I read a blog post that mentioned that direct SIP integration with OCS and Cisco Call Manager is now supported by Microsoft. This link also verifies that this is the case! http://technet.microsoft.com/en-us/office/bb735838.aspx
Has anyone found a good resource on the setup on the Call Manager side to make this happen? I know that Cisco has published documentation on how to integrate OCS with a 2800 or 3800 series ISR connected via T-1, but I haven't seen anything on the direct SIP integration setup. I would be very interested in seeing what the setup looks like on the Call Manager side.Tuesday, September 16, 2008 1:25 PM
After the OCS 2007 August updates I am able to make a call between MOC and Cisco IP Phone via a DirectSIP to CUCM6. I am still working on the Dual forking.
You need to read the following Microsoft KB and you shoud be able to make it work.
MobinTuesday, September 16, 2008 10:31 PM
Cisco has just released great documentation relating to Cisco Unified Presence with Microsoft OCS and RCC. The document is offically called "Integration Note for Configuring Cisco Unified Presence with Microsoft OCS for MOC Call Control" and can be found here:
I know you mentioned that you are looking for CUCM Direct SIP to OCS documentation above. Check out the following links:
Hope this helps,
KeenanThursday, September 25, 2008 5:22 PM
The link to Cisco document is for RCC feature only, which has nothing to do with Dual Forking.
Dual forking is supported (Microsoft OCS and Mediation server August updates) in 4.2, 5.x and 6.x, I have OCS 2007 with DirectSIP to CUCM6 working fine with Dual Forking.
CUCM 7.0.1 supports Dual Forking but not Dual Forking with RCC, that will be supported in upcoming CUCM 7.1.
I was in TechEd 2008 Sydney and both Microsoft and Cisco ppl mentioned that in their presentation.
Have a look at the MS Communications Server team blog site
MobinMonday, September 29, 2008 12:18 AM
Great information Mobin!
I am a little behind in testing dual forking, so I am hoping you can answer this. When you mentioned that dual forking IS SUPPORTED in Cisco Call Manager/UCM 4.2, 5.x and 6.x, you are referring to strictly an Direct SIP with Enterprise Voice (with no PBX integration option enabled) scenario, correct?
Basically..... CUCM 5.1 <--> Mediation server <--> OCS SE
Therefore, if I have my extension assigned to my Cisco IP Phone (ex. 9445) and I have the same extension assigned to my OCS profile (9445), dual forking will successfully occur? If I then answer the inbound PSTN call on my MOC client, my Cisco IP phone will stop ringing?
KeenanMonday, September 29, 2008 2:27 PM
I am not sure Mobin understands what dual call forking is but the only deployment model that supports it at the moment is with a Nortel CS1000. No version of Callmanager is currently supporting it. Direct SIP connectivity is different and it is supported with callmanager 4.2.3 and 5 and beyond.
Dual call forking is the ability for callmanger to fork the call and basically you can have a Cisco IP Phone ring and MOC ring using the same number as you described above and Callmanager cant do this yet. There are a number of ways you can do this without dual call forking of course using Cisco Callmanager and mobility manager but native dual call forking is not available yet.
ChrisMonday, September 29, 2008 3:45 PM
I did a bit more research on the subject and this is what I come across in the Cisco documentation for CUCM 7.0:
SIP URI Dialing
This feature supports Session Initiation Protocol (SIP) Universal Resource Identifier (URI) as an additional type of remote destination for Cisco Unified Mobility. When the DN is called, Cisco Unified Communications Manager extends the call to a SIP trunk that digit analysis selects with this SIP URI in the To: header.
This feature only allows routing that is based only on the domain name, not based on the full SIP URI.
When a remote destination of this type is configured, other Cisco Unified Mobility features, such as two-stage dialing, transformation to DN number when calling into Cisco Unified Communications Manager, Interactive Voice Response (IVR) support, caller ID match, or DTMF transfer and conferencing, do not get supported.
SIP URI Administration Details
The SIP URI dialing feature entails a relaxation of the business rules to allow the entry of a URI in the Destination Number field of the Remote Destination Configuration window. (From the Cisco Unified Communications Manager Administration menu bar, choose the Device > Remote Destination menu option.)
An additional requirement for this feature specifies that a SIP route pattern that matches the configured URI domain must be configured for the feature to work. To configure a SIP route pattern, from the Cisco Unified Communications Manager Administration menu bar, choose the Call Routing > SIP Route Pattern menu option.
SIP URI Example
For a remote destination, the SIP URI firstname.lastname@example.org gets configured. A SIP route pattern that specifies corporation.com must also get configured for the SIP URI remote destination to resolve correctly.
See the "Related Topics" section.
So instead of using a number as a remote destination you can use a SIP URI. kind of interesting. Its not free though. For those of you who are familar with Cisco licensing this will cost you two DLU's. This gives you the ability to have up to four remote destinations. Not sure if this is going to be Cisco's standard for dual call forking or not but the fact it costs you to use it kind of stinks.
ChrisTuesday, September 30, 2008 1:55 PM
Thanks to everyone for the information. I found this new blog post today that gets me exactly what I was looking for.Wednesday, October 01, 2008 2:32 PM
You may also want to check out their previous entry as it talks about how to load the patches on OCS and the file you will need to create. Its a good write up that I have used and it works with CCM 4.2.3.
Just make sure you test all the features on the Cisco and OCS side if you plan to roll it out.
ChrisWednesday, October 01, 2008 3:35 PM
I have read the whole post and I would like to ask if my understanding of forking is correct.
Currently I have setup where rcc is configured. (remote call control)
Callmanager ver 5.1 , presence 6, and no direct calling yet.
In this scenario when IP phone receives a call the MOC clienet rings also.
Would this be considered dual forking because both phones ring?
In future I am planning to enable MOC for dialing out and receiving calls (direct calling).
My question is this:
Is it possible for the MOC to answer the call so the media path is established between ccm and ocs (through mediation server) Is this this that feature that eveyone is waiting for?
If this is possible would I have to configure any extra extensions for MOC?
In the dicussion the Mobility was mentioned. How is that working?
Do we have to setup additional extensions for MOC client and the configure the ip phone to call that number also when the call is received? Can one extension be shared between MOC and IP Phone? How MOC registers the extesions? Does it act as a SIP client? Do we have to configure SIP phone in cmm?
BartWednesday, October 01, 2008 9:49 PM
We've recently been working on getting a direct sip connection working between mediation and CUCM. It seems this is working fine. Just wondering how do we achieve dual forking? You mentioned Cisco Callmanager and mobility manager? How do we have to see this? Can you please ellaborate a bit more?
DimitriMonday, October 13, 2008 2:58 PM