IP Phone Agent & One Button Login

        Certain Contact Centers utilizing UCCX will be running/utilizing the Standard version rather than the Enhanced or Premium versions because it makes sense from a feature and cost perspective. Other Contact Centers have no real need for the Agent Desktop and find it as another application that they have to worry about.

        For certain scenarios it makes sense to utilize IP Phone Agent (IPPA), and if IPPA makes sense then One Button Login may make sense as well. Here are the steps to configure both:

IP Phone Agent:

1. Make sure that you have your users/agents created, assigned the correct roles/skills and associated in both CUCM and UCCX.
2. In CUCM select Device > Device Settings > Phone Services
3. Select "Add New"
4. IP Phone Agent Configurations:
1. Service Name - IP Phone Agent
2. ASCII Service Name - IP Phone Agent
3. Service Description - Cisco IP Phone Agent
4. Service URL - http://<UCCX IP Addr>:6293/ipphone/jsp/sciphonexml/IPAgentInitial.jsp
1. Note: Port 6293 is the Tomcat web server port
5. Check Enable (or Enterprise Subscription if ALL users are agents)
6. Click Save
5. Go to Device > Phone and search for the Agent's phone
6. In the "Related Links" drop down menu in the top right corner, select "Subscriber/Unsubscribe Services"
7. Select IP Phone Agent and click Subscribe
8. The phone should reset on its own after clicking Save

One Button Login:

1. Make sure that you have your users/agents created, assigned the correct roles/skills and associated in both CUCM and UCCX.
2. In CUCM select Device > Device Settings > Phone Services
3. Select "Add New"
4. One Button Login Configurations:
1. Service Name - IP Phone Agent
2. ASCII Service Name - IP Phone Agent
3. Service Description - Cisco IP Phone Agent
4. Service URL - http://<UCCX IP Addr>:6293/ipphone/jsp/sciphonexml/IPAgentLogin.jsp
1. Note: Port 6293 is the Tomcat web server port
5. Check Enable (or Enterprise Subscription if ALL users are agents)
6. Click Save
7. Under the Parameters Section select "New"
1. Parameter Name - Ext
2. Display Name - Ext
3. Default Value - N/A
4. Parameter Description - ICD Extension
5. Checkmark "Parameter is Required"
6. Click "Insert"
8. Under the Parameters Section select "New"
1. Parameter Name - Pwd
2. Display Name - Pwd
3. Default Value - N/A
4. Parameter Description - Agent Password
5. Checkmark "Parameter is Required"
6. Click "Insert"
9. Under the Parameters Section select "New"
1. Parameter Name - ID
2. Display Name - ID
3. Default Value - N/A
4. Parameter Description - Agent ID
5. Checkmark "Parameter is Required"
6. Click "Insert"
10.  Click Save
5. Go to Device > Phone and search for the Agent's phone
6. In the "Related Links" drop down menu in the top right corner, select "Subscriber/Unsubscribe Services"
7. Select One Button Login
8. Enter the specific phone/users ID, Password, and Extension
9. The phone should reset on its own after clicking Save

.

UCCX - Application Users

Three Application Users on CUCM need to be configured for UCCX and CUCM to interoperate.

UCCX_AXL - Unified CM Administrative XML Layer User

        The  Cisco AXL Web Service must be enabled to allow other appliances or applications to request from AVVID or the Administrative XML Layer. This access will allow certain methods (list, add, update, get, remove) when those other applications access the CUCM database. This is a great way to get read access to users, phones, features, settings, permissions, etc. within the CUCM database.

        To utilize the AXL Database API, an Application User will need that specific Role. However, by default, there is no User Group with the AXL Database API role configured, so you'll have to set up a new User Group.

1. Go to User Management > User Group and click Add New
2. For the Name type in "Standard AXL API Users"
3. In the top right corner under Related Links select "Assign Role to User Group" and click Go
4. Click on "Assign Role to User Group" and Add "Standard AXL API Access"
5. Click Save
6. Go to User Management > Application User and click Add New
7. For the Name type in "UCCX_AXL"
8. Click on "Assign Role to User Group" and Add "Standard CTI Enabled"
9. Click Save

UCCX_CTI - Unified CM Telephony User
        The Cisco Contact Center Express Telephony Integration Protocol (CTI) is a TCP/IP socket based message protocol that can receive event information for UCCX agents and calls. It is used to provide agent and call control, call information and call statistics.

1. Go to User Management > Application User and click Add New
2. For the Name type in "UCCX_CTI"
3. Click on "Assign Role to User Group" and Add "Standard CTI Enabled"
4. Click Save

UCCX_RMCM - Resource-Manager Contact-Manager Provider User
       The Resource Manager - Contact Manager (RmCm) allows the monitoring of agent's phones, routing/queueing of calls, control of agent states and management of historical reporting.

1. Go to User Management > Application User and click Add New
2. For the Name type in "UCCX_RMCM"
3. Click on "Assign Role to User Group" and Add "Standard CTI Enabled"
4. Click Save

.

VMWare VM Tools

In all guest operating systems it is very important that you install VMware Tools. VMware Tools are many of the drivers needed which will boost support and performance of that VM.

To install VMware Tools, launch the console of the VM from vSphere and select VM > Install VMware Tools from the workstation menu.

If you are running CUCM 8.5 and below, please refer to the applicable Cisco documentation on installing VMware Tools.

.

UCS C-Series Firmware Upgrade

The first step is to verify the CIMC Firmware and BIOS and perform an upgrade on the Firmware first and then the BIOS via the UCS Server Configuration Utility. If you upgrade the BIOS first, you must also upgrade the CIMC firmware or the server will not boot. This is why it is better to upgrade the CIMC firmware first.

http://www.cisco.com/en/US/products/ps10493/products_configuration_example09186a0080b07b7c.shtml

1. Verify your current server hardware Firmware version and BIOS version in CIMC by going to Server (tab) > Summary.
2. Download the upgrade ISO from Cisco’s website (roughly 200 MB)
1. Products > Unified Computing and Servers > Cisco UCS C-Series Rack-Mount Standalone Server Software > Cisco UCS C200 M2 Rack-Mount Server Software
2. Select Unified Computing System (UCS) Server Firmware
3. Verify your current version release and the latest one available
1. Web GUI into CIMC via the DHCP assigned IP Address (admin/password)
2. The Server Summary first page will list the BIOS Version, and Firmware Version
3. Under Admin tab > Firmware Management will be the CIMC Firmware Version
4. Mount the ISO locally and search for the CIMC folder
5. Find the BIN files that are needed for the upgrade
3. Setup your TFTP server with this file location as the directory
4. In CIMC go to Admin (tab) > Firmware Management
5. Click on "Install CIMC Firmware through Browser Client" or "Install CIMC Firmware from TFTP Server"
1. Click Browse and locate the "full" BIN file and click "Install Firmware"
2. Press TAB and enter the IP Address (if using TFTP)
3. Press TAB and enter the "full" upgrade file (if using TFTP)
1. This file MUST be of type .BIN or it will NOT work
2. Click Install Firmware
4. Install time takes roughly 15 minutes
6. Watch the "Last Firmware Install" window and wait for it to Complete Successfully
1. You should see a Running Version, Backup Version, and Boot-loader Version.
2. Click on "Activate CIMC Firmware
1. You will be given an option for the upgraded firmware
3. The management controller will be rebooted immediately after you click on "Activate Firmware" (this does NOT reboot the server, it ONLY reboots CIMC)
7. When CIMC is back up, confirm the Firmware Version
8. mount the image and access the utility via the KVM Console.

.

08.05 - Other Supplementary Services

Other Supplementary Services

08.04 - Callback

Callback

08.03 - Barge

Barge

08.02 - Call Pickup

Call Pickup

08.01 - Call Park

Call Park

08.00 - Implement and TSHOOT Supplementary Servies

Implement and TSHOOT Supplementary Services

07.06 - Other CUCM Media Resources

Other CUCM Media Resources

07.05 - Media Resources Preference and Redundancy

Media Resources Preference and Redundancy

07.04 - Music On Hold (MOH)

Music On Hold (MOH)

07.03 - Transcoder (XCODE)

Transcoder (XCODE)

07.02 - Conference Bridges (CFB)

Conference Bridges (CFB)

07.01 - CODEC Selection and Flexibility

CODEC Selection and Flexibility

07.00 - Implement and TSHOOT Media Resources

Media Resources

06.02 - Automated Alternate Routing (AAR)

Automated Alternate Routing (AAR)

06.01 - Survivable Remote Site Telephony (SRST)

Survivable Remote Site Telephony (SRST)

06.00 - Implement and TSHOOT High Availability Features

High Availability Features

05.05 - Mobility and Single Number Reach

 Mobility and Single Number Reach

05.04 - Route Selection Preference and Redundancy

Route Selection Preference and Redundancy

05.03 - Class of Services

Class of Services

05.02 - Digit Manipulations and Translations

Digit Manipulations and Translations

05.01 - Route Patterns and Dial-Peers

Route Patterns and Dial-Peers

05.00 - Implement and TSHOOT Call Routing

Implement and TSHOOT Call Routing

04.07 - IP-IP Gateway / CUBE

IP-IP Gateway / CUBE

04.05 - H.323 RAS

H.323 RAS

04.05 - SIP

SIP

04.04 - MGCP

MGCP

04.03 - H.323

H.323

04.02 - T1/E1 CAS

T1/E1 CAS

04.01 - T1/E1 PRI

T1/E1 PRI

04.00 - Implement and TSHOOT Voice Gateways

Implement and Troubleshoot Voice Gateways

03.02 - CME SIP Endpoints

CME SIP Endpoints

03.01 - CME SCCP Endpoints

CME SCCP Endpoints

Create a Bootable ISO from an Cisco Upgrade ISO

        Cisco does not provide bootable ISO files for any UC Software by download from its website. This is extremely problematic when install discs are lost or when you need to restore from backup and the current version is an upgraded from the original install.

        Luckily, Cisco's upgrade ISO files that are not bootable contain all of the information for an upgrade AND an install. The only file missing is a BOOT file (.bif - Bootable Info File). If you have a bootable disc from Cisco, you can extract this BOOT.bif file and combine it with upgrade ISOs to make a bootable install disc of any version that is available for download from Cisco.com!

        This process is not supported by Cisco and we may run into a time when Cisco modifies their upgrade files so that simply adding a boot file will not make it bootable. But that time is not today.

        Use your favorite disc/image tool (I use UltraISO) and place a bootable Cisco install disc into your PC. Open up that disc with UltraISO and select from the menu bar "Bootable > Extract Boot File from CD/DVD...". This will extract the .bif file and prompt you to save it in a certain location. You can rename this file to something like "Boot.bif". 

        Next, download the UC upgrade file from Cisco.com and verify the MD5 hash. Open this ISO file with UltraISO. Note that in the upper left-hand corner that the type is of "Data CD/DVD". From the menu bar, select "Bootable > Generate Bootinfotable". Once that is check-marked, select "Bootable > Load Boot File..." and choose the Boot.bif you had extracted earlier.

       The image type in the upper left corner of UltraISO should now be "Bootable CD/DVD". Go to "File > Save As" and save the ISO as "Bootable_<ISO file name>". Feel free to burn this bootable image to a CD/DVD and install away!!

.

Music On Hold (MOH)

Music On Hold allows callers to hear music while on hold.

CTI Devices do not support Multicast

Audio Source Format

1. Most WAV files are supported
1. 16-bit PCM (stereo/mono)
2. 8-bit CCITT a-law or mu-law (stereo/mono)
3. 48kHz, 32kHz, 16kHz, 8kHz sample rates
2. MP3 format is NOT supported
3. 
   

Resource Selection

        If Phone A puts Phone B on hold, the Audio Source within Phone A's configuration are selected, but Phone B's Media Resource Group is utilized. This optimizes the Media Resource utilization as Phone B will use the resources closest to it.

MOH Audio Sources are selected in the following order:

1. Line
2. Device
3. Common Device Configuration (CDC)
4. Clusterwide Parameters

Media Resource Group Lists (MRGL) are selected in the following order:

1. Device
2. Device Pool
3. System Default
1. System default resources contain any Media Resources not utilized by ANY Media Resource Groups (MRG)
2. Always Unicast

.

Extension Mobility Configuration

1. Verify Extension Mobility Service is Running
1. From CUCM Serviceability Select  Tools > Control Center – Feature Services
2. Make sure that the Cisco Extension Mobility service shows status Activated
2. Configure Extension Mobility Service
1. From CUCM Administration select  Device > Device Settings > Phone Services
2. Click Add New
3. In the Service Name field, type Extension Mobility
4. In the Service Description field, type Login and Logout Service
5. In the Service URL field, Enter the following URL
1. http://<CUCM IP Address>/emapp/EMAppServlet?device=#DEVICENAME#
6. Click Save
3. Modify Enterprise Parameters to Reflect IP Address of CallManager
1. From CUCM Administration select  System > Enterprise Parameters
2. Under Phone URL parameters, change all fields from hostname to IP address. Change ONLY the host name, not the reset of the field.
3. Click Save
4. Create Device Profile Default for Each Phone Model that shall Support Cisco Extension Mobility
1. From CUCM Administration select  Device > Device Settings > Default Device Profile
2. From the drop down list, select the phone type, for example, Cisco 7965
3. Under Description, enter a description of this profile.
4. Under Phone Button Template, select Standard 7965 SCCP.
5. Click Save
6. Repeat for each model phone to be configured
5. Create Device User Profile for a User
1. From CUCM Administration select  Device > Device Settings > Device Profile
2. Click Add New
3. From the drop down list, select the phone type, for example, Cisco 7965
4. Click Next
5. Enter a Device Profile Name (username-EM)
6. From the Phone Button Template field, select Standard 79XX SCCP
7. Click Save
8. On the left hand side of the screen, click the link Line [1] – Add a new DN
9. Choose a valid DN, enter that DN in the Directory Number field.
10. Under Route Partition, select INTERNAL-PT
11. Enter any Call Forward and Call Pickup Settings as necessary.
12. Click Save
13. Select  Related Links > Subscribe/Unsubscribe Services
14. From Services, select Extension Mobility, then click Next.
15. Click Subscribe
16. Click Save
17. Repeat steps 7-13 for any additional lines.
6. Associate User Device Profile to a User
1. From CUCM Administration select User Management > End User.
2. Select the user from the list that matches the profile that was created.
3. Under Extension Mobility > Available Profiles, select the profile that was created in the previous exercise and move it to the Controlled Profiles selection.
4. Under Default Profile, select the profile.
5. Click Save.
7. Configure and Subscribe Cisco Unified IP Phones to Service and Enable it.
1. Select Device>Phone from the menu.
2. Select the phone from the list of devices.
3. In the Related Links: field, select Subscribe/Unsubscribe Services and click Go
4. In the pop-up window, under Service Information, in the Select a Service pull down menu, elect Extension Mobility.
5. Click Next
6. Click Subscribe
7. Click Save
8. Close the pop-up window.
9. Under Extension Information, check the Enable Extension Mobility box.
10. Under the Logout Profile field, select – Use Current Device Settings –
11. Click Save.
12. Click Ok from the pop-up warning.
13. Click Reset
14. In the pop-up window select Reset.
15. Click Close.

.

IP Manager Assistant (IPMA) - Shared Line Mode

        IP Manager Assistant (IPMA) provides better support, visibility, and control between a Manager and an Assistant or multiple Assistants. There are two ways of configuring IPMA. The first being Proxy Mode and the second being shared-line mode. 

Configuring Shared Line Mode

1. Configure the Manager phone and Assistant(s) phones with a shared line appearance.
2. In Cisco Unified Serviceability, activate the Cisco IP Manager Assistant service in the Service Activation window.
3. Configure Cisco IP Manager Assistant service parameters for shared line support
1. Clusterwide Parameters (applies to all servers)
1. In CM Administration go to System > Service Parameters
1. Choose the IPMA server and choose the CIsco IP Manager Assistant Service
2. Cisco IPMA Server (Primary) - up to 2500 managers and assistants
3. Cisco IPMA Server (Backup)
4. Cisco IPMA Server Port - Default Port 2912
5. IPMA Heartbeat Interval - Time for failover to occur (default is 30 seconds)
6. IPMA Assistant Console Request timeout - Default is 30 seconds
2. Advanced Clusterwide
1. Enable Multiple Active Mode - default is false. If true, both IPMA servers are used and up to 7000 managers and assistants can be used via multiple pools
2. Pool 2 - IPMA Server (Primary) - 2500 users here
3. Pool 2 - IPMA Server (Backup)
4. Pool 3 - IPMA Server (Primary) - 2500 users here
5. Pool 3 - IPMA Server (Backup)
3. IPMA Service Parameters (for each server)
1. CTIManager (Primary) IP Address - used for call control
2. CTIManager (Backup) IP Address
4. Restart the IPMA Service for the changes to take effect
5. In CM Administrator go to User Management > End User
6. Start with the Manager
1. Verify that he/she is associated with their phone
2. Under Related Links, select Manager Configuration
3. Configure the device, intercom, shared lines, and assistant(s)
4. Once the Manager is setup, their phone will reset with IPMA features on the phone
7. Configure the Assistant
1. Verify that he/she is associated with their phone
2. Under Related Links, select Assistant Configuration
3. Configure the device, intercom, primary line, managers, and manager lines
8. You can setup dial rules (prepend 9) for the IPMA
9. Download and install Cisco Unified Communication Manager Assistant
1. If running Windows 7, download through Cisco CallManager Updates
10. Install and login with the correct IPMA server IP address, username and password

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmfeat/fsipmasl.html#wp1288368

.

03.00 - Implement/TSHOOT CME Endpoints

Implement and Troubleshoot CME Endpoints

02.03 - CUCM Session Initiation Protocol (SIP)

Session Initiation Protocol (SIP)

02.01 - CUCM Skinny (SCCP) Endpoints

CUCM Skinny Protocol (SCCP)

02.00 - Implement/TSHOOT CUCM Endpoints

Implement/TSHOOT CUCM Endpoints

01.00 - Campus Infrastructure

Campus Infrastructure, Protocols, and Services

01.05 - Domain Name System

Domain Name System

01.04 - Power over Ethernet (PoE)

Power over Ethernet (PoE)

01.02 - Trivial File Transfer Protocol (TFTP)

TFTP

01.01 - Virtual Local Area Network (VLAN)

Virtual Local Area Network (VLAN)

CME as SRST

CUCM Configuration:

1. Go to the CM Administration page and select System > SRST
2. Click on Add New
3. Configure the Name as SiteCode_IPAddress (eg. NY_10.1.100.2)
4. Set the IP Address for both IP Address and SIP IP Address
5. Click Save
6. Go to System > Device Pool
7. Select the Remote Site Device Pool
8. Configure the corresponding SRST within the Device Pool
9. Click Save

VGW Configruation (CME-as-SRST):

1. license accept end user agreement (if running IOS Version 15.x or higher)
2. license modify priority cme-srst high (if running IOS Version 15.x or higher)
3. telephony-service
4. srst mode auto-provision none
5. srst dn template 1
6. srst dn line-mode octo
7. max-ephones 110
8. max-dn 300
9. ip source-address 10.1.89.2 port 2000
10. system message Life Alert Fallback
11. time-zone 5
12. max-conferences 4 gain -6
13. dn-webedit  
14. time-webedit  
15. transfer-system full-consult
16. transfer-pattern ....
17. transfer-pattern 9...........
18. transfer-pattern 9.......
19. secondary-dialtone 9

        Dial-peer configurations, conferencing or media resources, and MOH configuration would still be needed in this VGW configuration.




.

SRST versus CME-in-SRST

Comparison chart of CME-in-SRST and SRST:

http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/vcallcon/ps2169/prod_qas0900aecd8028d113.html

The main questions to ask yourself whether to go for CME-as-SRST or SRST are:

1. How many phones do I need to support?
2. What features do I need during failover
3. Do I need to support a VG248 during failover?
4. Do I need to support hunt groups or call park?
5. Is conferencing required?
6. Are we utilizing secure voice? 

 



.

Enhanced Survivable Remote Site Telephony (E-SRST)

Enhanced SRST (E-SRST) advantages over SRST:

• Automatic provisioning - it is integrated with CUCM and can be centrally configured/deployed from CUCM
• Automatic Synchronization - the Cisco Unified Messaging Gateway (required piece of hardware/software for E-SRST) synchronizes the configuration from the central site to the branch on a schedule. 
• Consistent Device Layout - phone displays, basic functions (extensions, soft-key templates, phone types, etc) are carried over in E-SRST.
• Full call control features during failover
•  Call forward no answer, call forward all, call forward busy
• Time of Day routing
• Class of Service restrictions for both incoming and outgoing
• Hunt Groups
• Call Park and Call Pickup

SRST advantages of Enhanced SRST (E-SRST):

• Lots cheaper
• No extra licenses required other than basic SRST licenses
• Unified Messaging Gateway not needed
• SIP SRST supported
• Who needs to pay the extra money to support call park/pickup, hunt groups, and TOD!?!

.

Survivable Remote Site Telephony (SRST)

        How do you design Unified Communications failover for your remote sites or your primary site with a single CUCM Publisher or CUCM Business Edition? Survivable Remote Site Telephony (SRST) is the answer! SRST is a Cisco IOS Software capability that acts as a backup call-processing agent in the event of a link (typically WAN) failure, providing users with a subset of capabilities. However, the choice is not that easy as there are multiple versions and configuration scenarios:

• SRST
• H.323
• MGCP
• SIP
• CME as SRST
• Enhanced SRST
• Dual SRST

        In order to decide on which one to select you have to decide on the foundation first. How many users are at each SRST site and what phone models are utilized? The number of users and phone types will help you decide which SRST to select, which router to utilize, and what code and licensing you'll need to deploy. 

Requirements:

• The SRST routers need sufficient PSTN paths (FXO, PRI, BRI, etc.) to support the remote site users
• Decide on a list of requirements for SRST
• Inbound and outbound PSTN connectivity to all phones
• Voicemail
• Call Forward No Answer or Busy
• Extension expansion to full 10-digit DIDs of other sites
• Music on Hold locally streamed from SRST router
• Conferencing

CUCM Configuration:

1. Go to the CM Administration page and select System > SRST
2. Click on Add New
3. Configure the Name as SiteCode_IPAddress (eg. NY_10.1.100.2)
4. Set the IP Address for both IP Address and SIP IP Address
5. Click Save
6. Go to System > Device Pool
7. Select the Remote Site Device Pool
8. Configure the corresponding SRST within the Device Pool
9. Click Save

 

VGW Configurations:

1. license accept end user agreement (if running IOS Version 15.x or higher)
2. license modify priority cme-srst high (if running IOS Version 15.x or higher)
3. clock timezone [zone] [hours-offset]
4. call-manager-fallback
5. ip source-address [ip address] [port]
6. max-dn [max-directory-numbers] [dual-line | octo-line] [preference] [number]
7. huntstop channel [1-8]
8. max-ephone [max-phones]
9. date-format [mm-dd-yy | dd-mm-yy | yy-dd-mm | yy-mm-dd]
10. time-format [12 | 24]
11. system message [primary | secondary] [custom message]
12. secondary-dialtone [digit-string]

Example

1. clock timezone PST -8
2. call-manager-fallback
3. ip source-address 10.1.100.2  2000
4. max-dn 300 octo-line
5. huntstop channel 6
6. max-ephone 100
7. date-format mm-dd-yy
8. time-format 12
9. system message primary CM Fallback
10. secondary-dialtone 9

Verifying that SRST is enabled:

1. Show running-config
2. show call-manager-fallback all
3. show ephone (while in fallback)
4. debug ephone [keepalive | register | state | detail | error | statistics | pak | raw]

.

Unity Connection SQL Commands

To find the Informix User ID of a user using a SQL query:

run cuc dbquery unitydirdb SELECT ObjectId from vw_User WHERE Alias = ‘aliasofunityuser’

To force delete that user:

run cuc dbquery unitydirdb EXECUTE PROCEDURE csp_UserDelete (pObjectId = ‘objectid that resulted from above query’)



.

UCS200 M2 Install

        Download ESXi version 4.1 or 5.x and burn the ISO to a CD (roughly 300MB). Place disc into UCS200 and follow the configuration steps and complete the install.

        When the UCS restarts after installing, make sure the install ISO is removed. Connect the management console port to your network and it will receive an IP address via DHCP (you must watch the boot up process for this IP Address information. Use your web browser to connect to that address to access the Cisco Integrated Management Controller (CIMC). The default credentials are “admin” and “password”.

        The first step is to verify the CIMC firmware, system BIOS, LAN on motherboard (LOM), LSI, and Cisco Virtual Interface Card (VIC). If there are newer versions, it is best to upgrade to the latest stable release. The great thing is that Cisco has a "Host Upgrade Utility" tool that upgrades all of the firmwares and in the correct order (this only works if your system firmware is at version 1.2 or higher).
http://www.cisco.com/en/US/docs/unified_computing/ucs/c/sw/lomug/1.3.x/install/HUUUG_chapter1.html

1. Verify your current server hardware Firmware version and BIOS version in CIMC by going to Server (tab) > Summary.
2. Download the Host Upgrade Utility ISO from Cisco’s website (roughly 200 MB)
1. Products > Unified Computing and Servers > Cisco UCS C-eries Rack-Mount Standalone Server Software > Cisco UCS C200 M2 Rack-Mount Server Software > Unified Computing System (UCS) Server Firmware
2. Compare your current version release and the latest one available
3. Download and burn the ISO (ucs-c200-huu-1.4.1c.iso - 11/06/2011)
3. Insert the disc into the C-Series chassis and reload the machine. Make sure you have KVM connectivity.
4. The Cisco Host Upgrade Utility loads. Follow the steps to upgrade all available options.
1. Cisco Integrated Management Controller (CIMC)
2. System BIOS
3. LAN on motherboard (LOM)
4. LSI (for any third-party LSI storage controllers that are installed)
5. Cisco UCS P81E Virtual Interface Card (VIC)
5. Eject the disc and reboot for all of the upgrades to take effect
6. Verify all upgrades

        Next mount the VMWare ESXi ISO and follow the install. Set a password (no password by default) along with a static management IP address and run your VMware vSphere Client to access that address.

        Download and install vSphere onto your computer. This is the software that lets you manage the ESXi host and work on virtual machines.

        You can now deploy OVA files and build your UC servers!




.

Software Conference Bridge Resources

        How many conference participants can you support with only software based (CallManager) resources?

Per the Cisco Unified Communications Manager System Guide, Release 8.5(1) the maximum number of audio streams for G.711 is 128. This means G.711 software conferencing can support up to any combination of conference(s) equaling 128 users (eg. 1 conference of 128, or 3 conferences of 48 participants). However, the default is set to 48 and it is NOT recommended that you increase that number if you are running both the Cisco IP Voice Media Streaming Application service on the same server as the Cisco CallManager service.

        When reviewing the maximum number of conferences on CUCM Service Parameters, the Call Count maximum is actually 256, rather than the max of 128 listed in the System Guide documentation.

        G.711 is the only codec supported at default. You can modify this setting within Cisco Unified CM Administration by going to System > Service Parameters, selecting the server and then selecting the Cisco IP Voice Media Streaming App. You can add additional Supported MOH Codecs (711 mulaw, 711 alaw, 729 Annex A, wideband). If you select G729 (729 Annex A), a warning pops up stating that “this codec is optimized for speech, and the audio fidelity of music is marginal”.

        To set the maximum participants for any single Ad Hoc and MeetMe Conference, go to the conference section with the CUCM Service Parameters. The default is 4 maximum participants for a single Ad Hoc or MeetMe conference which is low.



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International Dialing – VGW

When dialing internationally, each carrier can differ based on the ISDN type they accept.  To override the default ISDN type and plan generated by the router, use the command below within your Serial interface:

isdn map address [[address | reg-exp] plan plan type type | transparent]

The carrier may reject your default “International” ISDN type which you should notice from a Q.931 debug:

Called Party Number i = 0×91, ’0118704421234′
Plan:ISDN, Type:International
Aug 18 18:12:25: ISDN Se0/0/0:23 Q931: RX <- RELEASE_COMP pd = 8  callref = 0×8112
Cause i = 0x809C – Invalid number format (incomplete number)

Example:

interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-ni
isdn incoming-voice voice
isdn map address 011.* plan isdn type unknown
no cdp enable

…

dial-peer voice 90111 pots
corlist outgoing call-international    (CME ONLY)
preference 1
destination-pattern 9011T
port 0/0/0:23
prefix 011



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Block Toll Fraud Numbers – Route Filter

Toll Fraud List

        There are area codes that can be reached from within the United States or Canada without dialing an international code. This is where Toll Fraud is a huge issue and should be blocked on all systems. You may want to create an “Executive” Calling Search Space which will allow any/all calls to go through unfiltered, but only at the client’s request.

242     Bahamas

246     Barbados

264     Anguilla (split from 809)

268     Antigua and Barbuda

284     British Virgin Islands

340     US Virgin Islands: St Thomas, St John

345     Cayman Islands

441     Bermuda

473     Grenada

649     Turks and Caicos Islands

664     Montserrat

670     Northern Mariana Islands

671     Guam

758     St. Lucia

767     Dominica

784     St. Vincent and Grenada

787     Puerto Rico

809    Caribbean, Bermuda, Puerto Rico, Virgin Islands

868    Trinidad and Tobago

869    St. Kitts/Nevis

876    Jamaica

900    Pay_Per_Call Numbers

939    Puerto Rico

976    Pay_Per_Call Numbers



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CUCM - Automated Alternate Routing (AAR)

Automated Alternate Routing (AAR) is a high-availability feature that takes a call which would normally traverse the WAN, but is re-routed through the PSTN because the set bandwidth has been reached between two locations.

To be honest, I haven't deployed AAR at many sites. I guess this has a lot to do with the fact that I'm either really lazy, or that the cost of bandwidth has gone down so much that bandwidth and call admission control (CAC) hasn't been much of a customer requirement, or it's probably a combination of both.

If you would like to deploy AAR, the following steps and options are below, but keep in mind AAR ties directly to Call Admission Control (CAC) which means you'll have to configure and/or pay attention to Location and Region settings.


Configuring Automated Alternate Routing (AAR)

1. Turn the service on!!
1. System > Service Parameters
2. Set “Automated Alternate Routing Enabled” to True. It is False by default.
3. (Optional) Change “AAR Network Congestion Rerouting Text” to alert users that call is being re-routed. By default it is set to “Network Congestion. Rerouting.”
4. If you don't turn this on, callers will experience a re-order tone
2. Call Routing > Automated Alternate Routing Group
3. Create XXX-AAR for each site
1. You can actually create a single AAR group for NANP sites, but it depends on a few things. I would recommend creating an AAR Group for each site to keep everything clean and less confusing for administrators.
4. Set Prefix Digits Within to 9 (Local) or 91 (Long Distance) for AAR calls between phones within this group
5. Set Prefix Digits Between to 9 (Local) or 91 (Long Distance) for AAR calls between phones from AAR Group A to AAR Group B. (this field does not show up if you have ONLY 1 AAR group)
6. AAR Group and CSS
1. The AAR Group and CSS can be set on the Device Pool or the Phone, gateway, or endpoint.
7. Extension AAR Settings
1. Make sure that you have the correct External Phone Number Mask (EPNM) for each phone. This is how AAR figures out what number and how many digits to dial after prepending 91 or 9 within the AAR Group. 
2. If, for some reason, you want AAR triggered calls to go to voicemail or to a different number/group altogether, you can configure these options within "AAR Settings" on the extension
3. The AAR Settings options (voicemail or AAR Destination Mask) take precedence over the External Phone Number Mask
8. Lastly, the AAR triggered number is matched to a route pattern using the AAR Calling Search Space. If an outbound route is matched, the call is pushed out through the PSTN.

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Single Number Reach and MobileConnect

        Single Number Reach allows users to give out a single number and be reached on multiple devices (maximum of four). When calling a user’s Single Number Reach DID, four seconds (default) go by before ringing the user’s remote destinations (eg. cell phone, home phone, etc.). Once the remote destination is ringing, the Send Call to Mobile Menu Timer Service Parameter (default 60 seconds) defines the amount of time the remote destination(s) has to pick up the call.

        Users/Administrators can set the schedule (days/times) that SNR is active for each of their remote destinations though the ccmuser or appadmin GUI.

        MobileConnect allows users to receive a call on their primary device (desk phone) and press the Mobility soft-key or line button which calls their remote destination device (eg. cell phone). Once the user picks up their remote device, the call is connected on their remote device and the person they are talking to is none the wiser. MobileConnect also allows users to hang up their remote destination device and seamlessly pick the call up on their primary phone (default time to pick up is 10 seconds).

How to configure Single Number Reach and MobileConnect:

1. Create End User and verify that the user is associated to their phone and that “Enable Mobility” (uses 4 device licenses per user, and 2 if adjunct) and “Enable Mobile Voice Access” is checked along with their Primary User Device selected.
1. Selecting the Primary User Device takes the Mobility Enabled End User license and changes it into an Adjunct license.
2. Under Enable Mobile Voice Access is the Maximum Wait Time for Desk Pickup which is defaulted at 10 seconds. This is the time (in milliseconds) that the user has to pick up the MobileConnect call after hanging up their remote destination device. It is recommended that if users are not at their desk, that they let the other party hang up first as their call may be hijacked at their desk.
2. Create/modify a SoftKey Template to include Mobility.
1. If the user’s phone is a 8900/9900 series phone, a Phone Button Template must be created with a Mobility line key.
3. On the user’s phone, apply the new softkey (or phone button template) and select the user under the Owner User ID.
1. If the Owner User ID is not selected, MobileConnect will fail with a message stating “You are not a valid Mobile Phone User”.
2. If the Owner User ID is grayed out, it is because you have Extension Mobility selected. The Cisco Help Guide states that you should not have both EM and Owner User ID selected, but I’ve tested and it works fine.
4. Under Device > Device Settings > Remote Destination Profile create a new RDP for each MobileConnect/SNR user.
1. For the call to actually reach the external number, the Rerouting Calling Search Space is the CSS that controls this.
2. You must select a Rerouting CSS that can reach the PSTN or the SNR will fail for external numbers.
3. Make sure the Privacy setting is set to "Default"
5. Under Device > Destination Profile create a new RD for each MobileConnect/SNR user.
1. You must put the user’s remote destination here along with the out-dialing number(s) such as 91 or 81, unless it is an internally reachable extension. Always associate the line for SNR to work.
2. Instead of using 91 or 81 for dialing out, you can also set an Application Dial Rule which will be triggered for SNR.
6. Under System > Service Parameters, select the Cisco CallManager service.
1. Verify that "Privacy Setting" is set to True (default)
2. Verify that "Enforce Privacy Settings on Held Calls" is set to False (default)
3. You cannot transfer a call from a cell phone to a desk phone if the Remote Destination Profile Privacy specifies On, and the "Enforce Privacy Setting on Held Calls" service parameter specifies True.
7. Reset the user’s phone and test SNR and MobileConnect.

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