Configure Skype for Business Server SIP Trunk with Telnyx

This document includes a step-by-step for configuring Skype for Business Server SIP Trunking with Telnyx.

  1. Start Skype forBusiness Topology Builder: Click Startand type Skype for Business Server. Then click on Skype for BusinessServer Topology Builder.

2. Under Skype for Business Server, your site name, then Shared Components,right-click the PSTN gateways node, and then click NewIP/PSTN Gateway.

3. In Define New IP/PSTN Gateway, type,and click Next.

4. In the next page leave IPv4 – Use allconfigured IP addresses (default), andclick Next.

5. Define a root trunk for the PSTN gateway:
 -UnderListening Port for IP/PSTN Gateway, type 5060.
 -UnderSIP Transport Protocol, click TCP,and then click OK.
 -UnderAssociated Mediation Server, select the Mediation Server  pool toassociate with the root trunk of this PSTN Gateway.
 -Under Associated Mediation Server port, type the listening port that the Mediation Server will use for SIP messages from the gateway (5068 by default).
 -Be sure that the peer you defined is running and using the FQDN    or  IP address that you specified. Then click Finish.

6. Right-click the Skype for Business Server node, and then click Publish Topology.

7. To finish you must create your Dial Plan, Voice Policy, Route, PSTN Usage and Trunk Configuration. We recommend that you create a custom configuration based on your location and company policies, but we leave this script as an example to start with a basic configuration. Please use it only if you have a default environment with no voice settings.
You must copy and paste the following text into afile editor, save it with extension .ps1 and run it on Skype forBusiness Management Shell:

Write-Host "Starting Telnyx Enterprise Voice example settings..."

#Add PSTN Usage
Write-Host "Adding PSTN Usage..."
Set-CsPstnUsage -Force -Usage @{add="US-Basic-PSTN-Usage"}  | Out-Null

#Set Dial In Conference Region
Write-Host "Setting Dial Plan..."
Set-CsDialPlan -Identity "Global" -SimpleName "Global" -DialinConferencingRegion "US"  | Out-Null

#Add Normalization Rules
Write-Host "Adding Normalization Rules..."
New-CsVoiceNormalizationRule -Name 'US-National' -Parent Global -
Pattern '^1?([2-9]\d\d[2-9]\d{6})\d*(\D+\d+)?$' -Translation '+1$1' -
Priority 0 -Description "National number normalization for United States" -WarningAction:SilentlyContinue | Out-NullNew-CsVoiceNormalizationRule -Name 'US-Service' -Parent Global -Pattern '^([2-9]11)$' -Translation '$1' -Priority 1 -Description "Service number normalization for United States" -WarningAction:SilentlyContinue | Out-NullNew-CsVoiceNormalizationRule -Name 'US-International' -Parent Global -Pattern '^(?:\+|011)(1|7|2[07]|3[0-46]|39\d|4[013-9]|5[1-8]|6[0-6]|8[1246]|9[0-58]|2[1235689]\d|24[013-9]|242\d|3[578]\d|42\d|5[09]\d|6[789]\d|8[035789]\d|9[679]\d)(?:0)?(\d{6,14})(\D+\d+)?$' -Translation '+$1$2' -Priority 2 -Description "International number normalization for United States" -WarningAction:SilentlyContinue | Out-Null

#Set Voice Policy
Write-Host "Setting Voice Policy..."Set-CsVoicePolicy -Identity Global -AllowCallForwarding $true -AllowPSTNReRouting $true -AllowSimulRing $true -EnableBWPolicyOverride $false -EnableCallPark $true -EnableCallTransfer $true -EnableDelegation $true -EnableMaliciousCallTracing $true -EnableTeamCall $true -PstnUsages @{add="US-Basic-PSTN-Usage"}  | Out-Null

#Add Voice Route
Write-Host "Adding Voice Route..."New-CsVoiceRoute -Identity "US-Basic-Voice-Route" -NumberPattern ^+ -PstnGatewayList @{add=""} -PstnUsages @{add="US-Basic-PSTN-Usage"} -WarningAction:SilentlyContinue | Out-Null

#Update Trunk Configuration
Write-Host "Updating Trunk Configuration..."Set-CsTrunkConfiguration -Identity Global -MaxEarlyDialogs 20 -SRTPMode "Optional" -ConcentratedTopology $true -EnableReferSupport $false  | Out-Null

#Job is done
Write-Host "Configuration is done!"

IMPORTANT: Before running this script, make sure you've defined a PSTN gateway in your topology for the selected site, and that you've backed upyour existing Enterprise Voice configuration

Configure firewall
You must allow this traffic on your firewall to work with Telnyx SIP Trunk:

Cloud Connector Edition
If you need to deploy CCE (Cloud Connector Edition) you should follow the steps in the next article:
To connect CCE to the Telnyx SIP Trunk you must specify the following information in the .ini file:

-Voice Gateway 1 Make and Model = Telnyx Telephony Engine
-Voice Gateway 1 Name =
-Voice Gateway 1 IP Address =
-Voice Gateway 1 Port # = 5060
-Voice Gateway 1 Protocol for SIP Traffic = TCP
-Enable REFER support = $false
-Remove the section in the .ini file for the second gateway

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