Setting Office Communications Server R2 with a Mitel 3300

I've noticed some people trying to setup an OCS R2 Environment with a Mitel 3300 PBX, please note that this configuration is not in the supported list. However it is still feasible for you to connect your 3300 to OCSR2 via a Mediation server to test the functionality of your OCS R2 environment being able to make and take calls.

Once you want to go live with the solution, it is best to follow the Microsoft best practices and install a gateway between the 3300 and your mediation server so that it is a supported environment.

So all that being said here is what you need to do, I have and had this setup with release 9 and release 10 (MCD 4.0 of the 3300)

How to connect a 3300 to an OCS mediation server:

How to connect a 3300 to an OCS mediation server:

Release 9.0
1. Create a SIP network element to point to the IP address of your mediation server
2. Create a Class Of Service, which has Public Access via DPNSS Enabled
3. Create a Trunk service assignment
a. Ensure your Baud is set to 9600
b. Dial in Trunks Incoming Digit Modification Absorb 0
4.
5. Create a SIP Peer Profile , note that the number of simultaneous calls cannot exceed your licensed number of sip trunks.

SIP Peer Profile Label:

SIP Peer Profile Label: Mediation
Network Element: MEDIATION

Local Account Information
Registration User Name:
Address Type: IP Address:

Outbound Proxy Server:

Calling Line ID
Default CPN:
Restriction: False


Policies
Trunk Service: 2
Interconnect Restriction: 1
Maximum Simultaneous Calls: 10
Session Timer: 0
Zone: 1
SMDR Tag: 8888
NAT Keepalive: False
Enable Mitel Proprietary SDP: Yes
Use P-Asserted Identity Header: No
Use Restricted Character Set For Authentication: No
Disable Reliable Provisional Responses: Yes
Use Alternate Destination Domain: No
FQDN or IP Address:
Ignore Incoming Loose Routing Indication: No
Suppress Use of SDP Inactive Media Streams: No
Enable Special Re-invite Collision Handling: No
Enable sending '+' for E.164 numbers: Yes
Force sending SDP in initial Invite message: Yes
Use To Address in From Header on Outgoing Calls: No
Force Answer - send SDP in initial Invite: Yes
Prevent the Use of IP Address 0.0.0.0 in SDP Messages: Yes
Use P-Preferred Identity Header: No
Route Call Using To Header: Yes
Private SIP Trunk: No
Public Calling Party Number Passthrough: Yes
Use Diverting Party Number as Calling Party Number: Yes
Build Contact Using Request URI Address: No
Renegotiate SDP To Enforce Symmetric Codec: No
Repeat SDP Answer If Duplicate Offer Is Received: No
Allow Peer To Use Multiple Active M-Lines: No


Authentication
User Name:
Password: *******
Confirm Password: *******

Authentication Option for Incoming Calls: No Authentication


6. Create a SIP Peer Profile Assignment by Incoming DID
a. Add the Range for your Enterprise voice extensions, if they start with 1 you can add 1* as a range or if you only want to use 11001 to 11500, you can enter 11001-115000
7. Create a Digit Modification to Absorb 0
8. Create a Route Assignment
a. Select a Route Number, click Change
b. Routing Medium select SIP Trunk,
c. SIP Peer Profile Select the Peer Profile you’ve previously created
d. Digit Modification Select the Digit Modification you’ve previously created.
9. Create an ARS Digits Dialed Assignment
a. Create a digits dialed number and number of digits to follow, and set the termination number to the route you’ve previously created.
10. Log into your Mediation Server add the 3300 IP Address as your next hope IP Address on port 5060
11. To verify if your connection is alive, log into your 3300 and got to your maintenance commands and enter: SIP Link State All , you will get a response if it is in service or out of service.
12. To dial an OCS R2 Enterprice Voice extension dial your ARS number+extension your communicator will then begin to ring.


Release 10.0

1. The Steps are identical as release 9.0 the only thing that looks different is the SIP Peer profile, please see below.

SIP Peer Profile Label: OCSDCMED
Network Element: OCSDCMED

Local Account Information
Registration User Name:
Address Type: IP Address:


Call Routing and Administration Options
Interconnect Restriction: 1
Maximum Simultaneous Calls: 5
Outbound Proxy Server:
SMDR Tag: 9996
Trunk Service: 9
Zone: 1
Alternate Destination Domain Enabled: No
Alternate Destination Domain FQDN or IP Address:
Enable Special Re-invite Collision Handling: No
Private SIP Trunk: Yes
Route Call Using To Header: Yes


Calling Line ID Options
Default CPN:
CPN Restriction: No
Public Calling Party Number Passthrough: Yes
Use Diverting Party Number as Calling Party Number: No


Authentication Options
User Name:
Password: *******
Confirm Password: *******
Authentication Option for Incoming Calls: No Authentication


SDP Options
Allow Peer To Use Multiple Active M-Lines: No
Enable Mitel Proprietary SDP: Yes
Force sending SDP in initial Invite message: Yes
Force sending SDP in initial Invite - Early Answer: Yes
NAT Keepalive: No
Prevent the Use of IP Address 0.0.0.0 in SDP Messages: Yes
Renegotiate SDP To Enforce Symmetric Codec: No
Repeat SDP Answer If Duplicate Offer Is Received: No
RTP Packetization Rate Override: No
RTP Packetization Rate: 20ms
Special handling of Offers in 2XX responses (INVITE): No
Suppress Use of SDP Inactive Media Streams: No


Signaling and Header Manipulation Options
Session Timer: 0
Build Contact Using Request URI Address: No
Disable Reliable Provisional Responses: Yes
Enable sending '+' for E.164 numbers: Yes
Ignore Incoming Loose Routing Indication: No
Use P-Asserted Identity Header: No
Use P-Preferred Identity Header: No
Use Restricted Character Set For Authentication: No
Use To Address in From Header on Outgoing Calls: No

Please let me know if you have any questions regarding this setup and I will help as best I can.

Thanks
Habib