Deploying Polycom phones for Cloud PBX with on-premise PSTN Connectivity Part 1

In the previous article we upgraded our directory synchronization server from the older DirSync to the latest version of AAD Connect to prepare our Skype for Business hybrid environment for Office 365 E5 Cloud PBX with on-premise PSTN Connectivity. In this article, we will configure our Skype for Business on-prem environment with a simulated PSTN connectivity using an Asterisk IP-PBX. We will create the necessary PSTN Gateway and Trunk from the Skype for Business side, and configure the corresponding SIP trunks and inbound/outbound routes on the Asterisk side. Then we will register a Polycom VVX600 and Polycom Trio8800 using accounts homed on the on-premise Skype for Business and make sure they are able to call extensions on the Asterisk IP-PBX. Once this is completed, we will move those accounts to Office365 E5 and test the voice functionality again to confirm that on-premise PSTN connectivity is working. Below is the diagram of the initial setup before moving accounts to Office365:

Section 1/2: Configuring Skype for Business for on-premise PSTN Connectivity

We first need to configure our mediation server pool to listen on TCP 5060. By default this is not enabled and the mediation server only listens on TLS 5067. To do this we use topology builder and modify the properties of the Front End Pool to enable the TCP listening port 5060 on the mediation server as shown below:

Next we use topology builder again to define a new PSTN Gateway which is fairly straightforward. A corresponding Trunk will be automatically created and we need to modify the properties of the Trunk to listen on TCP 5060 as well as define the mediation server port as TCP 5060 also:

After publishing the topology we actually need to run the "Setup or Remove Skype for Business Server Components" on each Front End server to enable the changes. Simply start deployment wizard and run this step for each front end server in the pool.

Once completed we proceed to create a new voice route for the PSTN Gateway (Asterisk PBX). Using the Skype for Business Server Control Panel, we add a new voice route and specify the matching pattern for calls going to. As shown below, the numbers matching the pattern "+656445XXXX" will make use of this PSTN Gateway:

When defining the voice route, we also need to specify the Associated trunks and make sure the PSTN Gateway defined earlier is selected. We also want to associate the PSTN Usages for this route, which in the lab is the "Local" usage which in turn is part of the "Global" voice policy. These are shown below:

Once the route is defined, we can go to the PSTN Usage tab and see that the "Local" usage contains the route that we just defined:

Next we define and associate a new normalization rule for 4-digit extensions that will be routed through our PSTN Gateway. In this lab, extensions starting as 3XXX will be routed to the PSTN Gateway, and later we will define these 3XXX extensions in Asterisk IP-PBX. We will add the "+656445" digits to the 3XXX extension so that the Skype for Business server will match the called number to the route going to the PSTN Gateway:

Next, we modify the Global Trunk Configuration to associate the "Local" PSTN Usage so that this trunk will be used when calling our 3XXX extensions. We need to disable Refer Support and disable Media Bypass as it is not known if Asterisk will support these options:

In the trunk configuration, we also need to define a Called number translation rule so that the trunk will strip off the +656445 digits and just send the 3XXX extension to the PSTN Gateway. This is because we are going to terminate the call from Skype for Business to Asterisk on an extension defined in Asterisk itself, which is simply a 3XXX extension rather than a full E.164 number:

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Section 2/2: Configuring Asterisk IP-PBX as a PSTN Gateway for Skype for Business on-premises

Now that we have completed the Skype for Business configurations, we proceed to configure the Asterisk IP-PBX. For our lab we are using FreePBX-5.211.65 which is bundled with Asterisk IP-PBX. The full name of the iso image is FreePBX-5.211.65-21-i386-Full-1416756098.iso and the Asterisk version is 11.14.1. This is not the latest version of FreePBX/Asterisk at the time of this post, as the I find that the older version is much simpler to configure and manage. The basic steps of installing and configuring FreePBX/Asterisk on Windows Server Hyper-V are already covered in this article http://www.ucprimer.com/tech-blog/lync-asterisknow-20-integration-guide and won't be repeated here. Therefore, only the key configurations will be mentioned here.

After the FreePBX/Asterisk installation is complete, we need to create a SIP Trunk for the Skype for Business Front End Pool which in this lab consists of 3 Front End servers. The FreePBX GUI will allow us to define a SIP Trunk to the first Front End server as shown below. In the Dialed Number Manipulation Rules, we defined a new rule for normalizing 4 digit extension dialing from Asterisk so that the full E.164 number will be sent over to Skype for Business Mediation Server. We also need to define the PEER Details as shown:
host=10.250.27.52                          // This is the IP address of the first FE Server
transport=tcp
type=friend
port=5060
insecure=very
fromdomain=10.222.210.70           // This needs to match the IP address of the Asterisk IP-PBX
context=from-internal
promiscredir=yes
qualify=yes
canreinvite=yes

After defining the SIP trunk to the first Front End server, we need to manually add the SIP Trunks for the other 2 FE servers by editing the /etc/asterisk/sip_addtional.conf file as shown below. If we do not do this, then only the first FE server will be recognized by Asterisk which results in calls from Skype for Business users homed on the other FE servers to fail.

After modifying the conf file, we need to restart Asterisk by running 'asterisk restart' from the shell and to see that the SIP trunks are operational, we run run 'asterisk -r' to enter the CLI and run 'SIP Show Peers". We should see an "OK" status for all our SIP trunks going to each FE Server as shown below:

Finally, we define the extensions 3001 and 3002 in the Asterisk and register a soft phone such as X-Lite to these extensions as shown below:

With the above settings the X-Lite client will register to Asterisk and we can start calling to/from Skype for Business clients. In this lab, our Polycom Trio has an extension 8800 and the VVX phone has extension 1135. From the X-Lite client we can call to/from these extensions as shown below:

This concludes the 1st part of this 2 part series on deploying Polycom phones for Office 365 E5 Cloud PBX with on-premise PSTN connectivity.