IDT Express SIP Trunking On VitalPBX

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IDTExpress

With the aim of documenting how to integrate different VoIP providers with VitalPBX, today, we will talk about IDT Express, a division of IDT Corporation focused on delivering wholesale Voice and DID services from small to medium size businesses across the globe.

For testing IDT Express’s services, they give you a $25 trial, with this, you can buy a DID to perform or receive calls, so, you will be able to test all of its features before deciding to work with them or not.

Configuring IDT Express Portal

Something important to notice about this provider, is that it only works with IP authentication, so, the first thing you must do is to add on the switches section (Voice >> Switches) your PBX’s IP address.

Voice Management

The second step we need to perform is to purchase a DID number to receive and make calls, just as it is shown on the picture below. Once we have purchased the DID number, we must wait until the DID is available on the “My DIDs” section (DIDs >> My DIDs).

Purchase DIDs

To finish the configuration on the IDT Express portal, we will edit the trunk group assigned to the DID we have just purchased, and configure it with our PBX’s IP and the PJSIP port, as shown on the picture below.

Trunk Groups (IPs)

Configuring the Trunk (PJSIP)

To configure the trunk with this provider, we will use PJSIP, due that this new version of the SIP protocol has new features that can make our lives easier during the configuration of providers that have multiple originating and terminating IP addresses. So, we don’t need to create multiple trunks the way we once did with SIP.

In the case of IDT Express, we have two termination IPs (216.53.4.1, 66.33.147.149), so, we will use the contacts parameter on PJSIP trunks to define this IPs. Let’s check how our trunk must look like, in the picture below.

PJSIP Trunk

Something important to consider is that we have made this tutorial using VitalPBX 2.3.9, a release candidate version that we have launched a couple of days ago, and, that includes many improvements on PJSIP trunks.

This is the list of parameters we have configured in the trunk:

  • Trunk CID: YOUR COMPANY + YOUR DID
  • Overwrite CID: Yes
  • Local Username: IDTExpress
  • Contacts: sip:216.53.4.1:5060,sip:66.33.147.149:5060
  • Match:132.196.47,206.20.196.38
  • From Domain: YOUR PBX’s IP
  • Require Registration: No

With just those parameters above, we have successfully connected our VitalPBX with IDTExpress VoIP provider.

Configuring the Outbound Routes

IDT Express has a unique way for routing calls, because it comes with a feature called “Voice Divisions”. These voice divisions are different dialing prefixes that you may use to define what rates, and settings must be applied to your call. The available voice divisions are: Platinum, Gold, and, IDT Instant.

IDTExpress Voice Division

To configure these different voice divisions on our VitalPBX, we must create an outbound route as shown in the picture below.

Outbound Routes

With the dial patterns defined above you may dial any number in the following ways:

  • 13055605776 => Default Voice Division
  • 013055605776 => Gold Voice Division
  • 113055605776 => Platinum Voice Division
  • 213055605776 => IDT Instant Voice Division

Configuring the Inbound Routes

For finish, we are going to configure our DID (the one we’ve purchased at the beginning of this tutorial) on VitalPBX’s inbound routes, as shown in the picture below.

Inbound Routes

Conclusion

As you may see, IDT Express has a simple way to be configured with VitalPBX, some interesting features and a clean, and friendly user portal.

We hope this tutorial may help you on future implementations where you want to implement or try this VoIP provider.

mrivera

mrivera

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This project started with the objective of creating a system/interface for the administration of PBX systems based on Asterisk,  easy to use, totally adapted for different mobile devices (Fully Responsive Design), and with all the characteristics of an advanced telecommunications system; Combining the flexibility from Asterisk with concepts that have been satisfactorily used in traditional telephone systems, concepts that somehow were ignored by the new generations of IP telephony.

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