Information

Answer

To configure Switchvox to work with the Digium SIP Trunking service, open a web browser and log into your Switchvox Admin web interface. The following is a step-by-step tutorial to show you the way.

NOTE: If you do not need a tutorial, go to Quick Setup at the bottom of this article.

Tutorial

Log into your Switchvox PBX admin account.

Select Setup from the main screen.
Select VOIP Providers from the Setup screen.
Select Create SIP Provider from the VOIP Providers screen.
Input SIP Provider information.

Fill out the form as indicated above.
Username and Password are provided by DCS in the Digium SIP Trunking Order Details email.
Do NOT select Save SIP Provider button yet.

Select the Peer Settings tab.

User-added image

Confirm Host Type is set to Provider, which is the default setting.
Apply Incoming Call Rules to Provider should be set to YES, which is default.

Select the Connection Settings tab.
Proxy Host should be set as sip.digiumcloud.net
Always Trust this Provider should already be set to YES.
Change Qualify Hosts to YES.
Choose the Call Settings tab.
Confirm Audio Codecs you will be using are selected. ULAW and ALAW are defaulted ON. If you have purchased G.729 for Switchvox, change the codec to ON

Click Save SIP Provider button to save all entered information.
The screen below confirms a SIP Provider has been successfully added. Select the Go To Connection Status option. This allows you to see if your SIP trunk has successfully registered to the DCS network.

After all the VOIP Providers info has been properly configured, the Connection Status screen will display a Registered state.

If the VOIP Provider state does not immediately show a state of Registered, give the system 60-90 seconds to register. The connection status can be checked at anytime by accessing Server -> Diagnostics -> Connection Status menu or simply by refreshing your current Connection Status screen using your web browser’s refresh button.
If your Switchvox system sits behind a router that is performing NAT, then you will need to change an additional Network Setting in the system. Enable Allow NAT Port Forwarding in the Server -> Networking -> IP Configuration menu.

Input the External IP Address of your router. If you are not sure what your public IP is, click Look Up External IP to automatically find it.

In this screenshot, xxx.xxx.xxx.xxx should be replaced with your actual external IP address.

If you changed Allow Nat Port Forwarding and External IP Address, you will need to choose Save IP Configuration at bottom of page.

Note: Additional info concerning NAT configuration can be found at Purpose of Allow NAT Port Forwarding in IP Configuration

Your Switchvox system should now be registered to Digium Cloud Services SIP Trunking services. If you need assistance, please contact Digium Technical Support at 877.344.4861.

Note: If this is a new install, additional Switchvox configuration is required to complete your installation. This includes, but is not limited to, outgoing call rules, incoming call rules, IVR, voicemail setup, creation of extensions, etc.

Quick Setup

Point your web browser to your PBX web interface. The following is a sample login :

https://IP_OF_YOUR_PBX/admin

Then enter configuration values in the following areas of the interface as described below:

VOIP Providers

Go to Setup > Call Routing > VOIP Providers and click Create SIP Provider.
Add values as indicated under the following tabs:

SIP Provider Information tab

SIP Provider Name: Enter your provider name.
Your Account ID: Enter your_digium_username without @sip.digiumcloud.net; Switchvox will report the length of the account ID is too long if you use the extended name.
Your Password: Enter your_digium_password.
Hostname/IP Address: Enter sip.digiumcloud.net.
Callback Extension: Select an IVR or Extension or any other destination to which you want to send Inbound Calls on this Digium SIP Trunk.
Default Fax Extension: NA.
DTMF Mode: Use the default.

Caller ID Settings tab

Supports Changing Caller ID: Set to YES.
Caller-ID Method: Use the default.
Caller ID Name: NA.
Caller ID Number: NA.

Call Settings tab

ULAW: Set to ON.
ALAW: Set to ON.
G722: Set to ON.
G729: Set to ON if you have purchased the G729 add-on.
Always Send Early Media: Set to YES.

Incoming Calls

After a trunk has been created, create an Inbound Route to handle calls coming from the Digium SIP Trunking service to your Switchvox system and make these configuration changes by going to Setup > Incoming calls and click Create a Single DID Route. Add values as indicated below:

Rule Name: Enter calls-from-your_digium_number.
Incoming DID to Match: Enter your_digium_number; for example, 2565551234
Incoming Provider: Use the default.
Incoming Call Type Use the default.
Extension to Route Call: Select an IVR or Extension or any other destination to which you want to send Inbound Calls on this Digium SIP Trunk.

Outgoing Calls

Probably you already have several outgoing call rules set up on your system. The following steps guide you through the process of changing the provider that Switchvox uses to place the outbound call. Go to Setup > Outgoing Calls.

From the list of call rules, click the pencil icon (under Actions) of the rule you wish to modify.
Primary Call Through Provider: Change to Your_digium_sip_trunk_provider.
Click Save Outgoing Call Rule.
Repeat the process on the other outgoing calls that you wish to dial through your Digium SIP trunk

NOTE: Digium Trunk servers accept 10 or 1+10 digits dialing. If you wish to dial 7 digits, add your area code in the section that prepends the digits to the number in the call rule.

Title What is the recommended Switchvox configuration to connect to DCS SIP Trunking

URL Name What-is-the-recommended-configuration-for-a-Switchvox-PBX-to-connect-to-DCS-SIP-Trunking

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