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New Member

OCS 2007 integration with Call Manager


I have the following proposed design in mind for a customer:

Customer A

OCS 2007-->Cisco 3800 series router for PSTN calls

Customer B

Call Manager in a centralised topology with a voice gatway Cisco 2800 series at Customer A.

We would like to use a digital trunk between the Cisco 3800 series gateway and Cisco 2800 series router. Therefore calls to customers on the Customer B and other sites connected can be reached and treated as on-net calls.

Can you see any issues with this design? In particular the SIP conversion and also will there be a need for CUBE lic?

Hall of Fame Super Silver

OCS 2007 integration with Call Manager

When you say digital trunk, do you mean PRI/QSIG connection between the routers?

Are the routers at the same location?

Or did you mean IP connection such as SIP trunk between the 2 routers?  If so, you will obviously require IP routing between the 2 customers, you may be better off simply creating SIP trunk between OCS and CUCM directly which will not require CUBE licensing.



New Member

OCS 2007 integration with Call Manager

Hi Chris,

Let me explain:

There is a Head office location which has a CUCM6 deployment. We would like to connect one of their branch location that has OCS installed as the PBX. The branch location has OCS PBX with a Cisco gateway to the PSTN. We have also deployed a Cisco 2800 router at the same site for sole purpose to connect to the 3800 router. We are thinking about using digital trunk with T1 PRI configuration. This design will not require ip routing between locations. Can you advise if this is a good design and if CUBE lic will needed.

Hall of Fame Super Silver

OCS 2007 integration with Call Manager

Sure, it will work just like a traditional PBX integration and simply use QSIG as the PRI protocol to take advantage of path replacement, etc.

This will not require CUBE licenses as you are not running any IP-IP GW stuff here.



New Member

Re: OCS 2007 integration with Call Manager

Hi Chris,

We have finally started this implementation and so far we have one way calling working.

Call Flow

Phone A(2010)-->OCS Site A-->H323A GW->Dig PRI Trunk->H323B GW->>CUCM6-->IP Phones(Site B- 5302801)

Phone A dials 85302801-> matches dial-peer sends all 8 characters to the H323B GW which forwards this to the CUCM to match the IP Phone(5302801).


OCS Site A can call IP Phones

Not Working:

Extension (5302801) dials 85032010(Phone A)(8503 is the site code and gets stripped, only 2010 is sent to the H323A GW and all is heard is fast busy.


We have enabled the debugs but no logs is seen whether this call is being received on the router.

debugs used were:

debug h225 asn1

### debug h245 asn1

### debug cch323 all

### debug voip ccapi inout

### debug isdn q931

Any ideas what is happening here.

I am attaching the logs of both gws.