Home » Posts tagged 'google voice' (Page 7)

Tag Archives: google voice

The Most Versatile VoIP Provider: FREE PORTING

Best of Both Worlds: Marrying Asterisk to 3CX’s Free PBX with a $35 Raspberry Pi

blank

One of the real beauties of Asterisk® has always been its flexibility in talking to other PBXs, both commercial and open source. There are numerous reasons why you might want to try this. First, it makes it easy to migrate to a commercial platform where you can get support for mission critical telephony requirements. Second, you may want a hybrid setup where servers with on-site support personnel can run Asterisk while remote satellite offices can take advantage of a commercial PBX and the support options it offers. Third, you may want to take advantage of specific features that are only available by relying upon multiple PBX solutions. In the case of 3CX, their integrated softphone clients with one-click setup simplicity, conferencing and WebRTC apps, and Call Center offerings are the best in the business while providing unmatched VoIP security. Asterisk on the other hand is light-years ahead of almost everybody in the text-to-speech and voice recognition fields while offering the most powerful VoIP toolkit to build any custom VoIP application imaginable.

Today we thought it would be fun to walk you through the easy way to tie an Incredible PBX server with all its features to a powerful (free) 3CX platform with its virtually flawless softphone clients.1 When we’re finished, you’ll have a free 3CX server in the Cloud at a one-time total cost of $17.50. And you’ll be able to place and receive free U.S./Canada calls from any iPhone, Android phone, or PC using the 3CX client from anywhere in the world with nothing more than a WiFi connection. The Google Voice trunk supporting the calls will reside on Incredible PBX for the Raspberry Pi. When you’re sold on the power of the 3CX platform, you can upgrade to the 3CX 4-simultaneous call commercial offering with unlimited users and trunks at an annual cost of just $149. Maintenance and upgrades are included. Large organizations have relied upon back office servers for custom applications forever. And now you can take advantage of the same flexibility using a tiny $35 Raspberry Pi and our free (as in really free) Incredible PBX software. No Gotchas!

Initial Raspberry Pi Platform Setup

Before we can interconnect 3CX’s Free PBX with a Raspberry Pi, you obviously have to set up both PBX platforms. For the Raspberry Pi, our recent Nerd Vittles tutorial will walk you through the setup process. In lieu of a Raspberry Pi, you can use any legacy FreePBX®-based Asterisk platform including Incredible PBX 13, PIAF3, Elastix®, AsteriskNOW®, or FreePBX Distro®. The setup procedure is exactly the same.

Building a 3CX Server in the Cloud

Building a 3CX server in the Cloud is equally easy. Let’s go through the process once again. If you’re just experimenting, a lifetime Cloud-based server at CloudAtCost for a one-time charge of $17.50 cannot be beat. We would hasten to add that we don’t recommend this platform for production use, but it’s a terrific proof-of-concept option. When you’re actually ready to deploy 3CX for production use, the least costly Cloud solution is the $3.49 per month OVH RAID offering with 2GB of RAM and 10GB storage. The $5 per month offerings from Digital Ocean and Vultr are other alternatives worth a look. Both of these platforms come with free credits ($10 and $20, respectively) to let you try things out.

To get started, sign up for a $17.50 server at Cloud at Cost. They will send you credentials to log into the Cloud at Cost Management Portal. Change your password IMMEDIATELY after logging in. Just go to SETTINGS and follow your nose.

To build your free 3CX PBX, create a virtual machine by clicking on the CLOUDPRO button in the CloudAtCost control panel. Then click Add New Server. Choose 1 CPU, 512MB RAM, and 10GB storage for your server. Choose Debian 8 64bit as the OS Type and click Complete.

While CloudAtCost is building your server platform, obtain a free license key for 3CX.

blank

Once the Debian 8 server appears in your Control Panel, it will look something like what’s shown above, not CentOS obviously. The red arrow points to the i button you’ll need to click to decipher the password for your new virtual machine. You’ll need both the IP address and the password for your new virtual machine in order to log into the server which is now up and running with a barebones Debian 8 operating system. Note the yellow caution flag. That’s telling you that Cloud at Cost will automatically shut down your server in a week to save (them) computing resources. You can change the setting to keep your server running 24/7. Click Modify, Change Run Mode, and select Normal – Leave Powered On. Click Continue and OK to save your new settings.

blank

Finally, you’ll want to change the Host Name for your server to something more descriptive than c7…cloudpro.92… Click the Modify button again and click Rename Server to make the change. Your management portal then will show the new server name as shown above.

Next, log in to your new Debian server as root using SSH or Putty and issue the commands below. Step #1 is to change your root password. What appears as the fourth line below is actually part of the third line and needs to be run as a single command. The last line to install SendMail will actually be run after you elect to use the Web Interface Wizard to configure 3CX. Just run it from the SSH command line before you switch to a browser to complete the 3CX setup.

passwd
wget -O- http://downloads.3cx.com/downloads/3cxpbx/public.key | apt-key add -
echo "deb http://downloads.3cx.com/downloads/3cxpbx/ /" | tee /etc/apt/sources.list.d/3cxpbx.list
apt-get update
rm -f /zang-debian.sh
apt-get -y install 3cxpbx
apt-get -y install sendmail sendmail-bin

When the initial setup finishes, choose the Web Interface Wizard and complete the install using your favorite web browser. Enter your 3CX license key when prompted. Make up a very secure Username and Password to access your 3CX portal. Specify that your IP address is Dynamic when prompted (even though it isn’t). This tells 3CX to generate an FQDN for your server. Accept the default ports for HTTP (5000) and HTTPS (5001) access to your server. We recommend choosing 4-digit extensions numbers which will make it easy to distinguish 3CX extension numbers from 3-digit extension numbers of the RasPi platform. While logged into the 3CX management portal, adjust Settings → Email to Mail Server → 127.0.0.1 and Reply to → noreply@YourActual3CX-FQDN. Leave the other settings blank and click TEST then OK. Now download your favorite 3CX smartphone client, send yourself the Welcome Email for your default extension, and your 3CX initial setup is complete.

blank

Server Interconnection Overview

Now we’re ready to interconnect the two servers. What we’ll be doing is creating Trunks on both the Raspberry Pi and the 3CX server and tying them together. We’ll use this trunk to handle the call traffic between the two PBXs. Then we’ll add incoming and outgoing call routes on both servers to specify how the individual calls should be routed. Because the free version of 3CX limits the administrator to a single trunk, we’ll offload all of the provider trunks to the Raspberry Pi and reserve the one available 3CX trunk as the interconnect path to the Raspberry Pi. For today’s setup, we’ll use 3CX’s free softphone clients as the actual phone devices for end-users. Of course, you could also use your favorite SIP phones, and 3CX provides automatic configuration for dozens of devices. But we want to introduce the 3CX smartphone clients because they provide an incredibly easy way to get users connected without having to worry about punching holes in firewalls.

To place outbound calls on the 3CX side, 3CX provides enormous flexibility in call routing. Because we chose 4-digit local extensions when we set up the 3CX server, it will make it easy to route other calls through the outbound trunk to the Raspberry Pi using nothing more than the length of the dial string. For example, 3-digit calls line up perfectly with extension numbers on the Incredible PBX for RasPi platform. So 3CX users can easily reach extensions connected directly to the Raspberry Pi. And 10-digit 3CX calls will be forwarded to the Raspberry Pi as traditional outbound calls. They will be processed just as if you had dialed a 10-digit call from a Raspberry Pi extension. For example, if you have a registered Google Voice trunk to handle 10-digit calls on the Raspberry Pi, then the same call path would be used for calls originating from 3CX extensions. And, yes, calls to the U.S. and Canada would still be free and would display the CallerID associated with the Raspberry Pi’s Google Voice trunk. You could get more creative and add an additional dialing prefix on the 3CX side to route specific types of calls to a designated outbound trunk on the Raspberry Pi side based upon the dialing prefix, but we’ll leave that as a homework project for you.

For incoming calls on the 3CX side, in addition to 4-digit local extension-to-extension calling, we can define the destination for incoming calls that originate from either a Raspberry Pi extension or from outside calls coming in from one of the Raspberry Pi’s provider trunks. These are managed by assigning one or more DIDs in the 3CX trunk configuration and then creating 3CX Inbound DID Rules that tell 3CX where to route calls to each defined DID. For 3CX softphone clients registered to extensions, it means your cellphone will ring whenever a call is routed to that particular extension. On the Raspberry Pi side, we create Incoming Call Routes for each DID to be routed to 3CX and specify our defined 3CX trunk as the destination for incoming calls from those DIDs. Not all DIDs on the Raspberry Pi have to be routed to the 3CX server obviously. That is merely one of many call destination options available to the administrator on the Raspberry Pi server.

blank

Here’s a typical call path for an outside call that is placed to a Google Voice number registered with your Raspberry Pi. The Asterisk server running on the Raspberry Pi would answer the call placed to the Google Voice Trunk. Asterisk then would check for an Incoming Route on the Raspberry Pi with a DID matching the number of your Google Voice trunk. Finding a match, Asterisk would check for the desired destination of the call and would note that it is listed as the registered 3CX trunk. Asterisk would pass the call through this trunk to the 3CX server including its associated DID and CallerID info. The 3CX server would answer the incoming call and would check for an Incoming Route matching the DID passed from Asterisk. Finding a match, it would pass the call to the Extension specified in the Incoming Route. When 3CX rings the extension, it would also detect that a softphone was registered to that extension and would also ring the 3CX client on the user’s smartphone. The user answers the call on the 3CX client of their smartphone and begins a conversation. The free version of the 3CX server supports 8 simultaneous calls so you are unlikely to ever run out of call paths for calls in the home and small office environment.

Firewall Setup for Server Interconnection

Because the 3CX server is sitting in the Cloud, its firewall is configured automatically as part of the setup process. If your Raspberry Pi is sitting behind a NAT-based firewall, then you would need to map port UDP 5060 from the router on your public IP address to the private IP address of your Raspberry Pi. In addition, login to your Raspberry Pi as root using SSH and run /root/add-ip to whitelist the public IP address of your 3CX server in the cloud. Otherwise, the 3CX server cannot establish a connection to your Raspberry Pi.

Raspberry Pi Trunk Configuration

Using a browser, login to the web interface for FreePBX on your Raspberry Pi and choose Connectivity → Trunks → Add SIP (chan_sip) Trunk. Name the trunk remote. In the Outgoing Settings, make the entries shown below naming the trunk remote and using a secure secret that will be used to interconnect the two servers. The Register String looks like the following: main:secret@3CX-IP-Address where main is the 3CX server trunk name, secret is your secure secret, and 3CX-IP-Address is the 3CX public IP address.

blank

3CX Trunk Configuration

Using a browser, login to your 3CX server: https://3CX-IP-Address:5001 or http://3CX-IP-Address:5000. From your Dashboard, choose SIP Trunks → Add SIP Trunk. Create a Generic SIP Trunk and then fill in the blanks as shown below. For Registrar/Server/Gateway Hostname or IP, use the public IP address or FQDN of your Raspberry Pi. For Type of Authentication choose Outbound. The authentication credentials should be remote and the secure secret you chose, and the Main Trunk No should match the DID of the Google Voice trunk you set up on your Raspberry Pi. Then pick a default Destination for incoming calls.

blank

3CX Outbound Rules Configuration

Next, we need to tell 3CX which outgoing calls to send out through the Raspberry Pi trunk we just set up. In our example today, we’re going to send all 10-digit calls and 3-digit calls. The 10-digit calls will be routed out the Google Voice trunk on the Raspberry Pi side. And the 3-digit calls will be sent directly to Raspberry Pi extensions. So we’ll need two Outbound Rules.

For the first rule, choose Outbound Rules → Add. For the Rule Name, specify StandardOut. Apply the rule to Calls to Numbers with a length: 10. For Route 1, choose Generic SIP Trunk as the Destination. Click OK to save the new rule.

For the second rule, choose Outbound Rules → Add. For Rule Name, specify StandardInt. Apply the rule to Calls to Numbers with a length: 3. For Route 1, choose Generic SIP Trunk as the Destination. Click OK to save the new rule.

If you already have configured a 3CX smartphone client for one of your 3CX extensions, you now should be able to dial any 3-digit or 10-digit number and have the call processed through your new 3CX→RasPi trunk without any further setup assuming you’ve created a Google Voice trunk on the Raspberry Pi side. That wasn’t too hard, was it?

Routing Incoming Google Voice Calls to 3CX

Depending upon your own requirements, you may want to route incoming Google Voice calls or other trunks directly to an extension and/or softphone on your 3CX server. You obviously could set up multiple trunks of any type on the Raspberry Pi side and have the calls to each trunk routed to a different extension or softphone on the 3CX side. To enable this on the 3CX side, edit your Generic SIP Trunk and click the DIDs tab. Then Add each of the 10-digit DIDs of the Raspberry Pi trunks you wish to redirect. Next, create an Inbound Rule for every DID and tell 3CX where to route the calls.

On the Raspberry Pi side, add each of your Google Voice Trunks. Then create an Inbound Route for each DID and specify the Destination as Trunks → Remote (sip). The 3CX server will take care of routing the various incoming calls to each of the Google Voice trunks to its predefined extension and/or softphone. Enjoy!

Originally published: Monday, March 6, 2017




blank
Need help with Asterisk? Visit the PBX in a Flash Forum.


 

Special Thanks to Our Generous Sponsors


FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

blankBOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

blankThe lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

blankVitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
 

blankSpecial Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
 



Some Recent Nerd Vittles Articles of Interest…

  1. A simpler Bridge setup is available in the paid versions of 3CX. []

Cellphone Hell: 2017 Minefield Navigation Guide

blank

Well, it’s been an interesting week. RingPlus, a Sprint MVNO, has gone belly up after Sprint pulled the plug on them. Lawsuit details are here. Then, not surprisingly, Sprint announced a new "unlimited" talk, text, and data plan: 5 phones for $90 with a free iPhone with trade-in. After first year, Sprint price escalates to $160 for 4 lines or $190 for 5 lines. And then, Verizon surprised everyone with an "unlimited" plan of their own: 4 phones for $180. With both of these plans, you pay through the nose for the first phone, and then the remaining ones are either free or almost free. So you might as well have some more babies and give them each a phone. For our weary followers that have been with RingPlus, you are about to be introduced to the Sprint Gotcha. Unbeknownst to you, when you inserted that RingPlus SIM and turned on your phone, Sprint locked the phone to their network. And guess what? RingPlus can’t unlock it, and Sprint won’t claiming that you’re not "their customer." But, alas, if you’ve bought your phone, you’re still entitled to use it with a provider of your choice. And, if your phone supports other CDMA carriers such as Verizon or GSM carriers such as AT&T and T-Mobile, you’re in luck. There’s a terrific guy with a company called GSM Zambia, and he will unlock your Sprint phone for $10.84 assuming you have a Windows PC with a USB connector and cable to plug in your phone. For those lucky enough to have a Google-branded phone such as a Nexus or Pixel, you have no worries. Google unlocks it automatically when you insert a SIM card from a different provider.

There are more gotchas awaiting those with iPhones. You see Apple actually makes an iPhone that supports all four of the major U.S. carriers: Verizon, Sprint, AT&T, and T-Mobile. The problem is you probably didn’t get handed that phone. Instead, you got one that was locked to the Sprint network or the AT&T/T-Mobile GSM network, and both of them are missing the necessary radios to support other carriers. But there’s good news. If you’re a loyal customer and have AppleCare for your iPhone, chances are pretty good that Apple will work with you to swap out the phone for one that will work with the carrier of your choice. You have to say this for Apple. Nobody else in the cellphone business would even give you the time of day if you made such a request. So, yes, we are a FanBoy and for very good reason. Apple bends over backwards to help out its loyal customers. Just be advised that you probably will need to speak with an Apple Store manager, and he will probably have to call Cupertino to obtain the document explaining how to handle the transaction. In our case, it was several phones under Apple leases which made things even more complex. But Apple solved it, and they were pleasant about it.

AT&T has had a new "unlimited" plan for about a year, but there were several gotchas in addition to their fine print about what unlimited really means. First, you had to also be a DirecTV customer, but they eliminated that requirement today. And, second, tethering was prohibited. While we’ve previously noted that you could work around the tethering problem by purchasing a ZTE Mobley portable device for your car that could be used outside the vehicle with an adapter. But the wrinkle was AT&T wanted another $40+ a month to cover the device on your unlimited plan. While AT&T boasts that the fourth phone on the unlimited plan is free, it turns out the car device doesn’t meet their definition so, if you only need 3 phones, you still have to cough up the $40 for the mobile device.

T-Mobile also had an "unlimited" plan, but it also restricted tethering. However, T-Mobile is not one to leave money on the table, and they quickly removed the tethering limitation once the Verizon plan was announced. So the bottom line on the 4-phone unlimited plans as of today looks like this: Sprint $90 (10GB tethering), T-Mobile $160 (10GB tethering), AT&T $180 (no tethering), and Verizon $180 (10GB tethering). All four carriers describe their plans as "unlimited" while none truly are insofar as 4G data is concerned. The new buzzword is "deprioritization" which means the carrier reserves the right to slow your data speeds once you reach a certain threshold. Also be advised that zero-rating of certain services is likely to become less of an issue with the Trump administration. In T-Mobile’s case, you get unlimited streaming of certain music and video services at reduced bandwidth. With AT&T, you get streaming of DirecTV movies at reduced bandwidth. With Sprint, you get HD video streaming at no extra cost plus a free iPhone7 for the next 18 months when you trade-in certain older phones. Unless you live in a very busy metropolitan area, user reports suggest that deprioritization shouldn’t be a concern. Here’s the Reddit thread with everything you need to know.

Despite our extreme dislike for almost everything about the Sprint organization and the way they do business, if you happen to live in a city with good Sprint coverage, you really can’t beat their 5 phones for $90 "unlimited" deal at least for one year. After that, Sprint is no bargain at all. If you’re using RingPlus, then that probably means you already have endured Sprint so the change will be easy for you. Just be advised that there are plenty of Sprint reps out there that will try to tell you your phones don’t qualify because they were "prepaid" phones and the plan is only available for "postpaid" phones. A better approach is to visit a Sprint store and advise them that you wish to port your existing phones to the new Sprint unlimited plan. That seems to work although YMMV. Remember, it’s still Sprint you’re dealing with. Good luck!

Feb. 27 Update: The Unlimited Data Plan competition continues to escalate. Today, AT&T sweetened its unlimited plan offering by adding 10GB of free tethering to each phone on its plan beginning Thursday. And T-Mobile announced that customers now can register three phones on its unlimited plan for only $100/month. Unlike Sprint, the T-Mobile offering has no one-year discount cutoff for customers taking advantage of the special pricing. All four major carriers in the U.S. now offer 10GB/month of tethering for each phone on an unlimited data plan.

Published: Thursday, February 16, 2017  Updated: Monday, February 27, 2017


blank
Need help with Asterisk? Visit the PBX in a Flash Forum.


 

Special Thanks to Our Generous Sponsors


FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

blankBOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

blankThe lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

blankVitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
 

blankSpecial Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
 



Some Recent Nerd Vittles Articles of Interest…

R.I.P. RingPlus: Don’t Panic, Google Hangouts Is Your Friend

blank

Three months ago, we warned you that RingPlus appeared to be on its last leg and to begin planning your exit strategy. February 11 appears to be their Drop Dead Date although Ting apparently will now absorb all RingPlus accounts that aren’t ported out but under markedly different terms. For openers, there’s a $6 per month charge per phone before you even turn it on. After that, it’s pay-as-you-go for minutes, messages, and data. While the buckets are pooled between all of your phones, the pricing is not cheap. 1,000 minutes will run you $18/month while 2100 minutes will set you back $35/month. 1,000 text messages costs $5/month while 2,000 will run $8/month. 2 gigs of data is $20/month with an additional $10/month for each additional gigabyte or fraction thereof. For a family of four with four phones and modest usage sharing 2100 minutes, 2000 messages, and 2 gigs of data per person, the monthly tab (not including taxes and fees) would be $123 which works out to over $30 per phone. To add insult to injury, Sprint automatically locks RingPlus phones to their network, but you’ll have to move heaven and earth to get them to unlock the devices, even those that were paid in full, without moving to Sprint for several months of "service." Thinking you’ll switch to Sprint’s unlimited 5 phones for $100 unlimited plan, think again. They’ve ended that promotion. So what is a guy or gal to do? We’ve dusted off some of our previous recommendations to provide you an extremely inexpensive solution if you have WiFi access most of the time and do modest mobile calling and texting without WiFi.

Feb. 11 Update: We have some good news and bad news. First, the bad news. It’s still the Sprint network. And now the good news. RingPlus has alerted users that their Drop Dead Date has been postponed until February 21. The migration to Ting is expected next week, well before the Feb. 21 date. And more good news. Sprint is not one to leave money sitting on the table, and yesterday they announced a new "Unlimited" Plan offering 5 lines with unlimited calls, texts, and data for $90 a month. The pricing is good through the end of March, 2018. This is good for a couple of reasons. First, it will keep you on the Sprint network for a month or two in order to qualify all of your existing phones for unlocking. Second, it’s a great deal despite the fine print explaining that unlimited doesn’t really mean unlimited: "Data deprioritization applies during times of congestion. Reqs ebill and new account." Don’t think you’re going to save money by moving fewer phones. The first phone is $50/month. The second is $40. And the next three are free. Our best advice is to wait until after your numbers have been ported to Ting which is better equipped to handle porting than RingPlus while the Titanic is sinking.

Feb. 13 News Flash Verizon introduces unlimited data plans including tethering…


When we introduced Google’s new Pixel phone, we noted that it seemed like a perfect candidate to determine whether we could do everything a normal mobile phone could do using no cellular service. In other words, we wanted to set up the mobile phone basically in Airplane Mode with no SIM card usage and just WiFi connectivity. The goal was to determine what, if anything, we couldn’t do that we’d normally expect to do using a top-of-the-line mobile phone. There was one obvious prerequisite. The mobile phone needed an Internet connection. This could be a normal WiFi network connection, or it could be a connection using an LTE-powered WiFi HotSpot, or it could be a WiFi connection established through tethering to an existing smartphone. It turned out that all three options let us make calls, send and receive text messages, and surf the web with no monthly cost for the additional phone other than the cost, if any, for the WiFi data used.

Why Would You Do Such a Thing? We can think of a number of reasons. The most important considerations for RingPlus users are it’s considerably cheaper than adding another mobile phone to your cellular plan and you’re not tied to Sprint’s lousy network. Typically, adding a mobile phone to many cellphone plans can cost upwards of $50 a month before you make the first call. Second, some may like the flexibility of having BOTH an iPhone and an Android phone because of differences in features and functionality. Finally, it’s a perfect way to introduce younger children to mobile phone technology without spending an arm and a leg on cellular service.

blank

What’s a Typical Use Case for a Non-Cellular Mobile Phone? We can think of several scenarios where this makes perfect sense. We happen to have a Verizon HotSpot that’s still on an unlimited data plan. While it costs almost $100 a month, it lets 7 devices connect to blazing fast LTE service at zero additional cost. If you travel with a group of people that all need mobile phones and that typically travel or work together except when alternative WiFi service is available, this is a real deal. For those with a newer vehicle that includes a WiFi HotSpot or an OBD-II diagnostics port1 and AT&T’s $100 ZTE Mobley device, mobile phones and tablets in the car or truck work perfectly without a cellular connection. And AT&T now lets you add a vehicle’s WiFi hotspot or ZTE Mobley to their unlimited data plan for $40 a month.2 If you have four kids and a spouse, you can do the math. Finally, if you and your family or business associates spend 95% of the day either at home or in an office or car with WiFi, everyone now gets the flexibility of a mobile phone with no recurring cost just like the good ol’ days with RingPlus. Driving our daughter to the school bus stop in our old neighborhood recently, we happened to check for WiFi access because the cellular service was so horrible. There were 27 separate WiFi HotSpots, all of which were secured. Seems we weren’t the only ones having difficulty with cell service in the neighborhood.

blank

VoIP Requirements for a Non-Cellular Mobile Phone. As we’ve said many times, the beauty of VoIP technology is not having to put all your eggs in one basket. So there’s really no reason to deploy a single technology. In the Google world, that means you can take advantage of Google’s rich collection of messaging applications such as Hangouts and Allo and Duo while also deploying Skype, Facebook Messenger, WebRTC and SIP-based services to connect to traditional hosting providers and PBXs such as Incredible PBX and PIAF5 powered by 3CX (shown below). Today we’ll walk through the setup process for all of these. When we’re finished, you’ll have crystal-clear phone calls as well as SMS messaging with something you never got with RingPlus, multiple layers of redundancy.

blank

What Does All of this Really Cost? You obviously have to purchase a mobile phone but, if you’ve been with RingPlus, you already have one or more phones in hand. When we’re finished today, you’ll be able to make calls as well as send and receive SMS messages in multiple ways. Calls and SMS messages to U.S. and Canada destinations are free using Google’s services. Skype-to-Skype calls worldwide are free. SMS messages sent and received using Pinger/Textfree as well as Facebook Messenger are also free of charge. For calls made using a SIP softphone or WebRTC connection to an Incredible PBX or PIAF5 PBX, you only pay the standard VoIP tariff for the calls, typically less than a penny a minute for domestic calls. Calls to many international destinations are free using FreeVoipDeal.com.

blank

Numerous SIP softphones for Android devices are available at no cost including Zoiper, CSipSimple and many others. Still others are available for less than $10 and can be installed on as many Android devices as you happen to own, e.g. Acrobits and Bria. And, of course, the 3CX softphone above is free with PIAF5. Stick with softphones with 4 stars or better!

blank

Putting the VoIP Pieces in Place. Once you have your SIM-free phone in hand, switch to Airplane mode and then reactivate WiFi. Go through the basic setup to establish a WiFi connection in your home or office. Then it’s time to add the components you’ll need to turn your smartphone into a fully-functioning VoIP phone. If they’re not already on your phone, download the following apps from the Google Play Store or the iPhone App Store: Hangouts, Hangouts Dialer, Allo, Duo, Skype, Facebook Messenger, Textfree, Port Knocker,3 DynDNS or FreeDyn (not free!) Client,4 and the VoIP softphones of your choice.

We recommend reserving the Google Voice number associated with the primary Gmail account on your smartphone for use with Hangouts, Allo, and Duo. The reason is that you can’t really use these services satisfactorily while also using the same Google Voice number with Google Chat and the Asterisk® XMPP module. Our previous Nerd Vittles tutorial will walk you through obtaining multiple Google Voice numbers to use with with your smartphones or Incredible PBX and PIAF5.

Pictured above is the layout we actually use. Keep in mind that the bottom row stays in place as you scroll through other screens on your smartphone. Long-press on an existing icon on the bottom row and drag it off the row. Then long-press on the app you want to add and drag it onto the bottom row. We recommend replacing the default Phone and Messaging apps with the Hangouts Dialer and Allo (as shown). We also include a SIP softphone on the bottom row which gives you multiple ways to place and receive calls.


[soundcloud url="https://api.soundcloud.com/tracks/293184354″ params="auto_play=false&hide_related=false&show_comments=true&show_user=true&show_reposts=false&visual=true" width="400″ height="300″ iframe="true" /]

But I Really Want a Cellphone Provider. Yes, we hear you. Backup cellphone service has its virtues. Here are 3 Android phones from Google ranging in price from $199 to $649 with easy payment plans ranging from $8 to $27 a month. Each gives you unlimited domestic calling as well as unlimited domestic and international texting with multiple cellphone carriers. Rates start at $20 a month plus $10/GB for data. You even get bill credits for any data you don’t use. Project Fi is worth a careful look if you’re on a budget and limit most of your data usage to WiFi connections. Here’s a great article explaining the pro’s and con’s of Project Fi after six months of actual usage.

Bottom Line. On our smartphone we have the following services activated and functioning reliably: Google Voice with Hangouts, Allo and Pinger for SMS messaging, Bria for VoIP calling with Incredible PBX for XiVO, CSipSimple and Zoiper for SIP calling with Incredible PBX 13, Facebook Messenger, Skype, plus the 3CX Dialer for calling with PIAF5 powered by 3CX. That translates into 5 different phone lines supporting free incoming and outgoing voice calls, plus 2 additional lines for free SMS messaging, plus the Facebook and Skype services to reach over a billion people worldwide at no cost. And both the PIAF5 and XiVO lines can support calls via multiple trunks using customized dial prefixes. Even with all these services running, most smartphones have sufficient horsepower to make it through a busy day. What are you waiting for? Make the switch!

blank

Update: Another WiFi HotSpot Option. We’ve now had an opportunity to test yet another WiFi HotSpot solution. This one uses AT&T’s Unite Pro, available from Amazon for $65. Rather than sign up for AT&T’s $50 monthly pay-as-you-go plan, you can use this with StraightTalk with minimal effort. That lets you purchase 2 months of service and 4GB of BYOT data for $40. Be sure to purchase the Mobile HotSpot Plan and not a cellphone plan! To get started, visit your neighborhood WalMart and pick up the StraightTalk Bring Your Own Tablet SIM Activation Kit. Do NOT mistakenly buy the BYO Phone Activation Kit. It won’t work! The kit includes a 1GB Data service plan that’s good for 30 days from activation. With your Unite Pro in hand, insert the AT&T SIM card that came with the tablet activation kit. Turn on the HotSpot and select it as your WiFi connection using your desktop PC. The name of the HotSpot and its WiFi password will be displayed on the main screen of the HotSpot. From your desktop, use a browser to log into 192.168.1.1. The default administrator password is attadmin. Goto Settings -> Mobile Broadband -> APN and add a new APN with name: StraightTalk and APN: tfdata. After saving it, select it as active APN.

Now switch back to your desktop PC and change your default WiFi connection so that you can access the Internet. Visit http://STBYOT.com to register your StraightTalk SIM card and activate your 1GB data plan. You’ll need your SIM card number (at the bottom of the big card from which you removed the MicroSIM). Then you’ll need your scratch-off data service PIN. Once you complete the setup and register with StraightTalk, activate your service. Now turn off your HotSpot and turn it back on. It should display a 4G connection at the top left of the screen. At this point, reselect the HotSpot as your WiFi connection for your desktop PC. Then try to make a connection to a web site on the Internet, and you should be in business. For conservative data usage, 4GB for $40 with 2 months of service is the best deal we’ve been able to find for those that prefer pre-paid cellular service.

Published: Thursday, February 9, 2017  Updated: Friday, February 17, 2017


blank
Need help with Asterisk? Visit the PBX in a Flash Forum.


 

Special Thanks to Our Generous Sponsors


FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

blankBOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

blankThe lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

blankVitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
 

blankSpecial Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
 



Some Recent Nerd Vittles Articles of Interest…

  1. OBD-II port is mandatory on vehicles sold in the U.S. since 1996. But you may not need a vehicle at all. 🙂 []
  2. DirecTV service is required to take advantage of AT&T’s Unlimited Data Plan offering. []
  3. We strongly recommend setting up PortKnocker with the credentials found in /root/knock.FAQ on your Incredible PBX server. This will provide a back door to assure that you aren’t inadvertently locked out of your server by the Incredible PBX firewall while you are traveling. []
  4. You’ll need to set up a dynamic DNS client on your Android phone in order to keep your IP address updated and whitelisted with the Incredible PBX firewall. Unfortunately, this feature requires the FreeDyn paid app on the iPhone. []

Chasing Rainbows: The VoIP in the Cloud Trifecta


blank

Week after week, the VoIP landscape for Cloud Computing continues to improve. And today we have more terrific news. Not only is there a new release of Wazo with simplified support for WebRTC and FollowMe roaming, but the Wazo 17.02 release also is now available on the RentPBX platform worldwide. Coupling Wazo and RentPBX with a secondary Cloud platform to achieve total VoIP redundancy is the VoIP in the Cloud Trifecta if ever there were one. RentPBX has been a platinum sponsor of Nerd Vittles for many years and, while they may not be the cheapest Cloud provider, they are certainly the best when it comes to VoIP. The reason is simple. Their cloud platform is only used for VoIP so you’re not competing for server resources with a zillion customers that are compiling millions of lines of code all day long. You also get free support! Their worldwide hosting locations translate into crystal clear VoIP calling without jitter using your favorite VoIP providers. With the Nerd Vittles NoGotchas coupon code, monthly service is just $15. For mission critical VoIP platforms, we recommend you set up Wazo with RentPBX as your primary server and configure a secondary server at OVH or Vultr.com or Digital Ocean for an additional $3.50 to $5 per month. Using Wazo’s native High Availability feature, your business gets a fault-tolerant platform with automatic failover for less than $20 a month.

Installing Incredible PBX for Wazo at RentPBX

We want to quickly walk you through the installation procedure at RentPBX because it’s the easiest cloud platform to get up and running, period. First, sign up for an account at RentPBX and order Incredible PBX for Wazo which you’ll find under the PBX in a Flash section of their site. Next, choose your favorite hosting location. We strongly recommend their Miami site if you’re east of the Rockies. For example, ping times to atlanta.voip.ms are under 14 milliseconds. The LA node works great for those on the Left Coast. Then choose Incredible PBX Wazo (Debian 8 Asterisk 14) for your platform. Enter a hostname for your server (HINT: test.rentpbx.com works fine if you don’t have your own) and click Continue. Enter NoGotchas for your Promo Code. Click Validate Code and then Checkout. Once you receive your credentials, login to your new server as root using SSH or Putty. The RentPBX setup procedure is a two-step install. First, you get Debian up to date. Then you reboot and the main Incredible PBX installer will be run.

blank

Because of some new certificates, you will get an exim prompt during the initial phase of the install. Just type q to proceed. After initial reboot, log back in with your root credentials and complete the prompts to add your Wazo web password, a telephone reminders numeric password, and a PPTP username and password. Review your passwords carefully. Then press ENTER to proceed with the installation of Incredible PBX for Wazo. Set your time zone when prompted. After about 5-10 minutes, you will be prompted to verify that the Wazo base install completed successfully. It’s perfectly normal that some of the Wazo services are disabled at this juncture. If you see “Wazo fully booted” after the listing of services, you’re good to go. Just press ENTER to proceed. The installer then will run the Wazo Wizard. Within a minute or two, you will again be asked to verify that it completed successfully. If you see no error messages, press ENTER and go have a cup of coffee. The rest of the install will proceed without further prompting. In 10-20 minutes, your server will be ready to use.

Setting Up SIP and Google Voice Trunks with Wazo

When the installation is finished, you can make toll-free calls in the U.S. and Canada without doing anything except dialing "1″ and the 10-digit number from any phone connected to your server. For other calls, there are two steps in setting up trunks to use with Incredible PBX. First, you have to sign up with the provider of your choice and obtain trunk credentials. These typically include the FQDN of the provider’s server as well as your username and password to use for access to that server. Second, you have to configure a trunk on the Incredible PBX for Wazo server so that you can make or receive calls outside of your PBX. As with the platform tutorials, we have taken the guesswork out of the trunk setup procedure for roughly a dozen respected providers around the globe. In addition, Wazo Snapshots goes a step further and actually creates the trunks for you, minus your credentials, as part of the initial Incredible PBX install.

blank

For Google Voice trunks, log into your server as root and run ./add-gvtrunk. When prompted, insert your 10-digit Google Voice number, your Google Voice email address and OAuth 2 token. The native Google Voice OAuth tutorial explains how to obtain it.

blank

For the other providers, review the setup procedure below and then edit the preconfigured trunk for that provider by logging into the Wazo web GUI and choosing IPX → Trunk Management → SIP Protocol. Edit the setup for your provider (as shown above) and fill in your credentials and CallerID number in the General tab. Activate the trunk in the Register tab after again filling in your credentials. Save your settings when finished. No additional configuration for these providers is required using the Incredible PBX for Wazo Snapshot.

Directing Incoming Calls from Wazo Trunks

Registered Wazo trunks typically include a DID number. With the exception of CallCentric, this is the number that callers would dial to reach your PBX. With CallCentric, it’s the 11-digit account number of your account, e.g. 17771234567. In the Wazo web GUI, we use IPX → Call Management → Incoming Calls to create inbound routes for every DID and trunk associated with your PBX. Two sample DIDs have been preconfigured to show you how to route calls to an extension or to an IVR. To use these, simply edit their settings and change the DID to match your trunk. Or you can create new incoming routes to send calls to dozens of other destinations on your PBX.

Routing Outgoing Calls from Wazo to Providers

Outgoing calls from extensions on your Wazo PBX must be routed to a trunk provider to reach call destinations outside your PBX. Outgoing call routing is managed in IPX → Call Management → Outgoing Calls. You tell Wazo which trunk provider to use in the General tab. Then you assign a Calling Digit Sequence to this provider in the Exten tab. For example, if NXXNXXXXXX were assigned to Vitelity, this would tell Wazo to send calls to Vitelity if the caller dialed a 10-digit number. Wazo has the flexibility to add and remove digits from a dialed number as part of the outbound call routing process. For example, you might want callers to dial 48NXXNXXXXXX to send calls to a Google Voice trunk where 48 spells "GV" on the phone keypad. We obviously don’t want to send the entire dial string to Google Voice so we tell Wazo to strip the first 2 digits (48) from the number before routing the call out your Google Voice trunk. We’ve included two examples in the Wazo Snapshot to get you started. Skype Connect (shown below) is an example showing how to strip digits and also add digits before sending a call on its way:

blank

Setting Up a Softphone & WebRTC with Wazo

If you’re a Mac user, you’re lucky (and smart). Download and install Telephone from the Mac App Store. Start up the application and choose Telephone:Preference:Accounts. Click on the + icon to add a new account. To set up your softphone, you need 3 pieces of information: the IP address of your server (Domain), and your Username and Password. In the World of Wazo, you’ll find these under IPBX → Services → Lines. Just click on the Pencil icon beside the extension to which you want to connect. Now copy or cut-and-paste your Username and Password into the Accounts dialog of the Telephone app. Click Done when you’re finished, and your new softphone will come to life and should show Available. Dial the IVR (4871) to try things out. With Telephone, you can use over two dozen soft phones simultaneously on your desktop.

blank

For everyone else, we recommend the YateClient softphone which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for the Wazo Line. You’ll need the IP address of your server plus your Line username and password associated with the 701 extension. On the Wazo platform, do NOT use an actual extension number for your username with Wazo. Go to IPBX Settings → Lines to decipher the appropriate username and password for the desired extension. Click OK to save your entries.

blank


WebRTC allows you to use your Chrome or Firefox browser as a softphone. Extension 701 comes preconfigured for WebRTC access with Incredible PBX for Wazo. It shares the same password as the Line associated with extension 701, but the username is 701 rather than the username associated with the Line. You can decipher the password by accessing the Wazo Web GUI and then IPBX → Services → Users → Incredible PBX → XiVO Client Password. Or log into your server as root using SSH or Putty and run: /root/show-701-pw. Wazo introduces several new features to WebRTC including support for the awesome new Opus codec plus voicemail management and even Gravatar support. It’s all preconfigured!

Special Note: Beginning with this version of Wazo (17.02), WebRTC is fully integrated with NGINX on your server, and a simplified method for configuring WebRTC users has been added. When you create a new User account, simply choose the SIP (WebRTC) Protocol when creating a new user account, and all of the Advanced Line options required to support WebRTC will be preconfigured for you.

blank

To use WebRTC, you first need to accept the different SSL certificates associated with the WebRTC app. From your browser, go to the following site and click on each link to accept the certificates. Once you’ve completed this process, visit the Wazo WebRTC site.

Before logging in, click on the Gear icon in the lower right corner and then click on the Pencil icon to edit your Settings. Fill in the public IP address of your Wazo server and specify 443 for the Port. Leave the Backend field blank and click Save. Now login to your WebRTC account with Username 701. The Password is the one you obtained running show-701-pw. The IP Address (if required) is the address of your Wazo PBX.

blank

Implementing FollowMe Roaming with a CellPhone

In addition to ringing your SIP extension when incoming calls arrive, Wazo 17.02 can also ring your cellphone simultaneously. This obviously requires at least one outbound trunk. If that trunk provider also supports CallerID spoofing, then Wazo will pass the CallerID number of the caller rather than the DID associated with the trunk. Incredible PBX for Wazo comes with cellphone support for extension 702 ready to go. To enable it, access the Wazo Web GUI and go to IPBX → Services → Users → Incredible PBX and insert your Mobile Phone Number using the same dial string format associated with the trunk you wish to use to place the calls to your cellphone. You then can answer the incoming calls on either your cellphone or the registered SIP phone. If you answer on your cellphone, you will be prompted whether you wish to accept the call. If you press 1 after observing the CallerID, the caller will be connected. If you decline, the caller will be routed to the Wazo voicemail account of the extension.

Activating Voice Recognition for Wazo

Google has changed the licensing of their speech recognition engine about as many times as you change diapers on a newborn baby. Today’s rule restricts use to “personal and development use.” Assuming you qualify, the very first order of business is to enable speech recognition for your Wazo PBX. Once enabled, the Incredible PBX feature set grows exponentially. You’ll ultimately have access to the Voice Dialer for AsteriDex, Worldwide Weather Reports where you can say the name of a city and state or province to get a weather forecast for almost anywhere, Wolfram Alpha for a Siri-like encyclopedia for your PBX, and Lefteris Zafiris’ speech recognition software to build additional Asterisk apps limited only by your imagination. And, rumor has it, Google is about to announce new licensing terms, but we’re not there yet. To try out the Voice Dialer in today’s demo IVR, you’ll need to obtain a license key from Google. This Nerd Vittles tutorial will walk you through that process. Add your key to /var/lib/asterisk/agi-bin/speech-recog.agi on line 72.

Adding DISA Support to Your Wazo PBX

If you’re new to PBX lingo, DISA stands for Direct Inward System Access. As the name implies, it lets you make calls from outside your PBX using the call resources inside your PBX. This gives anybody with your DISA credentials the ability to make calls through your PBX on your nickel. It probably ranks up there as the most abused and one of the most loved features of the modern PBX.

There are three ways to implement DISA with Incredible PBX for Wazo. You can continue reading this section for our custom implementation with two-step authentication. There also are two native Wazo methods for implementing DISA using a PIN for security. First, you can dedicate a DID to incoming DISA calls. Or you can add a DISA option to an existing IVR. Both methods are documented in our tutorial on the PIAF Forum.

blank

We prefer two-step authentication with DISA to make it harder for the bad guys. First, the outside phone number has to match the whitelist of numbers authorized to use your DISA service. And, second, you have to supply the DISA password for your server before you get dialtone to place an outbound call. Ultimately, of course, the monkey is on your back to create a very secure DISA password and to change it regularly. If all this sounds too scary, don’t install DISA on your PBX.

1. To get started, edit /root/disa-xivo.txt. When the editor opens the dialplan code, move the cursor down to the following line:

exten => 3472,n,GotoIf($["${CALLERID(number)}"="701"]?disago1)  ; Good guy

2. Clone the line by pressing Ctrl-K and then Ctrl-U. Add copies of the line by pressing Ctrl-U again for each phone number you’d like to whitelist so that the caller can access DISA on your server. Now edit each line and replace 701 with the 10-digit number to be whitelisted.

3. Move the cursor down to the following line and replace 12341234 with the 8-digit numeric password that callers will have to enter to access DISA on your server:

exten => 3472,n,GotoIf($["${MYCODE}" = "12341234"]?disago2:bad,1)

4. Save the dialplan changes by pressing Ctrl-X, then Y, then ENTER.

5. Now copy the dialplan code into your Wazo setup, remove any previous copies of the code, and restart Asterisk:

cd /root
sed -i '\:// BEGIN DISA:,\:// END DISA:d' /etc/asterisk/extensions_extra.d/xivo-extrafeatures.conf
cat disa-xivo.txt >> /etc/asterisk/extensions_extra.d/xivo-extrafeatures.conf
/etc/init.d/asterisk reload

6. The traditional way to access DISA is to add it as an undisclosed option in an IVR that is assigned to one of your inbound trunks (DIDs). For the demo IVR that is installed, edit the ivr-1.conf configuration file and change the "option 0″ line so that it looks like this. Then SAVE your changes.

exten => 0,1(ivrsel-0),Dial(Local/3472@default)

7. Adjust the inbound calls route of one of your DIDs to point to the demo IVR by changing the destination to Customized with the following Command:

Goto(ivr-1,s,1)

A sample is included in the Wazo Snapshot. Here’s how ours looks for the Demo IVR:


blank

8. Now you should be able to call your DID and choose option 0 to access DISA assuming you have whitelisted the number from which you are calling. When prompted, enter the DISA password you assigned and press #. You then should be able to dial a 10-digit number to make an outside call from within your PBX.

SECURITY HINT: Whenever you implement a new IVR on your PBX, it’s always a good idea to call in from an outside number 13 TIMES and try every key from your phone to make sure there is no unanticipated hole in your setup. Be sure to also let the IVR timeout to see what result you get.

Implementing HA Redundancy with Wazo

With a business phone system, nothing is more important than never missing a call. Wazo’s High Availability (HA) option makes this a no-brainer, and it’s free! Just set up a second server either in the cloud or in your office and walk through our HA tutorial to set up the second server and activate HA. Even though located just across the border in Canada, OVH is hard to beat at $3.49 a month with 2 gigs of RAM and 10 gigs of storage. Vultr.com and Digital Ocean are also good candidates for a slave server, and the cost is still just $5 a month. Their 512MB platforms work fine with a drive cache, especially for a backup server. To get started, create a new Wazo platform using one of the highlighted links above. Be sure to use the same version of Wazo. Once the server is up and running, go to our Wazo HA tutorial and we’ll walk you through installing the NeoRouter Server and completing the Wazo setup. Be sure to configure Google Voice on the backup server before activating HA!



Test Drive Incredible PBX for Wazo

To give you a good idea of what to expect with Incredible PBX for Wazo, just pick up a phone and dial any toll-free number in the U.S. and Canada using a 1 prefix. We’ve also set up a sample IVR using voice prompts from Allison. Try it out from any phone on your PBX by dialing 4871 (IVR1):

Nerd Vittles Demo IVR Options
1 – Call by Name (say "Delta Airlines" or "American Airlines" to try it out)
2 – MeetMe Conference
3 – Wolfram Alpha (Coming Soon!)
4 – Lenny (The Telemarketer’s Worst Nightmare)
5 – Today’s News Headlines
6 – Weather Forecast (enter a 5-digit ZIP code)
7 – Today in History (Coming Soon!)
8 – Speak to a Real Person (or maybe just Lenny if we’re out)

What To Do and Where to Go Next?

Here are a boatload of projects to get you started exploring Wazo on your own. Just plug the keywords into the search bar at the top of Nerd Vittles to find numerous tutorials covering the topics or simply follow our links. Unless there is an asterisk (*) the components already are in place so do NOT reinstall them. Just read the previous tutorials to learn how to configure each component. Be sure to also join the PIAF Forum to keep track of the latest tips and tricks with Wazo. There’s a treasure trove of information that awaits.

Wazo and Incredible PBX Dial Code Cheat Sheets

Complete Wazo documentation is available here. But here are two cheat sheets in PDF format for Wazo Star Codes and Incredible PBX Dial Codes.

blank

blank

Published: Monday, January 30, 2017




blank
Need help with Asterisk? Visit the PBX in a Flash Forum.


 

Special Thanks to Our Generous Sponsors


FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

blankBOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

blankThe lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

blankVitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
 

blankSpecial Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
 



Some Recent Nerd Vittles Articles of Interest…

Deploying WebRTC with Incredible PBX for Wazo


blank

We continue our open source adventure with Wazo today by introducing Sylvain Boily’s latest masterpiece, WebRTC for Wazo. What started as a simple experiment has now become a full-featured WebRTC implementation that rivals any of the commercial alternatives. Did we mention it’s FREE! Better still, when you install the latest release of Incredible PBX for Wazo with all of its modules, the key components to support WebRTC are already in place thanks to Wazo Snapshots. If you have an earlier version of Incredible PBX for XiVO, we’ve already put together a tutorial on the PIAF Forum to walk you through installing WebRTC.

If you’re new to WebRTC, this slide from AT&T covers it all:

blank

Why WebRTC? Some of you may be asking, “What’s the big deal? Why would I want to deploy WebRTC?” The short answer is it eliminates the need to install and configure a proprietary softphone on every users’ desktop computer before they can communicate. Instead, all the user needs is a web browser that supports Real-Time Communications. By pointing their browser to https://phone.wazo.community/?serverIP=Wazo-ip-address, the user instantly gains a communications platform that’s as feature-rich as the most sophisticated softphone. Not only is it comparable to the dedicated clients of old, but there’s no associated cost nor the hassle of marrying a softphone to every user’s particular desktop operating system! And your web page could easily provide a directory of supported contact names and numbers as part of the user interface. In the case of the Wazo implementation, it does. To make a connection, all an end user needs is the latest Firefox or Chrome browser.


blank

WebRTC Admin Setup with Incredible PBX for Wazo

We’re getting ahead of ourselves. Let’s get WebRTC set up with Incredible PBX for Wazo so your users have something to play with. If you haven’t already installed the latest Incredible PBX for Wazo, start there. This puts all the pieces in place to support WebRTC. Write down the IP address of Incredible PBX for Wazo once you complete the install. You’ll need to provide this IP address to WebRTC users.

The other piece a WebRTC user will need is the random password assigned to their WebRTC extension. Incredible PBX comes with extension 701 preconfigured. You can create additional extensions as needed. Running the /root/show-701-pw script will display the password for the default 701 extension. If you’re missing that script, running the command below from the Linux CLI will display it. Or you can log into the Wazo CLI with your browser and go to IPBX → IPBX Settings → Users. Then edit the Incredible PBX 701 user account by clicking on the Pencil icon and write down the Password assigned to the 701 Wazo Client. By the way, this will be the same password assigned to the Default SIP/m1hqy5f3 Line for the Incredible PBX user.

export PGPASSWORD='proformatique'; psql -P pager=off -U asterisk -d asterisk \\
-c "SELECT secret FROM usersip WHERE id=1"

WebRTC User Setup with Incredible PBX for Wazo

The end user needs 3 pieces of information to get WebRTC running: the IP address of the Incredible PBX for Wazo PBX as well as the end user’s username and password for an extension to be used for WebRTC communications. With those 3 pieces in hand, the actual WebRTC setup is easy.

Here are the steps for the end-user to perform:

(1) Use the extension 701 user credentials as explained above or create a new user account and password choosing SIP (WebRTC) Protocol for the account type.

(2) Using Firefox or Chrome, go to the following link: https://phone.wazo.community/

(3) Before logging in, click on the Gear icon in the lower right corner and click the Pencil icon to edit your Settings. Fill in the public IP address of your Wazo server and specify 443 for the Port. Leave the Backend field blank and click Save.

(4) Login to your WebRTC account with Username 701. The Password is the one you obtained running /root/show-701-pw.

(5) When prompted, authorize WebRTC to use the camera and microphone on the user’s desktop computer.


blank

Once you’re logged in, at Enter number prompt, type in a phone number and click the Phone icon to dial.

There are loads of additional features in the Wazo WebRTC UI. Just follow your nose. Enjoy!

Published: Wednesday, October 26, 2016  Updated: Monday, May 29, 2017


blank
Need help with Asterisk? Visit the PBX in a Flash Forum.


 

Special Thanks to Our Generous Sponsors


FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

blankBOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

blankThe lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

blankVitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
 

blankSpecial Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
 



Some Recent Nerd Vittles Articles of Interest…

The Loneliest Number: One Remaining Open Source Distro for Asterisk 14

blank

With Asterisk® World just around the corner, this may come as a surprise to some of you. The Asterisk community that has championed open source software development for the past decade now has only one open source distro still standing. All the rest have morphed into closed source or commercial products. Can you guess which one is still carrying the Asterisk 14 open source banner? AsteriskNOW®? Nope. The FreePBX Distro®? Nope. Ombutel™? Nope. PBX in a Flash™? Nope. Elastix™? Nope. The answer is Wazo 17.01, and the latest Incredible PBX installer makes it a turnkey install in less than 15 minutes. If continuing the FOSS tradition is important to you, you really should take Wazo for a spin.


What Went Wrong? The answer is probably nothing. Reality simply set in. We all have to eat. As someone who has been involved in both the shareware and open source revolutions for more than 30 years, I can tell you that earning a living with open source software development is mostly a pipe dream. You can love open source software development and starve. Will some folks donate to the cause? Absolutely. Can you pay your mortgage from the proceeds? Not a chance. So you either find a "real job" that will pay the bills, or you change your business model and develop some sort of recurring revenue stream either through maintenance and support contracts, consulting, or hardware sales. Or you can write a technology blog and hope to find enough advertisers to keep the lights on. 🙂

We don’t mean to suggest that there’s anything wrong with commercial products per se. When it comes to VoIP telephony, commercial solutions make perfect sense. Businesses want their phones to ring when customers call. And the best way to achieve that is with commercially proven software and a support network that stands behind their products 24×7. So then it becomes a matter of comparison shopping for the best price and feature set. With this week’s release of the 3CX commercial product at zero cost to all PIAF users that participate in the PIAF Forum, that really should be a no-brainer. With a network of thousands of 3CX dealers worldwide for support, what have you got to lose? Zero.


Our New Year’s Resolution goes like this. For Nerd Vittles readers and for members of the PIAF community, we want you to have the best of both worlds. So we’ll be pushing our commercial provider to further enhance 3CX with features such as voice recognition and text-to-speech plus a robust API and programming language that makes expandability both simple and participatory. On the open source front, we will continue to work with the Wazo developers to make their platform even more flexible and feature rich than any FOSS product on the market. Please join us on both platforms as we continue our VoIP adventure.

In the meantime, come explore Wazo

Published: Tuesday, January 17, 2017


blank
Need help with Asterisk? Visit the PBX in a Flash Forum.


 

Special Thanks to Our Generous Sponsors


FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

blankBOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

blankThe lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

blankVitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
 

blankSpecial Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
 



Some Recent Nerd Vittles Articles of Interest…

2016, The Year of VoIP Choice: Meet Wazo and XiVO 16.15

blank

UPDATE: Wazo 17.01 has been officially released. The complete tutorial is available here.

And you thought the excitement was over for 2016. Well, not so fast. The core development team at XiVO has now forked the project so this will be the last XiVO-branded release until Wazo 16.16 hits the street. Nothing has changed except the name and a boatload of new features with more to come including a new GUI interface a little further down the road. And you’ll have a front row seat at Nerd Vittles. But lets save that discussion for coming weeks. For today, we’ll set the stage with the latest development release of XiVO featuring Incredible PBX and Asterisk® 14.1.2. Yes, there is an easy migration path for every existing XiVO server. That’s what the 2-minute xivo-upgrade is all about. In the meantime, anyone with the pioneering spirit can take a glimpse into the future. If you know XiVO, then you know that development releases normally are almost as stable as production releases because of their unique development methodology and enormous test suite which checks every change for naughty or nice. And, yes, the development team eats their own dog food! But please note that this is a Development Version of Wazo which means changes are happening regularly. The official release will be available in early December. For the pioneers installing now, be advised that there may be install hiccups from time to time as the developers migrate older components to Wazo. If an install fails for you, don’t get frustrated. Just wait 12 hours and try again.

Introducing the Opus Codec and Asterisk 14

We think you will enjoy this first release of Incredible PBX 14 featuring XiVO 16.15 and Asterisk 14 with integrated support for the Opus codec. If you haven’t heard of Opus, you’re in for a treat. You get the wideband voice quality of G.711U (ULAW) calls requiring 80-90kbps of bandwidth using only 16kbps. And, because it’s a variable bandwidth codec based upon your available Internet pipe, Opus can support narrowband calls with equivalent call quality to G.729 and Speex. Simply stated, you can squeeze FIVE wideband calls into the same bandwidth that one ULAW call used to consume. And, when you have the Internet capacity to support it, Opus calls can scale up to 128kbps for MP3-quality sound. Details.

blank

There’s more good news with Opus. XiVO’s WebRTC client now is preconfigured with the Opus codec when you deploy Incredible PBX 14. And, as if that weren’t enough, the WebRTC client with XiVO 16.15 now includes integrated voicemail support so you can play and delete voicemails without ever leaving the WebRTC client. See our WebRTC tutorial for more.

blank

Finally: A New CDR Reporting Module for XiVO

Here’s another important development that many have requested. The Incredible PBX 14 platform includes a terrific new CDR Reporting Module from Bart Fisher on the PIAF Forum. In the XiVO GUI, goto IPX → Call Management → Call Logs:

blank

FLITE TTS Implemented with New Voices

We’re pleased to announce that FLITE 1.4 is now included in Incredible PBX 14 builds on or after November 26. For the first time, you now have a choice of four different voices:

kal (American male)
rms (American male)
awb (Scottish male)
slt (American female)

While it’s a matter of personal taste, the RMS and SLT voices are dramatic improvements over the previous FLITE implementation. To change the voice, edit /etc/asterisk/flite.conf and replace voice=slt with your favorite. Then restart Asterisk. This post on the PIAF Forum includes dialplan code and will walk you through installing FLITE on existing servers. There’s more good news. You now can build your own FLITE voice for use with FLITE.

The Future Vision for Wazo

We don’t want to spill the beans on everything that lies ahead, but let’s talk briefly about the API Framework behind what will soon be the Wazo Telephony Business Engine. With Incredible PBX 14, you will note that you now have direct access to all available XiVO APIs with more to come. Using a browser, head over to https://ServerIPaddress/api/. A series of tutorials on how to use these APIs will be forthcoming now that we’ve gotten a few lessons from Sylvain Boily. Suffice it to say, the idea behind these APIs is that any developer will be able to quickly produce a customized web GUI for Wazo using nothing but API calls in conjunction with open source web development tools such as Bootstrap and Smarty. Think of it as OpenStack for the Telephony Cloud. And a new Wazo GUI is in the works as well. Here are a few examples to give you some idea of what’s possible in just a matter of hours:

blank

blank

blank

Rather than having a hard-coded GUI that uses spaghetti code to generate obscure Asterisk commands, you now will have a fully-documented development platform where the sky’s the limit. Think of it. You can actually contribute code back to the project while developing custom solutions for your organization. It’s what open source development is all about!

Update Your Address Book: New Wazo Links

Incredible PBX 14 for XiVO Installation Overview

Before we roll up our sleeves and walk you through the installation process, we wanted to provide a quick summary of the 10 Basic Steps in setting up Incredible PBX 14 for XiVO. By the way, the whole process takes less than an hour, half of that in the Cloud.

  1. Set Up Desired PBX Platform: Stand-alone PC, Virtual Machine, or Cloud-Based Server
  2. Run the Incredible PBX for XiVO installer and Activate All Options
  3. Set Up One or More SIP or Google Voice Trunks for Your PBX
  4. Tell XiVO Where to Direct Incoming Calls from Each Trunk
  5. Tell XiVO Which Trunk to Use for Every Outbound Calling Digit Sequence
  6. Set Up a SoftPhone or WebRTC Phone (or both)
  7. Decide Whether to Activate Simultaneous Ringing on your Cellphone
  8. Add Google Speech Recognition Key (if desired)
  9. Activating DISA with Incredible PBX for XiVO (if desired)
  10. Test Drive Incredible PBX for XiVO

1. Incredible PBX for XiVO Hardware Platform Setup

The first step is to choose your hardware platform and decide whether you want to babysit a server and network or leave those tasks to others. We’ve taken the guesswork out of the setups documented below. The last four options are cloud providers, each of whom provides a generous discount to let you kick the tires. So click on the links below to review the terms and our walkthrough of the setup process on each platform.

If your situation falls somewhere in between all of these, here’s a quick summary. For stand-alone systems and virtual machine platforms that you own (such as VirtualBox and VMware ESXi), download and install the 64-bit version of XiVO using the XiVO ISO. For most other virtual machine platforms in the Cloud, you’ll start by creating a 64-bit Debian 8 virtual machine with at least 1GB of RAM and a 20GB drive.

2. Running the Incredible PBX for XiVO Installer

Once you have your hardware platform up and running, the rest of the initial setup process is easy. Simply download and run the Incredible PBX 13 for XiVO installer. On some platforms, it first updates Debian 8 to current specs and reboots. Then log back in and rerun the installer a second time. You will be prompted whether to activate about a dozen applications for Incredible PBX. Choose Y for each option if you want to take advantage of the XiVO Snapshot with all components preconfigured. Otherwise, you’ll need to jump over to the original tutorial and manually configure all of the XiVO components.

cd /root
wget http://incrediblepbx.com/IncrediblePBX13-XiVO.sh
chmod +x IncrediblePBX13-XiVO.sh
./IncrediblePBX13-XiVO.sh

When you have completed the Incredible PBX 13 install, you then can log into your server as root and upgrade to Incredible PBX 14 with Asterisk 14 and the development version of XiVO/WAZO. Here are the steps:

xivo-dist xivo-dev
/etc/init.d/netfilter-persistent stop
xivo-upgrade
iptables-restart
# restore Incredible PBX module and ODBC configuration
cp -p /etc/asterisk/modules.conf /etc/asterisk/modules.conf-new
cp -p /etc/asterisk/res_odbc.conf /etc/asterisk/res_odbc.conf-new
cp -p /etc/asterisk/modules.conf.dpkg-old /etc/asterisk/modules.conf
cp -p /etc/asterisk/res_odbc.conf.dpkg-old /etc/asterisk/res_odbc.conf
# add Google Voice OAuth support for Asterisk 14
cd /usr/src
git clone https://github.com/sboily/asterisk-res-xmpp-oauth.git
cd asterisk*
make patch
make
make install
xivo-service restart
# put the Incredible PBX web add-ons back in place
cd /
wget http://incrediblepbx.com/incredible-nginx.tar.gz
tar zxvf incredible-nginx.tar.gz
rm -f incredible-nginx.tar.gz
ln -s /etc/nginx/locations/https-available/01_incrediblepbx /etc/nginx/locations/https-enabled/.
cd /etc/nginx
wget http://incrediblepbx.com/nginx-config.tar.gz
tar zxvf nginx-config.tar.gz
rm -f /etc/nginx/sites-enabled/default
/etc/init.d/nginx restart
sed -i 's|13.|14.|' /etc/pbx/.version

blank

While this may sound convoluted, there’s a reason for it. The WAZO Development Version is undergoing some major plumbing changes which affect the PostGreSQL database structure. Because Incredible PBX uses database snapshots to preconfigure a number of components, there would be major breakage if the Dev version database structure was different than the Incredible PBX snapshot. By performing an upgrade, we avoid the problem while preserving all of the Incredible PBX settings.

3. Setting Up SIP and Google Voice Trunks with XiVO

There are two steps in setting up trunks to use with Incredible PBX. First, you have to sign up with the provider of your choice and obtain trunk credentials. These typically include the FQDN of the provider’s server as well as your username and password to use for access to that server. Second, you have to configure a trunk on the Incredible PBX for XiVO server so that you can make or receive calls outside of your PBX. As with the platform tutorials, we have taken the guesswork out of the trunk setup procedure for roughly a dozen respected providers around the globe. In addition, XiVO Snapshots goes a step further and actually creates the trunks for you, minus credentials, as part of the initial Incredible PBX install.

For Google Voice trunks, log into your server as root and run ./add-gvtrunk. When prompted, insert your 10-digit Google Voice number, your Google Voice email address and OAuth 2 token. The native Google Voice OAuth tutorial explains how to obtain it.

blank

For the other providers, review the setup procedure below and then edit the preconfigured trunk for that provider by logging into the XiVO web GUI and choosing IPX → Trunk Management → SIP Protocol. Edit the setup for your provider (as shown above) and fill in your credentials and CallerID number in the General tab. Activate the trunk in the Register tab after again filling in your credentials. Save your settings when finished. No additional configuration for these providers is required when using the XiVO Snapshot.

4. Directing Incoming Calls from XiVO Trunks

Registered XiVO trunks typically include a DID number. With the exception of CallCentric, this is the number that callers would dial to reach your PBX. With CallCentric, it’s the 11-digit account number of your account, e.g. 17771234567. In the XiVO web GUI, we use IPX → Call Management → Incoming Calls to create inbound routes for every DID and trunk associated with your PBX. Two sample DIDs have been preconfigured to show you how to route calls to an extension or to an IVR. To use these, simply edit their settings and change the DID to match your trunk. Or you can create new incoming routes to send calls to dozens of other destinations on your PBX.

5. Routing Outgoing Calls from XiVO to Providers

Outgoing calls from extensions on your XiVO PBX must be routed to a trunk provider to reach call destinations outside your PBX. Outgoing call routing is managed in IPX → Call Management → Outgoing Calls. You tell XiVO which trunk provider to use in the General tab. Then you assign a Calling Digit Sequence to this provider in the Exten tab. For example, if NXXNXXXXXX were assigned to Vitelity, this would tell XiVO to send calls to Vitelity if the caller dialed a 10-digit number. XiVO has the flexibility to add and remove digits from a dialed number as part of the outbound call routing process. For example, you might want callers to dial 48NXXNXXXXXX to send calls to a Google Voice trunk where 48 spells "GV" on the phone keypad. We obviously don’t want to send the entire dial string to Google Voice so we tell XiVO to strip the first 2 digits (48) from the number before routing the call out your Google Voice trunk. We’ve included two examples in the XiVO Snapshot to get you started. Skype Connect (shown below) is an example showing how to strip digits and also add digits before sending a call on its way:

blank

6. Setting Up Softphone & WebRTC to Connect to XiVO

If you’re a Mac user, you’re lucky (and smart). Download and install Telephone from the Mac App Store. Start up the application and choose Telephone:Preference:Accounts. Click on the + icon to add a new account. To set up your softphone, you need 3 pieces of information: the IP address of your server (Domain), and your Username and Password. In the World of XiVO, you’ll find these under IPBX → Services → Lines. Just click on the Pencil icon beside the extension to which you want to connect. Now copy or cut-and-paste your Username and Password into the Accounts dialog of the Telephone app. Click Done when you’re finished, and your new softphone will come to life and should show Available. Dial the IVR (4871) to try things out. With Telephone, you can use over two dozen soft phones simultaneously on your desktop.

blank

For everyone else, we recommend the YateClient softphone which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for the XiVO Line. You’ll need the IP address of your server plus your Line username and password associated with the 701 extension. On the XiVO platform, do NOT use an actual extension number for your username with XiVO. Go to IPBX Settings → Lines to decipher the appropriate username and password for the desired extension. Click OK to save your entries.

blank

WebRTC allows you to use your Chrome or Firefox browser as a softphone. Extension 701 comes preconfigured for WebRTC access with Incredible PBX for XiVO. It shares the same password as the Line associated with extension 701, but the username is 701 rather than the username associated with the Line. You can decipher the password by accessing the XiVO Web GUI and then IPBX → Services → Users → Incredible PBX → XiVO Client Password. Or you can log into your server as root and run: /root/show-701-pw

To use WebRTC, you first need to accept the different SSL certificates associated with the WebRTC app. From your browser, go to the following site and click on each link to accept the certificates. Once you’ve completed this process, visit the Wazo WebRTC site. The Username is 701. The Password is the one you obtained above. The IP Address is the address of your XiVO PBX.

blank

7. Setting Up a CellPhone Extension with XiVO

In addition to ringing your SIP extension when incoming calls arrive, XiVO can also ring your cellphone simultaneously. This obviously requires at least one outbound trunk. If that trunk provider also supports CallerID spoofing, then XiVO will pass the CallerID number of the caller rather than the DID associated with the trunk. Incredible PBX for XiVO comes preconfigured with cellphone support for extension 701. To enable it, access the XiVO Web GUI and go to IPBX → Services → Users → Incredible PBX and insert your Mobile Phone Number using the same dial string format associated with the trunk you wish to use to place the calls to your cellphone. You can answer the incoming calls on either your cellphone or the phone registered to extension 701.

blank

8. Activating Voice Recognition for XiVO

Google has changed the licensing of their speech recognition engine about as many times as you change diapers on a newborn baby. Today’s rule restricts use to “personal and development use.” Assuming you qualify, the very first order of business is to enable speech recognition for your XiVO PBX. Once enabled, the Incredible PBX feature set grows exponentially. You’ll ultimately have access to the Voice Dialer for AsteriDex, Worldwide Weather Reports where you can say the name of a city and state or province to get a weather forecast for almost anywhere, Wolfram Alpha for a Siri-like encyclopedia for your PBX, and Lefteris Zafiris’ speech recognition software to build additional Asterisk apps limited only by your imagination. And, rumor has it, Google is about to announce new licensing terms, but we’re not there yet. To try out the Voice Dialer in today’s demo IVR, you’ll need to obtain a license key from Google. This Nerd Vittles tutorial will walk you through that process. Don’t forget to add your key to /var/lib/asterisk/agi-bin/speech-recog.agi on line 72.

9. Adding DISA Support to Your XiVO PBX

If you’re new to PBX lingo, DISA stands for Direct Inward System Access. As the name implies, it lets you make calls from outside your PBX using the call resources inside your PBX. This gives anybody with your DISA credentials the ability to make calls through your PBX on your nickel. It probably ranks up there as the most abused and one of the most loved features of the modern PBX.

There are three ways to implement DISA with Incredible PBX for XiVO. You can continue reading this section for our custom implementation with two-step authentication. There also are two native XiVO methods for implementing DISA using a PIN for security. First, you can dedicate a DID to incoming DISA calls. Or you can add a DISA option to an existing IVR. Both methods are documented in our tutorial on the PIAF Forum.

blank

We prefer two-step authentication with DISA to make it harder for the bad guys. First, the outside phone number has to match the whitelist of numbers authorized to use your DISA service. And, second, you have to supply the DISA password for your server before you get dialtone to place an outbound call. Ultimately, of course, the monkey is on your back to create a very secure DISA password and to change it regularly. If all this sounds too scary, don’t install DISA on your PBX.

1. To get started, edit /root/disa-xivo.txt. When the editor opens the dialplan code, move the cursor down to the following line:

exten => 3472,n,GotoIf($["${CALLERID(number)}"="701"]?disago1)  ; Good guy

2. Clone the line by pressing Ctrl-K and then Ctrl-U. Add copies of the line by pressing Ctrl-U again for each phone number you’d like to whitelist so that the caller can access DISA on your server. Now edit each line and replace 701 with the 10-digit number to be whitelisted.

3. Move the cursor down to the following line and replace 12341234 with the 8-digit numeric password that callers will have to enter to access DISA on your server:

exten => 3472,n,GotoIf($["${MYCODE}" = "12341234"]?disago2:bad,1)

4. Save the dialplan changes by pressing Ctrl-X, then Y, then ENTER.

5. Now copy the dialplan code into your XiVO setup, remove any previous copies of the code, and restart Asterisk:

cd /root
sed -i '\:// BEGIN DISA:,\:// END DISA:d' /etc/asterisk/extensions_extra.d/xivo-extrafeatures.conf
cat disa-xivo.txt >> /etc/asterisk/extensions_extra.d/xivo-extrafeatures.conf
/etc/init.d/asterisk reload

6. The traditional way to access DISA is to add it as an undisclosed option in an IVR that is assigned to one of your inbound trunks (DIDs). For the demo IVR that is installed, edit the ivr-1.conf configuration file and change the "option 0″ line so that it looks like this. Then SAVE your changes.

exten => 0,1(ivrsel-0),Dial(Local/3472@default)

7. Adjust the inbound calls route of one of your DIDs to point to the demo IVR by changing the destination to Customized with the following Command:

Goto(ivr-1,s,1)

A sample is included in the XiVO Snapshot. Here’s how ours looks for the Nerd Vittles XiVO Demo IVR:


blank

8. Now you should be able to call your DID and choose option 0 to access DISA assuming you have whitelisted the number from which you are calling. When prompted, enter the DISA password you assigned and press #. You then should be able to dial a 10-digit number to make an outside call from within your PBX.

SECURITY HINT: Whenever you implement a new IVR on your PBX, it’s always a good idea to call in from an outside number 13 TIMES and try every key from your phone to make sure there is no unanticipated hole in your setup. Be sure to also let the IVR timeout to see what result you get.

10. Test Drive Incredible PBX 14 for XiVO

To give you a good idea of what to expect with Incredible PBX for XiVO, we’ve set up a sample IVR using voice prompts from Allison. Give it a call and try out some of the features including voice recognition. Dial 1-843-606-0555.

Nerd Vittles Demo IVR Options
1 – Call by Name (say "Delta Airlines" or "American Airlines" to try it out)
2 – MeetMe Conference
3 – Wolfram Alpha (Coming Soon!)
4 – Lenny (The Telemarketer’s Worst Nightmare)
5 – Today’s News Headlines
6 – Weather Forecast (enter a 5-digit ZIP code)
7 – Today in History (Coming Soon!)
8 – Speak to a Real Person (or maybe just Lenny if we’re out)

What To Do and Where to Go Next?

Here are a Baker’s Dozen projects to get you started exploring XiVO on your own. Just plug the keywords into the search bar at the top of Nerd Vittles to find numerous tutorials covering the topics or simply follow our links. Note that all of these components already are in place so do NOT reinstall them. Just read the previous tutorials to learn how to configure each component. Be sure to also join the PIAF Forum to keep track of the latest tips and tricks with XiVO. There’s a treasure trove of information that awaits.

XiVO and Incredible PBX 14 Dial Code Cheat Sheets

Complete XiVO documentation is available here. But here are two cheat sheets in PDF format for XiVO Star Codes and Incredible PBX Dial Codes.

blank

blank

Published: Monday, November 28, 2016


blank
Need help with Asterisk? Visit the PBX in a Flash Forum.


Coming Soon to Nerd Vittles: The Autonomous Car




 

Special Thanks to Our Generous Sponsors


FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

blankBOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

blankThe lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

blankVitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
 

blankSpecial Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
 



Some Recent Nerd Vittles Articles of Interest…

2016, The Year of VoIP Choice: Introducing Ombutel


blank

Today we’re pleased to introduce our last (but not least) Unified Communications platform for 2016. Meet Ombutel, a terrific new GUI-based Asterisk® aggregation that was developed jointly by Telesoft and Xorcom, two familiar faces in the Asterisk community. We racked our brain trying to come up with a simple way to explain all of the things that Ombutel can do. We finally concluded "a picture [really] is worth a thousand words!"

Here’s some additional background on the Ombutel project:


blank

As with the other platforms we’ve introduced this year, we think the best way to get started is to install it yourself and kick the tires. For those familiar with FreePBX® or XiVO®, this will be a walk in the park. You set up an Extension and Device, configure a SIP Trunk to handle your calls, define an Inbound and Outbound Route to direct calls to their proper destination, load your extension credentials into a softphone or SIP phone, and you’re done. We were making calls after loading Ombutel into VirtualBox in less than 30 minutes.

To get started, download Ombutel from their web site. The ISO is approximately 1GB in size.

Installing Ombutel. Using the console interface in VirtualBox, we kicked off the install and went through the typical CentOS 7 setup choosing a language, choosing a keyboard, selecting an install destination, and setting up a root password. When the base install completes, you can log in as root to obtain Ombutel’s IP address. All of the remaining setup is completed using a browser pointed to Ombutel’s IP address. Set up an admin password for your server. Then login as admin with your new password. The Dashboard will display.

Creating an Extension. To get started, create an Extension and let Ombutel automatically populate an associated Device: (1) PBX → (2) Extensions → (3) Extensions. The only required entries are the (4) Extension Number and (5) Name. Be sure to set the NAT entry correctly for your network. Once you’ve completed the entries, click the Save button and then the red Reload icon. Notice the list icon in the right column of the window. Clicking on the List pull-down will show all of the extensions you created and allow you to edit them and decipher whether a particular extension is active.


blank

Adding a SIP Trunk. Adding Trunks is equally straight-forward: (1) PBX → (2) External → (3) Trunks. Then fill in the dozen items with your own credentials and settings. We’ve used a RingPlus SIP trunk as an example. NOTE: Be sure to set the From User field to your 10-digit RingPlus number even though this is not shown in the screenshot below. Once you’ve completed the entries, click the Save button and then the red Reload icon. As previously noted, the list icon in the right column will display all of the Trunks you’ve created.

blank

Configuring an Incoming Route. As with other PBXs, incoming routes define how calls from individual DIDs are routed once they arrive. The minimum requirements to set up an Incoming Route are a Description, a DID Pattern (usually the number associated with the DID), and a Destination for the incoming calls. Once you’ve completed the entries, click the Save button and then the red Reload icon. As previously noted, the list icon in the right column will display all of the Incoming Routes you’ve created and let you edit them.

blank

Configuring an Outgoing Route. As with other PBXs, outbound routes define how calls are routed out of your PBX based upon the dial string. You can choose one ore more trunks to associate with each Outbound Route. The dial string for each outbound route needs to be unique. Once you’ve completed the entries, click Save and then the red Reload icon.

blank

Just to Be Safe, Restart Asterisk. Ombutel still is fairly new code. We’ve found that a quirk occurs once in a while during all of the initial configuration. This typically can be squared away (e.g. extensions not connecting) by restarting Asterisk: /etc/init.d/asterisk restart.

Setting Up a Softphone to Connect to Ombutel. If you’re a Mac user, you’re lucky (and smart). Download and install Telephone from the Mac App Store. Start up the application and choose Telephone:Preference:Accounts. Click on the + icon to add a new account. To set up your softphone, you need 3 pieces of information: the IP address of your server (Domain), and your Username and Password. With Ombutel, choose PBX → Extensions → Extensions. Then click on the List icon and click on the extension to which you want to connect. Now copy or cut-and-paste your User Device number into Username and Password into Password on the Accounts dialog of the Telephone app. Click Done when you’re finished, and your new softphone will come to life and should show Available. Dial the same extension (7001) to test things out. With Telephone, you can use over two dozen soft phones simultaneously on your desktop.

blank

For everyone else, we recommend the YateClient softphone which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for the XiVO Line. As with the Telephone app above, you’ll need the IP address of your server plus your User Device and Password associated with the desired extension. Click OK to save your entries.

Some Thoughts on Network Security. We’ll have more to say about the Ombutel security model with FirewallD at another time. Suffice it to say, it’s not our preferred way of securing an Asterisk server. Here’s why. The following ports are all exposed by default:

1 - SIP udp and tcp 5060
2 - DNS tcp and udp 53
3 - NTP udp 123
4 - DHCP udp 67-68
5 - HTTP tcp 80
6 - SSH tcp 22
7 - RTP udp 10000-20000
8 - IAX2 udp 4569
9 - SwitchBoard tcp 4445
10 - mDNS udp 5353 224.0.0.251

You can check these for yourself in /etc/firewalld/services, and you can list the default firewall setup like this: firewall-cmd --list-all-zones. In fairness to Ombutel, their firewall design is no worse than what you will find with AsteriskNOW or the FreePBX Distro or Elastix. Incredible PBX and PBX in a Flash powered by 3CX take a different approach and don’t put all the responsibility for network security on the system administrator. We simply don’t have sufficient confidence in any Asterisk platform to risk exposing SIP, IAX2, HTTP, and SSH to the Big Bad Internet. For the time being until we can complete work on Incredible PBX for Ombutel, we recommend you run Ombutel behind a hardware-based firewall that does not expose these ports to the Internet for anyone and everyone.

Where To Go From Here. Ombutel has an awesome collection of video tutorials that should be the next stop in your Ombutel adventure. We’ve barely scratched the surface of this powerful platform, and there are still some missing pieces such as Google Voice. For the time being, you can use the Simonics SIP to Google Voice gateway to add this functionality. See this recent tutorial for some hints and a discount coupon.

blank

An Early Stocking Stuffer from Santa. We’ll leave you with a quick tutorial on how to install FLITE so that text-to-speech can be used in your Asterisk custom dialplan.1 In addition, we’re releasing the first of many Incredible PBX components for Ombutel with our Yahoo News application. After installing it, just dial *951 from any extension to listen to the latest Yahoo News Headlines. Both FLITE and the news application are GPL2 open source code. We’ll have more goodies to share with you in coming months.

yum -y upgrade
cd /usr/src
wget http://incrediblepbx.com/Asterisk-Flite-2.2-rc1-flite1.3.tar.gz
tar zxvf Asterisk-Flite*
cd Asterisk-Flite*
yum -y install gcc asterisk-devel
make
make install
make samples
ldconfig
/etc/init.d/asterisk restart
asterisk -rx "core show application like flite"
cd /
wget http://incrediblepbx.com/nv-news-ombutel.tar.gz
tar zxvf nv-news-ombutel.tar.gz
rm -f nv-news-ombutel.tar.gz
asterisk -rx "dialplan reload"

A final cautionary note to would-be Ombutel developers. You can’t use Feature Codes such as *951 as Destinations in the Ombutel GUI. Instead, you first will need to create a Custom Application as shown below. Then you can use Custom Applications → Yahoo News as a Destination in components such as IVRs and Inbound Routes. Enjoy!


blank

Published: Monday, November 21, 2016


blank
Need help with Asterisk? Visit the PBX in a Flash Forum.


 

Special Thanks to Our Generous Sponsors


FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

blankBOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

blankThe lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

blankVitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
 

blankSpecial Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
 



Some Recent Nerd Vittles Articles of Interest…

  1. Customizations to the dialplan can be made by creating files in /etc/asterisk/ombutel with the filename pattern “extensions__NN-*.conf” where NN defines the order in which to load the files. Numbers above 50 are strongly recommended! []