Fusion Hires Michael Fair as New Channel Chief

Fusion, a leading provider of cloud services, today announced the addition of Michael Fair to its team as Senior Vice President of Channel Sales and Alliances. Reporting to Dan Foster, Chief Revenue Officer, Fair is responsible for managing channel sales programs that drive revenue growth, as well as integrating and optimizing Fusion’s Partner Program following the company’s recent acquisitions.
Fair is an industry veteran with more than 30 years of experience creating, managing and integrating successful channel programs. Previously, he held key leadership roles at Charter, EarthLink, One Communications and Qwest. Most recently, Fair served as Managing Partner at MarketRace, a leading channel consulting company focused on ensuring the sales success of its clients through effectively designing, building and managing superior indirect channel programs and sales teams.
Fair’s hiring is a key step in Foster’s plan to build his sales and marketing organization and leadership team with a strong focus on the indirect channel. Fair will play a critical role in launching a new consolidated Partner Program as the company brings together the successful Partner Programs of Fusion, MegaPath and Birch following two recent acquisitions.
“A successful channel program is critical to Fusion’s ambitious plan to scale. As we combine our offerings, teams and programs following the recent acquisitions of MegaPath and Birch, it is essential that we build on each company’s existing success in the channel to create a unified sales and marketing organization with expertise that spans our combined companies,” said Foster. “Michael’s experience and knowledge of the industry make him perfectly suited to lead this effort and I am excited to welcome him to our sales team.”
“This is an exciting time to join the Fusion team,” said Fair. “Fusion’s partners continue to be at the forefront of our go-to-market strategy. I look forward to integrating Fusion’s Partner Program to deliver on our vision to serve as the industry’s most successful single source cloud provider.”
Fusion’s Partner Program offers total flexibility, product innovation, unparalleled support and programs to empower its growing partner network. To learn more about Fusion’s Partner Program, visit www.fusionconnect.com/partners.
About Fusion
Fusion (NASDAQ:FSNN), a leading provider of integrated cloud solutions to small, medium and large businesses, is the industry’s Single Source for the Cloud(R). Fusion’s advanced, proprietary cloud services platform enables the integration of leading edge solutions in the cloud, including cloud communications, contact center, cloud connectivity, and cloud computing. Fusion’s innovative, yet proven cloud solutions lower our customers’ cost of ownership, and deliver new levels of security, flexibility, scalability, and speed of deployment. For more information, please visit www.fusionconnect.com.
Fusion Contact:
Brian Coyne
1-212-201-2404
bcoyne@fusionconnect.com
Look for Fusion as exhibitor at ITEXPO Jan. 29 – Feb. 1, 2019 in Fort Lauderdale, Florida. Reserve your booth and sponsorship like DIDx and Fusion have now.

Emercoin Group Polozov & SIP Pulse Concalves | Mix of VoIP and Blockchain by OpenSIPS and DIDx

opensips emercoin enumer voip didx virtual phone numbersEmercoin Group’s Stan Polozov, with a strong background in the realm of finance and specializes in the technical aspects of blockchains and cryptography, talks with SIPPulse’s Flavio Concalves about blockchain projects during OpenSIPS Summit 2018 in Amsterdam.

Flavio asks Stan, “Let’s suppose I am an operator. I am sending a call to another operator. The other operator needs to charge 7 cents, not 7 cents, 7/10ths of a cent for each minute. Can we charge this directly using blockchain?”

Stan replies, “I think, uh…”

Flavio says, “Let’s say that they were supposed to buy coins to have a use of this network.”

“To publish the address?” Stan asks.

“They can publish but to complete the call on the other side…” says Flavio.

Watch the video to find out more of this fascinating conversation. Understanding the true potential of the blockchain-based technologies, Emercoin’s representatives, blockchain implementation specialist Oleg Khovayko and chief implementation officer Stan Polozov spoke about the unique ENUMER solution for VoIP at OpenSIPS Summit 2018.

Check out DIDx’s video of Stan of Emercoin team giving a presentation at OpenSIPS at https://youtu.be/kvlcYxeurK4. (Sippy Software‘s Max and DIDx’s Shaharyar edited these videos for the final products.)

 

Visit http://www.didx.net, https://www.opensips.org/events/index, and https://medium.com/@emer.tech.

Our website: www.DIDx.net.

To sign-up, visit https://www.DIDx.net/signup.

Our Official Facebook page https://www.Facebook.com/DIDxGlobal.

Our Official Twitter handle lhttps://www.Twitter.com/DIDxGlobal.

Our Linkedin page https://www.linkedin.com/company/didx

Who are we? Voiceland and DIDx.net! Clustering rulz.

Voiceland and DID do a look back at the latency of the 1990s to now… OpenSIPS’ clustering and high availability. There is much progress in VoIP. We recorded our video discussion via Zoom webRTC. A big part of our conversation is about how helping contact center startups is much of what both of our companies’ business.

Who uses Voiceland besides contact centers? Museums, taxi services, and load ferries are a few. During our talk with Vasilios , their founder and CTO, we leared they use no compression on end-to-end user calls for great quality.

Vasilios Tzanoudakis’s Internet telephony mentors include Mark Spencer, Daniel Mierla, Bogdan Iancu, Rehan Ahmed, the person who corrected the Zapata driver Jim Dixon. Suzanne mentioned hers are the same plus Stacy Higginbotham, Allison Smith, Anthony Minessalle, Brian West, Rich Tehrani and Jeff Pulver.

Jim Dixon Astricon 2008 Zapata and DIDx

The bravery of Mark, Daniel, Bodgan, Jim, Anthony and Brian to work hard on something that was brand new like voip …

“We owe them a lot,” says Visilios.

Emercoin Group’s Polozov was a presenter on blockchain at Open SIPs. “He was like from another planet,” says Vasilios. (We’ll share a video discussion about mixing blockchain and voIP between SIPPulse Flavio Concalves and Polozov on Youtube soon.)

Pretty cool though. Blockchain is something most people just want to understand how to make money from it. Extracredit.io realizes this and is obviously dealing with it be educating its community, so each person can more effectively take advantage of their cryptocurrency is something Suzanne believes.

In 2000, SegaNet released the first voice-chat-compatible browser for the Dreamcast. Internet services such as YahooChat! worked on the Java compatible web browsers with the ability of voicechat with the microphone, although it was already available for use in its HTML servers. This browser web integration became a standard in future game consoles. Vasilios and Suzanne talked about playing with different game systems that early on … integrated voice over Internet chat.

DIDx Suzanne Bowen and Voiceland Vasilios’s IP communications Part 1 and Part 2 discussion is ready for you to view on Youtube.

Ecommerce Giant Rakuten to Break up Japan’s 3 Part Oligopoly

telecommunications companies on the moveThe government is involved in the biggest Japan telecommunications shake-up ever. Until Rakuten proposed its plan to become an MNO in 2018, NTT Docomo (with over 40 million users), KDDI (48 million users) and SoftBank (with 39 million users), the Big 3 held what is sometimes called an organized oligopoly. How has and will Rakuten help to break up this “syndicate?”

Rakuten’s Rakuten Moble Network already has invested over 18 million USD to a pot that could total as much as 5.6 million to fully launch.

Other utilities’ companies that have the potential to assist Rakuten Mobile Network to achieve a complete Japan nation coverage include:

Chugoku Electric Power Company (CEPCO)
Chubu Electric Power (Chuden)
Hokuriku Electric Power Company (Hokuden)
Hokkaido Electric Power Company (HEPCO)
Kyushu Electric Power (Kyuden)
Okinawa Electric Power Company (Okiden)
Tokyo Electric Power Company (TEPCO)
Tohoku Electric Power (Tohokuden)
Shikoku Electric Power Company (Yonden)

According to Lumen Learning, “firms in an oligopoly may collude to set a price or output level for a market in order to maximize industry profits. … Collusive arrangements are generally illegal. Moreover, it is difficult for firms to coordinate actions, and there is a threat that firms may defect and undermine the others in the arrangement.”

And … according to Statista … “During the most recently reported period [late 2017], the number of members who had logged in at least once after registration amounted to approximately 95.2 million.’

That’s a good start to pull from as potential cellular subscribers. The Japan ministry of internal affairs and communications has approved of Rakutan’s plan.

Of course, DIDx is ready to assist Rakuten Mobile Network in making available local DID numbers of 130 nations for its customers. Such a move will be one more key resource to cutting costs, raising sales, competing better with the Big 3 and retaining customers.

Read recent news about KDDI, Softbank and NTT Docomo.

Distribute Data, Scale Capacity and Achieve High Availability with OpenSIPS 2.4 Release

OpenSIPS Summit 2018 Amsterdam, Netherlands | DIDx.net from Muntwo Productions on Vimeo.

The road map of OpenSIPS 2.4 is an ambitious one, addressing one of the most complex but vital capabilities of a SIP server. The ability to exceed the status of a single instance application in order to scale in a shape of a multi-node instance cluster.  The road map of OpenSIPS 2.4 is an ambitious one, addressing one of the most complex but vital capabilities of a SIP server. The ability to exceed the status of a single instance application in order to scale in a shape of a multi-node instance cluster.

What are the benefits?

A. Data distribution, capacity scaling

B. Geographical distribution

C. High availability as data and service

Being able to add clustering support services like registration, calling and presence requires complex and extensive work. Still, the team at OpenSIPS are almost at the end, just a day before the beta release.

Let’s get the status from the OpenSIPS developer team. Watch the video to learn more.

Bogdan Iancu, Liviu Chircu, Razvan Crainea and Vlad Patrascu describe their contributions and collaboration in the the exciting OpenSIPS 2.4 beta release that takes place March 28, 2018 and then the full release on April 30, 2018, one day before the OpenSIPS Summit 2018.

A big part of Liviu’s work was in the OpenSIPS and FreeSWITCH integration and total restructuring and revamp of the user location module. They are planning to build certain clustering modes into it to help you fit better with whatever platform you have.

Razvan is planning to bring some CGRates and RTP engine to OpenSIPS 2.4. (More information from his contribution soon!)

Vlad Patrascu says he’s been with the project for more than two years. His work is focused on improving the clustering under layer and to support important features like dynamic topologies and full data syncing. He’s implemented one model for distributed user location and highly available support for dialer tracking.

Discover the 2.4 capabilities from OpenSIPS Summit formal presentations on stage, in informal discussions in the lobby and via relaxing night-time activities. You’ll learn from interactive demos on building clusters and more. Be fully immersed in the one day workshop focused on clustering training. Be sure to follow OpenSIPS on Twitter, LinkedIN, Facebook and their blog.

International Telecoms Week since 2008 | DIDX Invite for 2018

DIDX.net has participated in International Telecoms Week every year since 2008. It’s a wonderful gathering of international professionals in telecom and IP comunications industry, often called ITW 2018. It attracts 6,500 of the most businesses from Americas, Asia, Europe, Australia, Africa and the Middle East.
Some examples are DIDX, Apple, Verizon, Samitel, CenturyLink, Telus, Telenor, Airtel, Facebook, Cox Business, Etisalat, Tawasal, and Google. They come from 140 nations, pretty much the same nations of which DIDX offers DID numbers. ITW 2018 is a win-win event with plenty of opportunities for developing business during the day and networking at night.DIDX meets Airtel at ITW!
This year 2018, this business-development rich event takes place 6-9 May at the  Hyatt Regency & Swissôtel Chicago, IL. Want to get the most out of International Telecoms Week? Sign up now like DIDX already has to get the best booth, the best meeting table and to get the most positive publicity as far in advance as possible.
Be sure to run the PCCW Global TSF Charity Run 5K with DIDX and other ITW attendees such as those fast runners from Telefonica, Sippy Software and PCCW. The funds go to great causes.
This year at ITW, learn about the IoT Boom for Carriers, Understanding Net Neutrality Implications, Age of Virtualization and Demands behind 5G! Visit internationaltelecomsweek.com now and register for a meeting table, booth, to give a presentation and to participate in all that ITW has to offer. Plus, DIDX will also be at Global Voice Congress / Digital Data Congress events on May 6 in the same area of Chicago. Be sure to schedule an appointment to meet with DIDX at sales at didx.net for both today.

Etisalat to Host World Teleport Associations Awards Event 2018 – Take a Peek at Finalists!

This is the property of etisalat. It does not belong to DIDX.WTA (World Teleport Association) announced in early March the finalists for its 2018 Teleport Awards for Excellence. From this group of finalists, WTA will name the winners at the 23rd annual Teleport Awards for Excellence Ceremony and Luncheon on March 13th during the SATELLITE conference in Washington, DC.

WTA selects its finalists from nominations submitted to the association by both members and non-members. So keep this in mind for 2018: even non-members can nominate companies that deserve to be recognized for their excellence in teleport, satellite and content distribution technolog The winning recipients are selected by a vote of the WTA Membership.

2018 Finalists for Independent Teleport of the Year

Santander Teleport (Spain) *

Telespazio (Italy) *

U.S. Electrodynamics – USEI (USA) *

2018 Finalists for Teleport Technology of the Year

Alusat, Always Up by Integrasys

DVB-S2X Wideband VSAT Modem Portfolio by Newtec

Satellite Network Virtualization with DataMiner Service and Resource Management (SRM) by Skyline Communications

In addition to honoring the winner from these two awards categories, WTA will present the 2018 Teleport Executive of the Year Award at the luncheon, hosted by Etisalat, a multinational Emirati based telecommunications services provider, currently operating in 16 countries across Asia, the Middle East and Africa. The over 150 attendees will include teleport, satellite and content distribution technology industry executives, as well as previous awards recipients from around the world. WTA’s Director of Development Louis Zacharilla will emcee the event, which is free for WTA members. Attendance is also available on a paid basis to non-members. Register at http://www.worldteleport.org/events/EventDetails.aspx?id=1031957.

*DIDX does offer DID number coverage of Spain, Italy, USA and 137 other countries. Join now at www.didx.net and begin a lucrative, new revenue stream that will enable you to gain and retain customers from any area of the world.

How to change the “Ring To” address at DIDX to your Asterisk SIP server

We’ll include a demo video here soon! Let’s learn more about configuring your Asterisk Server with DIDX. Today we will learn how to forward DID you have purchased via DIDX.net to your server.

For that, you will select the option “My Purchased DIDs”.

Here you will see all your purchased DIDs and their details. Now, in order to change the forwarding of your DID, just click on the “Ring To” address.

Next, set the “Ring To” address like this “DID@Your-Server-IP-Address”.

If you are using the default port that is # 5060, then there is no need to define it. If you are using any other port, then you have to define like “DID@Your-Server-IP Address:5060”. Here we are using the default port.

After completing the steps, click on “Update” button.

You have successfully configured the DID forwarding from DIDX.net to your SIP server. After that, you will want to configure SIP trunk on your Asterisk server. Thank you for enabling us to serve you. Welcome to DIDX DID number coverage of 140 nations, no pre-purchase required!

We will add a demo video to this blog post soon.

SIP clients typically use TCP or UDP on port numbers 5060 or 5061 for SIP traffic to servers and other endpoints. Port 5060 is commonly used for non-encrypted signaling traffic whereas port 5061 is typically used for traffic encrypted with Transport Layer Security (TLS).

Configure OpenSIPS to Receive Calls from DIDX DID Number Coverage

Hello to your new rich IP comunications business that includes OpenSIPS Open Source SIP proxy/server for voice, video, IM, presence and any other SIP extensions … multi-functional, multi-purpose signaling SIP server and DIDX direct inward dialing!


There is a misspelling of “Interop” at the beginning of this video. We apologize. We deeply appreciate Bogdan Iancu, founder of OpenSIPS for making it really easy to understand how to configure OpenSIPS to receive calls from DID of DIDX.net!

How To: Achieve Interoperability Between DIDX and OpenSIPS

 

In this tutorial, we will expand an existing OpenSIPS server configuration so that it will accept incoming traffic from a given list of DIDX servers.

 

To find out which particular DIDX IP address will send traffic for a given DID number, you may login to your DIDX account and visit the “DID INFO” page of that number. Alternatively, you may visit https://www.didx.net/pages/configs for the list of all DIDX IPs.

 

Regarding OpenSIPS, we assume that it is already running alongside an SQL database. Depending on your Linux OS, you may install and immediately start OpenSIPS from either https://apt.opensips.org or https://yum.opensips.org. For the database, a quick tutorial on how to import the OpenSIPS database schema into your SQL engine of choice is available at https://www.opensips.org/Documentation/Install-DBDeployment-2-4

 

The typical way to authenticate the SIP sender is via digest authentication (with username and password). As this mechanism is suitable for authenticating endpoints (users), it does not fit when comes to authenticate gateways, DID servers or other types of remote SIP Servers. For such purposes is it better to use IP authentication – the SIP sender is recognized and authenticated based on the source IP at the IP level.

 

Typically, most OpenSIPS config files (including the default config file) provide support for digest authentication, so we need to add to your config file the support for IP authentication in order to recognize and trust the calls sent by the DIDX servers.

 

First, open the /etc/opensips/opensips.cfg OpenSIPS configuration file using your favorite text editor. If OpenSIPS is installed from sources, the default path is /usr/local/etc/opensips/opensips.cfg. Within the initial section of the script, load the “permissions” module and configure a database URL for it:

 

loadmodule “permissions.so”

modparam(“permissions”, “db_url”,                  “mysql://opensips:opensipsrw@localhost/opensips”)

 

The “permissions” module is an in-memory storage for lists of IPs and network masks. We will use it to store the list of DIDX servers and validate all incoming calls against this list. With regards to the “validation” part, we only want to perform it when a call starts (i.e. initial INVITE receival). We recommend placing this check near your SIP digest authentication script code, and skipping the digest authentication altogether if the source is whitelisted:

 

 

if (!is_method(“REGISTER”)) {

if (check_source_address(“11”)) {

xlog(“Call from DIDX, skipping SIP digest authentication\n”);

} else if (is_myself(“$fd”)) {

# authenticate local subscriber

}

}

 

We apply the config file changes by restarting OpenSIPS:

 

opensipsctl stop

opensipsctl start

# or

/etc/init.d/opensips restart

 

Next, we provision the DIDX IPs under group 11 into the SQL database using opensipsctl. Note that you need to provision the DB support via opensipsctlrc file (typically under /etc/opensips/ directory):

 

# list the  current addresses from DB (should show nothing)

opensipsctl address show

# add the addresses one by one to DB

opensipsctl address add 11 198.199.87.53 32 0 udp

# list again to see all the addresses from DB

opensipsctl address show

 

Finally, we refresh the OpenSIPS “permissions” module cache with the new IPs:

 

# list the current in-memory addresses (should show nothing)

opensipsctl address dump

# instruct OpenSIPS to refresh in-memory cache with the DB content

opensipsctl address reload

# list again to see the addresses loaded from DB

opensipsctl address dump

And we’re done! OpenSIPS will now accept IP authenticated inbound traffic from the DIDX service. Awesome!

BTW, also don’t miss a single OpenSIPS Summit! Super informative, hands-on, welcoming event each year in Amsterdam, Netherlands.

Welcome to Asterisk Configuration with DIDX DID Coverage of 100 + Nations

Millions of individuals and businesses on planet Earth use a communications system in which the Asterisk and DIDX direct inward dialing are used for convenient, feature-rich, inexpensive calling. Here is how to configure the technical side of this empowering combination. (We will include a demo video on this blog that goes with this description soon.)

Let’s create SIP trunks in Asterisk with the IP addresses of DIDX SIP server. Use any
preferred editor to edit the “SIP.conf” file. The file path is “/etc/asterisk/sip.conf”.

Here you have to define the SIP trunks with the IP addresses of DIDX.

All the IP addresses of DIDX are mentioned on this link which is “www.didx.net/pages/asterisk”. If you are using USA DID, then the call can come from sip4, us1 and us2 addresses. If you are using UK DID, then call can come from eu2 and eu3 addresses. It is recommended to create the SIP trunk with all the IP addresses which are mentioned on this page.

Let’s create the SIP trunk with “sip4”.
The format for this follows:

[sip4]
type=peer
host=198.101.50.4
context=incoming

Let’s make a second SIP trunk of “sip8”.

[sip8]
type=peer
host=67.228.182.162
context=incoming

Similarly, in this format you will define the remaining IP addresses of DIDX.

Next, you will create an extension in this file. The format for creating the “10005” extension is like
this:

[10005]
type=peer
username=10005
secret=DIDx (Please make sure to use strong, complex password.)
host=dynamic
context=incoming
disallow=all (This will disallow all the codecs.)
allow=alaw (You will allow only “ulaw” and “alaw.”)
allow=ulaw

You have created the extension successfully in the file. Now save and exit the file.

Let’s go to the extension file using the preferred editor. The file path is same; just replace …
“sip.conf” with “extensions.conf”.

You will define your “Dial Plan” in this format where we are using DID 15672446030 as an example:

[incoming]
exten => 15672446030,1,Dial(SIP/10005)
exten => 15672446030,2,Hangup()

All incoming calls on this “DID” will land on this extension “10005”.

Instead of defining all the DIDs separately in the “Dial Plan”, you can customize your Dial plan like this:

exten => _X.,1,Dial(SIP/10005) (Here “X” means number from 0-9 and “.” means any length.)
exten => _X.,1,Hangup()

After this, save and exit the file.

Now you will go to Asterisk CLI to reload all your configuration. For that type “Asterisk -rvvvv”. Then hit Enter.

Now type “reload” and hit Enter. Your configuration is reloaded successfully.

Now you will register the extension on your softphone that you have created in your “sip.conf” file. You can use any softphone such as your own or Zoiper, X-lite, Jitsi, SwitchVox softphone, and Eyebeam because they are easily available and can be configured in no time. In our demo video, we use Zoiper.

In Zoiper, click on “Setting” and then use the SIP protocol. Click “next”.

Type your extension in the “User” box. Type the password that you have defined in your extension.

Last, type the IP address of your Asterisk server and then click “next”. Again “next”.

You will see: “Your account has been added to account list”.

Again, go to your Asterisk CLI. Check if your extension is registered on the softphone or
not by using this command “sip show peers”. Here you can see your extension has been registered.

Once your configuration is completed successfully, dial the DID that you have purchased and had provisioned from the DIDX marketplace, you will see the incoming call on this DID reach this extension successfully which is registered on your softphone.

Feel free to double-check the call on the “Asterisk CLI”. Welcome to your Asterisk server with DIDX direct inward dialing with a coverage of 100 + nations.