In the CUCM there are 3 main components to build a dial plan.
1. Route Pattern
If someone dials a number which matches a configured route pattern, that call is eligible for routing.
Ex:- XXX, 9.XXX, [1-2]XX, 234!#
If you want to know more about Route Patterns click here.
2. Dial Plan Hierarchy
This is what tells the CUCM to route the call to the specified destination.
Ex:- route 911 calls to external gateway..
3. Class of Control
This controls the calling abilities of phones.
Ex:- A lobby phone should not be able to take IDD calls..
If you want to know more about Class of Control click here.
In this post, let's discuss about Dial Plan Hierarchy.
Following diagram shows 2 independent companies which has 2 CUCMs and connected via a SIP trunk. And both of them have E1 connections to PSTN too.
Let's assume 011224 & 011225 are prefixes given to identify the area code and the specific company.
To dial 4642 from people outside the companies (from home - PSTN) they should dial the full number 0112254642.
But if 4641 of company B dials 4642, they can make an internal call.
Also to dial 4642 of company B from company A, 74642 is enough as they have a SIP trunk and the CUCM of company A is configured with a route pattern 7.XXXX to be routed out to the inter-company SIP trunk. But in this case, if a phone at company A dials only 4642, it should ring the 4642 of company A..
Let's see how the dial plan hierarchy works..
Route Patterns are pointed to Route Lists which are pointed to Route Groups which are pointed to Devices/Gateways/Trunks
When you are configuring it, you actually do it from right to left. You configure Devices (Gateways/Trunks), Route Groups, Route Lists & finally Route Patterns.
Above hierarchy is for the CUCM of company A. It should be configured to route calls to company B from company A which matches the route patterns in CB_RL route list which is associated with CB_RG route group which points to SIP_to_CB SIP trunk.
If the SIP trunk fails, the calls will be routed out via PSTN_RG route group which points to PRI-1 & PRI-2. But before sending out to PSTN, we should do digit manipulation in order to match the numbers of the outside world. This digit manipulation (changing numbers) is something we can do in dial plan hierarchy.
We have to change 2 numbers in the above scenario.
1. Calling Party number
2. Called Party number
If the SIP trunk fails, the calls will be routed out via PSTN_RG route group which points to PRI-1 & PRI-2. But before sending out to PSTN, we should do digit manipulation in order to match the numbers of the outside world. This digit manipulation (changing numbers) is something we can do in dial plan hierarchy.
We have to change 2 numbers in the above scenario.
1. Calling Party number
2. Called Party number
Let's assume 4641 of company A (Calling Party) wants to dial 4642 of company B (Called Party). So he dials 74642 which is the route pattern associated with CB_RL. But when the device (SIP_to_CB) of CB_RG fails, the CB_RL route list should consider PSTN_RG which is the second route group associated with it. When we add the PSTN_RG to the CB_RL is the place we do digit manipulation.
Company A should add prefix digits 011224 for the calling party and 011225 for the called party so that the number can be identified by PSTN. In the gateway configuration of company B, they can configure to accept only last 4 digits which will strip away the prefix digits company A added so that their Call Manager can route the call internally.
The above digit manipulation can be also done using a mask too like the following.
To change the Calling Party number, Company A need only to select on from drop down menu of Use Calling Party's Phone Number Mask which will use the default external mask we give while configuring extensions. If that mask is not enough to route the calls back from a long distance, we can add prefix digits too.
Note:-
Although the user dials 7.4642, I have not configured a discard predot at Route Lists because it is already stripped away at the route pattern.
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