Reason 1:
Let the DIDx wholesale DID marketplace help you kickstart your business with $25 worth of credit to activate your DIDx account.
Reason 2:
When you buy a minimum of 100 DIDs of USA, UK, and Canada, enjoy another 25% discount. (Conditions apply.)
With DIDx, you can become a telco or voIp operator in your country today. That’s exciting!
Contact sales at didx.net and get started right away!
A Pensacola, Florida headquartered VP of DIDX Suzanne Bowen, Karachi-based DIDX senior customer service exec Ali Muhammad and Australian James Hughes of Dove Gospel Global met up online via webRTC to discuss in short form how the nonprofit organization and DIDX work together to meet the needs of DGG and its community that is now as global as its name.
Dove Gospel Global global network director and pastor James Hughes says, “My heart is for a church without politics, and where Jesus moves in power and authority through his Holy Spirit.”
One major goal of Dove Gospel Global is to connect with religious leaders anywhere and of any religion. They searched for ways to make these leaders have an easy, efficient way to contact Dove Gospel Global, and for DGG to stay in touch. There are many communications issues such as poor cellular, landline and/or Internet networks in different areas of the world whether 3rd world, rural, emerging and also cost and security.
Because of the organization’s tireless search and testing of several solutions, they found happiness with a FreeSWITCH-based platform and DIDX for incoming calls. Now, the group has headquarters in Kampala, Uganda; Bhattiprolu, India; Lahore, Pakistan; Kent, United Kingdom; and with the main headquarters in Sydney, Australia. Anyone can call Dove Gospel Global via a local phone number of over 50 countries, and we have learned that those phone lines stay very busy. Call logs tell it all. Those 50 countries include USA, UK, Uganda, Turkey, Thailand, Sweden, Spain, South Korea, South Africa, Slovac Republic, Singapore, Russia, Rumania, Portugal, Poland, Pakistan, Nigeria, New Zealand, Netherlands, Mexico, Malaysia, Liberia, Latvia, Kenya, Japan, Italy, Israel, Ireland, Hong Kong, Ireland, Germany, France, Finland, Denmark, Czech Republic, Brazil, Belgium, Austria and Australia.
The number and scope of communications, new community members and collaborators and growth of their nonprofit movement in every way has grown and experienced more success, but that’s not to say, they have not had some tough times such as communications solution trials that never really worked, DIDs purchased from other virtual phone number service providers where price fluctuated, mostly upward, and service stopped for weeks at a time. The group’s growth was not matched by a flexible, scalable communications set-up. But once they developed in-house with paid guidance from a FreeSWITCH developer their own platform and integrated DIDX.net DID number service, they are now able to grow without the “growing pains.”
Watch Ali and James talk about Dove Gospel Global’s technological path to success.
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).
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.
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.
Whether, you are using 3CX, Asterisk, OpenSIPS, Kamailio, FreeSWITCH, Callision, Telinta, VoIPSwitch or other, DIDX is ready to help you move your DID number business and service forward to success. The following is set up a 3CX SIP server with DIDX for DID number provisioning.
After forwarding the DID to your SIP server from the DIDX portal, you are going to make a SIP trunk on 3CX web portal with the IPs of DIDX. We’ll add a demo video here on this blog post soon.
1. Select the option “SIP Trunks”.
2. Click on the “Add SIP Trunk” option.
3. Choose the “Generic” on “Select Country” section.
4. Select the “Generic SIP Trunk” on “Select Provider in your Country” option.
5. Type in your chosen DID number that you have bought from DIDX in “Main Trunk No”. For example: 15672441578. Then click on “Ok”.
6. After clicking the “Ok”, a new window will appear. On this page you will create the SIP trunk with all the IPs of DIDX. You can find these on this link “https://www.didx.net/pages/asterisk/”.
7. Copy the IP address and paste it on the section called “Register/Server/Gateway Hostname or IP”.
8. Please set the value of “Number of SIP Calls”. This means how many calls you want to receive on this DID simultaneously. Here we set the value 10.
9. If you want to add more DIDs, then select the “DIDs” option which is on the right of the “General” section.
10. Then click on the “Add DID” button, write the DID number and then press enter. If do not want to add more DIDs, let it remain the same.
11. Click on the “Ok” button. A new window will pop up and show the SIP trunk which is successfully created.
12. Select the “Extensions” option. After that select the “Add” button.
13. A new window will pop up. By default the 3CX will automatically provide the extension number. You can change the password that is given by the 3CX or you can use the same.
14. If you want to add the personal details, you can add that but it is not compulsory.
15. You will see that the extension is created successfully. In this case, it indicates the red dot because it was not registered on any softphone. The green dot would indicate that the extension is registered on the softphone.
16. Select the “Inbound Rules”. After that select the “Add DID” option. A new window will pop up.
17. Type any “Name” in the General area. But we are defining didx-incoming. Then select the DID to which you want to forward it.
18. In the “Route call to” section, you will see the Destination for calls during office hours and after office hours.
19. You can define one extension in both sections, if you want to receive call on your extension without the office hours restriction.
20. If your office is closed and you want to forward those calls to on Voicemail rather than softphone, then you can define voicemail extension as after office hours.
But here we are using same extension in both of the fields.
21. Click on the “Ok” button. Now the “Inbound rule” has been successfully define.
22. Open the extension on “Web Client” portal on 3CX or you can use the softphone. But for now we are showing testing on “Web Client”.
23. Test the call on the DID. We will able to receive the call on this extension which is “0000”.
24. In the demo video, we are able to receive the call on this extension. This means we have successfully set up the SIP trunk on the 3CX portal website! Welcome to DIDX!
Just back from ITEXPO East and completed new business development with top quality DID providers and really excited to announce new DID phone numbers added today that are available for our DID buyers to show as available to purchase and provision immediately. All DIDX buyers may use the DIDX API to show all DID numbers of up to 140 nations as available to their clients. They do not pay in advance for all these DIDs. They are billed from their account balance when their clients buy. That’s the reason so many DIDX buyers are growing fast.
Log in to your DIDX account today to take advantage of 120 new USA DIDs available of New York City, Cleveland,Philadelphia, Dallas Metro, Downtown Los Angeles, Stockton, Maine, Idaho; Champaign, Illinois; Duluth, Gary, Albany, Michigan, Akron, Cape Coral, San Antonio, Charleston, Washington, Frederick, Seattle, Central Alabama, Bridgeport, Central Texas, Tacoma, Rocky Mount, Charleston, Huntsville, Fort Wayne, Danville, Owensboro, Otsego, Philadelphia, Kenosha, Eugene,Toll-Free, Shreveport, Fresno, Fort Lauderdale,Washington,Philadelphia,Washington,Washington,Washington,New YorkCity, Philadelphia, Cincinnati, Silver Spring, Houston, and Pontiac.
In addition to new USA DIDs just added today and available for you to sell your clients: Sittingbourne (Southeast England) Mobile; St Petersburg, Moscow, Chelyabinsk, Russia; Indosat Mobile of Indonesia; Guatemala City, Guatemala; Cape Town, South Africa; and Prague, Czech Republic (largest City of Czech).
LinkedIn has a reputation for efficient use of social media to develop business. In general, people do not fill up their timelines with fluff. Most are genuine business invitations, advice, demos and discussions. After studying 1000s of of our customers and vendors’ LinkedIn profiles, we share some of our own tips.
We wish you well in business, and we definitely look forward to meet you online and offline to see how we can help you expand your business to up to 140 nations via something as wonderful and practical as local DID phone numbers. Visit http://www.didx.net to get started!
DID = direct inward dialing
Which conferences and exhibitions will DIDX partcipate in soon? OpenSIPS Summit May 1-4 in Amsterdam, Global Voice Congress May 6 in Chicago, ITW May 6-9 in Chicago and CommunicAsia June 26-28 in Singapore! Visit http://www.didx.net/events and message us to meet and discuss your business’s exciting accomplishments and unbelievable challenges!