and running even when your phone system is down. Stay communicating with your customers and don't miss critical phone calls. You can learn more about Accent and our SIP trunking services at https://www.AccentVoice.com You can see more of our videos at https://www.youtube.com/c/accentvoice
Views: 159 AccentVoice
How to setup your Trunk Carrier in Vicidial, Vicibox, VicidialNow.
 
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WARNING WARNING WARNING!!! www.datasoft.ws are Scammers and Spammers DO NOT TRUST THIS DIALER GROUP. THEY WILL RIP YOU OFF EVERY TIME!!! How to setup your Trunk/ Carrier in Vicidial, Vicibox, VicidialNow. Also how to sign up for a $2 Voip account and our hosted Vicidial solutions at www.callcentREvoip.com
Views: 45990 Aaron Mellick
Polycom SoundPoint IP 301 | Inbound Call | Asterisk PBX
 
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A call is recieved by a Polycom SoundPoint IP301 sip phone that is using an Asterisk server.
Views: 427 lre23
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.
Panasonic KX-HDV130 / KX-HDV100 firmware upgrade
 
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Firmware upgrade takes a bit more than 10 minutes, your SIP settings and phonebook are preserved after the upgrade but I recommend backing them up anyway. To update the firmware you basically have to check the connectivity to the firmware server from the phone (Menu-System Settings-Network Settings-Link Speed), then turn on the web interface (Menu-Basic Settings-Other Option-Embedded Web), login to the web interface from your PC (login: admin, password: adminpass), go to Maintenance-Upgrade Firmware and enter the URL to the first part of firmware. To upgrade to version 8.101 you can enter http://50.3.81.167/HDV130-08.101.fw.part1 The firmware is on my server, you should check if you can download it beforehand and if your phone pings the IP, this server is provided AS IS.
Views: 1251 Yury Grigoryev
How to setup a basic outgoing  trunk in Elastix.
 
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How to setup a basic outgoing trunk in Elastix.
Views: 104702 synapseglobal
SIP over TLS + SRTP: Decrypting Two Caller Traffic with Tshark
 
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SIP over TLS + SRTP: Decrypting Two Caller Traffic with Tshark Full course: https://www.pentesteracademy.com/course?id=43
Views: 2535 Pentester Academy TV
Homer Seven
 
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There's a new HOMER in town! Learn about HOMER Seven and its completely redesigned architecture, unleashing the power of HEP to new and unthinkable levels. Designed to integrate rather than segregate data, HOMER Seven expands its potential broadly and ships with multi I/O options ready to fit and feed out of existing Voice and RTC ecosystems, with native support for Asterisk and its many features.
Asterisk Tutorial 33 - Asterisk IVR Menu Looping [english]
 
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It's time to enhance our IVRs to account for "timeouts" by looping our IVR menus. Menu loops allow the IVR menu to be repeated should an option not be selected within a specific time frame, giving the caller the opportunity to make their selection. For more information regarding our Business Communications and VoIP telephony solutions, please check out our website: ► We upgrade business communications • https://www.pascom.net/en/ ► Free pascom cloud business phone system • https://www.pascom.net/en/voip-installation/ ► pascom phone system free download • https://www.pascom.net/en/downloads/ ► Our Blog • https://www.pascom.net/en/blog/
Views: 4572 pascom GmbH & Co. KG
✅ 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: 8310 Nick's Hardware
VoIP Supply | RenegadePBX Unboxing
 
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Join Marc Spehalski, our Senior VoIP Engineer, as we unbox the RenegadePBX by VoIP Supply and use it to make two Grandstream phone's ring using Elastix. http://www.voipsupply.com/renegadepbx-1u-appliance-with-elastix-open-source-software Hi, I’m Marc Spehalski, Senior VoIP Engineer at VoIP Supply and I’m here to unbox our brand new product the RenegadePBX. So lets get to it. Immediately you can see the USA sticker. This product is proudly made in the USA. So inside the box we have the PBX itself, and the power cord. With the power cord, there is also an included CAT5 cable, and the rack ears. On the back of the PBX, you see the power supply, keyboard and mouse, HDMI, VGA, two Ethernet gigabit ports, four USB, soundcard, and two PCI card slots. The renegadePBX ships with many different options, but they all ship with at least a sixty-one gigabyte, solid-state drive, two gigabytes of DDR3 Ram, and supports up to 75 concurrent calls. This particular PBX shipped with Elastix, an Asterisk based PBX, but for a complete list of options, please visit VoIP Supply dot com. Now that we have unboxed the RenegadePBX, lets put it to good use. So, next I’d like to register the Grandstream GXV3275, one of my personal favorites, and the super popular GXP2140. The first thing we need to do, is plug the PBX into the outlet. Then we can also plug it into the network. I’ll also add the two phones to the network. If you don’t have external power supplies for your phones, make sure you are plugging them into a POE enabled switch. Now that the RenegadePBX is fully powered on, we are ready to log in to the web interface via the IP address assigned by your DHP server. Our IP address in the lab is going to be ten dot ten dot ten dot fifty-six. Accept the self-sign certificate, and use the standard credentials: Admin, and VoIP Supply! Now it’s time to create a couple of extensions, and register the two phones. The first step is to click on the PBX tab, then to add an extension, click submit. We’ll call this first extension, 2001. And we’ll just call it, test. We will leave the rest default for now, but we will pay particular attention to the SIP Secret, which we’ll need later. We’ll scroll down, and click submit. We’ll also apply the config, and create the second extension. The next step is to log to each phone, and register to the RenegadePBX. On the GXV3275, it’s very easy. On the home screen, it already tells you what the IP address is, so we’ll just enter that into our browser. On the GXP2140, you hit the center button, go to status, and network status. Next we will register the accounts on the phones, to the extensions we built on the PBX. We’ll use the same SIP password that was originally presented to us, on the extension page. Now we have both extensions added. Now that both extensions are registered, we can place a test call using the extensions we created, 2001 and 2002. Hello, thank you for calling VoIP Supply. And it’s really that easy, to create a couple of extensions, and have them work, in only about ten minutes or so. For any further questions regarding the RenegadePBX or any of your VoIP Supply needs, please visit the brand new VoIP Supply dot com website. In the next video, we are going to be building a SIP trunk so we can make and receive calls from the outside. Once again, I’m Marc Spehalski, Senior VoIP Engineer at VoIP Supply, and thanks for watching.
Views: 1071 VoIP Supply
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,
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: 39670 nirvgorilla
Setup Own Asterisk VoIP Server with Android, iOS & Win Apps-Abhilash Nelson|Learnfly
 
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View more Learnfly instructors in this playlist: https://www.learnfly.com Explore the full course on Learnfly (special discount included in the link): https://www.learnfly.com/setup-own-asterisk-voip-server-with-android-ios-win-apps VoIP for Dummies - Asterisk VoIP Server setup with Android, iOS, Win Apps - Using Fully Open Source Server and Clients In this course, we will setup a VoIP server and the client devices and the clients can make calls in between them using the VoIP server. And no prior experience is required. VoIP or Voice over Internet Protocol is a technology that allows you to make phone calls across devices without using the normal analogue phone connection. So your calls will be placed across internet and the normal phone lines are not required. VoIP allows you to make calls from a computer, a mobile phone which is connected to internet, or a normal phone which is connected to a specific adapter called the 'VoIP Adapter'. The major benefit is that since this call is placed over internet, you don't need a separate line or a dedicated line in-order to make the call. Just an internet connection is more than enough to make a call. If you are a business owner trying to get down the cost of communication at your office, or you are trying to setup a vast call based call centre operation or you are a technical enthusiast who wants to host your own VoIP server and provide this service to your client users, then this is the course exactly for you. Let me now give you a brief overview of what are the topics that we are going to cover in this course. In the first session, which is basically a theory session, we will be covering the technology behind VoIP. And we will be covering the architecture and the working of VoIP technology compared to the normal PSTN (Public Switching Telephone Network) that has been there from the beginning. And after setting up the VPS server, we will installing Asterisk, which is a very popular open source VoIP server software available, and we will be installing it into our VPS server and we will also configure the ports, the specific number of ports that are required for the client devices to communicate with other client devices through the server. And in the next session, which is an important session, in which we will be configuring the dial plan and the extensions that we are going to use with our server. We will also make configuration to accept audio calls as well as video calls through our VoIP server. For our VoIP client applications we will be using an application called Linphone. It is also a completely open source application and the advantage is that it is available for all the platforms. For windows, Linux, Mac, Android and iOS device and the source code is completely open. You can download it and customize it as per your needs. The same configuration settings that you are going to make in this softphone, do the same configuration if you have a physical, hard wired, IP Phone with you. You can enter all these configuration into that and it will also work in the same way as we configure the soft phone clients. And in the next session, we will be configuring a windows based softphone, which is our linphone. We will be installing it in our windows and we will configure the option so that it can register with the server and make video and audio calls in between devices. And later, we will configuring it for Android. The same configurations. We will be installing it from the play store in an android device and we will make the configuration so that video and audio calls can be placed. And after that, we will have it for iPhone We will directly download it from the iPhone App Store and we will be installing it into our iPhone and then we will try to make calls from iPhone to other devices and we will be testing the video calls as well as audio calls across the devices. Learning and becoming an expert in VoIP technology is actually a very rewarding career because VoIP technology is very extensive so far and there is some time in future where we will discard all those analogue telephone lines and we will rely completely on VoIP based IP telephones. Because we need only a single internet channel , rather than having multiple channels. So world is evolving into that kind of a technology and VoIP experts are very much required in the market. And by the end of this course, we will be providing with you an experience certificate (Course Completion Certificate), which you will have great benefits, if you are trying for a VoIP based career. For more videos, subscribe to Learnfly Academy channel: https://goo.gl/dkNVnM Follow Learnfly Academy Facebook: https://www.facebook.com/learnfly/ Twitter: https://twitter.com/learnflySP Linkedin: https://www.linkedin.com/showcase/lea... Now, I hope to see you in this course
FreePBX 13 Made Easy - Part 2 - Initial Setup and Firewall
 
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In Part 2, we are going to discuss FreePBX initial setup and the FreePBX firewall. This covers best practices for FreePBX security and initial checklist of items to configure. FreePBX 13 Made Easy! playlist: https://www.youtube.com/playlist?list=PL1fn6oC5ndU8QTUpny7Gif9QeuN1fP2F9 Chris Sherwood with Crosstalk Solutions is available for best practice network, WiFi, VoIP, and PBX consulting services. Visit http://CrosstalkSolutions.com for details. Crosstalk Solutions is an authorized FreePBX and Sangoma partner and reseller. Connect with Chris: Twitter: @CrosstalkSol LinkedIn: https://goo.gl/j2Ucgg YouTube: https://goo.gl/g4G58M Amazon Wish List: https://amzn.com/w/M8KHAYD73CB4
Views: 58559 Crosstalk Solutions
6867i and 6865i Aastra Mitel SIP Phones Overview
 
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Brief overview of the Aastra-Mitel 6867i and 6865i SIP phones when used on Asterisk and FreePBX. Shows them making and receiving calls and accessing various screens.
Views: 3127 IP Office Techs