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The Most Versatile VoIP Provider: FREE PORTING
JUST RELEASED: Visit the Incredible PBX WikiIntroducing Cellular Extensions for Incredible PBX & Asterisk
The Holy Grail for many of us in the VoIP community has been a cellphone that functions like a traditional Asterisk® extension on your PBX. The typical use case would be a real estate agent, service technician, salesperson, or other mobile worker who interacts with a home office as part of their daily routine. The beauty of this for the mobile workforce is it allows both the home office receptionist and the mobile worker to not only exchange calls but also to transfer the calls and retrieve voicemail just as if the worker were using a phone in the office.
Some of you may recall that we introduced a service like this called vMobile from Vitelity about six years ago. vMobile had some growing pains not the least of which was total reliance upon Sprint for cellular coverage. The service has been discontinued.
Today we’re testing a new offering from Global Technologies (GTI Global) on the Incredible PBX® platform that addresses some of the shortcomings of the vMobile offering. First, it provides a SIM card that can automatically utilize all four of the major carrier networks in the United States. Second, it offers a dedicated DID, a SIP trunk with free U.S. calling, and an Asterisk extension for every cellphone. And third, it takes advantage of PJsip which permits multiple phones to be connected to and ring on the same Asterisk extension.
GTI Global provided promotional consideration to help defray costs of our Asterisk projects.
Prerequisites. System integrators will acquire SIM cards through the Incredible PBX project. Pricing for the recommended unlimited calling, unlimited messaging, and 2GB monthly data plan is $25/month. A NoData plan is available for $20/month, and other higher data plans also are available. The SIM is compatible with any iPhone® or Android phone device. Users must provide their own cellphones which need not be jailbroken. Dual-SIM smartphones allow users to dedicate one line to the GTI Global SIM and another line for personal use. GTI Global currently is exploring support for eSIMs which would facilitate iPhone use.
To use the SIM-based phone, a public-facing Asterisk platform with a dedicated IP address is required, and it must support PJsip extensions. Incredible PBX 2020, Incredible PBX 16-15, and Incredible PBX 13-13 all have been tested. Other FreePBX® 13, 14, and 15 platforms should also suffice. Accompanying each SIM is a dedicated DID assigned to each cellphone. A matching PJsip extension must be created on the Asterisk platform, and the SIP credentials must be provided to GTI Global to make the cellphone connection to the PBX. Additional phones may be connected to the same PJsip extension to support a desktop phone or receptionist. Simply adjust the Max Connects entry when creating the PJsip extension to support the number of phones desired. Each phone on an extension requires a unique port.
How Calling Works. GTI Global has positioned servers in the facilities of every carrier. When a call arrives from a carrier’s tower, the carrier processes 911 calls directly through its network. For all other incoming calls, the carrier verifies the credentials of the SIM. Once verified, the call is passed to the GTI Global server which sends incoming calls to the associated extension on your PBX. The Inbound Route associated with that DID then sends the call to the assigned destination. If the GTI Global server is unavailable, the carrier processes the incoming calls just as it would any other call on its network by sending it directly to the cellphone.
When the cellphone user places a call, it is processed just as if the call had been made from an internal extension on the PBX with the exception of 911 calls which are handled directly by the carrier. After SIM verification, GTI Global passes the outgoing call to the PBX for processing using the Outbound Routing rules of the PBX. Internal extensions and voicemail can be dialed directly. Outbound calls can utilize any trunk associated with the PBX including the free U.S. trunk provided by GTI Global. CallerID is determined by the outbound trunk processing the call.
Configuring a GTI Global Extension. After logging into the FreePBX GUI, create a PJsip extension with the following settings:
Extension: 12345 Display Name: 12345 GTIglobal Secret: yourSIPpassword Advanced -> Max Contacts: 4
Configuring a GTI Global Trunk. Create a chan_SIP Trunk in the FreePBX GUI using your GTIglobal-provided credentials:
Trunk Name: trunk4321 PEER Details: username=trunk4321 type=friend secret=yourTRUNKpassword host=xxxxxx.mvnoserver.com disallow=all allow=ulaw canreinvite=no dtmfmode=rfc2833 port=5062 Register string: trunk4321:yourTRUNKpassword@xxxxxx.mvnoserver.com:5062
IMPORTANT: When you sign up with GTI, you need to give them the IP address of your PBX, the credentials (extension and password) for your GTI PJsip extension, the outbound SIP port for your GTI trunk (UDP 5062), and the inbound PJsip port for your PJsip extension (UDP 5061).
Configuring an Inbound Route for GTI Global Calls. Replacing 16785551212 with your actual 11-digit DID assigned to each cellphone, create Inbound Routes that look like this:
Description: GTIglobal 16785551212 DID Number: 16785551212 Destination: Extension -> 12345 (your extension associated with each SIM)
You need to add the following context at the end of /etc/asterisk/extensions_custom.conf using your DID. Then reload your dialplan: asterisk -rx "dialplan reload"
[from-sip-external]
; GTI Global
exten => 16785551212,3,Goto(from-trunk,${DID},1)
Configuring an Outbound Route for GTI Global Calls. No special outbound routes are required unless you wish to use the GTI Global trunk to process U.S. calls at no cost. In this case, configure an Outbound Route pointing to your GTI Global Trunk. NOTE: Calls using this trunk can only be made from the extensions associated with GTI Global SIMs.
CallerID and Trunking Strategy. In setting up the GTI components in FreePBX, you need to consider how you wish to process outbound calls. If you want all outbound calls including those from the GTI extensions to reflect the company’s main phone number, then you can use your default outbound route making certain that CallerID override is set with the company number in the appropriate trunks. No CallerID should be set in the outbound route.
If you want calls from the GTI extensions to use the GTI trunk with the CallerID number of the individual cellphones, then the GTI trunk needs to be moved to the top of the default route sequence with no CallerID number specified in the GTI trunk or the outbound route. Instead, set the correct 11-digit CallerID numbers in each of the GTI extensions. This will not impact outbound calls from other extensions because the GTI trunk will indicate congestion forcing the calls out through the next trunk specified in the outbound route.
GTI Global Firewall WhiteList. On Incredible PBX servers and other IPtables platforms, add the following whitelist entries to your firewall:
/usr/sbin/iptables -I INPUT -p udp -m udp -s 99.198.122.166 --dport 5060:5069 -j ACCEPT /usr/sbin/iptables -I INPUT -p udp -m udp -s 99.198.110.51 --dport 5060:5069 -j ACCEPT /usr/sbin/iptables -I INPUT -p udp -m udp -s 96.127.174.38 --dport 5060:5069 -j ACCEPT /usr/sbin/iptables -I INPUT -p udp -m udp -s 37.18.129.170 --dport 5060:5069 -j ACCEPT /usr/sbin/iptables -I INPUT -p udp -m udp -s 37.18.129.171 --dport 5060:5069 -j ACCEPT /usr/sbin/iptables -I INPUT -p udp -m udp -s 37.18.129.172 --dport 5060:5069 -j ACCEPT /usr/sbin/iptables -I INPUT -p udp -m udp -s 37.18.129.173 --dport 5060:5069 -j ACCEPT
How to Proceed. If GTI Global cellular extensions are of interest for your customers, kindly contact us at support@incrediblepbx.com, and we’ll hook you up with the folks at GTI Global. There also is an extensive KnowledgeBase for those wanting more information.
Continue Reading: Last Chance to Jump onto Incredible PBX Cellular Bandwagon
Soup-to-Nuts Tutorial: Adding Cellular Extensions to Incredible PBX
Originally published: Monday, June 22, 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.
Interconnecting a Mobile PBX to the Asterisk Mothership
The Holy Grail for a mobile VoIP solution is a simple way to connect back to your primary Asterisk® PBX via Wi-Fi from anywhere in the world to make and receive calls as if you never left. Let’s tick off the potential problems. First, many home-based PBXs are sitting behind NAT-based routers. Second, almost all remote Wi-Fi connections are made through a NAT-based router. Third, chances are the remote hosting platform blocks outgoing email from downstream servers such as a mobile PBX. Fourth, deciphering the IP address of your remote connection can be problematic. Fifth, the chances of experiencing one-way audio or no audio on your VoIP calls is high because of NAT-based routers at both ends of your connection.
Last week we introduced OpenVPN as a solution for those with multiple VoIP sites to interconnect. But there’s a much simpler solution for those that travel regularly and want to avoid the complexity of configuring OpenVPN. Here is a quick thumbnail of the setup we recommend as your mobile companion, and you’ll never have a one-way audio problem again. In terms of hardware, you’ll need a Raspberry Pi 3B+ with its native WiFi support and a Windows or Mac notebook computer for traveling. You’ll also need a NeoRouter VPN server to make this process seamless. If you’ve already set up an OpenVPN server platform, it will work equally well. One advantage of NeoRouter is that clients can be added from the client side without having to create a config file on the VPN server. All you need is a username and password. But the choice of VPN platform is totally a matter of preference. The objective using either OpenVPN or NeoRouter is secure communications to your home base. We don’t want to have to reconfigure either your home PBX or your traveling PBX or your notebook PC based upon changes in your public and private IP addresses.
Today we’ll walk you through the easiest way to set up a (free) NeoRouter server on the Internet. It can be used to connect up to 254 devices on an encrypted private LAN. We’re delighted to have finally found a perfect use for the (free) Google Cloud instance.
Using a RaspberryPi 3B+, build an Incredible PBX 13-13.10 platform by following our previous tutorial. We’ll set this up on your home WiFi network so that you only have to throw the Raspberry Pi and its power supply in your suitcase when you travel. As part of the setup, we’ll download NeoRouter and activate private IP addresses for your notebook computer as well as both of your PBXs (using nrclientcmd). Next, we’ll interconnect the two PBXs using SIP trunks and the NeoRouter private LAN IP addresses. We’ll take advantage of a neat little Raspberry Pi trick by storing a wpa_supplicant.conf template on your PC for the remote WiFi setup even though we don’t yet know anything about the remote LAN. Once we know the SSID and password at the remote destination, we’ll use your notebook computer to edit the template and transfer the file to the /boot folder of your RasPi’s microSD card. When the card then is inserted and the RasPi is booted, it will automatically move the template to the proper /etc/wpa_supplicant folder to successfully activate your WiFi connection. We’ll also load links, a fast text-based browser, just in case you encounter a hotel that requires some sort of acknowledgement or password before establishing your WiFi connection to the Internet.
Setting Up a (free) NeoRouter Server in the Cloud
Because NeoRouter uses a star-based VPN architecture, that means the NeoRouter Server must always be available at the same IP address for all of the NeoRouter Clients (aka Nodes) to talk to. If you already have a cloud-based server that has a static IP address and can handle the traffic cop duties of NeoRouter Server, then that’s an ideal place to install NeoRouter Server. Simply download the Free flavor of NeoRouter Server that matches your existing platform and install it. Add an FQDN for your server’s IP address, and you’re all set. A detailed summary of available management options is included in our previous NeoRouter v2 article.
We devoted a couple weeks to Google Cloud instances last month, and it turned out to be a pretty awful platform for hosting Asterisk. But the free offering looks to be a perfect fit as a hosting platform for NeoRouter Server. You also won’t have to worry about Google going out of business anytime soon. So let us walk you through an abbreviated setup process on the Google Cloud platform. If you’re just getting started with Google Cloud, read our previous article to take advantage of Google’s generous $300 offer to get you started and to generally familiarize yourself with the mechanics of setting up an instance in the Google Cloud.
For NeoRouter Server, navigate to https://console.cloud.google.com. Click the 3-bar image 
in the upper left corner of your Dashboard. This exposes the Navigation Menu. In the COMPUTE section of the Dashboard, click Compute Engine -> VM Instances. Then click CREATE PROJECT and name it. Now click CREATE INSTANCE and Name it nrserver. The instance name becomes the hostname for your virtual machine. If you want to remain in the Free Tier, choose f1-micro instance as the Machine Type and choose a U.S. Region (us-central1, us-east1 or us-west1). For the Boot Disk, choose CentOS 6 and expand the disk storage to at least 20GB (30GB is available with the Free Tier). For the Firewall setting, leave HTTP and HTTPS disabled. Check your entries carefully and then click the Create button.
When your virtual machine instance comes on line, jot down the assigned public IP address. We’ll need it in a minute. Now click on the SSH pull-down tab and choose Open in a Browser Window. Now we need to set a root password and adjust the SSH settings so that you can login from your desktop computer using SSH or Putty:
sudo passwd root su root nano -w /etc/ssh/sshd_config
When the editor opens the SSH config file, add the following entries. Then save the file and restart SSH: service sshd restart
PermitRootLogin yes PasswordAuthentication yes
You now should be able to log in to your instance as root from your desktop computer using SSH or Putty. Test it to be sure: ssh root@server-IP-address
Before we leave the Google Cloud Dashboard, let’s make the assigned public IP address permanent so that it doesn’t get changed down the road. Keep in mind that, if you ever delete your instance, you also need to remove the assigned static IP address so you don’t continue to get billed for it. From Home on the Dashboard, scroll down to the NETWORKING section and choose VPS Network -> External IP Addresses. Change the Type of your existing address to Static and Name it staticip. Next, choose Firewall Rules in the VPS Network section and click CREATE FIREWALL RULE. Fill in the template like the following leaving the other fields with their default entries. Then click CREATE.
- Name: neorouter
- Target Tags: neorouter
- Source IP Range: 0.0.0.0/0
- Protocols/Ports: check tcp: 32976
CAUTION: Before this firewall rule will be activated for your instance, it also must be specified in the Network Tags section for your instance. Shut down your instance and add the neorouter tag by editing your instance. Then restart your instance.
Now we’re ready to install NeoRouter Free v2 Server on your instance. Be sure to choose the Free v2 variety. Log back into your server as root using SSH/Putty and issue these commands:
yum -y update yum -y install nano wget http://download.neorouter.com/Downloads/NRFree/Update_2.3.1.4360/Linux/CentOS/nrserver-2.3.1.4360-free-centos-x86_64.rpm rpm -Uvh nrserver-2.3.1.4360-free-centos-x86_64.rpm /etc/rc.d/init.d/nrserver.sh restart nrserver -setdomain <DOMAINNAME> <DOMAINPASSWORD> nrserver -adduser <USERNAME> <PASSWORD> admin nrserver -enableuser <USERNAME> nrserver -showsettings
Finally, add the following command to /etc/rc.local so that NeoRouter Server gets started whenever your instance is rebooted:
echo "/etc/rc.d/init.d/nrserver.sh start" >> /etc/rc.local
Installing Incredible PBX 13-13.10 on a Raspberry Pi
Configuring NeoRouter Client on Your Computers
On Linux-based (non-GUI) platforms, setting up the NeoRouter Client is done by issuing the command: nrclientcmd. You’ll be prompted for your NeoRouter Server FQDN as well as your username and password credentials. Perform this procedure on both your home PBX and the Raspberry Pi.
To add your Windows or Mac notebook to the NeoRouter VPN, download the appropriate client and run the application which will prompt for your NeoRouter Server FQDN as well as your NeoRouter credentials. Once completed, you should see all three machines in your NeoRouter Free Client Dashboard: your PC as well as your home PBX and Raspberry Pi-based Incredible PBX. Make note of the private VPN addresses (10.0.0.X) of both your home PBX and your Raspberry Pi. These VPN addresses never change, and we’ll need them to interconnect your PBXs and to set up a softphone on your notebook computer.
Admininistrative Tools to Manage NeoRouter
Here are a few helpful commands for monitoring and managing your NeoRouter VPN.
To access your NeoRouter Linux client: nrclientcmd
To restart NeoRouter Linux client: /etc/rc.d/init.d/nrservice.sh restart
To restart NeoRouter Linux server: /etc/rc.d/init.d/nrserver.sh restart
To set domain: nrserver -setdomain YOUR-VPN-NAME domainpassword
For a list of client devices: nrserver -showcomputers
For a list of existing user accounts: nrserver -showusers
For the settings of your NeoRouter VPN: nrserver -showsettings
To add a user account: nrserver -adduser username password user
To add admin account: nrserver -adduser username password admin
For a complete list of commands: nrserver –help
Interconnecting Your Raspberry Pi and Home PBX
To keep things simple, our setup examples below assume the following NeoRouter VPN addresses: Home PBX (10.0.0.1) and Raspberry Pi (10.0.0.2). Using a browser, you’ll need to login to the GUI of your Home PBX and Raspberry Pi and add a Trunk to each PBX. Be sure to use the same secret on BOTH trunk setups. We don’t recommend forwarding incoming calls from your Home PBX to your Raspberry Pi because most folks won’t be sitting in their hotel room all day to answer incoming calls. Instead, add the number of your smartphone to a Ring Group on the Home PBX and don’t forget the # symbol at the end of the number. On the Raspberry Pi side, we are assuming that whenever a call is dialed from a registered softphone with the 9 prefix, the call will be sent to the Home PBX for call processing (without the 9). For example, 98005551212 would send 800-555-1212 to the Home PBX for outbound routing and 9701 would send 701 to the Home PBX for routing to the 701 extension. You can obviously adjust your dialplan to meet your own local requirements.
On the Home PBX, the chan_sip trunk entries should look like this:
Trunk Name: raspi-remote PEER DETAILS host=10.0.0.2 type=friend context=from-internal username=home-pbx fromuser=home-pbx secret=some-password canreinvite=no insecure=port,invite qualify=yes nat=yes
On the Raspberry Pi, the chan_sip trunk entries should look like this:
Trunk Name: home-pbx PEER DETAILS host=10.0.0.1 type=friend context=from-internal username=raspi-remote fromuser=raspi-remote secret=some-password canreinvite=no insecure=port,invite qualify=yes nat=yes
On the Raspberry Pi, add an Outbound Route named Out9-home-pbx pointed to home-pbx Trunk with the following Dial Patterns. For each Dial Pattern, prepend=blank and prefix=9:
dial string: 1NXXNXXXXXX dial string: NXXNXXXXXX dial string: *98X. dial string: XXX dial string: XXXX dial string: XXXXX
Tweaking Your Raspberry Pi for WiFi Mobility
wpa_supplicant.conf config file to the /boot directory on the card once you arrive at your destination and know the SSID and password of the local WiFi network. When the Raspberry Pi is subsequently booted, the operating system will move the config file to the /etc/wpa_supplicant directory so that your WiFi network will come on line. Here’s what a typical wpa_supplicant.conf file should look like using your actual credentials. The last network section handles open WiFi network connections (think: McDonald’s) if you want to enable them:
country=US
update_config=1
network={
ssid="your-SSID"
psk="your-SSID-password"
key_mgmt=WPA-PSK
scan_ssid=1
priority=5
}
network={
key_mgmt=NONE
priority=1
}
The other gotcha is that some public WiFi networks require some type of web login procedure before you can actually access the Internet even though an IP address may have been assigned to your Raspberry Pi. To handle this situation, you’ll need a text-based web browser on the Raspberry Pi that can be accessed through your notebook PC using SSH and your Raspberry Pi’s VPN address. Our favorite is links which can be installed on your Raspberry Pi before you pack up.
apt-get install links -y
Once you arrive at your destination, connect both your notebook PC and Raspberry Pi to the same WiFi network, login to the RasPi with SSH at the VPN address assigned to your RasPi, and run links to start the browser. Press <esc> to access the links menu options. If you can’t access your RasPi at the VPN IP address, try its WiFi-assigned local IP address.
Adding a Softphone to Your Notebook PC
We recommend YateClient which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for an extension on your Raspberry Pi. Then enter the VPN IP address of your server plus your extension’s password. Click OK to save your entries.
If you are a Mac user, another great no-frills softphone is Telephone. Just download and install it from the Mac App Store.
Adding a Softphone to Your Smartphone
Enjoy your pain-free traveling!
Originally published: Monday, April 22, 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.
Back to Basics: Configuring Extensions, Trunks & Routes
With last week’s release of Incredible PBX 13-13 Lean with Asterisk® 13 and FreePBX® 13 GPL modules, it seemed like an opportune time to revisit the initial setup process of an Asterisk-based PBX. Configuring extensions, trunks, and routes are the fundamental steps in successfully interconnecting your PBX to the telecommunications network. So today we’ll walk through the initial setup process in some detail for those that are just getting started. And we think old-timers may find some hidden nuggets in the exercise as well.
Overview of the Initial Asterisk Setup Process
For those new to PBXs, here’s a two paragraph summary of how Voice over IP (VoIP) works. Phones connected to your PBX are registered with Extensions so that they can make and receive calls. When a PBX user picks up a phone and dials a number, an Outbound Route tells the PBX which Trunk to use to place the call based upon established dialing rules. Unless the dialed number is a local extension, a Trunk registered with some service provider accepts the call, and the PBX sends the call to that provider. The provider then routes the call to its destination where the recipient’s phone rings to announce the incoming call. When the recipient picks up the phone, the conversation begins.
Looking at things from the other end, when a caller somewhere in the world wishes to reach you, the caller picks up a telephone and dials a number known as a DID that is assigned to you by a provider with whom you have established service. When the provider receives the call to your DID, it routes the call to your PBX based upon destination information you established with the provider. Your PBX receives the call with information identifying the DID of the call as well as the CallerID name and number of the caller. An Inbound Route on your PBX then determines where to send the call based upon that DID and CallerID information. Typically, a call is routed to an Extension, a group of Extensions known as a Ring Group, or an IVR or AutoAttendant giving the caller choices on routing the call to the desired destination. Once the call is routed to an Extension, the PBX rings the phone registered to that Extension. When you pick up the phone, the conversation begins.
Configuring Asterisk to Support NAT-Based Routing
With a VoIP server, many PBXs and Extensions are housed behind a NAT-based router that is found in most homes and businesses. These routers assign private IP addresses that are not accessible from the Internet. This causes SIP routing headaches because there are actually two legs to every call, one on the private IP address of your server or extension and another on the public Internet with an entirely different IP address. Routers supposedly handle this handoff of the call using Network Address Translation (NAT) and SIP ALG. With Asterisk-based PBXs, we want the PBX itself to handle the NAT chores so it is critically important to do three things when setting up your PBX. First, turn off SIP ALG on every router used by your PBX and every extension connected to your PBX. Second, tell your PBX about your public and private IP address setup. Step #2 is done in the Incredible PBX GUI with a browser. Login as admin and choose Settings:Asterisk SIP Settings. In the NAT Settings section of the form, click Detect Network Settings. Make sure your public and private IP addresses are correctly listed. Then click Submit and reload your dialplan when prompted. Failure to perform BOTH of these steps typically results in calls with one-way audio, i.e. where either you or the called party can’t hear the other party in the conversation. The third rule to remember is to always configure SIP Extensions on your PBX with NAT Mode=YES. This is rarely harmful and failure to configure SIP extensions in this way typically causes one-way audio in calls as well. IAX extensions avoid NAT issues.
Configuring Extensions with Incredible PBX GUI
Extensions are created using the Incredible PBX GUI: Applications:Extensions. Many SIP phones expect extensions to communicate on UDP port 5060. If this is the case with your SIP phone or softphone, then always create Chan_SIP extensions which communicate on UDP 5060. If your SIP phone or softphone provide port flexibility, then you have a choice in the type of SIP extension to create: Chan_SIP or the more versatile PJSIP. Just remember to always configure SIP extensions with NAT Mode=YES in the Advanced tab. If your VoIP phones or softphones support IAX connectivity, you may wish to consider IAX extensions which avoid NAT problems.
When you create a new Extension, a new entry is automatically created in the PBX Internal Directory. If you wish to allow individual users to manage their extensions or use the WebRTC softphone, then you will also have to create a (very) secure password for User Control Panel (UCP) access. Choose Admin:User Management and click on the key icon of the desired extension to assign a password for UCP and WebRTC access.
Configuring SIP Phones with Incredible PBX GUI
SIP phones and softphones typically require three pieces of information: the IP address of your server, the extension number, and the extension password. If you’re using a PJSIP extension, you also will need to change the port to UDP 5061. If your server is behind a NAT-based router, SIP phones also behind the same router need to use the private LAN address rather than the public IP address. If the SIP phones are outside the router protecting the PBX, then use the public IP address and make certain that you also map ports 5060 and 5061 from your router to the private LAN address of your PBX.
The PIAF Forum can provide you with helpful information in choosing high quality SIP phones. Yealink phones are highly recommended with minimal issues. Cisco phones are the most difficult to configure. Insofar as free softphones, we recommend the Zoiper 3 offerings for Windows, Mac, iOS, and Android. Zoiper 5 still is experiencing some growing pains. A key advantage of the Zoiper softphone is it supports IAX extensions which eliminate the NAT issues entirely. On the Mac platform, we also recommend the Telephone app which is available in the App Store. For SRTP communications, use Grandstream Wave.
Configuring Trunks with Incredible PBX GUI
Perhaps the most difficult component to configure in the PBX is the Trunk. Almost every provider has a different way of doing things. We’ve taken some of the torture out of the exercise by providing a script which will configure settings for dozens of providers in seconds. Once installed, all you need to do is edit the desired Trunk (Connectivity:Trunks), change the Disable Trunk entry to No, and insert your credentials in both the PEER Details and Registration string of the SIP Settings Outgoing and Incoming tabs.
To install the Trunk setups on your PBX, log into your server as root and issue the following commands only once:
cd /root wget http://incrediblepbx.com/create-sample-trunks.tar.gz tar zxvf create-sample-trunks.tar.gz rm create-sample-trunks.tar.gz ./create-sample-trunks
Incredible PBX Wholesale Providers Access
Nerd Vittles has negotiated a special offer that gives you instant access to 300+ wholesale carriers around the globe. In lieu of paying the $650 annual fee for the service, a 13% wholesale surcharge is assessed to cover operational costs of TelecomsXchange. In addition, TelecomsXchange has generously offered to contribute a portion of the surcharge to support the Incredible PBX open source project. See this Nerd Vittles tutorial for installation instructions and signup details.
Configuring Google Voice with Incredible PBX GUI
The advantage of Google Voice trunks for those of you in the United States is that all of your calls within the U.S. and Canada are free. You can’t beat the price, and it has worked reliably for many, many years. There are three different ways to set up Google Voice trunks with Incredible PBX. For a one-time fee of $4.99 with this coupon, you can use the Simonics GV/SIP gateway to configure a Google Voice account using OAuth 2 authentication. Then just set up the Simonics SIP trunk on your PBX to point to the Simonics gateway. A second option is to choose the (recommended) OAuth 2 authentication method for Google Voice when you initially install Incredible PBX 13-13. Finally, you can choose plain-text passwords for Google Voice when you set up Incredible PBX. The drawback of this last option is Google has hinted that they may discontinue support of plain-text passwords.
Here are the initial setup steps on the Google side:
1. Set up a dedicated Gmail and Google Voice account to use exclusively for this Google Voice setup on your PBX. Head over to the Google Voice site and register. You’ll need to provide a U.S. phone number to verify your account by either text message or phone call.
2. Once you have verified your account by entering your verification code, you’ll get a welcome message from Mr. Google. Click Continue to Google Voice.
3. Provide an existing U.S. phone number for verification. It can be the same one you used to set up your Google account in step #1.
4. Once your phone number has been verified, choose a DID in the area code of your choice.
Special Note: Google continues to tighten up on obtaining more than one Google Voice number from the same computer or the same IP address. If this is a problem for you, here’s a workaround. From your smartphone, install the Google Voice app from iPhone App Store or Google’s Play Store. Then open the app and login to your new Google account. Choose your new Google Voice number when prompted and provide a cell number with SMS as your callback number for verification. Once the number is verified, log out of Google Voice. Do NOT make any calls. Now head back to your PC’s browser and login to https://voice.google.com. You will be presented with the new Google Voice interface which does not include the Google Chat option. But fear not. At least for now there’s still a way to get there. After you have set up your new phone number and opened the Google Voice interface, click on the 3 vertical dots in the left sidebar (it’s labeled More). When it opens, click Legacy Google Voice in the sidebar. That will return you to the old UI. Now click on the Gear icon (upper right) and choose Settings. Make sure the Google Chat option is selected and disable forwarding calls to whatever default phone number you set up.
5. When your DID has been assigned, click the More icon at the bottom of the left column of the Google Voice desktop. Click Legacy Google Voice. Now click the Settings icon on your legacy Google Voice desktop. Make certain that Forward Calls to Google chat is checked and disable calls to your forwarding number. Click on the Calls tab and select Call Screening:OFF, CallerID (Incoming):Display Caller’s Number, and Global Spam Filtering:checked. The remaining entries should be blank.
6. Google Voice configuration is now complete. Sign out of your Google Voice account.
The Simonics GV-SIP Gateway Solution. Here’s the quick thumbnail of the steps to put all the pieces in place. First, we set up a Google Voice account at Google as documented above. Next, we’ll set up an account at the Simonics site to link our Google Voice account to the Simonics SIP Gateway. Then we’ll plug our Simonics SIP credentials into the preconfigured Simonics trunk on Incredible PBX. Finally, we’ll add Incoming and Outgoing Routes to tell Incredible PBX how to process Google Voice calls.
Now you’re ready to set up an account on the Simonics site. With this Nerd Vittles link, there’s a one-time fee of $4.99.
1. Start by registering your new Google account.
2. After paying the $4.99 registration fee via PayPal, proceed through the setup process to link your Google Voice account and 11-digit Google Voice phone number to the Simonics SIP Gateway.
3. You then will be provided your SIP username and password as well as the gateway address, gvgw.simonics.com, to use in building your SIP trunk on your PBX.
4. If your SIP credentials ever get compromised, regenerate your password by logging back into the Simonics GW site.
Now it’s time to configure your Simonics trunk in Incredible PBX. Start by logging into the web interface as admin with your admin password from above. Click Connectivity:Trunks and choose the Simonics trunk in the PBX Configuration menu. The Simonics trunk template will display:
1. Untick the Disable Trunk check box.
2. In Outbound CallerID, insert your 10-digit Google Voice number.
3. In username, insert GV1 followed by your 10-digit Google Voice number.
4. In secret, insert your Simonics SIP password.
5. In the Registration String, insert GV1 followed by your 10-digit Google Voice number followed by a colon (:)
6. In the Registration String after the colon, insert your Simonics SIP password.
7. In the tail of the Registration String after the slash (/), insert your 10-digit Google Voice number.
8. Click Submit Changes and then Reload the Dialplan when prompted.
Configuring GV Trunk with Motif in the GUI. If you elect to configure your Google Voice trunk natively using the Incredible PBX GUI, you first will need to obtain a Refresh_Token if you elected to use OAuth 2 authentication.
1. Be sure you are still logged into your Google Voice account. If not, log back in at https://voice.google.com.
2. In a separate browser tab, go to the Google OAUTH Playground using your browser while still logged into your Google Voice account.
3. Once logged in to Google OAUTH Playground, click on the Gear icon in upper right corner (as shown below).
3a. Check the box: Use your own OAuth credentials
3b. Enter Incredible PBX OAuth Client ID:
466295438629-prpknsovs0b8gjfcrs0sn04s9hgn8j3d.apps.googleusercontent.com
3c. Enter Incredible PBX OAuth Client secret: 4ewzJaCx275clcT4i4Hfxqo2
3d. Click Close
4. Click Step 1: Select and Authorize APIs (as shown below)
4a. In OAUTH Scope field, enter: https://www.googleapis.com/auth/googletalk
4b. Click Authorize APIs (blue) button.
5. Click Step 2: Exchange authorization code for tokens
5a. Click Exchange authorization code for tokens (blue) button
5b. When the tokens have been generated, Step 2 will close.
6. Reopen Step 2 and copy your Refresh_Token. This is the "password" you will need to enter (together with your Gmail account name and 10-digit GV phone number) when you add your GV trunk in the Incredible PBX GUI. Store this refresh_token in a safe place. Google doesn’t permanently store it!
7. Authorization tokens NEVER expire! If you ever need to remove your authorization tokens, go here and delete Incredible PBX Google Voice OAUTH entry by clicking on it and choosing DELETE option.
Switch back to your Gmail account and click on the Phone icon at the bottom of the window to place one test call. Once you successfully place a call, you can log out of Google Voice and Gmail.
Yes, this is a convoluted process. Setting up a secure computing environment often is. Just follow the steps and don’t skip any. It’s easy once you get the hang of it. And you’ll sleep better.
Now you’re ready to configure your Google Voice account in Incredible PBX. You do it from within the Incredible PBX GUI by choosing Connectivity:Google Voice. Just plug in your Google Voice Username, enter your refresh_token from Step #6 above as your Google Voice Password, enter your 10-digit Google Voice Phone Number, and check the first two boxes: Add Trunk and Add Outbound Routes. Then click Submit and Apply Settings to save your new entries.
If you elected to use plain-text passwords for Google Voice, simply skip obtaining OAuth 2 credentials and substitute your plain-text password for the refresh_token when you create the Google Voice trunk above. If you have trouble getting Google Voice to work using a plain-text password, try this Google Voice Reset Procedure. It usually fixes connectivity problems. If it still doesn’t work, enable Less Secure Apps using this Google tool.
IMPORTANT: Once you’ve entered your credentials, you MUST restart Asterisk from the Linux command line, or Google Voice calls will fail: amportal restart
Configuring Outbound Routes in Incredible PBX GUI
Outbound Routes serve a couple of purposes. First, they assure that calls placed by users of your PBX are routed out through an appropriate trunk to reach their destination in the least costly manner. Second, they serve as a security mechanism by either blocking or restricting certain calls by requiring a PIN to complete the calls. For example, if you only permit 10-digit calls and route all of those calls out through a Google Voice trunk, there is zero risk of running up an exorbitant phone bill because of unauthorized calls unless you’ve deposited a lot of money in your Google Voice account. This raises another important security tip. Never authorize recurring charges on credit cards registered with your VoIP providers and, if possible, place pricing limits on calls with your providers. If a bad guy were to break into your PBX, you don’t want to give the intruder a blank check to make unauthorized calls. And you certainly don’t want to join the $100,000 Phone Bill Club.
To create outbound routes in the Incredible PBX GUI, navigate to Connectivity:Outbound Routes and click Add Outbound Route. In the Route Settings tab, give the Outbound Route a name and choose one or more trunks to use for the outbound calls. In the Dial Patterns tab, specify the dial strings that must be matched to use this Outbound Route. NXXNXXXXXX would require only 10-digit numbers with the first and fourth digits being a number between 2 and 9. Note that Outbound Routes are searched from the top entry to the bottom until there is a match. Make certain that you order your routes correctly and then place test calls watching the Asterisk CLI to make sure the calls are routed as you intended.
Configuring Inbound Routes in Incredible PBX GUI
Inbound Routes, as the name implies, are used to direct incoming calls to a specific destination. That destination could be an extension, a ring group, an IVR or AutoAttendant, or even a conference or DISA extension to place outbound calls (hopefully with a very secure password). Inbound Routes can be identified by DID, CallerID number, or both. To create Inbound Routes, choose Connectivity:Inbound Routes and then click Add Inbound Route. Provide at least a Description for the route, a DID to be matched, and the Destination for the incoming calls that match. If you only want certain callers to be able to reach certain extensions, add a CallerID number to your matching criteria. You can add Call Recording and CallerID CNAM Lookups under the Other tab.
Bug Fix for Incredible Fax Installer
If you installed Incredible PBX 13-13 during the past week, be advised that removal of a GitHub repo will prevent the Incredible Fax installer in /root from completing successfully. Here’s the fix:
sed -i 's|joshnorth|wardmundy|' /root/incrediblefax11.sh
Also note that, with the Incredible PBX 13-13 Lean install, you must manually create a Custom Destination using the GUI. This is the destination to use for receipt of incoming faxes. The settings for the new Custom Destination should look like this:
target -> custom-fax-iaxmodem,s,1 label -> Fax (HylaFax) return -> no
You now have a functioning PBX. Down the road, we’ll tackle some of the more esoteric features of Asterisk so… Stay Tuned!
Published: Monday, October 30, 2017
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? 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 Incredible PBX: Remote Phone Meets the Travelin’ Man
Ever wrestled with one of those thorny problems for weeks only to wake up in the middle of the night with the answer? Thus was born Travelin’ Man, a web- based, one-click Asterisk® application that automatically reconfigures your Asterisk PBX to enable remote SIP phone access from your cellphone, iPad, remote PC, NetBook, or desktop telephone.
News Flash: Be sure to read our latest article introducing Travelin’ Man 3, a completely new security methodology based upon FQDN Whitelists and DDNS. In a nutshell, you get set-it-and-forget-it convenience and rock-solid VoIP security for your Cloud-based PBX or any PBX in a Flash server that’s lacking a hardware-based firewall and you get both transparent connectivity and security for your mobile or remote workforce.
If you’ve read the Incredible PBX series of articles on Nerd Vittles, you already know what a thorny problem remote phone access is if you want to preserve the overall security of your server. Indeed, our recommendation has been to leave SIP access closed on your hardware-based firewall because of the dangers inherent in activating remote SIP access. Now we have a better idea!
Today’s new approach works like this. First, we’ll run a little script that secures all of your extensions with permit entries locking down all these connections to the IP address range within your private network. Then we’ll open the SIP and RTP ports on your hardware and software firewalls and map these ports to your Asterisk server’s private IP address. With this setup, no one can attempt remote SIP logins to your server because Asterisk blocks all SIP extension connection attempts except those originating inside your LAN. To manage external phone connections to your server, the install script creates a new virtual Apache web server on your Incredible PBX using port 83. We’ll enable and map TCP port 83 on your hardware and software firewalls to your server as well. Web access with port 83 is limited to running the Travelin’ Man app to activate external phones.
Now we’re ready to set up access to your server for remote devices. For each extension you wish to enable for remote access, we’ll create a special web directory using an obscure, random file name which will serve as the web link for the Travelin’ Man web app. For example, in the diagram above, directory 184778 manages extension 501, directory 2389957h manages extension 701, and directory 6993h5j manages extension 702. This is accomplished by simply changing the extension number in the index.php script stored in each directory.
When one of these web links is accessed remotely, the PHP script will automatically reconfigure Asterisk to enable access to the designated SIP extension on your server using the remote IP address from which the web page was accessed. And, of course, there’s an additional layer of SIP security as well. You still need your extension credentials to actually log in to your server with a softphone to place and receive calls. The Travelin’ Man installation process takes only a couple minutes, and the remote SIP activation procedure takes just a couple seconds each time you want remote access from a different location. Here’s a quick example of how it actually works.
Let’s assume we want to use the new $3.95 Bria SIP softphone on an iPad to connect as extension 501 on our Incredible PBX back at home. The problem is that the dynamic IP address of your iPad changes at each new site on your itinerary. Some locations have WiFi while others only have 3G connections.
First, we’ll generate an icon to run Travelin’ Man from your iPad desktop. Use the same procedure with an iPhone or iPod Touch, and there’s a similar procedure for Android devices.1 You only have to do this once. Start up Safari on the iPad to access the new port 83 web server at the random web address the installer created to support extension 501. That web address is something like this using your own FQDN2: http://myserver.dyndns.org:83/184778. After establishing the link once, we’ll hit the + button in Safari and choose Add to Home Screen. This creates the TravelMan icon on the iPad. See the screenshot below of our demo iPad setup which used extension 221 instead of 501.
Once configured, it’s just two clicks to enable your remote phone anywhere: click once on the TravelMan icon. When your IP address is confirmed, return to your Home Screen and click the Bria softphone icon to establish a SIP connection back to your server. Behind the scenes, the Travelin’ Man application will generate the required permit entry for your remote IP address mapping it to the designated extension on your server, and then it will reload your SIP settings to make your Asterisk server accessible to the Bria softphone in your hotel room. The entire process takes only a couple seconds.
If your company happens to have a dozen traveling salesmen, then you’d simply assign a dedicated extension to each employee and create secure directory names for each person (e.g. 2389957h and 6993h5j in diagram above) with a copy of the Travelin’ Man app configured for that employee’s extension number. Now your entire mobile workforce has connectivity back to the home office from any location on the globe. And, when an employee leaves the company and another arrives, just create a new name for the old employee’s web directory to preserve the security of your system (e.g. 184778 in our example becomes 78hd773). Keep in mind that each time the Travelin’ Man app is run for any extension, it wipes out any previously authorized IP address entry for that extension. Thus, the security of your Incredible PBX is always preserved.
Prerequisites. Before proceeding with today’s install, you must be running a stock install of Incredible PBX with PBX in a Flash behind a properly-secured, hardware-based firewall3. We recommend the latest version of Asterisk 1.4 because it addresses a SIP vulnerability that might cause you problems if malformed SIP packets are targeted at your server. The current release of PBX in a Flash (1.7.5.5 Silver) is ideal, but any version of PBX in a Flash can be brought current with Asterisk using the update-source and update-fixes tools. Travelin’ Man assumes that you have the Incredible PBX base install of extensions: 501 plus 701-715. You can obviously add more or remove some, but you’ll need to manually adjust sip_custom_post.conf to reflect your actual extension list after the install completes.
The installer has been encrypted for your/our own protection. In source form, the script would allow anyone to defeat the Incredible PBX requirement. Doing so would mean the required IPtables security component would not be in place and properly configured to protect the underlying system from attack. So we’ve opted to play Big Brother to avoid potential security problems for all of us down the road. This article clearly explains all the necessary components if some folks want to roll their own version. We just don’t want the responsibility if something goes horribly wrong. As Forrest Gump would say, "Shit Happens." 🙂 If you don’t believe it, check out the latest security scramble in the trixbox forums.
Installation. Now we’re ready to get started. So log into your Incredible PBX as root and issue the following commands:
cd /root
wget http://incrediblepbx.com/travelinman.tar.gz
tar zxvf travelinman.tar.gz
./travelinman.x
NOTE: If you’re using PIAF2 with CentOS 6.2, you’ll need to use the updated version of Travelin’ Man because of a syntax change in the Apache config file:
cd /root
wget http://incrediblepbx.com/travelinman2.tar.gz
tar zxvf travelinman2.tar.gz
./travelinman2
The first step in the install procedure is to lock down access to all of your extensions to your private LAN subnet. In case you ever want to do this on another server not running the Incredible PBX, here’s a link to our privip.sh shell script that shows how to do it. This should work on most FreePBX-based Asterisk systems.
Once the extensions are locked down, the script will modify your IPtables and Apache configurations to permit web access on port 83. Next, it will adjust your Asterisk setup to support the Travelin’ Man permit scheme. This involves reworking of sip_custom_post.conf so that permit settings for individual extensions can be stored in files named 501.inc, 701.inc, etc. Finally, the installation procedure will set up a single web site to support extension 501 with a randomized directory name for remote access.4 This setup will be stored in /var/www/travelman. To activate support for additional extensions, you would simply copy the subdirectory giving it a new random name: cp -r dir1 dir2. Then edit config.php in the new subdirectory and change the $extension entry.
To complete the install, you must reconfigure your hardware-based firewall and map the following ports to the private IP address of your server:
TCP 83
UDP 5060
UDP 10000-20000
When the installation is completed, it will show you how to access the new web site for extension 501 using either a fully-qualified domain name or a public or private IP address. Now just follow the steps at the beginning of this article to set up your Android or iDevice, and test things out. Enjoy!
Reminders: Be sure to review the comments to this article and the related support forum thread for a week or two for late-breaking enhancements and issues. Also, Incredible PBX comes preconfigured with call forwarding activated for extension 501. Don’t forget to either disable it or set up a real call forwarding number for extension 501 if you want your cellphone to ring. From any extension on your server, just dial *72501 to set up call forwarding. To cancel call forwarding and pass calls directly to the registered 501 softphone, dial *74 and enter 501. Also be aware that the default RingAll ring group (700) configuration on Incredible PBX systems does not include extension 501. So add 501 if you want your remote extension to ring for incoming calls.
The Incredible PBX: Basic Installation Guide
Adding Skype to The Incredible PBX
Adding Incredible Backup… and Restore to The Incredible PBX
Adding Multiple Google Voice Trunks to The Incredible PBX
Adding Remotes, Preserving Security with Incredible PBX
Continue reading Basic Installation Guide, Part II.
Continue reading Basic Installation Guide, Part III.
Continue reading Basic Installation Guide, Part IV.
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! We maintain a thread with the latest Patches and Bug Fixes for Incredible PBX. Please have a look. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of ordinary users just like you. So you won’t have to wait long for an answer to your questions.
Need help with Asterisk? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.
whos.amung.us If you’re wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our new whos.amung.us statistical web site and check out what’s happening. It’s a terrific resource both for us and for you.
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…
- To create a desktop icon for Travelin’ Man on Android devices, navigate to the link with your browser. Then save the link as a Bookmark by clicking the Star icon in your browser then click Add. Return to the Home Screen and, from the screen on which you wish to add the icon, touch and hold your finger on the screen. When the Add to Home Screen menu appears, choose Shortcuts then Bookmarks and select the link you previously saved. As with iDevices, you only have to do this once. [↩]
- FQDN = Fully-qualified domain name [↩]
- We recommend the dLink Router/Firewall. Low Cost: $35 WBR-2310 Best: DGL-4500 [↩]
- If you’d like to download the web site code independently from the Travelin’ Man install procedure, here’s the link. [↩]
VoIP Softphone Shootout for iPhone, iPad, & iPod Touch
We interrupt our Incredible PBX coverage this week to bring you a summer roundup of the best and worst VoIP softphones for use with an iPhone, iPad, or iPod Touch in conjunction with Asterisk®. We’ve tested all of these products with Asterisk sitting behind a NAT-based firewall/router which introduces some additional wrinkles unless your softphone and server are connected through a virtual private network. We’ll leave the VPN discussion for another day. None of these products has native support for the iPad although all will work with any iPad as will any standard iPhone app in either 1X or 2X mode.
The three four products we’ll be evaluating are Acrobits SIP Softphone, the WiFone from Snizmo.com Ltd., the Media5-fone, and CounterPath’s just-released Bria softphone. All support SIP dialing, and the WiFone provides IAX connectivity as well. We were a bit surprised that, despite their reliance on SIP to connect calls, SIP URI support was minimal to non-existent in all but the Bria product. Before diving into the individual products, we should note that, in conjunction with our product evaluations, we received no compensation or discounted/free software from any source. We are a beta site for CounterPath’s next Bria release.
Acrobits Softphone. The Acrobits Softphone requires iPhone OS 3.0 or later and was recently updated on June 3, 2010. The softphone only supports SIP but works with both WiFi and 3G connections which makes it a perfect complement to current generation iPhones as well as the iPad-3G. The softphone also supports push notifications for inbound calls until multitasking is available with iOS 4.0. Multiple SIP accounts can be registered, and the softphone has SIP proxy, VPN, and STUN server support, a must with Asterisk sitting behind most NAT-based routers. G.711, GSM, and iLBC audio codecs are supported in the standard configuration, and we experienced excellent call quality using WiFi with no DTMF issues. As with all of these VoIP phones, 3G call quality was all over the map depending upon the reliability of your nearest cell tower. SIP URI’s can be called by cutting-and-pasting dial strings from entries in the Contacts list email address fields provided the SIP URI destination name is numeric. Quirky but it works. There’s also a speed dial feature for your 12 favorite contacts. Flexible dial strings are supported to smooth the path for international calling. With iOS 3.1, a bluetooth headset can also be used. The application sells for $7.99 in the App Store, and G.729 support can be added for an additional $9.99. G.729 is a must-have if you’ll be using a 3G network for most of your VoIP calls.
While call quality is obviously subjective, the Acrobits Softphone was our personal favorite for daily use. We routinely use it on an iPad to check Asterisk voicemails and to make outbound calls through our home Asterisk server while traveling. Setup is as simple as entering the IP address or FQDN1 of your Asterisk server and an extension number and password to handle the calls. We added a public STUN server entry because of our NAT-based Asterisk setup.
Snizmo’s WiFone. A very close runner-up in voice quality was the WiFone from Snizmo.com Ltd. This softphone has the added advantage of supporting both SIP and IAX2 connections to Asterisk. If security and ease of use matter most to you, then you can’t go wrong with this softphone. IAX2 connections are much less vulnerable to attack from the Internet and are considerably easier to configure because of the elimination of thorny NAT issues. If we had found this softphone first, we probably would have looked no further. As you can see from the screenshot, this softphone supports multiple SIP and IAX connections and is easily set up using the configuration menu. For our European friends, it also supports SMS using a dozen different providers. Echo cancellation and STUN support are available, and G.711 and GSM codecs can be individually configured for SIP and IAX connections. An Outbound Proxy is also available as well as support for international dial strings and prefixes if you need it.
For SIP accounts, simply provide the server address, a username, and password. Authorization name, SIP port, and proxy server settings are optional. For IAX accounts, server address, username, and password are the only required entries. Each account can be toggled ON and OFF to meet your individual requirements. SMS Settings provides a listing of a dozen SMS providers. Simply add your username, password, and a CallerID and SMS just works. The contacts list also synchronizes with your Mac Address Book as well as MobileMe. The call quality of both SIP and IAX connections using WiFi was excellent. 3G support is not yet available. The web-based tutorial is excellent, and the application is available in the App Store for $6.99. An international version also is available.
We could not get the SIP URI functionality to work because the Contacts list phone numbers do not support SIP URI syntax, and there’s no way to manually enter or cut-and-paste a dial string from an email address in the Contacts list. While the polish of the application was not quite up to the Acrobits Softphone, the call quality was uniformly excellent with the SIP URI limitation that we’ve noted.
Media5-fone. Our final softphone in today’s roundup is Media5-fone from Media5 Corporation. It can be downloaded from the App Store for $4.99. While the application is exclusively a SIP phone, it does have preconfigured setups for dozens of providers in the event your requirements extend beyond the Asterisk universe. Unfortunately, there is no STUN support in the current version which makes it unsuitable for use with Asterisk implementations that sit behind NAT-based routers. Multiple SIP connections are supported as are second call, call waiting, and call toggle. In the current version, both SIP over WiFi and 3G are supported using iLBC, G.711, Enhanced G.711, G.722, and iSAC codecs. SIP Info, RFC 2833, and RTP Inband DTMF methods are configurable for each SIP account. Dialing prefixes are flexible and the phone has language support for English, Arabic, French, German, Italian and Spanish which facilitates international use. The phone also includes a nice implementation of visual voicemail; however, the SIP password and voicemail password would have to be the same to function properly with Asterisk. Automatic gain control and echo cancellation also are supported. With the addition of STUN and SIP URI support, Media5-fone would be a worthy competitor.
Update: CounterPath’s Bria. As luck would have it, CounterPath released their new Bria softphone for the iPhone today. It also is iPod Touch and iPad-compatible and supports both WiFi and 3G. The softphone is available at an introductory price of $3.99 in the App Store. It’s the best bargain in the softphone market. G.729 support can be added for an additional $8.99. G.722 wideband support reportedly is coming in August. You may recall CounterPath’s terrific and free X-Lite offerings for Windows, Macs, and Linux. They’ve been one of our favorite developers ever since, and we are actually serving as a beta tester for their next release. As usual, the Bria interface offers what is hands-down the best UI in the business. The voice quality of the calls is impeccable. Our only criticism is that out-of-the-box, Bria doesn’t work for placing outbound calls with Asterisk. Registration of credentials works fine, inbound calling works great, but outbound calls to either an extension, a phone number in the Address Book, or a SIP URI all just hang with no error message or notation in the log. Only after tracing down an obscure link on their web site did we discover the problem. It turns out that one simple change of a single default setting gets things working as they should. To make the change to support Asterisk, click Settings, Advanced Settings, Network Traversal Strategy, User Specified. Then change ICE:ON to ICE:OFF. Click the Advanced button, and then Apply Changes. Aside from this one default configuration glitch, the Bria softphone would be our Editor’s Choice. We highly recommend you make your purchase while the softphone still is available at the introductory price. For an excellent review, see Alec Saunder’s Blog today.
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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…
- FQDN = Fully-Qualified Domain Name [↩]
Free Asterisk Calls to Zillions of Phones with ENUM and Gizmo5’s Backdoor Dialing
It’s been a while since there’s been much to cheer about in the free calls department with Asterisk®. But today, to kick off the new school year, we have lots of good news and some simple tricks to add zillions of free phone numbers to your Asterisk repertoire. In fact, you’ll be able to call almost any non-AT&T cellphone or landline in the United States at no cost. Remember that when you buy your next cellphone! Special thanks to Cliff on the PBX in a Flash Forums for heads up.
Some early readers of Nerd Vittles may remember sipphone.com which morphed into Gizmo5.com. In January of this year, Gizmo5 struck peering deals with a number of telephone providers that already routed their calls over the Internet. And it’s a pretty impressive list that includes more than 10% of the phones and cellphones in the United States according to Gizmo5’s bean counters. There’s Access One, Airadigm, Allegiance, Alltel, Cablevision Lightpath, Cat Communications, Cbeyond, Cellcom, Cellular Properties, Centennial Wireless, Choice One, Cincinnati Bell Wireless, Cinergy Communications, Cingular, CityNet, Cleveland Unlimited, Comcast Digital Voice, Commpartners, Conversent Communications, Cox Communications, CP Telecom, CTC Communications, Dobson Cell, Eureka, Globalcom, Heartland Communications, Illinois Valley, ITC Deltacom, LDMI, McLeod, Metro PCS, Mpower, Nationsline, Nextel, Nextera Communications, Paetec, RCN, Sprint PCS, Talk America, Telnet Worldwide, T-Mobile, US Cellular, Verizon Wireless, and XO. Whew! And the program is constantly being expanded. Toll-free numbers and Gizmo5-to-Gizmo5 calls also are free using Gizmo5. You can check whether your frequently called numbers are free calls by simply entering the phone numbers at this link.
Thus was born what Gizmo5 calls Backdoor Dialing. Just dial 0101 and the 10-digit number of your choice. If it’s free, the call goes through. If not, you get a message that the number is not yet supported and click. The beauty of the program is that your total investment to use the free service with Asterisk is a one-time fee of $10 for a bucket of CallOut minutes to activate your account. Sometimes this takes a day for the credit to appear, particularly if you use PayPal to cover the cost. The good news is you can spend most of the $10 making calls to any phone in the world, many for under 2¢ per minute, using just about any computer on the planet. Just leave a few cents in the pot to keep your free Backdoor Dialing service enabled. From our testing, we’d rate the Gizmo5 call quality as excellent on both the free and the pay-per-minute calls! Complete rate tables are available here.
Gizmo5 provides free softphones for Windows, Macs, and Linux as well as numerous cell phones and mobile devices including Treo, Nokia, and many more (not the iPhone… yet!). All of the softphones make it extremely easy to place SIP calls, e.g. joeschmo@mypbx.dyndns.org. And you can place these calls all day long at no cost. See our tutorial for step-by-step instructions on setting up your own SIP addresses on your Asterisk server. The softphones also include Conferencing, SMS, and Instant Messaging with AIM, Yahoo, MSN, Google, and MySpace.
As with many of these services, they weren’t designed for Asterisk, but nothing in their fine print precludes Asterisk use so today we’ll show you how. Will the program last forever? Who knows, but it’s free for now. And the cost of admission is too good to resist. You’re obviously not going to dial every number you frequently call twice just to see if the call is free. That’s why you’ll want to use a robodialer such as AsteriDex for your outbound calling. Then it’s easy to adjust the phone numbers of your friends with Sprint, T-Mobile, or Verizon cellphones so that you never have to pay for those calls again. Just add a prefix of 0101 to the numbers, and you’re done. And they can call you on your Gizmo5 CallIn number through Asterisk if you’ve enabled the CallIn Service and chosen a number. It’s under $3 a month with an annual subscription. Or the calls can be returned using the CallerID number displayed by Gizmo5 when you call your friends. Toll charges may apply in this case due to the Gizmo5 area code.
So let’s get started. Step 1 is to download and install a free softphone of your choice and follow the prompts to sign up for your account. There’s really no reason not to install a Gizmo5 softphone on every computer you own. If you don’t use it, there’s no cost. If you ever need it, it’ll be there for you. Step 2 is to make a $10 purchase of CallOut minutes. While you’re waiting on the credit to appear (and it usually takes less than a day), let’s set up Asterisk. You’ll need your new account name, password, and phone number from Gizmo5 to get started.
Setting Up a FreePBX Trunk for Gizmo5. If you’re using a product such as PBX in a Flash that includes FreePBX, then open FreePBX in your browser and choose Setup->Trunks->Add SIP Trunk. Leave the General Settings blank. For the Dialing Rules, if you just want free calling through your Gizmo5 trunk, plug in values below. For regular calls as well, add 1NXXNXXXXXX or an entry that is suitable for each country you wish to call.
1800NXXXXXX
1822NXXXXXX
1833NXXXXXX
1844NXXXXXX
1855NXXXXXX
1866NXXXXXX
1877NXXXXXX
1888NXXXXXX
800NXXXXXX
822NXXXXXX
833NXXXXXX
844NXXXXXX
855NXXXXXX
866NXXXXXX
877NXXXXXX
888NXXXXXX
0101+NXXNXXXXXX
0101NXXNXXXXXX
Name the Trunk: Gizmo5. Make the following entries in Outgoing Settings Peer Details:
disallow=all
allow=ulaw
auth=md5
authuser=youracctnameNOTyourphonenumber
canreinvite=no
context=from-trunk
dtmfmode=auto
fromdomain=proxy01.sipphone.com
fromuser=youracctnameNOTyourphonenumber
host=proxy01.sipphone.com
insecure=very
nat=yes
qualify=yes
secret=yourpassword
type=peer
username=youracctnameNOTyourphonenumber
Clear out the Incoming Settings and use the following syntax for the Registration String. Then Save your setup and Reload Your Dialplan. NOTE: Don’t use any registration string unless you want incoming call support. By not registering, you can use your softphones whenever you need it to also make outbound calls. If you register with Gizmo5 using a registration string, then it knocks out use of a softphone since you can’t have two simultaneous registrations to the same account. But registering allows those you call with this service to call you back conveniently… although not necessarily for free from the caller’s phone.
youracctname:yourpassword@proxy01.sipphone.com/yourphonenumber
Setting Up a FreePBX Outbound Route for Gizmo5. While still in FreePBX, choose Setup->Outbound Routes->Add Route. Name the route: OutGizmo5. Then enter the following Dial Pattern: 0101NXXNXXXXXX. Choose SIP/Gizmo5 as your Trunk Sequence. Then click Submit Changes and Reload Your Dialplan.
Setting Up a FreePBX Inbound Route for Gizmo5. While still in FreePBX, choose Setup->Inbound Routes->Add Incoming Route. Name the route: Gizmo5 and plug in your 10-digit DID number in the appropriate field. Then Set a Destination for the incoming calls. That’s it. Save your entries by clicking the Submit button and then Reload Your Dialplan.
Making a Free Call with Gizmo5. Once your DialOut credit appears on your softphone or in your Gizmo5 web account, you’re ready to start making calls. From any phone connected to your Asterisk server, just dial 0101 plus the 10-digit phone number. On the Asterisk CLI, you should see the call routed out through your SIP/Gizmo5 trunk. If you get a congestion tone and you’re sure your DialOut credit has been posted to your account, then check your username and password entries in your Trunk setup. Be sure to use your account name and NOT your Gizmo5 phone number for your username, authuser, and fromuser entries. But, if that doesn’t work, try using your Gizmo5 phone number instead of your assigned user name. Some have reported quirks in which actually works. For us, the assigned user name did the trick. Also make certain that the disallow all entry is above the allow=ulaw in versions of FreePBX after 2.3, or no calls will ever be successful.
Turning Non-Free Numbers into Freebies. There’s always some enterprising individual that figures out a quick way to beat the system even when many calls already are free. Suppose the number you wish to call isn’t yet available through Backdoor Dialing. The only trick is to have a pool of numbers from a provider with a peering arrangement with Gizmo5… and, of course, an Asterisk or FreeSwitch server to forward the calls and handle the number translation. You can read about RingBranch’s implementation, and then you can sign up for the service here.
There’s another way to turn non-free calls into freebies. This is Gizmo5’s "All Calls Free" Plan which is available in 60 countries. Landlines and mobile phones are supported in 17 countries while landlines only are supported in 43 more. U.S., Canadian, and Chinese landlines and cellphones are included in the program in addition to those of the Pope and the other residents of Vatican City. God works in mysterious ways! Here’s the complete list of countries that are supported.
To qualify a landline or mobile number for free calling (by dialing with the usual country code prefixes), you both have to be "active" Gizmo5 subscribers, your landline and mobile numbers must be listed on your account, and you must enter each other in your respective Buddy Lists. Then free calls using your Asterisk Gizmo trunk can be made to the "regular" phone numbers of all your pals whether the called person is online with Gizmo or not. Be aware that you can’t call your own numbers for free, and there is lots of additional "fine print" in this program. Nothing precludes your spouse having his or her own Gizmo5 account, however. You’ll need to wade through the rules carefully to take advantage of the free calling. It is possible, but it’s not easy. If you have relatives in Europe, Australia, or the Far East, you might want to have a look here. Just do a search for "All Calls Free." Your Gizmo5 softphone also will report your current All Calls Free Status.
Add Free Calls to 40 Million Asterisk Servers with e164.org. While we’re on a roll of free calling, here’s a simple way to add free calling to 40 million Asterisk servers around the world. Just add your name and phone numbers to the e164.org registry at no cost and configure FreePBX with ENUM support. Then outbound calls to numbers in the e164 registry will always be free as well. The whole setup takes less than 10 minutes. Here’s how.
The first step in setting up ENUM is to create a SIP address for your Asterisk server. The format looks like this: myname@somedomain.com. You’ll need either a fully-qualified domain name (FQDN) if your server has a static IP address or an FQDN issued through a dynamic DNS service such as dyndns.org if you have a dynamic IP address, e.g. pbx.dyndns.org. In the latter case, your router keeps dyndns.org apprised of changes in your external IP address so that pbx.dyndns.org always resolves to the correct IP address of your Asterisk server. Incidentally, with any hosted domain using a registrar such as omnis.com, it’s easy to add a subdomain DNS entry and point it to your Asterisk server, e.g. sip.joeschmo.com. That won’t cost you a dime other than the annual $6.95 domain registration fee which you’re already paying anyway.
Step two is to add your new FQDN address with a name of your choice to your Asterisk server. Then Asterisk will know how to process incoming SIP calls to that address. Read the Rolling Your Own section of our article on SIP Proxies for the procedure using FreePBX. It only takes a minute or two to set up. Let’s assume for purposes of this tutorial that you’re going to use the following destination address on e164.org for your server: e164@pbx.dyndns.org. An advantage to this type naming scheme is you can always keep straight the source of your incoming SIP calls. Thus your /etc/asterisk/extensions_override_freepbx.conf file should include a line in the [from-sip-external] context that looks like this: exten => e164,1,Goto(from-trunk,e164,1)
This tells Asterisk to route incoming SIP calls to e164@pbx.dyndns.org to the FreePBX Incoming Route for e164. And to complete the routing of the inbound calls to this address, add an Inbound Route in FreePBX called e164 that includes a destination of your choice for these SIP calls, e.g. an extension, a ring group, or an IVR already configured on your system. Just a footnote that e164.org requires you to enter a confirmation PIN when you set up the SIP routing to your server. So, at least initially, make the destination for your e164 SIP calls an extension that you can answer to obtain your PIN. You can safely ignore the FreePBX warning that you’re entering an odd type of inbound route by clicking OK. But you knew that.
Now let’s get you signed up with an account on e164.org. Go to the web site and click the Sign Up tab. Go through the sign up drill and then log into your new account. Then click the Phone Numbers tab and Add your phone numbers to e164. For each number, enter the area code and number. Then click the Next button. You’ll be warned about not having the number you’ve specified redirected to an IVR. If you already have this DID redirected to an IVR, change the routing temporarily to an extension that you can answer to obtain your PIN before you press Next to proceed. You’ll then be prompted for the SIP address to contact your server. Leave the default SIP protocol and plug in the address you created, e.g. e164@pbx.dyndns.org (using your own FQDN, of course). As soon as you click the Next button, your phone should start to ring, but there may not be a message when you answer. Hang up and wait for the second call within 15 minutes. It will include your PIN. Now click on the Phone Numbers tab and update your phone entry by choosing Enter PIN and typing your assigned PIN. Your phone number now has been activated with the e164 service. To complete the setup, you’ll want to click on the Do Not Call option and make your selections. You also can decide whether to list yourself in the ENUM White Pages directory.
Remember that the real purpose of this drill was to avoid charges when you place outbound calls to numbers in the ENUM directory. We merely added your numbers to e164.org so that others could benefit as well. So the final step before you can start saving money is to configure FreePBX to handle ENUM lookups for outbound calls from your server. One more observation may be helpful. You’ll recall that one of the limitations of FreePBX has always been that once an outbound route was chosen for a call, if the call was completed using the first destination trunk in that route, then the call processing ended there. ENUM adds a new wrinkle because we basically want to connect to ENUM to check for a free route and, if no matching entry is found, then we want the next trunk to process the call. As luck would have it, FreePBX has been tweaked to allow this scenario. All you have to do is create an ENUM trunk and then place it first in your sequence of trunks for each of your outbound routes. If an ENUM entry is found for the number you’re calling, the call will be routed as a free call with a direct SIP connection. Otherwise, the call processing will continue and the call will be routed using the next trunk specified in your outbound route.
There are two steps in FreePBX to implement ENUM. First, we need to create a special ENUM trunk. And second, we need to adjust our outbound routes to use the ENUM trunk first, and then the series of trunks you already have specified in each outbound route. NOTE: You obviously wouldn’t do this for an emergency 911 outbound route.
In FreePBX, click Setup, Trunk, Add ENUM Trunk. Enter your desired CallerID for these calls. Set a maximum number of channels, if desired, and then leave the other entries blank in most cases. Save your settings and reload your dialplan. Now click Setup, Outbound Routes and adjust the sequence of trunks for each of your existing routes. Be sure to put ENUM in the top position of each desired route. We also recommend adding a new Free Calls route so that users on your system can dial 0 and then a number to place a call through ENUM and then Gizmo5. If neither has a route for calling the party for free, the call will fail. The dial patterns might look like this for U.S. calls:
0|1NXXNXXXXXX
0|NXXNXXXXXX
The trunk list would look like this:
0 ENUM
1 SIP/gizmo5
Continue reading Part II.
Today’s Must Read: 101 Things You Can Do With Asterisk
VPN in a Flash Update! We’ve had over 100 reservations for our new VPN in a Flash system since last week. We’re very close to having a manufacturer in place so hopefully we’ll have more good news in a week or two. We have begun the documentation for the new product, and we encourage you to take a look and offer any questions or comments you may have on our forums. The documentation is in the new Google Knol format and can be reviewed here. It’s not too late to get in the queue and place a reservation for a system. Just send us a note, and we’ll keep you posted as the release date approaches. It’ll hold your place in line with absolutely no obligation to purchase.
Coming Attractions. We’re very close to signing on a new VoIP provider for PBX in a Flash users that will provide penny-a-minute calls in the U.S. and Canada as well as all-you-can-eat plans for just over $10 a month with an annual contract. We’re also only a week or two away from a new version of AsteriDex with Outlook synchronization and a TTS dialer for AsteriDex queries from any connected Asterisk phone. Stay tuned!
Hosting Provider Deal of the Century. Just an FYI that the Nerd Vittles hosting provider, BlueHost, has raised the bar again on hosting services. For $6.95 a month, you can host unlimited domains with unlimited web hosting disk storage and unlimited monthly bandwidth. Free domain registration is included for as long as you have an account. It really doesn’t get any better than that. And their hosting services are flawless! Just use our link. You get a terrific hosting service, and we get a little lunch money.
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…
Free Asterisk Solutions
