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The Most Versatile VoIP Provider: FREE PORTING
JUST RELEASED: Visit the Incredible PBX WikiClearlyIP Introduces New Features for Incredible PBX Phones
We’re excited to announce a host of new features for the new Incredible PBX IP phones. For locations where dynamic IP addresses and NAT-based routers pose challenges, the Clearly Devices GUI now offers direct support for OpenVPN. For environments in which security is critically important such as banks and schools, a new Panic button provides unique protection for employees and workplaces. And, for those that have always wanted an "Answered Elsewhere" indication for business communications, your search is over.
Configuring Incredible PBX Phones with OpenVPN
To get started, log into the FreePBX GUI with your admin credentials and navigate to Admin -> Module Admin -> Check Online and update the Clearly Devices module, if necessary. Next, create an OpenVPN server and generate a client template using the MAC address of each of your Incredible PBX phones following the steps in our previous tutorial. Copy the new client templates to the tftpboot folder of your Incredible PBX server. Next, open this ClearlyIP tutorial in a separate window and follow the steps to set up each of your phones. NOTE: When you create a template in Clearly Devices, it will provide a default provision URL at the top of the template that should also be used as the Custom Client Location when you enable the VPN in the User Management template, e.g. http://abc:xyz@192.168.0.3:2580/%%MAC%%.ovpn
Adding a PANIC Button to Incredible PBX Phones
Adding ‘Answered Elsewhere’ As A Call Destination
One of the complaints of many administrators has been the destination entries made in call logs when a call to a ring group is answered on another extension. With the latest release, the "Answered Elsewhere" notation is included in the phone’s firmware. Give these phones a careful look when you are in the market for new SIP phones. There’s nothing quite like them for Asterisk® platforms. Enjoy!
Originally published: Tuesday, May 19, 2020
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.
BulkVS: A Bargain SIP Provider for Incredible PBX Platforms
At every opportunity I always tell new VoIP enthusiasts that one of the true advantages of switching to a VoIP platform is the fact that you don’t have to put all your eggs in one basket. Just this morning, I read a Facebook post from one of the elders in my family lamenting the fact that her MaBell landline had failed in the midst of this week’s snowstorm in North Carolina. Her local WiFi and cable TV still worked but not her landline or cellphone.
With that background, we are pleased to introduce BulkVS trunking as another option to add to your collection. Unlike Skyetel, ClearlyIP, Vitelity, and VoIP.ms, we receive no commissions from BulkVS so chalk this article up as a good example of biting off your nose to spite your face. There is a PayPal link to the right if you’re feeling grateful. 😉
Why does BulkVS matter? In the words of Alex Trebek, it’s The 3 P’s: Price, Price, and Price. An inbound US48 Tier0 phone number (DID) will set you back 6¢ a month with a 25¢ setup fee. And calls are billed at $.0003 per minute. Toll-free numbers in the U.S. and Canada are 14¢ a month with a per minute rate of $.0055. CNAM lookups are $.002. Outbound calls are $0.004/minute. E911 service is 49¢/month. Billing increment: 6 seconds. Those aren’t typos.
Getting Started with BulkVS
To get started, click the sign up link on the main BulkVS page. Then fund your account with $25 using PayPal. Or you can sign up for Net 15 billing and pay by check or credit card if you’re not in a rush to get started.
BulkVS offers two ways to set up your BulkVS trunking: IP-based authentication and SIP registration. If you don’t have a firewall which means you’re not using Incredible PBX, the first method is a little safer because nobody can spoof the IP address of your Asterisk® PBX. But it’s not for everyone. For example, if you’re behind a NAT-based firewall or if your server has a dynamic IP address, then IP-based authentication really isn’t an option. Similarly, if you don’t have control of the router that your PBX is sitting behind, then IP-based authentication won’t work since you have to forward both the SIP port (UDP 5060) and the RTP ports (10000-20000) to your PBX. The beauty of SIP registrations is they work from almost anywhere including double-NAT environments. So today, we’ll cover the SIP registration approach which will work for everyone.
There are three setup procedures: one using the BulkVS Control Panel, a second using the Linux CLI, and a third using the FreePBX® GUI included in Incredible PBX®.
BulkVS Setup with SIP Registration
Step 1: Go to Inbound -> DIDs – Purchase and buy one or more DIDs for your PBX.
Step 2: Go to Interconnection -> Host – Add and add your PBX’s public IP address. Leave the port as 5060 for both chan_sip and chan_pjsip setups.
Step 3: Go to Interconnection -> Trunk Group – Add and create a Trunk Group.
Step 4: Go to Interconnection -> Trunk Group – Manage and add the Primary IP Address for your new Trunk Group. Set Delivery Type to 11DIGITS.
Step 5: Go to Interconnection -> SIP Registration and write down the credentials for one of the SIP credentials you wish to use to register your new trunks.
Step 6: Go to Inbound -> DIDs – Manage and select each telephone number. Then set the Trunk Group to the SIPREG Trunk Group you chose in the previous step. Click Update button.
Step 7: Wait 15 minutes for the new IP and Trunk Group settings to propagate to SBC nodes.
Linux CLI Setup for BulkVS
First, log into your server as root and edit iptables-custom in /usr/local/sbin. Add the following just above the # End of Trusted Provider Section marker:
# BulkVS WhiteList /usr/sbin/iptables -A INPUT -p udp -m udp -s 162.249.171.198 --dport 5060:5069 -j ACCEPT /usr/sbin/iptables -A INPUT -p udp -m udp -s 76.8.29.198 --dport 5060:5069 -j ACCEPT /usr/sbin/iptables -A INPUT -p udp -m udp -s 69.12.88.198 --dport 5060:5069 -j ACCEPT /usr/sbin/iptables -A INPUT -p udp -m udp -s 192.9.236.42 --dport 5060:5069 -j ACCEPT /usr/sbin/iptables -A INPUT -p udp -m udp -s 52.206.134.245 --dport 5060:5069 -j ACCEPT
For chan_sip trunk implementations, while logged into your server as root, edit sip_custom_post.conf in /etc/asterisk. Add the following:
[bulkvs1](bulkvs); host=192.9.236.42 [bulkvs2](bulkvs); host=162.249.171.198 [bulkvs3](bulkvs); host=69.12.88.198 [bulkvs4](bulkvs); host=76.8.29.198 [bulkvs5](bulkvs); host=52.206.134.245
Finally, restart the IPtables firewall and reload Asterisk:
iptables-restart fwconsole reload
FreePBX PJsip Setup with SIP Registration
The PJsip alternative is considerably easier. First, you don’t need sip_custom_post.conf entries at all. To begin, navigate to Connectivity -> Trunks and choose Add a PJsip trunk. Name the trunk BulkVS and then click on the pjsip Settings tab. Fill out the form as shown below substituting the BulkVS registration account name you chose above. Any of the three SIP registrations offered for your account under Interconnection -> SIP Registration in the BulkVS Dashboard will work as long as you use the matching password.
Next, click on the Advanced tab and enter the following in the Match (Permit) field.
162.249.171.198,76.8.29.198,69.12.88.198,192.9.236.42,52.206.134.245
In the Codecs tab, enable ULAW and ALAW. Then click Submit and reload your dialplan.
With PJsip registrations, you may also need to add the following lines to the end of extensions_custom.conf in /etc/asterisk using your actual DID. Then reload your dialplan: asterisk -rx "dialplan reload"
[from-sip-external]
; BulkVS
exten => 18005551212,3,Goto(from-trunk,${DID},1)
FreePBX chan_sip Setup with SIP Registration
If you prefer to set up your BulkVS trunk the old-fashioned way, navigate to Connectivity -> Trunks -> Add chan_sip trunk and enter:
In the Incoming tab, enter a Registration String in the following format where 19991234567 is one of your actual BulkVS DIDs. Then Save the settings and reload the dialplan.
yourBulkVSacctname:yourBulkVSpassword@sip.bulkvs.com/19991234567
Finally, navigate to Settings -> Asterisk SIP Settings and the chan_SIP tab, then set the Registration Minimum Expiry and Registration Default Expiry entries to 25. Then click Submit and reload the dialplan.
FreePBX Inbound & Outbound Route Configuration
Finally, we need to tell FreePBX how to route BulkVS calls into and out of your PBX. In the FreePBX GUI under Connectivty -> Inbound Routes, add a new route for BulkVS specifying the 11-digit DID you purchased from BulkVS. Choose a Destination for the incoming calls, save your settings, and reload the dialplan. Repeat this process for each of your BulkVS DIDs. HINT: The monthly cost of the DIDs is inexpensive enough to assign a DID to every extension on your PBX.
Next, navigate to Connectivity -> Outbound Routes and create a new Outbound Route for calls you wish to process using BulkVS termination services. Name the Outbound Route BulkVS and assign the bulkvs trunk as the first entry in the call sequence. In the Dial Patterns tab, you would want match patterns for 1NXXNXXXXXX and NXXNXXXXXX. For the latter entry, be sure to add a Prepend entry of 1. Then save your settings and reload the dialplan.
SMS Message Delivery from BulkVS Trunks
BulkVS also supports SMS messaging on most of their DIDs. To deliver SMS messages from BulkVS, you’ll need a public-facing web server (not Incredible PBX). Assuming you already have that in place, delivery of SMS messages from BulkVS DIDs to your email address or smartphone’s messaging app is straight-forward. Begin by enabling SMS messaging on your DID: Inbound -> DIDs Manage. Next, assign a web address to process the incoming messages on your web server, e.g. http://yourdomain.com/bulkvs-sms/index.php. Then create the index.php file using the sample code below after inserting your email address for delivery of the incoming messages:
<?php
// Syntax for delivery from bulkvs.com SMS Forwarding Service
$deliverto = "yourname@yourdomain.org";
// $deliverto = "18431234567@txt.att.net";
$from = htmlspecialchars($_REQUEST['from']);
$to = htmlspecialchars($_REQUEST['to']);
$message = htmlspecialchars($_REQUEST['message']);
$subject="SMS Message from $from to $to";
$comment="SMS Message\\n\\nFROM: $from\\n\\nTO: $to\\n\\nMSG: $message\\n\\n";
mail("$deliverto", "$subject", "$comment", "$from");
echo "OK";
?>
To send an SMS message from one of your BulkVS DIDs, you’ll need your API credentials from the BulkVS web site. Simply insert them together with one of your 11-digit DIDs in the script below, and you can send SMS messages to your heart’s content.
from="18005551212"
apikey="aaabbbccc"
apisecret="dddeeefff"
if [ -z "$1" ]; then
echo 'Syntax: send-sms-bulkvs 18005551212 "Your SMS message"'
exit
fi
if [ -z "$2" ]; then
echo 'Syntax: send-sms-bulkvs 18005551212 "Your SMS message"'
exit
fi
to=$1
msg=$2
curl --header "Content-Type: application/json" --request POST --data \\
'{"apikey":"'"$apikey"'","apisecret":"'"$apisecret"'","from":"'"$from"'","to":"'"$to"'","message":"'"$msg"'"}' \\
https://portal.bulkvs.com/sendSMS
To send SMS messages from a Windows machine, see this post from @jerrm.
Originally published: Tuesday, May 12, 2020
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 the Incredible Redialer for Asterisk
If you’ve been a VoIP regular during the past decade then you’re probably already aware that the automatic redial function has disappeared from both Asterisk® and most SIP phones. In these difficult times of trying to connect with critical services and radio contests, we thought it was a good time to introduce the Incredible Redialer. It should perform well on most Asterisk platforms with FreePBX® as well as current releases of Incredible PBX®.
How It Works. The Incredible Redialer assumes you have an Outbound Route to process 10-digit calls. If not, you can adjust the code to meet your local or international calling patterns. For the default install, a caller simply dials 2 plus a 10-digit number, and the Incredible Redialer will repeatedly dial the 10-digit number every five seconds until the call completes without a busy signal. No CDR entries are logged to avoid clutter.
Installation. On FreePBX-based systems including Incredible PBX platforms, edit extensions_custom.conf in /etc/asterisk. Just below the [from-internal-custom] line at the top of the file, insert the following code:
;# // BEGIN Redialer
exten => _2NXXNXXXXXX,1,Answer
exten => _2NXXNXXXXXX,n(dialnow),Wait(5)
exten => _2NXXNXXXXXX,n,Set(NUM2CALL=${EXTEN:1})
exten => _2NXXNXXXXXX,n,Dial(local/${NUM2CALL}@from-internal)
exten => _2NXXNXXXXXX,n,Goto(dialnow)
exten => s-CANCEL,1,Hangup
exten => s-NOANSWER,1,Hangup
exten => s-BUSY,1(redial),Goto(dialnow)
exten => s-CONGESTION,1(congestion),Goto(dialnow)
exten => s-CHANUNAVAIL,1,Hangup
exten => s-,1,Hangup
exten => _s-.,1,Hangup
;# // END Redialer
If your dialplan requires 11-digit numbers beginning with a 1, then edit the five lines beginning with _2 and change the entries to _21. You can make similar changes to support international dialing prefixes. If you’d prefer a dialing prefix other than 2, then replace the 2 in the _2 lines with the prefix of your choice. Save the file and then reload your Asterisk dialplan: asterisk -rx "dialplan reload"
Testing Incredible Redialer. We’ve set up a test number in the United States that always rings busy. Feel free to call it to try things out from your own PBX: 843-606-0555. Enjoy!
Originally published: Monday, May 4, 2020
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 Plug-and-Play Incredible PBX IP Phones
Let’s face it. One of the most tedious tasks in setting up a new PBX is configuring all of the buttons on all of the SIP phones connected to your PBX. Finally, there’s a one-click solution with the new Incredible IP Phones from ClearlyIP. Using the Clearly IP Devices module included in every Incredible PBX 2020 platform, you can use a web browser to point-and-click your way through setting up one or multiple phone configurations which can be pushed to every phone by simply entering its MAC address and extension number. When changes are needed, simply modify the web configuration for the desired phones and the modifications are immediately pushed to the affected devices without ever rebooting any of the affected phones. If you only have a couple extensions attached to your PBX, this may not sound like a big deal; however, if you have hundreds of phones in dozens of locations, you’ve just saved yourself hundreds of hours and thousands of dollars in labor costs.
Unlike the Sangoma "solution" there’s no costly FreePBX module required to auto-provision Incredible PBX phones. It’s an integral component of Incredible PBX 2020.
But don’t take our word for it. Watch Chris Sherwood’s YouTube video above and chuckle to yourself knowing that the first two of the four setup steps are already in place with every new Incredible PBX 2020 install.
Better yet, sign up for one of the (free) Tony Lewis webinars currently scheduled for this Tuesday, January 7, at 2 p.m. Eastern time or Friday, January 10 at 9 a.m. Eastern time. You may remember Tony as the former Chief Operating Officer at Sangoma until he resigned and started the new ClearlyIP organization in which he now serves as the CEO. Come join us!
We’ll hold off the tutorial for a bit to give everyone an opportunity to watch the video and attend one of the webinars on Tuesday. Be sure to sign up to reserve your place. Then check back here soon for the Incredible IP Phones tutorial.
Continue reading: ClearlyIP Introduces New Features for Incredible PBX Phones
Originally published: Sunday, January 5, 2020
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.
The Next Best Thing to (formerly free) Google Voice
Today we want to once again shine the spotlight on LocalPhone, an oft-overlooked VoIP service that’s been around forever. You can call to and be called from any LocalPhone user at no cost. They also offer phone numbers (DIDs) of your choice almost anywhere in the world with free or almost free incoming calls. For those wanting a U.S. DID, the cost is 99¢ a month with a $3 setup fee. That gets you up to 100 free incoming calls a day to your PBX or any SIP phone. Additional calls are a penny per call. There are no limitations on the duration of the calls. If you prefer to forward the calls to your cellphone number in the contiguous U.S., there’s an additional fee of 0.5¢ per minute. But there’s little reason to do that when sending the calls to a SIP softphone on your Android device or iPhone is free. And now the mobile LocalPhone app supports PUSH Notifications. We’ll show you how.
FYI: Nerd Vittles receives a referral credit to keep the lights on when you sign up for service.
Deciphering Your SIP Credentials with LocalPhone
Once you have signed up for a LocalPhone account, the first thing you’ll want to do is make note of your Internet Phone credentials under My Account. These are what we typically refer to as SIP credentials consisting of a SIP ID, SIP password, and SIP server (localphone.com). That’s all you’ll need to configure an incoming LocalPhone trunk on any Incredible PBX® server. And these are the same settings you’d use to configure any SIP phone running on any Android or iOS device. As we noted, you and any other LocalPhone user can call any Internet Phone number worldwide at no cost without limitation. For world travelers, you’ll want to download the LocalPhone app for your smartphone (Android or iOS) and take advantage of their extremely competitive international calling rates.1
Ordering Incoming Numbers (DIDs) from LocalPhone
Begin by funding your account under My Account -> Add Credit. $10 will last you a long time.
The next step is to order one or more incoming phone numbers from LocalPhone.2 If you have friends in far away places that call you frequently, you can purchase DIDs in those locations to eliminate the cost of incoming calls both to them and to you. If you only want a dirt cheap U.S. DID for your home or small office, then LocalPhone is also a perfect fit. Navigate to My Account -> Incoming Numbers and choose the United States as the desired Country. Next, pick the State and City for the desired DID. For free incoming calls, set Call Forwarding and Caller ID for Internet Phone to your assigned Internet Phone SIP ID. You can also elect to forward calls to a SIP URI, if desired. Agree to the terms of use and make your purchase.
Configuring a LocalPhone Trunk with Incredible PBX
We’ve previously covered the LocalPhone trunk setup with Wazo. Most other releases of Incredible PBX include preconfigured LocalPhone trunks for incoming and outgoing calls. Login to the Incredible PBX GUI as admin using your favorite browser and navigate to Connectivity -> Trunks and edit the LocalPhone-In trunk. Set Disable Trunk to NO. Then click the sip-Settings tab. Insert your LocalPhone SIP ID in the username, fromuser, and authuser fields. Insert your LocalPhone SIP Password in the secret field. Change the context field entry to from-trunk. Click on the Incoming tab, and modify the Register String 9999999:yourpassword@localphone.com/9999999 replacing 9999999 with your LocalPhone SIP ID and yourpassword with your LocalPhone SIP Password. Click the Submit button and reload your dialplan when prompted.
Navigate to Connectivity -> Inbound Routes and create a new Inbound Route for LocalPhone using your SIP ID as the DID Number and choosing a desired Destination for incoming calls from your LocalPhone DID. Save your settings and reload the dialplan when prompted.
Configuring a LocalPhone Trunk with VitalPBX
Login to the VitalPBX GUI as admin using your favorite browser and navigate to PBX -> External -> Trunks. Create a new SIP trunk with the following settings replacing 999999 with your LocalPhone SIP ID and 1234 with the LocalPhone SIP Password. Leave the Device for Incoming Calls (User) section blank. Then click SAVE and reload your dialplan.
- Description: LocalPhone
- Codecs: ulaw,alaw
- Local Username: 999999
- Remote Host: localphone.com
- Remote Port: 5060
- Local Secret: 1234
- Insecure: Port,Invite
- Allow Inbound Calls: YES
- Username: [leave blank]
- Host: [leave blank]
- Local Secret: [leave blank]
- Remote Username: 999999
- Remote Secret: 1234
- From User: 999999
- From Domain: localphone.com
- Qualify: YES
- Insecure: [leave blank]
- IP Authentication: NO
- Qualify: [leave default]
- Register String: 999999:1234@localphone.com/999999
Navigate to PBX -> External -> Inbound Routes. Create a new Inbound Route for LocalPhone using your SIP ID as the DID Number and choosing a desired Destination for incoming calls from your LocalPhone DID. Save your settings and reload the dialplan when prompted.
Configuring a LocalPhone Trunk with FreePBX
Login to the FreePBX® GUI as admin using your favorite browser and navigate to Connectivity -> Trunks. Add a new chan_sip trunk named localphone. Then click on the sipSettings tab and enter the following replacing 999999 with your LocalPhone SIP ID and 1234 with the LocalPhone SIP Password.
- username=9999999
- type=friend
- secret=1234
- nat=no
- insecure=port,invite
- host=localphone.com
- fromuser=9999999
- fromdomain=localphone.com
- dtmfmode=rfc2833
- disallow=all
- context=from-trunk
- canreinvite=no
- authuser=9999999
- allow=ulaw&alaw
Next, click on the Incoming tab and enter the following Register String replacing 999999 with your LocalPhone SIP ID and 1234 with the LocalPhone SIP Password:
9999999:1234@localphone.com/9999999
Then click SUBMIT and reload your dialplan.
Navigate to Connectivity -> Inbound Routes and create a new Inbound Route for LocalPhone using your SIP ID as the DID Number and choosing a desired Destination for incoming calls from your LocalPhone DID. Save your settings and reload the dialplan when prompted.
Using Local Numbers for International Calls
LocalPhone has a unique feature that lets you dial a local number from a phone number you have whitelisted in your country and reach almost anyone in the world that you’ve added to your Contacts List. You only pay LocalPhone’s discounted international calling rate for the calls. For example, to call a landline in the U.K. from the U.S. using a LocalPhone-provided U.S. phone number, the calling rate is less than a penny a minute. A call to Cyprus by dialing a U.S. number assigned to your account for your whitelisted phone numbers is 4.5 cents per minute. To get started setting up your whitelisted phone numbers and contacts list, navigate to My Account -> Local Numbers in your LocalPhone account. In your Local Numbers list, first add and verify phone numbers you want to authorize to make calls on your nickel. Next, add the names and phone numbers of international destinations you wish to reach by dialing a local number. LocalPhone will immediately assign a local number for each destination. Simply add these local numbers to the contacts list on your smartphone, and you can call from anywhere in your country at the discounted LocalPhone international calling rates. There are no double-dialing or call menus to navigate. Dialing the assigned local number transparently connects you directly to your destination with no intermediate hurdles.
Using LocalPhone with Other Trunk Providers
So long as your PBX doesn’t have more than two incoming calls to a single DID at the same time, the most economical PBX design is to use LocalPhone DIDs as your published DIDs. This reduces the cost of incoming calls to less than a dollar a month per DID for up to 3,000 incoming calls of unlimited duration. Then use one of our Platinum Sponsors, Skyetel or our soon-to-be-available ClearlyIP SIP trunking service for outbound calls and spoof the outbound CallerID on those other trunks using your LocalPhone DID.
Enjoying the Best of All Worlds with LocalPhone
If you have an iPhone or Android smartphone in addition to a PBX, you can take advantage of LocalPhone’s ability to send incoming calls to multiple destinations. Just make sure your PBX isn’t routing the incoming calls to a destination that is automatically answered, e.g. an IVR. On your Android phone, download the VitalPBX Communicator from the Google Play Store and configure a SIP connection using your LocalPhone SIP credentials. Incoming calls from your LocalPhone DIDs and Internet Phone Number now will be sent to both destinations.
If you have followed one of our previous tutorials that document making SIP URI calls from either a PBX or a SIP client such as LinPhone on your smartphone, then you can take advantage of LocalPhone’s incoming SIP URI feature.3 Just dial 9999999@localphone.com where 9999999 is any LocalPhone SIP ID. You also can add Custom Extensions in Incredible PBX much like the Lenny extension using a Dial string of SIP/9999999@localphone.com to reach worldwide LocalPhone destinations from any PBX extension at no cost. Enjoy!
Originally published: Monday, December 9, 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.- Rates are based on the lowest pay as you go per-minute price to call a landline or a mobile. Skype is a registered trademark of Microsoft Corporation. [↩]
- LocalPhone advises that DID fulfillment can take up to 14 days although our orders always have been completed in less than an hour. [↩]
- LocalPhone offers call filtering for your Internet Phone number using either a blacklist or whitelist in addition to offering the option of blocking anonymous calls. [↩]
Some Tips & Tricks to Supercharge Incredible PBX 16-15
As September comes to a close, we wanted to offer up some simple tips and tricks to get the most out of Incredible PBX® 16-15. Many of these already have been covered on the PIAF Forum so, if you’re not already a subscriber, sign up to keep up with the latest information. Perhaps the most frequently asked question concerns the painfully slow restarts that some folks experience with SendMail and/or Fail2Ban. So let’s start there.
Assigning an FQDN to Your Server
Many system services depend upon identification of the fully-qualified domain name (FQDN) assigned to your server. If you don’t have one, services such as Fail2Ban and SendMail have a difficult time starting and restarting. This includes the iptables-restart script which includes a restart of the Fail2Ban service. With SendMail, you not only will have difficulty restarting the service, but outbound email delivery also will fail since SendMail always checks for a valid sender email address before sending out a message.
If you don’t have a domain that you control, you can always use a free dynamic DNS service such as No-IP which has the added advantage of managing changes in your host’s IP address if you don’t have a static IP address.
Once you have an FQDN for your server, here are the simple steps to assign it to your server. First, edit /etc/hosts, add the FQDN immediately after 127.0.0.1, and then save the file. Next, edit /etc/hostname and replace the default entry with your FQDN. Finally, issue the following command using your actual FQDN: hostname FQDN.
Now you can test restarting Fail2Ban and SendMail with the following commands:
systemctl restart sendmail systemctl restart fail2ban
And you can check the status of the two services with the following commands:
systemctl status sendmail systemctl status fail2ban
Configuring SendMail/Exim with Incredible PBX 16-15
You’re not out of the woods yet if you wish to use SendMail (CentOS) or Exim (Raspbian) to deliver email and voicemail messages. Unless your PBX is in the Cloud with a public IP address on the Internet, be advised that many hosting providers such as Comcast, Spectrum, and AT&T block downstream mail servers from sending email. If you’re using one of these services in your home or office, the solution is to use Gmail or your local ISP as a smart relay host to send mail from your server. Setup instructions for SendMail on the CentOS 7 platform are available here. Setup instructions for Exim on the Raspberry Pi are available here.
Blocking Call Scammers and Robocalls
As election season kicks into high gear, expect robocalls to go through the roof. Not that the diehard scammers care but Congress exempted all politicians from the rules pertaining to robocalls. And then there are those that spoof a phone number similar to yours in order to treat you to the latest car warranty deal or Caribbean vacation. So here’s a way to block 99% of these callers, almost all of whom depend upon autodialers and call center software to distribute calls to live operators once you answer the call.
The design is simple to implement with Incredible PBX. Instead of answering incoming calls with a standard AutoAttendant or IVR, we’ll play a brief announcement followed by a request that the caller "press 7″ or some other number to be connected. When the caller doesn’t press the requested number within a brief number of seconds, Incredible PBX will hangup the call.
Many SIP providers support the early media feature which lets two SIP user agents communicate before a call is actually answered. If your provider supports this and you pay by the minute for inbound call traffic, then we’ll show you how to use the early media feature to block these callers without ever answering the calls and incurring charges. The easiest way to determine whether your SIP provider supports early media is to implement the early media code below and try a test call. If you hear the greeting message after dialing your own number, then early media is supported. If not, use the other dialplan code.
Incredible PBX already includes all the tools you’ll need to implement this. We’ll use a little modified Announcement dialplan code to greet the caller with Allison’s Generic Welcome message: "Thank you for calling. Please hold a moment while we locate someone to take your call." Then we’ll tack on an extra message which says: "To continue in English, press 7." If you’d prefer a different number, you can modify the dialplan code below accordingly.
Add this early media dialplan code to the end of /etc/asterisk/extensions_custom.conf. If you prefer a number other than 7, then replace "7″ in both lines 4 and 6 below:
[hello-caller] exten => s,1,Progress exten => s,n(begin),Noop(Playing announcement Howdy as Early Media) exten => s,n,Set(TIMEOUT(response)=10) exten => s,n,Playback(custom/nv-GenericWelcome&silence/1&continue-in-english&press-7,noanswer) exten => s,n,WaitExten(,) exten => 7,1,Goto(ivr-1,s,1) exten => t,1,Hangup exten => i,1,Hangup exten => fax,1,Noop(Fax detected!) exten => fax,2,Goto(custom-fax-iaxmodem,s,1) ;--== end of [hello-caller] ==--;
In the FreePBX GUI, add a new Custom Destination:
Target: hello-caller,s,1 Description: Scam Blocker
In the FreePBX GUI, modify the Inbound Route for each DID and set the Destination to Custom Destination: Scam Blocker. Save your settings and reload your dialplan.
Now place a test call to your DID and see if you hear the greeting message. If so, you’re done.
If not, edit /etc/asterisk/extensions_custom.conf and replace the [hello-caller] context at the bottom of the file with the following. You can replace "7″ on lines 6 and 8, if desired. Then reload your dialplan: asterisk -rx "dialplan reload". Then place another test call.
[hello-caller]
exten => s,1,GotoIf($["${CHANNEL(state)}" = "Up"]?begin)
exten => s,n,Answer
exten => s,n,Wait(1)
exten => s,n(begin),Noop(Playing announcement Howdy)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n(play),Background(custom/nv-GenericWelcome&silence/1&continue-in-english&press-7,nm)
exten => s,n,WaitExten(,)
exten => 7,1,Goto(ivr-1,s,1)
exten => t,1,Hangup
exten => i,1,Hangup
exten => fax,1,Noop(Fax detected!)
exten => fax,2,Goto(custom-fax-iaxmodem,s,1)
;--== end of [hello-caller] ==--;
Separating Friends from Foes with Fail2Ban
If you’ve ever locked yourself out of your server when Fail2Ban mistakenly believed you were one of the bad guys, welcome to the club. Here’s the simple way to make sure it never happens again. First, deploy a NeoRouter Server and activate the NeoRouter Client on both your PBX and desktop machines. Always login to your PBX using the 10.0.0.x NeoRouter Client IP address of your PBX. Next, edit /etc/fail2ban/jail.conf. Scroll down to the [DEFAULT] section and edit the line which begins with ignoreip. Make certain the line includes the following entries. Then save the file and restart Fail2Ban: systemctl restart fail2ban
ignoreip = 127.0.0.1/8 10.0.0.0/24
You can always check who is currently banned with the command: iptables -nL
And you can unban an IP address by logging in to SSH from a different IP address and using the chain name and banned IP address shown in iptables -nL with the following command:
fail2ban-client set apache-forbidden unbanip xx.xx.xx.xx
Adding Outbound CNAM Data to CDRs
In many implementations, it’s useful in Call Detail Records (CDRs) to be able to associate names (CNAM) with outbound calls just as we do with incoming calls. One of our earliest Asterisk applications, CallerID Superfecta, provides an easy way to do that with just a little tweaking in Incredible PBX 16-15.
1. Open the FreePBX GUI in a browser and go to Admin -> CID Superfecta. There should be one Default setup but it’ll show as disabled. For some quirky reason, you can’t make enabling it stick so click on the third (COPY) option under Actions to create a second setup. Then go down to that one and click the first button (Enable) under Actions. Make future setup changes to CallerID Superfecta by clicking on that setup.
2. Next, log into your server as root and issue the following commands:
cd /root wget http://incrediblepbx.com/dialout-cnam.tar.gz tar zxvf dialout-cnam.tar.gz rm -f dialout-cnam.tar.gz
3. Run the script: /root/install-dialout-cnam.sh. Choose the number of the new CID Superfecta setup (probably will be a negative number which is fine). No idea why.
4. Once the script completes, make a call from extension 701 to an outside number. The new CNAM info should be shown in the ACCOUNT column of your CDR listing.
Originally published: Monday, September 30, 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.Desktop Dream Machine: Incredible PBX 16-15 for VirtualBox
If you’re new to the VoIP world and want to kick the tires to see what you’re missing, then today’s one minute setup is for you. You’ll get a $10 credit to try out some penny-a-minute calls and to purchase a $1 a month phone number in your choice of area codes. If you decide VoIP is not for you, you don’t have to buy anything ever. And you can use almost any desktop computer you already own to bring up the VirtualBox® edition of Incredible PBX® 16-15.
If you’ve followed Nerd Vittles over the years, you already know that VirtualBox from Oracle® is one of our favorite platforms. Once VirtualBox is installed on your desktop computer, adding Incredible PBX is a snap. Download the latest Incredible PBX image from SourceForge, double-click on the downloaded image, check the initialize MAC address box, and boom. In less than a minute, your PBX is ready to use. There are no hidden fees or crippleware to hinder your use of Incredible PBX for as long as you like. Just set up an account with our Platinum provider, Skyetel, and you can start making calls in minutes. Of course, the Incredible PBX feature set is included as well which brings you nearly three dozen applications for Asterisk® that will revolutionize your communications platform. Speech-to-text, voice recognition, and a Siri-like telephony interface are as close as your SIP phone.
Installing Oracle VM VirtualBox
Oracle’s virtual machine platform inherited from Sun is amazing. It’s not only free, but it’s pure GPL2 code. VirtualBox gives you a virtual machine platform that runs on top of any desktop operating system. In terms of limitations, we haven’t found any. We even tested this on an Atom-based Windows 7 machine with 2GB of RAM, and it worked without a hiccup. So step #1 today is to download one or more of the VirtualBox installers from VirtualBox.org or Oracle.com. Our recommendation is to put all of the 100MB installers on a 4GB thumb drive.1 Then you’ll have everything in one place whenever and wherever you happen to need it. Once you’ve downloaded the software, simply install it onto your favorite desktop machine. Accept all of the default settings, and you’ll be good to go. For more details, here’s a link to the Oracle VM VirtualBox User Manual.
Installing the Incredible PBX 16-15 Image
To begin, download the Incredible PBX 16-15 image (3.2 GB) onto your desktop.
Next, double-click on the Incredible PBX .ova image on your desktop. Be sure to check the box to initialize the MAC address of the image and then click Import. Once the import is finished, you’ll see a new Incredible PBX virtual machine in the VM List of the VirtualBox Manager Window. Let’s make a couple of one-time adjustments to the Incredible PBX configuration to account for possible differences in sound and network cards on different host machines.
(1) Click once on the Incredible PBX virtual machine in the VM List. Then (2) click the Settings button. In the Audio tab, check the Enable Audio option and choose your sound card. In the Network tab for Adapter 1, check the Enable Network Adapter option. From the Attached to pull-down menu, choose Bridged Adapter. Then select your network card from the Name list. Then click OK. That’s all the configuration that is necessary for Incredible PBX.
Running Incredible PBX 16-15 in VirtualBox
Once you’ve imported and configured the Incredible PBX Virtual Machine, you’re ready to go. Highlight the Incredible PBX virtual machine in the VM List on the VirtualBox Manager Window and click the Start button. The standard Linux boot procedure will begin and, within a few seconds, you’ll get the familiar Linux login prompt. During the bootstrap procedure, you’ll see a couple of dialogue boxes pop up that explain the keystrokes to move back and forth between your host operating system desktop and your virtual machine. Remember, you still have full access to your desktop computer. Incredible PBX is merely running as a task in a VM window. Always gracefully halt Incredible PBX just as you would on any computer.
Here’s what you need to know. To work in the Incredible PBX virtual machine, just left-click your mouse while it is positioned inside the VM window. To return to your host operating system desktop, press the right Option key on Windows machines or the left Command key on any Mac. For other operating systems, read the dialogue boxes for instructions on moving around. To access the Linux CLI, login as root with the default password: password. Change your root password immediately by typing: passwd. Then update your admin password for web access by typing: ./admin-pw-change. You’ll need the admin password to access the web GUI and manage your PBX. You can update all of your other passwords using the scripts provided in /root.
Setting the Date and Time with VirtualBox
On some platforms, VirtualBox has a nasty habit of mangling the date and time of your virtual machine. Typing date will tell you whether your VM is affected. If it’s a problem, manually set the date and time and then update the hardware clock. Here’s how assuming 08130709 is the month, day, and correct time of your server:
date 08130709 clock -w
Configuring Skyetel for Incredible PBX 16-15
If you’d like to try out the Skyetel service at no charge, here’s the drill. Sign up for Skyetel service to take advantage of the Nerd Vittles specials. First, complete the Prequalification Form here. You then will be provided a link to the Skyetel site to complete your registration. Once you have registered on the Skyetel site and your account has been activated, open a support ticket and request 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. Credit is limited to one per person, company, and address. Effective 10/1/2023, $25/month minimum spend required.
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 16-15:
- Name: MyPBX
- Priority: 1
- IP Address: PBX-Public-IP-Address
- Port: 5060
- 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) or purchasing new ones under the Buy Phone Numbers menu option.
Once you have acquired one or more DIDs, navigate to the Local Numbers or Toll Free Numbers tab and specify the desired SIP Format and Endpoint Group for each DID. Add SMS/MMS and E911 support, if desired. Call Forwarding and Failover are also supported. That completes the VoIP setup on the Skyetel side. System Status is always available here.
Configuring VoIP.ms for Incredible PBX 16-15
To sign up for VoIP.ms service, may we suggest you use our signup link so that Nerd Vittles gets a referral credit for your signup. Once your account is set up, you’ll need to set up a SIP SubAccount and, for Authentication Type, choose Static IP Authentication and enter your Incredible PBX 16-15 server’s public IP address. For Transport, choose UDP. For Device Type, choose Asterisk, IP PBX, Gateway or VoIP Switch. Order a DID in their web panel, and then point the DID to the SubAccount you just created. Be sure to specify atlanta1.voip.ms as the POP from which to receive incoming calls.
Configuring V1VoIP for Incredible PBX 16-15
To sign up for V1VoIP service, sign up on their web site. Then login to your account and order a DID under the DIDs tab. Once the DID has been assigned, choose View DIDs and click on the Forwarding button beside your DID. For Option #1, choose Forward to IP Address/PBX. For the Forwarding Address, enter the public IP address of your server. For the T/O (timeout) value, set it to 2o seconds. Then click the Update button. Under the Termination tab, create a new Endpoint with the public IP address of your server so that you can place outbound calls through V1VoIP.
Configuring Anveo Direct for Incredible PBX 16-15
To sign up for Anveo Direct service, sign up on their web site and then login. After adding funds to your account, purchase a DID under Inbound Service -> Order DID. Next, choose Configure Destination SIP Trunk. Give the Trunk a name. For the Primary SIP URI, enter $[E164]$@server-IP-address. For Call Options, select your new DID from the list. You also must whitelist your public IP address under Outbound Service -> Configure. Create a new Call Termination Trunk and name it to match your server. For Dialing Prefix, choose six alphanumeric characters beginning with a zero. In Authorized IP Addresses, enter the public IP address of your server. Set an appropriate rate cap. We like $0.01 per minute to be safe. Set a concurrent calls limit. We like 2. For the Call Routing Method, choose Least Cost unless you’re feeling extravagant. For Routes/Carriers, choose Standard Routes. Write down your Dialing Prefix and then click the Save button.
Before you can make outbound calls through Anveo Direct from your PBX, you first must configure the Dialing Prefix that you wrote down in the previous step. Log into the GUI as admin using a web browser and edit the Anveo-Out trunk in Connectivity -> Trunks. Click on the custom-Settings tab and replace anveo-pin with your actual Dialing Prefix. Click Submit and Apply Config to complete the setup.
By default, incoming Anveo Direct calls will be processed by the Default inbound route on your PBX. If you wish to redirect incoming Anveo Direct calls using DID-specific inbound routes, then you’ve got a bit more work to do. In addition to creating the inbound route using the 11-digit Anveo Direct DID, enter the following commands after logging into your server as root using SSH/Putty:
cd /etc/asterisk
echo "[from-anveo]" >> extensions_custom.conf
echo "exten => _.,1,Ringing" >> extensions_custom.conf
echo "exten => _.,n,Goto(from-trunk,\${SIP_HEADER(X-anveo-e164)},1)" >> extensions_custom.conf
asterisk -rx "dialplan reload"
Configuring a Softphone for Incredible PBX 16-15
We’re in the home stretch now. You can connect virtually any kind of telephone to your new PBX. Plain Old Phones require an analog telephone adapter (ATA) which can be a separate board in your computer from a company such as Digium. Or it can be a standalone SIP device such as ObiHai’s OBi100 or OBi110 (if you have a phone line from Ma Bell to hook up as well). SIP phones can be connected directly so long as they have an IP address. These could be hardware devices or software devices such as the YateClient softphone. We’ll start with a free one today so you can begin making calls. You can find dozens of recommendations for hardware-based SIP phones both on Nerd Vittles and the PIAF Forum when you’re ready to get serious about VoIP telephony.
We recommend YateClient which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for the 701 extension on Incredible PBX. You can find them by running /root/show-passwords. You’ll need the IP address of your server plus your extension 701 password. In the YateClient, fill in the blanks using the IP address of your Server, 701 for your Username, and whatever Password was assigned to the extension when you installed Incredible PBX. Click OK to save your entries.
Configuring Incredible PBX for VirtualBox
In order to take advantage of all the Incredible PBX applications, you’ll need to obtain IBM text-to-speech (TTS) and speech-to-text (STT) credentials as well as a (free) Application ID for Wolfram Alpha.
This Nerd Vittles tutorial will walk you through getting your IBM account set up and obtaining both your TTS and STT credentials. Be sure to write down BOTH sets of credentials which you’ll need in a minute. For home and SOHO use, IBM access and services are mostly FREE even though you must provide a credit card when signing up. The IBM signup process explains their pricing plans.
To use Wolfram Alpha, sign up for a free Wolfram Alpha API account. Just provide your email address and set up a password. It takes less than a minute. Log into your account and click on Get An App ID. Make up a name for your application and write down (and keep secret) your APP-ID code. That’s all there is to getting set up with Wolfram Alpha. If you want to explore costs for commercial use, there are links to let you get more information.
In addition to your Wolfram Alpha APPID, there are two sets of IBM credentials to plug into the Asterisk AGI scripts. Keep in mind that there are different usernames and passwords for the IBM Watson TTS and STT services. The TTS credentials will look like the following: $IBM_username and $IBM_password. The STT credentials look like this: $API_USERNAME and $API_PASSWORD. Don’t mix them up. 🙂
All of the scripts requiring credentials are located in /var/lib/asterisk/agi-bin so switch to that directory after logging into your server as root. Edit each of the following files and insert your TTS credentials in the variables already provided: nv-today2.php, ibmtts.php, and ibmtts2.php. Edit each of the following files and insert your STT credentials in the variables already provided: getquery.sh, getnumber.sh, and getnumber2.sh. Finally, edit 4747 and insert your Wolfram Alpha APPID.
Using Asteridex with Incredible PBX
AsteriDex is a web-based dialer and address book application for Asterisk and Incredible PBX. It lets you store and manage phone numbers of all your friends and business associates in an easy-to-use SQLite3 database. You simply call up the application with your favorite web browser: http://pbx-ip-address/asteridex4/. When you click on a contact that you wish to call, AsteriDex first calls you at extension 701, and then AsteriDex connects you to your contact through another outbound call made using your default outbound trunk that supports numbers in the 1NXXNXXXXXX format.
Taking Incredible PBX for a Test Drive
You can take Incredible PBX on a test drive by dialing D-E-M-O (3366) from any phone connected to your PBX.
With Allison’s Demo IVR, you can choose from the following options:
- 0. Chat with Operator — connects to extension 701
- 1. AsteriDex Voice Dialer – say "Delta Airlines" or "American Airlines" to connect
- 2. Conferencing – log in using 1234 as the conference PIN
- 3. Wolfram Alpha Almanac – say "What planes are flying overhead"
- 4. Lenny – The Telemarketer’s Worst Nightmare
- 5. Today’s News Headlines — courtesy of Yahoo! News
- 6. Weather by ZIP Code – enter any 5-digit ZIP code for today’s weather
- 7. Today in History — courtesy of OnThisDay.com
- 8. Chat with Nerd Uno — courtesy of SIP URI connection to 3CX iPhone Client
- 9. DISA Voice Dialer — say any 10-digit number to be connected
- *. Current Date and Time — courtesy of Incredible PBX
Originally published: Monday, September 23, 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.- Many of our purchase links refer users to Amazon when we find their prices are competitive for the recommended products. Nerd Vittles receives a small referral fee from Amazon to help cover the costs of our blog. We never recommend particular products solely to generate Amazon commissions. However, when pricing is comparable or availability is favorable, we support Amazon because Amazon supports us. [↩]
Mastering the Incredible PBX 16-15 Feature Set with Raspbian
This week we’ll finish up our introduction of Incredible PBX® 16-15 for the Raspberry Pi with a quick look at some of the additional features that are offered on this new platform and that were not covered in our first and second articles. These include text-to-speech apps for news, weather, and today in history as well as the sample ODBC apps for speed dialing and employee data base lookups. We’ll also walk you through the conferencing setup and document the easiest way to deploy an Interactive Voice Response (IVR) system or a Stealth AutoAttendant with or without Direct Inward System Access (DISA) which provides a way to call into your PBX from anywhere and obtain dial tone to make calls or access features just as if you were using a local phone connected to your PBX. We’ll finish up with a review of the Incredible PBX security model: Travelin’ Man 3 IPtables firewall, Fail2Ban, and PortKnocker.
Introduction to Incredible PBX TTS Apps
Text-to-Speech (TTS) applications are included in Incredible PBX for two reasons. First, they provide useful information by phone. Second, they document the procedure required to build your own TTS applications using Asterisk®. That process typically includes a dialplan code addition to /etc/asterisk/extensions_custom.conf plus a PHP/AGI script which must be stored in /var/lib/asterisk/agi-bin. The actual interaction with the caller is handled in the dialplan code using a TTS engine to convert the text results of the PHP/AGI query into audio suitable for playback over the telephone. The (free) PicoTTS engine is included in the Incredible PBX image for the Raspberry Pi. The commercial IBM TTS engine is also available.
The job of the dialplan code is to answer the incoming call and prompt the caller for any necessary information that needs to be passed to the PHP/AGI script to obtain the information sought by the caller. For example, with the Weather by ZIP code app, the caller dials Z-I-P (947) and the dial plan code prompts the caller for the ZIP code of the desired weather report. This ZIP code is then passed to the nv-weather-zip.php AGI script to retrieve the requested weather forecast. The text results of the query then are passed back to the dialplan code which plays back the results to the caller using the PicoTTS engine.
Three sample TTS applications are included. Dial 951 for the latest Yahoo News headlines. Dial 947 to obtain a weather report for any American city using its ZIP code. Dial T-O-D-A-Y to listen to Today in History events for the current day of the year.
Introduction to Incredible PBX ODBC Apps
As with TTS apps, sample ODBC apps are included in Incredible PBX to provide useful information to callers and to document the procedure required to build your own ODBC applications with Asterisk. For those unfamiliar with ODBC, it is a middleware component that lets you build generic database applications that will work with almost any data base management system. In our case, we are using the MySQL clone, MariDB, as the backend database. But the same ODBC API could be used with a database stored in SQLite, or SQL Server, or PostgreSQL. To interact with your own database, the first step is to install an ODBC connector for your particular database so that it can "talk" to Linux and to Asterisk. On the Linux side, take a look at /etc/odbc.ini for examples of how this is done for individual databases. If you’re using a backend database other than MySQL/MariaDB, then the driver must be installed and added to /etc/odbcinst.ini. On the Asterisk side, there are three pieces that need to be put in place in /etc/asterisk. res_odbc_custom.conf houses the actual linkages to the ODBC databases defined in /etc/odbc.ini. func_odbc.conf houses the actual ODBC queries that will be used to read and write information from and to your databases. Finally, odbc.conf contains the dialplan code that will be used to interact with the caller. It answers the incoming calls, prompts the caller for necessary data to complete the query, executes the query defined in func_odbc.conf, and then converts the text results to audio and passes the results back to the caller using the PicoTTS app.
Two sample ODBC applications are included. Dial 222 to obtain an employee name lookup from the employee timeclock database by entering the employee number, e.g. 12345. Dial 223 for a speed dial application using the AsteriDex dialcode (the first 3 letters of a name). For example, enter D-E-L to obtain phone number of Delta Airlines and optionally place the call.
Introduction to Incredible PBX Conference Bridge
The Incredible PBX platform includes a preconfigured conference application which makes it easy for two or more parties to confer regarding any subject matter of common interest. Those with a local extension on the PBX can join the conference by dialing C-O-N-F (2663). For callers outside the PBX to participate, you would need to add a DID that points to the conference number. We’ve made it easy by including this option in the sample IVR created by Allison Smith. Simply designate the IVR as the destination for a DID and tell users to choose option 2. Local users can call D-E-M-O (3366) and choose option 2.
Before using the conference application, you will want to reset the conference passwords. There’s one for users and a second one for the conference leader. After logging into your server as root, issue the command: ./reset-conference-pins. You can display most of the passwords on your PBX including the conference PINs: ./show-passwords.
The conference bridge setup is configured in the GUI: Applications -> Conferences. Here you can decide whether to require the conference administrator to be present before users can join the conference, you can force termination of the conference when the admin leaves, you can enable the menus for users and administrators by pressing *, you can choose whether to record the conference, you can set the maximum number of conference participants, and much more. Simply click on the ? icons for explanations of the various features. CAUTION: Be advised that saving new settings for the conference bridge will reset the conference PINs to the entries shown or entered into the template, e.g. 1234 and 4321 as shown above!
Configuring Incredible PBX IVRs and AutoAttendants
We’ve included a sample IVR and the Stealth AutoAttendant as part of the Incredible PBX install. The easiest way to master the process of building these is to examine the included samples and try them out: Applications -> IVR -> DemoIVR. The demo IVR comes with all the options preconfigured. Be very careful exposing this through a DID unless you have hardened the passwords, especially for the Telephone Reminders app since this application allows any caller to set up calls to external phone numbers which may cost you money!
The IVR options themselves are self-explanatory and well-documented under the ? icon. The IVR Entries at the bottom of the template define the destinations for caller button presses during a call. The Stealth AutoAttendant is worth examining further since it does not include predefined destinations. You would need to add these yourself. The idea behind a Stealth AutoAttendant is to provide options to a caller which are not explained when the AutoAttendant answers the call. In this way, it allows you to "hide" certain features of your PBX from the average caller. While standing alone, it’s obviously not secure since anyone can press a number on their phone after being connected, it does at least obscure the existence of the options. One good use for this is a DISA option which would let you call into your PBX to obtain dialtone to perform other functions on the PBX with an appropriate password, of course. This is documented in the next section and would need to be set up BEFORE adding the option as a choice on the AutoAttendant.
Configuring DISA with Incredible PBX
Before setting up a DISA option with Incredible PBX, be aware of the risks. Anyone that guesses your DISA password basically gets a blank check to perform any function that could be executed from any phone registered to your PBX. If you’ve decided to proceed anyway, access the GUI and choose Applications -> DISA -> Add DISA. Here’s what a typical DISA setup would look like. You’d obviously want a much more secure PIN!
Once you have saved the template and reloaded your dialplan, you then can add DISA as an option in your IVR or AutoAttendant. Be sure to test it carefully before exposing it for public access. You’ve been warned!
Incredible PBX Security Model Overview
Unlike most other free PBX offerings, Incredible PBX is always deployed as a secure platform. Attempts to access Incredible PBX from outside your local area network will fail unless the IP address has been whitelisted in the IPtables firewall using one of the Travelin’ Man 3 utilities: add-ip or add-fqdn. Repeated attempts to access the PBX will be blocked by Fail2Ban and subsequent attempts to whitelist a blocked IP address will not be successful until the Fail2Ban quarantine expires. Thus, it is important to set up Incredible PBX initially using a desktop PC from which you will subsequently manage the PBX. This assures that at least this desktop PC’s IP address is whitelisted.
To whitelist a static IP address, log into your server as root and issue the following command: ./add-ip my-log-cabin 12.34.56.78 where my-log-cabin is the descriptive name you wish to associate with the whitelisted IP address and 12.34.56.78 is the actual IP address.
Obviously, everyone doesn’t have a static IP address. That’s what the add-fqdn utility is for. It allows you to use a dynamic DNS service to assign an FQDN to a dynamic IP address and rely upon the dynamic IP address provider to keep the FQDN synchronized as the IP address changes. Search your favorite search engine, search for "free dynamic dns raspberry pi" to find available providers. On the Incredible PBX, the setup is much the same except you’ll use the FQDN assigned to the IP address: ./add-fqdn my-log-cabin logcabin.myip.com. Incredible PBX actually runs a script every 10 minutes to keep dynamic IP addresses synchronized. Don’t make any changes to /root/ipchecker. If you’d prefer to have the script run more frequently, adjust the 10 entry in the ipchecker line in /etc/crontab.
Last but not least, Incredible PBX includes the PortKnocker utility which provides an emergency "back door" into your PBX if you ever find yourself locked out by the firewall rules. The idea behind PortKnocker is that you send a packet to three random, pre-defined ports in a particular sequence and, if there’s a match, PortKnocker whitelists your IP address for further access to the server until the firewall is restarted or the server is rebooted. You’ll find your credentials and documentation in /root/knock.FAQ. If your PBX is sitting behind a hardware-based router or firewall, be sure to map the three TCP ports to the LAN IP address of your PBX. Enjoy!
Originally published: Wednesday, August 28, 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.
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