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The Most Versatile VoIP Provider: FREE PORTING

A Better Way to Deploy Incredible PBX in the Google Cloud

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Last week we introduced you to Incredible PBX 13-13.10 for the Google Cloud. This week we’ll take off the training wheels and show you how to deploy Incredible PBX in the Google Cloud in exactly 3 minutes using Google Cloud’s Image repository. And you can repeat the drill to deploy as many PBXs as you like at 3 minutes a pop. If you’re still cruising along on your $300 credit from Google, then your PBX should be cost-free for the entire first year. After that, you can decide which Google Cloud Machine Type best meets your requirements and those of your wallet. The free tier is an option, but don’t expect much more performance-wise than what you’d get with the original Raspberry Pi. As the saying goes, "It ain’t pretty, but it works." We would encourage you to move up to the Standard machine type for consistent performance.

Before we get started, let us just offer a little constructive criticism regarding Google’s methodology. If a developer builds an application as we have and wants to make it publicly available at no cost, wouldn’t it make sense to allow the developer to host the image in the Google Cloud (for a fee) so that other users could quickly deploy it on their own Google Cloud platforms? That apparently makes too much sense so Google requires you to jump through all sorts of hoops to use free software unless we’re willing to type in the Google email address of every user authorized to deploy the software. Sorry but we’ve got better ways to waste our time. This is the corporate mentality run amuck. Don’t Be Evil, Google. Remember?

So here’s the drill to get you to the place that Google already should have provided. Download the 3GB tarball image to your desktop from SourceForge. After you’ve created your Google Cloud account, create a Bucket (storage locker) on the platform to house your files and upload the tarball into your own Bucket. Next, transform the tarball into what Google calls an Image that can be used to quickly build VM Instances (5 minutes). Finally, start up the instance. The Incredible PBX installer will work its magic letting you set your passwords, and then your PBX platform is ready for use (3 minutes). The real install time is under 10 minutes, but Google has managed to turn it into a project of an hour or more depending upon the speed of your Internet connection. Our apologies, but it beats the tedium of last week’s methodology.

Downloading Incredible PBX for the Google Cloud

Unlike Google and to its eternal credit, SourceForge still hosts open source projects with tarballs of enormous size which can be downloaded at no cost other than what your Internet service provider may charge for bandwidth. Begin your Incredible PBX adventure by downloading the tarball image (3GB) which was designed specifically for the Google Cloud. Depending upon the speed of your Internet connection, this takes some time. Here’s the link.

Creating a Google Cloud Account

If you haven’t already done so, hop over to https://cloud.google.com/free and claim your $300 credit by signing up for a Google Cloud account.

Creating a Bucket in the Google Cloud

To begin, log in to your Google Cloud Console using your Google credentials. If you haven’t already done so, Create a Project from your Dashboard. This Project will house your Compute Engine VM Instances. In Plain English, a Google Cloud VM Instance is nothing more than an application that happens to run in the Google Cloud.

Next, click on the 3-bar image blank in the upper left corner of your Dashboard. This exposes the Navigation Menu. Scroll down to the STORAGE section and choose Storage -> Browser.

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Click on the CREATE BUCKET button. When the dialog window opens, Name your bucket something unique and creative in lower case letters. Fill in the rest of the form as shown and choose the Region in which you want to store your stuff. Then click Create.

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Uploading Incredible PBX into Google Cloud Bucket

Once you have created your Bucket, the Bucket Details dialog will open. Click on the Upload Files tab and choose the Incredible PBX tarball that you downloaded from SourceForge. Or you can simply drag the file to the area reserved for uploads in the dialog window.

Once the file upload completes, the Browser window will appear displaying your Bucket. You can click on the Bucket name to display the files in your Bucket which should now include the uploaded Incredible PBX tarball:

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Transforming Incredible PBX Tarball into an Image

Google Cloud can create Instances from Images, but not from tarballs in your Bucket. So the next step is to create an Image from the Incredible PBX tarball. Once that is done, you can delete the tarball and bucket from your Google Cloud platform so you don’t have to pay monthly storage fees. Up to this step is where Google could have handled setup transparently by simply allowing us to share our bucket with anonymous users without this knuckle drill, but…

So now we need to create an Image which will transform the Incredible PBX tarball into a format that can be used to create Instances.

Click on the Navigation Menu blank (the 3-bar image in the upper left corner of your Dashboard). Navigate to COMPUTE -> Compute Engine -> Images. Click CREATE IMAGE.

When the Create Image dialog opens, fill in the form as shown below and click on the Browse button to choose the Incredible PBX tarball from your Bucket. Then click Create.

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Creating an Instance from a Cloud Image

It takes about 5 minutes for Google Cloud to transmogrify the Incredible PBX tarball into an Image that can actually be used to create Instances. So be patient. Once your image has been created, it will appear at the top of the Images listing.

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Click on the checkbox blank to the left of the Image to select it as shown above. Then click CREATE INSTANCE at the top of the form.

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The Create Instance dialog window will appear. Fill in the form as shown above using a unique Name for your Instance. Adjust the Region to match your closest location. This choice may also affect the performance of your instance so picking the default is not a good idea if you want to stick with the freebie platform. Note that the Standard Machine Type (1vCPU) is selected by default. If you still have remaining credits, this won’t be a problem. Otherwise, you’ll have to pay about $25/month for this Machine Type level once your credits expire. We’ve had fair to good results using the Small Machine Type which costs under $15/month.

HINT #1: Never use the default zone for your PBX if you plan to use one of the shared vCPU machine types (micro or small). If you prefer the freebie which we strongly discourage because of performance issues, change the Machine Type to micro in the pull-down. Also note that the Boot Disk defaults to 10GB in size. This won’t work for long, and we’d recommend upping it to at least 20GB. Up to 30GB is provided at no cost using the micro Machine Type. Simply click the Change button to adjust the disk size. Once you’ve made your desired changes, click Create to build the Incredible PBX instance and bring it on line.

HINT #2: If you’re not going to move up to at least the small Machine Type, we would strongly urge you try one of our recommended $1/month VPS providers, all of whom offer considerably better performance at much less cost. In fact, you can bring up a redundant platform with a second VPS provider and still spend about the same money for a year that you would spend with a Google Cloud Standard VPS for one month.

While your Instance is being created and activated, navigate blank to COMPUTE -> Compute Engine -> VM Instances to display the status of your instances and to decipher the public IP address of your server. After you complete the next section, we’ll make a couple additional modifications using the Google Cloud Console by changing your public IP address from ephemeral (dynamic) to static and and adjusting the Google Cloud firewall. Delay making these changes at this time for the reason covered in the Word of Caution which follows.

A Word of Caution: Incredible PBX for the Google Cloud installs with a default root password of blank That obviously makes your running instance susceptible to compromise if someone else reads this article. So IMMEDIATELY after creating and activating a new Incredible PBX instance, make sure you complete the setup process in the next step during which you will be prompted to reset all of your passwords including the root password.

Completing the Incredible PBX Setup Process

Login to Incredible PBX as root using the default password at the public IP address of your instance using SSH or Putty. The Incredible PBX license agreement should display. If not, your server may have already been compromised. Accept the license agreement and enter very secure passwords for your server when prompted. Once the setup process finishes, reboot your server and wait about a minute for the reboot process to finish. Then log back into your server and allow the Automatic Update Utility to bring your server up to current specs. Once the pbxstatus screen displays, make sure everything is up and running. If not, wait another minute and rerun pbxstatus. Now issue the command user and make certain that you are the only root user on your server. If not, or if you didn’t see the license agreement when you first logged in, or if you couldn’t log in with the default root password, immediately shut down and destroy your instance and create a new one from your Google Cloud Image as documented in the previous section. TIP: If you see connection refused when you first attempt to log in, don’t be alarmed. Just count to 60 and try to log in again. The instance has to have time to boot up after activation before you can log in.

Finalizing Your Google Cloud Setup

Now that you’ve completed the Incredible PBX setup process, here are a couple of changes that need to be made using the Google Cloud GUI. First, you’ll need to permanently assign your IP address to your instance so that you don’t risk having Google change it when your server is rebooted. We also need to make a couple adjustments in the Google Cloud Firewall. Login again to the Google Cloud Console using your Google credentials.

From the Navigation Menu blank scroll down to the NETWORKING section and choose VPS Network -> External IP Addresses. Change the Type of your existing address to Static and Name it staticip. Next, choose Firewall Rules in the VPS Network section and click CREATE FIREWALL RULE. Fill in the template like the following leaving the other fields with their default entries. Then click CREATE.

Name: incrediblepbx-udp
Target Tags: udp-in
Source IP Range: 0.0.0.0/0
Protocols/Ports: udp: all

If you plan to use HTTPS with your server, you’ll also need to add another firewall rule similar to the existing default-allow-http rule. Simply change the Port to tcp:443 and Name it default-allow-https with a Target Tag of https-in.

CAUTION: Before these firewall rules will be activated for your instance, they also must be specified in the Network Tags section for your instance by adding the udp-in and https-in tags and restarting your instance.

It should be noted that Incredible PBX includes its own Travelin’ Man 3 firewall that manages a whitelist of IP addresses that are allowed ANY access to your server. So we will primarily use the firewall component of the Google Cloud instance to allow sufficient access to Incredible PBX to allow it to actually control server access.

Once you’ve verified that your instance is functioning properly, it’s safe to go back to your Bucket and delete it together with its contents. This will save you having to pay monthly storage fees even though they are quite reasonable.

Getting Started with Incredible PBX

Most of the configuration of your PBX will be performed using the web-based Incredible PBX GUI with its FreePBX® 13 GPL modules. Use a browser pointed to the IP address of your server and choose Incredible PBX Admin. Log in as admin with the password you configured above. HINT: You can always change it if you happen to forget it: /root/admin-pw-change

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Configuring Trunks with Incredible PBX

Before you can actually make and receive calls, you’ll need to add one or more VoIP trunks with providers, create extensions for your phones, and add inbound and outbound routes that link your extensions to your trunks. Here’s how a PBX works. Phones connect to extensions. Extensions connect to outbound routes that direct calls to specific trunks, a.k.a. commercial providers that complete your outbound calls to any phone in the world. Coming the other way, incoming calls are directed to your phone number, otherwise known as a DID. DIDs are assigned by providers. Some require trunk registration using credentials handed out by these providers. Others including Skyetel use the IP address of your PBX to make connections. Incoming calls are routed to your DIDs which use inbound routes telling the PBX how to direct the calls internally. A call could go to an extension to ring a phone, or it could go to a group of extensions known as a ring group to ring a group of phones. It could also go to a conference that joins multiple people into a single call. Finally, it could be routed to an IVR or AutoAttendant providing a list of options from which callers could choose by pressing various keys on their phone.

We’ve done most of the prep work for you with Incredible PBX. We’ve set up an Extension to which you can connect a SIP phone or softphone. We’ve set up an Inbound Route that, by default, sends all incoming calls from registered trunks to a Demo IVR. And we’ve built dozens of trunks for some of the best providers in the business. Sign up with the ones you prefer, plug in your credentials, and you’re done.

Unlike traditional telephone service, you need not and probably should not put all your eggs in one basket when it comes to telephone providers. In order to connect to Plain Old Telephones, you still need at least one provider. But there is nothing wrong with having several. And a provider that handles an outbound call (termination) need not be the same one that handles an incoming call (origination) and provides your phone number (DID). Keep in mind that you only pay for the calls you make with each provider so you have little to lose by choosing several. The PIAF Forum also has dozens of recommendations on VoIP providers.

With the preconfigured trunks in Incredible PBX, all you need are your credentials for each provider and the domain name of their server. Log into Incredible PBX GUI Administration as admin using a browser. From the System Status menu, click Connectivity -> Trunks. Click on each provider you have chosen and fill in your credentials including the host entry. Be sure to uncheck the Disable Trunk checkbox! Fill in the appropriate information for the Register String. Save your settings by clicking Submit Changes. Then click the red Apply Config button.

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Introducing Skyetel SIP Trunking for Incredible PBX

As frequent visitors already know, Skyetel is a Platinum Sponsor of Nerd Vittles and our open source projects including Incredible PBX. Their financial support keeps the lights on while all of our software remains free for the taking. Today we’re pleased to introduce a special new Skyetel offering for Nerd Vittles readers. If you loved BOGO deals at your favorite grocery store, then you’re going to love this new Skyetel offer which starts today. By signing up through this Nerd Vittles link, Skyetel will match any deposit originally made to your new account up to $250. For example, if you deposit $50, you’ll get $100 of SIP trunking service credit. Deposit $250, and you’ll get $500 of SIP trunking service credit. Basically, it’s half price service, and you get to choose how much you’d like. Skyetel also offers free porting of your DIDs for the first 60 days after you open your account plus a 10% reduction in your current origination rate and DID costs by presenting your last month’s bill.1 Complete details and configuration instructions on the Skyetel service are available in this tutorial. It only takes a minute or two to get up and running. Effective 10/1/2023, $25/month minimum spend at Skyetel is required.

Adding Skyetel Trunks to Incredible PBX

The Skyetel trunks were configured as part of the default install of Incredible PBX. All that’s required on your part is to sign up for Skyetel service to take advantage of the Nerd Vittles special offer. First, complete the Prequalification Form here. You then will be provided a link to the Skyetel site to complete your registration. Once you have registered on the Skyetel site and your account has been activated, open a support ticket and request the BOGO credit for your account by referencing the Nerd Vittles special offer. Greed will get you nowhere. Credit is limited to one per person/company/address/location. If you want to take advantage of the 10% discount on your current service, open another ticket and attach a copy of your last month’s bill. See footnote 1 for the fine print. If you have high call volume requirements, document these in your Prequalification Form, and we will be in touch. Easy Peasy!

Unlike many VoIP providers, Skyetel does not use SIP registrations to make connections to your PBX. Instead, Skyetel utilizes Endpoint Groups to identify which servers can communicate with the Skyetel service. An Endpoint Group consists of a Name, an IP address, a UDP or TCP port for the connection, and a numerical Priority for the group. For incoming calls destined to your PBX, DIDs are associated with an Endpoint Group to route the calls to your PBX. For outgoing calls from your PBX, a matching Endpoint Group is required to authorize outbound calls through the Skyetel network. Thus, the first step in configuring the Skyetel side for use with your PBX is to set up an Endpoint Group. A typical setup for use with Incredible PBX®, Asterisk®, or FreePBX® would look like the following:

  • Name: MyPBX
  • Priority: 1
  • IP Address: PBX-Public-IP-Address
  • Port: 5060
  • Protocol: UDP
  • Description: server1.incrediblepbx.com

To receive incoming PSTN calls, you’ll need at least one DID. On the Skyetel site, you acquire DIDs under the Phone Numbers tab. You have the option of Porting in Existing Numbers (free for the first 60 days after you sign up for service) or purchasing new ones under the Buy Phone Numbers menu option.

Once you have acquired one or more DIDs, navigate to the Local Numbers or Toll Free Numbers tab and specify the desired SIP Format and Endpoint Group for each DID. Add SMS/MMS and E911 support, if desired. Call Forwarding and Failover are also supported. That completes the VoIP setup on the Skyetel side. System Status is always available here.

Configuring a Skyetel Inbound Route

Because there is no SIP registration with Skyetel, incoming calls to Skyetel trunks will NOT be sent to the Default Inbound Route configured on your PBX because FreePBX treats the calls as blocked anonymous calls without an Inbound Route pointing to the 11-digit number of each Skyetel DID. From the GUI, choose Connectivity -> Inbound Routes -> Add Inbound Route. For both the Description and DID fields, enter the 11-digit phone number beginning with a 1. Set the Destination for the incoming DID as desired and click Submit. Reload the Dialplan when prompted. Place a test call to each of your DIDs after configuring the Inbound Routes.

With the included Incredible Fax add-on, you can enable Fax Detection under the Fax tab. And, if you’d like CallerID Name lookups using CallerID Superfecta, you can enable it under the Other tab before saving your setup and reloading your dialplan.

Configuring a Skyetel Outbound Route

If Skyetel will be your primary provider, you can use both 10-digit and 11-digit dialing to process outbound calls through your Skyetel account. It’s preconfigured to support Skyetel in Connectivity -> Outbound Routes -> Add Outbound Route. The recommended setup is shown below. Just add the CallerID Number you wish to associate with your outbound calls through Skyetel:

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Under the Dial Patterns tab, you’ll find the default rules as shown below. Adjust them to meet your own requirements.

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There are a million ways to design outbound calling schemes on PBXs with multiple trunks. One of the simplest ways is to use no dial prefix for the primary trunk and then use dialing prefixes for the remaining trunks.

Another outbound calling scheme would be to assign specific DIDs to individual extensions on your PBX. Here you could use NXXNXXXXXX with the 1 Prepend as the Dial Pattern with every Outbound Route and change the Extension Number in the CallerID field of the Dial Pattern. With this setup, you’d need a separate Outbound Route for each group of extensions using a specific trunk on your PBX. Additional dial patterns can be added for each extension designated for a particular trunk. A lower priority Outbound Route then could be added without a CallerID entry to cover extensions that weren’t restricted or specified.

HINT: Keep in mind that Outbound Routes are processed by FreePBX in top-down order. The first route with a matching dial pattern is the trunk that is selected to place the outbound call. No other outbound routes are ever used even if the call fails or the trunk is unavailable. To avoid failed calls, consider adding additional trunks to the Trunk Sequence in every outbound route. In summary, if you have multiple routes with the exact same dial pattern, then the match nearest to the top of the Outbound Route list wins. You can rearrange the order of the outbound routes by dragging them into any sequence desired.

Audio Issues with Skyetel

If you experience one-way or no audio on some calls, make sure you have filled in the NAT Settings section in the GUI under Settings -> Asterisk SIP Settings -> General. In addition to adding your external and internal IP addresses there, be sure to add your external IP address in /etc/asterisk/sip_general_custom.conf like the following example and restart Asterisk:

externip=xxx.xxx.xxx.xxx

If you’re using PJSIP trunks or extensions on your PBX, implement this fix as well.

Receiving SMS Messages Through Skyetel

Most Skyetel DIDs support SMS messaging. Once you have purchased one or more DIDs, you can edit each number and, under the SMS & MMS tab, you can redirect incoming SMS messages to an email or SMS destination of your choice using the following example:


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Sending SMS Messages Through Skyetel

We’ve created a simple script that will let you send SMS messages from the Linux CLI using your Skyetel DIDs. In order to send SMS messages, you first will need to create an SID key and password in the Skyetel portal. From the Settings icon, choose API Keys -> Create. Once the credentials appear, copy both your SID and Password. Then click SAVE.

Next, from the Linux CLI, issue the following commands to download the sms-skyetel script into your /root folder. Then edit the file and insert your SID, secret, and DID credentials in the fields at the top of the script. Save the file, and you’re all set.

cd /root
wget http://incrediblepbx.com/sms-skyetel
chmod +x sms-skyetel
nano -w sms-skyetel

To send an SMS message, use the following syntax where 18005551212 is the 11-digit SMS destination: sms-skyetel 18005551212 "Some message"

Configuring a Softphone for Incredible PBX

We’re in the home stretch now. You can connect virtually any kind of telephone to your new PBX. Plain Old Phones require an analog telephone adapter (ATA). With a cloud-based PBX, you need a standalone SIP device such as ObiHai’s OBi100 or OBi110 (if you have a phone line from Ma Bell to hook up as well). SIP phones can be connected directly so long as they have an IP address. These could be hardware devices or software devices such as the YateClient softphone. We’ll start with a free one today so you can begin making calls. You can find dozens of recommendations for hardware-based SIP phones both on Nerd Vittles and the PIAF Forum when you’re ready to get serious about VoIP telephony.

We recommend YateClient which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for the 701 extension on Incredible PBX. You’ll need the IP address of your server plus your extension 701 password. Choose Applications _> Extensions -> 701 and write down your SIP/IAX Password. You can also find it in /root/passwords.FAQ. Fill in the blanks using the IP address of your Server, 701 for your Username, and whatever Password you assigned to the extension when you installed Incredible PBX. Click OK to save your entries.

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Once you are registered to extension 701, close the Account window. Then click on YATE’s Telephony Tab and place some test calls to the numerous apps that are preconfigured on Incredible PBX. Dial a few of these to get started:

DEMO - Apps Demo
123 - Reminders
947 - Weather by ZIP Code
951 - Yahoo News
*61 - Time of Day
TODAY - Today in History

If you are a Mac user, another great no-frills softphone is Telephone. Just download and install it from the Mac App Store.

One cautionary note if you have multiple SIP softphones behind the same NAT-based router. Getting SIP packets routed back to the appropriate desktop machine can be problematic and typically results in missing audio on calls. The easy workaround is to set up the NeoRouter VPN on both your instance and each of your desktop computers. Then register the softphones to the NeoRouter private IP address of your instance. The NeoRouter client already is installed on your server, but you’ll need to set up a NeoRouter server somewhere and connect to it by running nrclientcmd.

Introducing the Incredible PBX Security Model

Incredible PBX includes one of the most secure turnkey PBX implementations on the planet. As configured, it is protected by both Fail2Ban and a hardened configuration of the IPtables Linux firewall. This release also includes Port Knocker for simple, secure access from any remote computer or smartphone. You can get up to speed on how the technology works by reading the Nerd Vittles tutorial. Your Port Knocker credentials are stored in /root/knock.FAQ together with activation instructions for your server and mobile devices. The NeoRouter VPN client also is included for rock-solid, secure connectivity to remote users. Read our previous tutorial for setup instructions. As configured, nobody can access your PBX without blank your credentials AND blank an IP address that matches the IP address of your server or the PC from which you installed Incredible PBX. You can whitelist additional IP addresses by running the command-line utility /root/add-ip. You can remove whitelisted IP addresses by running /root/del-acct. Incredible PBX is preconfigured to let you connect to many of the leading SIP hosting providers without additional firewall tweaking. The Google Cloud firewall adds an extra layer of protection.

The IPtables firewall is a complex piece of software. If you need assistance with configuring it, visit the PIAF Forum for some friendly assistance.

Incredible Backup and Restore

We’re pleased to introduce our latest backup and restore utilities for Incredible PBX. Running /root/incrediblebackup13 will create a backup image of your server in /tmp. This backup image then can be copied to any other medium desired for storage. To restore it to another Incredible PBX server, simply copy the image to a server running Asterisk 13 and the same version of the Incredible PBX GUI. Then run /root/incrediblerestore13. Doesn’t get much simpler than that.

Incredible PBX Automatic Update Utility

Every time you log into your server as root, Incredible PBX will ping the IncrediblePBX.com web site to determine whether one or more updates are available to bring your server up to current specs. We recommend you log in at least once a week just in case some new security vulnerability should come along.

In the meantime, we encourage you to sign up for an account on the PIAF Forum and join the discussion. In addition to providing first-class, free support, we think you’ll enjoy the camaraderie.

Upgrading to IBM Speech Engines

If you’ve endured Google’s Death by a Thousand Cuts with text-to-speech (TTS) and voice recognition (STT) over the years, then we don’t have to tell you what a welcome addition IBM’s new speech utilities are. We can’t say enough good things about the new IBM Watson TTS and STT offerings. With IBM’s services, you have a choice of free or commercial tiers. Let’s put the pieces in place so you’ll be ready to play with the Whole Enchilada.

Getting Started with IBM Watson TTS Service

We’ve created a separate tutorial to walk you through obtaining and configuring your IBM Watson credentials. Start there.

Next, login to your Incredible PBX server and issue these commands to update your Asterisk dialplan and edit ibmtts.php:

cd /var/lib/asterisk/agi-bin
./install-ibmtts-dialplan.sh
nano -w ibmtts.php

Insert your credentials in $IBM_username and $IBM_password. For new users, your $IBM_username will be apikey. Your $IBM_password will be the TTS APIkey you obtained from IBM. Next, verify that $IBM_url matches the entry provided when you registered with IBM. Then save the file: Ctrl-X, Y, then ENTER. Now reload the Asterisk dialplan: asterisk -rx "dialplan reload". Try things out by dialing 951 (news) or 947 (Weather) from an extension registered on your PBX.

Getting Started with IBM Watson STT Service

Now let’s get IBM’s Speech to Text service activated. Log back in to the IBM Cloud. Click on the Speech to Text app. Choose a Region to deploy in, choose your Organization from the pull-down menu, and select STT as your Space. Choose the Standard Pricing Plan. Then click Create. When Speech to Text Portal opens, click the Service Credentials tab. In the Actions column, click View Credentials and copy down your STT username and password.

Finally, login to your Incredible PBX server and issue these commands to edit getnumber.sh:

cd /var/lib/asterisk/agi-bin
nano -w getnumber.sh

Insert apikey as your API_USERNAME and your actual STT APIkey API_PASSWORD in the fields provided. Then save the file: Ctrl-X, Y, then ENTER. Update your Voice Dialer (411) to use the new IBM STT service:

sed -i '\\:// BEGIN Call by Name:,\\:// END Call by Name:d' /etc/asterisk/extensions_custom.conf
sed -i '/\\[from-internal-custom\]/r ibm-411.txt' /etc/asterisk/extensions_custom.conf
asterisk -rx "dialplan reload"

Now try out the Incredible PBX Voice Dialer with AsteriDex by dialing 411 and saying "Delta Airlines."

Using Gmail as a SmartHost for SendMail

Many Internet service providers including Google block email transmissions from downstream servers (that’s you) to reduce spam. The simple solution is to use your Gmail account as a smarthost for SendMail. Here’s how. Log into your server as root and issue the following commands:

cd /etc/mail
hostname -f > genericsdomain
touch genericstable
makemap -r hash genericstable.db < genericstable
mv sendmail.mc sendmail.mc.original
wget http://incrediblepbx.com/sendmail.mc.gmail
cp sendmail.mc.gmail sendmail.mc
mkdir -p auth
chmod 700 auth
cd auth
echo AuthInfo:smtp.gmail.com \\"U:smmsp\\" \\"I:user_id\\" \\"P:password\\" \\"M:PLAIN\\" > client-info
echo AuthInfo:smtp.gmail.com:587 \\"U:smmsp\\" \\"I:user_id\\" \\"P:password\\" \\"M:PLAIN\\" >> client-info
echo AuthInfo:smtp.gmail.com:465 \\"U:smmsp\\" \\"I:user_id\\" \\"P:password\\" \\"M:PLAIN\\" >> client-info
nano -w client-info

When the nano editor opens the client-info file, change the 3 user_id entries to your Gmail account name without @gmail.com and change the 3 password entries to your actual Gmail password. Save the file: Ctrl-X, Y, then ENTER.

Now issue the following commands:

chmod 600 client-info
makemap -r hash client-info.db < client-info
cd ..
make
service sendmail restart

Finally, send yourself a test message. Be sure to check your spam folder!

 echo "test" | mail -s testmessage yourname@yourdomain.com

Check mail success with: tail /var/log/mail.log. If you have trouble getting a successful Gmail registration (especially if you have previously used this Google account from a different IP address), try this Google Voice Reset Procedure. It usually fixes connectivity problems. If it still doesn’t work, enable Less Secure Apps using this Google tool.

Originally published: Monday, April 1, 2019


Continue Reading: Configuring Extensions, Trunks & Routes

Don't Miss: Incredible PBX Application User's Guide covering the 31 Whole Enchilada apps


blankSupport Issues. With any application as sophisticated as this one, you're bound to have questions. Blog comments are a difficult place to address support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forum. It's the best Asterisk tech support site in the business, and it's all free! Please have a look and post your support questions there. Unlike some forums, the PIAF Forum is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of users just like you. You won't have to wait long for an answer to your question.


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Need help with Asterisk? Visit the VoIP-info 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.
 



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  1. In the unlikely event that Skyetel cannot provide a 10% reduction in your current origination rate and/or DID costs, Skyetel will give you an additional $50 credit to use with the Skyetel service. []

Spring Is Sprung: Taking Incredible PBX to the Google Cloud

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Let’s chat about Google. We’ve had a love-hate relationship with Google for the past decade. For every good deed they do, they always manage to follow it up with a swift kick in the teeth… or lower. Business Insider recently catalogued all of the Google carnage over the years. And that ignores the train wreck that many VoIP users experienced with Google Voice. So we have mixed emotions about writing this column at all. But, suffice it to say, you won’t have to worry about Google’s Cloud Platform disappearing. It’s as well entrenched in the Google profit center as their advertising juggernaut.

I’ll begin with a story about a former neighbor of ours that was the IT Director at a major university. Maintaining their server farm with staff, labor, and hardware had simply become too expensive and too painful for the university to absorb so he made what at the time appeared to be a very brave decision. He decided to move all of the computing resources of the university to the Google Cloud. I haven’t spoken to him recently, but I can tell you the day it was completed was one of the happiest days of his life. Taking hardware acquisition, hardware maintenance, and facilities management out of the IT equation is great for your blood pressure. And the university actually has saved boatloads of money.

Is the Google Cloud right for everyone? Of course not. But you’ve got nothing to lose by trying it because Google is going to spot you $300 for the first year to get started. So we’d recommend you make the decision whether to continue AFTER you’ve spent the $300 you found lying on the sidewalk. Today we’ll show you how to build the always-free platform which probably will suffice for home users and small businesses in perpetuity. After your first year, the only charge would be a little chump change for bandwidth each month. If you decide not to use it as your PBX platform, it still would come in handy as a VPN server platform for an application such as NeoRouter. Pricing details here.

We want to start today by thanking Stewart Nelson on the DSLR Forum for his pioneering work on this beginning over a year ago. To start, hop over to https://cloud.google.com/free and claim your $300 by signing up for a Google Cloud account.

CAUTION: Before you embark on this adventure, we would encourage you to read through this article AND read our followup article which documented a much easier and simpler implementation scheme.

Creating a Google Cloud Instance for Incredible PBX

Once you have your account set up, it’s time to create your first project. Navigate to https://console.cloud.google.com. In the COMPUTE section of the dashboard, click Compute Engine -> VM Instances. Then click CREATE PROJECT and name it. Now click CREATE INSTANCE and Name it incrediblepbx. The instance name becomes the hostname for your virtual machine. If you want to remain in the Free Tier, choose f1-micro instance as the Machine Type and choose a U.S. Region (us-central1, us-east1 or us-west1). We strongly recommend installing your VPS using the N1-standard-1 as the Machine Type. It costs about 3 cents an hour and will save you several hours of tedious waiting. Once you complete the install, you then can shut down the server, downgrade to the f1-micro Machine Type, and restart your instance. For the Boot Disk, choose CentOS 6 and expand the disk storage to at least 20GB (30GB is available with the Free Tier). For the Firewall setting, enable HTTP and optionally HTTPS, if desired. Check your entries carefully and then click the Create button.

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When your virtual machine instance comes on line, jot down the assigned public IP address. We’ll need it in a minute. Now click on the SSH pull-down tab and choose Open in a Browser Window. Now we need to set a root password and adjust the SSH settings so that you can login from your desktop computer using SSH or Putty. This is important since the Incredible PBX installer will whitelist the IP address of your desktop PC as part of the setup process. You don’t want to lock yourself out of your virtual machine.

sudo passwd root
su root
nano -w /etc/ssh/sshd_config

When the editor opens the SSH config file, add the following entries. Then save the file and restart SSH: service sshd restart

PermitRootLogin yes
PasswordAuthentication yes

You now should be able to log in to your instance as root from your desktop computer using SSH or Putty. Test it to be sure: ssh root@server-IP-address

Before we leave the Google Cloud Dashboard, let’s make the assigned public IP address permanent so that it doesn’t get changed down the road. Keep in mind that, if you ever delete your instance, you also need to remove the assigned static IP address so you don’t continue to get billed for it. From Home on the Dashboard, scroll down to the NETWORKING section and choose VPS Network -> External IP Addresses. Change the Type of your existing address to Static and Name it staticip. Next, choose Firewall Rules in the VPS Network section and click CREATE FIREWALL RULE. Fill in the template like the following leaving the other fields with their default entries. Then click CREATE.

  1. Name: incrediblepbx-udp
  2. Target Tags: udp-in
  3. Source IP Range: 0.0.0.0/0
  4. Protocols/Ports: check udp: all

If you plan to use HTTPS with your server, you’ll also need to add another firewall rule similar to the existing default-allow-http rule. Simply change the Port to tcp:443 and Name it default-allow-https with a Target Tag of https-in.

CAUTION: Before these firewall rules will be activated for your instance, they also must be specified in the Network Tags section for your instance by adding the udp-in and https-in tags and restarting your instance.

It should be noted that Incredible PBX includes its own Travelin’ Man 3 firewall that manages a whitelist of IP addresses that are allowed ANY access to your server. So we will primarily use the firewall component of the Google Cloud instance to allow sufficient access to Incredible PBX to allow it to actually control server access.

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Installing Incredible PBX in the Google Cloud

If you’ve installed previous iterations of Incredible PBX, here is a thumbnail sketch of the install procedure. After logging into your server as root from a desktop PC using SSH or Putty, issue the following commands:

yum -y update
yum -y install net-tools nano wget tar
wget http://incrediblepbx.com/incrediblepbx-13-13-LEAN.tar.gz
tar zxvf incrediblepbx-13-13-LEAN.tar.gz
rm -f incrediblepbx-13-13-LEAN.tar.gz
# add swap file to your instance
./create-swapfile-DO
# kick off Phase I install
./IncrediblePBX-13-13.sh
# after reboot, kick off Phase II install
./IncrediblePBX-13-13.sh
# adjust TM3 firewall to block Google Cloud locals
sed -i 's|10.0.0.0/8|10.0.0.0/24|' /usr/local/sbin/iptables-custom
iptables-restart
# add Full Enchilada apps (see below)
./Enchilada-upgrade.sh
# add HylaFax/AvantFax (see below)
./incrediblefax13.sh
# after reboot, set passwords
./update-passwords
# set desired timezone
./timezone-setup
# fix permissions clobbered by Google Cloud install
chown -R asterisk:asterisk /var/lib/asterisk
amportal restart
# set up NeoRouter client, if desired
nrclientcmd
# check network speed
wget -O speedtest-cli https://raw.githubusercontent.com/sivel/speedtest-cli/master/speedtest.py
chmod +x speedtest-cli
./speedtest-cli

WebMin is also installed and configured as part of the base install. The root password for access is the same as your Linux root password. We strongly recommend that you not use WebMin to make configuration changes to your server. You may inadvertently damage the operation of your PBX beyond repair. WebMin is an excellent tool to LOOK at how your server is configured. When used for that purpose, we highly recommend WebMin as a way to become familiar with your Linux configuration.

Using the Incredible PBX 13-13 Web GUI

NOTE: If you plan to upgrade to the Whole Enchilada, you can skip this section. It’s for those that wish to roll their own PBX from the ground up.

Most of the configuration of your PBX will be performed using the web-based Incredible PBX GUI with its FreePBX 13 GPL modules. Use a browser pointed to the IP address of your server and choose Incredible PBX Admin. Log in as admin with the password you configured in the previous step. HINT: You can always change it if you happen to forget it.

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To get a basic system set up so that you can make and receive calls, you’ll need to add a VoIP trunk, create one or more extensions, set up an inbound route to send incoming calls to an extension, and set up an outbound route to send calls placed from your extension to a VoIP trunk that connects to telephones in the real world. You’ll also need a SIP phone or softphone to use as an extension on your PBX. Our previous tutorial will walk you through this setup procedure. Over the years, we’ve built a number of command line utilities including a script to preconfigure SIP trunks for more than a dozen providers in seconds. You’ll find links to all of them here.

Continue Reading: Configuring Extensions, Trunks & Routes

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Upgrading to Incredible PBX Whole Enchilada

There now are two more pieces to put in place. The sequence matters! Be sure to upgrade to the Whole Enchilada before you install Incredible Fax. If you perform the steps backwards, you may irreparably damage your fax setup by overwriting parts of it.

The Whole Enchilada upgrade script now is included in the Incredible PBX LEAN tarball. If you have an earlier release, you may need to download the Whole Enchilada tarball as documented below. Upgrading to the Whole Enchilada is simple. Log into your server as root and issue the following commands. Try issuing just the last command first to see if the enchilada upgrade script already is in place. Otherwise, execute all of the commands below. Be advised that the upgrade will overwrite all of your existing Incredible PBX setup including any extensions, trunks, and routes you may have created previously. You also will be prompted to reset all of your passwords as part of the upgrade.

cd /root
./Enchilada*

If you accidentally installed Incredible Fax before upgrading to the Whole Enchilada, you may be able to recover your Incredible Fax setup by executing the following commands. It’s worth a try anyway.

amportal a ma install avantfax
amportal a r

Installing Incredible Fax with HylaFax/AvantFax

You don’t need to upgrade to the Whole Enchilada in order to use Incredible Fax; however, you may forfeit the opportunity to later upgrade to the Whole Enchilada if you install Incredible Fax first. But the choice is completely up to you. To install Incredible Fax, log into your server as root and issue the following commands:

cd /root
./incrediblefax13.sh

After entering your email address to receive incoming faxes, you’ll be prompted about two dozen times to choose options as part of the install. Simple press the ENTER key at each prompt and accept all of the defaults. When the install finishes, make certain that you reboot your server to bring Incredible Fax on line. There will be a new AvantFax option in the Incredible PBX GUI. The default credentials for AvantFax GUI are admin:password; however, you first will be prompted for your Apache admin credentials which were set when you installed Incredible PBX 13-13 LEAN or the Whole Enchilada. Then you’ll be asked to change your AvantFax password.

Upgrading to IBM Speech Engines

If you’ve endured Google’s Death by a Thousand Cuts with text-to-speech (TTS) and voice recognition (STT) over the years, then we don’t have to tell you what a welcome addition IBM’s new speech utilities are. We can’t say enough good things about the new IBM Watson TTS and STT offerings. With IBM’s services, you have a choice of free or commercial tiers. Let’s put the pieces in place so you’ll be ready to play with the Whole Enchilada.

Getting Started with IBM Watson TTS Service

We’ve created a separate tutorial to walk you through obtaining and configuring your IBM Watson credentials. Start there.

Next, login to your Incredible PBX server and issue these commands to update your Asterisk dialplan and edit ibmtts.php:

cd /var/lib/asterisk/agi-bin
./install-ibmtts-dialplan.sh
nano -w ibmtts.php

Insert your credentials in $IBM_username and $IBM_password. For new users, your $IBM_username will be apikey. Your $IBM_password will be the TTS APIkey you obtained from IBM. Next, verify that $IBM_url matches the entry provided when you registered with IBM. Then save the file: Ctrl-X, Y, then ENTER. Now reload the Asterisk dialplan: asterisk -rx "dialplan reload". Try things out by dialing 951 (news) or 947 (Weather) from an extension registered on your PBX.

Getting Started with IBM Watson STT Service

Now let’s get IBM’s Speech to Text service activated. Log back in to the IBM Cloud. Click on the Speech to Text app. Choose a Region to deploy in, choose your Organization from the pull-down menu, and select STT as your Space. Choose the Standard Pricing Plan. Then click Create. When Speech to Text Portal opens, click the Service Credentials tab. In the Actions column, click View Credentials and copy down your STT username and password.

Finally, login to your Incredible PBX server and issue these commands to edit getnumber.sh:

cd /var/lib/asterisk/agi-bin
nano -w getnumber.sh

Insert apikey as your API_USERNAME and your actual STT APIkey API_PASSWORD in the fields provided. Then save the file: Ctrl-X, Y, then ENTER. Update your Voice Dialer (411) to use the new IBM STT service:

sed -i '\\:// BEGIN Call by Name:,\\:// END Call by Name:d' /etc/asterisk/extensions_custom.conf
sed -i '/\\[from-internal-custom\]/r ibm-411.txt' /etc/asterisk/extensions_custom.conf
asterisk -rx "dialplan reload"

Now try out the Incredible PBX Voice Dialer with AsteriDex by dialing 411 and saying "Delta Airlines." Check back next week for the Whole Enchilada apps tutorial.

Adding Skyetel Trunks to Incredible PBX

The Skyetel trunks were configured as part of the default install of Incredible PBX. All that’s required on your part is to sign up for Skyetel service and take advantage of the exclusive Nerd Vittles BOGO offer beginning April 1. Skyetel will match your original deposit of up to $250 which translates into as much as $500 of half-price SIP trunking service. Effective 10/1/2023, $25/month minimum spend required. First, complete the Prequalification Form here. You then will be provided a link to the Skyetel site to complete your registration. Once you have registered on the Skyetel site and your account has been activated, open a support ticket and request the BOGO credit for your account by referencing this Nerd Vittles special offer. Greed will get you nowhere. Credit is limited to one per person/company/address/location. If you want to take advantage of the 10% discount on your current service, open another ticket and attach a copy of your last month’s bill. See footnote 1 for the fine print.1 If you have high call volume requirements, document these in your Prequalification Form, and we will be in touch. Easy Peasy!

Unlike many VoIP providers, Skyetel does not use SIP registrations to make connections to your PBX. Instead, Skyetel utilizes Endpoint Groups to identify which servers can communicate with the Skyetel service. An Endpoint Group consists of a Name, an IP address, a UDP or TCP port for the connection, and a numerical Priority for the group. For incoming calls destined to your PBX, DIDs are associated with an Endpoint Group to route the calls to your PBX. For outgoing calls from your PBX, a matching Endpoint Group is required to authorize outbound calls through the Skyetel network. Thus, the first step in configuring the Skyetel side for use with your PBX is to set up an Endpoint Group. A typical setup for use with Incredible PBX®, Asterisk®, or FreePBX® would look like the following:

  • Name: MyPBX
  • Priority: 1
  • IP Address: PBX-Public-IP-Address
  • Port: 5060
  • Protocol: UDP
  • Description: server1.incrediblepbx.com

To receive incoming PSTN calls, you’ll need at least one DID. On the Skyetel site, you acquire DIDs under the Phone Numbers tab. You have the option of Porting in Existing Numbers (free for the first 60 days after you sign up for service) or purchasing new ones under the Buy Phone Numbers menu option.

Once you have acquired one or more DIDs, navigate to the Local Numbers or Toll Free Numbers tab and specify the desired SIP Format and Endpoint Group for each DID. Add SMS/MMS and E911 support, if desired. Call Forwarding and Failover are also supported. That completes the VoIP setup on the Skyetel side. System Status is always available here.

Configuring a Skyetel Inbound Route

Because there is no SIP registration with Skyetel, incoming calls to Skyetel trunks will NOT be sent to the Default Inbound Route configured on your PBX because FreePBX treats the calls as blocked anonymous calls without an Inbound Route pointing to the 11-digit number of each Skyetel DID. From the GUI, choose Connectivity -> Inbound Routes -> Add Inbound Route. For both the Description and DID fields, enter the 11-digit phone number beginning with a 1. Set the Destination for the incoming DID as desired and click Submit. Reload the Dialplan when prompted. Place a test call to each of your DIDs after configuring the Inbound Routes.

If you have installed the Incredible Fax add-on, you can enable Fax Detection under the Fax tab. And, if you’d like CallerID Name lookups using CallerID Superfecta, you can enable it under the Other tab before saving your setup and reloading your dialplan.

Configuring a Skyetel Outbound Route

If Skyetel will be your primary provider, you can use both 10-digit and 11-digit dialing to process outbound calls through your Skyetel account. From the GUI, choose Connectivity -> Outbound Routes -> Add Outbound Route. For the setup, we recommend the following using the CallerID Number you wish to associate with your outbound calls through Skyetel:

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Enter the Dial Patterns under the Dial Patterns tab before saving your outbound route. Here’s what you would enter for 10-digit and 11-digit dialing. If you want to require a dialing prefix to use the Skyetel Outbound Route, enter it in the Prefix field for both dial strings.

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There are a million ways to design outbound calling schemes on PBXs with multiple trunks. One of the simplest ways is to use no dial prefix for the primary trunk and then use dialing prefixes for the remaining trunks.

Another outbound calling scheme would be to assign specific DIDs to individual extensions on your PBX. Here you could use NXXNXXXXXX with the 1 Prepend as the Dial Pattern with every Outbound Route and change the Extension Number in the CallerID field of the Dial Pattern. With this setup, you’d need a separate Outbound Route for each group of extensions using a specific trunk on your PBX. Additional dial patterns can be added for each extension designated for a particular trunk. A lower priority Outbound Route then could be added without a CallerID entry to cover extensions that weren’t restricted or specified.

HINT: Keep in mind that Outbound Routes are processed by FreePBX in top-down order. The first route with a matching dial pattern is the trunk that is selected to place the outbound call. No other outbound routes are ever used even if the call fails or the trunk is unavailable. To avoid failed calls, consider adding additional trunks to the Trunk Sequence in every outbound route. In summary, if you have multiple routes with the exact same dial pattern, then the match nearest to the top of the Outbound Route list wins. You can rearrange the order of the outbound routes by dragging them into any sequence desired.

Audio Issues with Skyetel

If you experience one-way or no audio on some calls, make sure you have filled in the NAT Settings section in the GUI under Settings -> Asterisk SIP Settings -> General. In addition to adding your external and internal IP addresses there, be sure to add your external IP address in /etc/asterisk/sip_general_custom.conf like the following example and restart Asterisk:

externip=xxx.xxx.xxx.xxx

If you’re using PJSIP trunks or extensions on your PBX, implement this fix as well.

Receiving SMS Messages Through Skyetel

Most Skyetel DIDs support SMS messaging. Once you have purchased one or more DIDs, you can edit each number and, under the SMS & MMS tab, you can redirect incoming SMS messages to an email or SMS destination of your choice using the following example:


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Sending SMS Messages Through Skyetel

We’ve created a simple script that will let you send SMS messages from the Linux CLI using your Skyetel DIDs. In order to send SMS messages, you first will need to create an SID key and password in the Skyetel portal. From the Settings icon, choose API Keys -> Create. Once the credentials appear, copy both your SID and Password. Then click SAVE.

Next, from the Linux CLI, issue the following commands to download the sms-skyetel script into your /root folder. Then edit the file and insert your SID, secret, and DID credentials in the fields at the top of the script. Save the file, and you’re all set.

cd /root
wget http://incrediblepbx.com/sms-skyetel
chmod +x sms-skyetel
nano -w sms-skyetel

To send an SMS message, use the following syntax where 18005551212 is the 11-digit SMS destination: sms-skyetel 18005551212 "Some message"

Using Gmail as a SmartHost for SendMail

Many Internet service providers including Google block email transmissions from downstream servers (that’s you) to reduce spam. The simple solution is to use your Gmail account as a smarthost for SendMail. Here’s how. Log into your server as root and issue the following commands:

cd /etc/mail
hostname -f > genericsdomain
touch genericstable
makemap -r hash genericstable.db < genericstable
mv sendmail.mc sendmail.mc.original
wget http://incrediblepbx.com/sendmail.mc.gmail
cp sendmail.mc.gmail sendmail.mc
mkdir -p auth
chmod 700 auth
cd auth
echo AuthInfo:smtp.gmail.com \\"U:smmsp\\" \\"I:user_id\\" \\"P:password\\" \\"M:PLAIN\\" > client-info
echo AuthInfo:smtp.gmail.com:587 \\"U:smmsp\\" \\"I:user_id\\" \\"P:password\\" \\"M:PLAIN\\" >> client-info
echo AuthInfo:smtp.gmail.com:465 \\"U:smmsp\\" \\"I:user_id\\" \\"P:password\\" \\"M:PLAIN\\" >> client-info
nano -w client-info

When the nano editor opens the client-info file, change the 3 user_id entries to your Gmail account name without @gmail.com and change the 3 password entries to your actual Gmail password. Save the file: Ctrl-X, Y, then ENTER.

Now issue the following commands:

chmod 600 client-info
makemap -r hash client-info.db < client-info
cd ..
make
service sendmail restart

Finally, send yourself a test message. Be sure to check your spam folder!

 echo "test" | mail -s testmessage yourname@yourdomain.com

Check mail success with: tail /var/log/mail.log. If you have trouble getting a successful Gmail registration (especially if you have previously used this Google account from a different IP address), try this Google Voice Reset Procedure. It usually fixes connectivity problems. If it still doesn’t work, enable Less Secure Apps using this Google tool.

Finally, let's clean up Fail2Ban so you don't get bombarded by useless email messages. Issue the following commands and, when the editor opens, search for the nobody: line and change the destination from root to devnull. Save the file and then reload your aliases with the last command below:

sed -i 's|you@example.com|nobody@localhost|' /etc/fail2ban/jail.conf
nano -w /etc/aliases
newaliases

Continue Reading: A Better Way to Deploy Incredible PBX in the Google Cloud

Originally published: Tuesday, March 26, 2019


News Flash: Turn Incredible PBX into a Fault-Tolerant HA Platform for $1/Month

Continue Reading: Configuring Extensions, Trunks & Routes

Don't Miss: Incredible PBX Application User's Guide covering the 31 Whole Enchilada apps


blankSupport Issues. With any application as sophisticated as this one, you're bound to have questions. Blog comments are a difficult place to address support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forum. It's the best Asterisk tech support site in the business, and it's all free! Please have a look and post your support questions there. Unlike some forums, the PIAF Forum is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of users just like you. You won't have to wait long for an answer to your question.


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Need help with Asterisk? Visit the VoIP-info 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.
 



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  1. In the unlikely event that Skyetel cannot provide a 10% reduction in your current origination rate and/or DID costs, Skyetel will give you an additional $50 credit to use with the Skyetel service. []

Yowza! A Fault-Tolerant Incredible PBX Platform for $1/Mo.

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Not that you needed another reason to sign up for Skyetel’s triple-redundant SIP hosting platform, but we have a new one for you anyway. A fault-tolerant VoIP platform always has been the Holy Grail for many organizations. When your business depends on telephone calls, the very last thing you want to fail is your phone system. When most administrators think of fault-tolerant, high availability (HA) servers, the first word that comes to mind is expensive. For example, FreePBX® charges $1500/node just for HA software licenses. Using Skyetel trunks with Incredible PBX® 13-13.10, you now can deploy a fault-tolerant solution including the cloud-based secondary server for about a dollar a month. And, unlike the FreePBX design, the primary and secondary servers need not be colocated. Today we’ll get you up and running.

Many SIP providers require users to register their PBX with their service before you can make and receive calls. Skyetel does not.1 Instead, Skyetel relies upon the IP address of the PBX to determine whether to relay calls to and from your PBX. Dynamic IPs are supported. This is incredibly useful in designing a fault-tolerant PBX platform. If SIP registration is required with your SIP provider, then it obviously takes considerably more engineering effort to enable and disable the redundant server platform since you can’t be registered to two different servers using the same accounts at the same time. With Skyetel, that’s never a problem. Let’s quickly review the Skyetel setup process, and you’ll see why their design facilitates fault tolerance without spending a fortune on additional hardware and software. Effective 10/1/2023, $25/month minimum spend required.

Once your Skyetel account is set up, there are basically two steps to enable connections with your PBX. First, you create an Endpoint Group with the IP address of your PBX. Second, you purchase one or more DIDs and point each of them to the Endpoint Group you just created. With Incredible PBX platforms, the Skyetel trunk and firewall setup already is in place. All you need to do is create an Inbound Route for each of your Skyetel DIDs using its 11-digit phone number. Ten and eleven-digit outbound calls, by default, always are routed to Skyetel for processing. So how do we handle fault-tolerance on the Skyetel side? It’s easy. Just add the IP address of the secondary PBX to the Endpoint Group that you set up for your primary PBX. That’s it. When Skyetel can’t send inbound calls to your primary PBX, it now will send them to your secondary server.

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Now let’s address what has to happen at your end to bring up a fault-tolerant platform. We will assume that you already have set up and configured extensions, trunks, routes, and all desired bells and whistles on an Incredible PBX 13-13.10 platform somewhere. It really doesn’t matter where. We will also assume you have deployed SIP telephones for all of your users and registered them with your PBX. There are five simple steps to implement a fault-tolerant PBX architecture. First, you need a cloud-based Incredible PBX 13-13.10 server with static IP address and nightly-refreshed mirror image of your primary server. Second, SSH needs to be configured for SSH key-based authentication between the two servers. We’ll use rsync and SSH on the primary server to keep voicemails synchronized between your primary and secondary servers during the day. Third, we’ll add the IP address of the secondary PBX as the secondary registration IP address on each of your SIP phones. Fourth, you’ll need to copy over the firewall and whitelist rules from your primary server to your second server and restart IPtables after making certain that you’ve whitelisted the IP addresses of BOTH servers on the primary server. Fifth, we’ll add the IP address of the secondary PBX to the Endpoint Group of your primary PBX on the Skyetel site. That’s it. Once you’ve completed these five steps, your users will never miss a call. Since Skyetel doesn’t sit in the middle of the RTP stream on calls that are underway, callers may never even know their primary PBX failed.

So let’s address the worst case scenario. Your primary server has an unrecoverable system failure in the middle of the work day, and you have no backup. What do you lose? Not phone calls! And, assuming you’re making hourly dumps of your voicemails to the secondary PBX, you only would lose whatever voicemails arrived during the minutes preceding the system failure. If an hour is too painful in your operation, then make voicemail dumps more frequently. The only other data that would be lost is the CDR data since the last nightly backup was restored. If that’s too painful, that data also can be archived on a more frequent basis using a simple bash script. And, just to repeat, the total additional cost for this fault-tolerant platform is approximately $1 a month depending upon the VPS provider you choose, and we have a half dozen that we recommend AND USE at that price point. Let’s get started.

Building a Secondary Redundant Server

Start by signing up for a cloud-based OpenVZ VPS platform with one of our recommended providers. Server locations and special signup details are documented in our article. Average cost is about $1/month on an annual contract with 1Gbit port or *free 1Gbit port upgrade on request based upon LowEndBox offer. Protect yourself by paying with PayPal which gives you 6 months to dispute a charge if the provider happens to go belly up. NOTE: Performance is almost directly proportional to annual cost with our Tier 1 and Tier 2 providers.

ProviderRAMDiskBandwidthPerformance as of 12/1/19Cost
CrownCloud KVM (LA)1GB20GB +
Snapshot
1TB/month598Mb/DN 281Mb/UP
2CPU Core
$25/year
Best Buy!
Naranjatech KVM (The Netherlands)1GB20GB1TB/monthHosting since 2005
VAT: EU res.
20€/year w/code:
SBF2019
BudgetNode KVM (LA)1GB40GB RAID101TB/monthAlso available in U.K PM @Ishaq on LET before payment$24/year
FreeRangeCloud KVM (Ashburn VA, Winnipeg, Freemont CA)1GB20GB SSD3TB/monthPick EGG loc'n
Open ticket for last 5GB SSD
$30/year w/code:
LEBEGG30

If you can’t bear repeating this step in the event of a provider implosion, then consider one of the $5/month cloud platforms offered by Digital Ocean or Vultr.2 Ask yourself if it’s worth $4/month to eliminate the risk of having to spend an hour rebuilding your secondary PBX platform. Regardless of what you decide, here’s the drill. Set up a 64-bit CentOS 6 platform on your VPS. Since you already are familiar with the Incredible PBX 13-13 setup procedure, just follow last week’s quick-and-dirty checklist to set up your new secondary server. If your primary server also has the Whole Enchilada or Incredible Fax installed, we recommend you install the same components on your secondary server. If your secondary server is CentOS 7-based, issue the following command to assure that key-based authentication will work: chmod 711 /root. When you’re finished, write down the IP address of your secondary server.

Setting Up SSH Key-Based Authentication

In order to automate the process of loading updates from your primary server to your secondary one, we need a way to login between the servers without being prompted for a password. Key-based authentication provides that feature. Begin by whitelisting the IP addresses of BOTH your primary and secondary PBXs on BOTH of your servers. Using add-ip in /root, add server1 entry: ./addip server1 12.34.56.78. Then add server2 entry: ./addip server2 23.45.67.89. When prompted, choose the 0 option to whitelist all ports.

Next, store the IP addresses of server1 and server2 on your primary server so we can find them when we need them for future tasks:

echo 12.34.56.79 > /etc/pbx/server1
echo 23.45.67.89 > /etc/pbx/server2

Next, on your primary server, generate an SSH key-pair: ssh-keygen. Press Enter when prompted for the key’s file name. Press Enter when prompted (twice) for a passphrase.

Then copy the new key to your secondary server using the IP address of your secondary server. Enter the root password for your secondary server when prompted.

ssh-copy-id root@$(cat /etc/pbx/server2)

Now test logging into your secondary server using SSH from your primary server. You should not be prompted for a password. Close the secondary server SSH session by typing exit.

ssh root@$(cat /etc/pbx/server2)
exit

Restoring Backup Image to Secondary Server

We will automate these steps down the road, but let’s first test the procedure to make sure things work as expected. Begin by creating a /backup folder on both servers. Then install the latest backup and restore scripts on both of your servers:

mkdir /backup
cd /root
rm -f incrediblebackup
rm -f incrediblerestore
wget http://incrediblepbx.com/incrediblebackup13.tar.gz
tar zxvf incrediblebackup13.tar.gz
rm -f incrediblebackup13.tar.gz

On BOTH servers, verify that you have the latest backup and restore scripts. The following command should return the filenames of your backup and restore scripts:

grep PROCEEDNOW= /root/inc*

On BOTH servers, issue the following commands so that we can automate the backup and restore process without being prompted:

sed -i 's|PROCEEDNOW=false|PROCEEDNOW=true|' /root/incrediblebackup13
sed -i 's|PROCEEDNOW=false|PROCEEDNOW=true|' /root/incrediblerestore13

Finally, on both servers, make sure you have logged out and back in at least once so that the Automatic Update Utility has loaded the latest patches on both platforms.

Now let’s take a backup snapshot of your primary server: /root/incrediblebackup13

Copy the backup file to the secondary server at the server2 IP address below:

scp /backup/$(ls -t /backup | head -n 1) root@$(cat /etc/pbx/server2):/backup/.

Next, login to your secondary server as root using SSH. We want to restore the backup file that was just copied from the primary server. Keep in mind that our restore methodology overwrites files with the same names on the secondary server but it does not erase files on the secondary server which were not contained in the primary’s backup. With the exception of the /root directory, the latest iteration of Incredible Restore cleans out the other affected directories including voicemail files on the secondary server. This is especially important because voicemails may have been erased on the primary server since the last restore, and voicemail files get renumbered each time a voicemail is removed. Thus, failure to remove previous collections of voicemail files before restoring a new backup could produce a royal mess on the secondary server especially where the primary server processes numerous voicemails for multiple users every day. One other item of interest concerns primary servers behind NAT-based routers. Ordinarily, you would set the external IP address and local networks in SIP Settings to avoid NAT problems such as one-way audio. Since we may be copying a NAT-based backup to a public facing secondary PBX, we don’t want those settings carried over. We’ve addressed this by deleting the entries whenever you choose automated restores using the PROCEEDNOW=true flag. The full backup including these NAT settings is preserved if you execute a restore using the PROCEEDNOW=false flag. Finally, if you were using the backup and restore tools for normal server recovery, we obviously would want to include the /etc/crontab file in the restore. However, for fault tolerant deployments, we want to preserve separate cron jobs on the primary and secondary PBXs. We handle this using the PROCEEDNOW flag as described above. If you set it to true, we preserve /etc/crontab. Otherwise, we restore it from the backup file. Now restore the backup on the secondary PBX:

/root/incrediblerestore13 /backup/$(ls -t /backup | head -n 1) 

Although not recommended, we’ve been successful restoring a CentOS6-based primary server to a CentOS7-based secondary platform. Take a minute to actually examine the contents of the backup on the secondary server by logging into the FreePBX GUI with a browser. Be sure to check the extensions, trunks, and routes to verify that they match the entries on your primary server. If your primary server is behind a NAT-based router, check the SIP Settings to make certain the NAT entries for external IP address and localnets weren’t carried over to the public-facing secondary PBX.

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Refreshing Spool Data to Secondary Server with rsync

Now we want to test updating voicemails and recordings from your primary server to the secondary. Using a SIP phone, connect to *701 and leave yourself a new voicemail. Verify that it exists in /var/spool/asterisk/voicemail/default/701/INBOX. Simulate the procedure that will be used periodically to copy new voicemail messages and recordings to your secondary server. Execute the following commands to verify the new voicemail was transferred:

rsync -a -e "ssh" /var/spool/* root@$(cat /etc/pbx/server2):/var/spool
ls -all /var/spool/asterisk/voicemail/default/701/INBOX

Adding the Secondary Server to Skyetel

As noted above, we need to add the IP address of your secondary server to the Endpoint Group of your primary server on Skyetel. After logging into your Skyetel account, choose Endpoints -> Endpoint Groups and click the gear icon beside your primary server’s endpoint group. Add an additional entry to the Group with a Priority of 2. Your entries should look similar to this:

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Within a few minutes, the Endpoint Health dashboard should display three green dots for both of your servers blank signifying that all three Skyetel data centers can connect to both of your servers.

Adding Server2 Key-Based Authentication

On your secondary server, let’s set up key-based authentication to the primary PBX. Here are the commands to set it up and test it. You should get logged in and back out without a password prompt.

ssh-copy-id root@$(cat /etc/pbx/server1)
ssh root@$(cat /etc/pbx/server1)
exit

Choosing Compatible Phones for High Availability

Now that we have our HA platform in place, we need to configure your SIP phones to continue to work when the primary server fails and things switch over to the secondary PBX. For this to work, you’ll need SIP phones that are compatible with HA technology. Most are but some are not. Look through the registration menu on your SIP phone and enter the IP address of the secondary PBX in the field provided. Adjust the server timeout value to reflect the number of seconds your users can tolerate without a working phone. On the primary PBX, don’t forget to whitelist the public IP addresses associated with each of your SIP phones using add-ip! As for supported phones, here’s a hint. Most Digium, Snom, Yealink, Grandstream, Fanvil, and Aastra 6700i and 9000i series phones are safe bets. And here’s what a SIP extension setup would look like on Yealink’s popular T46G.

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Reminder: Do NOT use a private IP address (as shown) for the secondary SIP registration, and only use a private IP address for the primary PBX if both the PBX and the SIP phone are behind the same NAT-based router.

Migrating Travelin’ Man 3 Firewall to Secondary PBX

You’ll find it much easier to manage one firewall and then copy those firewall settings from your primary server to your secondary PBX. There are 3 sets of files that need to be copied from the primary PBX before restarting the secondary’s firewall: (1) iptables and ip6tables from /etc/sysconfig, (2) all iptables* files in /usr/local/sbin, and (3) *.iptables files in /root. Make certain you have whitelisted the IP addresses of the primary and secondary PBXs using /root/add-ip before issuing these commands on the primary server:

cd /etc/pbx
scp /etc/sysconfig/iptables root@$(cat server2):/etc/sysconfig/.
scp /etc/sysconfig/ip6tables root@$(cat server2):/etc/sysconfig/.
scp /usr/local/sbin/iptables* root@$(cat server2):/usr/local/sbin/.
scp /root/*.iptables root@$(cat server2):/root/.
ssh root@$(cat server2) "service iptables restart; service fail2ban restart"
ssh root@$(cat server2) "iptables-custom"

Automating the Daily Synchronization Process

We’ll use three cron scripts to keep the data on your primary and secondary servers in sync. These scripts automatically manage the backup sets by removing all but the latest backup on both servers. Keep in mind that, if you add new packages to your primary server, you also will need to add the same packages to your secondary server. Aside from that caveat, everything else should remain synchronized using these three simple scripts. On the primary server, we’ll use backup1restore2 to make the nightly backup and copy it to the secondary server, and we’ll use rsync-spool-1to2 to execute the hourly spool sync to catch changes in voicemails, recordings, and fax data. On the secondary server, we’ll use restore2backup1 to restore the backup from the primary server each night. Let’s first put the cron scripts in place, and then we’ll set up the cron jobs.

On the primary server, issue the following commands to install the two scripts:

cd /root
wget http://incrediblepbx.com/backup1restore2
chmod +x backup1restore2
wget http://incrediblepbx.com/rsync-spool-1to2
chmod +x rsync-spool-1to2

On the secondary server, issue the following commands to install the restore script:

cd /root
wget http://incrediblepbx.com/restore2backup1
chmod +x restore2backup1

Before setting up the cron entries, you have a few decisions to make. First, decide how often you wish to run the backup/restore scripts. If your PBX configuration rarely changes, you may not need to run the backup and restore scripts every night. Second, decide what time to run the backup and restore scripts. The restore script on the secondary server should always be run about 30 minutes AFTER the backup script runs on the primary server. The reason should be obvious. Remember to take into consideration the time zones of your two servers when making the time calculations. Third, decide how frequently to run the spool synchronization script from the primary server. This will depend upon how busy your primary server is and how much data (voicemails) you’re willing to lose in the event of a catastrophic failure that occurs after the last sync operation. There’s probably little need to run the synchronization script while everyone is sleeping unless you get frequent fax deliveries overnight. Finally, the backup of the primary server requires that Asterisk be shut down for at most a few minutes so schedule the backups during a time when there is no server activity on your primary server. Your test backup that we ran above should provide some idea of how long the operation will take on your PBX. Once you have tackled these issues, edit /etc/crontab on both your primary and secondary servers. Add the cron jobs to the end of the files after verifying server time on BOTH servers using date.

NOTE: We strongly recommend setting the time zone on the secondary server to match the primary server’s time zone using the /root/timezone-setup script.

On the primary server, here is a sample entry to run the backup script as the root user once a day at 2:15 am (local time on primary server):

15 2 * * * root /root/backup1restore2 >/dev/null 2>&1

On the secondary server, this entry will run the restore script once a day at 2:45 am assuming secondary server is in the same time zone as primary server:

45 2 * * * root /root/restore2backup1 >/dev/null 2>&1

On the primary server, this entry runs rsync spool script every hour on the half hour:

30 * * * * root /root/rsync-spool-1to2 >/dev/null 2>&1

Testing Fault Tolerance on Your Platform

Once you’ve put all the pieces in place, the easiest way to test the HA functionality is to shut down Asterisk on the primary PBX: amportal stop. Also disable the rsync cron job in /etc/crontab. Depending upon the timeout seconds you configured on your SIP phones, you should be able to make calls through the secondary server shortly. Don’t forget to restart Asterisk on your primary PBX: amportal start. Note also that you will need to shut down Asterisk on the secondary server for a few minutes after restoring the primary server. This will force all of your SIP phones to re-register with the primary server. And, in a real-world outage of considerable duration, it may be necessary to do a reverse-rsync of the spool directories from the secondary server (using the template below as an example) to assure that no voicemails and other spool files were lost during the outage. Then it’s safe to once again enable the rsync cron job on the primary server.

amportal stop
rsync -a -e "ssh" /var/spool/* root@$(cat /etc/pbx/server1):/var/spool
sleep 900
amportal start

Originally published: Thursday, March 7, 2019


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Need help with Asterisk? Visit the VoIP-info 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.
 



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  1. Skyetel is a platinum sponsor of Nerd Vittles and the open source projects of Ward Mundy & Associates, LLC. March is the final month to take advantage of the Nerd Vittles $50 usage credit when you sign up with Skyetel. []
  2. Digital Ocean and Vultr provide modest referral credits to Nerd Vittles for those that use our referral code. It in no way colors our recommendations regarding these two providers, both of whom we use extensively. []

In Search of a Better Mousetrap: Meet Incredible PBX 13-13.10

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There are lots of choices in the VoIP marketplace these days, and everyone’s requirements are different. So one size certainly doesn’t fit all. So let me tick off our most important criteria, and you can compare our list to your own. At the top of our list are reliability, stability, performance, feature set, and ease of deployment and use. Next would be cost of ownership which includes not only the cost of initial deployment but also the recurring costs for both the hardware and software platforms as well as support. Finally, we think ease of migration, backups, and upgradeability are important issues to consider. Keep in mind that proprietary phone systems were designed to operate for a decade or more with only minor software tweaks. While that doesn’t sell more product, it certainly provides less aggravation for end-users as well as those that have to deploy and maintain these systems. We believe CentOS 6 and Asterisk 13 are equally stable. From our experience working with large organizations, the only thing worse than swapping out a user’s desktop computer was replacing their telephone.

Since the early days of Asterisk®, FreePBX®, and PBX in a Flash™, the Achilles’ heel of all of these platforms has always been the difficulty in upgrading. The primary cause was the number of moving parts in any open source platform where literally thousands of developers are always "improving" things. It’s one of the real advantages of a closed source product such as 3CX® which has the luxury of making changes under the covers without ever exposing any weaknesses in prior releases of their software. With today’s introduction of Incredible PBX® 13-13.10, we are hopeful that those days are behind us.

The upgradeability improvements are fairly obvious. First, Incredible Backup and the FreePBX backup module have both matured into easy-to-use tools to take snapshots of your PBX to accommodate migration to other server platforms. Second, FreePBX now offers the Bulk Handler utility which provides snapshots in CSV (spreadsheet) format for extensions, DIDs, User Manager users and groups, contacts, and Conferences so that these settings can be easily imported and exported. While this doesn’t encompass every setting in Asterisk and FreePBX, it gives you a leg up in moving to newer releases. And, as with 3CX, the FreePBX developers have provided an upgrade migration tool at least for their proprietary distro.

Having said all of that, we would caution everyone with a functioning PBX to ask yourself these three questions before migrating to a newer platform. First, is my current PBX secure and reliable? Second, are there functions my current PBX doesn’t provide that my users actually need? Third, if/when my current hardware platform dies or my current cloud provider goes out of business, do I have a tested recovery option to get back into operation quickly? If your answers are yes, no, and yes, our strongest recommendation is SIT TIGHT.

If you flunked the test, then you owe it to yourself to give today’s release of Incredible PBX 13-13.10 a try. Why? Because it’s free and always will be. Because it’s incredibly easy both to configure and to use. Because it provides the most comprehensive feature set in the VoIP industry. And because it will provide years of worry-free telephone service with free support.

Introducing 2019 Edition of Incredible PBX

Today marks our third major release of our flagship Incredible PBX 13-13 platform featuring 70+ new FreePBX GPL modules plus integrated support for free SIP URI worldwide calling plus all the latest components for OSS Endpoint Manager making SIP phone deployment a breeze. There also are terrific new backup and restore utilities which make migration and restoration of Incredible PBX platforms a snap. Finally, we’ve incorporated Skyetel SIP trunking in the build. It literally makes configuration of outbound and incoming calling a one-minute process. On the Skyetel side, create an Endpoint Group pointing to the IP address of your PBX, order one or more DIDs and point them to the new EndPoint Group. Done. On the Incredible PBX side, add Inbound Routes specifying the 11-digit numbers of your Skyetel DIDs and point each of them to the desired destination for incoming calls. Done. Outbound calls are automatically configured to use your Skyetel account. Our complete Skyetel tutorial is available here and includes an exclusive BOGO usage credit of up to $250.1 You now can deploy a fault-tolerant, high availability (HA) platform using Incredible PBX 13-13.10 and Skyetel. Here’s how. Effective 10/1/2023, $25/month minimum spend required.

Installing Incredible PBX 13-13.10 Locally

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As with prior releases, the Incredible PBX ISO installer eases deployment on stand-alone hardware. For those that are new to Incredible PBX, head over to our detailed tutorial to get started. New releases for the Raspberry Pi and Ubuntu 18.04.2 are also available now.

Installing Incredible PBX 13-13.10 in the Cloud

Start by signing up for a cloud-based OpenVZ VPS platform with one of our recommended providers. Server locations and special signup details are documented in our article. Average cost is about $1/month on an annual contract with 1Gbit port or *free 1Gbit port upgrade on request based upon LowEndBox offer. Protect yourself by paying with PayPal which gives you 6 months to dispute a charge if the provider happens to go belly up. NOTE: Performance is almost directly proportional to annual cost with our Tier 1 and Tier 2 providers.

ProviderRAMDiskBandwidthPerformance as of 12/1/19Cost
CrownCloud KVM (LA)1GB20GB +
Snapshot
1TB/month598Mb/DN 281Mb/UP
2CPU Core
$25/year
Best Buy!
Naranjatech KVM (The Netherlands)1GB20GB1TB/monthHosting since 2005
VAT: EU res.
20€/year w/code:
SBF2019
BudgetNode KVM (LA)1GB40GB RAID101TB/monthAlso available in U.K PM @Ishaq on LET before payment$24/year
FreeRangeCloud KVM (Ashburn VA, Winnipeg, Freemont CA)1GB20GB SSD3TB/monthPick EGG loc'n
Open ticket for last 5GB SSD
$30/year w/code:
LEBEGG30

If you’ve installed previous iterations of Incredible PBX, here is a thumbnail sketch of the install procedure. Begin by installing a minimal CentOS 6 or CentOS 7 (64-bit) platform. Then log into your server as root and issue the following commands:

passwd
yum -y update
yum -y install net-tools nano wget tar
wget http://incrediblepbx.com/incrediblepbx-13-13-LEAN.tar.gz
tar zxvf incrediblepbx-13-13-LEAN.tar.gz
rm -f incrediblepbx-13-13-LEAN.tar.gz
# to add swap file on non-OpenVZ cloud platforms
./create-swapfile-DO
# kick off Phase I install
./IncrediblePBX-13-13.sh
# after reboot, kick off Phase II install
./IncrediblePBX-13-13.sh
# add Full Enchilada apps, if desired
./Enchilada-upgrade.sh
# add HylaFax/AvantFax, if desired
./incrediblefax13.sh
# set passwords
./update-passwords
# set desired timezone
./timezone-setup
# remember to enable TUN/TAP if using VPS Control Panel
# reconfigure PortKnocker if installing on an OpenVZ platform
echo 'OPTIONS="-i venet0:0"' >> /etc/sysconfig/knockd
service knockd restart
# fix pbxstatus for NeoRouter VPN support, if desired
cd /usr/local/sbin
sed -i "s|cat /etc/hostip|cat /etc/hostip \\| cut -f 3 -d ' ' |" pbxstatus
# set up NeoRouter client, if desired
nrclientcmd
# check network speed
wget -O speedtest-cli https://raw.githubusercontent.com/sivel/speedtest-cli/master/speedtest.py
chmod +x speedtest-cli
./speedtest-cli

Planning Ahead for That Rainy Day

If you haven’t already learned the hard way, let us save you from a future shock. Hardware fails. All of it. So spend an extra hour now so that you’ll be prepared when (not if) disaster strikes. First, once you have your new PBX configured the way you plan to use it, make a backup of your PBX by running the Incredible Backup script: /root/incrediblebackup13

Copy down the name of the backup file that was created. You’ll need it in a few minutes.

Second, build yourself a VirtualBox platform on your desktop PC. Then download, install, and run the new Incredible PBX 13-13.10 vbox image from SourceForge. Our tutorial makes it easy.

Next, create a /backup folder on your VirtualBox PBX and copy the backup file from your main server to your VirtualBox server and restore it after logging in to VirtualBox PBX as root:

mkdir /backup
scp root@main-pbx-ip-address:/backup/backup-file-name.tar.gz /backup/.
/root/incrediblerestore13 /backup/backup-file-name.tar.gz

Verify that everything looks right by using a browser to access and review the settings in your new VirtualBox PBX. At a minimum, verify extensions, trunks, and routes. Sleep well.


News Flash: Turn Incredible PBX into a Fault-Tolerant HA Platform for $1/Month

Adding Asterisk Apps: Introducing Incredible PBX 13-13 Whole Enchilada

Continue Reading: Configuring Extensions, Trunks & Routes

Don’t Miss: Incredible PBX Application User’s Guide covering the 31 Whole Enchilada apps

Check out the new Incredible PBX 13-13 ISO. Complete tutorial available here.

Originally published: Friday, March 1, 2019


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Need help with Asterisk? Visit the VoIP-info 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.
 



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  1. Skyetel is a Platinum Sponsor of Nerd Vittles and open source projects of Ward Mundy & Associates, LLC. []

Big Kahuna: 70 New FreePBX GPL Modules for Incredible PBX


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We don’t change the mix of FreePBX® GPL modules in Incredible PBX® 13-13 often although you can easily add or update any of particular interest at any time using the gpl-install scripts included in the distribution. So we’re excited to introduce the 2019 collection of 70 FreePBX GPL modules for those that want to keep their Asterisk® PBX platform loaded with the latest and greatest. We’ve included a batch installer which means ALL of the existing modules get updated with the latest releases from GitHub. Depending upon the speed of your Internet connection, it’s a 5 or 10-minute procedure. Schedule it for a time when the PBX is idle.

Upgrading the FreePBX GPL Modules

The upgrade procedure couldn’t be easier. Log into your server as root. We recommend you make a backup first using the incrediblebackup script in /root. Next, make sure you have at least 50MB of free disk space: df -h. Then issue these commands and have a cup of coffee:

cd /tmp
wget http://incrediblepbx.com/modules13.tar.gz
tar zxvf modules13.tar.gz
rm -f modules13.tar.gz
cd modules13
./update-modules.sh


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When you return, your Incredible PBX 13-13 server will be all shiny and new. You can review the license terms for each module by referencing the table below and calling up the GPL license provisions with a browser pointed to http://server-IP/admin/licenses.

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FreePBX GPL Modules Documentation

The FreePBX Dev Team has generously provided excellent documentation for all of the modules. We have arranged them in the same order as the GUI’s menus for ease of use.

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Admin Modules

Administrators Module
Asterisk CLI Module
Asterisk Phonebook Module
Backup and Restore Module
Blacklist Module
Bulk Handler
CID Superfecta
CallerID Lookup Sources
Certificate Management Module
Config Edit
Custom Destinations Module
Custom Extensions Module
Feature Codes Module
Module Admin Module
Phone Restart Module
Presence State Module
REST API
Sound Languages
System Recordings Module
User Management Module

Applications Modules

Announcements Module
Call Flow Control Module
Call Recording Module
Callback Module
Conferences Module
DISA Module
Directory Module
Extensions Module
Follow Me Module
IVR Module
Languages Module
Misc Applications Module
Misc Destinations Module
Paging and Intercom Module
Parking Module
Queue Priorities Module
Queues Module
Ring Groups Module
Set CallerID Module
Text to Speech Module
Time Conditions Module
Time Groups Module
Voicemail Blasting Module
Wakeup Calls Module

Connectivity Modules

DAHDI Channel DIDs
Inbound Routes Module
OSS End Point Manager (Disabled)
Outbound Routes Module
Trunks Module

Dashboard

PBX System Status

Reports Modules

Asterisk Info Module
Asterisk Logfiles
CDR Reports Module
Call Event Logging (CEL) Module
Print Extensions
Rest API Report
Weak Password Detection

Settings Modules

Advanced Settings
Asterisk IAX Settings
Asterisk Logfile Settings
Asterisk Manager Interface
Asterisk SIP Settings
Extension Settings
Fax Configuration
Music on Hold Module
Pin Sets
Route Congestion Messages
Text to Speech Engines Module
Voicemail Admin

Third Party Addons

AsteriDex
Reminders

UCP

User Control Panel

Installing OSS Endpoint Manager

If you have dozens of SIP phones to configure, then you’ll appreciate Andrew Nagy’s terrific OSS Endpoint Manager Module. Here’s how to install it once your Incredible PBX 13-13 server is updated with the new modules above:

cd /
wget http://incrediblepbx.com/epm.tar.gz
tar zxvf epm.tar.gz
./install-epm.sh

You will also need to install and configure a TFTP server. Here’s the CentOS procedure:

cd /root
wget http://incrediblepbx.com/setup-tftp
chmod +x setup-tftp
./setup-tftp

Pay particular attention to the firewall instructions which display at the end of the TFTP install procedure. Complete documentation for OSS Endpoint Manager is available here. Enjoy!

Originally published: Monday, February 18, 2019


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Need help with Asterisk? Visit the VoIP-info 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.
 



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UC on Steroids: Incredible PBX for Issabel Joins the Cloud


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We’re pleased to introduce the 2019 edition of Incredible PBX® for Issabel featuring new VPS cloud provider support and one-minute setup for Skyetel SIP trunking. One of the limitations of Issabel 4 has been the required use of the ISO installer to deploy Asterisk® 13. This 2019 release addresses that limitation and lets you do a fully scripted install using one of our four recommended $7 to $15 a year VPS cloud providers.

This new release includes our next generation Incredible PBX 13 platform with a preconfigured Travelin’ Man 3 firewall, additional text-to-speech engines (FLITE, GoogleTTS, PicoTTS, and IBM TTS), voice recognition with IBM’s state-of-the-art STT engine, turnkey trunks and extensions, SMS messaging, telephone reminders, turnkey fax support, an AsteriDex phone book with both voice and speed dialing, Wolfram Alpha, sample ODBC apps, and a boatload of dialplan code and AGI scripts to help anyone wanting to learn how to develop custom applications with Asterisk. This is one fantastic UC platform!


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Installing Issabel on a Cloud-Based VPS Platform

If you wish to install Issabel 4 on a cloud-based OpenVZ server, here’s the drill. Start by creating a CentOS 7/64 platform. Once the platform is ready, log in to your server as root and immediately change your root password. Then execute the remaining commands in the order listed below. Don’t worry if you cannot access the Issabel web GUI when the install finishes. We’ll fix this up during the Incredible PBX install shortly. Now jump down to the Incredible PBX installation steps to continue.

passwd
yum -y install wget nano
wget -O - http://repo.issabel.org/issabel4-netinstall.sh | bash
yum -y erase asterisk
yum -y install asterisk13
reboot

Installing Issabel with Asterisk 13.22.0 from ISO

If you’re using your own hardware or a platform that lets you upload an ISO and deploy, begin by downloading the October 2, 2018 Issabel ISO from SourceForge. On the platform of your choice, install Issabel 4 specifying your Keyboard and Installation Destination with Asterisk 13 as your Software Selection. Add the Sangoma WANPIPE component if desired. Set your Root password and have a cup of coffee. After a reboot, you’ll be prompted to set your MySQL/MariaDB root password (must be passw0rd with a zero) and the admin password of your choice to login to the Issabel web GUI. Be sure to use the new October 2018 Issabel ISO for the base Issabel install. It includes support for Asterisk 13.22.0. We will update things from there as part of the new Incredible PBX install below.

Installing Issabel with VirtualBox

For those using VirtualBox, we’ve uploaded a new Issabel 4 .ova image to SourceForge which will save you some time in getting Issabel up and running. Once you’ve downloaded and installed the image in VirtualBox, you can log in as root using the default password: password. Then you can set your admin password for the Issabel GUI by running /root/admin-pw-change.

Installing Incredible PBX 13 for Issabel 4


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As with all Incredible PBX builds, running the Incredible PBX installer will erase ALL of your existing Issabel configuration so start with a fresh install of Issabel.

Begin the Incredible PBX install by logging into your Issabel server as root from a desktop PC using SSH or Putty and execute the following commands:

cd /root
wget http://incrediblepbx.com/IncrediblePBX13-Issabel4.sh
chmod +x IncrediblePBX13-Issabel4.sh
./IncrediblePBX13-Issabel4.sh

The Travelin’ Man 3 firewall is installed and configured as part of the install. It whitelists certain IP addresses and blocks everyone else from even seeing your server on the Internet. For this reason, it is critically important that you perform the Incredible PBX install using SSH or Putty from a PC that you will use to manage your Issabel server. Otherwise, you risk locking yourself out of your own server. Whitelisted IP addresses include the Issabel server itself, the public and private IP addresses of your desktop PC, all non-routable, private LAN addresses, and the Nerd Vittles collection of recommended SIP hosting providers. You can add as many additional providers or users to the whitelist using the simple tools provided as part of the install and further documented below.

As part of the install process, you’ll be prompted during both passes to create a password for MySQL/MariaDB and an admin password for the Issabel web GUI. The MySQL password MUST be passw0rd (with a zero), or you will get a permanent mess. The admin password can be anything you like. Passwords can be updated by running /root/admin-pw-change. Many of the Incredible PBX apps depend upon this MySQL password so don’t change it. Your MySQL databases remain secure and can only be accessed on localhost or after a successful root login to your server from a whitelisted IP address.

WhiteListing IP Addresses in Fail2Ban

We also strongly recommend that you whitelist the IP addresses of computers you plan to use to access your new Issabel PBX. The reason is because Fail2Ban jails take precedence over IPtables settings. So even if your IP address has been whitelisted with IPtables using the Travelin’ Man 3 utilities, it’s still possible to lock yourself out of your server by entering the root or admin passwords incorrectly. Here’s how to avoid that. Edit /etc/fail2ban/jail.conf. Scroll down to line #50 which begins with the word "ignoreip." WhiteListed IP addresses are entered here with a space separating each entry. Once you have entered one or more addresses, save the file. Then restart Fail2Ban: service fail2ban restart.


Introducing the (new) Travelin’ Man 3 Firewall

Issabel 4 includes an IPtables firewall component. Do NOT activate it because Incredible PBX includes its own preconfigured IPtables firewall, better known as Travelin’ Man 3. With the Issabel 4 firewall, the administrator is responsible for setting all of the firewall rules. With Travelin’ Man 3, all the heavy lifting is done for you. The design is also markedly different. Issabel 4 opens ports which you define, but it gives worldwide access to those ports by any user. Travelin’ Man 3 employs a WhiteList rather than opening ports for everyone. If you’re on the WhiteList, you get access to the limited collection of ports assigned to that IP address. If you’re not on the WhiteList, you cannot even see the Issabel PBX from the Internet. For those without remote telephones or traveling employees, this provides total protection of your server with virtually no further firewall management.

If you have remote users of your PBX or if you wish to deploy softphones on mobile devices and rely upon WiFi facilities at random locations, Travelin’ Man 3 provides several utilities to assist. If the remote users have static IP addresses, then those IP addresses can be added to the WhiteList by running /root/add-ip. Better yet, a NeoRouter VPN is provided that lets remote users access Issabel using NeoRouter private LAN addresses that already are WhiteListed as part of the installation process. These require little to no configuration with static or dynamic IP addresses even when switching between WiFi networks. For those with dynamic IP addresses and no VPN, FQDNs can be assigned using a service such as dyn.com and a dynamic DNS client can be loaded on the smartphone to keep the current IP address synchronized with the FQDN. On the Incredible PBX side, these FQDNs can be added using /root/add-fqdn, and the IP addresses will be updated automatically every 10 minutes. The final option to provide remote users the 3-digit PortKnocker codes from knock.FAQ and let them automatically whitelist their own IP addresses by running the PortKnocker client from any smartphone or Linux server. When the Issabel server detects a successful knock sequence, the source IP of the knock sequence is whitelisted until the next reload of the firewall. If an administrator prefers to allow permanent additions to the WhiteList that survive a reboot or restart of the firewall, the administrator need only run the following command one time: iptables-knock activate. WhiteListed entries can be removed using the /root/del-acct utility. Further details on the new Travelin’ Man 3 design are available here.

We have modified the security methodology to access the AsteriDex and Reminders pages in the web GUI. We have added another layer of security by requiring Apache htaccess credentials before you can access these pages on your Issabel server. What this means is you will be prompted for Apache admin credentials when you attempt to access these pages. As the last step of the Incredible PBX installation procedure, you will be asked to specify your admin password again. This becomes your Apache admin password, and we recommend keeping it the same as your Issabel password so you don’t get confused. In this way, the username admin and the admin password will be used BOTH for Apache authentication AND Issabel GUI authentication. Should you ever need to change your Issabel admin password, run /root/admin-pw-change. You will need to execute the following command to change the Apache admin password: htpasswd -c /etc/pbx/wwwpasswd admin.

Overview of Issabel 4 Configuration Steps

Almost all PBXs employ a similar design to get calls flowing in and out of your PBX. Extensions are the hooks that let phones on your PBX make a connection to the PBX. Trunks are the hooks that connect your PBX to the outside world so that you can make and receive external calls. Inbound routes tell the PBX how to route incoming calls from the outside world. Outbound routes tell the PBX which trunk providers to use for various types of outgoing calls. And trunk providers are outside businesses that let you terminate calls to telephones all over the world. They also provide phone numbers (DIDs) to you so that the rest of the world has a way to call you.


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Incredible PBX for Issabel makes configuring your PBX easy enough for a fifth grader. We’ve provided two extensions (501 and 502) to give you a simple way to connect your first two phones. We’ve also provided over a dozen sample trunk setups to make it easy to set up trunks once you’ve registered with one or more providers of your choice. If you choose to use our Platinum Sponsor, Skyetel, their trunk setup is already activated and whitelisted on the Issabel platform so all you’ll need to do is collect your $50 signup credit, enter the IP address of your PBX as a Skyetel EndPoint, pick a phone number for your PBX, and point that phone number to your PBX endpoint. On the Issabel side, simply create an Inbound Route for your Skyetel calls by specifying the 11-digit phone number to associate with the inbound route. Finally, we’ll revise the Default Outbound Route to send outgoing calls out through Skyetel.

Getting Started with a $50 Skyetel Credit

To take advantage of the Nerd Vittles specials, begin by completing the Prequalification Form here. You then will be provided a link to the Skyetel site to complete your registration. Once you have registered on the Skyetel site and your account has been activated, open a support ticket and request a $50 credit for your account by referencing the Nerd Vittles special offer. Credit is limited to one per person/company/address/location. If you want to take advantage of the 10% discount on your current service, open another ticket and attach a copy of your last month’s bill. See footnote 1 for the fine print.1 If you have high call volume requirements, document these in your Prequalification Form, and we will be in touch. More details here. Effective 10/1/2023, $25/month minimum spend required.

Skyetel Endpoint Group Configuration

Unlike many VoIP providers, Skyetel does not use SIP registrations to make connections to your PBX. Instead, Skyetel utilizes Endpoint Groups to identify which servers can communicate with the Skyetel service. An Endpoint Group consists of a Name, an IP address, a UDP or TCP port for the connection, and a numerical Priority for the group. For incoming calls destined to your PBX, DIDs are associated with an Endpoint Group to route the calls to your PBX. For outgoing calls from your PBX, a matching Endpoint Group is required to authorize outbound calls through the Skyetel network. Thus, the first step in configuring the Skyetel side for use with your PBX is to set up an Endpoint Group. A typical setup for use with Incredible PBX®, Asterisk®, or FreePBX® would look like the following:

  • Name: Issabel
  • Priority: 1
  • IP Address: Issabel-Public-IP-Address
  • Port: 5060
  • Protocol: UDP
  • Description: issabel.incrediblepbx.com

Skyetel DID Configuration

To receive incoming PSTN calls, you’ll need at least one DID. On the Skyetel site, you acquire DIDs under the Phone Numbers tab. You have the option of Porting in Existing Numbers (free for the first 60 days after you sign up for service) or purchasing new ones under the Buy Phone Numbers menu option.

Once you have acquired one or more DIDs, navigate to the Local Numbers or Toll Free Numbers tab and specify the desired SIP Format and Endpoint Group for each DID. Add SMS/MMS and E911 support, if desired. Call Forwarding and Failover are also supported. That completes the VoIP setup on the Skyetel side. System Status is always available here.

Incredible PBX Inbound Routing with Skyetel

Next we need to tell your PBX how to route incoming calls from Skyetel. Using a browser, log into the IP address of your PBX using your admin credentials. Because there is no trunk registration with Skyetel trunks, you will need to create an Inbound Route for every Skyetel DID. You cannot rely upon a Default inbound route because Issabel treats the calls as blocked anonymous calls without an Inbound Route pointing to the 11-digit number of each Skyetel DID. From the Issabel GUI, choose PBX -> PBX Configuration -> Inbound Routes -> Add Incoming Route. For both the Description and DID fields, enter the 11-digit phone number beginning with a 1. Set the Destination for the incoming DID as desired, e.g. IVR:IVR Demo. Click Submit. Reload the Dialplan when prompted. Place a test call to each of your DIDs from an external phone or cellphone after configuring the Inbound Routes.

Incredible PBX Outbound Routing to Skyetel

If Skyetel will be your primary provider, you can use both 10-digit and 11-digit dialing to process outbound calls through your Skyetel account. From the GUI, choose PBX -> PBX Configuration -> Outbound Routes -> Default. Scroll down to the Trunk Sequence section of the template. Choose these 3 trunks in this order: Skyetel-1, Skyetel-NW, and Skyetel-SE. Next, click Submit Changes and reload the dialplan when prompted.


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Setting Up a Softphone with Issabel 4

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. You can decipher your server’s IP address by running pbxstatus. If you wish to use one of the preconfigured extensions (501 and 502), you’ll find the randomized passwords in /root/passwords.FAQ. 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 (D-E-M-O) to try things out. With Telephone, you can use over two dozen soft phones simultaneously.

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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 Issabel extension. You’ll need the IP address of your server plus your extension number and password associated with either the 501 or 502 extension.

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Adding Speech Recognition Support to Incredible PBX

To support many of our applications, Incredible PBX has included Google’s speech recognition service. These applications include AsteriDex Voice Dialing by Name (411) and Wolfram Alpha for Asterisk (4747), all of which use Lefteris Zafiris’ terrific speech-recog AGI script. Unfortunately (for some), Google now has tightened up the terms of use for their free speech recognition service. Now you can only use it for "personal and development use." If you meet those criteria, keep reading. Here’s how to activate speech recognition on Incredible PBX. Don’t skip any steps!

If you like Siri, you’ll love Wolfram Alpha. To use Wolfram Alpha by phone, you first must obtain a free Wolfram Alpha APP-ID. Then issue the following command replacing APP-ID with your actual ID. Don’t change the yourID portion of the command:

sed -i "s|yourID|APP-ID|" /var/lib/asterisk/agi-bin/4747

Now you’re ready to try out the speech recognition apps. Dial 411 and say "American Airlines" to be connected to American.

To access Wolfram Alpha by phone, dial 4747 and enter your query, e.g. "What planes are overhead now?" Read the Nerd Vittles tutorial for additional examples and tips.

Implementing IBM TTS and Voice Recognition

While Google voice recognition originally was free, it has been a hit and miss platform for the last couple years. If you’re really serious about text-to-speech (TTS) and voice recognition (STT) quality, then you owe it to yourself to make the switch to the IBM platform. For most deployments, the IBM platform will be nearly free. Our recent tutorial will walk you through the process of getting your IBM credentials and setting up the TTS and STT functions with IBM Watson. Be advised that you will have two sets of credentials, one for TTS applications and another for STT applications. Once you have your credentials, here are the steps to reconfigure Issabel to use the IBM TTS and voice recognition services. Begin by logging into your server as root and switching to the /var/lib/asterisk/agi-bin directory. Then install the IBM components:

cd /var/lib/asterisk/agi-bin
wget http://incrediblepbx.com/ibm-issabel.tar.gz
tar zxvf ibm-issabel.tar.gz
rm -f ibm-issabel.tar.gz
mv custom/* /var/lib/asterisk/sounds/custom

Implementing IBM STT with Incredible PBX’s Voice Dialer. With this application, a user dials 411 and speaks the name of a person or company to call. The app searches for a match in the AsteriDex directory and places the call. To get started, edit getnumber.sh and insert your IBM STT credentials in the API_USERNAME and API_PASSWORD fields. Then save the file. Replace the Call by Name context by running the following script: ./install-ibm411.sh. Place a test call by dialing 411 and saying "American Airlines."

Implementing IBM STT with Incredible PBX’s SMS Dictator. With this application, a user dials 767, enters the 10-digit number for the recipient of an SMS text message, and then speaks the message to be sent. To get started, edit smsgen.sh and insert your IBM STT and Google Voice credentials using your plain-text Google password. Then save the file. Replace the SMS Dictator context by running the following script: ./install-sms767-dialplan.sh. Place a test call to 767, and the app will send your text message to the recipient’s phone number using the gvoice application. If you experience failed calls, try executing the Unlock Captcha procedure using your Google Voice credentials. Then try again.

Implementing IBM STT with Incredible PBX’s Wolfram Alpha. With this Siri-like app, a user dials 4747 and speaks a query to be sent to Wolfram Alpha for processing. The results then are played back to the caller. To begin, edit wolfram.sh and insert your IBM STT credentials as well as your Wolfram Alpha APPID. Then save the file. Replace the Wolfram Alpha dialplan code by running the following script: ./install-wolfram4747-dialplan.sh. Place a test call by dialing 4747. When prompted for your query, say "What planes are flying overhead now?"

Implementing IBM TTS with Incredible PBX’s News and Weather Apps. With these apps, a user dials 951 for the latest News Headlines from Yahoo or 947 to retrieve the latest weather report by ZIP code. To begin, edit ibmtts.php and insert your IBM TTS credentials in the IBM_username and IBM_password fields. Then save the file. Replace the news and weather by zip code contexts by running the following script: ./install-ibmtts-dialplan.sh.

Generating IBM Voice Prompts to Use with Issabel. We’ve included a script that will let you generate IBM voice prompts that are suitable for use with Issabel and Incredible PBX. To begin, edit ibmprompt.php and insert your IBM TTS credentials in the IBM_username and IBM_password fields. Then save the file. Next, we need to add MP3 support to the SOX application before we can create voice prompts reliably with IBM’s Bluemix TTS service. Here’s how:

yum -y remove sox
yum -y install libmad libmad-devel libid3tag libid3tag-devel lame lame-devel flac-devel
cd /usr/src
wget https://sourceforge.net/projects/sox/files/sox/14.4.2/sox-14.4.2.tar.gz
tar zxvf sox-14.4.2.tar.gz
rm -f sox-14.4.2.tar.gz
cd sox*
./configure
make -s
make install
ldconfig
ln -s /usr/local/bin/sox /usr/bin/sox

Generate voice prompts using the following syntax: ./ibmprompt.php "Hello world."

Configuring the Issabel Fax Server

Incredible PBX for Issabel includes turnkey fax support with Issabel. Once you have added a trunk that supports VoIP faxing (HINT: Skyetel trunks work great!), fax configuration with Issabel only takes a minute. Start by logging into the Issabel web interface as admin. First, navigate to PBX:PBX Configuration:Extensions:Fax and obtain your password for extension 329. Next, navigate to Fax:Virtual Fax:New Virtual Fax. Fill in the form as shown below using your actual email address and phone number for receiving faxes as well as your actual extension 329 secret. Then click SAVE. Assuming you typed your secret correctly, you will see a status notification showing virtual fax machine "Running and idle on ttyIAX1."


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Assuming you already have set up a Skyetel trunk as outlined above, the next step is to modify the Inbound Route for this trunk to support fax detection. In that way, incoming fax calls will automatically be redirected to extension 329 and the received faxes will be emailed to you in PDF format. Set the email address in Fax:Fax Master. In addition, the faxes can be downloaded and managed from Fax:Virtual Fax:Fax Viewer. Modify your Inbound Route to match the #3 settings shown below. Then save/reload your changes.


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To receive the incoming faxes by email, navigate to Fax:Fax Master and enter your email address. Then click SAVE.

The final step is to designate the IP addresses of those authorized to send faxes using Issabel. Navigate to Fax:Fax Clients and specify the public and private IP addresses (one per line) authorized to send faxes. Then click SAVE. Hylafax clients can be used remotely, or you can use the web utility included with Issabel: Fax:Virtual Fax:Send Fax.


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The best way to test things out is to send yourself a test fax. FaxZERO lets you send 5 free faxes of up to 3 pages every day. Give it a whirl.

To send a fax out from your server from the Linux CLI using either a text document or PDF file, the syntax looks like the following:

sendfax -n -d 8005551212 smsmsg.txt


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Replacing MeetMe Conferencing with ConfBridge

The only serious limitation we’ve found with the Issabel implementation of FreePBX is the continued reliance upon MeetMe for conferencing which requires a timing source unlike the newer ConfBridge module. Particularly on OpenVZ VPS platforms, this causes issues because of the inability to directly access the kernel. Fortunately, Issabel has included the functioning ConfBridge module in their implementation so the workaround is fairly simple. By default, we’ve included a 2663 (C-O-N-F) conference setup in the Issabel GUI configuration so simply remove it. Then add a 2663 Misc Destination with a description of CONF. Finally, while still in the GUI, edit the IVR Demo and change the destination for option 2 to Misc Destination:CONF and save the file. Next, log into the Linux CLI as root and change to the /etc/asterisk directory. Edit confbridge_custom.conf and insert the following code. Then save the file.

[general]
;This section reserved for future use

[default_user]
type = user
quiet = no
announce_user_count = yes
announce_user_count_all = yes
wait_marked = no
end_marked = no
dsp_drop_silence = yes
announce_join_leave = yes
admin = no
marked = no
startmuted = no
music_on_hold_when_empty = yes

[admin]
type = user
quiet = no
announce_user_count = yes
announce_user_count_all = yes
wait_marked = no
end_marked = no
dsp_drop_silence = yes
announce_join_leave = yes
admin = yes
marked = no
startmuted = no
music_on_hold_when_empty = yes

[default_bridge]
type = bridge
record_conference = no
sound_only_person =    conf-onlyperson
sound_has_joined =     conf-hasjoin
sound_has_left =       conf-hasleft
sound_kicked =         conf-kicked
sound_muted =          conf-muted
sound_unmuted =        conf-unmuted
sound_there_are =      conf-thereare
sound_other_in_party = conf-otherinparty
sound_place_into_conference = conf-placeintoconf
sound_wait_for_leader =       conf-waitforleader
sound_get_pin =        conf-getpin
sound_invalid_pin =    conf-invalidpin
sound_locked =         conf-locked
sound_unlocked_now =   conf-unlockednow
sound_lockednow =      conf-lockednow
sound_error_menu =     conf-errormenu

[admin_menu]
type = menu
* = playback_and_continue(conf-adminmenu)
*1 = toggle_mute
*2 = admin_toggle_conference_lock
*3 = admin_kick_last
*4 = decrease_listening_volume
*5 = reset_listening_volume
*6 = increase_listening_volume
*7 = decrease_talking_volume
*8 = reset_talking_volume
*9 = increase_talking_volume
*# = leave_conference
*0 = admin_toggle_mute_participants

[user_menu]
type = menu
* = playback_and_continue(conf-usermenu)
*1 = toggle_mute
*4 = decrease_listening_volume
*5 = reset_listening_volume
*6 = increase_listening_volume
*7 = decrease_talking_volume
*8 = no_op
*9 = increase_talking_volume
*# = leave_conference

Now edit extensions_custom.conf and insert the following code below the [from-internal-custom] label replacing the 1234 and 4321 PINs in lines 6 and 7 with user and admin PINs of your choice (up to 8 numbers each). Then restart Asterisk: amportal restart.

;# // BEGIN Conf1
exten => 2663,1,Answer
exten => 2663,2,Wait(1)
exten => 2663,3,Playback(conf-getpin)
exten => 2663,4,Read(MYPIN,beep,8)
exten => 2663,5,GotoIf($["${MYPIN}" = "1234"]?userpin)
exten => 2663,6,GotoIf($["${MYPIN}" = "4321"]?adminpin)
exten => 2663,7,Playback(goodbye)
exten => 2663,8,Hangup
exten => 2663,n(adminpin),Set(CONFBRIDGE(user,template)=admin)
exten => 2663,n,ConfBridge(1)
exten => 2663,n,Hangup
exten => 2663,n(userpin),Set(CONFBRIDGE(user,template)=default_user)
exten => 2663,n,ConfBridge(1)
exten => 2663,n,Hangup
;# // END Conf1

Backup and Restore with Issabel

Issabel ships with the most full-featured Backup and Restore options of any of the Asterisk distributions. Ask us how we know. Yes, we managed to wipe out the entire Dashboard menu system on one of our early builds. Restoring from an image took only a couple minutes. To get started, navigate to System -> Backup/Restore. You can create backups locally and then drag and drop them onto a remote FTP server if desired. There is enormous flexibility in choosing what to backup or restore. And there’s even an option to automatically generate periodic backups. You’ll find your backups in /var/www/backup should you ever need to copy them to a new server. Now would be a good time to create your first backup. 🙂

Sampling Other Incredible PBX Applications

As installed, Incredible PBX includes dozens of additional applications for Asterisk. Here’s how to sample some of them using a softphone connected to your Issabel PBX. A good place to start is Allison’s Demo IVR (dial D-E-M-O) using any phone connected to your PBX:

Nerd Vittles Demo IVR Options
1 – 411 -Call by Name (say "American Airlines")
2 – 2663 – MeetMe/ConfBridge Conference
3 – 4747 – Wolfram Alpha
4 – 53669 – Lenny (The Telemarketer’s Worst Nightmare)
5 – 951 – Today’s News Headlines
6 – 947 – Weather Forecast (enter a 5-digit ZIP code)
7 – 86329 – Today in History
8 – 501 – Speak to a Real Person

For ODBC demos, dial 222 and enter 12345 for the employee number for a sample database application. Or dial 223 for a sample ODBC dialer using AsteriDex. Enter 263 (first three letters of American Airlines) to place the call. Sample dialplan code is stored in /etc/asterisk/odbc.conf. Dial L-E-N-N-Y (53669) to call or forward telemarketer calls to Lenny. Dial T-I-M-E (8463) for Time of Day. Dial *88HHMM to set an Alarm for HH:MM where HH is the hour of the day in military time. Dial C-O-N-F (2663) for MeetMe conference. Conference credentials are in /root/passwords.FAQ. Voice Dialer (411) works with any database entry in AsteriDex. Access AsteriDex with a browser at https://Issabel-IP-Address/asteridex4. Telephone Reminders can be scheduled by phone (123) or via the web: https://Issabel-IP-Address/reminders. Sample code for the FLITE, GoogleTTS, and PicoTTS engines is in 951 (Yahoo News) context of /etc/asterisk/extensions_custom.conf. All of your FreePBX "old favorites" including blacklists, call transfers and forwarding, dictation, recordings and more are still available as well: PBX:PBX Config:Feature Codes.

Continue Reading: Configuring Extensions, Trunks & Routes.

Don’t Miss: Incredible PBX Application User’s Guide covering the 31 Incredible PBX apps.

Published: Friday, October 5, 2018  Updated: Friday, February 1, 2019


blankSupport Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a difficult place to address support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forum. It’s the best Asterisk tech support site in the business, and it’s all free! Please have a look and post your support questions there. Unlike some forums, the PIAF Forum is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of users just like you. You won’t have to wait long for an answer to your question.


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Need help with Asterisk? Visit the VoIP-info 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.
 



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  1. In the unlikely event that Skyetel cannot provide a 10% reduction in your current origination rate and/or DID costs, Skyetel will give you an additional $50 credit to use with the Skyetel service. []

One Minute Cloud VPS: Meet Incredible PBX for HiFormance

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Dec. 17 Update: Before signing up for HiFormance service, read the latest update on this provider here.

Part of the challenge of deploying an Asterisk®-based server on a cloud platform is getting all of 1,000+ pieces in place without a hiccup during the installation. Particularly for first-time users of a VoIP platform, this can be problematic. HiFormance is one of our favorite low-cost cloud providers, and today they’re introducing something that no other cloud provider offers: a configured Incredible PBX 13-13 VPS that’s ready to accept calls in under a minute. We’ve been inching toward a new low-cost plateau for VoIP cloud providers, and today we have a new milestone that finally makes running VoIP servers out of your home or office look like the horse-and-buggy days.

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$10 to $13 a year now buys you a cloud platform that is less expensive than the cost of electricity to run a server on premise. You get 2GB of RAM, 20GB of SSD storage, two virtual core processors, and 2TB of monthly bandwidth. If you prepay for 3 years, you can double either the RAM or SSD storage by simply opening a ticket after you sign up. The $10/year offering is in limited supply and gives you your choice of the following sites: L.A., Buffalo, or Dallas. The $13/year offering adds the following additional sites: Chicago, Atlanta, and New York. It’s a near perfect platform for Incredible PBX 13-13 with CentOS 6.9. Add a Google Voice trunk and you get unlimited calling in the U.S. and Canada combined with a feature set that you’ll be hard-pressed to find on any PBX at any price. Putting all the pieces in place is about as simple as preparing slice-and-bake cookies, and you’ll be up and running before the cookies come out of the oven. Skip that hamburger lunch and come join the party!


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So what’s the catch? Well, there’s no catch with Incredible PBX 13-13 and CentOS 6.9. This HiFormance platform uses OpenVZ with SolusVM. Someone always asks, "If the platform is so great, why aren’t you using it?" And our answer is we are. We have deployed HiFormance cloud-based VoIP servers running Incredible PBX 13-13 in Atlanta, Buffalo, Chicago, and Los Angeles without any hiccups in service. Performance is excellent. Support is excellent. So run, don’t walk, to sign up for one of these before they’re all gone. You won’t be disappointed. Just fill out the entries as shown above once you log into the HiFormance site. Nerd Vittles receives no commissions from signups.

June 20 Update: For those of you on the West Coast and readers in the Far East, we are pleased to announce that HiFormance has extended the Incredible PBX offer to their new $1/month VZ-2019 platform featuring 4GB RAM, 20GB SSD storage, 4TB of bandwidth, and DDOS protection at the QuadraNet Los Angeles data center. As with the other offers, you can double the bandwidth or storage by prepaying for three years in advance and opening a ticket. For those in the Far East, you can upgrade to the Asia Optimized CN2, Los Angeles offering for an additional $10.99 a year.

Getting Started with Incredible PBX 13-13

Once you’ve signed up for an account, choose a location for your server and select Incredible PBX 13-13 as your "operating system." It’s a preconfigured implementation that uses the latest CentOS 6.9 platform. Once you receive your credentials by email, login to your SolusVM Control Panel with a browser. Click the Manage tab for your virtual machine. In the Settings tab at the bottom of the control panel, turn TUN/TAP ON if you plan to use the NeoRouter VPN. Next, click the Root/Admin Password tab and set a default root password for your virtual machine. Take a look around in the control panel while you’re there. This is where you’d return if you need to restart your virtual machine or if you wish to reinstall your virtual machine with the same or a different operating system.




 

Now that your virtual machine is up and running, log into your server with SSH as root and type YES to get started. Be sure to log in from a desktop machine that you plan to use to manage your server since this IP address will be whitelisted. You’ll be prompted to immediately change all of your passwords. When the setup finishes, just type "reboot" to reboot your server. Log back in as root and the Automatic Update Utility will bring your server up to current specs. If you’re not in the Eastern U.S. time zone, type: /root/timezone-setup. Then you’re ready to go.

One of the unique features of Incredible PBX 13-13 is that most of the major components of the aggregation including Asterisk were compiled from source, and the source code is actually on your server. The advantage is you can quickly make any future modifications desired to meet your own unique requirements. You won’t find this in any other VoIP implementation. For example, here’s all it takes to upgrade to the very latest release of Asterisk. As with the current version, both support SRTP encrypted communications!

cd /usr/src
rm -rf asterisk-13.21.0
wget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-13.21.1.tar.gz
tar zxvf asterisk-13.21.1.tar.gz
mv asterisk-13.21.1.tar.gz /tmp
cd asterisk-13.21.1
contrib/scripts/get_mp3_source.sh
 mv res/res_xmpp.c res/res_xmpp.c.orig
 wget http://incrediblepbx.com/res_xmpp-13.tar.gz
 tar zxvf res_xmpp-13.tar.gz
 cp -p res/res_xmpp.c res/res_xmpp.c.oauth
make distclean
wget http://incrediblepbx.com/menuselect-incredible13.8.tar.gz
tar zxvf menuselect-incredible*
rm -f menuselect-incredible*
wget https://issues.asterisk.org/jira/secure/attachment/54233/srtp_aead_and_big_aes.patch
wget http://incrediblepbx.com/srtp_aead_and_big_aes.patch.patch
patch -p0 < ./srtp_aead_and_big_aes.patch.patch
patch -p0 <./srtp_aead_and_big_aes.patch
CFLAGS='-DENABLE_SRTP_AES_256 -DENABLE_SRTP_AES_GCM' ./configure --libdir=/usr/lib64 --with-pjproject-bundled
 make menuselect.makeopts
 menuselect/menuselect --enable-category MENUSELECT_ADDONS menuselect.makeopts
 menuselect/menuselect --enable-category MENUSELECT_CODECS menuselect.makeopts
 menuselect/menuselect --disable-category MENUSELECT_TESTS menuselect.makeopts
 menuselect/menuselect --enable CORE-SOUNDS-EN-GSM --enable MOH-OPSOUND-WAV --enable EXTRA-SOUNDS-EN-GSM --enable cdr_mysql menuselect.makeopts
 menuselect/menuselect --disable app_mysql --disable app_setcallerid --disable func_audiohookinherit --disable res_fax_spandsp menuselect.makeopts
 menuselect/menuselect --disable test_named_lock --disable test_res_pjsip_scheduler --disable test_file --disable test_bridging --disable test_res_pjsip_scheduler menuselect.makeopts
 sed -i 's|march=native|mtune=native|' Makefile.rules
expect -c 'set timeout 60;spawn make menuselect;expect "Save";send "\t\t\r";interact'
make
make install
ldconfig
cd /usr/src/Asterisk-Flite-2.2-rc1-flite1.3
make clean
make
make install
amportal restart

WebMin is also installed and configured as part of the base install. The root password for access is the same as your Linux root password. We strongly recommend that you not use WebMin to make configuration changes to your server. You may inadvertently damage the operation of your PBX beyond repair. WebMin is an excellent tool to LOOK at how your server is configured. When used for that purpose, we highly recommend WebMin as a way to become familiar with your Linux configuration.

Using the Incredible PBX 13-13 Web GUI

Most of the configuration of your PBX will be performed using the web-based Incredible PBX GUI with its FreePBX 13 GPL modules. Use a browser pointed to the IP address of your server and choose Incredible PBX Admin. Log in as admin with the password you configured in the previous step. HINT: You can always change it if you happen to forget it: /root/admin-pw-change. You can safely ignore the warning about a missing swap file. You have plenty of RAM, and OpenVZ platforms don't permit swap files. If you're worried about it, choose the 3-year prepayment option and double your ram from 2GB to 4GB which is more than ample for even the busiest PBXs.

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To get a basic system set up so that you can make and receive calls, you'll need to configure a VoIP trunk, create one or more extensions, set up an inbound route to send incoming calls to an extension, and set up an outbound route to send calls placed from your extension to a VoIP trunk that connects to telephones in the real world. You'll also need a SIP phone or softphone to use as an extension on your PBX.

Continue Reading: Configuring Extensions, Trunks & Routes

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Introducing Incredible PBX Whole Enchilada

The Whole Enchilada upgrade already is included on the HiFormance platform. You can review the tutorial for the three dozen apps here.

Reconfiguring NeoRouter VPN for OpenVZ

On OpenVZ platforms including HiFormance, you'll need to enable TUN/TAP in the Control Panel for your VPS. After adjusting the setting, reboot your server. Then the NeoRouter VPN client will function properly.

Installing Incredible Fax with HylaFax/AvantFax

To install Incredible Fax, log into your server as root and issue the following commands. Install time: 2 minutes.

cd /root
./incrediblefax13.sh

After entering your email address to receive incoming faxes, you'll be prompted about two dozen times to choose options as part of the install. Simple press the ENTER key at each prompt and accept all of the defaults. When the install finishes, make certain that you reboot your server to bring Incredible Fax on line. There will be a new AvantFax option in the Incredible PBX GUI. The default credentials for AvantFax GUI are admin:password; however, you first may be prompted for your Apache admin credentials which were set when you installed your server. Then you'll be asked to change your AvantFax password.


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Be advised that HiFormance blocks outbound email ports including port 587 by default. You'll need to open a ticket with them to use email from your server. You can test whether port 25 and 587 are blocked with the following command:

telnet gmail-smtp-msa.l.google.com 587

Backing Up Your Server

One of the reasons we love the CentOS 6.9 platform is the ease with which you can make full system backups, particularly on cloud platforms. There's no MySQL InnoDB databases to worry about so you can treat MySQL databases just like any other Linux file. We're offering the following script for your use pursuant to the GPL2 license. By using the script at no cost, you agree to assume all risks and absolve us from any liability regarding bugs, performance, or any other failure in the code. If that's acceptable to you, copy the commands below and create a backup-full script in the /root folder of your primary server. After saving the script, make it executable: chmod +x backup-full.

#!/bin/bash
# backup-full for Incredible PBX, Copyright (c) 2008-2018, Ward Mundy & Associates, LLC
# Licensed pursuant to GPL2. See /root/COPYING on any Incredible PBX server for details
amportal stop
service mysqld stop
service httpd stop
service sendmail stop
cd /
tar -cf /tmp/backup.tar /bin /etc /home /lib /lib64 /media /mnt /opt /root /sbin /usr /var
service sendmail start
service httpd start
service mysqld start
amportal start
tar --delete -f /tmp/backup.tar etc/udev/rules.d
tar --delete -f /tmp/backup.tar etc/sysconfig/network-scripts
tar --delete -f /tmp/backup.tar var/lib/dhclient
tar --delete -f /tmp/backup.tar etc/fstab
tar --delete -f /tmp/backup.tar etc/resolv.conf
tar --delete -f /tmp/backup.tar etc/hosts
tar --delete -f /tmp/backup.tar etc/hostname
gzip /tmp/backup.tar
echo "Your backup is available: /tmp/backup.tar.gz"
echo "Copy it and test it in a safe place OFF SITE"
echo " "


To run the script, execute the following command: /root/backup-full

We recommend running the backup-full script during hours when your PBX is not in active use since Asterisk and other services typically are shut down for 2-10 minutes. Depending upon the size and performance of your server, the complete backup process typically takes 5-20 minutes. Once the backup script finishes, copy /tmp/backup.tar.gz to a safe place away from the primary server every week. You can automate the backup and the copying procedure with a cron job if desired.

To restore, copy your backup.tar.gz file back to the /tmp directory on the new server and then:

#!/bin/bash
# restore-full for Incredible PBX, Copyright (c) 2008-2018, Ward Mundy & Associates, LLC
# Licensed pursuant to GPL2. See /root/COPYING on any Incredible PBX server for details
amportal stop
service mysqld stop
service httpd stop
service sendmail stop
cd /
tar zxvf /tmp/backup.tar.gz
reboot


Upgrading to IBM Speech Engines

NOV. 1 UPDATE: IBM has moved the goal posts effective December 1, 2018:

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If you've endured Google's Death by a Thousand Cuts with text-to-speech (TTS) and voice recognition (STT) over the years, then we don't have to tell you what a welcome addition IBM's new speech utilities are. We can't say enough good things about the new IBM Watson TTS and STT offerings. While IBM's services are not free, that’s really theoretical for most of our readers. Your first month on the platform is entirely free. And, after that, you get 1,000 minutes a month of free STT voice recognition services. And the first million characters of text-to-speech synthesis are FREE every month as well. So let's put the pieces in place so you'll be ready to play with the Whole Enchilada. Here's our tutorial that will walk you through the one-time IBM setup.

Next, login to your Incredible PBX server and issue these commands to update your Asterisk dialplan and edit ibmtts.php:

cd /var/lib/asterisk/agi-bin
./install-ibmtts-dialplan.sh
nano -w ibmtts.php

Insert your credentials in $IBM_username and $IBM_password. Verify that $IBM_url matches the entry provided when you registered with IBM. Then save the file: Ctrl-X, Y, then ENTER. Now reload the Asterisk dialplan: asterisk -rx "dialplan reload". Try things out by dialing 951 (news) or 947 (Weather) from an extension registered on your PBX.

To get IBM's Speech to Text service configured, while still logged in to your Incredible PBX server, issue these commands to edit getnumber.sh:

cd /var/lib/asterisk/agi-bin
nano -w getnumber.sh

Insert your API_USERNAME and API_PASSWORD in the fields provided. Then save the file: Ctrl-X, Y, then ENTER. Update your Voice Dialer (411) to use the new IBM STT service:

sed -i '\\:// BEGIN Call by Name:,\\:// END Call by Name:d' /etc/asterisk/extensions_custom.conf
sed -i '/\\[from-internal-custom\]/r ibm-411.txt' /etc/asterisk/extensions_custom.conf
asterisk -rx "dialplan reload"

Now try out the Incredible PBX Voice Dialer with AsteriDex by dialing 411 and saying "Delta Airlines."

Configuring Google Voice with Incredible PBX

The advantage of Google Voice trunks for those of you in the United States is that all of your calls within the U.S. and Canada are free. You can't beat the price, and it has worked reliably for many, many years but it may disappear any day now. If so, we'll have a fresh tutorial with a workaround for you soon. Unless Google pulls the plug on XMPP support for Google Voice, there are three different ways to set up Google Voice trunks with Incredible PBX. For a one-time fee of $4.99 with this coupon, you can use the Simonics GV/SIP gateway to configure a Google Voice account using OAuth 2 authentication. Then just set up the Simonics SIP trunk on your PBX to point to the Simonics gateway. A second option is to choose the (recommended) OAuth 2 authentication method for Google Voice when you initially install Incredible PBX 13-13. Finally, you can choose plain-text passwords for Google Voice when you set up Incredible PBX. The drawback of this last option is Google has hinted that they may discontinue support of plain-text passwords.

Here are the initial setup steps on the Google side:

1. Set up a dedicated Gmail and Google Voice account to use exclusively for this Google Voice setup on your PBX. Head over to the Google Voice site and register. You'll need to provide a U.S. phone number to verify your account by either text message or phone call.


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2. Once you have verified your account by entering your verification code, you'll get a welcome message from Mr. Google. Click Continue to Google Voice.


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3. Provide an existing U.S. phone number for verification. It can be the same one you used to set up your Google account in step #1.


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4. Once your phone number has been verified, choose a DID in the area code of your choice.


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Special Note: Google continues to tighten up on obtaining more than one Google Voice number from the same computer or the same IP address. If this is a problem for you, here's a workaround. From your smartphone, install the Google Voice app from iPhone App Store or Google's Play Store. Then open the app and login to your new Google account. Choose your new Google Voice number when prompted and provide a cell number with SMS as your callback number for verification. Once the number is verified, log out of Google Voice. Do NOT make any calls. Now head back to your PC's browser and login to https://voice.google.com. You will be presented with the new Google Voice interface which does not include the Google Chat option. But fear not. At least for now there's still a way to get there. After you have set up your new phone number and opened the Google Voice interface, click on the 3 vertical dots in the left sidebar (it's labeled More). When it opens, click Legacy Google Voice in the sidebar. That will return you to the old UI. Now click on the Gear icon (upper right) and choose Settings. Make sure the Google Chat option is selected and disable forwarding calls to whatever default phone number you set up.

5. When your DID has been assigned, click the More icon at the bottom of the left column of the Google Voice desktop. Click Legacy Google Voice. Now click the Settings icon on your legacy Google Voice desktop. Make certain that Forward Calls to Google chat is checked and disable calls to your forwarding number. Click on the Calls tab and select Call Screening:OFF, CallerID (Incoming):Display Caller's Number, and Global Spam Filtering:checked. The remaining entries should be blank.

6. Google Voice configuration is now complete. Sign out of your Google Voice account.


The Simonics GV-SIP Gateway Solution. Here's the quick thumbnail of the steps to put all the pieces in place. First, we set up a Google Voice account at Google as documented above. Next, we'll set up an account at the Simonics site to link our Google Voice account to the Simonics SIP Gateway. Then we'll plug our Simonics SIP credentials into the preconfigured Simonics trunk on Incredible PBX. Finally, we'll add Incoming and Outgoing Routes to tell Incredible PBX how to process Google Voice calls.

Now you're ready to set up an account on the Simonics site. With this Nerd Vittles link, there's a one-time fee of $4.99.

1. Start by registering your new Google account.

2. After paying the $4.99 registration fee via PayPal, proceed through the setup process to link your Google Voice account and 11-digit Google Voice phone number to the Simonics SIP Gateway.

3. You then will be provided your SIP username and password as well as the gateway address, gvgw.simonics.com, to use in building your SIP trunk on your PBX.


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4. If your SIP credentials ever get compromised, regenerate your password by logging back into the Simonics GW site.

Now it's time to configure your Simonics trunk in Incredible PBX. Start by logging into the web interface as admin with your admin password from above. Click Connectivity:Trunks and choose the Simonics trunk in the PBX Configuration menu. The Simonics trunk template will display:

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1. Untick the Disable Trunk check box.

2. In Outbound CallerID, insert your 10-digit Google Voice number.

3. In username, insert GV1 followed by your 10-digit Google Voice number.

4. In secret, insert your Simonics SIP password.

5. In the Registration String, insert GV1 followed by your 10-digit Google Voice number followed by a colon (:)

6. In the Registration String after the colon, insert your Simonics SIP password.

7. In the tail of the Registration String after the slash (/), insert your 10-digit Google Voice number.

8. Click Submit Changes and then Reload the Dialplan when prompted.


Configuring GV Trunk with Motif in the GUI. If you elect to configure your Google Voice trunk natively using the Incredible PBX GUI, you first will need to obtain a Refresh_Token if you elected to use OAuth 2 authentication.

1. Be sure you are still logged into your Google Voice account. If not, log back in at https://voice.google.com.

2. In a separate browser tab, go to the Google OAUTH Playground using your browser while still logged into your Google Voice account.

3. Once logged in to Google OAUTH Playground, click on the Gear icon in upper right corner (as shown below).

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  3a. Check the box: Use your own OAuth credentials
  3b. Enter Incredible PBX OAuth Client ID:

466295438629-prpknsovs0b8gjfcrs0sn04s9hgn8j3d.apps.googleusercontent.com

  3c. Enter Incredible PBX OAuth Client secret: 4ewzJaCx275clcT4i4Hfxqo2
  3d. Click Close

4. Click Step 1: Select and Authorize APIs (as shown below)

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  4a. In OAUTH Scope field, enter: https://www.googleapis.com/auth/googletalk
  4b. Click Authorize APIs (blue) button.

5. Click Step 2: Exchange authorization code for tokens

  5a. Click Exchange authorization code for tokens (blue) button

  5b. When the tokens have been generated, Step 2 will close.

6. Reopen Step 2 and copy your Refresh_Token. This is the "password" you will need to enter (together with your Gmail account name and 10-digit GV phone number) when you add your GV trunk in the Incredible PBX GUI. Store this refresh_token in a safe place. Google doesn't permanently store it!

7. Authorization tokens NEVER expire! If you ever need to remove your authorization tokens, go here and delete Incredible PBX Google Voice OAUTH entry by clicking on it and choosing DELETE option.

Switch back to your Gmail account and click on the Phone icon at the bottom of the window to place one test call. Once you successfully place a call, you can log out of Google Voice and Gmail.

Yes, this is a convoluted process. Setting up a secure computing environment often is. Just follow the steps and don't skip any. It's easy once you get the hang of it. And you'll sleep better.

Now you’re ready to configure your Google Voice account in Incredible PBX. You do it from within the Incredible PBX GUI by choosing Connectivity:Google Voice. Just plug in your Google Voice Username, enter your refresh_token from Step #6 above as your Google Voice Password, enter your 10-digit Google Voice Phone Number, and check the first two boxes: Add Trunk and Add Outbound Routes. Then click Submit and Apply Settings to save your new entries.

If you elected to use plain-text passwords for Google Voice, simply skip obtaining OAuth 2 credentials and substitute your plain-text password for the refresh_token when you create the Google Voice trunk above. If you have trouble getting Google Voice to work using a plain-text password, try this Google Voice Reset Procedure. It usually fixes connectivity problems. If it still doesn’t work, enable Less Secure Apps using this Google tool.

IMPORTANT: Once you’ve entered your credentials, you MUST restart Asterisk from the Linux command line, or Google Voice calls will fail: amportal restart

Incredible PBX Wholesale Providers Access

Nerd Vittles has negotiated a special offer that gives you instant access to 300+ wholesale carriers around the globe. In lieu of paying the $650 annual fee for the service, a 13% wholesale surcharge is assessed to cover operational costs of TelecomsXchange. In addition, TelecomsXchange has generously offered to contribute a portion of the surcharge to support the Incredible PBX open source project. See this Nerd Vittles tutorial for installation instructions and signup details.

Continue Reading: Configuring Extensions, Trunks & Routes

Don't Miss: Incredible PBX Application User's Guide covering the 31 Whole Enchilada apps

Originally published: Monday, June 18, 2018


blankSupport Issues. With any application as sophisticated as this one, you're bound to have questions. Blog comments are a difficult place to address support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forum. It's the best Asterisk tech support site in the business, and it's all free! Please have a look and post your support questions there. Unlike some forums, the PIAF Forum is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of users just like you. You won't have to wait long for an answer to your question.


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2018 TREAT: If you could use one or more free DIDs in the U.S. with unlimited inbound calls and unlimited simultaneous channels, then today's your lucky day. TelecomsXChange and Bluebird Communications have a few hundred thousand DIDs to give away so you better hurry. You have your choice of DID locations including New York, New Jersey, California, Texas, and Iowa. The DIDs support Voice, Fax, Video, and even Text Messaging (by request). The only requirement at your end is a dedicated IP address for your VoIP server. Once you receive your welcome email with your number, be sure to whitelist the provider's IP address in your firewall. For Incredible PBX servers, use add-ip to whitelist the UDP SIP port, 5060, using the IP address provided in your welcoming email.

Here's the link to order your DIDs.

Your DID Trunk Setup in your favorite GUI should look like this:

Trunk Name: IPC
Peer Details:
type=friend
qualify=yes
host={IP address provided in welcome email}
context=from-trunk

Your Inbound Route should specify the 11-digit DID beginning with a 1. Enjoy!


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Need help with Asterisk? Join our new MeWe Support Site.


 

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.
 



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Cloud 9: Free Incredible PBX in the Cloud Hosting until 2019


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These deals don’t come along every day so we’re interrupting our regular programming to alert you to a terrific, limited time cloud hosting offer for first-time users of Vultr. If you hurry, you can take advantage of a $25 credit on Vultr which translates into 10 free months of cloud hosting service. We can’t say enough about Vultr. They’ve been one of our key resources for development and testing of new releases of Incredible PBX for many years. Historically, they’ve supported our open source projects through generous referral revenue although that does not apply with this special offer. If you’ve always wondered whether cloud hosting was a viable alternative to on-premise solutions, now’s your chance to kick the tires at zero cost. And the other good news is you have your choice of the following Incredible PBX offerings. Simply load the required OS or upload the ISO for the platform of your choice and follow the linked tutorials below. Enjoy!

Originally published: Friday, May 25, 2018


blankSupport Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It’s the best Asterisk tech support site in the business, and it’s all free! Please have a look and post your support questions there. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of users just like you. You won’t have to wait long for an answer to your question.


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.
 



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