Cisco IP Telephony | Voice Over IP | VOIP Gateway | UC | Call Centre

Reinitial UCCX 7.0

2010-07-25T14:54:00.003+07:00

Sometimes you may want to get back to UCCX initial page. You can do this by the following step.

1. From the UCCX server, Start> Run and type cet. When you get the warning select No.

2. Select com.cisco.crs.cluster.config.AppAdminSetupConfig, then double click in Row 0 and select com.cisco.crs.cluster.config.AppAdminSetupConfig tab and change Setup State from "Done" to "FRESH_INSTALL"

UCCX Initial Configuration

2010-07-25T14:36:00.003+07:00

This article is based on UCCX7.0.

After you have installed UCCX, the first thing to do is initial the configuration.

1. Use browser to access http://{UCCX IP Address}/appadmin.

The default login is Administrator, ciscocisco

2. Since this is standalone server I'll use Single Node.

3. Enter CM IP Address that has enabled AXL Service (Normally is Publisher) and user account that has Standard AXL API Access Roles in Callmanager.

4. Upload your license files.

5. Wait for System Components Activation and click next.

6. Enter user and password for Telephony Subsystem and RmCm Subsystem. These users will be created in Callmanager Application User automatically.

Note:

6.1 User in Telephony Subsystem is used to control CTI Ports and CTI Route Point in Callmanager.

6.2 User in RmCm Subsystem is used to control Agent Phones.

7. Enter System Parameters Configuration.

Number of HR session licenses = The maximum sessions that HR client can connect to server

Recording Count = The maximum concurrent on demand recording.

Number of Outbound seats = The number of outbound agent.

Codec = Codec that will be used in UCCX.

8. Select language you want to use (can choose more than 1) and check Country Specific if you want to customize your system audios for that language.

9. Select end user(Sync from Callmanager) to be used as an administrator in UCCX.

10. See Cisco Unified CCX Setup Result Information if these is any error.

Call Detail Records (CDR)

2010-07-25T13:44:00.005+07:00

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.

csim start command

2010-07-25T01:18:00.019+07:00

csim start {number} is a hidden command to initiate simulated calls. It allows debugging for call routing from voice gateways and CUCME.

When command was issued it will looks into dial-peer configured on the router and send call to destination-pattern configured in dial-peer.

Note that this command also bypass COR configured in dial-peer.

Router#csim start 7101 ---> this case dial-peer was found and success to call

csim: called number = 7101, loop count = 1 ping count = 0

csim err csimDisconnected recvd DISC cid(4885)

csim: loop = 1, failed = 1

csim: call attempted = 1, setup failed = 1, tone failed = 0

Router#csim start 8001 ---> this case dial-peer was not found

csim: called number = 8001, loop count = 1 ping count = 0

csim err:csim_do_test Error peer not found

Remote SPAN (RSPAN) for Voice Recorder

2010-07-25T00:40:00.000+07:00

Some Voice Recorders require the copy of signal and RTP from IP Phone to theirs interfaces.

This example will show you how to configure remote span session to send the copy of signal and RTP from IP Phone to Voice Recording Server.

Access Switch Configuration

Access-Switch(config)#vlan 999
Access-Switch(config-vlan)#remote-span
Access-Switch(config-vlan)#exit
Access-Switch(config)#monitor session 1 source vlan 100
Access-Switch(config)#monitor session 1 destination remote vlan 999

Access-Switch(config)#int gi0/1
Access-Switch(config-if)#switchport mode trunk

Core Switch Configuration

Core-Switch(config)#vlan 999
Core-Switch(config-vlan)#remote-span
Core-Switch(config-vlan)#exit
Core-Switch(config)#monitor session 1 source remote vlan 999
Core-Switch(config)#monitor session 1 destination interface gi0/1

Core-Switch(config)#int gi0/3
Core-Switch(config-if)#switchport mode trunk

Voice Vlan and Auto QoS

2010-07-25T00:12:00.010+07:00

Implementation of IP Telephony, voice vlan should be used to separate data and voice traffic.

Auto QoS Advantage.

1. Easy to implement.
2. Follow Cisco Best Practices for QoS.

This will show you the example configuration for voice vlan and QoS in switch.

Switch(config)#int fa0/1
Switch(config-if)#switchport mode access
Switch(config-if)#switchport access vlan 200
Switch(config-if)#switchport voice vlan 100
Switch(config-if)#spanning-tree portfast
Switch(config-if)#auto qos voip cisco-phone ---> this will generate many commands

Note:

1. CDP need to be enabled on switch to allow switch to detect Cisco IP Phones.
2. For switch trunk port use "auto qos voip trust"

Understanding Dial Peer

2010-07-24T23:34:00.003+07:00

Dial peers are the primary used to implement a dial plan on Cisco voice gateways. Dial peers are used to determine which calls are routed and the path a call should take.

POTS dial peers are used to process calls that are connected to traditional telephony equipment such as PBX, Analog Phone and PSTN.

VOIP dial peers are used to process calls to or from the data network.

Pots Dial Peers

Configuration

Router(config)#dial-peer voice 1 pots
Router(config-dial-peer)#destination-pattern 2222
Router(config-dial-peer)#port 0/0/0

VOIP Dial Peers

Configuration

Router(config)#dial-peer voice 100 voip
Router(config-dial-peer)#destination-pattern 3333
Router(config-dial-peer)#session-target ipv4:192.168.1.1

Dial Peer Matching

Inbound and Outbound Dial Peers.

Inbound Dial Peer Matching

Priority1 "incoming called-number" ---> Defines the called number or DNIS string

Priority2 "answer-address" ---> Defines the originating calling number or ANI string

Priority3 "destination-pattern" ---> Uses the calling number (originating or ANI string) to match the incoming call leg to an inbound dial peer.

Priority4 "port" ---> Attempts to match the configured dial-peer port to the voice port associated with the incoming call (POTS dial peers only).

Priority5 "default" dial-peer or dial peer 0, is matched when no other inbound dial peer is used. It is better to configure a specific dial peer to match incoming calls.

Outbound Dial Peer Matching

For outbound dial peer matching, the called number is matched against the configured destination-pattern in the dial peers with the longest matched.

Consider the following scenario.

R1 has 4 dial-peers configuration, 2 pots and 2 voips.

When phone 2222 call 3333,

dial-peer 2 pots will match as an inbound dial-peer because "incoming called-number" is the 1st priority when matching inbound dial-peer and the call will be processed as one-stage dialing(you will not get the second dial tone to enter additional digits).

dial-peer 101 voip will match as an outgoing dial-peer because 3333 is the longest matched and codec g.729r8 will be used for RTP media stream over ip network.

ISDN PRI Gateway Configuration

2010-07-24T21:08:00.003+07:00

Environment

1. Cisco Router 3825 with VWIC2-1MFT-T1/E1 card in router HWIC0 slot.
2. PVDM2-32
3. Link to PSTN with ISDN PRI E1 EURO Protocol.

Router(config)#card type e1 0 0
Router(config)#network-clock-participate wic 0
Router(config)#network-clock-select 1 E1 0/0/0
Router(config)#isdn switch-type primary-net5
Router(config)#controller E1 0/0/0
Router(config-controller)#pri-group timeslots 1-31
Router(config-controller)#interface serial 0/0/0:15
Router(config-if)#isdn switch-type primary-net5
Router(config-if)#isdn incoming-voice voice
Router(config-if)#end
Router#show isdn status

Global ISDN Switchtype = primary-net5
ISDN Serial0/0/0:15 interface
        dsl 0, interface ISDN Switchtype = primary-net5
    Layer 1 Status:
        ACTIVE
    Layer 2 Status:
        TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
    Layer 3 Status:
        0 Active Layer 3 Call(s)
    Active dsl 0 CCBs = 0
    The Free Channel Mask: 0xFFFF7FFF
    Number of L2 Discards = 0, L2 Session ID = 1

Now use dial-peer to create the numbering plan.

MGCP Gateway Auto Configuration

2010-07-24T20:40:00.004+07:00

MGCP was created for a centralized architecture, where most of the configuration and call-control intelligence resides on a call agent, such as Cisco Callmanager.

Configuration on IOS Router,

MGCP-Gateway(config)#ccm-manager mgcp
MGCP-Gateway(config)#ccm-manager music-on-hold
MGCP-Gateway(config)#ccm-manager config server
MGCP-Gateway(config)#ccm-manager config
---> After this command Gateway will download configuration from Callmanager.

Configuration Callmanager

Note:

1. Domain Name must be the same with router hostname.

2. If you have command "ip domain-name" in router configuration for example "ip domain-name uclab.com" in Callmanager Gateway Configuration Page use Domain Name = MGCP-Gateway.uclab.com

3. If you have command "no ip domain-name" in router configuration just use MGCP-Gateway in Callmanager Gateway Configuration Page.

Local Route Groups

2010-07-23T22:24:00.004+07:00

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)

2010-07-22T22:31:00.009+07:00

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

2010-07-21T19:57:00.011+07:00

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

2010-07-21T14:36:00.019+07:00

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

2010-07-21T01:07:00.004+07:00

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

2010-07-20T23:37:00.007+07:00

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.

Time-of-Day Routing

2010-07-11T22:05:00.001+07:00

What is Time-of-Day Routing used for?

Time-of-day routing routes calls to different locations based on the time of day when a call is made.

Time-of-day routing blocks calls to destinations based on the time of day or on a specific day when a call is made.

How to configure it?

Time-of-day routing use the concept of Partition(PT), Calling Search Space(CSS) and Time Schedules to control the call.

Example,

Manager would like the incoming call to his extension number 1001 ring to IP Phone during 8.30-17.30 and the rest of time to his mobile phone 0891111111

Step1. Add a Manager Phone with DN 1001 in Manager_PT

Step2. Create Time Period named WorkTime, Select Call Routing> Class of Control> Time Period.

Step3. Create Time Schedule named WorkTimeSchedule and associate WorkTime Period to this schedule.

Step4. Go to Partition Page and associate WorkTimeSchedule to Manager_PT

Step5. Create another DN 1001 with OutOfWork_PT and set Call Forward All to 90891111111 (I use 9 for access code) select CSS that can access mobile call.

Step6. Create Time Period named OutOfWorkTime1 and OutOfWorkTime2.

Step7. Create Create Time Schedule named OutOfWorkTimeSchedule and associate OutOfWorkTime1 and OutOfWorkTime2 to this schedule.

Step8. Associate OutOfWorkTimeSchedule to OutOfWork_PT.

This just only an example, you can apply this concept to any application where you want to control CSS based on Time-of-Day.

Partition and Calling Search Space

2010-07-11T17:54:00.003+07:00

Partition and Calling Search Space are used together to implement class-of-service permissions.

For example,

1. Phones in IT department can dial anywhere within the company and all local area codes, but not long distance or international.

2. Phones in HR department can place toll calls during normal business hours, but not at other times.

Use the following logic,

Partition(PT)=Where you are
Calling Search Space(CSS)=Where (List of PT) you can call.

Partition can be associated with DN, Route Pattern, Hunt Pilot, etc...
CSS can be associated with DN, Phone, Gateway, etc...

General conclusions,

Note: When I said PhoneX in X_PT it means DN of PhoneX in X_PT.

1. Phones in the same PT not mean that they can call each other, if PhoneA and PhoneB are in for example IT_PT but CSS of both phones=None PhoneA cannot call PhoneB, and vice-versa.

2. Phones can or cannot call to destination depend on its associated CSS not its PT.

As you can see PhoneA in IT_PT but CSS = None cannot call PhoneC and Local Area Call, but PhoneB without PT but CSS=CSS_B (with contain IT_PT, LOCAL_AREA_PT) can call PhoneC and Local Area Call.

3. Device/Destination (Phone, Route Pattern, Hunt Pilot, etc...) with PT=None can be accessed from any devices even those devices have CSS=None.

4. List of Partitions in CSS are determined in sequence.

In the picture PhoneA has CSS that can access both IT_PT and HR_PT but IT_PT is in the first list, when PhoneA dial 1002 it will always ring to PhoneB.

5. When you associate CSS in both Device and Line, the result is the combination of both CSS

Phone Designer

2010-07-11T15:39:00.006+07:00

Phone Designer is a free-to-download Cisco Unified Communications Widget for Cisco IP Phones.

You can quickly customize your Cisco Unified IP Phone display with wallpaper of your choice and create or change your ring tones to a preferred melody.

Prerequisite

1. Enable Phone Personalization in System> Enterprise Parameters

2. Create User Account and Associate to Phone. Select User Management> End User, click Add New enter User ID, Password and Last name. Then click save.

3. Click Device Association and associate your phone to this account.

4. Assign Standard CCM End Users Role to this account.

5. Download Phone Designer from Cisco Website (Require CCO Login) and install.

6. Open Phone Designer Program enter Account information, User ID, Password and Callmanager IP Address.

7. Now you can change your phone background by clicking image in the right and click Save to Phone, you can also import your own image using Import button.

8. Choose Ringtones tab and change your phone ring, you can also import your own ringtone using Import button.

Call Pickup in Callmanager hunt group

2010-07-11T14:03:00.003+07:00

Cisco Callmanager by default doesn't allow phone in hunt group to pickup each other even they're also in the same Call Pickup Group.

To enable this you need to edit some configuration.

1. From the CM Administration Page, Select System> Enterprise Parameters.
2. Find the section Cisco Support use, enter CSCsb42763 in Cisco Support Use1 and save.

3. In your hunt pilot configuration, choose Call Pickup Group in drop down list. So phones in this hunt group that have the same Call Pickup Group (both phone and hunt group) can pickup each other.

Bulk Administration Tool

2010-07-11T13:09:00.007+07:00

The Bulk Administration Tool (BAT), a web-based application, performs bulk transactions to the Cisco Unified Communications Manager database.

This article will show you how to add a large number of phones using BAT.

For easy to understand imagine that,

Completed Phones = Phone Template(Common Settings) + CSV File(Specific Settings)

Common Settings = Device Pool, Softkey Template, etc...
Specific Settings = MAC Address, Directory Number(DN), Name Display, etc...

Step1. At Cisco Unified CM Administration Page select Bulk Administration> Upload/Download Files.

Step2. Download bat.xlt from the server, use MS Excel to open this file (Need to enable macro).

Step3. Select Phones sheet and click Create File Format, we will generate csv file for specific setting.

Step4. Click Create and overwrite the existing format, now enter your phone details here, you need to key the MAC Address and Directory Number columns the others are optional.

Step5. After enter your phone details click Export to BAT Format, this will generate csv file with .txt extension.

Step6. Upload this csv file (phone.txt in my case) to Callmanager using Bulk Administration> Upload/Download Files menu.

Step7. Create Phone Template. Select Bulk Administration> Phones> Phone Template.

Step8. Click Add New and select phone type, protocol and others common settings, then Save. Note that these settings can be updated later using BAT menu.

Step9. Click Add a new DN to create a line template. Enter Line Template Name and other line common settings like Voice Mail Profile, Calling Search Space, Busy Trigger, etc...

Step10. Select Bulk Administration> Phones> Insert Phones, choose your csv file and template name.

Step11. At the bottom of the page select Run Immediately and click submit.

Step12. Select Bulk Administration> Job Scheduler to see the progress.

Step13. In case you've got the Number Of Records Failed, click Log File Name to see what error occur.

Step14. Select Device> Phone to check that your phone are added correctly.

Understanding Callmanager Services

2010-06-27T12:15:00.006+07:00

There are 2 groups of services in Callmanager.

1. Networks Services, these services are required for basic functionality you cannot activate or deactivate them. But you can stop/start/restart them in case of troubleshooting. The expected status of these services should be "Running".

2. Feature Services, these services are required for application feature, you can activate/deactivate/stop/start/restart them. Depending on your environment, you should activate only necessary services to reduce resource consumption.

Before activate these services I would recommend you to change CM hostname to IP Address in System> Server page configuration. This will tell the IP Phones to use IP Address instead when they're registering to Callmanager (DNS Server is required if you set it to hostname).

CM Services

Cisco CallManager - This service is used for call control function including features such as Call Pickup, Call Park, Hunt Group, etc...

Cisco TFTP - When IP Phone is booting up it will contact this service to get the list of Callmanager that it should register.

Cisco Messaging Interface - This service use the serial port of Callmanager to send SMDI message to 3rd party voice-mail system. In most case you don't need to activated this service.

Cisco Unified Mobile Voice Access Service - This service is required if you plan to use features like the following

1. Mobile Connect - Incoming call to IP Phone also ring to your mobile phone simultaneously

2. Enterprise Feature Access or Mobile Voice Access - You dial-in from your mobile phone to Callmanager and use the call control function as if you're using the IP Phone (I'll explain how to configure it in another article).

Cisco IP Voice Media Streaming App - This service is used for the application like music on hold (MoH), Annunciator.

Cisco CTIManager - This service is required for JTAPI/TAPI applications will connect such as Cisco UCCX, Arc Console Connect, Cisco Attendant Console, Cisco IPMA, QRT.

Cisco Extension Mobility - Activate this service if you plan to use extension mobility feature (You can login to IP Phone anywhere in the network to get your phone profiles).

Cisco Extended Functions - Activate this service if you plan to use Quality Report Tool (QRT) softkey to submit voice quality feedback from IP Phone back to the system.

Cisco Dialed Number Analyzer - This service is used for Numbering Plan Diagnostic. You can use this tool to analyze what the number pattern matched for a certain dialed-number.

Cisco DHCP Monitor Service - Activate this service if you use CM Server act as a DHCP Server.

CTI Services

Cisco CallManager Attendant Console Server - Activate this service if you plan to use attendant console software (free version). Note that beginning with V7.0 attendant console software will not be available to download from Callmanager plug-in page. But you can still use the software from V6.x to connect to this service in V7.x, Cisco has a plan to remove this service in V8.x.

Cisco IP Manager Assistant - Activate this service if you plan to use IPMA feature. If you don't know what it is just do not activate it.

Cisco WebDialer Web Service - Activate this service if you plan to use click-to-call feature (You can make an outgoing call by click the number on the web page or click-to-call program).

CDR Services

Cisco SOAP - CDRonDemand Service - Activate this service if you have 3rd party billing application that query CDR record directly to CM Server via SOAP.

Cisco CAR Web Service - Activate this service if you want to access CDR Analysis and Reporting page to see call data such as user call log, gateway utilization, etc...

Database and Admin Services

Cisco AXL Web Service - AXL Service is used for accepting the AXL client SOAP request to update Callmanager configuration. For example in hotel business will use PMS to send AXL SOAP request to update CSS to guest room phone, when guest checked-in they can call outside, when guest checked-out phone cannot call out.

Cisco UXL Web Service - Activate this service if you allow user to use Personal Address Book.

Cisco Bulk Provisioning Service - This service is used for administrator to add/update many phones at the same time, import phones from csv file, update all phone configuration, etc...

Cisco TAPS Service - This service is used to update phone MAC-Addresses when you create dummy MAC addresses for Callmanager auto-register. This service require Cisco CRS Server to function.

Performance and Monitoring Services

Cisco Serviceability Reporter - Activated this service to generate daily report for particular statistics, you access the archived report in Tools> Serviceability Reports Archive.

Cisco CallManager SNMP Service - Activated this service for server performance monitoring like CPU usage, memory usage, etc...

Security Services

Cisco CTL Provider - This service work with CTL Provider Utility to change security mode for the cluster. If you don't know what it is just do not activate it.

Cisco Certificate Authority Proxy Function - This service is used for issue/upgrade/retrieve/delete phone certificate, the example for phone certificate using is SRTP. If you don't know what it is just do not activate it.

Directory Services

Cisco DirSync - This service is used to integrate Callmanager user database with external directory service, for example you can integrate Callmanager user directory to Microsoft Active Directory user.

How to install cisco callmanager in VMware

2010-06-27T02:21:00.005+07:00

This article will show you the step by step to install Cisco Callmanager V7.0. It also apply to V6.x and V8.0.x with a little bit different menu.

Hardware Setup: VMware with 1GB of RAM, HDD 80GB, 1CPU.
Note: For Callmanager V8.0 2GB of RAM is required.

Step1. Media check, select No to skip (media checking quite take time to complete).

Step2. For V7.0 you will see the warning about VMware is not supported platform in production environment, select Agree to continue.

Step3. Product Deployment Selection, select Cisco Unified Communications Manager and click OK.

Step4. The installation will show you the current version on hard drive and the version that are going to be installed, just select Yes to continue.

Step5. Apply Patch, select No. (If you have patch you can apply it later).

Step6. Import Windows Data, select No.

Step7. Timezone Configuration, select your timezone.

Step8. NIC auto negotiation, select Yes.

Step9. MTU Size, select No to keep the default MTU.

Step10. DHCP Configuration, select No we will use static IP Address.

Step11. Enter Host name, IP Address, Subnet Mask and Gateway for your CM Server.

Step12. DNS Configuration, If you have DNS Server enter your DNS Server IP Address and domain otherwise select No.

Step13. Administration Login Configuration, enter your administrator id and password, this account will be used in console administration interface (command line).

Step14. Certificate Information, enter your detail to generate self-sign certificate for CM Server.

Step15. First Node Configuration, since this is our CM publisher server select Yes.

Step16. NTP Configuration, If you have NTP Server to synchronize time select Yes otherwise choose No.

Step17. If you don't use NTP Server the installation will prompt you to manually enter your date&time.

Step18. Security Password, enter password that used for secure communication between cluster (when you're installing CM subscriber you have to enter the same password here).

Step19. SMTP Configuration, If you have mail server select Yes otherwise choose No.

Step20. Application User, enter your administrator id and password, this account will be used in web administration.

Step21. After enter all information, Select OK to begin the installation.

Step22. Wait about 1 hour, finished.

Note: After install V7.0 in VMware I've found some issues.

1. Every time the image is booting up it'll always ask you to agree to used it in VMware.
2. CM Server never sync to NTP Server.

Upgrade to V7.1.3 will eliminate these issues and this version also install VMware tool during upgrade to improve performance.