Call Detail Records (CDR)

Call Detail Records (CDR) logs call activity in the system and can be sent out or be queried to/from billing application.

Before you begin make sure the following services have been activated.
1. Cisco CAR Web Service  ---> For access CDR details in Publisher web page.
2. Cisco SOAP - CDRonDemand Service  ---> For billing application that query CDR directly from Publisher

Enable Service Parameters

1. Enable CDR in Callmanager Service Parameters (Enable all server in the cluster).

2. If you want to include FAC in CDR, set Display FAC in CDR = True.

To access CDR details in Publisher web page use the following URL.

https://{Publisher IP Address}:8443/car/Logon.jsp

User has to be assigned in Standard CAR Admin Users Group to access this page.
  
To send CDR details to external server using FTP/SFTP.

1. In Cisco Unified Serviceability Page, Select Tools> CDR Management

2. In Billing Application Server Parameters, click Add new

3. Enter IP Address of FTP/SFTP server, account to login that server and Directory Path.

4. If you've got the error message, make sure that server already start FTP/SFTP service.
   

Local Route Groups

Local Route Groups(LRG) feature reduce the complexity of the numbering plan when you have a large number of locations.

Consider the following scenario.

In the picture, we have Callmanager at HQ and phone from each branch register to Callmanager.
 
Each branch has their own PSTN gateway, so in Route Pattern configuration we need 5 Route Patterns with 5 Partitions and phones from each branch have CSS to access their own PT and then to their own gateway.

With Local Route Group we can simplify this route plan. Consider the following picture

As you can see, each branch has their own Device Pool associated with their own LRG and then their own gateway.

With this configuration, we achieve the same result with only 1 Route Pattern.

When a phone initiate a call to outside and then match Route Pattern --> Route List

The Route List contain special route group that is named Standard Local Route Group.

At this point system will check the phone's Device Pool and see which Route Group associated.

Then call will be sent out to gateway associated to that Route Group.

Detail Configuration,

Step1. Create 5 Route Groups and associate gateways to Route Groups.

Step2. Create 5 Device Pools and associate Local Route Group in Device Pool with Route Group created in Step1.

Step3. Create Route List and associate Standard Local Route Group to this Route List.

Step4. Create Route Pattern and associate Route List to this Route Pattern.

Step5. Associate each Device Pool to phones.

Step6. If call sending out to each local gateway require different CLID, we need to use Calling Party Transformation Patterns to manipulate Calling Number for each barnch.

For example BR1 with DID number 021267XXX and extension number 7XXX.

Step6.1 Create Calling Party Transformation Patterns (Partition is required)

Step6.2 Create CSS that can access Calling Party Transformation Patterns Partition in Step6.1 in this case CSS_BR1_CLID

Step6.3 Associate CSS_BR1_CLID to Calling Party Transformation CSS in Device Pool or Gateway.

If you associate CSS_BR1_CLID to Gateway make sure to uncheck Use Device Pool Calling Party Transformation CSS.

     

Single Number reach(SNR) and Enterprise Feature Access(EFA)

SNR associates IP Phone number with the destination number(Up to 10) such as mobile phone. Incoming calls to a user IP Phone also ring to mobile phone at the same time.

With EFA user can originate calls from the remote destination phone through the Callmanager and instead of sending the mobile number to the called party, the user enterprise number gets sent to the called party.

Users can also use EFA to enable or disable SNR (Mobile connect).

Example configuration.

In this example incoming call to IP Phone DN 1001 will also ring to mobile phone number 0891111111.

Note:

1. In this example Route Pattern for mobile number will be 9.08XXXXXXXX with pre-dot discard digit.

2. CLID from PSTN to Callmanager is 0891111111.

3. You'll see in the example that application dial rule need to be used to make both SNR and EFA feature working.

Before you begin make sure that Cisco Unified Mobile Voice Access Service already activated.

Step1. Create User Account(user1001 in this example), associate physical phone or EM profile to this user, check Enable Mobility and assign Standard CCM End User Role.

Step2. Enable EFA in Service Parameter.

Step3. Define EFA Number in Call Routing> Mobility Configuration.

Step4. Create Remote Destination Profile, Select Device> Device Settings> Remote Destination Profile and Add New.

For Calling Search Space enter CSS that can access EFA number Partition in Step3.
For Rerouting Calling Search Space enter CSS that can access PSTN Mobile Call.

Step5. Associate DN 1001 to this RDP, Note that this DN already associated with physical phone.

Step6. Add a New Remote Destination in RDP.

In Destination Number enter mobile number in this case 0891111111. This number must match CLID from PSTN to Callmanager, otherwise EFA will fail.

Delay Before Ringing Timer is the delay time before ring to mobile phone after IP Phone ring.

Define Schedule for this Remote Destination such as you may want to enable SNR only on weekends.

When receiving a call during the above ring schedule is the list of caller that you want to send call to mobile phone(Need to create Access List in Call Routing> Class of Control> Access List)

Save and check Line Associate in Association Information(Important!!!)

Step7. Go to Phone Configuration, assign softkey template that has Mobility softkey and assign Owner User ID(For EM assign this user to Device Profile).

Step8. Since we have to use 9 for access code but in Remote Destination has only 0891111111(why? see Step6 for reason to not include access code). So we need to use Application Dial Rule to help.

Select Call Routing> Dial Rule> Application Dial Rules and Add New.

With this rule system will prepend the digit 9 to number begin with 08 and digit length=10.

Step9. Test call to number 1001, call should be ring both IP Phone and mobile phone.

If you answer call from IP Phone, press Mobility Softkey and select send call to mobile phone, call should be sent to mobile phone.

If you answer call from mobile phone, IP Phone line status will show busy(red) and when you drop the call from mobile phone system still hold the line for a few second before disconnect the call so during this time you can pick up the call from IP Phone.

When you use mobile phone(Same number in Remote Destination) call in to EFA number and press PIN#1#Number to Call# this will use Callmanager to call out to destination and CLID will be user enterprise number(IP Phone) not mobile number.

   

Extension Mobility

Cisco Extension Mobility(EM) allow users login to other phones and get their profiles such as DN, Services, speed dial. So users can use any phone in the network just like their own phones.

Configuration

Before you begin make sure that EM service already enabled in Serviceability Page.

1. Create Phone Service, Select Device> Device Settings> Phone Services and Add New

In Service URL enter
http://{EM Server}:8080/emapp/EMAppServlet?device=#DEVICENAME#

2. Create Device Profile, Select Device> Device Settings> Device Profile and Add New enter your device type and protocol.

Enter the other option like Softkey Template, Add DN, etc.. just like you're configuring Phone (Imagine that your Device Profile is your Physical Phone that will be loaded when you logged in).

3. Subscribe EM Service that you created in step1 to Device Profile. At Related Links select Subscribe/Unsubscribe Services, then click Go

4. Create User for login, Select User Management> End User and Add New.

Enter User ID, Password, PIN, Lastname

In Extension Mobility Section select your device profile created in step2 to Controlled Profiles

Click Save, then assisgn role to this user

5. Go to phone configuration, enable EM and subscribe EM service to physical phone

6. Go back to Device Profile Configuration and associate user to Device Profile.

7. Test login, Press Service button on phone enter User ID, PIN.

8. In Phone Configuration you'll see the details for logged in user.

9. You may want to tuning some EM Paramter. Go to Service Parameter and see.

     

Call Display Restriction

The Call Display Restriction feature allows you to selectively display or restrict the calling and connected (called party) line information. A hotel environment frequently requires this functionality.

In Callmanager we use translation pattern to implement this feature.

In Translation Pattern use these field to control call display

Example,

In this example call between room will restrict both number and name, calling phone will show "To Private" and called phone will show "From Private".

In this example call from room to room service, calling phone(room) will show only name "To RoomService" and called phone(room service) will show both number and name.

Note:

1. Make sure that phone cannot call directly to other phones, otherwise you cannot control call display.

2. Make sure that Translation Pattern has the CSS access to phone partition.

If you don't understand Partition and Calling Search Space read my article at

http://cisco-uc-learning.blogspot.com/2010/07/partition-and-calling-search-space.html

    

Conference Bridge Resource, Region and Transcoder

This article will explain the relationship between CUCM Conference Bridge Resouce(CFB), Region and Transcoder.

First of all let's see the normal call within the same Region.

Then what happen when IP Phones use conference call in the same Region.

As you can see RTP stream will be mixed to CFB and passed to IP Phones(Callmanager CFB support only G.711 codec).

Next let's see the normal call between Region.

In this example call between Region Default(HQ) and Region BR_RG(BRANCH) will use G.729 codec.

So, when happen if we have conference call between Regions.

Because CUCM CFB support G.711 codec only, so conference call in this case will fail.

There are 2 solutions to make this work.

1. Using the Transcoder Resource.

This solution require DSP Resource in Router to register to Callmanager and system will automatically detect if Transcoder require or not.

2. What if I don't have Transcoder Resource(DSP quite expensive), another solution is to create Region for CFB and configure all other Regions connect to CFB Region using G.711 codec.

Note that normal call between 2 Regions still use G.729 codec but when IP Phone start to use conference call between 2 Regions it will switch to G.711 codec.

Region and Location

Region can be used to control codec between regions, Location can be used to implement call admission control(CAC) which enables you to regulate audio quality and video availability by limiting the amount of bandwidth that is available for audio and video calls over links between the locations.


See example below,

In this example, our customer has 2 buildings, HQ and Branch with the leased line 128 kbps for Data+Voice. With this slow link our customer decide to use G.729 codec and limit 2 concurrent calls between 2 sites.

Note: 1. IP Phones at branch also register to HQ's Callmanager.
         2. Call within region use G.711 codec

Configure,

1. Create 2 Regions named HQ_RG and BR_RG, Select System> Regions and Add New.

2. Configure codec between 2 regions, HQ_RG Configuration.

3. Configure codec between 2 regions, BR_RG Configuration.

4. Create 2 Locations named HQ_LOC and BR_LOC and limit audio bandwidth to 2calls x 24kbps(G.729) = 48 kbps, Select System> Location and Add New.

Use this list to calculate codec bandwidth.

5. HQ_LOC Configuration.

6. BR_LOC Configuration.

7. Create 2 Device Pool named HQ_DP and BR_DP, Select System> Device Pool and Add New, then assign Region and Location to Device Pool.

8. HQ_DP Configuration.

9. BR_DP Configuration.

10. The last step is to assign Device Pool to each phone region. For HQ IP Phone use HQ_DP for branch IP Phone use BR_DP.

11. After testing I found that IP Phones in the same region used G.722 codec which may have some problem with voice recorder, you can disabled G.722 codec to force IP Phones to used G.711 codec in System> Enterprise Parameters and change Advertise G.722 codec to disabled.

12. When you place the 3rd call between 2 sites, this call will not success and IP Phone screen will show "Not Enough Bandwidth", you can customize this message in Callmanager Service Parameter Page.