Configure Skype For Business Enterprise Voice With A SIP-Trunk (DeutschlandLAN SIP Trunk) and The Office Master Gate From Ferrari Electronic AG - Step by Step

12/9/2020 Configure Skype for Business Enterprise Voice with a SIP-Trunk (DeutschlandLAN SIP Trunk) and the Office
he Office Master Gate from Ferrari el…
log.matrixpost.net  Step-by-Step Tutorials  Con gure Skype for Business Enterprise Voice with a SIP-Trunk
(DeutschlandLAN SIP Trunk) and the O ce Master Gate from Ferrari electronic AG | Step by Step
Con gure Skype for Business Enterprise

Voice with a SIP-Trunk (DeutschlandLAN SIP
Trunk) and the O ce Master Gate from
Ferrari electronic AG | Step by Step
by Marcus Rath 17. December 2017
STEP-BY-STEP TUTORIALS
Today I want go through the steps to activate enterprise voice on Skype for
Business Server with a SIP Trunk from Telekom, DeutschlandLAN SIP-Trunk.
Add a Mediation Server to your Deployment
First we had to add a Mediation Server in the Toplogy Builder from Skype for
Business, in this case we add to the single Front End Server the Mediation
Server Role. Right click on Mediation Pools and select New Mediation Pool.
https://blog.matrixpost.net/configure-skype-for-business-enterprise-voice-with-a-sip-trunk-deutschlandlan-sip-trunk/ 1/41
12/9/2020 Configure Skype for Business Enterprise Voice with a SIP-Trunk (DeutschlandLAN SIP Trunk) and the Office Master Gate from Ferrari el…
Here put in a FQDN for this pool, in my case I use a single Front End Server
and had to put in the FQDN of this Frontend Server. Also select here “This
pool has one Server” as we used a single server deployment. Next you have
to select the existing Frontend- and Edge Pool.
The Mediation Server is responsible for inbound and outbound calls to the
Public Switched Telephone Network (PSTN) and dial-in Conferencing.
The listening ports depends on the SIP Trunk Provider or if you use a Session
Border Controller (SBC) like here, you must set the ports con gured on the
SBC.
Under https://windowsitpro.com/lync/demystifying-mediation-server-role-
lync-server-2013 is a good explanation about the Mediation Server Role.
If you have a single Front End Server Deployment, the server can handle up
to 150 calls whereas a standalone Mediation Server can handle up to 1.100
calls.
We deploy here the Mediation Server Role on the single Front End Server, in
this case we go to the settings of the Standard Edition Front End Server and
select at the left menue the point Mediation Server. Here select the
checkbox for Collocated Mediation Server enabled. The default Port for
encrypted tra c with TLS is 5067. You can also enable only TCP without
encryption, default Port here is 5060. This settings depends on your SBC. In
my case I use TLS Port 5067.
If you want to deploy a standalone Mediation Server you must right click on
Mediation Pools and select new Mediation Pool, here you can add one
standalone or multiple servers to the Pool.
Add a PSTN Gateway to your Deployment
Next we have to add a PSTN Gateway through which we route

inbound/outbound calls to and from our Skype for Business Server.

You must put in a FQDN for the SBC, it is not possible to enter the ip address
of the SBC, so you have to register this FQDN at your internal DNS server.
At this step also the trunk is created for this PSTN Gateway. Here we have to
set a name and the ports for this trunk, I used here the FQDN of the SBC but
you can use any name you want. The Trunk is the phone line with the sip
protocol. And this trunk connects the SFB with the SBC.
The ports must match the ports you set on your SBC!
Con gure your SBC
Next step is to con gure the SBC in your network or of course you can
con gure the SBC rst and than the SFB, it doesn’t matter which sequence!
In my case I use here the O ce Master Gateway installation image 4.1 from
Ferrari electronics.
https://www.ferrari-electronic.de/produkte/o cemaster-gate
Download
https://www.ferrari-electronic.de/downloads.html
Version 4.1 and above supports DeutschlandLAN SIP Trunk
This Image is based on Linux CentOS 6.8
I run this as a hyper-v vm
A good installation guide of the O ceMaster Gate you will nd under

https://www.y outube.com/watch?v=sWciI0hX1pg
First you have to con gure the network settings of the SBC. You can see the
actual ip of the SBC appliance from the console when you press i for info. 
Here I already had set a static IP from my test network. You can set this with
the O ceMasterGate Con guration utility which can run on a di erent VM in
your network.
O ceMaster Gate Kon gurations-Tool

https://www.ferrari-electronic.de/downloads/ les/320002/6.1.3/Setup-
O ceMaster-Gate-Con g-6.13.1102.exe
As you can see in the following gure, I already had intalled an older version
and now update to the actual 6.113.1102 version. I get a warning that my
installed service O ceMaster-Syslog must deinstalled and the new version
of it reinstalled in order to work with the new version of the con guration
utility.
With the O ceMaster-Syslog service you can debug the tra c of the
O ceMaster Gate when it doesn’t work as expected and is optional.
You may wonder to see the ISDN protocol in the logs when con gured a sip
trunk which uses VOIP with the SIP and RTP protocol, the reason is, that
ferrari electronics comes from the ISDN world and used on the SBC itself
for routing the calls, the ISDN protocol and translates it to VOIP when
forwarding the calls to the internal Skype for Business Server and also when it
forwards the calls to the SIP Trunk Provider. The bene ts are that ferrari can
use the existing code but will change this to native VOIP and SIP/ RTP
protocol in the future and further versions.
Now as we had installed the O ceMaster Gate con guration utility, we can
con gure rst the network settings of the SBC. 
Press the connect button and enter the ip address the Appliance get from
the DHCP Server and the password, default is omc and can be changed
over the console menue of the Appliance.
You can put the SBC to your internal network or your perimeter network, he
doesn’t need a public IP address assigned directly on the SBC. The SBC
established in my case of DeutschlandLAN SIP Trunk a connection from
internal to the SBC of the SIP Trunk Provider and only needs allowed tra c
outbound. If your Firewall allows all outbound tra c and do not lter the
tra c, you don’t have to change the con guration of the DeutschlandLAN SIP
Trunk, in this case Registered Mode is per default the best choice. If your
Firewall lters outbound tra c and you must allow which tra c can ow
outbound, best choice is to change DeutschlandLAN SIP Trunk to Static
Mode where the used outbound Ports can be set. You can change this
settings in the telefonie center https://telefoniecenter.t-online.de/fcc
If your internal SFB Clients not in same Network as the OMG, you should
modify the Routing on the OMG and add the Routes to the Client Subnets,
otherwise if Media Bypass is enabled, the OMG can’t route tra c to the
clients.
Con gure the SIP Trunk
Now we have to con gure the connection from the SBC to the SIP Trunk
Provider, in my case DeutschlandLAN SIP Trunk.
First the O ceMaster Gate needs to register the Trunk at the SBC of the
Provider.
This information you get from your SIP Trunk provider.
DEUTSCHLANDLAN SIP-TRUNK
https://www.ferrari-
electronic.de/downloads/ les/102004/2016/DE_QS_1TR118_w.pdf
The screenshots from Ferrari use for SIP Trunk Registration the TCP protocol
on port 5060 instead the secure TLS on port 5061. A documentation from
Telekom for the SIP Trunk DeutschlandLAN in combination with Skype for
Business is so far not available and the support told me that they had no
experience with this combination. They shipped this Trunk with a LANCOM
883 VOIP Router and had only a documentation and experience with this.
After doing a DNS NAPTR Record Lookup on the FQDN of the Registrar, I saw
that their was a SRV Record entry with SIPS (SIP over TLS). So I tried to
con gure as you can see on the screenshots above the registration for this

trunk and the voice routing resp. the trunk connection itself over tls and port
5061 (screenshots below) and it works perfect for outbound calls.
Unfortunately it works at the moment with Firmware 4.1 .380 (201 8-01 -05) not
for inbound calls over TLS and Port 5061 so far.
Ferrari electronic xed this problem with the DeutschlandLAN SIP TRUNK
and had an internal pre-release demo which works for outbound and
inbound calls over TLS and Port 5061. If you need this pre-release demo and
can’t wait for the next o cial release which will include this x, please
contact the hotline of ferrari electronic.
The connection from the O ce Master Gate to the internal Mediation Server
resp. Skype for Business FrondEnd Server works over TLS and Port 5067 or
what tls port you set on the Mediation Server. The only thing to keep in mind
that this works is to con gure a X.509 Certi cate on the O ce Master Gate. In
my case I also had a internal Microsoft PKI in my test network and
requested a certi cate with the CSR from the O ce Master Gate here. You
can also import a Root Certi cate to the O ce Master Gate to be sure that he
trusts the certi cate you con gured on the internal Mediation Server.
This root certi cate is only for trusting the certi cate from the internal
Mediation server. The certi cate from the O ce Master Gate again is only
important that your internal Mediation Server can establish a secure tls
connection to him. And of course you should be aware that the Mediation
Server trusts the certi cate on the O ce Master Gate.
You can check the connection over tls from the O ce Master Gate to the
Mediation Server with the Verify … button in the Certi cates menue
DNS NAPTR Records from the

DeutschlandLAN SBC
Here you can see my tests regarding the DNS NAPTR Records for the SIP
Proxy FQDN.
If you do a DNS SRV Lookup on the second NAPTR Record with the TLS SRV
Records you can see that for TLS on port 5061 three SRV Records are
registered.
Now after a DNS A-Record Lookup on the SRV Record with the lowest
priority we get the IP of one of the SBC from Telekom with the SIP
registration service on TLS.
Trying to connect to this service with telnet works as you can see

Con gure Rules on your SBC for Call Routing
After this you must con gure the Calls for the Trunk, here click on Change
Setttings …
Here you can see two network adapter symbols at the top, normally they
named PCM 1 an PCM 2 and comes from the history of ferrari and their
relation to the ISDN world. I changed this for a better understanding to Lync
and SIP, because the rst adpater is connected to the internal Skype for
Business Server and the second adapter is connected to the SIP Trunk
Provider. So calls to and from Skype for Business traverse to the rst adapter
and calls from and to the PSTN traverse to the second adapter.
For each Adapter you have to add two call processing rules, incoming and
outgoing rules.
Let’s do this for the rst Adapter PCM 1 in my case labeled SFB which is
responsible for the connection from the O ceMaster Gate to the Skype for
Business Server.

We need to add a rule for calls from ISDN (calls from the PSTN resp. SIP
Trunk Provider which converted to the ISDN protocoll from the O ceMaster
Gate).
This calls we route here to the internal Skype for Business Mediation Server
resp. the Mediation Server Role.
Protocol and Port must be the same as con gured on the Mediation
Server.
Also we need to con gure at this adapter a second rule for outbound calls
from our Skype for Business Server. Here you have to enter the IP Address
from the Skype for Business Mediation Server or Role.
These are VOIP SIP Calls from the internal Skype for Business Server which

were converted to ISDN from the O ceMasterGate and here terminated for
further routing to the SIP Trunk Provider for which the second adapter is
responsible.
Now we come to the con guration of the second adapter PCM 2 or in my

case labeled SIP.
We must also con gure here two call rules, one for calls from the
O ceMaster Gate which are converted into ISDN to the SIP Trunk Provider
and reconverted in VOIP SIP and one for all incoming Calls from the SIP
Trunk Provider which rst must converted from the O ceMaster Gate into
ISDN protocol.
First rule is for calls from the OMG to the SIP Trunk Provider. Since OMG
Version 4.1 you can select the DeutschlandLAN SIP Trunk 1TR118 Pro le, on
which all parameter con gured for this trunk or many other SIP Trunk
Provider Pro les. In my case I need the DeutschlandLAN SIP Trunk Pro le. As
we use TLS as discussed above we need to set the protocol to TLS and the
port to 5061. The FQDN of the registrar is reg.sip-trunk.telekom.de.
The second rule are for all incoming calls from the SIP Trunk Provider. Here
you had only to select the Provider Pro le in my case the DeutschlandLAN
SIP Trun which also set the correct paramters for the incoming VOIP calls.
Con gure Enterprise Voice in Skype for

Business
Now after con guring the SBC and the connection with Skype for Business
Server, we have to switch to the SFB Control Panel to con gure the rest.
First we need to enable the users for Enterprise Voice. You can enable this
in the user menu.
Also you need to enter the telephone number and the extension number in
Germany the MSN (Multible Subscriber Number) number. in the E.164 format
https://en.wikipedia.org/wiki/E.164
Skype for Business Server needs to know how to route calls outside to the
PSTN. Therefore we go to the Voice Routing menue in the control panel.
You can edit the Global Dial Plan or create a separate Dial Plan which I
prefer. In case of multiple o ce locations you can create here for each
location a separate Dial Plan and the corresponding normalization rules.
In my test environment I only had one SIP Trunk with two lines and one
phone number block so I only need one Dial Plan for the location in
Stuttgart.
At this location I created 5 normalization rules, the rst for international calls
outside germany, the second for calls within germany, the third for calls
within Stuttgart so you do not have to dial the area code +49 711, the fourth
rule are for calls within the company at this location and the last rule do not
normalize the dialed number.
International
Pattern to match: ^00(\ d{2}\ d+)$
National
Pattern to match: ^0(\ d{3}\ d+)$
Ortskennzahl Stuttgart
Pattern to match: ^(\ d{3}\ d+)$
Intern Stuttgart

Pattern to match: ^(\ d{1 })$
Keep All
Pattern to match: ^(\ d+)$
We also need to modify the Global Voice Policy or create a separate one as I
did. If you want to allow di erent features or PSTN Usages for di erent
locations or users, you can create more Policies.
I named it like the SIP Trunk.
As you can see I created one PSTN usage record and named it “Allow all
Calls” and so I added all created Routes to this so that all users are allowed
to use all routes. Over these PSTN usages you can control which routes the
users can use or are allowed to. Before you can add here the routes you
must rst create them, you will see this at the next step.
The Routes are created automatically from the Associated Normalization

Rules you added to the Dial Plan Policy.
Route for Interne Durchwahl Stuttgart
Route for Ortskennzahl Stuttgart
Route for National
Route for International
Next step is to con gure in Voice Routing the register Trunk Con guration,
here we can set some further options for the SIP Trunk. I will add in the
Global Policy the PSTN Usages Allow all Calls which I created previously.
Don’t forget to select the con gured Voice Policy and Dial Plan Policy in the
steps before for the users who should use this policy and should be able to
make calls to the PSTN.
Now Users are able to call from Skype for Business to the PSTN Public
Switched Telephone Network and get calls from.
I will describe all the settings and options more in detail when I have time and
also the normalization rules to translate the dialed numbers from the users
into correct E.164 numbers.
Links
Palaver mit Telekom Deutschland zu SIP-Trunks

http://netzpalaver.de/2017/08/08/palaver-mit-telekom-deutschland-zu-
sip-trunks/
Grundlegende Informationen zu STUN und NAT bei SIP-Anschlüssen der

Deutschen Telekom (DeutschlandLAN)
http://faq.bintec-elmeg.com/index.php?
title=Grundlegende_Informationen_zu_STUN_und_NAT_bei_SIP-
Anschl%C3%BCssen_der_Deutschen_Telekom_(DeutschlandLAN)&mobileacti
on=toggle_view_desktop
Tags In
DEUTSCHLAND LAN ENTERPRISE VOICE FERRARI ELECTRONIC
OFFICE MASTER GATE SFB SIP SIP TRUNK SKYPE FOR BUSINESS
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12/9/2020 Configure Skype for Business Enterprise Voice with a SIP-Trunk (DeutschlandLAN SIP Trunk) and the Office

he Office Master Gate from Ferrari el…

Con gure Skype for Business Enterprise

Add a Mediation Server to your Deployment

Add a PSTN Gateway to your Deployment

Next we have to add a PSTN Gateway through which we route

Con gure your SBC

Version 4.1 and above supports DeutschlandLAN SIP Trunk

This Image is based on Linux CentOS 6.8

I run this as a hyper-v vm

A good installation guide of the O ceMaster Gate you will nd under

O ceMaster Gate Kon gurations-Tool

Con gure the SIP Trunk

This information you get from your SIP Trunk provider.

DNS NAPTR Records from the

Con gure Rules on your SBC for Call Routing

Now we come to the con guration of the second adapter PCM 2 or in my

Con gure Enterprise Voice in Skype for

Pattern to match: ^00(\ d{2}\ d+)$

Pattern to match: ^0(\ d{3}\ d+)$

Pattern to match: ^(\ d{3}\ d+)$

Pattern to match: ^(\ d{1 })$

Pattern to match: ^(\ d+)$

I named it like the SIP Trunk.

The Routes are created automatically from the Associated Normalization

Route for Interne Durchwahl Stuttgart

Route for Ortskennzahl Stuttgart

Route for National

Route for International

Palaver mit Telekom Deutschland zu SIP-Trunks

Grundlegende Informationen zu STUN und NAT bei SIP-Anschlüssen der

DEUTSCHLAND LAN ENTERPRISE VOICE FERRARI ELECTRONIC

Set up a Central Log Server with Migration from onPremise to O ce

WLAN RADIUS Authenti zierung

OFFICE 365, TROUBLESHOOTING

OFFICE 365, SESSION BORDER CONTROLLER, STEP-BY-STEP TUTORIALS

LINUX, STEP-BY-STEP TUTORIALS, TROUBLESHOOTING

ACTIVE DIRECTORY ASSEMBLY AZURE AZURE AD BGP C#

ENTERPRISE VOICE EXCHANGE EXCHANGE HYBRID CLASSIC FULL

EXCHANGE ONLINE GAC GLOBAL ASSEMBLY CACHE IKEV2 IPSEC LYNC

MIGRATION MYSQL OFFICE 365 OFFICE MASTER GATE OPENBGPD PAAS

PFSENSE PKI POLICY-BASED POWERSHELL RADIUS RDS

ROUTE-BASED S2S SFB SIP TRUNK SITE-TO-SITE SKYPE FOR BUSINESS

SQL MANAGEMENT STUDIO SQL SERVER SQL SERVER DATA TOOLS

SQL SERVER INTEGRATION SERVICES SSDT SSIS SYSLOG SYSLOG-NG

TROUBLESHOOTING UBUNTU VNET VPN

SESSION BORDER CONTROLLER

OFFICE 365, TROUBLESHOOTING

OFFICE 365, SESSION BORDER CONTROLLER, STEP-BY-STEP TUTORIALS

LINUX, STEP-BY-STEP TUTORIALS, TROUBLESHOOTING

ACTIVE DIRECTORY ASSEMBLY AZURE AZURE AD BGP C#

ENTERPRISE VOICE EXCHANGE EXCHANGE HYBRID CLASSIC FULL

EXCHANGE ONLINE GAC GLOBAL ASSEMBLY CACHE IKEV2 IPSEC LYNC

MIGRATION MYSQL OFFICE 365 OFFICE MASTER GATE OPENBGPD PAAS

PFSENSE PKI POLICY-BASED POWERSHELL RADIUS RDS

ROUTE-BASED S2S SFB SIP TRUNK SITE-TO-SITE SKYPE FOR BUSINESS

SQL MANAGEMENT STUDIO SQL SERVER SQL SERVER DATA TOOLS

SQL SERVER INTEGRATION SERVICES SSDT SSIS SYSLOG SYSLOG-NG

TROUBLESHOOTING UBUNTU VNET VPN

© 2020 matrixpost Imprint | Privacy Policy

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