How to Interdomain Dialing on the Vodia Phone System
Published on:
July 14, 2023
Vodia gives users the ability to dial other tenants within the system using full NANPA numbers, extensions or extension alias numbers. To access this feature, a multi-tenant license is essential; it empowers you to establish multiple tenants within the system, each resolving to its designated domain. This functionality streamlines communication and fosters an interconnected network within the Vodia Phone System, ensuring efficient and flexible calling across tenants.
Vodia gives you the ability to dial across the phone system to other tenants with a full NANPA number, or with an extension or an alias number for the extension. To use this feature you will need to be running the multi-tenant license; this will enable you to create multi-tenants and have them resolve to that particular domain. For those familiar with the Vodia PBX, here's how to set it up properly.
Dial Plans
Vodia uses its dial plan to call outbound - it's usually paired with a SIP trunk but, in this case, we will use the "Loopback" mode”
Optional: You could deploy the same dial plan you already have on the system, but you will still need to create the "Loopback" in your dial plan.
If you want other users to be able to dial across domains, you will need to create the entry in the dial plan; if it's a global dial plan you can just create it there.
Assuming you already have some entry information in your dial plan, you will need to specify the preference.
Order of Operation
Your dial plan has to be clear: do you want to dial another tenant number without leaving the SIP trunk? If so, you will have to define when you want the call to terminate to a SIP trunk.
In our test, if anyone dials a tenant number, it should go to a local SIP trunk which points to the local IP; in this case it's 127.0.0.1 (see below).
As you can see, we have created a dummy number on compro.audiomercy.com on extension 4001; in this screenshot extension, 4001 from pbx.audiomercy.com calls number 781-555-6666 on pbx.compro.audiomercy.com.
Try the following:
Create a SIP trunk and rename it. Here are the settings you will need to activate:
Name to interdomain "keep it simple" ;)
Type: SIP Gateway
Global Enabled
Inter-office - Trunk settings set to yes
Dial plan for outbound calls: here you can choose any dial plan; for now we will need to create a new dial plan, so let's continue building the trunk
Registration settings
Under proxy enter 127.0.0.1
Routing/Redirection as is
Number / Call Identification
Request: URI "let the system decide" which is the default
From "Based on incoming call"
To " Same as Request - URI
Remote-Party-ID "Based on incoming call"
Everything else stays as default
Save
Dial plans
For the dial plan, you can create a new plan or use your existing one
For the Trunk: use "Try Lookback"; for the pattern use * and pattern *
You will need to go back into the SIP trunk and choose the new dial plan under the setting "Dial plan for outbound calls"
Vodia’s cloud PBX system enhances the Do-Not-Disturb (DND) function for hotels, allowing guests to manage their privacy through buttons in the room or on the phone. The system syncs with PMS for seamless functionality, enabling hotel operators to manually override DND when guests struggle to disable it. For critical situations, like room service follow-ups, the system allows specific extensions to bypass DND, ensuring guests are reachable when needed. VIP guest management features, guest-to-guest call restrictions and automatic resets upon checkout ensure privacy, security, and convenience, improving the guest experience while maintaining operational efficiency.
As businesses increasingly adopt Wireless Local Area Networks (WLAN), understanding its impact on Voice over IP (VoIP) is essential. While WLAN offers flexibility and mobility, it can lead to call quality issues due to packet loss and bursts during access point switching. To combat these challenges, organizations can utilize robust codecs like OPUS which are designed to handle packet loss effectively and employ Session Border Controllers (SBCs) to enhance jitter buffers. By incorporating these solutions, companies can ensure a more reliable VoIP experience that meets the demands of modern workplaces, allowing seamless communication without interruptions.
In the late 1980s, prank calls were a common form of entertainment but often caused confusion, similar to today’s call forwarding glitches. Recently, a client migrating to a new PBX system experienced disruptions when users mistyped numbers for call forwarding. While user errors are common, there's a bigger concern: calls being misdirected to emergency numbers or costly lines, especially since many VoIP phones lack security features. Solutions include restricting call forwarding to internal numbers, blocking access or allowing only specific, pre-approved numbers. Ultimately, balancing control over settings is essential to prevent glitches while keeping systems user-friendly.
.svg)
