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The Most Versatile VoIP Provider: FREE PORTING

Introducing Digium’s Awesome SIP Phones for Asterisk



If you’ve been waiting for a low-cost, feature-rich SIP phone that meshes perfectly with your Asterisk® PBX, your prayers have been answered. Digium has just released not one, but four, new SIP phones with prices starting at $59. No, that’s not a typo. Digium gave us a couple of early models to play with, and today we’ll walk you through the incredibly simple setup. We would begin by noting that, despite the pricing, these phones are configured with nothing resembling a bargain basement feature set. All four models have color displays, HD Voice, POE for use without the $15 power adapter, and at least two lines. The phones can be configured using the phones themselves, or through a slick web interface, or with auto-provisioning by MAC address. Beginning with the $89 A22, the top three models support gigabit Ethernet. With the $119 A25, you get four line registrations as well as a second LCD supporting six Rapid Dial keys or up to 30 BLF entries. The top-of-the-line $169 A30 supports six line registrations and an LED setup that closely matches our previous VoIP Phone of the Year, Yealink’s T46G. While the phones were not designed for use with Switchvox®, we found them to be plug-and-play with 3CX® which is probably also true with Switchvox even though we have not tested them on that platform. We have been using our A22 phone with one line connected to Incredible PBX® for the Raspberry Pi and the second connected to VitalBox. We’ve had zero issues with the phone, and sound quality is excellent.



Connecting Digium’s A-Series IP Phone

To get started, you’ll need a power source for the phone which can be either a POE network connection or a power adapter. You’ll also need to connect to a network that can provide DHCP or VLAN configuration data. Once the phone boots up, press the checkmark button (✓) twice to display the IP address assigned to the phone. Using a desktop browser, navigate to that IP address and enter admin:789 as the default login credentials.

Configuring a SIP Extension on Your IP Phone

Once you’re logged in, click on the Line tab and fill in the blanks for the SIP1 account using the desired extension number, extension password, and IP address of your Asterisk server. Be sure Activate is checked. It should look something like the following. Then click Apply.

This one-minute setup is all that’s required to put your new phone into production with Asterisk. You’re ready to make and receive calls. The L1 button on the A20 or A22 phone (pictured above) should now be lit. To light up the L2 button, add a second SIP connection by repeating the drill after choosing the SIP2 Line from the pull-down menu. If you have redundant PBXs, fill in the IP address of the Backup server, and the phone will automatically failover when the primary PBX goes down. It doesn’t get any easier than that.

With 3CX extensions, the setup is virtually identical except the phone’s Authentication Name field should reflect the Authentication Name chosen when setting up the 3CX extension.

Customizing Your SIP Phone Settings

VoiceMail Setup. The voicemail button can be activated for one or both SIP lines in the Advanced Settings tab under each of the SIP connections. Check the Subscribe to Voice Message box and enter the Voice Message Number to retrieve your voicemails, e.g. *98701 for extension 701 on an Asterisk PBX or 999 for a 3CX extension’s voicemail.

Customizing Phone Display. If you’d like to customize the branding and background image on your phone, navigate to Phone Settings and click the Advanced tab. Here’s a link to download one of our favorite beach scenes (pictured above), or you can use your own 320×240 BMP image on the A20 and A22. The high end phones use a 480×272 BMP image. The Asterisk label at the top of the phone’s display can also be adjusted in the Greeting Words field. We’re Enchilada fans personally. 🙂

Changing Passwords and PINs. You also can adjust the passwords and PINs for the phone device itself under the Phone Settings:Advanced tab. The default is 789. To modify the admin credentials for the browser interface or to add new accounts, go to System and click on the Account tab. Because the phone can be configured using either the phone itself or the browser interface, you’ll need to change both sets of passwords to secure your phone.

Adjusting Codecs. Depending upon your PBX setup, you may need to adjust or reorder the codecs for one or both of your SIP lines. Simply navigate to Line:SIP1:Codec Settings and make any necessary changes. HINT: You’ll rarely have a problem if you make G.711U (U.S.) or G.711A (elsewhere) your primary codec although G.722 is what you’ll want for HD Voice. This is especially important if you’re using Google Voice trunks or 3CX client software.

Auto-Provisioning Your A20, A22, and A25 Phones

Let’s get to the fun stuff now. Everything we’ve covered (and much more) can be scripted with these new phones. You can read all about it here. For today, let’s get your Phonebook Contacts populated using your AsteriDex database entries. And then you can press the Down button on the phone to retrieve your Contacts.

Setting Up Phone Provisioning. Before you can auto-provision your phone, both your phone and your Asterisk server need a little navigation information. Let’s start with the phone so login as admin:789 to get started. Click on the System option and then the Auto Provision tab. Write down the last 12 digits of your phone’s MAC address (CPE Serial Number highlighted above). Check the DownloadDeviceConfig option (as shown). Disable the DHCP Option and the SIP Plug and Play options by clicking on the respective tabs. Then open the Static Provisioning Server option (as shown). Enter the local IP address of your server assuming your phone and server are both behind a firewall. For the Protocol Type, choose HTTP. For the Update Mode, choose Update After Reboot. Then click the Apply button.

Next, let’s configure the phone so that you can press the Down arrow button to access your Phonebook Contacts. Click on the Function Key option in the left margin. Then look in the Programmable Keys section and locate the row with the settings for the Down button. Change the entry in the Desktop column to Phonebook. Then click the Apply button.

Configuring Asterisk for Phone Provisioning. Now we need to get your server set up to support phone provisioning. The way provisioning works is we will set up a provisioning profile for each phone which will be processed by your web server whenever a phone is rebooted. This profile will also tell the phone where to find your Phonebook Contacts XML file. To get started, navigate to /var/www/html and create a new .cfg file for each of your phones using the 12-character MAC address of the phone, e.g. 000123456789.cfg. The file should look like the following with the exception of the Auto Pbook Url entry which should reflect the local IP address of your server:

<<VOIP CONFIG FILE>>Version:2.0.0.0

<PHONE CONFIG MODULE>
LCD Title          :IncredblePBX

<AUTOUPDATE CONFIG MODULE>
Download CommonConf:0
Download DeviceConf:1
Check FailTimes    :5
update PB Interval :720
Clr PB B4 Import   :1
Trust Certification:0
Enable Auto Upgrade:0
Upgrade Server 1   :
Upgrade Server 2   :
Auto Upgrade intval:24
Auto Pbook Url     :http://192.168.0.108/phonebook.xml

<<END OF FILE>>

Populating Phonebook Contacts with AsteriDex. Now we’re ready to build the Phonebook Contacts file (phonebook.xml) using the AsteriDex 4 database. Just issue the following commands and then reboot each of your phones (Menu+8+Yes):

cd /var/www/html/asteridex4
wget http://incrediblepbx.com/asterisk-phonebook.tar.gz
tar zxvf asterisk-phonebook.tar.gz
rm -f asterisk-phonebook.tar.gz
php asterisk-phonebook.php

Digium A-Series IP Phone User Guide

Last but not least, take a look at Digium’s A-Series IP Phone User Guide (PDF) for more tips.

Final Thoughts on A-Series IP Phones

If you couldn’t already tell, we’re quite impressed with the new A-Series phones from Digium. If you’re on a budget, the $59 model is one terrific bargain for home or SOHO use. The only thing you’re really forfeiting with this phone is the gigabit Ethernet port which will have zero impact on small and medium-sized network implementations of a VoIP server. Rather than buying power adapters for your phones, drop by your favorite WalMart and purchase a network switch that includes POE support. They start at about $30. Then pick one of these phones up from your favorite provider and let us know what you think. You’ll also be helping to fund Digium’s open source Asterisk project. Enjoy!

Originally published: Friday, April 13, 2018





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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.
 



Free Worldwide VoIP Calling with SIP URIs and Issabel 4

SIP URIs make the VoIP World go ’round. They’re the email-like addresses that carry VoIP calls between SIP servers to reach their destination. But there’s gold in them hills if you know how to use SIP URIs because SIP URI calls are free even if the calls travel all the way around the world. We previously documented how to deploy SIP URI calling with PIAF5 and 3CX, and today we’ll show you how to make SIP URI calls from and to your Issabel™ server using Incredible PBX®. More importantly, we’ll show you how to do it safely without opening up the anonymous calling floodgates and compromising your Asterisk® server.

Now that we’ve gotten the price of cloud-based servers down to a respectable $1.50 to $2.50 per month, it’s time to cut the cord and kiss your home-grown server goodbye. The babysitting headaches and maintenance costs of running your own server and paying for electricity simply aren’t worth it. There’s another reason. NAT-based routers and firewalls complicate things when it comes to VoIP. Not only do you have to wrestle with SIP headers and ALG, but you also have to troubleshoot thorny one-way audio issues with VoIP calling. So bite the bullet and play along today. Skip that Starbucks coffee this week and you’ve all but paid for a full year’s worth of VoIP server hosting in the Cloud.

Getting Started with Vultr

If you just want to experiment in a cloud-based sandbox, then there’s no better option than Vultr. For less than a penny an hour, you can build a VoIP platform, tear it down, and build another one for less than the cost of a nickel candy bar. You remember those, don’t you? I actually tried to think of something that still costs a nickel, but that was the best I could do… and that was 50+ years ago.

After you’ve created an account on Vultr with our referral link, the first step is to create your new cloud instance. Choose New York or Miami as your desired hosting site (they both have $2.50/month availability) and select 64-bit CentOS 7 as your server platform. An additional 50¢ a month buys you automatic daily, weekly, or monthly backups to a separate, fault tolerant storage system in the same data center. HINT!

(1) Once you’ve built and started your new virtual machine, log into your server as root using SSH/Putty and immediately change your root password: passwd.

(2) With the $2.50 size VULTR virtual machine, you must create a swapfile before beginning the Issabel installation. Here are the commands:

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

(3) Now skip down to the Issabel installation section to continue.

Getting Started with WootHosting

If $2.50 a month is too rich for your blood, there actually are two $1.50 a month options at WootHosting if you sign up for a year. With the New York special, you get a single VPS platform. With the twofer special, you actually get two VPS platforms in your choice of cities. WootHosting also offers considerably more horsepower with quadruple the RAM and more storage space. You can read our review of WootHosting here.

(1) Start by creating a CentOS 7 Minimal VPS platform in New York, Miami, or Los Angeles. If you opted for the WootHosting twofer special, then you’ll need to create a user and then a virtual server platform that looks something like this:

(2) Set a root password in the Root/Admin Password tab and enable TUN/TAP (needed for NeoRouter) in the Settings tab.

(3) Login to your server as root using SSH Terminal or Putty.

(4) Now continue with the Issabel installation as documented below.

Installing Issabel on Your VPS Platform

Issue the following commands to install Issabel. When prompted for a MariaDB (MySQL) and admin password, make certain to use passw0rd (with a zero) for your MariaDB password and a very secure password for your admin password, the one you’ll use to login as admin to Issabel’s web interface.

yum -y update
yum -y install wget nano
wget -O - http://repo.issabel.org/issabel4-netinstall.sh | bash

When the Issabel install is complete, your server will automatically reboot.

Installing Incredible PBX for Issabel on Your VPS

After the reboot, log back into your server as root and issue the following commands to install Incredible PBX for Issabel. You will again be prompted for MariaDB and admin passwords. Do exactly as you did above using passw0rd as your MariaDB password. At the conclusion of the install, you will again be prompted for the same admin password you used above. This is actually used for Apache web security and will be the first prompt you see when you attempt to login to any web application including Issabel, AsteriDex, and Reminders.

wget http://incrediblepbx.com/IncrediblePBX11-Issabel4.sh
chmod +x IncrediblePBX11-Issabel4.sh
./IncrediblePBX11-Issabel4.sh

When the installation finishes, reboot your server once again and then log back in as root. The Automatic Update Utility will load current patches and then display pbxstatus.

Using a browser, login to the Issabel web client at the IP address shown in pbxstatus. You’ll be prompted twice (http and https) for your Apache admin credentials which should be the same as your Issabel GUI admin credentials. Save your Apache credentials in your browser when prompted to do so. Then you won’t have to provide Apache credentials again. Next, login to the Issabel GUI with admin and your admin password.

That completes the basic install of Incredible PBX and Issabel. Our previous tutorial will walk you through the basics of setting up your trunks, extensions, and routes in Issabel.

Overview of SIP URI Implementation with Issabel

There are any number of ways to implement incoming SIP URI support on Asterisk-based servers. Most are terribly insecure and provide an easy target for the bad guys to make free calls using your paid VoIP provider accounts. The traditional method to permit SIP URI access to your server would require poking a hole in your firewall to allow unrestricted access to the SIP port of your server, UDP 5060. In addition, it would require enabling unrestricted anonymous calling access to Asterisk via FreePBX®. After all, that’s similar to the way the Ma Bell telephone system operated. Anyone in the world could call you provided they had your number. The major deterrent was that most of the calls incurred costs to the caller with no monetary benefits being derived. VoIP changed all of that. Using a SIP client and SIP URIs, anonymous individuals now can place unlimited calls to unlimited VoIP servers at no cost. And, if they get lucky, they can decipher a way to call into your PBX via SIP URI and then call out using phone trunks that you actually have to pay for. Bad idea!

We have a better way that’s entirely secure and won’t incur calling charges for incoming anonymous SIP URI calls. The solution is to set up a trunk with a hosting provider that supports anonymous SIP URI access and then leave it to the VoIP provider to manage the thorny SIP security problems which is not Asterisk’s strong suit. Once we’ve set up the SIP URI with the provider, we will register a trunk with that provider on our Issabel server. Then all of the anonymous SIP URI calls will come into the SIP provider and be rerouted to Issabel through our registered trunk with that provider. No firewall puncturing is required because we will be using a registered trunk and tunnel between our server and the provider.

Implementing SIP URI Support with VoIP.ms

Our favorite VoIP provider to implement this is VoIP.ms in Canada. They have POP servers throughout the world so you can pick a server that is close to your cloud-based Issabel server. VoIP.ms POPs are available in Tampa, New York, and Los Angeles among others worldwide. Step one is to set up an account at VoIP.ms if you don’t already have one. Step two is to set up a SubAccount with a difficult-to-guess VoIP.ms Internal Extension Number. Be sure to jot down the Username and Password you set up for your SubAccount. You’ll need them in a minute. In our example today, we’re using 4772235642 as the internal extension number. This means other VoIP.ms account holders can reach this account by dialing 10+ 4772235642. And anyone on the Internet can reach this account by dialing your VoIP.ms account number + 4772235642 at the POP to which you are registering a VoIP.ms DID associated with this SubAccount. Clear as mud? Hang in there a bit longer.

Step three is to sign up for a VoIP.ms DID. This could be a free iNUM DID or a commercial DID (traditional 10-digit NANPA number) that your PBX could actually use to receive traditional calls. Commercial DIDs range in price from under $1 a month with incoming calls costing under a penny a minute to $4.25 a month with unlimited incoming (residential) calls. For our purposes today, the type of DID and its commercial cost really don’t matter. When any of these DIDs are connected to a SubAccount with an associated Internal Extension Number, SIP URI calls to that DID’s internal extension number are free! So… the cheaper, the better.

The final step on the VoIP.ms side of things is to associate your DID with a SubAccount and choose a POP server to process the calls coming to you. This is done under the Manage DIDs tab in the VoIP.ms web interface.

So let’s review what we’ve done. We set up a VoIP.ms account. We created a SubAccount in their web interface and created an internal extension number for that subaccount. Next, we ordered a DID. And finally, we associated that DID with the subaccount we created and chose a POP server to deliver the inbound calls to our server.

Now we’re ready to set up a VoIP.ms trunk on our Issabel server and test things out.

Implementing SIP URI Support with Issabel

Incredible PBX makes setting up a VoIP.ms trunk easy. The template is already in place in the Issabel GUI. All you’ll need are your VoIP.ms credentials (SubAccount Username and Password), your DID number that you ordered from VoIP.ms, and the name of the VoIP.ms POP server (from Manage DIDs) that will be delivering the incoming calls. You’ll also want to jot down your Internal Extension Number (without leading 10) that you set up in your VoIP.ms SubAccount. You’ll need that and the FQDN of the VoIP.ms POP in order to decipher the SIP URI (phone number) to reach your server.

While logged into the Issabel GUI, navigate to PBX:PBX Config:Trunks:VoIPms. Insert your DID in the Outbound CallerID field. Uncheck the Disable Trunk box. Under PEER Details, insert your VoIP.ms username in the username and fromuser fields. Insert your VoIP.ms password in the password field. Insert the FQDN of the VoIP.ms POP server in the host field. Under Register String, insert your username, followed by a colon, your password, followed by @, your POP FQDN, followed by /DID, e.g. johndoe:secret@tampa.voip.ms/8005551212.

Next, we need to create an Inbound Route to process the incoming calls from VoIP.ms. Navigate to PBX:PBX Config:Inbound Routes. Click Add Incoming Route. In the Description field, enter VoIPms-Incoming. In the DID Number field, insert your DID number. In the Source field, choose OpenCNAM. In the Set Destination dialog, choose a destination for the incoming calls, e.g. an extension, ring group, or IVR. Then click Submit and reload dialplan.

Finally, we need to adjust a SIP setting to support SIP URI calls from VoIP.ms. Navigate to Security:Advanced Settings. Set Enable Direct Access ON. Set Allow Anonymous Calls OFF. Enter your admin password twice. Click SAVE.

Next, navigate to PBX:PBX Config:Unembedded IssabelPBX. When the new window opens, navigate to Settings:Asterisk SIP Settings. In the External IP field, insert the IP address of your Issabel server. Click Auto Configure button immediately below that. Scroll to the bottom and, in Other SIP Settings, insert match_auth_username = yes in the two fields provided. Click Submit Changes and reload dialplan. Click Logout: Admin at the top of the browser window and then close the browser tab to return to the main Issabel GUI.

Deciphering the SIP URI for Your Issabel PBX

From the information you wrote down above, here’s how to assemble the SIP URI for your Issabel PBX. Start with your VoIP.ms account number, e.g. 101595. Add your Internal Extension Number, e.g. 4772235642. Add the @ symbol followed by the VoIP.ms POP routing calls to Issabel, e.g. tampa.voip.ms. You can give ours a try if you’d like to interact with Allison’s Demo IVR: 1015954772235642@tampa.voip.ms. Most SIP clients support SIP URI calling including Zoiper (PCs) and Telephone (Macs).

Placing Outbound SIP URI Calls from Issabel PBX

The easiest way to place outbound SIP URI calls from your Issabel PBX is to set up Custom Extensions for the destinations you wish to reach.

Navigate to PBX:PBX Config:Extensions:Other (Custom) Device. Assign an extension number and display name to the extension and insert the SIP URI in the dial field using the syntax shown below. Then click Submit and reload your dialplan.





We’ve barely scratched the surface of what you can do with Incredible PBX for Issabel. Head over to our introductory article where we’ve documented dozens of Asterisk® applications that await your exploration. Enjoy!

Published: Thursday, August 24, 2017  



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…

XiVO Nirvana: Cloud Hosting with SIP Service for 15¢ a Day



Unlike in the Ma Bell era, the real beauty of VoIP technology is being able to experiment with different providers and never having to put all your eggs in one basket. And today we’re pleased to introduce a new XiVO hosting and trunk combination that provides first-class service at truly incredible price points. For the XiVO cloud provider, we’ve chosen OVH, a cloud platform that was recommended to us by our friends at 3CX. It also works great with the new PIAF5 platform for those exploring a (free) commercial alternative.

$3.49 a month at OVH (No, that is not a typo!) gets you an OpenStack KVM with 2GB RAM, 10GB of SSD storage, and RAID10 redundancy plus a 99.95% uptime SLA. And you have your choice of worldwide data centers with many more on the way. For those in the United States, the closest location for the time being is Québec which also happens to be the hometown of the XiVO developers. Ping times on both U.S. coasts are well under 100 milliseconds so you won’t have to worry about voice quality and latency.

For our XiVO SIP provider today, we’ll walk you through setting up inbound and outbound calling with Anveo Direct, one of the least expensive SIP providers in the world. In addition to great pricing, Anveo also provides SIP URI failover for your Anveo trunks. Just follow our previous tutorial to set up a SIP URI address for your XiVO server. Or you could use the SIP URI of your RingPlus mobile phone if you followed our previous tutorial. Anveo also happens to give you total control over call routing with their highly configurable LCR technology. Pay-by-the-minute incoming SIP calls in the U.S. are a penny per minute. Outgoing U.S. calls typically range from one-tenth to three-tenths of a cent per minute depending upon the destination.



Installing Incredible PBX for XiVO at OVH

To get started with OVH, order the VPS SSD 1 package and choose Debian 8 as your operating system. Once your credentials arrive, log into your server as root using SSH/Putty and immediately change your root password: passwd.

While still logged into your server as root using SSH/Putty, issue the following commands to kick off the base install:

cd /root
wget http://incrediblepbx.com/IncrediblePBX13-XiVO.sh
chmod +x IncrediblePBX13-XiVO.sh
./IncrediblePBX13-XiVO.sh

After rebooting (it takes about 2 minutes on the OVH platform), log into your server again as root and issue the following command to complete the XiVO and Incredible PBX installation and configuration:

./IncrediblePBX13-XiVO.sh

You now can proceed to Incredible PBX Initial Configuration tutorial to continue your setup. Much of this initial configuration already has been put in place using our XiVO Snapshot technology. Just review the settings and make sure they meet your requirements. Then you’ll be ready to set up your Anveo Direct trunk and routes to handle your SIP calls.

Getting Started with Anveo Direct

We previously have documented how to set up Anveo Direct for Outbound Calling from your XiVO PBX so we won’t repeat it here. Today we’ll show you how to obtain and configure an Anveo Direct DID to enable Inbound Calling to your XiVO PBX. We’re going to walk you through the procedure to install a U.S. DID, but Anveo Direct offers worldwide DIDs. And we’ll show you how to modify the default XiVO setup to support international DIDs should you wish to use them.

After you’ve set up Outbound Calling with Anveo Direct as previously documented, log back into the Anveo Direct portal with your credentials.

Under the Inbound Service tab, choose Order Anveo Direct DID and click Geographic. Then select the United States, your desired State, and your desired City to obtain a DID. Select one or more DIDs as desired and then click ORDER PHONE NUMBERS SELECTED. Choose either the pay-by-the-minute or all-you-can-eat option for your DID depending upon your needs and complete your purchase. There’s a 3-month minimum charge for all DIDs.

Once you complete your DID purchase, choose the Inbound Service tab again and choose Configure Destination SIP Trunks. ADD A NEW SIP TRUNK following the example below and specifying the IP address of your XiVO PBX. Include a Failover SIP URI if you’ve set one up. Don’t confuse the SIP URI entry with the Failover entry. The first is mandatory while the second is not. The SIP URI entry tells Anveo how to send out the SIP calls to your XiVO PBX. It should look like this using your own server’s IP address or FQDN: $[E164]$@1.2.3.4 or $[E164]$@ovh.yourdomain.com. Click SAVE when finished.

Next, choose the Inbound Service tab again and choose Configure AnveoDIDs. Every DID you purchased should already have an entry here. Click the EDIT button to open the options window. Then click the Call Options tab. From the pull-down, choose the Destination SIP Trunk entry that you created in the previous step:

Finally, under the CallerID tab, choose {E164} (no prefix). Then click SAVE. That completes the DID setup process on the Anveo Direct side. Now you simply have to configure XiVO to accept the incoming calls from your Anveo DID.

Configuring Anveo DIDs on Your XiVO PBX

Unlike most SIP providers, Anveo Direct does not require (nor permit) registration of your Anveo trunks. Calls to Anveo DIDs will simply be routed to your XiVO PBX based upon the SIP URI you specified above. If your Incredible PBX for XiVO server was built on or after November 9, 2016, then the Anveo trunk and dialplan are already in place. All you’ll need to do on the XiVO side is to add an Incoming Call route for each DID telling XiVO where to send the calls. If you have an existing Incredible PBX for XiVO server, there’s a little more work to do, and we’ve documented the steps to support Anveo DIDs on the PIAF Forum.

For U.S. DIDs, the DID format is 11 digits beginning with a 1. The example below would route incoming calls from the Anveo DID to the Demo IVR. You could just as easily have specified an extension or ring group to take the calls.

Published: Wednesday, November 9, 2016



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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…

Integrating SIP URIs into XiVO for Free Worldwide Calling

It’s been a while since we’ve explored SIP URIs and all of the advantages that SIP URI calling brings to your PBX. Number one on that list is FREE calling to and from anyone on the planet so long as both of you have an Internet connection with a SIP phone or a VoIP server such as Incredible PBX for XiVO. SIP URIs are the fundamental building blocks for VoIP technology. Consider this. If everyone in the world had a SIP address instead of a phone number, every call to every person in the world via the Internet would be free. That pretty much sums up why SIP URIs are important. The syntax for SIP URIs depends upon your platform. With Asterisk® they look like this: SIP/somebody@FQDN.yourdomain.com. On SIP phones, SIP URIs look like this: sip:somenameORnumber@FQDN.yourdomain.com. Others use somenameORnumber@FQDN.yourdomain.com. Assuming you have a reliable Internet connection, once you have “dialed” a SIP URI, the destination SIP device will ring just as if the called party had a POTS phone. Asterisk® processes SIP URIs in much the same way as other calls originating from trunks and, as noted, SIP URI calls of any duration to anywhere are free. Today we’ll show you how to set things up on your XiVO PBX without exposing any ports to the Internet in a way that would jeopardize your server’s security.

Placing Outbound SIP URI Calls with a SIP Softphone

There are two ways to place outbound SIP calls. You can use a SIP phone or softphone that supports SIP URI calling to dial SIP URIs directly. If you have a Mac, the best free softphone for SIP URI calling is Telephone which you can download from the App Store. On other platforms as well as Macs, Zoiper is a great no-cost option. Both of these softphones support the sip:someone@FQDN.yourdomain.com syntax. An excellent way to test this is to call our friend Lenny and strike up a conversation: sip:2233435945@sip2sip.info.

Configuring Outbound SIP URIs with XiVO

The major drawback of SIP URIs is they’re difficult both to remember and to dial. It’s much simpler to dial a short number using a traditional phone. And, with Incredible PBX for XiVO, it’s easy to create custom extensions that can be accessed simply by dialing a few digits from any phone connected to your server. Here’s how to set it up in the XiVO GUI.

1. Create a User and assign the Customized Protocol and an Extension Number to that user:

TIP: If you’d prefer to use a different series of numbers for speeddials so you don’t get them mixed up with your standard extension numbers, just add a new range of numbers for XiVO: IPX Configuration → Contexts → Default → Users. Then choose one of them above.

2. Access the new Line that was generated for the new User:

3. Replace the Interface entry for the Line with the desired SIP URI for your speeddial, e.g. SIP/2233435945@sip2sip.info. Then SAVE your new Line settings.

4. Dial 750 from an Extension on your XiVO PBX to try out Lenny using your new SIP URI.

A Better Way to Create SpeedDials with XiVO

We’ve gone through the XiVO GUI approach to demonstrate that it is indeed possible to create speeddials for SIP URIs. However, there is a better way unless you’re one of the naysayers that believes everything is better in a GUI. If you have dozens or even hundreds of speeddials to create, you may change your mind. The GUI approach could obviously become tedious. Instead, with one line of Asterisk dialplan code, you can create as many speeddials as you like keeping in mind that it’s your responsibility to assure that SIP URI extension numbers don’t conflict with existing extensions on your server. Insert a new section of code at the bottom of /etc/asterisk/extensions_extra.d/xivo-extrafeatures.conf and reload your dialplan: asterisk -rx "dialplan reload".

You can also insert this code from within the XiVO GUI itself: IPX Configuration → Configuration Files. Edit xivo-extrafeatures.conf and insert the following code snippet at the end of the file and Save your entries. The dialplan will be reloaded automatically.

Some of our favorites include the following:

;# // BEGIN SpeedDials
exten = 882,1,Dial(SIP/200901@login.zipdx.com)     ; V-U-C on Fridays at noon EST
exten = 8378,1,Dial(SIP/thetestcall@getonsip.com)  ; T-E-S-T everything VoIP
exten = 53669,1,Dial(SIP/2233435945@sip2sip.info)  ; L-E-N-N-Y
exten = 68742,1,Dial(SIP/0289304@zero-nine.biz)    ; M-U-S-I-C
exten = 3733411,1,Dial(SIP/411@ideasip.com)        ; F-R-E-E-4-1-1 Directory Asst
;# // END SpeedDials

Creating a SIP URI Address for Your XiVO PBX

Free calls to other folks is only half of the story, of course. You’re also going to want a way for people to call you without incurring charges for the calls. There are many SIP URI approaches for inbound calls. Most of them are not safe with Asterisk. Let me say that again. Most of them are not safe with Asterisk. The reason is because most of them force you to open SIP access to your server for everybody in the world. Unfortunately, that means they can not only call you, but they can also attempt to use your extensions and trunks to place very expensive calls to others. Don’t even think about opening the SIP floodgate by exposing port 5060 unless Bill Gates sends you a check every week. You’ve been warned!

Setting Up an iNum SIP URI Trunk with XiVO

The better and safer way to add SIP URI connectivity to your XiVO server is to first obtain a freely available iNum DID from one of the many providers that support iNum and then use that provider as a SIP intermediary. All SIP calls pass only over your registered trunk with your provider. Our favorites in no particular order are VoIP.ms, LocalPhone and CallCentric. There are many, many others. In order to obtain a free iNum DID, you will need an account with one of these providers. All require some sort of minimal deposit, but you usually can get back unused funds if you decide to close your account down the road. Our XiVO tutorials for VoIP.ms, LocalPhone, and CallCentric will walk you through creating your SIP account and registering it with your XiVO server. Then verify that your SIP account is registered:

asterisk -rx "sip show registry"

Configuring an iNum DID with VoIP.ms

Our trunk tutorials for LocalPhone and CallCentric will walk you through their setup procedures for iNUM DIDs. VoIP.ms provides more flexibility in redirecting trunks so let us quickly walk you through their procedure. Log in to your VoIP.ms account and then order your free iNum DID at this link. Your iNum DID then will appear in your DID Listing here. Write down your iNum DID which you’ll need in a minute to configure the XiVO side of things. Then click on the Edit DID icon beside your iNum DID and assign the DID to your registered Main Account or the SubAccount that you’ve already registered with XiVO. Be sure to use the same DID POP that you used when you registered your VoIP.ms account with XiVO. Don’t enable VoiceMail and set the ring time to 60 seconds just to keep things simple.

Configuring XiVO to Support Your iNum DID

Now for the XiVO part. Using a browser, log into the XiVO GUI. Navigate to IPX Configuration → Contexts → Default → Users. For VoIP.ms and LocalPhone, add a new Number Range starting and ending with your iNum DID. Then click Save. For CallCentric, do the same thing but substitute your CallCentric username which will be an 11-digit number starting with 1777.

Repeat the above in IPX Configuration → Contexts → from-extern (Incalls) → Users.

For CallCentric only, also click on the Incoming Calls tab and add a new Number Range. For the Starting value, use your 11-digit LocalPhone username. For the DID length, set it to 11. You do NOT need to include a Number Range ending value. Click Save when you’re finished.

For VoIP.ms, navigate to IPX Settings → Users. Then Add a new User for your iNum DID. In the General tab, name the User VoIP.ms iNum. In the Lines tab, provide your actual iNum DID number. This must be the same number you added to the Number Range in the Default context above. In the No Answer tab, set the Fail option to the Destination of your choice, e.g. an extension, a ring group, an IVR, etc. Then click Save.

For LocalPhone, navigate to Call Management → Incoming Calls and Add a new Inbound Route for your DID specifying the destination for the calls using your iNum DID number:

For CallCentric, navigate to Call Management → Incoming Calls and Add a new Inbound Route using your 11-digit CallCentric username as the DID. Then specify the destination for the calls and click Save.

Calling Your XiVO PBX Using Your iNum SIP URI

To receive SIP URI calls safely on your iNum DID, your SIP URI is your iNum DID number followed by @sip.inum.net, e.g. 883510012345678@sip.inum.net. Neither the identity of your XiVO PBX or your SIP service provider is ever exposed. Enjoy your safe, free calling!

Originally published: Monday, September 26, 2016





Need help with Asterisk? Come join 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…

Wazo Trunks Tutorial: Installing a CallCentric SIP Trunk



Setting Up a SIP Trunk at CallCentric

1. Sign up for a CallCentric account.

2. Login to your CallCentric account. You can order a DID here. Follow this link for a free iNum DID. You also can order a free DID from one of several New York area codes.

Test call quality by dialing Allison’s Demo IVR courtesy of free DID available to all from CallCentric: 1-631-440-3057

3. Click the Extensions tab to obtain and/or set your SIP username and password. You’ll need these to register your CallCentric trunk with Wazo. Anyone with your credentials can make and receive calls!

Setting Up a CallCentric SIP Trunk on Wazo

SIP trunks are different than traditional Ma Bell phone lines. With SIP trunks, you need not use the same provider to process incoming and outgoing calls. With some SIP providers including CallCentric, incoming and outgoing calls are managed on the same server. To place outgoing calls with CallCentric, all you need are your credentials. You do NOT need a DID. To receive calls from Plain Old Telephones, you will need a CallCentric DID. Free iNum DIDs and some free New York DIDs are among the choices.

In the Wazo GUI, create a new CallCentric SIP Trunk by choosing IPBX:Trunk Management:SIP Protocol. Click on + Add to open a new template.

In the General tab, fill in the blanks using your CallCentric credentials in the template below:

Next, click on the Register tab and fill in the blanks using your SIP credentials from CallCentric. Name, Authentication username, Remote Server and Contact are all your CallCentric account number. Password entry is not required.

In the Signalling tab, set DTMF to RFC2833, Monitoring to Yes, and add the ULAW and ALAW Codecs.

In the Advanced tab, set Insecure = ALL and Port = 5060.

Click SAVE when you’ve finished.

Wazo will not actually process incoming and outgoing calls through this CallCentric trunk until you configure an outgoing route in IPBX:Call Management:Outgoing Calls and an incoming route using IPBX:Call Management:Incoming Calls if you have a DID. Outgoing and Incoming call routing are covered in separate tutorials.

Wazo Trunks Tutorial: Installing a LocalPhone SIP Trunk



Setting Up a SIP Trunk at LocalPhone

1. Sign up for a LocalPhone account.

2. Login to your LocalPhone account. You can order a DID here. For a free iNum DID, choose International (iNum) as your desired country.

3. Click Settings to configure your account with a very secure password.

4. Click on the Internet Phone icon to obtain your SIP credentials.

Setting Up a LocalPhone SIP Trunk on Wazo

SIP trunks are different than traditional Ma Bell phone lines. With SIP trunks, you need not use the same provider to process incoming and outgoing calls. With some SIP providers including LocalPhone, incoming and outgoing calls are managed on the same server. To place outgoing calls with LocalPhone, all you need are your credentials. You do NOT need a DID. To receive calls from Plain Old Telephones, you will need a LocalPhone DID (aka Incoming Number).

In the Wazo GUI, create a new LocalPhone SIP Trunk by choosing IPBX:Trunk Management:SIP Protocol. Click on + Add to open a new template.

In the General tab, fill in the blanks using your LocalPhone credentials in the template below:

Next, click on the Register tab and fill in the blanks using your SIP credentials from LocalPhone. Name and Authentication username are your LocalPhone account number. Password is your LocalPhone SIP password.

In the Signalling tab, set DTMF to RFC2833 and add the ULAW and ALAW Codecs.

In the Advanced tab, set Insecure = ALL and Port = 5060.

Click SAVE when you’ve finished.

Wazo will not actually process incoming and outgoing calls through this LocalPhone trunk until you configure an outgoing route in IPBX:Call Management:Outgoing Calls and an incoming route using IPBX:Call Management:Incoming Calls if you have a DID. Outgoing and Incoming call routing are covered in separate tutorials.

Wazo Trunks Tutorial: Installing a Skype Connect SIP Trunk



Skype has been a free calling favorite for many of us for a very long time. Don’t confuse that history with Microsoft’s Skype Connect SIP trunks. These are pure SIP trunks that happen to be hosted by Microsoft under the Skype name. They’re not only not free. They’re expensive. Here’s a quick overview from Microsoft:

Skype Connect provides connectivity between your business and the Skype community. By adding Skype Connect to your existing SIP-enabled PBX, your business can save on communication costs with little or no additional upgrades required.

With Skype Connect, your business can make great value Skype calls and receive calls from your customers using your desk phones. Customers can also contact your business for free by using Skype from a browser with Skype buttons, by calling [not for free] the Skype business accounts associated with your SIP-enabled PBX, or [by placing PSTN calls to Skype Numbers you may have purchased].

Before you can make or receive calls from Skype Connect on your Wazo PBX, you’ll need to set things up on both the Skype side and the Wazo side. Here’s how.

Setting Up a Skype Connect Trunk with Microsoft

To get started, sign up for a Skype Manager account if you don’t already have one. It’s easy and it’s the only thing that’s free. Once you’re signed up and logged in, you’re going to need cash in your Skype credit account to get things going. $30 will get you started but finish reading before you invest.

First, click Features in the toolbar, choose Skype Connect and click Set up a SIP Profile. Give the profile a name "XiVO" and click Next. Next, choose the number of Channels you need at $6.95 per month. A channel gets you one simultaneous call in or out of Skype. Two channels gets you one call in and one call out simultaneously for $13.90 per month. You can take it from there but, sorry, you can only buy 300 channels at this time. You can also add the U.S. Minute Bundles if you make lots of calls in the United States.

Don’t buy your channels just yet. For now, cancel out of the dialog by clicking Back. Microsoft will set up your profile anyway:

The money deposited into your Skype Manager account will be needed to fund Skype Connect in three separate ways: (1) monthly payments for Channels at $6.95 each,1 (2) monthly payments for Phone Numbers associated with those Channels at $6.30 each, and (3) allocation of funds in advance to pay for outbound calls from each profile you create. You’ll need at least one phone number (a.k.a. DID) to receive any inbound calls from POTS phones to the Skype Connect SIP account on your Wazo server. You’ll also need at least one phone number before you can assign a CallerID to your outbound calls.2 Otherwise, they go out as Anonymous calls. Outgoing and incoming calls using traditional Skype Names are not supported!

Once you get your finances in order, it’s time to set up your SIP credentials for your new profile. Click on Authentication Details to display the dialog. Leave the Registration tab highlighted, and click on Generate a New Password, and a new SIP password will be sent to the email address you used to register when you set up your Skype Manager account.

Write down your credentials including the SIP password that was emailed to you by Microsoft. You’ll need them to configure your Skype Connect trunk on your Wazo server.

Configuring a Skype Connect SIP Trunk with Wazo

If you’ve already set up Incredible PBX for Wazo, then you can skip this first step which you should have already performed. In Services:IPBX:General settings:SIP Protocol under the General tab, make sure Match users with ‘username’ field is checked and SAVE your settings.

Next, create a new SIP trunk by click + Add after choosing Services:IPBX:Trunk Management:SIP Protocol. Make the settings under each of the tabs look like the following using your Skype Connect credentials. Then Save your settings.

Configuring an Outbound Route for Skype Connect with Wazo

Create an outgoing route for your Skype Connect trunk by clicking + Add after choosing Services:IPBX:Call Management:Outgoing Calls. Make the settings under the first two tabs look like the following. If you wish to use a dialing prefix for Skype calls other than 759 (S-K-Y), change the dial string accordingly. Save your new outgoing route.

Configuring an Incoming Route for a Skype Connect DID with Wazo

Create an incoming route for each of your Skype Connect DIDs by clicking + Add after choosing Services:IPBX:Call Management:Incoming Calls. Make the settings under the first tab look like the following. Enter the Skype Connect DID assigned to your account. Choose the destination for incoming calls to this DID based upon your specific requirements. Click Save when you complete the setup.

  1. Only in the Land of Micro$oft is a month equal to 27 days. If you multiply that by 12, you’ll see how an extra month of fees can be generated annually out of simple math "errors." []
  2. According to this article, phone numbers registered to your company can also be used as a CallerID number. []

Wazo Trunks Tutorial: Installing a FreeVoIPDeal (Betamax) SIP Trunk



FreeVoipDeal is one of dozens of Betamax companies that offer free international calling to dozens of countries in exchange for a modest deposit every four months. The deposit can be used to call everywhere else at very competitive rates. With FreeVoipDeal, for every 10 euros ($11.27) you deposit into your account, you get 300 minutes a week of free calls to 44 countries for 120 days. FreeVoipDeal provides outbound SIP calling only. The most recent Nerd Vittles article on FreeVoipDeal will tell you everything you ever wanted to know.

Setting Up Service at FreeVoipDeal and Obtaining SIP Credentials

1. Register for an account on the FreeVoipDeal portal.

2. The only credentials you will need to set up your Wazo SIP trunk are your username, password, and the CallerID number you would like to use to make outgoing calls. You must own the number under U.S. law.

Setting Up a FreeVoipDeal SIP Trunk on Wazo

As noted, you can only make outbound calls with FreeVoipDeal so there is no need to register your SIP trunk.

In the Wazo GUI, create a new FreeVoIP SIP Trunk by choosing IPBX:Trunk Management:SIP Protocol. Click on + Add to open a new template.

In the General tab, fill in the blanks using your FreeVoipDeal username and password. For the gateway address, use sip.freevoipdeal.com.

Skip the Register tab since registration is not required to make outbound calls.

In the Signaling tab, DTMF=RFC2833 and Monitoring=Yes. Customize Codec to G.711 u-law.

In the Advanced tab, Insecure=All and Port=5060.

Save your settings when done.

Wazo will not actually process outgoing calls through the FreeVoipDeal trunk until you configure an outgoing route in IPBX:Call Management:Outgoing Calls. Outgoing call routing is covered in a separate tutorial.