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.
Let’s see firsthand how to configure a FreePBX server with DIDX. (We will add a demo video soon.)
Please log in to your DIDX account. Let’s forward the DID to a FreePBX server. First, select the “My Purchased DID”.
Here you can see all the DIDs you have purchased from DIDX. For forwarding the DID, click on the “Ring To” address.
In the “New SIP” section, you will want to define the forwarding in this format “DID@Server:port”. If you are using the “Channel SIP,” then you need to define the port “5160” in the forwarding. Otherwise, if you are using “PJ SIP,” then you don’t need to define the port in the forwarding.
After defining the forwarding, click on the update button.
You have successfully configured the DID forwarding. All the configuration is done on the DIDX portal.
Now, let’s create the SIP trunk on FreePBX. All the SIP server IPs of DIDX are available on this link “www.didx.net/pages/asterisk”. For configuring the SIP server on FreePBX, you will need to define it one by one. It is recommended to create the SIP trunk with all IP addresses on this link.
On the FreePBX server, just click on the “Connectivity” option, and then select the “Trunks” option.
Now click on the “Add Trunk” option, and then select the SIP (Chan_SIP).
Now fill up the “Trunk Name”. Here we are defining “DIDX”.
Now select the “SIP Setting” option. In that you need to define the “Trunk Name”. We are defining “didx-incoming”
For defining the peer details, you do not need to define the “username” and “secret”. In the host, just define the IP address of SIP server. After this, just click on the submit option. Then click on Ok.
After that, click on the “Apply Config”. Now the trunk has been created successfully. You will need to create the remaining IPs SIP trunks in the same manner that you have created this. After creating all the SIP trunks, just click on the “Application” option and last, select the “Extensions” option.
Now click on the “Add Extension” button. In this area, select the “Add new channel SIP extension”.
Here, you will need to define the “User Extension.” You can use any preferred digits. In this case, we are defining “1234”. In the display name, we are defining “didx”. Let’s make sure to choose a strong password, so that no one can access your extension. After that, just click the “Submit “button. Then click on the “Apply Config”. You have successfully created the extension.
Now you will need to define the inbound routes. Just click on “Connectivity” in that area and select the option “Inbound Routes”.
Then click on the “Add Inbound Routes”. Now you will need to define the DID that you have bought from DIDX in the “DID Number” section. Next, select the “Set Destination” which is 1234. Then click on the “Submit” button. Then click on the “Apply Config”. All the configuration of FreePBX is successfully created.
Whenever someone dials this DID, the call will land on this extension “1234”. Welcome to DIDX DID Number Coverage of 100 + nations with the awesome FreePBX. Feel free to sign up at http://www.didx.net to connect your FreePBX service to direct inward dialing coverage of millions of numbers. You don’t have to buy them before your customers buy them from you. Really freeing!
The “Free” in FreePBX stands for freedom. The site offers paid support, a wiki, video lessons and training classes. You can get involved by bug reporting, resolving bugs, becoming a certified ecosystem partner like Allision Smith and Vitelity, and more. Their site is powered by a free Atlassian Confluence Open Source Project License granted to FreePBX.org.
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).