Can CUCM manipulate the Caller ID name that is presented to a called number when calling to the PSTN? If so is there good documentation on how to do it? Thanks!
I am trying to manipulate the Caller ID Name that Displays out when users go out a certain PRI.
Your original post said caller ID (i.e. ANI). Now you're talking about Caller ID Name (i.e. CNAM). On the telco side these are two very different things. For example, with ANI you can specify this in the ISDN SETUP or FACILITY IE of an outbound call. The telco switch will copy this value, assuming it matches the filter list applied (i.e. the DIDs you own) to that trunk group, into the SS7 messages for the call. The called party's telco switch will use the ANI value specified in the SS7 message for display purposes unless it is flagged for end-user blocking.
In the case of CNAM this process is very different. CNAM isn't carried in the SS7 messages. The called party's telco switch must perform a database query based on the calling ANI to get the name. This database is not updated in real-time and most telcos set it to the name on the bill. Only the telco can change this and it must be done per TN. Because of this it doesn't matter what you send for CNAM; it won't get passed along anyways*.
*The possible exception is intra-telco calls. In some cases carriers do pass this information inside their own network. If this is the case (the telco will actually advertise this as a feature to you) then you can play some tricks on the gateway. The prerequisite to this trick is doing SIP between CUCM and the gateway. You would then need to create two separate SIP trunks (i.e. separate route patterns and the like) so the router can differentiate between company A and B calls. On the gateway you can use a voice class sip-profile to manipulate the FROM header of the incoming SIP INVITE on the incoming VoIP dial-peer. CUCM will set it to the Alerting Name per-phone; however, you can override it once it gets to the router this way. I believe it would look like this; however, I'm writing this from memory so this may not be exactly correct:
voice class sip-profiles 1
request INVITE sip-header From modify "^\"\([A-Za-z0-9 ]+\)\"[ \t]+\(<.*\)$" "\"COMPANY A\" \2"
! What this is attempting to do is turn this:
! From: “JOHN DOE” ;tag=12345
! into this:
! From: “COMPANY A” ;tag=12345
! Then apply this to your incoming VoIP dial-peer.
You can keep things separate on the gateway by just prefixing a unique digit sequence on the CUCM route pattern. For example company A's route pattern may prefix 9A while company B prefixes 9B. You can use this to match two unique incoming dial-peers.