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The Most Versatile VoIP Provider: FREE PORTING

Going Public with Incredible PBX 16 and VitalPBX 2.3.8



As part of our ongoing development efforts, we maintain about a dozen honeypot servers across the U.S. and Canada to monitor the latest adventures of the bad guys. Security becomes especially important for those wishing to live on the bleeding edge and deploy a cloud-based, public-facing VoIP server. Today we want to walk you through our latest suggestions to set up and secure a VitalPBX platform using just the built-in FirewallD, IPset, and Fail2Ban components. If you opt to deploy VitalPBX in the Cloud, a KVM-based VPS is absolutely essential in order to take advantage of the security mechanisms we will introduce today.

Here are 6 Key Security Features in today’s public design:

  • SIP Registration Lockdown by FQDN
  • Extension Lockdown by IP Address
  • Trunk Provider Lockdown by IP Address
  • Web Access Lockdown by WhiteList
  • Disguised Ports for SIP and SSH Access
  • 100,000+ VOIP Blacklist for FirewallD

Is it 100% safe? Nothing ever is. That’s what backups are for. 😉

FYI: The CentOS folks reintroduced a previous FirewallD bug on October 22 which (again) broke new VitalPBX installs. On October 23, the VitalPBX developers fixed the bug (again). There should be no problems with new installs. For previous installs, see this thread on the PIAF Forum for the fix.

Taking Incredible PBX with VitalPBX to the Cloud

Because Incredible PBX with VitalPBX 2.3.8 was originally distributed as an ISO, getting it installed in the cloud was a challenge. A few cloud providers let you bring your own ISO to install on their VPS platforms, but it was still a tedious process. So today we’re pleased to introduce a new install script that can be run on any CentOS 7 platform.

We have a few cloud providers that we recommend without reservation. Both Vultr and Digital Ocean provide referral credits to Nerd Vittles to support our VoIP project development efforts. We’ve used both of them for many years with no problems. Either of the platforms works well using the $5 a month option in your choice of cities. Just be sure to choose the CentOS 7 platform, not CentOS 8. For an extra buck, you can add automatic backups.

Our favorite bargain is now CrownCloud in Los Angeles. For $25 a year, they offer a KVM VPS that is ideal as a VoIP platform. And the offering includes a free snapshot image as well. As you might imagine, it’s very popular and goes Out of Stock from time to time so check back often. For our international friends, CrownCloud offers similar platforms at the same price point in both Germany and the Netherlands.

Installing Incredible PBX with VitalPBX on CentOS 7

Once your CentOS 7 platform is up and running, here’s how to install Incredible PBX for VitalPBX. Log into your server as root using SSH or Putty. Then issue these commands:

cd /root
passwd
yum -y install net-tools wget nano tar
wget http://incrediblepbx.com/incrediblepbx.sh
chmod +x incrediblepbx.sh
./incrediblepbx.sh

Incredible PBX Cloud Setup Recipe for VitalPBX

We think the easiest way to configure your new VitalPBX platform is to follow the simple steps outlined below. This will avoid your having to jump back and forth between tutorials to get all the pieces in place. When you’re finished, you’ll have a secure VitalPBX cloud platform. Don’t be intimidated by the number of steps. If you can handle slice-and-bake cookies, you can do this!

1. Point your browser to the IP address of your server. You’ll be prompted to set a password for admin access to the GUI. Fill in the blanks to proceed. Should you ever forget your admin password, here’s how as root user to force a reset on your next login from a browser:

mysql ombutel -e 'update ombu_settings set value = "yes" where name = "reset_pwd"'


2. Register your server when prompted. The VitalPBX Dashboard will appear.

3. Decipher the public IP address of your desktop machine and any other PCs that will be used to manage your server.

4. From the VitalPBX Dashboard, navigate to Admin:Security:Firewall:WhiteList. Enter each of your IP addresses from step #3 and click Save button.

5. From the VitalPBX Dashboard, navigate to Admin:Security:Intrusion Detection:WhiteList. Enter each of your IP addresses from step #3 and click Save button.

6. Modify the default SSH port by logging in to your server as root and issuing the following commands using the year you were born in the first line replacing 2000:

sed -i 's|#Port 22|Port 2000|' /etc/ssh/sshd_config
systemctl restart sshd

 
7. From the VitalPBX Dashboard, navigate to Admin:Security:Firewall:Services. Change the SIP port to 5080 or some other port number not in the 5060-5065 range. Change the SSH port to a 4-digit number matching the year you were born. Click Save button. Monitor your SSH log for attempted breaches and change your port if necessary:

cat /var/log/secure | grep password

 
8. Verify that you can log back into your server with SSH using the new SSH port number you assigned in step #6: ssh -p 2000 root@server-IP-address

9. From the VitalPBX Dashboard, navigate to Admin:Security:Firewall:Rules. Delete the HTTP and HTTPS items by clicking the Trash icon beside each entry. In the GENERAL tab, set Block ICMP Requests to YES. Click Save button. This blocks web access to everyone except those you’ve whitelisted in step #4 above. If you ever lock yourself out of web access, login to your server as shown in step #8 and temporarily whitelist the public IP address desired. This gets removed automatically the next time you save your Firewall settings from within the VitalPBX GUI.

iptables -A vpbx_white_list -s 12.34.56.78 -j ACCEPT

10. Before we get too far along, let’s put another layer of security in place for your new server. We’re going to add the VoIP Blacklist which blocks about 100,000 bad guys from around the globe. We’ll also add a cron job to update the blacklist every night. Log back into your server as root and issue these commands to put the pieces in place and enable the VoIP Blacklist.

TIP: The cron job below is scheduled to run at 20 minutes after 3 a.m. Change the time to something else so we don’t all bombard the VoIP Blacklist site for downloads at exactly the same time every night.

cd /etc
wget http://incrediblepbx.com/voipbl-firewalld.tar.gz
tar zxvf voipbl-firewalld.tar.gz
rm -f voipbl-firewalld.tar.gz
echo "20 3 * * * root /etc/update-voipbl.sh >/dev/null 2>&1" >> /etc/crontab
/etc/update-voipbl.sh

11. From the VitalPBX Dashboard, navigate to Admin:Add-Ons:Add-Ons. Click Check Online button. Click Install button beside Custom Contexts. Click Install button beside Phonebooks. Click Install button beside Domotic.

12. From the VitalPBX Dashboard, navigate to Settings:Tech Settings:SIP Settings.

  a. In the GENERAL tab, set the Bind Address port to 5080 or whatever port you chose in step #7 above. This is the port number together with the FQDN of your PBX (set in the next step) that any SIP phone will need to successfully register to an extension.

  b. In the SECURITY tab, set Allow Guest to NO, set Auto-Domain to NO, set Allow External Domains to NO, and enter a fully-qualified domain name (FQDN) pointing to the IP address of your server in the Domain field. We cannot stress enough how important this FQDN is to the security of your cloud-based server. It limits SIP registrations to this FQDN only, and all SIP registration attempts by IP address are automatically blocked. Don’t skip this step!

  c. In the NETWORK tab, enter the IP address of your server in External Address. Click the ADD button in the Local Networks section and enter the private IP addresses associated with your LAN and VPN, e.g. 192.168.0.0/255.255.0.0 and 10.0.0.0/255.240.0.0. Change NAT to Force,Comedia if your server is behind a NAT-based router.

  d. In the CODECS tab, enable ULAW, ALAW, G722, and G729.

  e. In the OTHERS tab, set SRV LOOKUPS to Yes. Click SAVE button.

13. From the VitalPBX Dashboard, navigate to Settings:Tech Settings:Profiles. Click Show All Profiles bar and choose Default PJSIP Profile. In the GENERAL tab, set the following entries to YES: Force rport, Rewrite Contact, Direct Media, RTP Symmetric, and Send Diversion Header. Click UPDATE button.

14. From the VitalPBX Dashboard, navigate to PBX:Applications:Parking. Click Show All Parking Profiles bar and choose Default. Change Code from 700 to 7000 and click Update button. This changes your Parking Lot extensions to the 7000 range so that 700 range can be used for Extensions, just like other versions of Incredible PBX.

15. Log out of your Dashboard and then log back in so that the menus get refreshed with the Custom Contexts addition.

16. From the VitalPBX Dashboard, navigate to PBX:Applications:Custom Contexts. Create the new sample IVR context with the following entries. Then click Save button.

  • Description: IncrediblePBX
  • Context: incrediblepbx
  • Extension: s
  • Priority: 1
  • Destination: Terminate Call -> Hangup

17. From the VitalPBX Dashboard, navigate to PBX:Applications:Custom Applications. Create the custom application for the sample IVR and Save it.

  • Code: 3366
  • Name: DEMO
  • Enabled: YES
  • Destination: Custom Contexts -> IncrediblePBX

18. From the VitalPBX Dashboard, navigate to PBX:Applications:Conferences. Create the new sample conference application and Save it.

  • Code: 2663
  • Description: CONF
  • Music on Hold When Empty: YES
  • User Count: YES
  • Announce Join/Leave: YES
  • Announce Only User: YES
  • User PIN: 1234
  • Leader PIN: 4321
  • Drop Silence: YES

19. If you didn’t read last week’s article on Custom Contexts, now would be a good time to do so. Here are the commands to put all those pieces in place on your new cloud-based server:

cd /
yum -y install dialog wget nano tar mailx
cp -p /etc/crontab /etc/crontab.bak
wget http://incrediblepbx.com/incrediblepbx-vitalpbx.tar.gz
tar zxvf incrediblepbx-vitalpbx.tar.gz
rm -f incrediblepbx-vitalpbx.tar.gz
chown asterisk:asterisk /var/lib/asterisk
cd /etc/asterisk/ombutel
echo "[cos-all-custom](+)" >> extensions__80-custom.conf
echo "exten => 412,1,NoOp(Voice Dialer)" >> extensions__80-custom.conf
echo " same => n,Answer" >> extensions__80-custom.conf
echo " same => n,Goto(incrediblepbx,1,1)" >> extensions__80-custom.conf
echo " same => n,Hangup()" >> extensions__80-custom.conf
echo "" >> extensions__80-custom.conf
echo "exten => 951,1,NoOp(News)" >> extensions__80-custom.conf
echo " same => n,Answer" >> extensions__80-custom.conf
echo " same => n,Goto(incrediblepbx,5,1)" >> extensions__80-custom.conf
echo " same => n,Hangup()" >> extensions__80-custom.conf
echo "" >> extensions__80-custom.conf
echo "exten => 947,1,NoOp(Weather by ZIP)" >> extensions__80-custom.conf
echo " same => n,Answer" >> extensions__80-custom.conf
echo " same => n,Goto(incrediblepbx,6,1)" >> extensions__80-custom.conf
echo " same => n,Hangup()" >> extensions__80-custom.conf
echo "" >> extensions__80-custom.conf
echo "exten => 3172,1,NoOp(DISA Voice Dialer)" >> extensions__80-custom.conf
echo " same => n,Answer" >> extensions__80-custom.conf
echo " same => n,Goto(incrediblepbx,9,1)" >> extensions__80-custom.conf
echo " same => n,Hangup()" >> extensions__80-custom.conf
echo "" >> extensions__80-custom.conf
echo "exten => 4747,1,NoOp(Wolfram Alpha)" >> extensions__80-custom.conf
echo " same => n,Answer" >> extensions__80-custom.conf
echo " same => n,Goto(incrediblepbx,3,1)" >> extensions__80-custom.conf
echo " same => n,Hangup()" >> extensions__80-custom.conf
echo "" >> extensions__80-custom.conf
echo "exten => 8463,1,NoOp(Time of Day)" >> extensions__80-custom.conf
echo " same => n,Answer" >> extensions__80-custom.conf
echo " same => n,Goto(incrediblepbx,*,1)" >> extensions__80-custom.conf
echo " same => n,Hangup()" >> extensions__80-custom.conf
echo "" >> extensions__80-custom.conf
echo "exten => 53669,1,NoOp(Lenny)" >> extensions__80-custom.conf
echo " same => n,Answer" >> extensions__80-custom.conf
echo " same => n,Goto(incrediblepbx,53669,1)" >> extensions__80-custom.conf
echo " same => n,Hangup()" >> extensions__80-custom.conf
echo "" >> extensions__80-custom.conf
echo "exten => 86329,1,NoOp(Today in History)" >> extensions__80-custom.conf
echo " same => n,Answer" >> extensions__80-custom.conf
echo " same => n,Goto(incrediblepbx,7,1)" >> extensions__80-custom.conf
echo " same => n,Hangup()" >> extensions__80-custom.conf
echo "" >> extensions__80-custom.conf
systemctl restart asterisk
chown asterisk:asterisk /var/lib/asterisk
chown asterisk:apache /var/lib/asterisk/agi-bin

20. Create new Extensions for your PBX by navigating to PBX:Extensions:Extensions. You only need to fill in the Extension, Name, and Email Address fields. We recommend extension numbers beginning with 701. If the extension will be used from a phone behind a NAT-based router, change the NAT entry to Force,Comedia. If the phone associated with the extension has a static IP address, enter it in the Permit field for an extra layer of security. In the VOICEMAIL tab, you will note that voicemail is enabled by default with a password matching the extension number. This forces the user to set the voicemail password the first time they access voicemail with their phone. We recommend the YES setting for Attach Voicemail, Ask Password, Say CID, Say Duration, and Envelope. Then press SAVE.

21. Once you have created your extensions, you can create Ring Groups to assign multiple extensions and external numbers to a designated number which will ring all of the extensions and external numbers in the ring group either simultaneously or serially. Navigate to PBX:Call Center:Ring Groups to set this up.

22. Trunk Setup. While we don’t recommend it, if you just want to play around with some toll-free calls using option 1 in the DEMO IVR to see how everything works, here’s a simple trunk setup to get you started. First, navigate to Settings:Telephony:Channel Groups and save a group named Default with no entries. Then navigate to PBX:External:Trunks:CUSTOM. Create TollFree trunk with this Dial String: SIP/1${EXTEN}@ovh.starcompartners.com. No other entries are required. Click SAVE and reload your dialplan. Finally, create an Outbound Route for these calls in PBX:External:Outbound Routes like this:

  • Description: TollFree
  • Trunks: TollFree
  • Dial Pattern: Pattern=NXXNXXXXXX

Save your settings and reload the dialplan. You now can skip down to step #25. NOTE: You will not be able to receive outside calls or make calls to numbers other than toll-free ones.

Our preference is that you use our Platinum Provider, Skyetel, for your default trunk and DID because they offer quadruple redundancy so you never miss a call. Sign up for Skyetel service and take advantage of the Nerd Vittles specials which include a $10 credit to kick the tires. First, complete the Prequalification Form here. You then will be provided a link to the Skyetel site to complete your registration. Once you have registered on the Skyetel site and your account has been activated, open a support ticket and request the $10 credit for your account by referencing the Nerd Vittles special offer. Once you are satisfied with the service, fund your account as desired, and Skyetel will match your deposit of up to $250 simply by opening another ticket. That gets you up to $500 of half-price calling. You can also port in your DIDs at no cost for 60 days after funding your account. Credit is limited to one per person/company/address/location. Effective 10/1/2023, $25/month minimum spend required.

We don’t recommend trunk registrations with a publicly exposed server because it creates a potential attack vector for intruders and any intrusion would be undetectable from the PBX since the attacker could make unauthorized calls after registering directly with your SIP provider. For this reason, Skyetel does not use SIP registrations to make connections to your PBX. Instead, Skyetel utilizes Endpoint Groups to identify which servers can communicate with the Skyetel service. An Endpoint Group consists of a Name, an IP address, a UDP or TCP port for the connection, and a numerical Priority for the group. For incoming calls destined to your PBX, DIDs are associated with an Endpoint Group to route the calls to your PBX. For outgoing calls from your PBX, a matching Endpoint Group is required to authorize outbound calls through the Skyetel network. Thus, the first step in configuring the Skyetel side for use with your PBX is to set up an Endpoint Group. Here’s a typical setup for Incredible PBX 16 for VitalPBX:

  • Name: IncrediblePBX
  • Priority: 1
  • IP Address: IncrediblePBX-Public-IP-Address
  • Port: 5062
  • Protocol: UDP
  • Description: my.incrediblepbx.com

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

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

If you’d like additional details on why we recommend Skyetel, see this Nerd Vittles article.

On the VitalPBX side, we need to add a new Skyetel trunk. Navigate to PBX:External:Trunks:PJSIP. The VitalPBX Trunk setup should look like the following for Skyetel. If you’d like to cut-and-paste the entries for the Match field, here you go:

52.41.52.34,52.8.201.128,52.60.138.31,50.17.48.216,35.156.192.164


[popup url="https://pbs.twimg.com/media/EGDhgsXWsAIbmw1?format=jpg&name=medium" width="1200″ height="700″][/popup]

In Admin:Security:Firewall:WHITELIST, you’ll need to individually Add the five Skyetel IP addresses used in the Match field above and then SAVE your firewall settings.

Finally in PBX:Incoming Calls:CID Modifiers, add a new entry for Skyetel with Skip/Length = 2/10 and Save your settings.

23. Before your PBX can receive calls, you’ll need at least one Inbound Route. This tells the PBX how to route calls from one or more phone numbers (DIDs) that you own to a destination on your PBX, e.g. an extension, a ring group, an IVR, or custom context. Navigate to PBX:External:Inbound Routes to get started. Let’s set up a default inbound route for all the DIDs you have acquired from Skyetel in step #22. Fill in the fields shown below. Then SAVE.

  • Routing Method: Default
  • Description: Default Skyetel
  • DID Pattern: [leave blank for ALL DIDs]
  • CallerID Modifier: Skyetel
  • Inbound Destination: Custom Contexts -> IncrediblePBX

24. Before you can make outbound calls from extensions on your PBX, you’ll need at least one Outbound Route. This tells the PBX which provider to use to complete calls dialed with a certain sequence of numbers. For example, you probably would want 10-digit numbers routed to Skyetel. And, if users dial 1 and then a 10-digit number, you’d probably want those calls routed to Skyetel as well. To create this outbound route, navigate to PBX:External:Outbound Routes. Fill in the fields shown below. Click ADD to add a second Dial Pattern. Click SAVE and Reload Dialplan when finished.

NOTE: While you can "spoof" any CallerID number here, it is only legal to assign CallerID numbers that you actually own. Most carriers do not forward CallerID names to destinations regardless of what you enter here. The CallerID name and number will be shown in your CDR logs: Reports:CDR Reports:CDR.

  • Description: Skyetel-OUT
  • Trunks: Skyetel
  • Outbound CID: [Your Name and CallerID Number]
  • Overwrite CID: YES
  • Dial Pattern: Prepend=1 Pattern=NXXNXXXXXX
  • Dial Pattern: Pattern=1NXXNXXXXXX

25. For the time being, we strongly recommend disabling IPv6 simply because we don’t have the necessary confidence that all of the security mechanisms are in place for IPv6. Here’s how on the CentOS 7 platform:

echo "net.ipv6.conf.all.disable_ipv6 = 1" >> /etc/sysctl.conf
echo "net.ipv6.conf.default.disable_ipv6 = 1" >> /etc/sysctl.conf
sysctl -p
sed -i 's|#AddressFamily any|AddressFamily inet|' /etc/ssh/sshd_config
systemctl restart sshd
sed -i 's|inet_protocols = all|inet_protocols = ipv4|' /etc/postfix/main.cf
systemctl restart postfix

 
26. Outbound email functionality is essential on your PBX. You’ll need it to be alerted to potential issues with VitalPBX, and you’ll need it for delivery of voicemail messages to users. There are a couple ways to implement it, and both are easy. If you want to use the native capabilities of Postfix to send the emails assuming your provider is not blocking outbound SMTP mail from downstream servers, then follow these steps:

  • Insert your FQDN from step #12b into /etc/hosts immediately after 127.0.0.1
  • Replace the contents of /etc/hostname with the same FQDN
  • Issue the following command using your actual FQDN: hostname FQDN
  • Sending yourself an email: echo "test" | mail -s test you@your-domain.com

If you don’t receive the test email message, then the easiest solution is to configure PostFix as an SMTP Relay using a Gmail account. You can do this easily from within the VitalPBX GUI. Navigate to Admin:System Settings:Email Settings and click the External Mail Server tab. Be sure that Gmail is selected and enter your Gmail name and password in the fields provided. Save your settings and send yourself an email using the field provided.

27. Once you get outbound email flowing, jump down to the next section and obtain IBM TTS and STT passwords. Now set up Voicemail Transcription with Email Message Delivery:

  a. After logging into your VitalPBX server as root using SSH/Putty:

cd /tmp
mkdir sendmail
cd sendmail
wget http://incrediblepbx.com/sendmailibm-vitalpbx.tar.gz
tar zxvf sendmailibm-vitalpbx.tar.gz
rm -f sendmailibm-vitalpbx.tar.gz
mv usr/sbin/sendmailibm /usr/sbin
cd /etc/asterisk/ombutel
echo "[general](+)" > voicemail__60-1-transcript.conf 
echo "; format=wav|wav49|gsm" >> voicemail__60-1-transcript.conf
echo "mailcmd=/usr/sbin/sendmailibm" >> voicemail__60-1-transcript.conf
chown apache:apache voicemail__60-1-transcript.conf
rm -rf /tmp/sendmail

 
  b. Restart Asterisk core services: asterisk -rx "core reload"

  c. Edit /usr/sbin/sendmailibm and insert your IBM Watson STT APIkey on line 23. Change the language on line 31 if you don’t want en-US. Then save the file.

  d. Log back into the VitalPBX GUI and configure the extensions desired for email delivery of voicemail. For each extension in PBX:Extensions:General, enter an Email Address for delivery of voicemails. In PBX:Extensions:Voicemail, verify the VM settings from step #20.

28. We hesitate to even mention (free) Festival TTS as a text-to-speech alternative because it is so bad compared to IBM TTS. But for those that like always free, here’s how to install it. Once installed, you can issue Festival commands in your dialplan using the keyword Festival followed by the text to be spoken in parentheses.

yum -y install festival
echo "[general]" > /etc/asterisk/festival.conf
asterisk -rx "dialplan reload"
festival_server &
systemctl restart asterisk
echo "/usr/bin/festival_server &" >> /etc/rc.d/rc.local

 

29. If you’d like to test the performance of your cloud-based server, here’s how to deploy and run SpeedTest:

cd /root
wget -O speedtest-cli https://raw.githubusercontent.com/sivel/speedtest-cli/master/speedtest.py
chmod +x speedtest-cli
/root/speedtest-cli

 
30. Associating CallerID Names (CNAM) with inbound calls for display on SIP phones and in the CDR logs is an often-requested PBX feature. There are a few ways to do it. First, for less than a penny a call, you can activate the feature with your DIDs in the Skyetel Dashboard. Or, for about half the cost, you can acquire an OpenCNAM account and activate it in VitalPBX by navigating to PBX:Incoming Calls:CID Lookup. Choose OpenCNAM as the Source and enter your credentials. Then SAVE your settings and reload the dialplan. Then, for each of your Inbound Routes, add OpenCNAM as the CID Lookup source and Update your configuration.

31. Unless you want a full-time job monitoring the size of your logs, remove the fail2ban Asterisk log which grows every 5 seconds. Navigate to Settings:PBX Settings:Log Files and click the Trash icon beside fail2ban. It’s probably a good idea to turn OFF the Notice option for the full log while you’re at it. Then SAVE your changes.

32. Before you do anything else, navigate to Admin:Admin:Backup & Restore, configure and run a Full Backup, and then download the file and keep it in a safe place. Be advised that Backup/Restore doesn’t restore Add-Ons, /var/lib/asterisk/agi-bin, custom contexts (extensions__80*.conf) in /etc/asterisk/ombutel, custom MySQL databases (mysqldump -u root yourDB > yourDB.sql), custom and lenny sound directories in /var/lib/asterisk/sounds, phpMyAdmin, /usr/local/sbin, and /etc/crontab.

Obtaining IBM Watson TTS and STT Credentials

Incredible PBX uses IBM Watson® for TTS and STT support. 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 generally is FREE even though you must provide a credit card when signing up. Details are provided when you sign up. If you ever forget your passwords, you can retrieve them by navigating to Resource List:Services:TTS or STT:View Full Details:Show Credentials.

Obtaining Wolfram Alpha Credentials

When people ask what exactly Wolfram Alpha is, our favorite answer was provided by Ed Borasky.

It’s an almanac driven by a supercomputer.

That’s an understatement. It’s a bit like calling Google Search a topic index. Unlike Google which provides links to web sites that can provide answers to queries, Wolfram Alpha provides specific and detailed answers to almost any question. Here are a few examples (with descriptions of the functionality) to help you wrap your head around the breadth of information. For a complete list of what’s available, visit Wolfram Alpha’s Examples by Topic. Type a sample query here. Some of our favorites include:

Weather in Charleston South Carolina
Weather forecast for Washington D.C.
Next solar eclipse
Otis Redding
Define politician
Who won the 1969 Superbowl? (Broadway Joe)
What planes are flying overhead now? (flying over your server’s location)
Ham and cheese sandwich (nutritional information)
Holidays 2012 (summary of all holidays for 2012 with dates and DOW)
Medical University of South Carolina (history of MUSC)
Star Trek (show history, air dates, number of episodes, and more)
Apollo 11 (everything you ever wanted to know)
Cheapest Toaster (brand and price)
Battle of Gettysburg (sad day 🙂 )
Daylight Savings Time 2012 (date ranges and how to set your clocks)
Tablets by Motorola (pricing, models, and specs from Best Buy)
Doughnut (you don’t wanna know)
Snickers bar (ditto)
Weather (local weather at your server’s location)

Before you can actually use our TTS implementation of Wolfram Alpha, you’ll need to obtain a free Wolfram Alpha account. As you can imagine, there have to be some rules when you’re using someone else’s supercomputer for free. So here’s the deal. It’s free for non-commercial, personal use once you sign up for an account. But you’re limited to 2,000 queries a month which works out to almost 70 queries a day. Every query requires your personal application ID, and that’s how Wolfram Alpha keeps track of your queries. Considering the price, we think you’ll find the query limitation generous compared to other web resources.

To get started, 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.

Configuring Your Incredible PBX Credentials

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 passwords for the IBM Watson TTS and STT services. The TTS credentials will look like the following: $IBM_password. The STT credentials look like this: $API_PASSWORD. Don’t mix them up. The username for both TTS and STT is now the single word: apikey

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.

If you ever want to learn how to develop applications for Asterisk, these scripts coupled with the dialplan code included in /etc/asterisk/ombutel/extensions__80-1-incrediblepbx.conf will point you in the right direction with easy to follow examples.

Managing the AsteriDex SQLite3 Database

We’ve alluded to the AsteriDex database in a couple of VitalPBX articles but never mentioned how to access it. Using a browser, point it to http://server-ip/asteridex4. You can add, edit, display, and delete entries from there. Before you can make changes in the database, issue the following command after logging into your server as root:

chown asterisk:apache /var/lib/asterisk/agi-bin

Taking Incredible PBX for a Test Drive

You can take Incredible PBX for VitalPBX on a test drive in two ways. You can call our server, and then you can try things out on your own server and compare the results. Call our IVR by dialing 1-843-606-0555. For our international friends, you can use the following SIP URI for a free call: 10159591015959@atlanta.voip.ms. For tips on setting up your own secure, hybrid SIP URI with VitalPBX, see our original tutorial. The FreePBX® setup is virtually identical except for the location of the custom SIP setting for match_auth_username=yes. On a VitalPBX server, you will enter it here: Settings:Technology Settings:SIP Settings:CUSTOM.

With Allison’s Demo IVR, you can choose from the following options:

  • 0. Chat with Operator — connects to extension 701
  • 1. AsteriDex Voice Dialer (412) – say "Delta Airlines" or "American Airlines" to connect
  • 2. Conferencing (2663) – log in using 1234 as the conference PIN
  • 3. Wolfram Alpha Almanac (4747) – say "What planes are flying overhead now?"
  • 4. Lenny (53669) – The Telemarketer’s Worst Nightmare
  • 5. Today’s News Headlines (951) — courtesy of Yahoo! News
  • 6. Weather by ZIP Code (947) – enter any 5-digit ZIP code for today’s weather
  • 7. Today in History (86329) — courtesy of OnThisDay.com
  • 8. Call Extension 701 — on your local PBX
  • 9. DISA Voice Dialer (3172) — say any 10-digit number to be connected
  • *. Current TIME and Date (8463) — courtesy of VitalPBX

CAUTION: We have intentionally disabled outbound calls using Option #9 and redirected callers to Lenny. The reason is that an unscrupulous caller could easily run up your phone bill by entering a number with expensive destination charges. If you wish to enable the feature, despite the risks, you can edit extensions__80-1-incrediblepbx.conf and make the change.

You can call your own IVR in a few ways. From an internal VitalPBX phone, dial D-E-M-O (2663) to be connected. Or simply dial the number of the DID you routed to the Incredible PBX Custom Context. Local users can also dial the individual feature codes shown in parentheses above. Be sure that you heed AND test the CAUTION documented above.

Originally published: Monday, October 21, 2019





Need help with Asterisk? Visit the VoIP-info Forum.


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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



Introducing Incredible PBX Custom Contexts for VitalPBX



Last week we introduced Incredible PBX for VitalPBX 2.3.8 to the Asterisk® community. And today we’re pleased to present the first of many Incredible PBX® add-ons, Allison’s DEMO IVR featuring text-to-speech and voice recognition utilities that take your Asterisk PBX to a whole new level. When we’re finished, you’ll have Headlines News, Weather Reports, Today in History, Wolfram Alpha, Voice Recognition apps to call people in your AsteriDex phonebook and to call anyone else by simply speaking the number to dial, and good ol’ Lenny, the Telemarketer’s Worst Nightmare. We’ll utilize a single custom context to harness the power of 11 Asterisk utilities using an easily configurable Interactive Voice Response (IVR) interface. Last week, VitalPBX released a new Custom Contexts module that has no limitations on the number of contexts you can create for free. But, for today, we’ll do it the old-fashioned way just to demonstrate what’s possible. And it’s pure GPL code.

Prerequisites. Before deploying the Incredible PBX components, you’ll need to install VitalPBX on a platform of your choice behind a hardware-based firewall. You’ll need to add an extension 701 where you can receive calls on a softphone or SIP phone, and you’ll need at least one SIP trunk to make and receive calls from those not connected to your PBX. The Outbound Route for this trunk must support 10 and 11-digit calls in the following formats: NXXNXXXXXX and 1NXXNXXXXXX. You’ll also need to install the Custom Contexts add-on to VitalPBX and then configure the (free) Custom Context for use by the Incredible PBX add-ons. The Inbound Route for at least one of your trunks must send calls to this new Incredible PBX Custom Context. Our previous tutorial will walk you through the setup steps for all of this.

In the VitalPBX GUI, navigate to PBX:Applications:Custom Applications and add a new 3366 (D-E-M-O) application for the Incredible PBX Demo IVR. Make it look like the following. Save your entries and Reload your dialplan when you’re finished.



In the VitalPBX GUI, navigate to PBX:Applications:Conference and add a 2663 (C-O-N-F) conference that looks like the following using PINs of your choice for participants and the conference leader. Save your entries and Reload your dialplan when you’re finished.



Finally, navigate to Settings:Technology Settings:SIP Settings:OTHERS and set SRV Lookups to YES. Save your settings and reload the dialplan.

Obtaining IBM Watson TTS and STT Credentials

Incredible PBX uses IBM Watson® for TTS and STT support. 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 generally is FREE even though you must provide a credit card when signing up. Details are provided when you sign up. If you ever forget your passwords, you can retrieve them by navigating to Watson:ExistingServices:TTS or STT:View Full Details:Show Credentials.

Obtaining Wolfram Alpha Credentials

When people ask what exactly Wolfram Alpha is, our favorite answer was provided by Ed Borasky.

It’s an almanac driven by a supercomputer.

That’s an understatement. It’s a bit like calling Google Search a topic index. Unlike Google which provides links to web sites that can provide answers to queries, Wolfram Alpha provides specific and detailed answers to almost any question. Here are a few examples (with descriptions of the functionality) to help you wrap your head around the breadth of information. For a complete list of what’s available, visit Wolfram Alpha’s Examples by Topic. Type a sample query here. Some of our favorites include:

Weather in Charleston South Carolina
Weather forecast for Washington D.C.
Next solar eclipse
Otis Redding
Define politician
Who won the 1969 Superbowl? (Broadway Joe)
What planes are flying overhead now? (flying over your server’s location)
Ham and cheese sandwich (nutritional information)
Holidays 2012 (summary of all holidays for 2012 with dates and DOW)
Medical University of South Carolina (history of MUSC)
Star Trek (show history, air dates, number of episodes, and more)
Apollo 11 (everything you ever wanted to know)
Cheapest Toaster (brand and price)
Battle of Gettysburg (sad day 🙂 )
Daylight Savings Time 2012 (date ranges and how to set your clocks)
Tablets by Motorola (pricing, models, and specs from Best Buy)
Doughnut (you don’t wanna know)
Snickers bar (ditto)
Weather (local weather at your server’s location)

Before you can actually use our TTS implementation of Wolfram Alpha, you’ll need to obtain a free Wolfram Alpha account. As you can imagine, there have to be some rules when you’re using someone else’s supercomputer for free. So here’s the deal. It’s free for non-commercial, personal use once you sign up for an account. But you’re limited to 2,000 queries a month which works out to almost 70 queries a day. Every query requires your personal application ID, and that’s how Wolfram Alpha keeps track of your queries. Considering the price, we think you’ll find the query limitation generous compared to other web resources.

To get started, 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.

Installing Incredible PBX Components

To install the Incredible PBX components on your VitalPBX server, login to your server as root and issue the following commands:

# on VitalPBX 3.x servers only, issue next 2 commands:
ln -s /etc/asterisk/vitalpbx /etc/asterisk/ombutel
ln -s /usr/share/vitalpbx /usr/share/ombutel

cd /
yum -y install dialog wget nano tar mailx
cp -p /etc/crontab /etc/crontab.bak
wget http://incrediblepbx.com/incrediblepbx-vitalpbx.tar.gz
tar zxvf incrediblepbx-vitalpbx.tar.gz
rm -f incrediblepbx-vitalpbx.tar.gz
chown asterisk:asterisk /var/lib/asterisk
chown asterisk:apache /var/lib/asterisk/agi-bin
/etc/init.d/asterisk restart

The GPL3 license for all of the Incredible PBX components is available for review in /root/COPYING.

Configuring Your Incredible PBX Credentials

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 passwords for the IBM Watson TTS and STT services. The TTS credentials will look like the following: $IBM_password. The STT credentials look like this: $API_PASSWORD. Don’t mix them up. The username for both TTS and STT is now the single word: apikey

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.

If you ever want to learn how to develop applications for Asterisk, these scripts coupled with the dialplan code included in /etc/asterisk/ombutel/extensions__80-1-incrediblepbx.conf will point you in the right direction with easy to follow examples.

Using Asteridex with VitalPBX

AsteriDex is a web-based dialer and address book application for Asterisk and VitalPBX. 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://vitalpbx-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.

Before AsteriDex Click-to-Call will work, you must authorize AsteriDex to access Asterisk from your browser. After logging into your server as root, edit the following file in /etc/asterisk/ombutel: manager__50-ombutel-user.conf. For each browser IP address you wish to authorize, add an entry like the following immediately below the existing permit entry in the file. Then reload the Asterisk dialplan: asterisk -rx "dialplan reload"

permit=12.34.56.78
permit=192.168.0.0/255.255.255.0

Taking Incredible PBX for a Test Drive

You can take Incredible PBX for VitalPBX on a test drive in two ways. You can call our server, and then you can try things out on your own server and compare the results. Call our IVR by dialing 1-843-606-0555. For our international friends, you can use the following SIP URI for a free call: 10159591015959@atlanta.voip.ms. For tips on setting up your own secure, hybrid SIP URI with VitalPBX, see our original tutorial. The FreePBX® setup is virtually identical except for the location of the custom SIP setting for match_auth_username=yes. On a VitalPBX server, you will enter it here: Settings:Technology Settings:SIP Settings:CUSTOM.

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 now?"
  • 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 VitalPBX

CAUTION: We have intentionally disabled outbound calls using Option #9 and redirected callers to Lenny. The reason is that an unscrupulous caller could easily run up your phone bill by entering a number with expensive destination charges. If you wish to enable the feature, despite the risks, you can edit extensions__80-1-incrediblepbx.conf and make the change.

You can call your own IVR in two ways. From an internal VitalPBX phone, dial D-E-M-O (2663) to be connected. Or simply dial the number of the DID you routed to the Incredible PBX Custom Context. Either way, you should be connected to the Incredible PBX IVR running on your VitalPBX server. Be sure that you heed AND test the CAUTION documented above.

Using Incredible PBX Features Independently of IVR

Now that we have some of the Incredible PBX applications in place using the IVR, there may be situations in which you want callers to be able to access the individual components without navigating through the IVR. For example, you may want to let users contact LENNY (53669) directly. There are three steps to do this: (1) add a new Custom Context for the feature desired, (2) add a new Custom Application with the number to associate with the application, and (3) add a snippet of dialplan code to extensions__80-custom.conf redirecting the Custom Context to the matching IVR priority. Then reload Asterisk dialplan.

For the new Lenny Custom Context, here are the entries:

Description: Lenny
Context: lenny
Extension: s
Priority: 1
Destination: Terminate Call -> Hangup

For the new Lenny Custom Application, here are the entries:

Code: 53669
Name: Lenny
Enabled: Yes
Destination:  Custom Contexts -> Lenny

In /etc/asterisk/ombutel/extensions__80-custom.conf, add the following dialplan code and reload the Asterisk dialplan: asterisk -rx "dialplan reload"

[lenny]
exten => s,1,NoOp(Lenny)
 same => n,Answer
 same => n,Goto(incrediblepbx,53669,1)

Let’s walk through one more example to be sure you get the hang of it. To mimic the News Headlines (951) setup on other Incredible PBX platforms, here are the steps:

For the News Custom Context, here are the entries:

Description: News
Context: news
Extension: s
Priority: 1
Destination: Terminate Call -> Hangup

For the News Custom Application, here are the entries:

Code: 951
Name: News
Enabled: Yes
Destination:  Custom Contexts -> News

In /etc/asterisk/ombutel/extensions__80-custom.conf, add the following dialplan code and reload the Asterisk dialplan: asterisk -rx "dialplan reload"

[news]
exten => s,1,NoOp(News)
 same => n,Answer
 same => n,Goto(incrediblepbx,5,1)

Coming Events. We still plan to address security for those that want to deploy this terrific platform in the cloud. And we’re already working on a new version of CallerID Superfecta for VitalPBX that will let you use OpenCNAM to associate names with CallerID numbers. You can follow our progress and offer suggestions on the PIAF Forum.

Originally published: Monday, October 14, 2019





Need help with Asterisk? Visit the VoIP-info Forum.


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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



FusionPBX on Steroids: Text-to-Speech Apps Have Arrived


SECURITY ALERT: https://securityboulevard.com/2019/06/rce-using-caller-id-multiple-vulnerabilities-in-fusionpbx/

And you thought you needed an Asterisk® PBX for your users to enjoy FREE text-to-speech applications such as current News Headlines and Weather reports from the convenience of their telephone. Well, move over Asterisk. FusionPBX™ for FreeSWITCH™ now offers virtually identical functionality with all of the terrific advantages that FusionPBX provides: reliability, updates, performance, security and an unmatched UC platform with no rivals. To get started, make sure you have completed the steps in our FusionPBX introductory tutorial.

Intuitive support in FusionPBX for interactive TTS or STT applications is not (yet) available. So we’re doing the next best thing. Once or more a day, we will use cron jobs to retrieve the latest News Headlines and Weather reports for your local area. Then anyone using your PBX can pick up a phone and listen to the News Headlines by dialing 951 or U.S. weather forecasts by dialing 947, or worldwide weather forecasts from ApiXU by dialing 949.1 We’ll be using IBM’s awesome TTS engine to handle the text-to-speech chores. We think you will agree that IBM’s offering is the best in the business. And you can’t beat the price. After your first free month, you get a million characters of FREE text-to-speech synthesis every month forever! For ApiXU worldwide weather data, your first 2,500 queries are also FREE every month.

Here’s a sample from the 3CX implementation of these identical applications:


[soundcloud url="https://api.soundcloud.com/tracks/364353344″ params="auto_play=false&hide_related=false&show_comments=true&show_user=true&show_reposts=false&visual=true" width="80%" height="350″ iframe="true" /]

 

Getting Started with IBM Bluemix TTS Service

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

You can start your free, 30-day trial of IBM Bluemix services without providing a credit card. Just sign up here. Once your account is activated, here’s how to obtain credentials for the TTS service to use with FusionPBX. Start by logging in to your IBM Bluemix account. Once you’re logged in, click on your account name (1) in the upper right corner of your web page to reveal the pull-down to select your Region, Organization, and Space. Follow the blue links at the bottom of the pull-down menu to create an Organization and Space for your TTS service.



Next, click the Menu icon which is displayed as three horizontal bars on the left side of the web page. Choose Watson. Click Create Watson Service and select Text to Speech from the applications listing. Watson will generate a new TTS service template and display it. Make certain that your Region, Organization, and Space are shown correctly. Then verify that the Standard Pricing Plan is selected. When everything is correct, click the Create button.

When your Text to Speech application displays, click Service Credentials and then click New Credential (+). When the Add New Credential dialog appears, leave the default settings as they are and click Add. Your Credentials Listing then will appear. Click View Credentials beside the new entry you just created. Write down your URL, username, and password. You’ll need these in Step #4 below to configure the IBM Bluemix TTS service. Logout of the IBM Cloud by clicking on the little face in the upper right corner of your browser window and choose Log Out. Confirm that you do, indeed, wish to log out.

Getting Started with ApiXU Weather

Finding free worldwide weather forecasts has been a difficult nut to crack. So we’re pleased to introduce ApiXU. Your first 5,000 API calls every month are free, but our Worldwide Weather application for FusionPBX actually makes two API calls to retrieve the latest weather conditions AND the weather forecast. What that means is you can make 2,500 free queries a month with the Nerd Vittles application. One or two a day should suffice. While the U.S. weather reports are retrieved by ZIP code, the ApiXU queries are retrieved by city. So long as you don’t choose small towns, the city names should be sufficiently unique to work well with the WorldWide Weather application. HINT: Nicosia in Cyprus (home of 3CX) works great! 😉




Before you can obtain worldwide weather reports, you’ll need to sign up for an account at ApiXU.com. Once you’re registered, log into your account and copy down your API Key. You’ll need it in a minute.

5 Steps to TTS Paradise with FusionPBX

Once you have your IBM TTS credentials in hand, there are only five simple steps to get everything set up for TTS application support on FusionPBX. When we’re finished, anyone on your PBX can pick up a phone and listen to the News Headlines by dialing 951, a U.S. Weather Forecast by dialing 947, or Worldwide Weather for most international cities by dialing 949.

  1. Download WAV file placeholders to FusionPBX
  2. Set up TTS Extensions in FusionPBX
  3. Install the Linux components to support TTS Applications
  4. Insert IBM and ApiXU Credentials, Email Address and Weather Locations
  5. Run the News Headlines and Weather Update Scripts

1. Downloading WAV File Placeholders

Login to your FusionPBX server as root using SSH or Putty. Change to /var/lib/freeswitch/recordings directory. List its contents to decipher the names of any subdirectories that have been created for your various FusionPBX domains. Change to each subdirectory under /var/lib/freeswitch/recordings and issue the following commands to install the TTS placeholders:

wget http://incrediblepbx.com/freeswitch/placeholders.tar.gz
tar zxvf placeholders.tar.gz
rm -f placeholders.tar.gz

IMPORTANT: Once you’ve copied the placeholders into position, use a browser to open the FusionPBX Dashboard for each of your domains. Navigate to Apps then Recordings and play each of the three placeholder files that were uploaded: News-update, Weather-forecast, and Weather-zip. Otherwise, they won’t be available for use in the next step of the setup.

2. Setting Up TTS Apps in FusionPBX

Before you can implement the Nerd Vittles TTS Apps for News Headlines, Weather by ZIP Code, and Worldwide Weather, we first need to create the proper environment on the FusionPBX side to support the new applications. We’ll be using the FusionPBX Dialplan Manager for this purpose. We need to set up three extensions to handle the calls: one for the News Headlines and one for each of the Weather applications.

Login to your FusionPBX Dashboard with a browser.

News Headlines: From the FusionPBX Dashboard, navigate to DialPlan, then Dialplan Manager, and click the Add (+) icon. Using your default Context, insert the following new entry into the Dialplan for News Headlines (951) by filling in the Name, Condition1, Action1, and Description fields as shown below. Leave the other defaults. Then click SAVE.



When the Dialplan listing reappears, find the NewsHeadlines entry in the list and click the pencil icon to Edit the entry. Add 951 in the Number field as shown below. Then click SAVE and BACK.



Now let’s add the Dialplan entries to support the two Weather applications.

Weather by ZIP Code: From the FusionPBX Dashboard, navigate to DialPlan, then Dialplan Manager, and click the Add (+) icon. Using your default Context, insert the following new entry into the Dialplan for Weather by ZIP Code (947) by filling in the Name, Condition1, Action1, and Description fields as shown below. Leave the other defaults. Then click SAVE.



When the Dialplan listing reappears, find the WeatherZIP entry in the list and click the pencil icon to Edit the entry. Add 947 in the Number field as shown below. Then click SAVE and BACK.



Worldwide Weather: From the FusionPBX Dashboard, navigate to DialPlan, then Dialplan Manager, and click the Add (+) icon. Using your default Context, insert the following new entry into the Dialplan for Worldwide Weather (949) by filling in the Name, Condition1, Action1, and Description fields as shown below. Leave the other defaults. Then click SAVE.



When the Dialplan listing reappears, find the WorldwideWeather entry in the list and click the pencil icon to Edit the entry. Add 949 in the Number field as shown below. Then click SAVE and BACK.



Try things out by dialing 947, 949, and 951 from any FusionPBX extension. Be sure these work before proceeding!

3. Installing Linux Components for TTS

First, we need to get the missing pieces in place to support TTS applications using IBM Bluemix TTS and the Nerd Vittles scripts. We want to add PHP support from the Linux CLI only so there will be no security issues. And we want to add support for SQLite 3 so we can look up latitude and longitude data for U.S. zip codes. Just issue the following commands to get everything set up:

apt-get update
apt-get -y install php-fpm php-curl php-cli php-pear php-db php-gd sqlite3 libsqlite3-dev
apt-get -y install sox lame libsox-fmt-mp3
sed -i 's|;cgi.fix_pathinfo=1|cgi.fix_pathinfo=0|' /etc/php/7.1/fpm/php.ini
systemctl restart php7.1-fpm

Next, we need to put the Nerd Vittles scripts and ZIP code database for SQLite 3 in place:

cd /
wget http://incrediblepbx.com/freeswitch/fusionpbx-tts-linux.tar.gz
tar zxvf fusionpbx-tts-linux.tar.gz
rm -f fusionpbx-tts-linux.tar.gz

Finally, we need to add cron jobs to run the three update scripts at least once a day. You can run them more often depending upon your needs. We have these configured to run at 6:15 am and 6:20 am every day. Adjust to meet your own requirements. On a busy PBX, you probably don’t want to run them during the workday.

echo "15 6 * * * root /root/nv-weather-update.sh >/dev/null 2>&1" >> /etc/crontab
echo "20 6 * * * root /root/nv-news-update.sh >/dev/null 2>&1" >> /etc/crontab
echo "25 6 * * * root /root/nv-wwweather-update.sh >/dev/null 2>&1" >> /etc/crontab

4. Adding TTS Credentials to FusionPBX

Now we need to add your IBM TTS and ApiXU credentials, email address, a local ZIP code for Weather by ZIP code reports, and a city for Worldwide Weather reports. Edit the credentials file and save it with your information:

cd /root
nano -w ibm-credentials.php

5. Running the News & Weather Update Scripts

Finally, we need to run the News Headlines and two Weather update scripts once to put current data in place for FusionPBX callers. After the initial setup, the cron jobs will update the News Headlines and Weather reports every day moving forward. Press ENTER as each of the scripts finishes to get back to a command prompt.

cd /root
./nv-news-update.sh
./nv-weather-update.sh
./nv-wwweather-update.sh

Taking the News & Weather Apps for a Spin

Now you’re ready to try things out. From any phone connected to your PBX, dial 951 for current News Headlines. Then dial 947 for a local Weather Report matching your zip code. Finally, dial 949 to retrieve a worldwide weather forecast for most international cities.

If you don’t yet have a FusionPBX server set up but would like to sample the voice quality of the TTS applications running on our FusionPBX server in New York, here are several ways to try them out using an IVR we set up using an IBM voice prompt from last week’s tutorial. Airport codes reflect (PROVIDER LOCATION-SERVER LOCATION-DID LOCATION).

  • Skyetel DID: 843-970-9997 (SEA-BUF-CHS)
  • Vitelity DID: 646-666-5997 (DEN-BUF-NYC)
  • VoIPms DID: 843-606-0444 (ATL-BUF-CHS)
  • Free iNUM Call: 883510009901997 (ATL-BUF-ATL)
  • Free SIP Call: 883510009901997@sip.inum.net (ATL-BUF-ATL)

Originally published: Monday, September 24, 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 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…

  1. The included applications are licensed pursuant to GPL2 with the exception of nv-worldwide-weather.php which is licensed pursuant to The MIT License. Terms and conditions of both licenses are included in /root/COPYING. []

Creating Free IBM Voice Prompts for FusionPBX/FreeSWITCH

SECURITY ALERT: https://securityboulevard.com/2019/06/rce-using-caller-id-multiple-vulnerabilities-in-fusionpbx/

One of the first things you’ll need if you choose to migrate to FusionPBX and FreeSWITCH is voice prompts. You can record your own in FusionPBX using the Recordings application by dialing *732. Of course, your PBX will probably sound like you recorded your own prompts. 🙂 Our first recommendation is to always direct folks to Allison Smith whose voice prompts for Asterisk are legendary. But, for those on a tight budget, recordings by a professional voice talent may not be a viable option. So, today, we want to deliver the next best thing with synthesized voice prompts that are second to none. For regular readers of Nerd Vittles, you’re already aware of our enthusiasm for IBM’s new TTS offerings. You can try them out for yourself by clicking View Demo here. Or you can sample the Weather Report for 3CX that we uploaded to SoundCloud:


[soundcloud url="https://api.soundcloud.com/tracks/364353344″ params="auto_play=false&hide_related=false&show_comments=true&show_user=true&show_reposts=false&visual=true" width="80%" height="350″ iframe="true" /]

Cost issues aside, other users may need voice prompts for VoIP applications that require a language other than English. With IBM’s TTS offerings, you have quite a selection of voices and languages from which to choose:

There are many occasions where you may want to build IVRs or applications that require custom voice prompts to obtain information from callers such as requests for a name, a phone number, a part number, a location, a department, or many other pieces of data that are used to formulate data base queries. And now there’s a simple, professional, and free way to create these voice prompts using IBM’s TTS platform and FusionPBX. The first million characters of TTS synthesis and the resultant voice files are free every month. That will be more than ample for almost all of our users. To get started, you’ll need to set up a free account with IBM Bluemix and install FusionPBX on a platform of your choice: a dedicated server, a virtual machine on your desktop PC using VirtualBox, or a cloud-based server.

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

Getting Started with IBM Bluemix TTS Service

You can start your free, 30-day trial of IBM Bluemix services without providing a credit card. Just sign up here. Once your account is activated, here’s how to obtain credentials for the TTS service to use with FusionPBX.

Getting Started with FusionPBX

We won’t repeat the tutorial that walks you through installation of FusionPBX. Just follow the steps outlined here. Once your server is up and running, log into your server as root using SSH or Putty. We need to add MP3 support to the SOX application before we can create voice prompts reliably with IBM’s Bluemix TTS service. Here’s how:

apt-get update
apt-get install libsox-fmt-mp3 -y

Installing the Voice Prompts Script for TTS

Now we’re ready to install the Nerd Vittles Voice Prompts script that we’ll use to actually create the custom voice prompts. While you’re still logged into your server as root with SSH or Putty, issue the following commands:

cd /root
wget http://incrediblepbx.com/freeswitch/ibmprompt-fusion.tar.gz
tar zxvf ibmprompt-fusion.tar.gz
rm -f ibmprompt-fusion.tar.gz

Adding Your Credentials to the Script

Using your favorite editor, it’s time to add your IBM TTS credentials to the Voice Prompt script: nano -w ibmprompt.php. Simply replace the x’s in $API_KEY with your credentials from above. If you prefer a different voice for your voice prompts, update the $IBM_voice option using the examples shown below. For example, for the Brazilian Portuguese voice, use $IBM_voice = "pt-BR_IsabelaVoice". Verify that the $IBM_url matches what was provided with your credentials. Once you’ve updated the entries, save the file: Ctrl-X, Y, and ENTER.

Taking Voice Prompts Script for a Test Drive

Now we’re ready to try things out. The syntax while logged into the /root folder looks like this. If creating a prompt in a different language, text should be in native language, not English.

./ibmprompt.php "Text of your voice prompt"

Once the voice prompt is generated, you’ll find voiceprompt.wav in the /root folder. We’ve added a second script to actually move the new voice prompts into place to use with FusionPBX. So, once you’ve created the voice prompt above, issue the following command to assign a prompt name and copy the prompt into your FusionPBX recordings folders. Don’t forget the .wav extension on the prompt name you choose, and don’t put any spaces in the prompt name. Use hyphens.

./ibmprompt-to-fusionpbx prompt-name.wav

Once you have put the recordings in place in FusionPBX, you can edit and play them back within the FusionPBX GUI by navigating to Apps -> Recordings. Enjoy!

Originally published: Wednesday, September 19, 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 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 or 3CX? 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…

Back to School: Introducing FusionPBX for FreeSWITCH

SECURITY ALERT: https://securityboulevard.com/2019/06/rce-using-caller-id-multiple-vulnerabilities-in-fusionpbx/

It’s been quite a week with the surprise acquisition of Digium® and Asterisk® by Sangoma®. It became official on Wednesday, September 5. You can read all about it here, and you can read our cautious optimism here. As with the recent Google Voice transformation, we hope it serves as a gentle reminder to the VoIP community not to put all your eggs in one basket. With the start of the new school year, we could think of no better time to explore an excellent alternative. And today we’re pleased to introduce FusionPBX™ for FreeSwitch™.

9/10 EDIT: We’ll be updating this article in coming days to add tutorials on additional features rather than releasing new articles that force you to jump around. So mark your place at the end of the article and come back soon to see the new additions.

FreeSWITCH is an open source softswitch that’s been around for over a decade. The lead designer is Anthony Minessale, who originally worked on the Asterisk project. FusionPBX is a GUI front end for FreeSWITCH that performs many of the same functions that FreePBX® performs for Asterisk. It’s the brainchild of Mark J. Crane. With that background, let’s dive right in.

Today we’ll get a functioning server set up with trunks and extensions so that you can begin making calls. We’ll also show you how to interconnect with an Incredible PBX server in the Cloud to add Google Voice GVSIP functionality for free calling in the U.S. and Canada. Once you get that far, we’d recommend you pick up a good book on FreeSWITCH, review the excellent FusionPBX documentation, and roll up your sleeves. There’s virtually nothing that FusionPBX and FreeSWITCH can’t do with a telephone.

Creating the Debian 8 Minimal Platform

We’ll be building FusionPBX atop a minimal install of Debian 8 (Jessie). If you’re creating your server in the Cloud with 1GB or less of RAM (such as the $3.50/month Vultr platform), we strongly recommend creation of a swap file after you set up the Debian 8 platform:

dd if=/dev/zero of=/swapfile bs=1024 count=1024k
chown root:root /swapfile
chmod 0600 /swapfile
mkswap /swapfile
swapon /swapfile
echo "/swapfile          swap            swap    defaults        0 0" >> /etc/fstab
sysctl vm.swappiness=10
echo vm.swappiness=10 >> /etc/sysctl.conf
free -h
cat /proc/sys/vm/swappiness

Next, create a very secure root password: passwd

Now put the missing pieces in place to support your FusionPBX install:

apt-get update
apt-get upgrade
apt-get install nano -y
apt-get install dialog -y
apt-get install ca-certificates -y
apt-get install systemd -y
apt-get install systemd-sysv -y
reboot

Installing FusionPBX and FreeSWITCH

Now we’re ready to install FusionPBX with FreeSWITCH. Issue the following command on a single line. Be advised that FusionPBX currently uses FreeSWITCH 1.6, not 1.8. If you buy a book about FreeSWITCH 1.8, just be aware that there may be some features that are not yet available with FusionPBX.

wget -O - https://raw.githubusercontent.com/fusionpbx/fusionpbx-install.sh/master/debian/pre-install.sh | sh; cd /usr/src/fusionpbx-install.sh/debian && ./install.sh

When the install completes, you’ll see a message that looks something like this:

Installation has completed.

   Use a web browser to login.
      domain name: https://45.76.249.125
      username: admin*
      password: D6pHudQGqeYsQUWK

   *The browser domain name is used as part of the authentication.

   If you need to login to a different domain then use username@domain.
      username: admin@45.76.249.125

   Official FusionPBX Training
      Fastest way to learn FusionPBX: https://www.fusionpbx.com.
      Available online and in person. Includes documentation and recording.

      Location:               Online
      Admin Training:          7 -  9 August 2018 (3 Days)
      Advanced Training:      21 - 22 August 2018 (2 Days)
      Continuing Education:        30 August 2018 (1 Day)
      Timezone:               https://www.timeanddate.com/weather/usa/boise

   Additional information.
      https://fusionpbx.com/training.php
      https://fusionpbx.com/support.php
      https://www.fusionpbx.com
      http://docs.fusionpbx.com

If you’re coming from the FreePBX world and you’re new to FusionPBX and FreeSWITCH, be advised that your browser login name is NOT admin. It’s admin@some-IP-address. The reason is because FreeSWITCH supports multiple domains, unlike FreePBX. The default domain will be the IP address from which you performed the installation. On a server in the cloud, it will be your public IP address. On a private LAN, it will be the localhost private IP address, e.g. 127.0.0.1 or 127.0.0.2.

Locking Down Your Server

FusionPBX includes a basic IPtables firewall setup. Those that have followed Nerd Vittles over the years know that we view a firewall whitelist (Travelin’ Man 3) as absolutely essential to avoid security problems down the road. In the case of FusionPBX, we recommend changing the SSH access port from 22 to a random number above 1000. Then it can remain exposed so long as you check regularly to make certain no one is attempting to access your server via SSH: cat /var/log/auth.log. We also recommend locking down HTTP and HTTPS to your whitelisted IP addresses rather than leaving those ports open to the world. Finally, we recommend closing off IPv6 access to your server except from localhost. Here’s how.

Let’s assume you want to change the SSH access port from 22 to 1789. Simply issue the following commands and restart SSH. WARNING: Be careful not to log out of your server until we update the firewall, or you will lock yourself out of your server!

sed -i 's|#Port 22|Port 22|'  /etc/ssh/sshd_config
sed -i 's|Port 22|Port 1789|' /etc/ssh/sshd_config
/etc/init.d/ssh restart

To reconfigure IPtables using a WhiteList of allowed IP addresses, you first need to decipher what those IP addresses actually are. You’ll need the public and private IP addresses of any PCs from which you wish to access FusionPBX. Depending upon your pain threshold and bank account, SIP access can remain open. However, you’ll still need the IP addresses of your hosting providers and the IP addresses of each of the locations where you plan to install a SIP phone for direct access to properly configure FusionPBX. Once you have those IP addresses in hand, it’s time to edit /etc/iptables/rules.v4. The filter section of the default install looks like:

*filter
:INPUT DROP [1:40]
:FORWARD DROP [0:0]
:OUTPUT ACCEPT [58:8069]
-A INPUT -i lo -j ACCEPT
-A INPUT -m state --state RELATED,ESTABLISHED -j ACCEPT
-A INPUT -p udp -m udp --dport 5060:5091 -m string --string "friendly-scanner" --algo bm --to 65535 -j DROP
-A INPUT -p udp -m udp --dport 5060:5091 -m string --string "sipcli/" --algo bm --to 65535 -j DROP
-A INPUT -p udp -m udp --dport 5060:5091 -m string --string "VaxSIPUserAgent/" --algo bm --to 65535 -j DROP
-A INPUT -p tcp -m tcp --dport 22 -j ACCEPT
-A INPUT -p tcp -m tcp --dport 80 -j ACCEPT
-A INPUT -p tcp -m tcp --dport 443 -j ACCEPT
-A INPUT -p tcp -m tcp --dport 7443 -j ACCEPT
-A INPUT -p tcp -m tcp --dport 5060:5091 -j ACCEPT
-A INPUT -p udp -m udp --dport 5060:5091 -j ACCEPT
-A INPUT -p udp -m udp --dport 16384:32768 -j ACCEPT
-A INPUT -p icmp -m icmp --icmp-type 8 -j ACCEPT
-A INPUT -p udp -m udp --dport 1194 -j ACCEPT
COMMIT

1. Modify the SSH rule (–dport 22) replacing 22 with your new SSH port number, e.g. 1789.

2. Using #, comment out the HTTP (–dport 80) and HTTPS (–dport 443) rules. There simply are too many zero day vulnerabilities with PHP and SQL injection to leave web ports exposed to the public.

3. Just above the COMMIT line, whitelist your private LAN IP addresses. Do NOT whitelist the 172 subnet if you’re deploying on Amazon! Amazon treats these as routable IP addresses on their network.

-A INPUT -s 127.0.0.0/8 -j ACCEPT
-A INPUT -s 10.0.0.0/8 -j ACCEPT
-A INPUT -s 172.16.0.0/12 -j ACCEPT
-A INPUT -s 192.168.0.0/16 -j ACCEPT

4. If you’re planning to use NeoRouter VPN, add the following above the COMMIT line:

-A INPUT -p tcp -m tcp --dport 32976 -j ACCEPT

5. Add rules above the COMMIT line for each IP address you wish to WhiteList, e.g.

-A INPUT -s 8.8.8.8 -j ACCEPT

6. Save the file.

7. Edit /etc/iptables/rules.v6 to look like this:

*filter
:INPUT DROP [0:0]
:FORWARD ACCEPT [0:0]
:OUTPUT ACCEPT [0:0]
-A INPUT -s ::1 -j ACCEPT
COMMIT

8. Restart IPtables and Fail2Ban:

/etc/init.d/netfilter-persistent restart
/etc/init.d/fail2ban restart
iptables -nL
ip6tables -nL

9. If your server is on the public Internet and you’d like to add SSL security, which is required for WebRTC deployments, we’re adding a separate tutorial below as part of the WebRTC implementation to show you the easy way to do this. Keep reading.

Finally, a cautionary note. If you leave your SIP ports exposed to the Internet, then you’ll need to regularly monitor your FreeSWITCH log for attempted attacks. You can download the Incredible Utilities scripts including update-blacklist that we run regularly as a cron job to blacklist all of the most recent bad guys. Please note that IP addresses detected as "bad guys" with this script take precedence over whitelist entries you may have made in step #5 above so be sure to also add the IP addresses from step #5 to this script’s WHITELIST table before running the script, or you may inadvertently lock yourself out of your own server.

cd /
wget http://incrediblepbx.com/freeswitch/incredible-utils-FS.tar.gz
tar zxvf incredible-utils-FS.tar.gz
rm -f incredible-utils-FS.tar.gz

Getting Started with FusionPBX

Using the account credentials displayed after your installation completed, login to FusionPBX using your favorite browser. Don’t forget to include the IP address in the admin field:


Before you do anything else, navigate to Advanced -> Access Controls. Here you will want to whitelist all of the IP addresses of SIP service providers and other PBXs to which you want to interconnect. Simply add Allow entries in the Domains category for each IP address/CIDR entry. HINT: Single IP addresses have a CIDR entry of /32. WARNING: We don’t recommend using FQDN/Domain entries. Despite legitimate FQDNs, all of our entry attempts resulted in "cannot locate" alerts in the FreeSWITCH CLI (fs_cli). This means that future connection attempts from those providers would fail without any indication of what caused the failures. Also, do NOT add entries for IP addresses of phones/softphones that will register to extensions or calls to and from those extensions will fail. This is anything but intuitive but, trust us, you will save hours of hair-pulling.

Creating Extensions in FusionPBX

While you’re still logged into the FusionPBX GUI, let’s add an extension to demonstrate how easy it is. Choose Accounts -> Extensions and click on the + symbol to add a new extension. Here is a sample to get you started, but you really only need the extension number and voicemail PIN entries:



Unlike in FreePBX, the default extension password is not displayed on the template. Once you SAVE the extension, you then have to edit it and click on the Password field to display the default entry. This can be changed, if desired.

Configuring a Softphone for FusionPBX

You can connect virtually any kind of telephone to your new PBX, and FusionPBX includes terrific provisioning tools for dozens of SIP phones. We’ll start with a free SIP softphone today so you can begin making calls. You can find dozens of recommendations for hardware-based SIP phones both on Nerd Vittles and the PIAF Forum when you’re ready to get serious about VoIP telephony.

We recommend YateClient which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for the extension you created above. You’ll need the IP address of your server plus your extension’s password. Fill in the Yate Client template using the IP address of your Server, the extension number for your Username, and whatever Password you assigned to the extension when you created it. Click OK to save your entries.

Once the Yate softphone shows that it has registered with FusionPBX, try a test call by dialing *9664 which should begin playing the default Music on Hold.

Creating Trunks/Providers/Gateways in FusionPBX

In FusionPBX and FreeSWITCH, what FreePBX and Asterisk users call Trunks are referred to as Providers or Gateways. These are commercial outfits that offer to interconnect your PBX with the rest of the telephones in the world using a SIP connection. The first step is to register with the providers of your choice and obtain your SIP credentials and the FQDN(s) of the provider’s servers to which you should register. Most allow authentication by either username/password or by IP address. If you have a static IP address for your server, that is the safer method since you don’t have to worry about someone guessing your password. The only difference in the setup is the Register field should be changed to False.

As luck would have it, there is thorough documentation on the FreeSWITCH site to configure literally hundreds of Providers. Here’s the link.

Before you actually set up your new Provider in FusionPBX, we first need to add the provider’s server to FusionPBX’s Access Control List (ACL). As noted, we’ve encountered problems attempting to register FQDNs in the ACL so we strongly recommend you ping the FQDN of your provider’s server and obtain its actual IP address. Once you have it, navigate to Advanced -> Access Controls -> domains. Click on the Pencil icon to edit the ACL list for domains. Next, click on the + icon at the top of the Nodes listing. Change Type to allow. Enter the IP address of your provider’s server in CIDR. Leave the Domain field blank. Enter your Provider in the Description field. Click the SAVE button.

Now we’re ready to add your Provider. Navigate to Accounts -> Gateways and click the + symbol to add a new one. Click on the Advanced button to expose all of the available fields. Now find your provider in the FreeSWITCH listing and copy the sample entries using your own credentials to the appropriate fields in the FusionPBX template. SAVE your settings when you’re finished. If you chose username/password authentication with Register=True, then your new gateway’s Status should display as Running with a State of REGED.

If you want to take advantage of free calling in the U.S. and Canada using Google Voice, then you’ll need to interconnect your FusionPBX server with an Incredible PBX GVSIP gateway as described in this Nerd Vittles article. On the FusionPBX side, the first step is to add the IP address of the Incredible PBX GVSIP gateway to the ACL (as described above). Next, assuming you followed the tutorial and created a trunk on the Incredible PBX server named FusionPBX in step #2, here’s what the corresponding Gateway should look like on the FusionPBX side:

Gateway: GVSIP-Host
Username: FusionPBX
Password: same as on GVSIP-Host
From User: FusionPBX
From Domain: FusionPBX
Realm: IP address of GVSIP-Host
Expire Seconds: 90
Register: True
Retry Seconds: 30
Auth Username (in Advanced): FusionPBX
Domain: default setting
Context: Public
Profile: external
Description: GVSIP-Host

Be advised that you still need to WhiteList the IP addresses of the two servers on the corresponding sites using IPtables. And you need to whitelist the public IP addresses even if you choose to register the trunk using private VPN addresses. The reason is because FreeSwitch uses the public IP addresses internally, and the registration will fail without the whitelist entries.

Creating Inbound Routes in FusionPBX

As with all PBXs, Inbound Routes define how incoming calls from Trunks/Gateways are routed to destinations on your PBX. Creating inbound routes in FusionPBX (Dialplan -> Inbound Routes) is much the same as the process with FreePBX except the search Conditions are considerably broader than merely a DID or CallerID match and may include Time Conditions to accommodate after-hours calling:



As with FreePBX, the Action can be any destination available on your PBX including an extension, voicemail, company directory, or an IVR:



Typically, inbound calls should be routed to the public Context. And, unlike FreePBX where the first matching inbound route wins, with FusionPBX, you can prioritize the routes numerically to assign a certain search Order.



Creating Outbound Routes in FusionPBX

Outbound Routes tell your PBX how to route calls to destinations outside your PBX using Trunks/Gateways available on your PBX. Creating outbound routes in FusionPBX (Dialplan -> Outbound Routes) is equally flexible offering virtually limitless combinations to assist PBX designers in setting up scenarios for processing outbound calls. As with inbound routes, outbound routes can be prioritized by assigning an Order. And each outbound route can include a primary Gateway as well as up to two Alternates for routing the calls.



Unlike FreePBX which used NXXNXXXXXX and similar combinations as Dialplan Expressions, FusionPBX uses more powerful RegEx coding with many predefined options:



Choosing Providers for FusionPBX

Here’s a shameless plug for our Platinum Sponsor, Vitelity, if you’re looking for an incredible deal on a DID with unlimited inbound calling. You’ll find the offer at the end of this article. If dirt-cheap outbound calls are of interest and Google Voice isn’t an option where you’re calling from or to, then you can’t beat Anveo Direct. The AnveoDirect provider setup for FusionPBX isn’t included in the link we posted above, but it couldn’t be simpler.



To make outbound calls with Anveo Direct, you dial a number with the country code preceded by a special 6-character code starting with 0 which you create on the Anveo Direct web site. You also must whitelist the IP address of your PBX as part of the setup on the Anveo side. Once configured, a call to a number in the U.S. would look like this: 04He9x18005551212@sbc.anveo.com. When creating the Outbound Route for 10-digit dialing using the tutorial above, the AnveoDirect setup would define the Dialplan Expression as 10-digit dialing with a Prefix of 04HE9x1 assuming your 6-character secret code was 04He9x. The trailing 1 in the Prefix converts the 10-digit dialed number to 11-digits as required by Anveo’s international dial code requirement. We think you’ll like their pricing:



Using Gmail as SMTP Smarthost with FusionPBX

Be sure to test sending an email to yourself from the command line to be sure Exim is working properly. Here’s how:

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

If you don’t receive the email, be advised that many providers block downstream SMTP mail servers in which case you may want to use your Gmail account as an SMTP Smarthost with FusionPBX. Here’s how. Begin by reconfiguring Exim: dpkg-reconfigure exim4-config

  • Type Mail Server: Mail sent by smarthost using SMTP
  • System Mail Name: Your server’s FQDN (see /etc/hostname)
  • Allowed Senders: accept defaults
  • Other Destinations: accept default
  • Relay Mail: leave blank
  • Outgoing SmartHost: smtp.gmail.com::587 (note the double colons)
  • Hide local name: no
  • Keep DNS-queries minimal: no
  • Delivery method local mail: Maildir format in home directory
  • Split config into small files: no
  • Root and Postmaster recipient: root

After exim4 restarts, add the following entries to the end of /etc/exim4/passwd.client using your Gmail credentials:

gmail-smtp.l.google.com:YOUR-NAME@gmail.com:PASSWORD
*.google.com:YOUR-NAME@gmail.com:PASSWORD
smtp.gmail.com:YOUR-NAME@gmail.com:PASSWORD

Finally, issue the following commands to update exim4 and implement your changes:

update-exim4.conf
/etc/init.d/exim4 restart

Send yourself another test email to verify that everything is working properly. If the mail still doesn’t make it, be sure your provider (HiFormance, for example) is not also blocking port 587. You’ll need to open a ticket with them if this is the case. You can test whether the port is blocked with the following command:

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

Solving NAT and Audio Problems with FusionPBX

If you experience one-way audio, no audio, or calls that won’t disconnect when the called party hangs up, you’ve probably entered NAT Hell. First, make sure that SIP ALG is turned off on your router. If that doesn’t solve it, edit /etc/default/freeswitch from the Linux CLI and remove -nonat. Save the file and then systemctl daemon-reload. Switch to the FusionPBX GUI and navigate to Advanced -> SIP Profiles. Edit BOTH the internal and external profiles. Then modify BOTH the ext_rtp_ip AND ext_sip_ip entries and change them to autonat:XXX.XXX.XXX.XXX replacing XXX.XXX.XXX.XXX with your server’s public IP address. Then SAVE both profiles. Finally, return to the Linux CLI and restart FreeSWITCH: service freeswitch restart.

Congratulations! You now should have a working PBX. We’ll get deeper into the weeds down the road, but today’s tutorial coupled with the HTML FusionPBX Documentation or PDF version should be sufficient to get you started with a functioning PBX. Take some time to explore all of the Applications that are included in FusionPBX. Enjoy!



9/10 EDIT: New additions begin here…

Implementing WebRTC with FusionPBX

The first step in deploying WebRTC is to add SSL security to your server. The easiest way to do this is to take advantage of the free offering from LetsEncrypt. Begin by assigning a fully-qualified domain name (FQDN) to the public IP address of your server. Wait a few minutes for DNS propagation. Then you’re ready to install your LetsEncrypt certificate. Unlike many of the other LetsEncrypt implementations, the FusionPBX folks have made this a walk in the park. While logged into your server as root, issue the following commands:

cd /usr/src/fusionpbx-install.sh
cd debian/resources
./letsencrypt.sh
service freeswitch restart




Once the certificate is installed and you’ve restarted FreeSWITCH, close your browser and then restart it. Go to the FQDN of your server, and the lock should appear signifying that your site is now fully encrypted. Don’t proceed with the WebRTC steps until this is working.

To get a successful WebRTC implementation where you can make and receive phone calls from a browser, you’re going to need to use the Chrome or Firefox browser. We’ve also had success using the latest Safari browser.

For those that despise implementing complex procedures by viewing video tutorials, we offer the following regurgitation of the steps documented by Mark Crane in his ClueCon video below. This isn’t hard, but it is tedious so don’t skip any steps.



 

While you’re still logged into your server as root, let’s put the FusionPBX WebRTC client in place so you’ll have that option as one of several WebRTC clients to use down the road. The advantage of the FusionPBX WeRTC client is that it can handle your login automatically.

cd /usr/src
git clone https://github.com/fusionpbx/fusionpbx-apps
cd fusionpbx-apps
cp -R sipjs/ /var/www/fusionpbx/app/
chown -R www-data:www-data /var/www/fusionpbx/

Now let’s switch back to your browser and login to FusionPBX using your superadmin credentials. A word of caution… To get WebRTC working, your default Domain must be the FQDN of your server, not an IP address. Once you add this domain, you must switch to it and enter new extensions, trunks, and routes to that domain before proceeding. Begin by adding the domain: Advanced -> Domains -> Add (+). Switch to the domain in the upper right column that’s showing your current domain by clicking on it. It doubles as the Domain Selector.

First, let’s tell FreeSwitch to use your secure SSL setup. Navigate to Advanced -> Variables. Go to the SIP Profile: Internal section and change the false setting of internal_ssl_enable to true. Click SAVE. Go to the SIP Profile: External section and change the false setting of external_ssl_enable to true. Click SAVE. Navigate to Status -> SIP Status and click FLUSH CACHE. Switch back to your SSH session as root and restart FreeSWITCH: service freeswitch restart. Back in your browser, return to Status -> SIP Status, click REFRESH, and verify that both the Internal and External interfaces show TLS enabled.

Navigate to Advanced -> SIP Profiles -> Internal and set wss-binding to true. Switch back to your SSH session as root and restart FreeSWITCH: service freeswitch restart. Back in your browser, return to Status -> SIP Status, click FLUSH CACHE and then REFRESH. You now should see an internal entry for Secure Web Sockets (WSS) in your internal SIP Profile. Finally, to do video with WebRTC, navigate to Advanced -> Variables and add H264 to the list of supported codecs in both outbound_codec_prefs and global_codec_prefs: ULAW, ALAW, H264. Click SAVE. Navigate to Status -> SIP Status and click FLUSH CACHE then RESCAN the internal profile. Clicking on sofia status profile internal will let you verify that the H264 codec has been added successfully. That completes the required pieces to support WebRTC with FusionPBX.

To use the FusionPBX WebRTC client that we installed earlier, we first need to update the FusionPBX menus in the browser: Advanced -> Upgrade -> Menu Defaults and EXECUTE.

Now create an extension to use with WebRTC: Accounts -> Extensions -> Add (+). Once you’ve created the new Extension, drop down to the fourth item (Users) and click on the pull-down menu. Choose the Admin user and click the ADD button followed by SAVE. Next, log out and back into FusionPBX to associate the extension with your user account.

We’re now ready to try out the FusionPBX WebRTC client. Navigate to Apps -> SIPjs which will activate the WebRTC client with your extension credentials. In a separate window, you can verify that SIPjs is registered to your extension by navigating to Status -> Registrations. Verify that you can make a call by dialing *9664 for some nice Hold Music.

Adding Free IBM Voice Prompts to FusionPBX

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

One of the first things you’ll need with your new FusionPBX server is voice prompts for IVRs and custom applications. We’ve now added a tutorial which will walk you through setting up a platform to obtain free IBM voice prompts for your server. Here’s the link.

Blocking SIP Access by IP Address

If you’ve implemented SSL security with an FQDN as recommended above, then you’ll reduce the hammering your server takes from the bad guys by blocking those that attempt SIP registrations or calls using the IP address of your server. This, of course, means that all of your SIP registrations must be made using the FQDN of your server, not by IP address. For providers, you MUST whitelist their IP addresses in the ignoreip field of /etc/fail2ban/jail.conf and restart Fail2Ban, or they will be blocked when they attempt to send data by IP address. We’ve included a script in /root which will tell you which IP addresses currently are blocked: sip-attackers-blocked.

cd /
wget http://incrediblepbx.com/freeswitch/block-sip-by-ip.tar.gz
tar zxvf block-sip-by-ip.tar.gz
rm -f block-sip-by-ip.tar.gz
service fail2ban restart

Adding Free News/Weather TTS Apps

We’ve rolled out the first three Incredible PBX text-to-speech applications for FusionPBX: Yahoo News Headlines, Weather Reports by ZIP Code, and Worldwide Weather Forecasts. This new Nerd Vittles tutorial will walk you through the simple installation steps.

Originally published: Monday, September 3, 2018  Updated: Monday, September 24, 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 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 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…

Beginner’s Navigation Guide to VoIP PBXs and Nerd Vittles

Here at Nerd Vittles, we cover a lot of VoIP territory over the course of a year. To kick off the new year, we thought it might be helpful for those just beginning their VoIP adventure to sketch out the VoIP lay of the land for you. We’re assuming that you came to our site because you wanted a VoIP solution that gives you something to play with and to learn from. That’s not for everybody, and there are less flexible, turnkey VoIP solutions that function pretty much like a toaster. At the top of that short list would be the Ooma Telo and OBi200. Both offer (almost) free calling in the U.S. and Canada.

Keep in mind that all of us started as beginners so there’s no reason to be intimidated if you choose to deploy your own PBX. We’ve gotten a dozen years of enjoyment out of our adventures with VoIP telephony, and there’s no reason you can’t do the same. Let’s begin.

Choosing a Hardware Platform for Your VoIP PBX

First, you’ll need to choose a platform for your VoIP-based PBX: dedicated hardware, virtual machine, or cloud-based PBX. In no small part, this choice depends upon the target audience for your PBX. If it’s for home use or a SOHO business, a $35 Raspberry Pi may suffice. On the other hand, if your PBX will be supporting more than a dozen users or more than a handful of simultaneous calls, we’d look elsewhere. Many of Intel’s Atom-based PCs work very well. And a VirtualBox virtual PBX running atop an iMac or beefy Dell PC can support dozens of users if you have the necessary Internet bandwidth to handle your call volume. Cloud-based servers come in all shapes and sizes as well. As prices have plummeted, cloud solutions have become our favorite. For $3 to $6 a month, you now can host your PBX in the cloud with automatic image backups of your entire server every week. If you’re willing to forfeit backups, here is a cloud solution that will only set you back about a dollar a month. If your server is primarily for business use, we strongly recommend our Platinum Sponsor, RentPBX, that offers dozens of VoIP choices for $14.99 a month with coupon code: NOGOTCHAS.

Choosing the Best PBX to Meet Your Requirements

Once you’ve nailed down your hardware platform, the next step is choosing an operating system and PBX to support your individual requirements. As you might have guessed, there are dozens from which to choose. In both the open source and commercial PBX world, most systems require a specific version of Linux so your operating system choice typically is dependent upon the PBX you choose. In the open source world, the PBX learning curve is often related to the feature set being offered. More sophisticated feature sets typically have a steeper learning curve. If you’re just getting started with VoIP and you want a platform for learning, experimenting, or home use, you can’t beat Incredible PBX 13-13 Whole Enchilada. It was designed by us to be a turnkey PBX for first-time users with rock-solid security and all of the features you will ever need. It includes 31 applications for Asterisk® that cover every imaginable function that can be performed with a telephone including faxing, voice dialing, SMS messaging, wakeup calls and telephone reminders, free calling, conferencing, text-to-speech applications such as News Headlines and Weather Forecasts, Wolfram Alpha for Siri-like queries, plus all the usual PBX features: blacklists, call forwarding, call waiting, call transfer, call parking, call recording, intercom, voicemail including voicemail transcription with email delivery, IVRs, paging, AutoAttendants, DISA, and many more.

If you’re an experienced Asterisk developer that just wants a lean PBX where you can customize it to meet individual customer’s requirements, then Incredible PBX 13-13 Lean should be just the ticket. All of its components are configurable including Asterisk which can be recompiled from the included source code.

At the sophisticated end of the spectrum is Incredible PBX for Wazo which is based upon the Wazo PBX, an Asterisk 15 realtime implementation with full support for High Availability redundancy, multi-party videoconferencing, WebRTC, and automatic nightly backups. It includes API libraries from which you literally can build your own customized PBX from the ground up. The Incredible PBX feature set provides a platform with virtually identical applications to those found in Incredible PBX 13-13.

Sandwiched in between Incredible PBX 13-13 and Incredible PBX for Wazo is Incredible PBX 13 for Issabel. Issabel is an enhanced fork of the previous Elastix 4.0 PBX. The 2018 release includes Asterisk 13, the LTS version of the Asterisk platform. With the new Incredible PBX 13 add-on, you get the best of all worlds with Google Voice support and dozens of applications for Asterisk. Issabel provides a Unified Communications platform that is second to none in the open source world.

Thus far, all of our recommendations have been to open source, GPL-licensed PBX platforms. But you’d be making a mistake to limit your search for business telephony platforms to open source offerings. Our corporate sponsor, 3CX, offers a full year of their commercial PBX running in the Google Cloud at no cost. It’s incredibly simple to install and configure. And the beauty of the 3CX commercial platform is it can scale to any size as your business grows. And the 3CX feature set can be expanded geometrically as your business requirements mature. We added free text-to-speech applications for News and Weather reports just last week. Our favorite open source deployment strategy is to install a 3CX PBX alongside Incredible PBX which yields literally the best of both worlds. The 3CX clients for Windows and Macs, Android, and iOS make VoIP telephony available from anywhere with a couple of button clicks, and 3CX users experience none of the traditional communications problems that invariably crop up on platforms deployed by novice VoIP users running Asterisk.

Getting Started with Extensions, Trunks, and Routes

The Big 3 when it comes to PBX configuration are extensions, trunks, and routes. Extensions carry calls between phones on the PBX and other phones either inside or outside your home or office. Trunks actually provide the links between your PBX and the outside telephony world. Inbound routes tell your PBX where to send incoming calls while Outbound routes tell your PBX which trunk to use when calls are made to numbers outside your PBX. We’ve covered this in more detail including dozens of trunk setups in this Nerd Vittles tutorial.

Making Free U.S./Canada Calls within the United States

There are three ways to make free calls using your PBX. If you’re in the United States, you can use Google Voice to make free calls to the U.S. and Canada if your PBX supports Google Voice trunks, e.g. Incredible PBX 13-13 Whole Enchilada and Incredible PBX for Issabel. An alternative, if your PBX does not directly support Google Voice trunks, e.g. Incredible PBX for Wazo and 3CX, is to use the Simonics SIP to Google Voice Gateway service. For Nerd Vittles users, there is a one-time $4.99 signup fee with no additional charges ever. Whether you live in the U.S or not, all the PBXs we’ve covered today can make free SIP calls to anyone in the world that has a SIP URI address and a SIP phone. Most SIP softphones are free.

Mastering the Incredible PBX Feature Set

Incredible PBX 13-13 Application User’s Guide

Configuring the Travelin’ Man 3 Firewall

All Incredible PBX servers include a firewall that is configured automatically as part of the installation process. On the 3CX platform, you’ll need to add the Travelin’ Man 3 firewall after installing your 3CX PBX. Here’s how:

3CX in the Cloud: 8 Great Ways to Secure Your Server

Configuring a Firewall WhiteList:

WhiteListing Users with Travelin’ Man 3 and IPtables Firewall

Learning to Build Effective IVRs

Interactive Voice Response (IVR) systems and AutoAttendants are the bread-and-butter applications for businesses. If you’ve ever called a business and actually spoken to a live person without encountering an IVR, lucky you! But, believe it or not, IVRs can actually be a useful tool including our Stealth AutoAttendant which lets you intercept incoming calls with a greeting which provides a slight delay to allow the caller (or you) to reroute the call to a specific destination before the default destination kicks in. Nerd Vittles and the Incredible PBX offerings provide all of the tools you’ll need to build any type of IVR imaginable. Mastering Allison Smith’s Top 15 is an excellent starting point.

Harnessing Nerd Vittles Resources

Google is your friend when it comes to finding tutorials of interest in the VoIP world. To narrow searches to just Nerd Vittles, use the following syntax:

stealth autoattendant site:nerdvittles.com

And the Nerd Vittles site itself provides several powerful ways to drill down into topics of interest. In the upper right column of any article, you’ll find a search function which will return a list of matching articles to peruse. At the bottom of every article, check out the all-new Articles of Interest section of Nerd Vittles arranged by topic. Also in the right column of Nerd Vittles, you’ll find a listing of Categories with Nerd Vittles articles conveniently grouped by topic. And, finally, you can quickly jump to the lead article on every major Incredible PBX implementation in the color-coded tabs labeled: GPL VOIP SOLUTIONS FOR ALL.

Happy New Year!

Originally published: Monday, January 1, 2018


Support Issues. With any application as sophisticated as a VoIP PBX, 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 and 3CX gurus and thousands of users just like you. You won’t have to wait long for an answer to your question.



NEW YEAR’S 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 or 3CX? 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…

Introducing IVRs in a Flash with Incredible PBX for Wazo

Last week we introduced the all-new, API-driven Admin User Interface for Wazo 17.07. But pretty pictures are only the icing on the cake. It’s what’s under the hood that really matters. And today we want to walk you through the new IVR User Interface with Incredible PBX for Wazo. The Wazo developers heard from many of you that were missing the GUI-driven IVR tools available in other products. And we’re pleased to tell you it’s another Home Run for Wazo. The UI is well-documented, easy to use, and incredibly simple to implement.

We should mention that our focus today is on the mechanics of using the new Wazo IVR Builder. We won’t be covering how to design a good IVR. If you want some great examples of poor IVR design, look no further than the main number of your favorite cellular or cable provider. Insofar as good design tips go, you can’t do better than learning from the master, Allison Smith. Here’s a link to her previous articles on Nerd Vittles.

We’re assuming you followed last week’s tutorial and added the basic components for the Admin user interface after using your browser to login to https://IPaddress/admin/. If not, start there and install the following plugins to begin: Users, Extensions, Contexts, and Devices. Next, add the plugins for Incalls, Outcalls, Trunks, and IVR. The whole setup procedure takes less than a minute. When you’re finished, your Navigation Sidebar should look like what’s shown above.

You’re obviously going to need a SIP or Google Voice trunk to handle incoming IVR calls so start there if you haven’t already set up at least one trunk that can be dedicated to your IVR. For the time being, add an Inbound Route for the Trunk that points to an extension.

Now we’re ready to create your first IVR using the new IVR template in the Admin UI. Using your web browser, log back in to https://IPaddress/admin/ and (1) click IVR in the Navigation bar. Then (2) click + Add to create a new IVR.

Since the pieces already are in place for the hard-coded Demo IVR that comes with Incredible PBX for Wazo, let’s use those components to demonstrate how easy it is to build your own IVR with the new GUI. As you can see from the completed form below, there are two main sections in setting up an IVR. There are some basic settings that handle answering the call, playing the welcome message, and managing error conditions. Then there are IVR options that correspond to the Digit pressed by the caller using a touchtone phone. Each of these options has a destination. Currently, the available options include playing a sound file, ringing an extension, running custom dialplan code, or hanging up a call. There also is the ability to nest IVRs. Once you have built the secondary IVRs, the options list will also include an IVR option.

Since our sample IVR uses custom contexts for each of the Incredible PBX applications, we’ll rely heavily on the Custom option to route calls to extensions that were created when these applications were installed: Call by Name (411), Join Conference (2663), Lenny (53669), Yahoo News (951), and Weather Forecasts by ZIP Code (947). We’ll also show you how to direct a call to an extension using the default WebRTC extension (701). To get things going, simply copy the entries shown below and then Save/Update the template. The Wazo syntax to call a Custom destination looks like this: Dial(Local/951@default).

Before you attempt to build your own IVR, keep in mind that you’ll need at least a greeting message that tells callers which buttons to press. This file should be placed in /var/lib/xivo/sounds/playback. You can also use any of the default sound recordings found in /usr/share/asterisk/sounds/en for error messages and retry alerts.

Once you have your IVR built, the last step is to adjust an incoming route for one of your DIDs so that it points to your newly created IVR. We do this using the Incalls option in the Navigation bar. If you’re adjusting an existing inbound route, just call it up and make the changes as shown below:

You’re good to go now. Call your DID from another phone and try out your first custom IVR.

Enjoy!

Published: Monday, May 22, 2017  


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



Need help with Asterisk? Visit the 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…

The Stealth AutoAttendant for Incredible PBX and PIAF5

This week we’re dusting off an oldie but goodie, The Stealth AutoAttendant. If you missed our original column 8 years ago, here’s a quick refresher. When a call comes into your PBX, a generic greeting is played: "Thanks for calling. Please hold a moment while we locate someone to take your call." Then the call is transferred to an extension or ring group. Stealth comes into play because this is really an AutoAttendant and, while the greeting is played, a caller can press a preassigned key to transfer the call to some other destination. While it’s obviously not a secure method for providing additional phone features to certain callers, it’s nevertheless helpful in opening up additional PBX functionality without making callers feel like they’re dealing with yet another IVR when they call your home or office. Using the 12-button keypad and clever design, features such as conferencing and DISA can be offered while still providing security through added prompts for passwords or PINs.


https://youtu.be/dB-Tr3WnrKE

In addition to releasing GPL voice prompts from Allison Smith, today we’ll show you how to implement this on three different platforms: Incredible PBX 13 with its FreePBX® GPL modules, Incredible PBX for XiVO®, and PBX in a Flash 5 powered by 3CX®.

Stealth AutoAttendant Voice Prompts

Let’s start by getting you the voice prompts to support the Stealth AutoAttendant. What’s included is a copy of the GPL3 license and versions of the voice prompts in both GSM and WAV format. The prompts are identical except one has a two-second pause at the beginning of the prompt. This is helpful to avoid premature playing of the voice prompt when using trunk providers such as Google Voice that take a couple seconds to set up audio on a call. You can experiment with both and see which best meets your own requirements.

For the techies, it’s probably worth documenting how these prompts were created. We started with GSM versions of both our Generic Greeting prompt and the 2-second silence prompt that can be found on most Asterisk® servers. These were then converted to WAV format: sox 2.gsm -s -b 16 2.wav. Then the WAV files were combined like this:

sox -m 2.wav nv-GenericWelcome.wav nv-GenericWelcome2.wav

Now let’s download the Stealth AutoAttendant prompts to your desktop. The files are available in stealth.zip and stealth.tar.gz format so choose the download that is easiest for you to work with. Once downloaded, unzip or untar the compressed files.

Stealth AutoAttendant with Incredible PBX 13

On the Incredible PBX 13 platform with its FreePBX GPL modules, there are 3 steps to implement the Stealth AutoAttendant. The tricky part in setting up the Stealth AutoAttendant is getting the prompt installed so that it can be used as an Announcement when building an IVR in the GUI. So let’s start there.

1. FreePBX provides a facility for importing existing voice prompts in the GUI. Login as admin to get started. Then go to Admin → System Recordings → Upload and import the generic greeting desired from your desktop. Name the recording and then click Save to add it to your server.

2. Next choose Application → IVR and create a new IVR using the generic greeting as your Announcement. You can follow the IVR Demo template provided with Incredible PBX if you need some hints on how to set up an IVR. Or, better yet, review the Nerd Vittles IVR tutorial.

3. The final step is to point the Inbound Route for one or more of your DIDs to the Stealth AutoAttendant to take incoming calls. Go to Connectivity → Inbound Routes and set the Destination for the DID to the IVR you just created.

Stealth AutoAttendant with Incredible PBX for XiVO

For those using the Incredible PBX for XiVO platform, the voice prompt for the Stealth AutoAttendant already is in place in /usr/local/share/asterisk/sounds. Just copy it to /var/lib/xivo/sounds/playback in order to use it as a voice prompt with your IVRs:

cp -p /usr/local/share/asterisk/sounds/nv-GenericWelcome.wav \\
/var/lib/xivo/sounds/playback/.

Currently, there is no GUI to create IVRs and AutoAttendants with XiVO, but an IVR GUI is in the works so stay tuned. In the meantime, our tutorial and IVR template (ivr-1.conf in /etc/asterisk/extensions_extra.d) will show you how easy it is to create the dialplan code. Just copy it (cp -p) to stealth-aa.conf as a starting point. To use the Stealth AutoAttendant voice prompt, simply change the third line from ivr-Allison and replace it with the new voice prompt: nv-GenericWelcome. It doesn’t get much easier than that.

Finally, using the XiVO GUI, navigate to IPBX → Call Management → Incoming Calls and create a new incoming route for your DID that points to the IVR template you created.

Stealth AutoAttendant with PIAF5 powered by 3CX

On the PIAF5 and 3CX platform, implementing the Stealth AutoAttendant is the easiest of all. It’s all done on a single screen. You get what you pay for. 🙂 Start by logging into the 3CX web portal. When the Dashboard appears, click on Digital Receptionist in the left toolbar. Next click Add. 3CX will automatically assign an extension number for your new IVR, e.g. 800. Assign a new friendly name: Stealth AutoAttendant. Then click the Upload button to upload the WAV prompt from your desktop. Choose the options desired for your Stealth AutoAttendant and designate the DID(s) to use for incoming calls to the Stealth AutoAttendant. In the Destination for no or invalid input, set the Timeout to 2 seconds and choose where to route incoming calls when the caller presses no keys or when the input is invalid. This means the caller has the duration of the greeting plus 2 seconds to press a key and divert the call to another destination. Click OK to save your settings. Told you it’s easy.

Published: Monday, October 31, 2016



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