Get Clued In to FreeSWITCH via Fred, Man of Many Talents at Cluecon 2018 and on Youtube

Who does not admire Fred? He’s got that mesmerizing voice, and “voice” is much of what FreeSWITCH and the world of open source communications tech developers care about. DIDx invites you to subscribe to the FreeSWITCH with Fred YouTube playlist and learn from some exciting vidfred muteesa freeswitch cluecon picture by suzanne boweneo online workshops. Fred Muteesa and Joshua Young will be offer a free half hour in-person FreeSWITCH workshop during ClueCon July 23-26, 2018.

The workshops available on the FreeSWITCH with Fred YouTube channel workshops will include FS topics such as installing FS on a mikrotik router, Lua ESL (Electronic Structure Library), getting started with Verto (what’s behind FreeSWITCH’s Verto Communicator for WebRTC) and more. Josh and Fred will be around after the workshops during Cluecon to answer any questions you have about FreeSWITCH. No need to sign up in advance for FreeSWITCH with Fred. Just drop in.

Please direct your questions about FreeSWITCH workshops before, during or after Cluecon ClueCon, at the registration desk or by messaging sharon at freeswitch.com

Register today for Cluecon 2018.  Want to stay at the Cluecon venue of Swissotel? The deadline to get the Clueon fred muteesa james body freeswitch clueconhotel reservation discount is 6/30/18. Subject to availability, room rates are $245/night for a single or double, $275/night for a triple, and $305/night for a quad. Reservations can be booked online HERE and any pricing questions or concerns can be resolved by calling the Hotel at +1-312-565-0565 or 1-866-840-8096.

 

 

 

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FreePBX Configuration + DIDX DID Number Coverage of 100+ Nations = IP Communications Freedom

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.

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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.

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