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The Most Versatile VoIP Provider: FREE PORTING

The New Gold Standard: Incredible PBX 13-13.10 for Raspbian



Today we are pleased to introduce the 2019 update for Incredible PBX® and the Raspberry Pi® 2 and 3 featuring 70+ new FreePBX® GPL modules and a native Skyetel SIP trunking platform with a $10 service credit and up to $500 of half-price service. In addition to dozens of under-the-covers tweaks, there also are new backup and restore utilities which should ease the pain of backups and future migrations. In fact, today’s build was created using those tools because the image now is perilously close to filling up a 4GB microSD card. Crossing that threshold would mean future images would literally double in size. As always, for diehard users of legacy features, we’ve retained the terrific features we all know and love.

07/01/2019 NEWS FLASH: Please note that this version is not compatible with the Raspberry Pi 4. However, we have just released Incredible PBX LITE for the Raspberry Pi 2, 3, and 4 featuring Raspbian 10 Buster. Tutorial here.

08/07/2019 UPDATE: And, for the pioneers, Incredible PBX 16-15 for the Raspberry Pi 2, 3, and 4 is now available as well.

In addition to becoming a Nerd Vittles Platinum Provider, we have chosen Skyetel as our recommended SIP provider for several reasons that will be important to you. First, their triple-redundant platform has no equal. Not only have they never had an outage affecting customers, but they also are unlikely to ever have such an outage because their servers are scattered across the entire country (and soon the entire world). Let’s put it this way. If Skyetel’s servers all fail, you’ll have a lot more to worry about than restoring your VoIP service. A second reason we chose the Skyetel platform was introduced by us just last week. You now can bring up a fault-tolerant HA server platform using the Skyetel backbone and a cloud-based redundant server for about $1 a month. For Nerd Vittles readers, you can snag up to a $250 usage credit with Skyetel’s new BOGO deposit match. Read our Skyetel article and sign up soon to claim your BOGO service credit. Effective 10/1/2023, $25/month minimum spend required.

Raspberry Pi 3 Performance. Gone are the days of worrying about Raspberry Pi performance. Both the user interface and call quality now match what you’d expect to find on a $300-$500 VoIP server. Even with a Raspberry Pi 2, we have detected no performance degradation thanks to the latest Raspbian 8 OS and a virtually flawless Asterisk 13 platform. For best results, we recommend 32GB Class 10 microSD cards which now are plentiful for under $10.1



Incredible PBX Feature Set. Where to begin? Let’s start with the Alphabet Stew: IAX, SIP, SMS, Opus, and SRTP functionality. Voice Recognition and Text-to-Speech VoIP application support using FLITE, GoogleTTS, PicoTTS, and IBM TTS. SIP URI support for free worldwide calling. And all of your Nerd Vittles favorites: Fax, AsteriDex, Click-to-Dial, News, Weather, Reminders, and Wakeup Calls. Plus hundreds of features that typically are found in commercial PBXs: Conferencing, IVRs and AutoAttendants, Email Delivery of Voicemail, Voicemail Blasting…

10-Layer Network Security Model. Most phone calls cost money. Unlike many of the other "free" VoIP solutions, our most important criteria for VoIP is rock-solid security. If your free server ends up costing you thousands of dollars in phone bills due to fraud, guess what? It wasn’t free at all. Once you plug into a network, there’s a bullseye painted on your checkbook.

No single network security system can protect you against zero-day vulnerabilities that no one has ever seen. Deploying multiple layers of security is not only smart, it’s essential with today’s Internet topology. It works much like the Bundle of Sticks from Aesop’s Fables. The more sticks there are in your bundle, the more difficult it is to break them apart. If a vulnerability suddenly appears in the Linux kernel, or in Asterisk, or in Apache, or in your favorite web GUI, you can continue to sleep well knowing that other layers of security have your back. No one else in the telecommunications industry has anything close. Ours is all open source GPL code so we would encourage everyone to get on board and do your part to make the Internet a safer place!

Do your homework, too. Comparison shop as if your phone bill matters! 😉 Here’s what the latest Incredible PBX release provides at a software cost of exactly zero:

  1. Preconfigured IPtables Linux Firewall
  2. Preconfigured Travelin’ Man 3 WhiteLists
  3. Randomized Port Knocker for Remote Access
  4. TM4 WhiteListing by Telephone (optional)
  5. Fail2Ban Log Monitoring for SSH, Apache, Asterisk
  6. Password Customization
  7. Automatic Update Utility for Security & Bug Fixes
  8. Asterisk Manager Lockdown to localhost
  9. Apache htaccess Security for Vulnerable Web Apps
  10. Security Alerts via RSS Feed in the Incredible PBX GUI


Assembling the Required Raspberry Pi Components

Before you can deploy Incredible PBX, you’ll first need the necessary Raspberry Pi hardware. Here’s the short list and, if you’re in a hurry, the $35 Raspberry Pi 3B+ will cost you 10% more to get it quickly from Amazon using our referral link. It remains one of the world’s best bargains! Assuming you already own an HDMI-compatible monitor and a USB keyboard

  • $35* Raspberry Pi 3B+ from Newark or Amazon
  • $10 Power Adapter (2.5 amps minimum!)
  • $7 32GB microSDHC Class 10 card
  • £12.95 Rainbow or Ninja Pibow case or $7.99 Official RasPi 3B+ case
  • Getting Started with Incredible PBX

    Here’s everything to know about installation and setup. "Automatic" means just watch. Steps #1 and #2 are self-explanatory. For the remaining steps, we’ll further document the procedures in the sections below.

    1. Download and unzip Incredible PBX 13-13.10 image from SourceForge
    2. Transfer Incredible PBX image to microSD card
    3. Boot Raspberry Pi from new microSD card (16GB or larger)
    4. Login to RasPi console as root:password to initialize your server (Automatic)
    5. In raspi-config Advanced Options, Expand FileSystem to fill your SD card
    6. Reboot after writing down your server IP address (Automatic)
    7. Login via SSH or Putty as root:password to set passwords & setup firewall (Automatic)
    8. Register for and configure Skyetel for Incredible PBX, if desired
    9. Add Inbound Route for Skyetel, if desired
    10. Install Incredible Fax: /root/incrediblefax13_raspi3.sh (Credentials: admin:password)

    First Boot of Incredible PBX Using Wi-Fi

    Incredible PBX requires Internet connectivity to complete its automated install. If you’re using a wired network connection, you can skip to the next section. With the Raspberry Pi 3B+, WiFi is built into the hardware. But you still have to insert your SSID name and SSID password to make a connection to your WiFi network. To do so, follow these next steps carefully. Insert the Incredible PBX microSD card into your Raspberry Pi 3B+ and apply power to the hardware. When the bootup procedure finishes, login as root with the default password: password. At the first prompt, DO NOT PRESS THE ENTER KEY. Instead, press Ctrl-C to break out of the setup script. At the command prompt, issue the following commands to bring up the WiFi config file:

    cd /etc/wpa_supplicant
    nano -w wpa_supplicant.conf
    

    If your WiFi network does not require a password, then insert the four line below and save the file: Ctrl-X, Y, then Enter. Now restart your server: reboot. When the reboot finishes, you now should have network connectivity.

    network={
     key_mgmt=NONE
     priority=1
    }
    

    If your WiFi network requires a password, scroll down to the SSID entry and replace YourSSID with the actual SSID of your WiFi network. Make sure you preserve the entry with the quotes as shown. Next, replace YourSSIDpassword with the SSID password of your WiFi network. Save the file: Ctrl-X, Y, then Enter. Now restart your server: reboot. When the reboot finishes, you now should have network connectivity.

    Once the reboot process finishes, you should see an entry on about the middle line displayed on your monitor which reads: "My IP address is…". Write down the IP address shown. You’ll need it in a minute. Skip the next section since you are using a WiFi connection.

    If you don’t see an IP address assigned to your server, then correct the network deficiency (invalid WiFi credentials, DHCP not working, Internet down), and reboot until you see an IP address assigned to your server. DO NOT PROCEED WITHOUT AN ASSIGNED IP ADDRESS.

    First Boot of Incredible PBX Using Wired Connection

    Incredible PBX requires Internet connectivity to complete its automated install. After connecting your server to your local network with a network cable, insert the Incredible PBX microSD card into your Raspberry Pi 3B+ and apply power to the hardware. When the bootup procedure finishes, you should see an entry on about the middle line displayed on your monitor which reads: "My IP address is…". Write down the IP address shown. You’ll need it in the next step.

    If you don’t see an IP address assigned to your server, then correct the network deficiency (cable not connected, DHCP not working, Internet down), and reboot until you see an IP address assigned to your server. DO NOT PROCEED WITHOUT AN ASSIGNED IP ADDRESS.

    Completing the Incredible PBX Initialization Procedure

    The remainder of the install procedure should be completed from a desktop PC using SSH or Putty. This will assure that your desktop PC is whitelisted in the Incredible PBX firewall. Using the console to complete the install is NOT recommended as your desktop PC will not be whitelisted in the firewall. This may result in your not being able to log in to your server. Once you have network connectivity, log in to your server as root from a desktop PC using the default password: password. Accept the license agreement by pressing ENTER. You then will be redirected to raspi-config. This is the utility used to expand your Incredible PBX image to use your entire microSD card. If you fail to complete this step, your microSD card will be restricted to 4GB which already is 95% full. In the raspi-config utility, choose item 7 (Advanced Options). All of the defaults should be satisfactory with the exception of the first item: Expand Filesystem. Choose this option and activate the resizing directive. Review the other items and then exit and reboot your server.

    Once your server reboots and you log back in as root, you will be prompted to change all of your passwords. Write them down and put your cheat sheet in a safe place. It’s your only way back into your server without starting over.

    Finally, if your PBX is sitting behind a NAT-based router, you’ll need to redirect incoming UDP 5060 traffic to the private IP address of your PBX. While this isn’t technically necessary to complete calls with registered trunk providers, there are others such as Skyetel that don’t use SIP registrations where failure to redirect UDP 5060 would cause inbound calls to fail.



    The First Login to the Incredible PBX GUI

    The Incredible PBX GUI is accessed using a web browser pointed to the IP address of your server. As part of the password setup, you created an admin password for the Incredible PBX GUI, a.k.a. FreePBX GUI. Login now using your favorite browser. If you have forgotten your admin password, you can reset it by logging into your server as root using SSH: /root/admin-pw-change. Once you’ve logged into the GUI, your first task is to record the public and private IP addresses for your server. This eliminates 99% of the problems with one-way audio on calls where your server is sitting behind a NAT-based router. Navigate to Settings -> SIP Settings and click on Detect Network Settings in the NAT Settings section of the template. Verify that the entries shown are correct and then click Submit followed by Apply Config.

    Managing a PBX with a Dynamic IP Address

    Many Internet service providers assign dynamic IP addresses to customers. This poses issues with a PBX because SIP phones positioned outside your LAN need to be able to connect to the PBX. It also complicates SIP routing which needs both the public IP address and the private IP address of the PBX in order to route calls properly. In the previous section, you configured your PBX with these two IP addresses. The problem, of course, is that this public IP address may change when your ISP assigns dynamic IP addresses. Luckily, many ISPs rarely update dynamic IP addresses of their customers. For example, our home network has had the same dynamic IP address for more than four years. If this is your situation, then you have little to worry about. If the IP address ever changes, you can simply repeat the steps in the previous section. However, if your ISP regularly changes your public IP address, then you need an automatic way to keep your PBX configured properly. Otherwise you will start experiencing calls with one-way audio or no audio, and remote phones will no longer be able to connect to the PBX. We’ve developed a script to update the public IP address of your PBX. Depending upon your situation, all you need to do is run it hourly or daily to keep your PBX configured properly. To begin, first download the updater script after logging into your server as root:

    cd /root
    wget http://incrediblepbx.com/update-externip.tar.gz
    tar zxvf update-externip.tar.gz
    rm -f update-externip.tar.gz
    

    Try running the script once to make sure it correctly identifies the public IP address of your server: /root/update-externip. Then add an entry to the end of /etc/crontab that schedules the script to run at 12:30 a.m. each night:

    30 0 * * * root /root/update-externip > /dev/null 2>&1
    

    Enabling OPUS Codec with Incredible PBX

    @JoeOIVOV on the PIAF Forum has documented a method to activate the OPUS Codec on the Raspberry Pi. From the Linux CLI, issue the following commands while logged in as root:

    cd /usr/lib/asterisk/modules
    wget http://incrediblepbx.com/codec_opus_open_source.so
    

    Then, use a browser to open the Incredible PBX GUI as admin and navigate to Settings -> Asterisk SIP Settings and scroll down to the Audio Codecs section of the template. Place a check mark beside the opus codec option. Then click Submit and Apply Settings.

    Return to the Linux CLI and issue the following commands to complete the setup and verify:

    fwconsole restart
    asterisk -rx "core show codecs"
    

    Special Thanks to: Walter Sonius on SourceForge

    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. The next section of this tutorial will show you the easier way, using Skyetel.

    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.

    Using Skyetel with Incredible PBX

    On the Raspberry Pi platform, all of the Skyetel trunks are preconfigured. All you need to do is sign up for Skyetel service to take advantage of the $10 free credit and Nerd Vittles BOGO 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 a $10 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. Once you’ve had a chance to kick the tires, fund your account with up to $250, and Skyetel will match your deposit. That gets you up to $500 of half-price VoIP service. Once you have funded your account, you can port in your phone numbers for 60 days at no cost. And you can also take advantage of a 10% discount on your current service. Just open another ticket and attach a copy of your last month’s bill. See footnote 2 for the fine print.2 If you have high call volume requirements, document these in your Prequalification Form, and we will be in touch.

    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. You will note that we already have configured a Skyetel template for you. Simply edit the existing entry and plug in the 11-digit phone number (beginning with a 1) of your Skyetel DID . Set the Destination for the incoming DID as desired and click Submit. It defaults to extension 701.

    If your PBX is sitting behind a NAT-based router, you’ll need to redirect incoming UDP 5060 traffic to the private IP address of your PBX. Then 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, it is preconfigured by default on the Raspberry Pi platform so you can use both 10-digit and 11-digit dialing to process outbound calls through your Skyetel account. If you prefer another setup, choose Connectivity -> Outbound Routes.

    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:



    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) which can be a separate board in your computer from a company such as Digium. Or it can be 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.

    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.

    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."

    Transcribing Voicemails with IBM Watson STT Service

    We’ve included the necessary script to transcribe your incoming voicemails using IBM’s STT service. Navigate to the /usr/local/sbin folder and edit sendmailmp3.ibm. Insert your APIKEY in the password field and save the file. Now copy the file to sendmailmp3 and make the file executable: chmod +x sendmailmp3.

    Running Incredible PBX from an External USB Drive

    CAUTION: If you wish to use an external USB-powered drive with your Raspberry Pi to get better performance and enhanced reliability, then you’ll want to stick with the Raspberry Pi B for the time being because the B+ does not yet support booting from an external drive that lacks an independent power source. See this thread for details.

    With older versions of the Raspberry Pi, you may wish to consider an external USB drive to supplement your Incredible PBX for Raspberry Pi setup. If this is a production system on which you depend for important calls, we would highly recommend it. Begin by formatting the USB drive as a DOS FAT32 drive. Then install the Incredible PBX image on the USB drive using the same procedure outlined above for your microSD card. Be sure you choose the correct drive! Now boot your Raspberry Pi with the USB drive plugged in. Login as root and issue the command: mount /dev/sda2 /mnt. Using nano, edit /mnt/etc/fstab. Change /dev/mmcblk0p2 to /dev/sda2 and save the file. Edit /boot/cmdline.txt and change /dev/mmcblk0p2 to /dev/sda2. Then add the following to the end of the line: rootdelay=5. Save the file and reboot your server leaving the microSD card in place.

    As configured, your server will now boot to the external USB drive, but the usable space on the drive will be the original 4GB partition. To expand it, do the following carefully. Log back into your server as root. Issue the command: fdisk -cu /dev/sda. List the partitions on your external drive by typing p. Write down the starting sector number for the sda2 partition. For example, on a 1 terabyte drive, it will be something like 131072. Now delete the sda2 partition by typing d and then choosing 2. Create a new primary partition by typing n then p then 2. When prompted for the starting sector, enter the number you wrote down for the sda2 partition above. Press ENTER. When prompted for the ending sector, just press ENTER to accept the default. Now type w to write your changes to the drive. Reboot. Log back into your server as root and issue the following command to expand the primary partition to use the entire disk: resize2fs /dev/sda2. Verify the new size of your drive: pbxstatus.

    Using Gmail as a SmartHost for SendMail

    Many Internet service providers 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 RasPi 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. In the last step, press ENTER to accept all of the default prompts:

    chmod 600 client-info
    makemap -r hash client-info.db < client-info
    cd ..
    sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/sendmail.mc
    sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/sendmail.mc
    sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/Makefile
    sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/sendmail.cf
    sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/databases
    sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/sendmail.mc.gmail
    sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/sendmail.cf.errors
    make
    sendmailconfig
    

    Finally, stop and restart SendMail and then send yourself a test message. Be sure to check your spam folder!

    /etc/init.d/sendmail stop
    /etc/init.d/sendmail start
    apt-get install mailutils -y
    echo "test" | mail -s testmessage you@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.

    The last step is to add the following command to /etc/rc.local to send you an email with your PBX's IP addresses whenever the RasPi is rebooted. Insert the following one-line command just above the exit 0 line at the end of the file. Replace yourname@yourdomain.com with an email address to which you always have access.

    echo LAN: $(ifconfig | grep "inet addr" | sed 's/^[[:space:]]*//' | sed 's/  .*$//g' | cut -f 2 -d " ")  NET: $(curl -s -S --user-agent "Mozilla/4.0" http://myip.incrediblepbx.com | awk 'NR==2') |  mail -s "Incredible PBX 13-13.10 RasPi IP Address" yourname@yourdomain.com
    

    WebMin: Wherefore Art Thou?

    Some of you may have noticed that WebMin is missing in this new release. For newcomers, WebMin is the Swiss Army Knife of Linux. You can do almost anything to Linux from the convenience of a browser. Unfortunately, in the PBX environment, WebMin is a two-edged sword. You can also permanently ruin your PBX in a matter of seconds if you don't know what you're doing because WebMin hides most of its magic under the covers so you really can't decipher what's been changed. Our best advice to those wanting to use WebMin is to use it as a tool to look (but don't touch) the Linux setup. The other major dilemma for us was that the current Incredible PBX build comes perilously close to filling a 4GB microSD card. And moving to an 8GB card to build a PBX would have doubled the size of the download image. Once you have installed Incredible PBX on a larger microSD card and expanded the filesystem to fill the new card, the commands below will get WebMin installed. Once installed, you can access WebMin with a browser pointed to https://ip-address:9001 using the same root credentials used to login with SSH.

    echo "Installing WebMin..."
    echo "deb http://download.webmin.com/download/repository sarge contrib" \\ 
    > /etc/apt/sources.list.d/sarge.list
    cd /root
    wget http://www.webmin.com/jcameron-key.asc
    apt-key add jcameron-key.asc
    apt-get update
    apt-get install webmin -y
    sed -i 's|10000|9001|g' /etc/webmin/miniserv.conf
    service webmin restart
    

    Implementing Bluetooth Proximity Detection

    You may find it convenient to have your calls transferred when you're away from your desk. The RasPi can do it automatically if you have a smartphone and a RasPi 3B+ with built-in bluetooth support.

    1. Decipher the MAC address of your RasPi's Bluetooth adapter: hcitool dev

    2. Turn on Bluetooth and enable discovery on your smartphone.

    3. Search for your smartphone's MAC address from the RasPi CLI: hcitool scan

    4. Install our Bluetooth Proximity Detection Utility on your RasPi:

    cd /root
    wget http://nerdvittles.com/trixbox123/proximity.zip
    unzip proximity.zip
    chmod +x proximity
    

    5. Edit the proximity script and fill in the blanks using the extension you want to forward when you're not "at home" and the 10-digit number of the smartphone to forward the calls to:

    deviceuser=YourName
    devicemac=Mac:Address:Of:Your:Smartphone (with the colons from step #3)
    myextension=701
    mycellphone=8435551212
    

    6. Add a cron job to /etc/crontab to check for the presence of your cellphone every minute between 6 am and 9 pm:

    * 6-21 * * * root /root/proximity > /dev/null
    

    When you're home, your cellphone obviously must be within range of your Raspberry Pi and you need a working outbound trunk for outbound 10-digit calls for this to work while away.

    /root/proximity:
    
    WARD.now IN - Update Required
    Sat Mar  9 13:51:07 EST 2019
    Database entry removed.
    

    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 13-13.10 server is up and running:

    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. We've included a setup script to make it easy:

    cd /root
    ./tftp-setup
    

    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. Helpful tips on implementation can be found in this PIAF Forum thread.

    Configuring a SIP URI Address for Your PBX

    Setting up a SIP URI is a simple way to let anyone with a SIP phone call you from anywhere in the world and talk for as long and as often as you like FOR FREE. The drawback of SIP URIs is typically the security risk accompanying the SIP exposure you must provide to receive these calls. Here's the safe way using what we call a hybrid SIP URI. It works like this. Sign up for a VoIP.ms account and create a subaccount that you will register using the VoIPms trunk included in Incredible PBX. As part of the setup in the VoIP.ms portal, assign an Internal Extension Number to your subaccount, e.g. 789123. Make it random so you don't get surprise calls from anonymous sources. The extension can be up to 10 digits long. Next, sign up for a free iNUM DID, e.g. 883510009901234, in your VoIP.ms account. Using Manage DIDs in the portal, link the iNUM DID to your subaccount and assign one of the VoIP.ms POP locations for incoming calls, e.g. atlanta.voip.ms. Next, write down your VoIP.ms account number, e.g. 12345. Once you've completed these three steps and registered the VoIP.ms subaccount on your PBX, you now have two SIP URIs that are protected by your VoIP.ms credentials and don't require you to expose your SIP port to the outside world at all. These SIP URIs can be pointed to different destinations by setting up Inbound Routes using your VoIP.ms account number as one DID and setting up your iNUM number as the second DID. To reach your PBX via SIP URI, callers can use 12345789123@atlanta.voip.ms to reach the DID you set up for your VoIP.ms subaccount where 12345 is your VoIP.ms account number and 789123 is the Internal Extension Number for your subaccount. Or callers can use 8835100099012234@inum.net to reach the DID you set up using your iNUM number that was assigned by VoIP.ms. Don't forget to whitelist the VoIP.ms POP's FQDN for SIP UDP access to your PBX:

    /root/add-fqdn voipms atlanta.voip.ms

    If you wish to make SIP URI calls yourself, the easiest way is to first set up a free LinPhone SIP Account. You can find dozens of recommendations for hardware-based SIP phones both on Nerd Vittles and the PIAF Forum. For today we'll get you started with one of our favorite (free) softphones, YateClient. It's available for almost all desktop platforms. Download YateClient from here. Run YateClient once you’ve installed it and enter the credentials for your LinPhone account. You’ll need LinPhone's FQDN (sip.linphone.org) plus your LinPhone account name and password. Fill in the Yate Client template and click OK to save your entries. Once the Yate softphone shows that it is registered, try a test call to one of our demo SIP URIs: sip:weather@demo.nerdvittles.com or sip:news@demo.nerdvittles.com.

    Adding the NeoRouter Virtual Private Network

    We've made it easy to set up a virtual private network between your PBX and your other computers. NeoRouter is a free VPN for up to 256 machines. It requires that you first set up a server for NeoRouter using a static IP address and preferably a fully-qualified domain name. This is covered in this Nerd Vittles tutorial. Once you have your NeoRouter server operational, adding your PBX to the VPN is easy. Simply run nrclientcmd and enter the FQDN of your VPN server together with your credentials. All clients on the VPN have an encrypted tunnel with private LAN addresses in the 10.0.0.x range. HINT: Setting up a NeoRouter VPN provides an easy way to get back into your server if the firewall ever locks you out since the 10.0.0.x subnet is automatically whitelisted as part of the initial install.

    Using PortKnocker to Regain Access to Your PBX

    And speaking of getting locked out of your server because you've forgotten to whitelist the IP address of your computer, there's another easy way to regain access: PortKnocker. The way the service works is you send sequential pings to 3 randomized TCP ports that are known only by you. They are listed in /etc/knock.FAQ. When your server detects a match, it will whitelist your new IP address allowing you to login using SSH or Putty. There also are PortKnocker utilities for both iOS and Android devices. Complete implementation details are available in this Nerd Vittles tutorial. If your PBX is sitting behind a router or firewall, don't forget to forward the three TCP ports from your router to the private LAN address of your PBX.

    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. There's an Incredible PBX 13-13.10 Vbox image already available on SourceForge. Don't use the Vbox image for Raspbian. It has insufficient available disk space to support the new backups. Once you've downloaded the Vbox image, double-click on it to install. Then fire up the virtual machine, login as root with password as your password and install the latest Incredible Backup and Restore scripts

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

    Next, create a /backup folder on your new 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.

    The Million Dollar Question, of course, is whether you can put Humpty back together again by installing a fresh Incredible PBX 13-13.10 Raspbian image to a new microSD card, going through the basic initialization steps 1-7 on your Raspberry Pi, and then copying the backup image from the VirtualBox desktop machine back over to the new Raspbian PBX and restoring it. And the answer is A-B-S-O-L-U-T-E-L-Y. In fact, you can even make changes in the VirtualBox GUI, create a fresh backup, and then restore that image to your Raspberry Pi. Keep in mind our original caveat that, if you add components, packages, or applications to your primary server, those same additions need to be made to the secondary platform since they will not get picked up as part of the backup. Try it for yourself. And sleep well.

    Originally published: Monday, March 11, 2019



    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.

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    The 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.

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    Special 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.
     



    1. Many of our purchase links refer users to Amazon when we find their prices are competitive for the recommended products. Nerd Vittles receives a small referral fee from Amazon to help cover the costs of our blog. We never recommend particular products solely to generate Amazon commissions. However, when pricing is comparable or availability is favorable, we support Amazon because Amazon supports us. []
    2. 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. []

    Keep On Trunkin’: Free International VoIP Calling Returns



    Today we’re taking a fresh look at the international calling marketplace by updating the best VoIP deals available. FreeVoipDeal once again takes the prize with the best selection of "free" international calling destinations at the lowest prices. Below we’ll provide a quick tutorial to transform your Incredible PBX server into an international calling platform at minimal cost.

    Here’s How It Works. For every 10 euros ($10.72) you deposit into your account, you’ll get 300 minutes a week of free calls to a specific list of countries for 120 days. After you exhaust your free minutes, calls to the "free" countries revert to their standard VoIP rates. You can also call anywhere else in the world at very reasonable per minute rates that compare favorably with other SIP providers around the world. The beauty of a PBX and SIP trunks is you can mix and match as many providers as you like to take advantage of favorable calling rates to multiple countries. We’ll walk you through the FreeVoipDeal trunk setup below.

    Betamax 101. There are a few things you need to know about the so-called Betamax VoIP services up front. Most importantly, they change rates and free countries more frequently than college kids change partners. The calling rate to some country from some Betamax provider changes almost every day because Betamax has dozens of companies offering similar services with differing rates and freebies. Here’s an very old spreadsheet that will give you a good idea of what you’re up against. Don’t depend upon it for the current rates. You’ll need to visit the actual site(s) for their current rate tables or visit this site (not) maintained by Betamax for a country-by-country comparison by provider. That’s another way of saying DON’T BLAME US IF YOUR 3-HOUR CALL TO ANTARCTICA CHANGED FROM 20¢ PER MINUTE TO $1 PER MINUTE OVERNIGHT. IT PROBABLY WON’T, BUT IT MIGHT.

    One other word of warning. Some Betamax sites (marked with a red asterisk in the Betamax country table) such as powervoip.com have good calling rates, but they tack on a 3.9¢ connection fee to every call. If you make lengthy calls, it’s not a big deal. If you make numerous short calls, it drives your discount calling rates through the roof. Before making a lengthy call to a remote destination, spend the two minutes it takes to look up the current rate on the actual Betamax web site and take a snapshot of the page for your records. Here’s another tip. If you make frequent calls to Antarctica, spend a little time doing your homework. Review the latest Betamax spreadsheet to track down the cheapest rates. Then double-check the actual sites for the current rates. There’s a $100+ difference in the cost of a 3-hour call at €.20/minute from some Betamax sites versus the €.70/minute rate at some other Betamax sites. THIS OFTEN CHANGES! HINT: Don’t use FreeVoipDeal for Antarctica.

    Today we’ll be focusing on the company we’ve tracked for many years, FreeVoipDeal.com. Except for the domain name, the setup with other Betamax providers is similar but not identical. And, of course, you’ll have to kick in another deposit to make free calls from each site. The length of the Freebie period also may vary so read the terms carefully. FreeVoipDeal actually hasn’t changed much since our first visit about five years ago. In fact, we still had most of our ten euro credit so we could play all we wanted even though the calls were no longer free since our four month window has long since expired.

    Here’s the February 23, 2019 Freebie list by country. Don’t depend upon it! Check their actual web site or the Betamax country summary for current freebies and current rates. Here’s a great trick to remember. When you visit the FreeVoipDeal Rate Table, click on the Out of Minutes tab for a quick listing of all the Free Calling Countries as well as the rates once you’ve used up your four months or 300 weekly minutes of free calls. With few exceptions, most of the "free countries" still have a rate of 1.1¢ per minute even after you run out of minutes.

     

    How Free International Calling Works

    Placing international calls through FreeVoipDeal can be done in a number of ways. That’s the real beauty of a PBX. First, you can either load an app to make the calls if your smartphone or PC supports it. With Incredible PBX, you can use a SIP phone to dial a FreeVoipDeal number directly through your PBX, or you can dial a DISA access number or SIP URI from anywhere to connect to your PBX and then enter your DISA password after which you will get a second dial tone to place an international call using your FreeVoipDeal trunk. The beauty of the DISA approach is you can call into your PBX from any telephone to place free or dirt cheap international calls.

    Using Incredible PBX 13 and DISA for Calling

    On the Incredible PBX platform, you can use the DISA application to provide secondary dialtone for processing international calls. A phone number and trunk will receive incoming calls bound for DISA from your cellphone. An inbound route will only forward incoming calls to DISA that match your cellphone number. A secondary trunk from FreeVoipDeal or other providers will be used to process outgoing international calls that are dialed using DISA. We’ll create an outbound route or rule for every country to which you want to authorize international calling. Each of these outbound routes will point to the least expensive (or free) trunk to complete the call. In the VoIP world, you actually could have dozens of outbound trunks that handle international calls based upon the country codes of each international call. This lets you take advantage of the best calling rates for each country. We will block international calls to country codes not specifically authorized.

    Just to restate the obvious, a misconfigured DISA application that allows the world to make international calls on your nickel can get expensive quickly. We’ll protect today’s DISA setup for Incredible PBX with three layers of protection. First, we’ll require that the CallerID of the incoming call match your cellphone number. While this isn’t failsafe since CallerID numbers can be spoofed, it does reduce the risk considerably. Second, to make DISA calls, you’ll have to know the incoming phone number or SIP URI managing DISA on your PBX. And third, you’ll have to enter the correct DISA PIN before being prompted for an international number to dial. Without all three, nobody gets to make an international call on your nickel. Just remember, compromising DISA on your PBX is just as risky as handing out your credit card to a stranger so follow the setup steps below carefully. And then TEST, TEST, TEST to make sure strangers can’t access your DISA setup. We’ll show you how.

    Here’s an overview of the DISA setup drill once you have Incredible PBX running. We’ll walk through each of the six steps below. Don’t get frustrated. There are a number of steps, but none of them are difficult. Just pretend you’re baking cookies and don’t skip any steps.

    1. Set Up Your Trunk to Process Incoming DISA Calls
    2. Set Up Your Trunk(s) to Process Outgoing International Calls
    3. Configure DISA with a Very Secure Password
    4. Configure an Inbound Route to Limit Incoming DISA Calls to Your Cellphone #
    5. Configure an Outbound Route for Each International Country Code
    6. Test, Test, Test

    1. Setting Up Incoming DISA Call Trunk

    Before you can make calls to your PBX, it’ll need a phone number (known affectionately as a DID). As installed, Incredible PBX includes preconfigured SIP trunks from about a dozen SIP providers. All you’ll need is credentials from the company you wish to use. You can obtain a free DID here. To obtain your own SIP URI, read our tutorial.

    2. Trunk Setup for International Calling

    We’re going to walk you through setting up a trunk with FreeVoipDeal to handle free international calls to certain countries documented above. This may not be the best fit for you depending upon the international destinations you wish to call. Figure that out first! Then adjust the trunk settings below to match each SIP provider trunk you wish to create. There’s no limit to the number you can have. And, with most of these providers, you pay by the minute for international calls anyway so there is no harm in configuring multiple trunks to take advantage of the best rates calling the countries of your choice. The same applies to all-you-can-eat and "free" trunks except there are varying fees for using the services so you’re probably not going to want a dozen of them even if some of the calls are free after making a periodic deposit. Start with the pink and green entries on the old spreadsheet we referenced for the cheapest historical rates and then visit the actual sites and read the fine print.

    To add new trunks to Incredible PBX, use a browser to access the IP address of your server. Login with the default username of admin and the password that you set when your install completed. You can change it with the admin-pw-change script in /root. Once the dashboard appears, click the Connectivity tab and choose Trunks -> Add SIP (chan_sip) Trunk.

    For Trunk Name, enter FreeVoipDeal. In the Dialed Number Manipulation Rules section, add a rule for each country code you wish to activate. You can decipher the Country Code for any country at this link. For example, for the United Kingdom, you’d enter a rule like this where 44 is the Country Code and each X represents a required digit in the local area code and phone number. The trailing period means the number includes one or more additional digits. NOTE: DISA calls will not have to be prefixed with 011 to place international calls. Just enter the country code and number to be called. And, we are told that only 441, 442, and perhaps 443 calls to the U.K. are free since those are the designated landline prefixes.




    If there are other countries, you wish to support with this trunk provider, you’d click Add More Dial Pattern Fields and insert an additional rule for each country following the example above. If you’ll be using this trunk to make calls in the U.S. and Canada as well, the correct Match Pattern is 1NXXNXXXXXX, and calls will need to be dialed with the 1 to avoid conflicts with international dialing.

    Next, we need to enter the Outgoing Settings. For the Trunk Name, enter freevoipdeal. Clear out the entries in Peer Details section and enter the following using your actual FreeVoipDeal credentials for yourusername and yourpassword:

    authuser=yourusername
    username=yourusername
    secret=yourpassword
    type=peer
    qualify=yes
    nat=yes
    insecure=port,invite
    host=sip.freevoipdeal.com
    fromdomain=sip.freevoipdeal.com
    dtmfmode=auto
    disallow=all
    canreinvite=no
    allow=alaw&ulaw
    

    Finally, clear out the default entries in User Details and click the Submit Changes button and then red Apply Config button to save your new trunk.

    Spoofing Your CallerID. When setting up your FreeVoipDeal account, you can set up one or more numbers to use as your CallerID number on FreeVoipDeal calls. You simply verify the number with a code sent by SMS or phone call from their service. Once you’ve gone through the verification procedure, you can spoof the outbound CallerID on FreeVoipDeal calls using your actual cellphone number. Just add the following entries to your Trunk settings replacing 9991234567 with your cellphone number. Special thanks to @hillclimber on the PIAF Forum for the tip.

    fromuser=0019991234567
    sendrpid=yes
    

    3. Configuring DISA for International Calling

    In the Incredible PBX GUI, we’ll set up DISA by clicking the Applications tab and choosing DISA. Add your new DISA configuration by following this sample. Use a VERY secure password. It’s your phone bill. Once you’ve finished, click the Submit Changes button and then the Apply Config button to save your new DISA setup.



    4. Inbound Routing of DISA Calls

    Here’s where we lock down your setup so that Incredible PBX only accepts DISA calls from your cellphone number. If you want to allow additional people to use your DISA setup or if you have multiple cellphones, then simply create multiple inbound routes with the 10-digit numbers of each phone to be supported.

    In the Incredible PBX GUI, we’ll set up a new Inbound Route by clicking the Connectivity tab and choosing Inbound Routes. If you plan to support multiple phones, then create multiple inbound routes and give each of them a unique Description and CallerID Number that matches the phone number of the cellphone to be supported. Be sure to check the CID Priority Route checkbox and set the correct Destination for your incoming calls. Just fill in the blanks appropriately using this template as a guide. Once you’ve finished, click the Submit button and then the Apply Config button to save your new Inbound Route.



    5. Outbound Routing by Country Code

    The DISA application is going to obtain the phone number to be dialed and will pass that to the Outbound Routes module. The job of the Outbound Routes module is to examine the phone number passed to it from DISA to figure out which trunk to use to make the outbound call. It then will pass the call to the appropriate trunk which sends the outgoing call on its way to the destination.

    For each Dialed Number Manipulation Rule in every Trunk that you set up in Step #2 above, you’ll need a matching Outbound Route if your PBX is used to place calls using multiple trunks. If you’re only using one provider for all of your outbound calls, then we can use a more generic Outbound Route. It’s always a good idea to create the one-to-one match between Outbound Routes and Trunks to make certain that outbound calls are sent to the correct Trunk for processing. So let’s do that using the U.K. trunk we created above.

    In the Incredible PBX GUI, we’ll set up a new Outbound Route by clicking the Connectivity tab and choosing Outbound Routes. When the template appears, notice in the far right column that there’s a listing of all your existing Outbound Routes. Calls are actually processed sequentially using the order that these Outbound Routes appear in the list. If there’s no number match in the top route, processing drops to the next route in the list until there is a match AND a successful connection. You can adjust the sequence by dragging the Outbound Routes to a different position in the priority list.

    It’s important to use specificity in your Outbound Routes (especially with International calling) to make certain that a call isn’t inadvertently processed by some other trunk. The easiest way to do this is to require the Outbound Route Match Pattern for U.K. calls to be at least 11 digits, e.g. 44XXXXXXXX. (the trailing period is important in that it requires at least one more digit for a match). And we can force a Hangup if the FreeVoipDeal trunk is not available for some reason by adjusting the Destination on Congestion setting. This keeps the call routing from dropping down to the next available Outbound Route in the list if FreeVoipDeal happens to be off-line at some point. So our Outbound Route for U.K. calls should look something like this:



    The final step is to move the new Outbound Route for U.K. calls to the top of the Outbound Routes listing in the right column to assure that it is processed first. Once you’ve done that, click the Submit Changes button and then the Apply Config button to save your new Outbound Route AND the adjusted Outbound Route Priority List.

    Another alternative in creating Outbound Routes is to use a Dial Prefix that never matches a real phone number to direct calls to a particular trunk. For example, you might use *8 as a dial prefix for FreeVoipDeal calls. By placing *8 in the Prefix column of the Dial Pattern, it will get stripped off before the number is actually passed to the FreeVoipDeal trunk for processing. We actually prefer this setup because it adds an additional layer of security for international calls. If someone were to break into your DISA application by knowing your cellphone number AND your DID AND your DISA password, it’s unlikely they’d also know to prefix outgoing international calls with some arbitrary dial prefix. Just don’t use *8 in case they’re a Nerd Vittles reader. 😉

    6. Test, Test, Test!

    The easiest way to test the new setup is to place a couple of calls and to watch the Asterisk CLI (asterisk -rvvvvvvvvvv) and see how the calls are processed and who answers at the other end. Then you can apologize for reaching the wrong number.

    You can make up your own test methodology, but here’s one that works for us. There are several tests you need to make. First, call your Incredible PBX DID from your authorized cellphone and enter a correct DISA password to see if you get dial tone to make an international call. Then repeat the drill with an invalid password and make sure you don’t get a dial tone. Next, call your Incredible PBX DID from a phone other than your authorized cellphone. You should not get a prompt for a DISA password. Finally, we use the first three digits of a U.K. number to identify a matching NANPA area code. Then, we find hotels in the two matching cities. For example, one might attempt to call a hotel in Bath, England (44 1… ……) and a hotel in Bermuda (441-…-….). The U.K. call should go through, and the Bermuda call should fail. If you pass all three tests with flying colors, you’re good to go.

    Using FreeVoipDeal’s MobileVoIP App

    FreeVoipDeal also offers a MobileVoIP app that can be used directly on your smartphone (Android, iOS, and Windows phone versions available) using any Wi-Fi, UMTS, 4G/LTE, 3G, GPRS or EDGE connection. The drawback is the lack of the three extra layers of security protection that Incredible PBX using DISA offers. MobileVOIP lets you log in with your registered Betamax credentials and offers the option to use your existing VoIP credit from your smartphone. The downside is that anyone with the app and your credentials can call anywhere and talk for as long as they like on your nickel using any of your registered CallerIDs. You’ve been warned. For more information or to download the app for your mobile device, go here. Remember to dial the "+1″ country code prefix for U.S./Canada calls.

    Originally published: Monday, April 24, 2017  Updated: Monday, February 25, 2019


    Support 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.



    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.

    BOGO 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.

    The 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.

    VitalPBX 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!
     

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



    Some Recent Nerd Vittles Articles of Interest…

    Big Kahuna: 70 New FreePBX GPL Modules for Incredible PBX



    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
    




    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.



    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.

       

    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



    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.

    BOGO 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.

    The 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.

    VitalPBX 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!
     

    Special 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.
     



    Adding SIP URI Dialing to Asterisk for Free Worldwide Calling



    Since giving up on Google Voice, we’ve been extolling the virtues of SIP URI dialing which gives you unlimited free calls to anyone else in the world that happens to have their own SIP URI address. SIP URIs look very much like email addresses except they’re used to share phone conversations instead of email messages. And, as we’ve mentioned previously, if everyone in the world had their own SIP URI, paying for phone calls would become a thing of the past. We hope you’ll join us in making that happen. As a fallback, give our $50 credit at Skyetel a try.

    One of the drawbacks of Asterisk® PBXs using the FreePBX® GUI has been the inability to place outbound SIP URI calls from SIP phones registered as extensions on the PBX. Today we first want to address that shortcoming. Our SIP URI dialing solution for Asterisk should work with any FreePBX-based implementation including Incredible PBX® and Issabel as well as on Raspberry Pi platforms. We’ll wrap things up by providing some tips on obtaining and deploying your own SIP URI at little or no cost and pointing you to some excellent resources that facilitate calling millions of SIP phones around the world at zero cost. All you need is an Internet connection, and we’ll point you to a terrific softphone to begin your adventure.

    Let’s begin by examining why SIP URI dialing is a problem with FreePBX. The reason is pretty simple. FreePBX interprets dial strings by matching them against some rules to determine whether you’re making an internal call or a call outside your PBX. It matches internal calls against a list of available internal extensions. External calls are matched against rules defined in your outbound routes which are associated with trunks. Since SIP URI calls don’t match any extension or outbound route, the caller receives a congestion tone.

    The traditional workaround has been to define a custom extension using the FreePBX GUI which points to a SIP URI. Then the user can dial the custom extension, and the call will be routed to the defined SIP URI. These custom extensions also can be defined in extensions_custom.conf within the from-internal-custom context. For example, the following dialplan code would let users dial 411 to reach AT&T’s Toll-Free Directory Assistance: exten => 411,1,18005551212@switch.starcompartners.com.

    But there’s a better way. Wouldn’t it be nice to be able to dial any SIP URI from a softphone or to store SIP URI addresses in the phonebook of your SIP phone?1 Well, now you can. Before we actually put the dialplan code in place, let us explain how this will work. First, FreePBX still needs to be able to distinguish a SIP URI call from a "regular call." The reason this gets tricky is because Asterisk typically throws away the destination hostname when you place a call. For example, calls to 8005551212 and 8005551212@sip2sip.info are processed by Asterisk in exactly the same way, i.e. dropping the host address before dialing.

    Using the new dialplan code in the next section, here’s how calls will be processed:

    User dials                    Asterisk processes call as
    ------------------------      ---------------------------------------------
    701                           internal call to local extension 701
    4045551212                    external call using NXXNXXXXXX outbound route
    2233435945@sip2sip.info       SIP URI call to Lenny by acct at sip2sip.info
    lennybgood@sip2sip.info       SIP URI call to alias lennybgood@sip2sip.info 
    

    Cautionary Notes: Our code should work fine with any Asterisk 13 and FreePBX 13 or Incredible PBX deployment on any Linux platform; however, with servers other than Incredible PBX, make sure you have added the following entries to sip_general_custom.conf, or you can configure them in the GUI by making the changes in Settings -> Asterisk SIP Settings -> Chan SIP Settings:

    srvlookup=yes
    allowguest=yes
    

    You also need to test a traditional outbound call (e.g. 8005551212) immediately after you finish the install procedure. Monitor the Asterisk CLI (asterisk -rvvvvvvvvvv) and observe the first few lines of the log after you place a call. The second line will show SIPDOMAIN which should be either the FQDN of your server or an IP address depending upon how you registered your softphone extension. The first line should display the MyDomain variable. If it is empty or doesn’t match the SIPDOMAIN entry, the outbound call will fail. To fix it, add an entry to the Asterisk database from the Asterisk CLI using syntax like the following: database put MyDomain FQDN 10.0.0.11 or database put MyDomain FQDN sip.me.com where 10.0.0.11 or sip.me.com matches the SIPDOMAIN entry shown on the second line. Then retry your outbound call, and it should complete successfully. We’ve tested this back to the early Asterisk 11 days with FreePBX 2.11 without any problems. If your calls still fail, then you will probably need to remove the new code from your platform until you upgrade to a more current version of Asterisk and FreePBX. The code hasn’t been tested with FreePBX 14 and 15.

    Finally, you may want to manually set the CallerID for your outgoing SIP URI calls. From the Asterisk CLI, issue a command for every extension from which you will be placing SIP URI calls, e.g. extension 701 syntax: database put 701 user_sipname "Nerd Uno"

    Enabling SIP URI Dialing with FreePBX

    To enable SIP URI dialing from phones registered with your Asterisk PBX, we’ll modify the dialplan in order to detect SIP URI dial strings entered into a softphone or retrieved from a phonebook associated with almost any SIP phone. When a SIP URI dial string is detected, we’ll send the call out as requested rather than passing the call through the outbound routes and trunks associated with your PBX. All of this dialplan code is open source and is licensed pursuant to the GPL2 license.

    SECURITY ALERT: Never use the SIP URI MOD on a server with a publicly-exposed SIP port as it is possible for some nefarious individual to spoof your FQDN in the headers of a SIP packet and easily gain outbound calling access using your server’s trunk credentials.

    FEB. 21 UPDATE: There was a bug in the original code which caused some internal calls to fail including calls to a DISA extension. Simply install the application again, and it will overwrite the previous version.

    MAR. 5 UPDATE: A bug was discovered in previous releases that treated 911 and 933 calls as internal calls when, in fact, they should have been routed out using your outbound trunks. Simply install the application again, and it will overwrite the previous version.

    MAR. 13 ALERT: This software is not compatible with the Debian, Raspbian, and Ubuntu platforms.

    To begin or update your installation, log in to your PBX as root using SSH or Putty and issue these commands:

    cd /tmp
    wget http://incrediblepbx.com/sipuri-mod.tar.gz
    tar zxvf sipuri-mod.tar.gz
    rm -f sipuri-mod.tar.gz
    ./install-sip-uri-mod.sh
    

    Obtaining Your Own SIP URI

    There are a number of ways to obtain your own SIP URI. Perhaps the easiest is to set up the open Incredible PBX cloud platform that we introduced several weeks ago. Then you can create as many SIP URIs as you like, and they can be used to perform any task that’s available with Asterisk. If you’re not quite ready to make that leap, a free or almost free SIP URI is available from the following sources. VoIP.ms provides a SIP URI for every subaccount you create. Just set up an internal extension number for the subaccount, and that becomes a SIP URI to connect back to your registered server or SIP phone. In the alternative, VoIP.ms will also provide you with a free iNUM DID which can be reached at the following IP address: 81.201.82.50. CallCentric provides a SIP URI matching your account number which can be reached @in.callcentric.com. CallCentric will also provide you with a free iNUM DID which can be reached at the following IP address: 81.201.82.50. LocalPhone provides the same two options as CallCentric: you can be reached by your account number @localphone.com. Or the LocalPhone-assigned iNUM DID can be reached @81.201.82.50. Then there’s pbxes.org. Your account name can be used for SIP URI access @pbxes.org. And, of course, if you’re a 3CX user, you can set up a SIP URI for each extension on your PBX. Just navigate to the Options tab of the desired extension(s) and enter a unique SIP ID for each extension. The SIP URI becomes SIPID@YOUR-3CX-FQDN. SIP URI calls to 3CX Clients on smartphones are also free! This list is not exhaustive. There are now more than 2,000 VoIP networks that support SIP URI access. Using a SIP URI dialing prefix, call any of the referenced networks @sipbbroker.com.2

    Choosing a SIP Phone or Softphone

    You can find dozens of recommendations for hardware-based SIP phones both on Nerd Vittles and the PIAF Forum. For today we’ll get you started with one of our favorite (free) softphones, YateClient. It’s available for almost all desktop platforms. Download YateClient from here. Run YateClient once you’ve installed it and enter the credentials for an extension on your PBX. You’ll need the IP address of your server plus your extension number and its password. Fill in the Yate Client template using the IP address of your PBX as well as your extension credentials. Click OK to save your entries.

    Once the Yate softphone shows that it is registered, try a test call to Lenny using one of the following SIP URIs: 2233435945@sip2sip.info or 883510001198938@81.201.82.50. Better yet, try out a few Incredible PBX samples from the public server we previously deployed:

    Yahoo News Headlines    - news@demo.nerdvittles.com
    Weather by Zip Code     - weather@demo.nerdvittles.com
    Directory Assistance    - information@demo.nerdvittles.com
    Lenny for Telemarketers - lenny@demo.nerdvittles.com
    

    Originally published: Monday, February 11, 2019



    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.

    BOGO 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.

    The 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.

    VitalPBX 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!
     

    Special 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.
     



    1. Special thanks to Olivier Adler and voip-info.org for their early work on SIP URI dialing with Asterisk. []
    2. Some of our links refer users to sites or service providers when we find their prices are competitive for the recommended products. Nerd Vittles receives a small referral fee from these providers to help cover the costs of our blog. We never recommend particular products solely to generate commissions. []

    Introducing Skyetel: A VoIP Provider for All Seasons

    Having been around the block more times than we can remember, suffice it to say it takes a lot to get us excited about a VoIP provider. Let us tick off some criteria to even get our attention: terrific pricing, failsafe reliability, and first class performance. So just imagine our excitement to discover that an early follower of Nerd Vittles now provides one of the most compelling VoIP services we’ve ever tested with triple redundancy in multiple data centers. And Skyetel now has added what, for some, was the most important piece: support for VoIP servers with dynamic IP addresses. While it’s still beta code, it’s easy to use and reliable. There’s yet another hidden benefit. Incredible PBX coupled with Skyetel makes a perfect platform for redundant servers. We’ll cover it in a future article, but here’s the basic design.

    Let’s sweeten the pot a bit more. We were looking for a service provider that could offer a compelling price for the hobbyist and home user while also having the depth to provide millions of minutes to organizations and resellers that actually have such a need. Skyetel now offers Nerd Vittles readers two special offers. First, you can claim a $10 credit for your new account simply by opening a ticket once you sign up. Once you have kicked the tires and are satisfied with the service, you won’t want to miss the Nerd Vittles BOGO offer. Skyetel will match your original deposit up to $250. Deposit $50 and Skyetel will double it. Or plan ahead with a $250 deposit and Skyetel will still double it. That translates into $500 of half-price VoIP service! Once you have funded your account with your money, Skyetel will provide 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 Effective 10/1/2023, $25/month minimum spend required. For resellers and high volume users, document your requirements on your Nerd Vittles signup form and let us put you in touch with someone at Skyetel that will make you a deal you can’t refuse. And what does Nerd Vittles get out of this? Glad you asked. We’re delighted to have Skyetel as a platinum sponsor to keep the lights burning and the deals flowing for another decade of articles and open source offerings for our dedicated followers.

    Original Skyetel DepositSkyetel Deposit MatchAvailable SIP Service $'s
    $20$20$40
    $50$50$100
    $100$100$200
    $200$200$400
    $250$250$500

    We want to also address the elephant in the room. Some have asked about our relationship with Vitelity, a long time sponsor of Nerd Vittles and our open source projects. They’re alive and well. However, the company has gone through several acquisitions in the past few years, and their focus now has shifted more to the reseller and wholesale market. ALL EXISTING VITELITY CUSTOMERS ARE UNAFFECTED BY THIS CHANGE IN DIRECTION. And we are more than happy to put new resellers and wholesalers in touch with someone at Vitelity that can address your requirements. The good news is that you’ll now have two companies to compare while new home users and small businesses have a viable alternative moving forward.

    Skyetel’s State-of-the-Art Network Design

    Because Skyetel’s system architecture is radically different from most other VoIP providers, we wanted to spend a minute documenting their setup. Typically, a VoIP provider may offer a failover server in case their primary server fails. But all calls flow through the primary server unless there is a system failure. As we noted previously, Skyetel’s current setup includes three redundant data centers, all of which receive incoming calls while being firewalled from each other. Once you place or receive a call from the Skyetel network, their data center is completely removed from the audio path of the call which flows directly between your server and the outside party. Thus, even if the data center experienced a total system failure in the middle of your call, neither you nor the other party would ever know it. This design also eliminates the potential of a man-in-the-middle attack from your VoIP provider’s server.

    Skyetel Pricing Overview

    This summary is not intended to be an exhaustive listing of all Skyetel services. Follow this link for a complete summary of fees and services. Traditional DIDs are $1 per month. Toll free numbers an additional 20¢ per month. Outbound conversational calls are $0.012 per minute. DIDs can be SMS/MMS enabled for 10¢ per month. E911 service is $1.50 per month. Incoming conversational calls are a penny a minute. CallerID lookups are $0.004 per call. Voicemail transcription is available for 10¢ per message.

    Signing Up for Skyetel Service

    So here’s the drill to sign up for Skyetel service and take advantage of the Nerd Vittles specials. 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 your free $10 credit to kick the tires. You cannot port in numbers at no cost until you actually fund your account out of your own pocket. Once you have funded your account, open another ticket for the BOGO credit for your account by referencing the Nerd Vittles special offer. You then can initiate your free number porting requests on the portal and request a credit for the porting fees. BOGO credit is limited to one per person/company/address/location. If you want to take advantage of the 10% discount on your current service, 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!

    For those that may be concerned that one day, after your credit expires, you could be paying a penny a minute for phone calls, let me provide a little Ma Bell history lesson for you. When my roommate and I were in law school, our typical phone bill often exceeded $200 a month because we both had girlfriends a couple hundred miles up the road. In today’s dollars, that phone bill translates into roughly $1,200 a month. That would have been 120,000 minutes a month at a penny a minute in today’s dollars. So, yes, VoIP is having a profound influence on the AT&T and Verizon Bell Sisters.

    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: MyPBX
    • Priority: 1
    • IP Address: PBX-Public-IP-Address
    • Port: 5060
    • Protocol: UDP
    • Description: server1.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 Firewall Setup for Skyetel

    The Travelin’ Man 3 firewall included with all Incredible PBX platforms limits access to your server based upon whitelisted IP addresses of outside providers and users. In order to receive calls from the multiple Skyetel data centers, the following entries need to be included in the whitelist of your PBX. For new installs of Incredible PBX 13-13 for CentOS, the entries already are included. Otherwise, issue the following commands from the Linux CLI and choose the 0 option using the add-ip utility in /root:

    • /root/add-ip Skyetel-NW 52.41.52.34
    • /root/add-ip Skyetel-SW 52.8.201.128
    • /root/add-ip Skyetel-NE 52.60.138.31
    • /root/add-ip Skyetel-SE 50.17.48.216
    • /root/add-ip Skyetel-EU 35.156.192.164

    NOTE: If your PBX is sitting behind a NAT-based router, then you will also need to forward UDP port 5060 from your router to the internal IP address of your PBX. Otherwise, incoming calls from Skyetel will fail. You also may need to add a NAT=yes entry to each of the Skyetel trunk configurations using the GUI. The telltale sign that the NAT entry is required will be incoming calls with one-way or no audio.

    Incredible PBX Trunk Setups for Skyetel

    Because Skyetel uses multiple data centers without trunk registrations, you’ll actually need to configure 6 separate Skyetel trunks in the Incredible PBX GUI. The same setup applies for those using generic FreePBX aggregations. We’ve created a script to create all of the trunks for you. Just issue the following commands. The last command assures that you don’t accidentally run the script a second time which would cause all sorts of issues. Feel free to review the code if you want to learn how to create trunks in FreePBX from the command line.

    cd /root
    wget http://incrediblepbx.com/add-skyetel
    chmod +x add-skyetel
    # uncomment next line if your incoming calls all have 10-digit numbers
    # sed -i 's|from-trunk|from-pstn-e164-us|' add-skyetel
    ./add-skyetel
    chmod -x add-skyetel
    

    Incredible PBX Inbound Routing for 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 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.

    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 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:

    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.

    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:



    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 a 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 in 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"

    SMS and MMS Messaging with Postcards

    Skyetel now has released a terrific, open source Docker app, Postcards, that lets you build an SMS and MMS messaging platform for your entire organization. Suffice it to say, anything you ever wanted to do with SMS and MMS messaging, you can do with Postcards. We won’t repeat Skyetel’s excellent tutorial, but you certainly need to visit their site and take Postcards for a spin.

    NEW: Skyetel Support for Dynamic IP Addresses

    You asked for it, and Skyetel has delivered. For Nerd Vittles users running servers with dynamic IP addresses, Skyetel now provides support for your platform. Log into your server as root and cd /usr/src. Then review this tutorial which describes the steps to put the pieces in place. Be advised that this is beta software at this juncture. If you run into issues, please post your questions on the PIAF Forum. Here are the actual steps:

    (1) Log in to your Skyetel portal and Add a New Endpoint Group for your server giving it the name and current public IP address of your server.

    (2) While still logged in, tap the Gear icon to open Settings dialog and choose API Keys tab.

    (3) Add a new API key and write down your new SID and SID password.

    (4) If your server is behind a router or firewall, log into that device and map UDP 5060 and UDP 10000-20000 to the private LAN address of your server.

    NOTE: If your server is on the Debian, Ubuntu, or Raspbian platform, substitute the following command for the first two yum commands in step #5 below:

    apt-get -y install coreutils curl git jq
    

    (5) Log into your server and issue the following commands to install the EndPoint Updater:

    yum -y install coreutils curl git epel-release
    yum -y --enablerepo=epel install jq
    cd /usr/src
    git clone https://bitbucket.org/skyetel/ip-endpoint-group-update.git
    cd ip-endpoint-group-update
    ./ip-update-endpointgroup.sh
    

    (6) Fill in your credentials when prompted, and the cron script will be installed to keep your server’s dynamic IP address registered with Skyetel.

    Introducing Skyetel’s New Fax Platform

    Every time we read an article predicting the demise of fax technology, we have to chuckle. We’ve been reading the articles for about 30 years now, and fax still is the goto solution for many organizations. Can you spell HIPPA? Finally, Skyetel has dipped its toes in the fax waters by offering an easy-to-use fax solution for receipt of traditional and T.38 faxes. Simply purchase a Skyetel DID and configure it for vFax routing. Enter an email address for delivery of the faxes, and you’re done.


    Sending faxes from the Skyetel portal still is on the drawing boards, but it’s coming. In the meantime, Incredible Fax™ which is bundled with all Incredible PBX® platforms will let you send faxes ’til the cows come home with our easy-to-use Hylafax/AvantFax implementation.

    Implementing the New Spam Call Filter

    One of the most often requested features for any PBX is spam call filtering. Skyetel takes it to the next level by dealing with the spammers before the calls ever reach your PBX. For each of your Skyetel phone numbers, click on the Features tab and set the Spam Call Filter as desired.

    Recording and Transcribing Skyetel Calls

    As with spam call filtering, recording and/or transcribing Skyetel calls is only a click away. For each of your Skyetel phone numbers, click on the Features tab and set the option desired for Recording and/or Transcribing calls. Recordings and Transcriptions can be managed from your Skyetel Dashboard. Storage is free for up to 30 days, after which they are deleted.

    Skyetel Monitoring of Endpoint Health

    In addition to monitoring and reporting the health of all Skyetel services in your web portal, this latest addition allows you to configure Skyetel to not only monitor the State of every registered endpoint but also its Health with realtime metrics of the Latency, Packet Loss, and Jitter of each of your endpoints. Simply check the Network QOS options desired.

    Skyetel Expansion for Canadian Users


    Here’s some great news for our Canadian friends. Skyetel has been listening!

    • Porting to Skyetel in Canada now is significantly easier and faster
    • Awesome reductions in audio round trip times
    • Epic reductions in time-to-deliver
    • Faster response times to technical issues (and fewer of them!)
    • Audio for Canadian calls will now originate from Canadian data centers
    • SMS and MMS available on Canadian ported numbers

    Originally published: Thursday, November 1, 2018  Updated: Wednesday, June 12, 2019


    Support 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.



    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.

    BOGO 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.

    The 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.

    VitalPBX 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!
     

    Special 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.
     



    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. []

    A Sobering Look at Asterisk and the 2019 VoIP Landscape




    Every six months or so we like to gaze into our crystal ball for a quick look at the VoIP landscape. 2018 has been quite the transformative year with the acquisition of Digium® and Asterisk® by Sangoma®. Unfortunately, as we predicted, the Digium layoffs have already begun, and 2019 may only get worse. While we have no inside information, we wouldn’t be surprised to see Digium’s headquarters in Huntsville closed within six months in an effort to balance the books. Part of the problem may be attributable to the terms of the purchase itself. However, we sense there’s a more troubling development. And that is the reality that VoIP is becoming less and less appealing to home users and small businesses as more and more folks migrate purely to cell phones. Those with teenagers already know this transformation is underway. With services such as Google Fi starting at $20 for unlimited calling and texting, it’s difficult to justify VoIP services even at bargain basement prices. Making the cellular switch even more appealing are offers such as a $400 credit with the purchase of an LG G7 smartphone from Google or a free LG G7 with new Sprint service.

    What you lose with a pure cellular platform are many of the features that have made PBXs popular in the VoIP space: call routing, text-to-speech and voice recognition applications, conferencing, SPAM call blocking, and much more. But 2018 also was the year that Google finally pulled the plug on free calling through your PBX. Instead, you now have to purchase and configure a $50 OBi200 to continue with Google Voice, and the integration is painful to put it charitably. The demise of Google Voice added one more nail to the free VoIP coffin. And, as many of you know, Vitelity, our long-time platinum sponsor, now has bowed out of the VoIP retail business due to a change in focus from Voyant, the company’s new owner. Finally, our bargain-basement cloud provider for experimentation, HiFormance, appears to have bitten the dust. Details here. Suggestions here. Reminder: "You get what you pay for."

    It’s not all bad news for 2019. First, all of the Incredible PBX platforms are still alive and well. And they will remain open source GPL code. Second, we’ve found a terrific new VoIP provider, Skyetel, that will give you a $50 credit so you can kick the tires for a good long while. Effective 10/1/2023, $25/month minimum spend required. Third, if you’re looking for a robust Cloud platform, Digital Ocean still is offering a $100 signup credit for your first 60 days of service, and Incredible PBX runs swimmingly on their $5/month platform with CentOS. Spend another $1 a month, and you get automatic backups of your cloud-based server. It’s cheap insurance for something as important as your phone system.

    If you’re like us, you may be getting a little nervous about the future of Asterisk. We’ve already provided a series of articles on FusionPBX for FreeSWITCH. Our original tutorial and the follow-on articles showing how to create voice prompts using IBM Watson and how to create and deploy TTS applications such as news and weather reports are worth a careful read. And, if you consider yourself a pioneer, then you owe it to yourself to try out the FreeSWITCH developers’ new cloud-based platform, SignalWire. Here’s the $55 Promo code that worked for us: ITEXPO2019. That should get you off to a great start. And check out the pricing: U.S. DIDs are $0.08 per month, U.S. Origination rate (incoming) is $0.00325 per minute, U.S. Termination rate (outgoing) is $0.0072 per minute, U.S. SMS Outbound is $0.0009 per message, and U.S. SMS Inbound messages are free. MMS also available. Once verified, you can spoof any CallerID name and number that you own! What’s not to like? Asterisk Trunk setup example available here.

    CAUTIONARY NOTE: SignalWire should be considered EXPERIMENTAL SOFTWARE and is not yet suitable for production use.

    That should be enough excitement to keep all of you entertained over the holidays. We’re planning a few days off to be with family and friends. Let us be the first to wish each of you a very Merry Christmas. We’re looking forward to an exciting 2019!

    Originally published: Monday, December 17, 2018



    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.

    BOGO 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.

    The 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.

    VitalPBX 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!
     

    Special 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.
     



    Skyetel Smorgasborg: SMS Blasting, SMS Dictator, and more




    Just in time for Santa, we’ve got a great treat for those of you that have taken advantage of the Nerd Vittles special offer from Skyetel which gets you a $50 credit on their powerful VoIP platform. Today we’re adding not one, but three, SMS messaging utilities to the Incredible PBX UC platform. Effective 10/1/2023, $25/month minimum spend required. In addition to a command line utility to send SMS messages, we’re also introducing SMS Message Blasting which lets you send an SMS message to as many recipients as you would like. It’s perfect for sports team and community group messaging. To round out the trifecta, we’ve updated our SMS Dictator utility by integrating Skyetel messaging with IBM’s powerful voice recognition software.1 Simply dial S-M-S (767) from any extension on your PBX and dictate an SMS message to send to a recipient of your choice. Gone are the days of wrestling with Google’s ever-changing voice recognition platform. Good riddance!

    To get started, you’ll need to have an IBM Watson account with an APIkey for their Speech-to-Text (STT) engine. Next, you will need a Skyetel SMS-enabled DID. Before we install today’s SMS scripts, it should be noted that SMS messages must be sent from the PBX registered as the Skyetel Endpoint Group for the SMS-enabled DID specified in the Skyetel SMS scripts. So let’s begin with the configuration steps to put all the pieces in place.

    Getting Started with IBM Watson STT Service

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

    Now let’s get IBM’s Speech to Text service activated. Log back in to the IBM Cloud. Click on the (upper left) Menu icon and select Dashboard. 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 or LITE 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 APIkey. Then logout of IBM Watson.

    Getting Started with Skyetel Messaging

    If you haven’t already signed up for a Skyetel account, read our tutorial and take advantage of the $50 coupon for free service. Sign up for a DID and activate the SMS feature for your number. Create an Endpoint Group with the public IP address of your PBX. Then edit your phone number and link it to the Endpoint Group of your server. If you want to forward incoming SMS messages to either an email address or to your smartphone’s messaging service, configure it under the SMS & MMS tab. Finally, click on the settings icon beside your account name in the upper right corner of the Skyetel portal and then click the API Keys tab. Click the Create button and copy down your SID and SECRET for Skyetel’s API service. This secret is not retrievable once you close the window so put the credentials in a safe place for subsequent use. Then logout of the Skyetel portal.

    Installing the SMS Components on Your PBX

    There are three separate applications which we will install on your PBX: (1) a stand-alone utility that lets you send SMS messages from the Linux CLI by entering a recipients 11-digit phone number and an SMS message surrounded by quotes, (2) an SMS message blasting utility that lets you send a previously prepared SMS message to a group of recipients whose 11-digit SMS numbers have been entered into a text file, and (3) the SMS Dictator application which lets you pick up any phone on your PBX and dial S-M-S (767) to dictate a message and send it to a recipient whose number you’ve key in from your phone. For those not residing in North America, the number of phone number digits can easily be changed in all of the scripts. After we install the three applications, we’ll edit each of the scripts to insert your IBM STT and Skyetel API credentials. Then you’re ready to start messaging.

    First, let’s install the stand-alone and message blasting SMS utilities. Log into your server as root and issue the following commands:

    cd /root
    mkdir sms-skyetel
    cd sms-skyetel
    wget http://incrediblepbx.com/smsblast-skyetel.tgz
    tar zxvf smsblast-skyetel.tgz
    rm -f smsblast-skyetel.tgz
    

    Next, let’s install the SMS Dictator application while still logged into your server:

    cd /var/lib/asterisk/agi-bin
    wget http://incrediblepbx.com/sms-767-skyetel.tgz
    tar zxvf sms-767-skyetel.tgz
    rm -f sms-767-skyetel.tgz
    ./install-sms767-dialplan.sh
    

    Configuring the Skyetel SMS Components

    While still positioned in the agi-bin directory, edit smsgen.sh. Insert apikey as your API_USERNAME and your actual STT APIkey as API_PASSWORD in the fields provided. Insert your Skyetel SID, SECRET, and 11-digit DID in the fields provided. Then save the file.

    Next, change directories to /root/sms-skyetel and edit BOTH sms-skyetel and smsblast and insert your Skyetel credentials and DID in the fields provided at the top of both files.

    Finally, when you’re ready to use the message blasting application (smsblast), first insert your SMS message in the smsmsg.txt file. Then insert the list of SMS numbers in smslist.txt.

    Testing the Skyetel SMS Components

    To try out the SMS Dictator application, dial S-M-S (767) from a phone connected to your PBX. When prompted, enter the 11-digit number of the SMS recipient. When prompted, dictate the message to be sent and press #.

    To try out the stand-alone SMS application, navigate to /root/sms-skyetel and issue the following command using the 11-digit number of the SMS recipient followed by a space and an SMS message to be sent surrounded by quotes: ./sms-skyetel 18005551212 "Howdy."

    To try out the message blasting SMS application, navigate to /root/sms-skyetel. Enter the message to be sent in smsmsg.txt and enter the list of SMS numbers in smslist.txt. Kick off the message blast by entering the command: ./smsblast.

    Originally published: Monday, December 10, 2018


    Support 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.



    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.

    BOGO 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.

    The 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.

    VitalPBX 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!
     

    Special 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.
     



    1. Skyetel outbound SMS messages are billed at 1¢/message plus a monthly SMS surcharge of 10¢ per SMS-enabled DID. With IBM’s STT service, users have a choice of the LITE tier providing 100 minutes a month of free transcription or the STANDARD tier providing unlimited message transcription at a cost of 2¢/minute. []

    Spam Phone Call Blocker and CNAM Caching for FreePBX




    Blocking spam phone calls has been a challenge to put it charitably. Thanks to some earlier work by Stewart Nelson on the DSLR forum as well as Stewart’s considerable hand-holding in the development of today’s tutorial, we want to introduce a new approach to blocking these calls. The way it works is first time callers that pass the TrueCNAM SPAM check will be prompted to "press 5 to connect." Since most spam calls sit in a queue for several seconds before a live person chimes in, that person won’t hear the prompt. After 10 seconds or an invalid response, a SIT tone is played and the call is disconnected. If you’d prefer, you can send the failed calls to voicemail by uncommenting a single line in your dialplan. When a successful caller calls again, the caller will be connected without encountering the press 5 prompt.1 While today’s approach won’t block every robocaller, our testing suggests that, in combination with TrueCNAM, it will catch more than 95% of the spam callers. Using CallerID Superfecta with CNAM lookups from OpenCNAM coupled with AsteriDex and the Asterisk® Phonebook will provide an extremely low-cost solution both for blocking spammers AND for displaying accurate CNAM data for incoming calls since you’ll only pay for CNAM and TrueCNAM lookups from legitimate callers once.

    Here’s the actual dialplan addition that will monitor your incoming calls:

    [sub-log-caller]
    exten => s,1,NoOp(*** begin sub-log-caller ***)
    exten => s,n,GotoIf(${DB_EXISTS(cidname/${CALLERID(num)})}?CNAMCHECK)
    exten => s,n,GotoIf($[${DB_EXISTS(SPAMCHECK/deactivate)} = 0]?ACTIVATE)
    exten => s,n,GotoIf($[${DB(SPAMCHECK/deactivate)} = 1]?CONNECTNOW)
    exten => s,n(ACTIVATE),NoOp(Not yet WhiteListed)
    exten => s,n,AGI(truecnam.sh,${CALLERID(number)})
    exten => s,n,GotoIf($["${SPAM}"="SPAM"]?FLUNKED)
    exten => s,n,Playback(silence/1)
    exten => s,n,Playback(to-call-num-press)
    exten => s,n,Playback(digits/5)
    exten => s,n,Read(MYCODE,beep,1,n,1,10)
    exten => s,n,GotoIf($["${MYCODE}" = "5"]?ANONTEST)
    exten => s,n(FLUNKED),NoOp(*** Caller FLUNKED screening ***)
    ;exten => s,n,Dial(local/*701@from-internal) ; uncomment to send to 701 VM
    exten => s,n,Zapateller()
    exten => s,n,Hangup
    exten => s,n,Return()
    exten => s,n(ANONTEST),GotoIf($[${CALLERID(num)} > 0]?WHITELIST:CONNECTNOW) 
    exten => s,n(CNAMCHECK),Set(CNAM1=${CALLERID(name)})
    exten => s,n,Set(CNAM2=${DB(cidname/${CALLERID(number)})})
    exten => s,n,GotoIf($["${CNAM1}" = "${CNAM2}"]?WHITELISTED
    exten => s,n(WHITELIST),Set(DB(cidname/${CALLERID(number)})=${CALLERID(name)})
    exten => s,n,Set(CALLERID(all)="${CALLERID(name)} < ${CALLERID(number)}>")
    exten => s,n(WHITELISTED),NoOp(WhiteListed: ${CALLERID(all)})
    exten => s,n(CONNECTNOW),NoOp(*** end of sub-log-caller ***)
    exten => s,n,Return()
    

    We first introduced some of the CallerID caching concepts in our previous article last May. That article also documented the procedure for adding inbound call processing logic into FreePBX. If you already have implemented the steps outlined in that article, then the only modification required to deploy today’s new spam blocking technique is to replace the [sub-log-caller] context and reload the Asterisk dialplan. NOTE: Some deployments of CallerID Superfecta have an incorrect database password in the Default setup for AsteriDex. The original article will walk you through making the necessary change.

    If you’re starting from scratch, stop here for a bit and follow all of the steps in our previous article which now incorporates the spam blocking code as well. Here’s the link to get started. Return here once you’ve completed the initial setup.

    If you’re updating a previous deployment, here are the steps. Edit extensions_custom.conf in /etc/asterisk and remove the [sub-log-caller] context toward the end of the file. Then save the file. Next, issue the following commands to move the TrueCNAM script into place and insert the updated [sub-log-caller] context as well as the new [macro-dialout-trunk-predial-hook] context. Then reload your Asterisk dialplan. The dialplan additions will populate the Asterisk Phonebook and also whitelist calls from your PBX as well as incoming calls making it through the Spam Blocker.

    cd /tmp
    wget http://incrediblepbx.com/sub-log-caller.tar.gz
    tar zxvf sub-log-caller.tar.gz
    rm -f sub-log-caller.tar.gz
    mv truecnam.sh /var/lib/asterisk/agi-bin
    cd /etc/asterisk
    cat /tmp/sub-log-caller.txt >> extensions_custom.conf
    asterisk -rx "dialplan reload"
    

     

    Rotary Dial Phones & Blocked Numbers

    If someone you know and love still has a rotary dial phone, then you will need to manually add their number to either AsteriDex or your Asterisk Phonebook. Otherwise, the calls will never make it through the Spam Catcher. You can do this within the FreePBX GUI by accessing Admin -> Asterisk Phonebook. Click + Add Phonebook Entry and enter the 10-digit number for Grandma as well as her name. Add a second entry with Grandma’s 11-digit number in case some of your VoIP providers happen to send 11-digit CallerID numbers. We hasten to add you should normalize the formatting of your CallerID numbers as quickly as you can to avoid double entries. For those in the U.S. and Canada, we recommend the from-pstn-e164-us context for all of your trunks.

    If you have lots of friends with rotary dial phones or if you get calls from important, but unknown numbers such as medical offices where Caller ID numbers are blocked, then you probably should consider uncommenting the voicemail option in [sub-log-caller]. Then you at least will get voicemail notifications when one of these callers attempts to contact you. You still will have to manually add them to AsteriDex or the Asterisk Phonebook so they can contact you directly in the future. HINT: Most medical office calls now spoof the main number of the office so you only need to add the office number just as you did with grandma.

    Toggling Spam Blocker On and Off

    We’ve also included the ability to turn off the Spam Blocker should you ever wish to do so. To disable the Spam Blocker, issue the following command at the Asterisk CLI:

    database put SPAMCHECK deactivate 1

    To once again enable the Spam Blocker, issue the following command at the Asterisk CLI:

    database deltree SPAMCHECK

    WhiteListing Previous Callers

    We appreciate that you may not want to aggravate callers that have been calling you for years by making them jump through hoops the next time they call. So here’s a quick way to populate your Asterisk Phonebook with the names and numbers of previous callers. For entries where the CNAM is merely the CallerID Number, future calls from these numbers still will be looked up with OpenCNAM to obtain an actual CNAM match. We’ve made a couple of assumptions that you are more than welcome to adjust to meet your own needs. First, we’ve limited the list to callers from the past two calendar years. Second, we’ve only captured calls that lasted more than 15 seconds. We’ll drop down to the Linux CLI to build the list of callers to import. Then we’ll use the FreePBX GUI to import the list into the Asterisk Phonebook. While we’re building the import list, you’ll have two opportunities to prune the list using your favorite text editor. To get started, issue the following commands from the Linux CLI:

    mysql -u root -ppassw0rd asteriskcdrdb -Ns -e "select distinct src, clid \\
    from cdr where calldate > '2017/01/01' and duration > 15 \\
    order by clid asc" > 2YR-full
    

    Now edit the 2YR-full file and remove any complete lines you don’t want to import.

    Next, we’ll reformat the CallerID Numbers and Names into a format needed for the import:

    cat 2YR-full | cut -f 1 -d '"' | sed 's|[[:space:]]||' > 2YR-numbers
    cat 2YR-full | cut -f 2 -d '"' > 2YR-names
    paste 2YR-numbers 2YR-names | awk '{print $1,$2,$3,$4}' > 2YR-all
    awk '{print $2 " " $3 $4 ";" $1";"}' 2YR-all > 2YR-freepbx.csv
    

    Now we should have our 2YR=freepbx.csv file in its final form for import. Open the file in your favorite editor. The syntax of the entries should be CallerID Name, then a semicolon, then CallerID Number, and then a semicolon. Discard any additional lines you wish to exclude from the import. Once you have all the entries squared away, copy the file to your desktop PC and open FreePBX in your browser. Navigate to Admin -> Asterisk Phonebook. Click Import Phonebook and then Browse. Select the 2YR-freepbx.csv file from your desktop. Then click Upload. Take a final look at the new entries in your Asterisk Phonebook to make sure nothing came unglued, and you’re all set.

    TrueCNAM: The Icing on the Spam Cake



    A couple years ago we introduced TrueCNAM, a service that provides not only CNAM data but also Caller Reputation scoring. Those that flunk using the revolving caller reputation matrix get disconnected automatically. We strongly encourage you to add the TrueCNAM service to your PBX. The service includes a free tier as well as incredibly reasonable commercial tiers. For background on the service, here’s a link to our previous TrueCNAM tutorial. For today, start by signing up for a TrueCNAM account and obtain an APIkey and APIpassword. Then register at least one of your DIDs with the service. Once you have your credentials and your DID number in hand, edit truecnam.sh in /var/lib/asterisk/agi-bin. Insert these three items at the top of the file and save it to activate TrueCNAM. It doesn’t get much easier than that.

    Now make a few test calls to your PBX to assure that everything is working as documented. Enjoy!

    Originally published: Monday, November 26, 2018


    Support 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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    1. Once installed, you can change the voice prompt to a number other than 5 by modifying lines 10 and 12 of the context sub-log-caller which you will find in extensions_custom.conf in the /etc/asterisk directory at the completion of this install. []