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.



.

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



.

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



.

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.

.

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.

.