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The Most Versatile VoIP Provider: FREE PORTING
JUST RELEASED: Visit the Incredible PBX WikiDesktop Dream Machine: It’s Incredible PBX for VirtualBox
If you’re new to the VoIP world or just getting started with Google’s latest Google Voice creation, then we have a one minute setup solution today that doesn’t require you to buy anything ever. You can use almost any desktop computer you already own to bring up the VirtualBox® edition of Incredible PBX® in less than 60 seconds. Take another minute or two to install a Google Voice trunk, and you’ll have free calling in the U.S. and Canada until the cows come home. If you’ve followed Nerd Vittles over the years, you already know that VirtualBox from Oracle® is one of our favorite platforms. Once VirtualBox is installed on your desktop computer, adding Incredible PBX is a snap. Download our new GVSIP vbox image from SourceForge, double-click on the downloaded image, check the initialize MAC address box, and boom. In less than a minute, your PBX is ready to use. The really nice thing about playing along today is it won’t cost you a dime to try things out for yourself. And, if you really love it and we think you will, there’s no hidden fee or crippleware to hinder your continued use of Incredible PBX for as long as you like. Of course, the Incredible PBX feature set is included as well which brings you nearly three dozen applications for Asterisk® that will revolutionize your communications platform. Just add your credentials and speech-to-text, voice recognition, and a Siri-like telephony interface are as close as your nearest SIP phone.
Installing Oracle VM VirtualBox
Oracle’s virtual machine platform inherited from Sun is amazing. It’s not only free, but it’s pure GPL2 code. VirtualBox gives you a virtual machine platform that runs on top of any desktop operating system. In terms of limitations, we haven’t found any. We even tested this on an Atom-based Windows 7 machine with 2GB of RAM, and it worked without a hiccup. So step #1 today is to download one or more of the VirtualBox installers from VirtualBox.org or Oracle.com. Our recommendation is to put all of the 100MB installers on a 4GB thumb drive.1 Then you’ll have everything in one place whenever and wherever you happen to need it. Once you’ve downloaded the software, simply install it onto your favorite desktop machine. Accept all of the default settings, and you’ll be good to go. For more details, here’s a link to the Oracle VM VirtualBox User Manual.
Installing Incredible PBX 13-13 with GVSIP
To begin, download the Incredible PBX vbox image (2.3 GB) onto your desktop.
Next, double-click on the Incredible PBX .ova image on your desktop. Be sure to check the box to initialize the MAC address of the image and then click Import. Once the import is finished, you’ll see a new Incredible PBX virtual machine in the VM List of the VirtualBox Manager Window. Let’s make a couple of one-time adjustments to the Incredible PBX configuration to account for possible differences in sound and network cards on different host machines.
(1) Click once on the Incredible PBX virtual machine in the VM List. Then (2) click the Settings button. In the Audio tab, check the Enable Audio option and choose your sound card. In the Network tab for Adapter 1, check the Enable Network Adapter option. From the Attached to pull-down menu, choose Bridged Adapter. Then select your network card from the Name list. Then click OK. That’s all the configuration that is necessary for Incredible PBX.
Running Incredible PBX in VirtualBox
Once you’ve imported and configured the Incredible PBX Virtual Machine, you’re ready to go. Highlight the Incredible PBX virtual machine in the VM List on the VirtualBox Manager Window and click the Start button. The standard Linux boot procedure will begin and, within a few seconds, you’ll get the familiar Linux login prompt. During the bootstrap procedure, you’ll see a couple of dialogue boxes pop up that explain the keystrokes to move back and forth between your host operating system desktop and your virtual machine. Remember, you still have full access to your desktop computer. Incredible PBX is merely running as a task in a VM window. Always gracefully halt Incredible PBX just as you would on any computer.
Here’s what you need to know. To work in the Incredible PBX virtual machine, just left-click your mouse while it is positioned inside the VM window. To return to your host operating system desktop, press the right Option key on Windows machines or the left Command key on any Mac. For other operating systems, read the dialogue boxes for instructions on moving around. To access the Linux CLI, login as root with the default password: password. Change your passwords immediately by typing: /root/update-passwords.
Setting the Date and Time with VirtualBox
On some platforms, VirtualBox has a nasty habit of mangling the date and time of your virtual machine. Typing date will tell you whether your VM is affected. If it’s a problem, manually set the date and time and then update the hardware clock. Here’s how assuming 08130709 is the month, day, and correct time of your server:
date 08130709 clock -w
Obtaining Google Voice Credentials for GVSIP
You’ll need at least one dedicated Google Voice account to use the new Incredible PBX implementation of GVSIP with Asterisk. If you’re new to all of this, our Getting Started with Google Voice tutorial will walk you through setting up an account and obtaining your OAuth 2 refresh token for GVSIP.
Creating GVSIP Trunks with Incredible PBX
With your Google Voice refresh token and 10-digit phone number in hand, you’re ready to add the GVSIP trunk to your PBX. While logged into the Linux CLI as root, issue the following command: gvsip-naf/install-gvsip
You’ll be prompted whether you wish to create a GVSIP trunk, Type y and enter your refresh_token and 10-digit phone number for your existing Google Voice trunk. For each trunk, the installer will create the necessary code to support a PJSIP trunk and a GVSIPn Custom Trunk to use for outbound routing. To add additional trunks, simply run the installer again. Adding a new trunk takes about 10 seconds.
Should you ever want to refresh the patched version of Asterisk, copy pjsip_custom.conf from /etc/asterisk to a safe place, delete the contents of pjsip_custom.conf, rerun the installer, and then copy your version of pjsip_custom.conf back to /etc/asterisk and restart Asterisk: amportal restart. That way you won’t lose your previously configured trunks.
If you ever need to delete a GVSIP trunk that you previously have added, we’ve included a script which will perform the task for you. Just run del-trunk and specify the trunk to delete.
Once you have multiple GVSIP trunks on your PBX, we’ve added a utility to make it easy to decipher which trunk number is associated with each Google Voice trunk. Simply run: gvsip-naf/show-trunks
Configuring an Inbound Route for GVSIP Trunks
By default, incoming calls to GVSIP trunks on Incredible PBX servers will be sent to the Default Inbound Route configured on your PBX. As initially installed, that Default route points to Allison’s Demo IVR. This can be changed easily in the FreePBX GUI by modifying the Destination for the Default inbound route in Connectivity:Inbound Routes.
On other server platforms, you may not have a Default inbound route configured so you will need to create an inbound route to handle calls from each GVSIP trunk. Regardless of your server platform, we strongly recommend adding an Inbound Route for every GVSIP trunk using the 10-digit GVSIP phone number as the DID for the inbound route. Here’s an example of an Inbound Route created in Connectivity:Inbound Routes:Add Inbound Route:
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 an Outbound Route for GVSIP Trunks
By default, you cannot place outbound calls using your new GVSIP trunks. For each trunk, you first will need to create an Outbound Route specifying a Dial Pattern to use with each GVSIP trunk in Connectivity:Outbound Routes:Add Outbound Route. If you only have a single Google Voice trunk on your PBX and no other trunks, then you would probably want to specify that outbound calls be routed out the GVSIP1 trunk with a Dial Pattern of NXXNXXXXXX with 1 as the Prepend. This tells FreePBX to dial 18005551212 using the GVSIP1 trunk when a PBX user dials 8005551212. Google only accepts calls that include a country code (1=US/CAN).
HINT: If you ever forget which GVSIP trunks are associated with which phone numbers, simply run /root/gvsip-naf/del-trunk for a list of your trunks. Just press ENTER to exit without deleting any of your trunks.
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. As part of the install, the dialing prefixes of *41 through *49 were reserved for GVSIP trunks if you would like to use them. That’s totally up to you. Here’s what an Outbound Route would look like using this scheme for the GVSIP2 trunk:
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.
Configuring Incredible PBX for Incredible PBX
In order to take advantage of all the Incredible PBX applications, you’ll need to obtain IBM text-to-speech (TTS) and speech-to-text (STT) credentials as well as a (free) Application ID for Wolfram Alpha.
NOV. 1 UPDATE: IBM has moved the goal posts effective December 1, 2018:
This Nerd Vittles tutorial will walk you through getting your IBM account set up and obtaining both your TTS and STT credentials. Be sure to write down BOTH sets of credentials which you’ll need in a minute. For home and SOHO use, IBM access and services are FREE even though you must provide a credit card when signing up. The IBM signup process explains their pricing plans.
To use Wolfram Alpha, sign up for a free Wolfram Alpha API account. Just provide your email address and set up a password. It takes less than a minute. Log into your account and click on Get An App ID. Make up a name for your application and write down (and keep secret) your APP-ID code. That’s all there is to getting set up with Wolfram Alpha. If you want to explore costs for commercial use, there are links to let you get more information.
In addition to your Wolfram Alpha APPID, there are two sets of IBM credentials to plug into the Asterisk AGI scripts. Keep in mind that there are different usernames and passwords for the IBM Watson TTS and STT services. The TTS credentials will look like the following: $IBM_username and $IBM_password. The STT credentials look like this: $API_USERNAME and $API_PASSWORD. Don’t mix them up. 🙂
All of the scripts requiring credentials are located in /var/lib/asterisk/agi-bin so switch to that directory after logging into your server as root. Edit each of the following files and insert your TTS credentials in the variables already provided: nv-today2.php, ibmtts.php, and ibmtts2.php. Edit each of the following files and insert your STT credentials in the variables already provided: getquery.sh, getnumber.sh, and getnumber2.sh. Finally, edit 4747 and insert your Wolfram Alpha APPID.
Using Asteridex with Incredible PBX
AsteriDex is a web-based dialer and address book application for Asterisk and Incredible PBX. It lets you store and manage phone numbers of all your friends and business associates in an easy-to-use SQLite3 database. You simply call up the application with your favorite web browser: http://pbx-ip-address/asteridex4/. When you click on a contact that you wish to call, AsteriDex first calls you at extension 701, and then AsteriDex connects you to your contact through another outbound call made using your default outbound trunk that supports numbers in the 1NXXNXXXXXX format.
Taking Incredible PBX for a Test Drive
You can take Incredible PBX on a test drive by dialing D-E-M-O (3366) from any phone connected to your PBX.
With Allison’s Demo IVR, you can choose from the following options:
- 0. Chat with Operator — connects to extension 701
- 1. AsteriDex Voice Dialer – say "Delta Airlines" or "American Airlines" to connect
- 2. Conferencing – log in using 1234 as the conference PIN
- 3. Wolfram Alpha Almanac – say "What planes are flying overhead"
- 4. Lenny – The Telemarketer’s Worst Nightmare
- 5. Today’s News Headlines — courtesy of Yahoo! News
- 6. Weather by ZIP Code – enter any 5-digit ZIP code for today’s weather
- 7. Today in History — courtesy of OnThisDay.com
- 8. Chat with Nerd Uno — courtesy of SIP URI connection to 3CX iPhone Client
- 9. DISA Voice Dialer — say any 10-digit number to be connected
- *. Current Date and Time — courtesy of Incredible PBX
Originally published: Tuesday, August 14, 2018
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Special Thanks to Our Generous Sponsors
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One Minute Cloud VPS: Meet Incredible PBX for HiFormance
Dec. 17 Update: Before signing up for HiFormance service, read the latest update on this provider here.
Part of the challenge of deploying an Asterisk®-based server on a cloud platform is getting all of 1,000+ pieces in place without a hiccup during the installation. Particularly for first-time users of a VoIP platform, this can be problematic. HiFormance is one of our favorite low-cost cloud providers, and today they’re introducing something that no other cloud provider offers: a configured Incredible PBX 13-13 VPS that’s ready to accept calls in under a minute. We’ve been inching toward a new low-cost plateau for VoIP cloud providers, and today we have a new milestone that finally makes running VoIP servers out of your home or office look like the horse-and-buggy days.
$10 to $13 a year now buys you a cloud platform that is less expensive than the cost of electricity to run a server on premise. You get 2GB of RAM, 20GB of SSD storage, two virtual core processors, and 2TB of monthly bandwidth. If you prepay for 3 years, you can double either the RAM or SSD storage by simply opening a ticket after you sign up. The $10/year offering is in limited supply and gives you your choice of the following sites: L.A., Buffalo, or Dallas. The $13/year offering adds the following additional sites: Chicago, Atlanta, and New York. It’s a near perfect platform for Incredible PBX 13-13 with CentOS 6.9. Add a Google Voice trunk and you get unlimited calling in the U.S. and Canada combined with a feature set that you’ll be hard-pressed to find on any PBX at any price. Putting all the pieces in place is about as simple as preparing slice-and-bake cookies, and you’ll be up and running before the cookies come out of the oven. Skip that hamburger lunch and come join the party!
So what’s the catch? Well, there’s no catch with Incredible PBX 13-13 and CentOS 6.9. This HiFormance platform uses OpenVZ with SolusVM. Someone always asks, "If the platform is so great, why aren’t you using it?" And our answer is we are. We have deployed HiFormance cloud-based VoIP servers running Incredible PBX 13-13 in Atlanta, Buffalo, Chicago, and Los Angeles without any hiccups in service. Performance is excellent. Support is excellent. So run, don’t walk, to sign up for one of these before they’re all gone. You won’t be disappointed. Just fill out the entries as shown above once you log into the HiFormance site. Nerd Vittles receives no commissions from signups.
June 20 Update: For those of you on the West Coast and readers in the Far East, we are pleased to announce that HiFormance has extended the Incredible PBX offer to their new $1/month VZ-2019 platform featuring 4GB RAM, 20GB SSD storage, 4TB of bandwidth, and DDOS protection at the QuadraNet Los Angeles data center. As with the other offers, you can double the bandwidth or storage by prepaying for three years in advance and opening a ticket. For those in the Far East, you can upgrade to the Asia Optimized CN2, Los Angeles offering for an additional $10.99 a year.
Getting Started with Incredible PBX 13-13
Once you’ve signed up for an account, choose a location for your server and select Incredible PBX 13-13 as your "operating system." It’s a preconfigured implementation that uses the latest CentOS 6.9 platform. Once you receive your credentials by email, login to your SolusVM Control Panel with a browser. Click the Manage tab for your virtual machine. In the Settings tab at the bottom of the control panel, turn TUN/TAP ON if you plan to use the NeoRouter VPN. Next, click the Root/Admin Password tab and set a default root password for your virtual machine. Take a look around in the control panel while you’re there. This is where you’d return if you need to restart your virtual machine or if you wish to reinstall your virtual machine with the same or a different operating system.
Now that your virtual machine is up and running, log into your server with SSH as root and type YES to get started. Be sure to log in from a desktop machine that you plan to use to manage your server since this IP address will be whitelisted. You’ll be prompted to immediately change all of your passwords. When the setup finishes, just type "reboot" to reboot your server. Log back in as root and the Automatic Update Utility will bring your server up to current specs. If you’re not in the Eastern U.S. time zone, type: /root/timezone-setup. Then you’re ready to go.
One of the unique features of Incredible PBX 13-13 is that most of the major components of the aggregation including Asterisk were compiled from source, and the source code is actually on your server. The advantage is you can quickly make any future modifications desired to meet your own unique requirements. You won’t find this in any other VoIP implementation. For example, here’s all it takes to upgrade to the very latest release of Asterisk. As with the current version, both support SRTP encrypted communications!
cd /usr/src rm -rf asterisk-13.21.0 wget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-13.21.1.tar.gz tar zxvf asterisk-13.21.1.tar.gz mv asterisk-13.21.1.tar.gz /tmp cd asterisk-13.21.1 contrib/scripts/get_mp3_source.sh mv res/res_xmpp.c res/res_xmpp.c.orig wget http://incrediblepbx.com/res_xmpp-13.tar.gz tar zxvf res_xmpp-13.tar.gz cp -p res/res_xmpp.c res/res_xmpp.c.oauth make distclean wget http://incrediblepbx.com/menuselect-incredible13.8.tar.gz tar zxvf menuselect-incredible* rm -f menuselect-incredible* wget https://issues.asterisk.org/jira/secure/attachment/54233/srtp_aead_and_big_aes.patch wget http://incrediblepbx.com/srtp_aead_and_big_aes.patch.patch patch -p0 < ./srtp_aead_and_big_aes.patch.patch patch -p0 <./srtp_aead_and_big_aes.patch CFLAGS='-DENABLE_SRTP_AES_256 -DENABLE_SRTP_AES_GCM' ./configure --libdir=/usr/lib64 --with-pjproject-bundled make menuselect.makeopts menuselect/menuselect --enable-category MENUSELECT_ADDONS menuselect.makeopts menuselect/menuselect --enable-category MENUSELECT_CODECS menuselect.makeopts menuselect/menuselect --disable-category MENUSELECT_TESTS menuselect.makeopts menuselect/menuselect --enable CORE-SOUNDS-EN-GSM --enable MOH-OPSOUND-WAV --enable EXTRA-SOUNDS-EN-GSM --enable cdr_mysql menuselect.makeopts menuselect/menuselect --disable app_mysql --disable app_setcallerid --disable func_audiohookinherit --disable res_fax_spandsp menuselect.makeopts menuselect/menuselect --disable test_named_lock --disable test_res_pjsip_scheduler --disable test_file --disable test_bridging --disable test_res_pjsip_scheduler menuselect.makeopts sed -i 's|march=native|mtune=native|' Makefile.rules expect -c 'set timeout 60;spawn make menuselect;expect "Save";send "\t\t\r";interact' make make install ldconfig cd /usr/src/Asterisk-Flite-2.2-rc1-flite1.3 make clean make make install amportal restart
WebMin is also installed and configured as part of the base install. The root password for access is the same as your Linux root password. We strongly recommend that you not use WebMin to make configuration changes to your server. You may inadvertently damage the operation of your PBX beyond repair. WebMin is an excellent tool to LOOK at how your server is configured. When used for that purpose, we highly recommend WebMin as a way to become familiar with your Linux configuration.
Using the Incredible PBX 13-13 Web GUI
Most of the configuration of your PBX will be performed using the web-based Incredible PBX GUI with its FreePBX 13 GPL modules. Use a browser pointed to the IP address of your server and choose Incredible PBX Admin. Log in as admin with the password you configured in the previous step. HINT: You can always change it if you happen to forget it: /root/admin-pw-change. You can safely ignore the warning about a missing swap file. You have plenty of RAM, and OpenVZ platforms don't permit swap files. If you're worried about it, choose the 3-year prepayment option and double your ram from 2GB to 4GB which is more than ample for even the busiest PBXs.
To get a basic system set up so that you can make and receive calls, you'll need to configure a VoIP trunk, create one or more extensions, set up an inbound route to send incoming calls to an extension, and set up an outbound route to send calls placed from your extension to a VoIP trunk that connects to telephones in the real world. You'll also need a SIP phone or softphone to use as an extension on your PBX.
Continue Reading: Configuring Extensions, Trunks & Routes
Introducing Incredible PBX Whole Enchilada
The Whole Enchilada upgrade already is included on the HiFormance platform. You can review the tutorial for the three dozen apps here.
Reconfiguring NeoRouter VPN for OpenVZ
On OpenVZ platforms including HiFormance, you'll need to enable TUN/TAP in the Control Panel for your VPS. After adjusting the setting, reboot your server. Then the NeoRouter VPN client will function properly.
Installing Incredible Fax with HylaFax/AvantFax
To install Incredible Fax, log into your server as root and issue the following commands. Install time: 2 minutes.
cd /root ./incrediblefax13.sh
After entering your email address to receive incoming faxes, you'll be prompted about two dozen times to choose options as part of the install. Simple press the ENTER key at each prompt and accept all of the defaults. When the install finishes, make certain that you reboot your server to bring Incredible Fax on line. There will be a new AvantFax option in the Incredible PBX GUI. The default credentials for AvantFax GUI are admin:password; however, you first may be prompted for your Apache admin credentials which were set when you installed your server. Then you'll be asked to change your AvantFax password.
Be advised that HiFormance blocks outbound email ports including port 587 by default. You'll need to open a ticket with them to use email from your server. You can test whether port 25 and 587 are blocked with the following command:
telnet gmail-smtp-msa.l.google.com 587
Backing Up Your Server
One of the reasons we love the CentOS 6.9 platform is the ease with which you can make full system backups, particularly on cloud platforms. There's no MySQL InnoDB databases to worry about so you can treat MySQL databases just like any other Linux file. We're offering the following script for your use pursuant to the GPL2 license. By using the script at no cost, you agree to assume all risks and absolve us from any liability regarding bugs, performance, or any other failure in the code. If that's acceptable to you, copy the commands below and create a backup-full script in the /root folder of your primary server. After saving the script, make it executable: chmod +x backup-full.
#!/bin/bash # backup-full for Incredible PBX, Copyright (c) 2008-2018, Ward Mundy & Associates, LLC # Licensed pursuant to GPL2. See /root/COPYING on any Incredible PBX server for details amportal stop service mysqld stop service httpd stop service sendmail stop cd / tar -cf /tmp/backup.tar /bin /etc /home /lib /lib64 /media /mnt /opt /root /sbin /usr /var service sendmail start service httpd start service mysqld start amportal start tar --delete -f /tmp/backup.tar etc/udev/rules.d tar --delete -f /tmp/backup.tar etc/sysconfig/network-scripts tar --delete -f /tmp/backup.tar var/lib/dhclient tar --delete -f /tmp/backup.tar etc/fstab tar --delete -f /tmp/backup.tar etc/resolv.conf tar --delete -f /tmp/backup.tar etc/hosts tar --delete -f /tmp/backup.tar etc/hostname gzip /tmp/backup.tar echo "Your backup is available: /tmp/backup.tar.gz" echo "Copy it and test it in a safe place OFF SITE" echo " "
To run the script, execute the following command: /root/backup-full
We recommend running the backup-full script during hours when your PBX is not in active use since Asterisk and other services typically are shut down for 2-10 minutes. Depending upon the size and performance of your server, the complete backup process typically takes 5-20 minutes. Once the backup script finishes, copy /tmp/backup.tar.gz to a safe place away from the primary server every week. You can automate the backup and the copying procedure with a cron job if desired.
To restore, copy your backup.tar.gz file back to the /tmp directory on the new server and then:
#!/bin/bash # restore-full for Incredible PBX, Copyright (c) 2008-2018, Ward Mundy & Associates, LLC # Licensed pursuant to GPL2. See /root/COPYING on any Incredible PBX server for details amportal stop service mysqld stop service httpd stop service sendmail stop cd / tar zxvf /tmp/backup.tar.gz reboot
Upgrading to IBM Speech Engines
NOV. 1 UPDATE: IBM has moved the goal posts effective December 1, 2018:
If you've endured Google's Death by a Thousand Cuts with text-to-speech (TTS) and voice recognition (STT) over the years, then we don't have to tell you what a welcome addition IBM's new speech utilities are. We can't say enough good things about the new IBM Watson TTS and STT offerings. While IBM's services are not free, that’s really theoretical for most of our readers. Your first month on the platform is entirely free. And, after that, you get 1,000 minutes a month of free STT voice recognition services. And the first million characters of text-to-speech synthesis are FREE every month as well. So let's put the pieces in place so you'll be ready to play with the Whole Enchilada. Here's our tutorial that will walk you through the one-time IBM setup.
Next, login to your Incredible PBX server and issue these commands to update your Asterisk dialplan and edit ibmtts.php:
cd /var/lib/asterisk/agi-bin ./install-ibmtts-dialplan.sh nano -w ibmtts.php
Insert your credentials in $IBM_username and $IBM_password. Verify that $IBM_url matches the entry provided when you registered with IBM. Then save the file: Ctrl-X, Y, then ENTER. Now reload the Asterisk dialplan: asterisk -rx "dialplan reload". Try things out by dialing 951 (news) or 947 (Weather) from an extension registered on your PBX.
To get IBM's Speech to Text service configured, while still logged in to your Incredible PBX server, issue these commands to edit getnumber.sh:
cd /var/lib/asterisk/agi-bin nano -w getnumber.sh
Insert your API_USERNAME and API_PASSWORD in the fields provided. Then save the file: Ctrl-X, Y, then ENTER. Update your Voice Dialer (411) to use the new IBM STT service:
sed -i '\\:// BEGIN Call by Name:,\\:// END Call by Name:d' /etc/asterisk/extensions_custom.conf sed -i '/\\[from-internal-custom\]/r ibm-411.txt' /etc/asterisk/extensions_custom.conf asterisk -rx "dialplan reload"
Now try out the Incredible PBX Voice Dialer with AsteriDex by dialing 411 and saying "Delta Airlines."
Configuring Google Voice with Incredible PBX
The advantage of Google Voice trunks for those of you in the United States is that all of your calls within the U.S. and Canada are free. You can't beat the price, and it has worked reliably for many, many years but it may disappear any day now. If so, we'll have a fresh tutorial with a workaround for you soon. Unless Google pulls the plug on XMPP support for Google Voice, there are three different ways to set up Google Voice trunks with Incredible PBX. For a one-time fee of $4.99 with this coupon, you can use the Simonics GV/SIP gateway to configure a Google Voice account using OAuth 2 authentication. Then just set up the Simonics SIP trunk on your PBX to point to the Simonics gateway. A second option is to choose the (recommended) OAuth 2 authentication method for Google Voice when you initially install Incredible PBX 13-13. Finally, you can choose plain-text passwords for Google Voice when you set up Incredible PBX. The drawback of this last option is Google has hinted that they may discontinue support of plain-text passwords.
Here are the initial setup steps on the Google side:
1. Set up a dedicated Gmail and Google Voice account to use exclusively for this Google Voice setup on your PBX. Head over to the Google Voice site and register. You'll need to provide a U.S. phone number to verify your account by either text message or phone call.
2. Once you have verified your account by entering your verification code, you'll get a welcome message from Mr. Google. Click Continue to Google Voice.
3. Provide an existing U.S. phone number for verification. It can be the same one you used to set up your Google account in step #1.
4. Once your phone number has been verified, choose a DID in the area code of your choice.
Special Note: Google continues to tighten up on obtaining more than one Google Voice number from the same computer or the same IP address. If this is a problem for you, here's a workaround. From your smartphone, install the Google Voice app from iPhone App Store or Google's Play Store. Then open the app and login to your new Google account. Choose your new Google Voice number when prompted and provide a cell number with SMS as your callback number for verification. Once the number is verified, log out of Google Voice. Do NOT make any calls. Now head back to your PC's browser and login to https://voice.google.com. You will be presented with the new Google Voice interface which does not include the Google Chat option. But fear not. At least for now there's still a way to get there. After you have set up your new phone number and opened the Google Voice interface, click on the 3 vertical dots in the left sidebar (it's labeled More). When it opens, click Legacy Google Voice in the sidebar. That will return you to the old UI. Now click on the Gear icon (upper right) and choose Settings. Make sure the Google Chat option is selected and disable forwarding calls to whatever default phone number you set up.
5. When your DID has been assigned, click the More icon at the bottom of the left column of the Google Voice desktop. Click Legacy Google Voice. Now click the Settings icon on your legacy Google Voice desktop. Make certain that Forward Calls to Google chat is checked and disable calls to your forwarding number. Click on the Calls tab and select Call Screening:OFF, CallerID (Incoming):Display Caller's Number, and Global Spam Filtering:checked. The remaining entries should be blank.
6. Google Voice configuration is now complete. Sign out of your Google Voice account.
The Simonics GV-SIP Gateway Solution. Here's the quick thumbnail of the steps to put all the pieces in place. First, we set up a Google Voice account at Google as documented above. Next, we'll set up an account at the Simonics site to link our Google Voice account to the Simonics SIP Gateway. Then we'll plug our Simonics SIP credentials into the preconfigured Simonics trunk on Incredible PBX. Finally, we'll add Incoming and Outgoing Routes to tell Incredible PBX how to process Google Voice calls.
Now you're ready to set up an account on the Simonics site. With this Nerd Vittles link, there's a one-time fee of $4.99.
1. Start by registering your new Google account.
2. After paying the $4.99 registration fee via PayPal, proceed through the setup process to link your Google Voice account and 11-digit Google Voice phone number to the Simonics SIP Gateway.
3. You then will be provided your SIP username and password as well as the gateway address, gvgw.simonics.com, to use in building your SIP trunk on your PBX.
4. If your SIP credentials ever get compromised, regenerate your password by logging back into the Simonics GW site.
Now it's time to configure your Simonics trunk in Incredible PBX. Start by logging into the web interface as admin with your admin password from above. Click Connectivity:Trunks and choose the Simonics trunk in the PBX Configuration menu. The Simonics trunk template will display:
1. Untick the Disable Trunk check box.
2. In Outbound CallerID, insert your 10-digit Google Voice number.
3. In username, insert GV1 followed by your 10-digit Google Voice number.
4. In secret, insert your Simonics SIP password.
5. In the Registration String, insert GV1 followed by your 10-digit Google Voice number followed by a colon (:)
6. In the Registration String after the colon, insert your Simonics SIP password.
7. In the tail of the Registration String after the slash (/), insert your 10-digit Google Voice number.
8. Click Submit Changes and then Reload the Dialplan when prompted.
Configuring GV Trunk with Motif in the GUI. If you elect to configure your Google Voice trunk natively using the Incredible PBX GUI, you first will need to obtain a Refresh_Token if you elected to use OAuth 2 authentication.
1. Be sure you are still logged into your Google Voice account. If not, log back in at https://voice.google.com.
2. In a separate browser tab, go to the Google OAUTH Playground using your browser while still logged into your Google Voice account.
3. Once logged in to Google OAUTH Playground, click on the Gear icon in upper right corner (as shown below).
3a. Check the box: Use your own OAuth credentials
3b. Enter Incredible PBX OAuth Client ID:
466295438629-prpknsovs0b8gjfcrs0sn04s9hgn8j3d.apps.googleusercontent.com
3c. Enter Incredible PBX OAuth Client secret: 4ewzJaCx275clcT4i4Hfxqo2
3d. Click Close
4. Click Step 1: Select and Authorize APIs (as shown below)
4a. In OAUTH Scope field, enter: https://www.googleapis.com/auth/googletalk
4b. Click Authorize APIs (blue) button.
5. Click Step 2: Exchange authorization code for tokens
5a. Click Exchange authorization code for tokens (blue) button
5b. When the tokens have been generated, Step 2 will close.
6. Reopen Step 2 and copy your Refresh_Token. This is the "password" you will need to enter (together with your Gmail account name and 10-digit GV phone number) when you add your GV trunk in the Incredible PBX GUI. Store this refresh_token in a safe place. Google doesn't permanently store it!
7. Authorization tokens NEVER expire! If you ever need to remove your authorization tokens, go here and delete Incredible PBX Google Voice OAUTH entry by clicking on it and choosing DELETE option.
Switch back to your Gmail account and click on the Phone icon at the bottom of the window to place one test call. Once you successfully place a call, you can log out of Google Voice and Gmail.
Yes, this is a convoluted process. Setting up a secure computing environment often is. Just follow the steps and don't skip any. It's easy once you get the hang of it. And you'll sleep better.
Now you’re ready to configure your Google Voice account in Incredible PBX. You do it from within the Incredible PBX GUI by choosing Connectivity:Google Voice. Just plug in your Google Voice Username, enter your refresh_token from Step #6 above as your Google Voice Password, enter your 10-digit Google Voice Phone Number, and check the first two boxes: Add Trunk and Add Outbound Routes. Then click Submit and Apply Settings to save your new entries.
If you elected to use plain-text passwords for Google Voice, simply skip obtaining OAuth 2 credentials and substitute your plain-text password for the refresh_token when you create the Google Voice trunk above. If you have trouble getting Google Voice to work using a plain-text password, try this Google Voice Reset Procedure. It usually fixes connectivity problems. If it still doesn’t work, enable Less Secure Apps using this Google tool.
IMPORTANT: Once you’ve entered your credentials, you MUST restart Asterisk from the Linux command line, or Google Voice calls will fail: amportal restart
Incredible PBX Wholesale Providers Access
Nerd Vittles has negotiated a special offer that gives you instant access to 300+ wholesale carriers around the globe. In lieu of paying the $650 annual fee for the service, a 13% wholesale surcharge is assessed to cover operational costs of TelecomsXchange. In addition, TelecomsXchange has generously offered to contribute a portion of the surcharge to support the Incredible PBX open source project. See this Nerd Vittles tutorial for installation instructions and signup details.
Continue Reading: Configuring Extensions, Trunks & Routes
Don't Miss: Incredible PBX Application User's Guide covering the 31 Whole Enchilada apps
Originally published: Monday, June 18, 2018
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.
2018 TREAT: If you could use one or more free DIDs in the U.S. with unlimited inbound calls and unlimited simultaneous channels, then today's your lucky day. TelecomsXChange and Bluebird Communications have a few hundred thousand DIDs to give away so you better hurry. You have your choice of DID locations including New York, New Jersey, California, Texas, and Iowa. The DIDs support Voice, Fax, Video, and even Text Messaging (by request). The only requirement at your end is a dedicated IP address for your VoIP server. Once you receive your welcome email with your number, be sure to whitelist the provider's IP address in your firewall. For Incredible PBX servers, use add-ip to whitelist the UDP SIP port, 5060, using the IP address provided in your welcoming email.
Here's the link to order your DIDs.
Your DID Trunk Setup in your favorite GUI should look like this:
Trunk Name: IPC
Peer Details:
type=friend
qualify=yes
host={IP address provided in welcome email}
context=from-trunk
Your Inbound Route should specify the 11-digit DID beginning with a 1. Enjoy!
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.
VoIP 101: Developing a Cost-Effective SIP Strategy
In the lead up to the demise of Google Voice XMPP service next week, we wanted to offer what we have found to be a cost-effective SIP strategy which takes advantage of the best of all worlds. We would divide SIP offerings into five broad categories: business-class unlimited SIP trunks, Old Faithful SIP providers, Mom-and-Pop SIP services, dirt-cheap termination services, and Gee Whiz SIP providers. As we have said many times, the beauty of setting up an Asterisk® PBX such as Incredible PBX® is you don’t have to put all your VoIP eggs in one basket. In our particular case, that has included a mix of Google Voice trunks plus all five of the SIP categories above. Today we want to document why we’ve personally made the selections we’ve made and hope that it provides a roadmap for your own VoIP setup while encouraging you to venture out of your safe zone and try some new VoIP options.
The all-you-can-eat business plans, which we previously have covered, make little sense for most home and small business users. Then there are the rock-solid, long term pay-as-you-go providers such as Vitelity and CallCentric that make perfect sense as your primary DID and SIP provider. While they may not always be the cheapest VoIP providers, the tradeoff is dependability and long-term reliability for your VoIP platform. In the case of Vitelity, it turns out the Nerd Vittles DID special (detailed below) from our Platinum Sponsor is perhaps one of the best VoIP deals on the planet.
The third category of SIP providers and our personal favorite is what we would call the mom-and-pop providers. These are typically one or two-person operations that offer incredible deals on all-you-can-eat VoIP plans for home users. Included in this category are Vestalink (available to existing customers only), Future-Nine and CircleNet. VestaLink originally began as OBiVoice and morphed over trademark issues. While the service is no longer available to new customers, it remains the best bargain at $72 for two years of unlimited inbound and outbound residential calling services. A close second goes to Future-Nine and their "Future 5 Grey" plan which provides 1,500 inbound and 1,500 outbound minutes a month for only $5. You can sign up here. Be sure to read the Terms of Services carefully, especially item #18. The New Kid on the Block is CircleNet. In addition to very attractive pay-by-the-minute offerings of $.005 per minute to most of the U.S. and Canada, they also have an $8 a month all-you-can-eat plan for residential customers that includes a very reasonable 5,000 minutes a month for calls to the following countries: United States, Canada, Australia, Bangladesh, Belgium, Brazil, Chile, Cyprus, Denmark, Finland, France, Germany, Greece , Guam, Hungary, India,Ireland, Italy, Japan, Latvia, Mexico, Netherlands, New Zealand, Norway, Poland, Puerto Rico, Singapore, Spain, Sweden, Taiwan, Thailand, United Kingdom, and Vatican City. Just let them know that you plan to use it with an Asterisk-based PBX. CircleNet also is offering Nerd Vittles readers a free month of the $8/month service to kick the tires. Simply send an email to sales@circlenet.us with your valid email address to take advantage of the offer. One free trial per customer/email address. CircleNet also offers a $15 a month business plan with even more minutes.
A fourth class of VoIP providers is the dirt-cheap termination services including Anveo Direct, TelecomsXchange, V1VoIP and the Betamax companies for low-cost international calling. These providers make terrific additions for supplementing your other VoIP services. TelecomsXchange is our personal favorite because of the special deal they have extended to Incredible PBX users. You get access to 300 VoIP wholesalers and can read about their services in this Nerd Vittles article. V1VoIP also has some terrific deals with 15¢/mo. DIDs from 13,000 Rate Centers and incoming and outgoing U.S. call pricing as low as $.003 per minute (not a typo!). Anveo Direct was perhaps the first provider to offer wholesale pricing to consumers, and they remain a terrific service both for DID and origination services with T.38 fax support as well as many of the lowest cost SIP terminations worldwide featuring user-configurable least-cost routing. Check out their pricing and rates here.
Finally, there are the SIP providers such as VoIP.ms that offer a rich collection of special features that you won’t find in many places and certainly not under the same roof. These features include SMS messaging, SIP URI proxying and iNUM for free worldwide calling, and fax support. Every one of these features is free when you sign up for an account at VoIP.ms. We encourage you to take advantage of these little known free services to enhance your PBX.
Putting It All Together. Now that we’ve covered the options, let’s go over how we would actually implement this. For the inbound trunk and primary DID, we’d recommend a SIP trunk from either Vitelity, VoIP.ms, or CallCentric. If you have multiple, simultaneous inbound calls, then the Nerd Vittles Vitelity special below can’t be beat because it provides four call paths. In addition, you get SMS support on the same trunk. Many people now assume your primary number supports SMS. We actually get dozens of unsolicited SMS messages on our home number from schools, churches, and political groups. If incoming call volume isn’t an issue, then VoIP.ms and CallCentric also offer a free iNUM number for your account. And VoIP.ms throws in a SIP URI as well.
For outbound calling for home and SOHO deployments, we recommend at least one of the mom-and-pop, all-you-can-eat providers: Future-Nine or CircleNet. If international calling is a requirement, you can’t beat the CircleNet offering. In addition to using your primary incoming provider, we also recommend you set up SIP accounts with a couple of the dirt-cheap termination providers. These don’t cost you anything other than a modest deposit unless you actually use them to place calls. And, when your primary outbound service has an outage, your PBX will never miss a beat.
The icing on the cake always has been several Google Voice trunks which work well for IVRs, Stealth AutoAttendants with DISA support, and faxing. While this may change with the demise of XMPP support, it appears that Bill Simon’s SIP Gateway to Google Voice will live on. With the Nerd Vittles sign-up link, you can migrate your existing Google Voice XMPP connections to the Simonics gateway for $4.99 each should the need arise. Enjoy!
Originally published: Monday, June 11, 2018
CircleNet SIP Setup for FreePBX/IncrediblePBX/VitalPBX/Issabel:
username=acct-id type=friend trustrpid=yes sendrpid=yes secret=acct-pword qualify=yes nat=yes insecure=port,invite host=sip.circlenet.biz fromuser=acct-id context=from-trunk disallow=all allow=ulaw Registration String: acct-id:acct-pword@sip.circlenet.biz:5060/did-num
Future-Nine SIP Setup for FreePBX/IncrediblePBX/VitalPBX/Issabel:
username=acct-num type=friend trustrpid=yes sendrpid=yes secret=acct-pword qualify=yes nat=yes insecure=port,invite host=incoming.future-nine.com fromuser=acct-num context=from-trunk canreinvite=no disallow=all allow=ulaw Registration String: acct-num:acct-pword@incoming.future-nine.com/acct-num
Need help with Asterisk? Visit the PIAF 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…
Some Fresh CallerID Magic for Incredible PBX 13-13
It’s been more than 10 years since we first introduced CallerID Trifecta for Asterisk® and the FreePBX® platform. A few months later it morphed into CallerID Superfecta and, as they say, the rest is history. Today CallerID Superfecta is used by over a million people around the globe to obtain CallerID Name (CNAM) information from over 70 different lookup sources. WOW! Just call me the Proud Papa. What a journey it has been, and our special thanks to the dozens of contributors.
Unfortunately, for those in the United States, most of the lookup sources now are commercial enterprises much like the original Bell Sisters that monopolized CallerID information by keeping it proprietary instead of forwarding CallerID Name data together with the CallerID number of incoming calls. In fact, for many users, the cost of CallerID lookups now is more expensive than the cost of the VoIP trunks carrying the calls. While we can’t solve the problem, we can make the cost of these CallerID lookup services considerably cheaper using a neat little trick with CallerID Superfecta.
Here’s the problem. Many cell phone users don’t subscribe to any service that keeps track of the caller’s identity. The same is true for those using services such as Google Voice. When you receive a call from one of these users, the CNAM provided for the caller will either show something like "Mobile Phone User" or the city and state of the caller’s cellphone provider. This doesn’t keep the commercial CNAM providers from dinging you every time you receive one of these calls. And if your girlfriend calls you a hundred times a month, the commercial CNAM providers charge you for a hundred lookups even when the calls are from the same caller. That ends today. Once you implement this new CallerID add-on, you’ll only get billed for a single CNAM lookup regardless of how many times that person calls your PBX. Better yet, with a little effort, you can quickly replace the CNAM entries of the mystery cellphone and Google Voice callers by inserting their actual names into your Asterisk phonebook using either the FreePBX GUI or the Asterisk CLI. We’ll show you how.
Here’s the trick. CallerID Superfecta lets you prioritize source lookups when incoming calls hit your PBX. For example, you can lookup the CallerID number in AsteriDex and the Asterisk Phonebook in a fraction of a second at no cost. Once there’s a match on the CallerID number of the caller, additional lookups stop. So what we want to do is prioritize the lookups so that the free sources are used before any commercial lookup service is triggered. For the commercial CNAM source in the U.S., we’re going to use OpenCNAM in our examples because their service is one of the easiest and least expensive to implement. All OpenCNAM lookup options are less than a half a cent per lookup. By placing OpenCNAM third in the Default Superfecta lookup sequence, it will only be used when there is no match in either AsteriDex or the Asterisk Phonebook.
Getting Started. It would obviously be a hassle to have to add every caller to one of these phonebooks every time a call arrives on your PBX. So the other half of today’s magic is to automatically add an entry to the Asterisk Phonebook every time a call hits your PBX. Then, the next time you receive a call from that same number, CallerID Superfecta will be able to find it in your free databases instead of querying OpenCNAM for yet another commercial lookup. As a general rule, in the United States, phonebook data is not copyrightable. However, you should consult an attorney for questions regarding the legality of this methodology as it applies to any particular CNAM lookup service.
We’re aware that CallerID Superfecta has had a caching option which provides similar functionality; however, users have reported mixed results and data corruption with that feature so we’ll be implementing a slightly different solution that relies on Asterisk’s extremely reliable SQLite3 database. This also provides the flexibility to customize the entries after completion of a call for more accurate CNAM information on subsequent calls. Unfortunately, a CNAM entry of ATLANTA, GA doesn’t tell you much about the caller even though you paid for the lookup. By inserting these call entries in the Asterisk Phonebook, you then have the option of easily customizing them in multiple ways. Not only will this provide better CNAM data on subsequent inbound calls, but the phonebook can also be used to make outgoing calls and to produce an alphabetized phonebook listing as well.
cd /var/lib/asterisk sqlite3 astdb.sqlite3 "select key,value from astdb where key LIKE '%cidname/%' order by value"
Upgrading Asterisk Phonebook Module. Before we get too deep in the weeds, let’s get your Asterisk Phonebook module in FreePBX upgraded to resolve a bug in the version that ships with Incredible PBX which won’t let you edit phonebook entries without also adding a speed dial number for each entry. For generic FreePBX platforms, you can use Module Admin to upgrade the Phonebook module. For Incredible PBX platforms, log into your server as root and issue these commands:
cd /root ./gpl-install-fpbx phonebook
Activating CallerID Superfecta. There are three steps in activating CallerID Superfecta. First, configure the Default Scheme for CallerID Superfecta in Admin:CID Superfecta using the settings we previously displayed. Use the Arrow keys to properly order the three lookup sources: AsteriDex, Asterisk Phonebook, and OpenCNAM.
Be advised that there is a credentials mismatch in the Default setup for AsteriDex. To correct this, open the Default configuration for Superfecta and click on the Tool icon to the left of AsteriDex. The username must be root, and the password is passw0rd with a zero. Save your settings.
Next, you need to create and fund an account at OpenCNAM.com. Once your account is established, copy your credentials and insert them into the OpenCNAM lookup source by clicking on OpenCNAM’s tool icon :
Once you have saved the Default Scheme, then you need to enable CallerID Superfecta for each of your Inbound Routes under the Other tab in Connectivity:Inbound Routes:
Adding Hook for Inbound Call Processing. One of the decade-old shortcomings of FreePBX has always been the inability to insert dialplan code into the Inbound Call process without forever damaging your ability to add additional DIDs moving forward. While extensions_override_freepbx.conf lets you extract and manipulate large chunks of FreePBX code, the drawback is that once you do that, the flexibility provided by the GUI goes up in smoke unless you remember to always move the newly generated dialplan code back into the override context.
We much prefer a simpler way of inserting custom dialplan code into the Inbound Call processing routine, but it requires a change in the FreePBX Core module. We would caution you about using this except on Incredible PBX 13-13 servers running under CentOS 6 or 7 or Ubuntu 18.04 with Signature Checking Disabled. It will trigger nasty signature checking errors on the FreePBX Dashboard. See the next section for a workaround on generic FreePBX 13 platforms. On Incredible PBX 13 servers, login as root and issue the following commands making sure that you expand the last two lines into a single command before executing it:
cd /var/www/html/admin/modules/core
sed -i "s|new ext_noop('CallerID Entry Point')|new ext_gosub('1','s','sub-log-caller')|" \\
functions.inc.php
mysql -u root -ppassw0rd asterisk -e "update admin set value='true' \\
where variable='need_reload'"
amportal a r
Next, we need to add some supplemental dialplan code to extensions_custom.conf. This code actually populates the Asterisk Phonebook with the new call entries each time a call is received by your PBX:
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 cd /etc/asterisk cat /tmp/sub-log-caller.txt >> extensions_custom.conf fwconsole reload amportal restart
Disabling Module Signature Checking. If you’re running FreePBX 13 behind a secure firewall that blocks web access to FreePBX for anonymous users, then here’s a simple way that you can apply the patch above and take advantage of the sub-log-caller functionality without having to endure module signature errors within FreePBX. Simply disable Module Signature Checking. Here’s how:
cd /var/www/html/admin/libraries sed -i 's|"SIGNATURECHECK", true|"SIGNATURECHECK", false|' modulefunctions.class.php mysql -u root -ppassw0rd asterisk -e "UPDATE notifications SET candelete = '1' WHERE true;" mysql -u root -ppassw0rd asterisk -e "DELETE FROM notifications WHERE true;" mysql -u root -ppassw0rd asterisk -e "UPDATE freepbx_settings SET value='0' WHERE \\ keyword='SIGNATURECHECK';" amportal restart
Adding CNAM data to CDR Listings. One of our concerns with the current CDR implementation in FreePBX is the lack of display of CNAM data for most entries in CDR listings. Currently, the only way to decipher the CNAM information is to use your mouse to hover over each of the CallerID numbers. We much prefer displaying both CallerID Name and Number in the CallerID column of CDR Reports. Here’s how to fix it:
cd /var/www/html/admin/modules/cdr sed -i "s|cdr_formatSrc($row\['src'], $row\['clid']);|cdr_formatSrc($row\['clid'], \\ $row\['clid']);|" page.cdr.php amportal a r
Managing Your Asterisk Phonebook. As we mentioned, there are a couple of ways to display and/or update entries in your Asterisk Phonebook that have been populated with OpenCNAM lookups. From the Asterisk CLI, you can display entries like this: database show cidname.
You also can add or update entries using the Asterisk CLI like this:
database put cidname 8431234567 "Mundy, Ward"
The simplest method to manage your Asterisk Phonebook is from within the FreePBX GUI. Simply navigate to Admin:Asterisk Phonebook. There you can edit individual entries and replace generic CNAM entries such as Atlanta, GA within something more descriptive of the caller’d identity. You also have the ability to export the data and even create speed dial entries for frequently called parties. Once Speed Dial numbers have been created, you can reach the party from any phone connected to your PBX by dialing *0 plus the speed dial number.
Finally, you need to complete some additional steps to implement the Spam Call Blocker as documented in this tutorial. Enjoy!
Originally published: Wednesday, May 30, 2018 Updated: Monday, November 26, 2018
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 PBX in a Flash Forum.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
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…
Dare to Compare: The Best (free) VoIP Offerings for 2018
Last week we showed you how to get 10 months of free hosting for your Incredible PBX® in the Cloud. And today we present our semi-annual survey of the latest and greatest VoIP offerings for 2018. The beauty of the cloud platform is you can try all of them for less than a penny an hour and decide for yourself which free offering best meets your needs. This year we’ve ushered in new Asterisk® 13 LTS releases of Incredible PBX® on the CentOS, Ubuntu, and Raspberry Pi platforms as well as new versions for Issabel 4 and VitalPBX. To sweeten the pot even further, we nailed down a new Cloud-based offering for $10 a year that makes a perfect VOIP sandbox for our CentOS platform. For 2018, we also secured new (free) DID offerings in the U.S. and announced a Nerd Vittles exclusive providing access to 300+ VoIP providers worldwide, all at wholesale prices. And, last but not least, we introduced Digium’s newest IP phones for Asterisk including a $59 model that makes a perfect VoIP companion.
Choosing the Best VoIP Platform for Your Needs
Choosing a VoIP platform is partially a subjective decision, but there also are some glaring red flags to consider. We suggest you begin by deciding whether your preferences include any must-have’s. Do your requirements mandate an open source solution? Do you need text-to-speech and voice recognition? Does the operating system have to be Linux-based and, if so, must it be CentOS, Debian, or Ubuntu? If you’ll be using SIP phones, must the platform include phone provisioning software for your phones, or is the ability to purchase it as an add-on sufficient? Is paid support important in making your platform decision and how much are you prepared to pay? Are automatic or pain-free software updates critical in making your selection? Is migration from an existing platform a factor? Does a preconfigured, secure firewall matter, or are you prepared to do it yourself or take your chances? Before choosing to ignore security, read this RIPS analysis of FreePBX®. Here’s a snippet from the article. Read it carefully. It’s your phone bill.
Since FreePBX is written completely in PHP, we decided to throw it into our code analysis tool RIPS. The results were more than surprising and should tell you why a rock-solid firewall is absolutely essential.
The total amount of detected vulnerabilities is very high. Luckily, the majority of the detected vulnerabilities are inside the administration control panel, such that attackers either need to steal a valid account or they have to trick an administrator into visiting a malicious website that triggers one of the critical vulnerabilities. For example, a remote command execution vulnerability could be triggered by a less critical cross-site scripting vulnerability. By chaining both vulnerabilities, the severity is increased drastically and can lead to full server compromise.
In choosing which platforms to include today, we eliminated platforms which we considered too complicated for the average new user to configure. We also eliminated any platform that did not offer at least a free tier of service with a reasonably complete feature set as part of their offering. So here’s our Pick of the Litter.
We must confess that we are partial to the Incredible PBX offerings because they provide a turnkey GPL platform with minimal configuration required on your part. Regardless of platform, all come standard with a preconfigured firewall and about three dozen applications for Asterisk that will help you learn everything there is to know about VoIP telephony.
VoIP Platform Feature Summary
Aggregation: Incredible PBX 13-13 for CentOS/SL
License: Open Source GPL
VoIP Platform: Asterisk 13
GUI: FreePBX 13 GPL modules
O/S: CentOS/SL 6.9 or 7
Phone Provisioning: Open Source
Text-to-Speech/Voice Recognition: Yes/Yes
Software Updates: Automatic Update Utility included
Migration Tools: No
Security: Fail2Ban + Preconfigured Firewall Whitelist
Security Rating (as delivered): Secure
Comments: Lean & Mean or Whole Enchilada installers as well as ISO available
Aggregation: Incredible PBX 13-13 for Raspbian
License: Open Source GPL
VoIP Platform: Asterisk 13
GUI: FreePBX 13 GPL modules
O/S: Raspbian 7
Phone Provisioning: Open Source
Text-to-Speech/Voice Recognition: Yes/Yes
Software Updates: Automatic Update Utility included
Migration Tools: No
Security: Fail2Ban + Preconfigured Firewall Whitelist
Security Rating (as delivered): Secure
Aggregation: Incredible PBX 13-13 for Ubuntu
License: Open Source GPL
VoIP Platform: Asterisk 13
GUI: FreePBX 13 GPL modules
O/S: Ubuntu 18.04
Phone Provisioning: Open Source
Text-to-Speech/Voice Recognition: Yes/Yes
Software Updates: Automatic Update Utility included
Migration Tools: No
Security: Fail2Ban + Preconfigured Firewall Whitelist
Security Rating (as delivered): Secure
Comments: Lean & Mean or Whole Enchilada installers
Aggregation: VitalPBX
License: Closed Source
VoIP Platform: Asterisk 13
GUI: Free and Commercial modules
O/S: CentOS 7
Phone Provisioning: Free
Text-to-Speech/Voice Recognition: Optional/Optional
Software Updates: Automatic
Migration Tools: Yes
Security: Fail2Ban + User-Configurable Firewall
Security Rating (as delivered): Insecure
Comments: Incredible PBX add-on now available including TM3 firewall.
Aggregation: Incredible PBX for Issabel 4
License: Open Source GPL
VoIP Platform: Asterisk 13
GUI: FreePBX 11 GPL modules
O/S: CentOS 7
Phone Provisioning: Open Source
Text-to-Speech/Voice Recognition: No/No
Software Updates: Semi-Automatic
Migration Tools: No
Security: Fail2Ban + Unconfigured Firewall
Security Rating (as delivered): Secure with Incredible PBX add-on
Comments: Incredible PBX add-on provides secure platform
Aggregation: FusionPBX for FreeSWITCH
License: Open Source MPL 1.1
VoIP Platform: FreeSWITCH 1.6
GUI: FusionPBX
O/S: Debian 8
Phone Provisioning: Free
Text-to-Speech/Voice Recognition: Optional/Optional
Software Updates: Automatic
Security: Fail2Ban + User-Configurable Firewall
Security Rating (as delivered): Secure with mods below
Comments: Incredible PBX firewall add-on now available .
Aggregation: Incredible PBX for Wazo
License: GPL3 Open Source
VoIP Platform: Asterisk 15 RealTime
GUI: Wazo GPL3 modules
O/S: Debian 9
Phone Provisioning: Extensive Open Source
Text-to-Speech/Voice Recognition: Yes/Yes
Software Updates: Automatic or 2-minute Manual
Migration Tools: No
Security: Fail2Ban + Preconfigured Firewall
Security Rating (as delivered): Secure WhiteList with Incredible PBX add-on
Comments: High Availability & Call Center GPL3 Modules
Aggregation: FreePBX Distro a.k.a. AsteriskNOW
License: Closed Source
VoIP Platform: Asterisk 13/14/15
GUI: FreePBX GPL and Commercial modules
O/S: Closed-source CentOS fork
Phone Provisioning: Open Source (minimal) or Commercial
Text-to-Speech/Voice Recognition: Optional/No
Software Updates: Manual from Hidden Repo
Migration Tools: Yes
Security: Fail2Ban + User-Configurable Firewall
Security Rating (as delivered): Insecure
Comments: Extensive commercial NagWare preinstalled
Deploying a Local Server vs. Cloud Platform
We’ve always been big fans of local servers because you have almost total control of your own destiny. This was especially true when the Raspberry Pi came along to take the financial pain out of the server equation. But the price of Cloud-based servers has continued to plummet. For 2018, you can run any of our favorites on the least expensive platform at Vultr or Digital Ocean for $2.50 a month. And, if you hurry, your first 10 months are free at Vultr. Spending another 50 cents buys you automatic backups.1 And, for the Incredible PBX 13-13 build with CentOS 6.9 (64-bit), we’ve found a deal at HiFormance that offers a high-performance OpenVZ platform at an annual cost of just $10. The technical specs are impressive (even better if you sign up for 3 years), and we don’t think you’ll find a comparable deal with anything near comparable performance and specs anywhere, period. You get your choice of hosting sites including New York, Chicago, Los Angeles, Buffalo, Atlanta, and Dallas. Complete tutorial available here.
NOTE: OpenVZ/SolusVM platforms not suitable for CentOS 7, Debian 9, or Ubuntu 18 implementations, and some providers do not yet support Ubuntu 18.04 platform although Vultr and Digital Ocean both do.
Available Free Trunks for VoIP Servers
For many years, we’ve offered free Google Voice connectivity with our VoIP platforms. And that remains true at least for a few more weeks. On all of the Incredible PBX platforms, Google Voice trunks can be set up to make free calls in the U.S. and Canada provided you have a U.S. residence and a U.S. cellphone number to verify that you are who you say you are. There’s even a ray of hope that the Simonics gateway may allow you to continue using Google Voice after Google Voice’s mid-June drop-dead date for XMPP. Details here. But what about the rest of the world. For 2018, we solved the problem by offering free DID trunks for inbound calls and a collection of 300 wholesale VoIP carriers worldwide to make outbound calls at the same wholesale rates offered to the very largest resellers. Simply pay a 13% surcharge in lieu of the $650 annual fee, and TelecomsXchange (TCXC) will provide you access to their entire suite of wholesale carriers together with state-of-the-art tools to manage all of the services.2 The Nerd Vittles setup tutorial is available here. Enjoy!
Published: Monday, March 5, 2018 Updated: Sunday, May 27, 2018
Need help with Asterisk? Visit the PBX in a Flash Forum.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
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…
- On the Vultr and Digital Ocean $2.50 platforms, be sure to (1) create a 1GB swapfile once you’ve chosen your operating system. (2) Then, for Vultr, issue the following command before beginning the Incredible PBX install: apt-get install cloud-init.
(3) Now complete the steps outlined in your preferred Nerd Vittles tutorial, and you’ll be all set in about 15 minutes. [↩] - Our special thanks to TelecomsXchange. They have generously offered to contribute a portion of the wholesale surcharge to support the Incredible PBX open source project. [↩]
Cloud 9: Free Incredible PBX in the Cloud Hosting until 2019
These deals don’t come along every day so we’re interrupting our regular programming to alert you to a terrific, limited time cloud hosting offer for first-time users of Vultr. If you hurry, you can take advantage of a $25 credit on Vultr which translates into 10 free months of cloud hosting service. We can’t say enough about Vultr. They’ve been one of our key resources for development and testing of new releases of Incredible PBX for many years. Historically, they’ve supported our open source projects through generous referral revenue although that does not apply with this special offer. If you’ve always wondered whether cloud hosting was a viable alternative to on-premise solutions, now’s your chance to kick the tires at zero cost. And the other good news is you have your choice of the following Incredible PBX offerings. Simply load the required OS or upload the ISO for the platform of your choice and follow the linked tutorials below. Enjoy!
- Incredible PBX for CentOS 6 or 7
- Incredible PBX with Incredible PBX 13-13 ISO
- Incredible PBX for Ubuntu 18.04
- Incredible PBX with VitalPBX 2.0 ISO
- Incredible PBX with Jan. 2018 Issabel ISO
- Incredible PBX with Wazo 17.17 (Debian 8)
Originally published: Friday, May 25, 2018
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 PBX in a Flash Forum.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
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…
Creating an OBi200 Google Voice Trunk to Use with Asterisk
Since Asterisk® will no longer be able to "talk" to Google Voice after June 17, we promised to hold our nose and document how to salvage your Google Voice trunks. Our exercise for today is to show you how to deploy an OBi 200-series device which can speak the new Google Voice language and use it as a traditional SIP bridge between Google Voice’s proprietary SIP platform and your Asterisk server. We will skip the editorializing on why Google is making a terrible mistake by discarding XMPP and forcing users to a proprietary solution necessitating a hardware purchase without first offering an open standards solution as Google’s Community Manager promised here. Promises, of course, don’t keep your phones ringing. For the whole story, see our article from last Saturday. For today, you’ll need to shell out $50 for an OBi 200 device. Once you have it in hand, feel free to read on and we’ll get you back in business. For security reasons, it should be noted that today’s setup assumes you are running an Incredible PBX® server and OBi device locally behind a NAT-based router. This will work equally well with the Incredible PBX-enhanced versions of Issabel and VitalPBX. We’ll leave it to the FreePBX® folks to figure out a solution for their proprietary distro.
Everything we’re covering below will work just as well using any of the OBi 200-series devices. We’ve simply chosen to use an OBi202 in our examples today because it supports an extra phone port. But an OBi200 works just as well if you will only be deploying Google Voice trunks (up to 3 and perhaps more) for your PBX. They retail for approximately $50 and are readily available at Amazon through the link in the right column which also provides a few shekels for Nerd Vittles to keep the lights on. As mentioned last week, Obihai crippled the OBi 110-series devices which will no longer work with the new Google Voice setup. Such a fine company that we once praised for producing our Device of the Year. And don’t worry. If you ever visit their forum, you can expect a cheery reception from the Obihai forum moderator. Here’s the response we got1 when raising concerns about the demise of Google Voice XMPP:
Registering Your OBi2x Device with OBiTALK
A Quick Start Guide accompanies your OBi hardware. Following along in the tutorial will get your OBi set up using a free (so far) OBiTALK account. When you get to Step 5, you’ll be ready to set up your Google Voice account by clicking the Google Voice Set-Up button.
Before you begin the Google Voice setup, we strongly recommend that you plug a POTS phone into your OBi device and dial ***6 to update your firmware to the latest release. Depending upon where you purchased your device, it may or may not have the latest firmware which is required to communicate with Google Voice on or after June 17.
We also recommend that you dial ***1 and obtain the DHCP-assigned IP address for your OBi. You’ll need this in a few minutes. And, while you’re at it, be sure to set the OBi up behind a NAT-based router to protect it from intrusion. Once someone gains access to your OBi, they’ve essentially got the keys to your telecom castle. So always deploy an OBi behind a hardware-based firewall that is on the same private LAN as your Asterisk PBX. Finally, on your router, be sure to reserve the DHCP-assigned IP address of your OBi for permanent use by the OBi hardware. Otherwise, the IP address of your OBi may change, and this will break the SIP gateway connection to your Asterisk server.
Finally, a word about the new OBi setup. All of your settings are now stored and managed in the OBiTALK cloud. Obihai then pushes the configuration to your OBi device. To put it charitably, this usually works but sometimes it doesn’t, and you end up with a quirky OBi setup that looks correct in the cloud but simply doesn’t work. We’ve found the simplest solution is to unplug the device and then restart it. Then check all of your cloud-based settings when the OBi device comes back to life to be sure none of your settings disappeared. Sometimes they do! In the old days, you had the option of configuring your OBi device locally; however, Obihai (now Polycom) has disabled that functionality with the new Google Voice setup presumably to disguise what they are doing under the covers to connect to Google.
Configuring a Google Voice Trunk on OBi200
To give credit where credit is due, configuring a Google Voice trunk on the OBi 200-series devices is dead simple. Login into your OBiTALK account, click on your OBi device, and then click the Google Voice Set-Up button.
Enter your Google Voice credentials when prompted, give Obihai permission to control your Google Voice account, and you’re done. Within a few seconds, the connections dialog box should show Google Voice connected on service provider SP1.
If you haven’t already done so, plug a POTS phone into your OBi device and place a call to somebody by dialing a 10-digit number. Then use another phone and call the Google Voice number you assigned to your OBi device. The POTS phone should ring. Don’t continue until you get these calls working in both directions. You’d be wasting your time.
Now we need to adjust the destination for incoming calls to your OBi device and redirect them from the POTS phone to the SP3 trunk we’ll be using to connect to your Asterisk server. We’ll leave SP2 unoccupied in case you wish to add another Google Voice trunk down the road.
To make this change, click the OBi Expert Configuration button at the bottom of the Device Configuration window. Then click OK to confirm that you know what you’re doing. Next click the Enter OBi Expert button at the top of the next form. In the left column, click Voice Services and then SP1 Service. The fifth parameter is called X_InboundCallRoute. Beside it, uncheck both the OBiTALK Settings and Device Default checkboxes. Now enter sp3(6781234567) in the Value field for X_InboundCallRoute where 6781234567 is your actual Google Voice phone number (DID). Scroll to the bottom and click the Submit button.
Finally, at the top of the left column of the form, click Return to OBi Dashboard.
Configuring OBi SIP Trunk for Asterisk
1. Login to your OBi Dashboard using a web browser . After signing up for an account and registering your OBi device, click on the OBi 200 device in the My OBi Devices list.
2. In the Device Configuration dialog, click OBi Expert Configuration button. When prompted whether you’re sure, click OK.
3. In the OBi Expert Configuration Menu, click Enter OBi Expert button.
4. In the Production Information (left) column, click Service Providers.
5. In the Service Providers listing, click ITSP Profile C General.
6. For each of these fields, uncheck OBiTALK Settings and then uncheck Device Default:
- General:Name
- Service Provider Info:Name
- Service Provider Info:URL
7. Fill in the ! field Values as shown below using the private IP address of your PBX:
8. Click Submit button after checking your entries carefully.
9. In the Service Providers listing on the left, click ITSP Profile C SIP.
10. In the ITSP Profile, enter the private IP address of your PBX in the Proxy Server, Registrar Server, and Outbound Proxy fields after first unchecking both the OBiTALK Settings and Device Default checkboxes.
11. Scroll down the form to X_SpoofCallerID and uncheck both the OBiTALK Settings and Device Default checkboxes. Then check the Value field for X_SpoofCallerID.
12. Scroll down the form to X_DiscoverPublicAddress and uncheck both the OBiTALK Settings and Device Default checkboxes. Then uncheck the Value field for X_Discover PublicAddress.
13. Click Submit button after checking your entries beside the 5 red exclamation points.
14. In the Production Information (left) column, click Voice Services
15. In the Voice Services listing on the left, click SP3 Service.
16. In the SP3 Service Profile, fill in the 5 fields in which the OBiTALK Settings checkbox is unchecked. The AuthUsername and AuthPassword entries will be used to authenticate to your PBX so be sure to choose a very secure password. It’s your phone bill. The URI field actually makes the trunk connection to your PBX so replace the 192.168.0.82 entry shown with the actual IP address of your PBX.
17. In the SIP Credentials section of the form, make certain that X_EnforceRequestUserID is unchecked. If not, uncheck both the OBiTALK Settings and Device Default checkboxes and then uncheck X_EnforceRequestUserID.
18. If you do not want to pass the CallerID number with your calls, in the Calling Features section of the form, be sure to check AnonymousCallEnable after unchecking both the OBiTALK Settings and Device Default checkboxes.
19. In the Service Providers listing on the left, click ITSP Profile A SIP.
20. Be sure X_SpoofCallerID is checked.
21. Click Submit button after checking your entries carefully.
Configuring Incredible PBX GUI for an OBi200
On the Incredible PBX side, log into the GUI using a web browser. We’ll be adding a SIP trunk, an outbound route, and an inbound route to process calls to and from the OBi device.
Add a SIP Trunk with a Trunk Name matching whatever you used in your OBi SIP credentials, e.g. obi200 or obi202. Plug in your Outbound CallerID to match your Google Voice phone number. In the Dialed Number Manipulation Rules tab, add a Match Pattern of NXXNXXXXXX. In the SIP Settings tab for Outgoing, the Trunk Name should match whatever you used on the OBi side, e.g. obi200 or obi202. In the PEER DETAILS, enter the following using the default username and password you assigned on the OBi side. Normally, port 5061 is the default port assigned on the OBi side. If you get a failed registration, try 5060 and then 5062 and 5063. Click Submit and reload your dialplan when finished.
type=friend defaultuser=obi200 secret=your-password qualify=yes port=5061 nat=yes host=dynamic dtmfmode=rfc2833 disallow=all context=from-trunk canreinvite=no allow=ulaw insecure=port,invite
For Outbound Call Routing, we recommend an Outbound Route using the 624 (OBI) prefix and 10-digit numbers. For example, if a user dials 624-888-1234567, your Incredible PBX server would place a call using the OBi’s Google Voice trunk to 1-888-1234567. When your Outbound Route setup looks like the following, click Submit and reload your dialplan.
For Inbound Call Routing, create an Inbound Route specifying a DID Number to match your Google Voice number. Choose a Call Destination to meet your own requirements, e.g. an extension, ring group, or IVR. Then click Submit and reload your dialplan.
Now you’re ready to test an outgoing call by dialing the OBi prefix (624) plus a 10-digit number. Then place a call to your Google Voice number using your cellphone and be sure Asterisk routes it to the destination you specified in your inbound route above.
Configuring VitalPBX to Use an OBi200
Truth be told, we weren’t bright enough to figure out how to configure the VitalPBX Trunk using credentials so we simply set up the SIP trunk using IP address authentication with the IP address of the OBi device. It works just as well and just goes to prove there’s always more than one way to skin a cat. So here’s the Trunk configuration on the VitalPBX side. The only entry you will need to change is the Host IP address for your OBi device. If you don’t know it, plug a phone into the OBi and dial ***1.
NOTE: For the Username and Description fields below, be sure to match what you used on the OBi side (above) for your SIP credentials, i.e. obi200 or obi202. If they don’t match on both devices, you won’t get a successful connection. Our apologies for mixing apples and oranges in the screenshots.
For Outbound Call Routing, we recommend an Outbound Route using the 624 (OBI) prefix and 10-digit numbers. For example, if a user dials 624-888-1234567, the VitalPBX server would place a call using the OBi’s Google Voice trunk to 1-888-1234567. Here’s the Outbound Route setup to make that happen:
For Inbound Call Routing, go to PBX:External:Inbound Routes and add an inbound route and destination for calls from your 10-digit Google Voice number. Or you can use the Default Inbound Route which we explained in our previous VitalPBX tutorial. Basically, you set up an Inbound Route with a Description and Routing Method of Default. All the other fields should be left as is except for the Inbound Destination. For the destination, you can choose an IVR, Extension, Ring Group, etc. to meet your own requirements.
Originally published: Monday, May 14, 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.
SPECIAL TREAT: If you could use one or more free DIDs in the U.S. with unlimited inbound calls and unlimited simultaneous channels, then today’s your lucky day. TelecomsXChange and Bluebird Communications have a few hundred thousand DIDs to give away so you better hurry. You have your choice of DID locations including New York, New Jersey, California, Texas, and Iowa. The DIDs support Voice, Fax, Video, and even Text Messaging (by request). The only requirement at your end is a dedicated IP address for your VoIP server. Once you receive your welcome email with your number, be sure to whitelist the provider’s IP address in your firewall. For Incredible PBX servers, use add-ip to whitelist the UDP SIP port, 5060, using the IP address provided in your welcoming email.
Here’s the link to order your DIDs.
Your DID Trunk Setup in your favorite GUI should look like this:
Trunk Name: IPC
Peer Details:
type=friend
qualify=yes
host={IP address provided in welcome email}
context=from-trunk
Your Inbound Route should specify the 10-digit DID. Enjoy!
Need help with Asterisk? Visit the PBX in a Flash Forum.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
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.
- You can always find a little humor in insults if you dig deep enough. Ironically and unbeknownst to our pal, Steve, it was Sherman Scholten and his OBi development team that were among the first Google Voice "freeloaders." Only years later after Google Voice was integrated into FreeSwitch did Josh Culp at Digium perfect a clean way to integrate Google Voice into the Asterisk platform. [↩]
VoiceMail Transcription for VitalPBX Using IBM Watson STT
Our VitalPBX adventure resumes today with one of the most requested PBX features regardless of platform. VoiceMail Transcription simply means that recorded voicemail messages are transcribed using a speech-to-text (STT) engine before being delivered in both written and recorded formats via email. The good news is we’ll show you how to harness IBM Watson’s STT to do the heavy lifting. Their platform is hands-down the best in the industry. And today we’ll walk you through the 5-minute setup procedure for your VitalPBX server.
IBM Watson’s STT solution is a real game-changer for one simple reason. Their STT API performs more accurately than any speech recognition engine in the world. As an added bonus, you won’t have to worry about Google breaking our middleware every month. On the standard plan, voicemail transcription is 2 cents per minute, or you can opt for the LITE plan which provides 100 free minutes every month. It’s worth noting that IBM doesn’t round up minutes. Transcribing two 30-second messages counts as one minute.
https://youtu.be/JWnLgZ58zsw
Obtaining IBM Watson STT Credentials
NOV. 1 UPDATE: IBM has moved the goal posts effective December 1, 2018:
If you’ve already installed the Incredible PBX add-on for VitalPBX, then IBM Watson STT already is in place. All you need is your STT (not TTS) credentials. If you haven’t installed the Incredible PBX add-on, you have two choices to get started. You can either install the Incredible PBX Custom Context now, or you can skip Incredible PBX and set up an IBM Watson account and obtain STT credentials. So start there and write down your STT credentials. You’ll need them in a minute.
Outgoing SMTP Email Setup
You obviously can’t receive voicemail messages by email if your server can’t send emails. So the next step is to configure VitalPBX to assure reliable delivery of outbound email. We strongly recommend using a Gmail account for email relay for the simple reason that many ISPs (such as Comcast) block downstream SMTP mail messages. By using Gmail as a relay host for messages sent from VitalPBX, you avoid the problem. Here’s a simple test to determine whether your server can send emails reliably. Just substitute your email address for yourname@your-email-domain.com.
echo "test" | mail -s testmessage yourname@your-email-domain.com
To configure Gmail as an SMTP relay on your VitalPBX server, login to the GUI and go to Admin:System Settings:Email Settings. Click Use External Mail Server in the Server options. Choose Gmail as the Provider. Insert the From Address to match your Gmail account name. And then enter your Gmail credentials. If you use two-step authentication with your Gmail account, you’ll first need to Obtain an Application Password to use in lieu of your regular Gmail password. Once you’ve completed all of the entries, Save your settings and Reload the Dialplan when prompted. Then send yourself a test email using the fields provided. Don’t proceed until you get this working reliably.
Installing VitalPBX Voice Recognition Engine
1. After logging into your VitalPBX server as root using SSH/Putty:
cd / wget http://incrediblepbx.com/sendmailibm-vitalpbx.tar.gz tar zxvf sendmailibm-vitalpbx.tar.gz rm -f sendmailibm-vitalpbx.tar.gz
2. Now restart Asterisk core services: asterisk -rx "core reload"
3. Edit /usr/sbin/sendmailibm and insert your IBM Watson STT credentials on lines 30 and 31. Change the language on line 34 if you don’t want en-US. Then save the file. NOTE: For new deployments, your API Username should be apikey. And your API Password will be your actual APIkey.
4. Log back into the VitalPBX GUI and configure the extensions desired for email delivery of voicemail. In PBX:Extensions:General, enter an Email Address for each extension. In PBX:Extensions:Voicemail, enter the following data using the password and timezone for each extension. Don’t enable the Delete tab until you have first tested things out.
5. If you’re using Google Voice trunks with an inbound route connecting to one or more extensions, you’ll also need to adjust the Ring Time for incoming calls, or Google Voice’s voicemail may pick up the calls before VitalPBX does. You’ll find the Ring Time setting in PBX:Extensions:Advanced for each extension. We’ve found that 20 seconds works reliably.
Originally published: Monday, April 23, 2018
Got Friends? 7 Countries Have Never Visited Nerd Vittles. 2018 Is Calling! https://t.co/wMfmlhAr16 #asterisk #freepbx #wazo #issabel #IncrediblePBX #3CX pic.twitter.com/kAmAEnwVIw
— Ward Mundy (@NerdUno) January 9, 2018
Need help with VitalPBX? Visit the VitalPBX Forum.
Sad Day. Today we say goodbye to an old friend. Feedjit has been an informative piece in the Nerd Vittles landscape for many years providing a real-time snapshot of the location of our site’s visitors and what they were reading. The following was posted on their web site today: "Due to emerging cyber risks and regulatory requirements, it is not possible to continue to operate Feedjit as a not-for-profit fun service without incurring significant costs. For this reason we are regrettably shutting down the service." We want to join the multitudes who have thanked Mark Maunder and his partner, Kerry, for their tireless efforts in providing this incredible service. We, of course, hope they will reconsider even if it means converting the site into a commercial endeavor. It was a one-of-a-kind offering that will be sorely missed in the blogosphere.
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.
Free Asterisk Solutions