Direct SIP: Cisco Unified Communications Manager 6.1
If you're looking for an introduction about how to configure Cisco Unified Communications Manager (CUCM) version 6.1.2 with Office Communications Server 2007 R2, this article will help you. The proper way to connect and interoperate the two products is to set up a Direct SIP connection in Office Communications Server. On the Cisco side, this is referred to as a SIP trunk. This article covers how to configure a SIP trunk in CUCM.
Authors: Jerome Berniere, Rui Maximo
Original publication date: December 2009
Product version: Office Communications Server 2007 R2
Interoperating Office Communications Server and CUCM
If you're looking for an introduction about how to configure Cisco Unified Communications Manager (CUCM) version 6.1.2 with Office Communications Server 2007 R2, this article will help you. The proper way to connect and interoperate the two products is to set up a Direct SIP connection in Office Communications Server's parlance. In Cisco's parlance, this is referred to as a SIP trunk. So, when discussing the configuration of CUCM, we'll refer to this interop connectivity as SIP trunk. When explaining the configuration of the Office Communications Server, we'll refer to this interop connectivity as a Direct SIP.
If you're trying to operate CUCM with Office Communications Server 2007 R2, you probably already have CUCM deployed in your environment for all your IP phones and your CUCM configured to route all external calls to the PSTN through a PSTN trunk. In this case, you'll likely want to configure Office Communications Server 2007 R2 Enterprise Voice to route external calls to the PSTN through the CUCM PSTN trunk. Calls between Communicator users and Cisco IP phone users should be possible using each user's unique extension number, and users are accessible via externally routable direct inward dialing (DID). This is illustrated in Figure 1.
Figure 1. Routing of calls between Communicator users and Cisco IP phone users
There are two parts to interoperating the two products: configuring CUCM, and configuring Office Communications Server. The high level steps for configuring CUCM are as follows:
- Create partition
- Create calling search space
- Define translation patterns
- Provision SIP trunk
- Set up route pattern
The following sections describe the CUCM configuration steps in more detail. For details about how to configure Direct SIP on Office Communications Server 2007 R2 Mediation Server, see Direct SIP: Configuring Mediation Server.
Configuring Cisco Unified Communications Manager 6.1.2
Let's dive into the configuration steps for CUCM. The connectivity between CUCM and the Mediation Server is referred to as SIP trunk to conform to Cisco's terminology.
To read the rest of the article, go to Direct SIP: Cisco Unified Communications Manager 6.1 in the TechNet Library.
Comments
Anonymous
June 13, 2010
this is great - very helpful - I did something very similar in my CUCM 7.x / Exchange 2010 environment - hadn't tried to bring my OCS 2007 R2 in the mix yet. Now I might! :)Anonymous
June 13, 2010
The comment has been removedAnonymous
June 13, 2010
Why CUCM 6.1?? Current release is 8.0...Anonymous
June 13, 2010
Good article - thanks!Anonymous
June 13, 2010
Awesome. A great starting point, and very relevant to what I may have to configure soon! :-)Anonymous
June 21, 2010
One thing you should mention up front is the effect of the MTP on the SIP trunk - i.e. forcing all audio through CUCM. This can cause horrible WAN routing for audio if the customer has deployed centralised or regional CUCM clusters. Should be able to mitigate this by switching phones to SIP rather than SCCP (avoids need for MTP) -- but that's a big change for customers. Also - what about the user experience lost, e.g. caller name display. Direct SIP trunking means that the Callers Name is not displayed on incoming calls to CUCM from OCS. CUCM users are used to seeing name rather than number for internal users. Using an SIP-SIP gateway between Mediation and CUCM (e.g. NET VX1200) allows you to modify SIP traffic and replace caller name in the SIP header. These sorts of things are really important for user adoption of such hybrid solutions.