If this was a sip itsp circuit, Calling Party Name could be changed using voice class sip-profiles.
In the CUCM the only that the calling name can be changed maybe through the SIP Normalization Script.
Hope this is helpful.
Hi, Thank you very much for your kind reply.
We are receiving calls through 2 gateways.
1. E1 s terminating on PSTN Voice gateway (2951 router)
2. MPLS terminated to CUBE (2951)
So with this infrastructure will I be able to do these changes?
We want to have separate names displayed on the agent phones for the calls coming through E1s and MPLS without loosing ANI information.
Any insight here will be very helpful to me.
Correct me if I'm wrong:
2 Call Scenarios:
Customer(Annie) --> TELECO --> E1 --> CUCM -->Phone Display(Annie)
Customer(Annie) -->ISP --> CUBE --> CUCM --> Phone Display (James)
If a call comes from the TELECO, the displayed name should be " Annie Walker".
But if the calls comes from the ISP from the same number the name should be altered to "James"
Is that the sort of configuration you need.
If so. The incoming calls from CUBE can be modified, hence the calling name will not be the same
Calls from E1 = not modified (default name)
Calls from CUBE = Modified Name will be used.
Thanks again for replying.
We have separate set of calls being received through E1s and CUBE (Via MPLS).
We have UCCE setup where calls from UK comes through E1 circuits( We want all these calls to display as "for ex: UK Call" ) and
Calls from US will be coming through MPLS-->MPLS Router-->CUBE (We want all these calls to display as " for ex: US Call ") on phone.
This will help agents to identify calls(US & UK) and address the customers with appropriate verbiage (Agents are cross skilled to handle both US and UK Customers).
I know this can all be done using sip translations, I'm assuming that the CUBE is using sip to connect to cucm.
If both your routers are using sip to route calls to cucm but if your using h.323 or mgcp i can't.
Please post your incoming and outgoing dial-peer on both our gateways and a sample debug for a typical incoming call only from the gateway that is connected to cucm as a sip trunk.(Debug ccsip messages)
I have attached following info for CUBE with this reply.
1. Outgoing Dial-peer (translation profile will just prefix 8 with DNIS)
2. Debug ccsip messages
Typical call scenario is as below.
1. Call reaches CUBE(10.255.23.168) via MPLS(from 10.0.128.167)
2. From CUBE it reaches CVP (comprehensive model) call server (10.255.6.162)
3. Call server contacts with ICM for agent and extension info
4. ICM receives request and provides target extension info to call server.
5. Call server will inform CUCM via SIP trunk configured to call server.
6. Once the call answered RTP will flow between extension and CUBE and to caller.
I have done some research on voice-class sip profiles but still not sure which header to modify.
am able to modify Remote party ID using voice class sip profiles and achieve desired result.
thanks for your kind help and assistance.