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Gatekeeper

pcromwell
Level 3
Level 3

Can anyone help me clarify the use of gatekeepers.

The design is for over 23 remote sites hub and spoke. Each Site has a Gateway. A gatekeeper is at the Hub site.

cisco design/config guides say, that each gateway should have a dial peer with a dest pattern that matches all outbound voip calls(not PSTN)

and the session target is RAS.

So in other words all gateways are to register with gatekeeper for cac etc. O.k I get that bit.

It also reccomends that a ICT trunk (gatekeeper controlled) is setup on the callmanager that registers with the gatekeeper.

I then create a route pattern that routes all VOip calls to the gatekeeper.

So for the Hub site...

1. Why do I need to point Dial-peers to the gatekeeper, when all calls will go straight to gatekeeper from Cucm via the ICT trunk and bypass voice Gateway?

Or do I need to point the route pattern onto the voice gateway which would then forward to the gatekeeper, (seems inefficient)

2. I thought the call flow would be as follows.. A call made form Hub site to x remote site, it first goes to the gatekeeper, where it is resolved to an ip address this is then forwarded to the Wan router and onto the remote site.

3.Any calls incoming to the Hub site would go to the gatekeeper and from there onto the callmanager, am I missing something? surely the hub site  voice gateway is only needed for PSTN calls?

4.Based on this thinking If i had an IP enabled PBX at the remote end, I wouldn't need to go through that gateway either.

1 Accepted Solution

Accepted Solutions

Rick Arps
Level 4
Level 4

Hope I can get some of these answers for you

It also reccomends that a ICT trunk (gatekeeper controlled) is setup on the callmanager that registers with the gatekeeper.

I think you want to be using an h.225 gatekeeper controlled trunk.  ICT is for CUCM to CUCM.

So for the Hub site...

1. Why do I need to point Dial-peers to the gatekeeper, when all calls will go straight to gatekeeper from Cucm via the ICT trunk and bypass voice Gateway?

Or do I need to point the route pattern onto the voice gateway which would then forward to the gatekeeper, (seems inefficient)

This is for gateway to gateway calls.  On the originating gateway you point a match all dial-peer at the gatekeeper which will route the call to the best terminating gateway.  It sounds like (correct me if I'm wrong) that all of your calls will either be from your gateways to your callmanager, or from your callmanager to your gateways.  If this is the case, then the gatekeeper will only be doing cac.

2. I thought the call flow would be as follows.. A call made form Hub site to x remote site, it first goes to the gatekeeper, where it is resolved to an ip address this is then forwarded to the Wan router and onto the remote site.

Correct.  Each router will register with the gatekeeper.  The gatekeeper will route the calls from the callmanager to the best gateway based on what destination patterns each gateway registered with.

3.Any calls incoming to the Hub site would go to the gatekeeper and from there onto the callmanager, am I missing something? surely the hub site  voice gateway is only needed for PSTN calls?

I don't completely understand the question here, calls from the gateways to the cucm will go through the gatekeeper for cac.

4.Based on this thinking If i had an IP enabled PBX at the remote end, I wouldn't need to go through that gateway either.

If you had an IP PBX, you would need to create another trunk with the callmanager.

If you're managing your dialplan on your callmanager, and you can do cac on your callmanager, you can probably get away without a gatekeeper.  If I understand your situation properly, it sounds like you could get rid of it and take care of your cac and dialplan on your callmanager.

Hope this helps

Rick

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2 Replies 2

pcromwell
Level 3
Level 3

Any info would be much appreciated

Rick Arps
Level 4
Level 4

Hope I can get some of these answers for you

It also reccomends that a ICT trunk (gatekeeper controlled) is setup on the callmanager that registers with the gatekeeper.

I think you want to be using an h.225 gatekeeper controlled trunk.  ICT is for CUCM to CUCM.

So for the Hub site...

1. Why do I need to point Dial-peers to the gatekeeper, when all calls will go straight to gatekeeper from Cucm via the ICT trunk and bypass voice Gateway?

Or do I need to point the route pattern onto the voice gateway which would then forward to the gatekeeper, (seems inefficient)

This is for gateway to gateway calls.  On the originating gateway you point a match all dial-peer at the gatekeeper which will route the call to the best terminating gateway.  It sounds like (correct me if I'm wrong) that all of your calls will either be from your gateways to your callmanager, or from your callmanager to your gateways.  If this is the case, then the gatekeeper will only be doing cac.

2. I thought the call flow would be as follows.. A call made form Hub site to x remote site, it first goes to the gatekeeper, where it is resolved to an ip address this is then forwarded to the Wan router and onto the remote site.

Correct.  Each router will register with the gatekeeper.  The gatekeeper will route the calls from the callmanager to the best gateway based on what destination patterns each gateway registered with.

3.Any calls incoming to the Hub site would go to the gatekeeper and from there onto the callmanager, am I missing something? surely the hub site  voice gateway is only needed for PSTN calls?

I don't completely understand the question here, calls from the gateways to the cucm will go through the gatekeeper for cac.

4.Based on this thinking If i had an IP enabled PBX at the remote end, I wouldn't need to go through that gateway either.

If you had an IP PBX, you would need to create another trunk with the callmanager.

If you're managing your dialplan on your callmanager, and you can do cac on your callmanager, you can probably get away without a gatekeeper.  If I understand your situation properly, it sounds like you could get rid of it and take care of your cac and dialplan on your callmanager.

Hope this helps

Rick