I have noticed in recent years Cisco is migrating away from SCCP and H323 and doing everything via SIP protocol. While I have not used the gatekeeper functionality personally yet, I have seen it implemented in other deployments, for inter-site dialling.
eg. - Create SCCP extension, add a secondary number which registers with GK, to call in-between two sites, dial secondary number instead of primary ext.
I may need to implement the above example functionality soon. While I have SCCP deployed on 2900 routers, it could be done easily. But it doesn't sound future proof the way things are. - I have also seen that ISR 4000 doesn't support GK functionality.
What is the new/modern alternative to my above example? I have searched but cannot seem to find a simple answer. Is there a way to register secondary numbers on both SIP and SCCP extensions for inter-site dialling without a H323 GK?
Do I understand correctly that you are talking about extensions/phones in CME? If so, dial-peers and voice translation rules should provide the functionality you describe without a GK. If this is the case, can you provide an example of the primary and intersite extensions you would like to accommodate? It may be that an "overlay" number can give a phone button on a CME phone a secondary extension in the way you describe.
Or, if you are talking about CUCM, have you looked into Enterprise Alternate Numbers?
Yes you are correct. I am referring to phones in CME. While I understand that dial-peers and translation rules can indeed provide this functionality, I was interested in a more future-proof solution in case of growth. If we did it between just these two sites, then yes peers and rules would be fine. But, if we grew and needed to deploy more than a few extra locations, that would likely require logging in to each router every time and adding translations for all the extensions at other sites. However, if each site/extension could just be pointed out to GK and register call processing/destination info there, then I would prefer to handle it that way. Fine on 2900/3900 hardware, but 4000 ISR's seem to have done away with this functionality - hence I am now wondering what the "modern" alternative to this functionality is.
Understood. I would suggest looking at a SIP Proxy Server. A proxy server can provide registration, centralized dialplan, and can be programmed for call admission control and other policy-based call processing. They can do pretty much anything the H323 gatekeepers could.
Cisco's entry into the SIP Proxy Server market is the CUSP (Cisco Unified SIP Proxy) but there are others that are as good or better. Cost may be your deciding factor.
Let us know if this helps or if you have additional questions.
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