EzCloudInfo

After upgrade to Skype4Business incoming calls to phones might not show the original  caller ID (Might be hidden or unknown)

This is a small misconfiguration which we might have left out during the upgrade.

This is due to a value called Forward PAI which might be set to false

What is this Forward PAI ?

Its a value that  sends the  P-Asserted-Identity (PAI) header  along with the call. This P-Asserted-Identity (PAI) will have headers through which it will verify the original  identity of the caller.

When the call is being processed by the SIP network, a P-Asserted-Identity header will be part of all SIP messages for that complete call transaction (i.e. INVITE, ACK, BYE).

In-order to check this value in your settings you can run the below command

Get-CstrunkConfiguration

I just ran Get-CsTrunkConfiguration | Fl “*Forward*” to filter the appropriate value

In my case it was set to false. You have to set this value to True

Run the below command to set this value to true

Set-CsTrunkConfiguration -Identity Site:ExchangeQuery  ForwardPAI $True

You can enable this value through edit trunk configuration settings through control panel  also

I just explored the other below options as well and thought of adding them up in this blog itself .Below are them

Enable Media by Pass :

If we enable this option bypass will be attempted for all PSTN calls. You can enable this if there is a full connectivity strength between clients and PSTN gateways.Typically by enabling this option we can minimize the number of Mediation Servers deployed.This improves the voice quality by reducing the latency since the number of hops gets reduced.

Centralized Media Processing :

By enabling this Media bypass can improve voice quality by reducing latency, needless translation, possibility of packet loss, and the number of points of potential failure.Enabling Centralized Media Processing is a useful feature in that it allows the CircuitID Gateway to handle as much of the SIP responsibility as possible.

Enable forward call history: If we enable this value then all the call history information will be forwarded through the SIP trunk.

Enable RTP latching: Indicates whether or not the SIP trunks support RTP latching. RTP latching is a technology that enables RTP/RTCP connectivity through a NAT (network address translator) device or firewall.

Enable forward call history: Indicates whether call history information will be forwarded through the trunk.

Note:

All of the above will not be standard configuration setting in all deployments.For each UI setting in the Trunk Configuration we need to understand and plan accordingly to your PSTN connectivity , SIP configuration and your current Lync setup.

Hope this is useful

Thanks

Sathish Veerapandian

MVP – Exchange Server