Tuesday, February 28, 2012

My 2nd voice lab attempt failed

I failed and the worst part is I think I am doing well and I can't recall I have done anything wrong.  Now the most important thing is to revisit my work and see what I have missed.  Hope to make my 3rd attempt in a couple of months time.

Sunday, February 26, 2012

Ready to make my second voice lab exam attempt tomorrow, wish me luck!

MGCP Gateway Fallback Configuration Example

When you are deploying MGCP in branch office and the WAN connection between the MGCP gateway and UCM is broken, you can enable fallback on IOS gateway and route call based on your H.323 configuration.

These are the commands that you need to configure on your MGCP gateway.

R1(config)#ccm-manager fallback-mgcp

R1(config-app-global)#service alternate Default

Also configure the necessary incoming and outgoing voip and pots dial-peer as well as SRST configuration to handle the fallback situation.

VoiceView Express - Authentication Error

If you try to setup VoiceView Express for a CUE + UCM integration, and get the error:

Playback failed
Authentication error.  Report this error to your system administrator.

when you try to playback a message on Voiceview express, very likely you don't have the JTAPI user associated with the phone that use this service.  Make sure the JTAPI user is associated to the phones that are configured to use VoiceView Express.

Cisco UCM - Voice Mail Box Mask on Voice Mail Profile

The mask only mask the ReDir number.  Not the caller number, therefore:

When branch phone 5002 calls 5001 redir to 1220 (VM Pilot number, CUC in hub site) in SRST mode, it will mask the redir number from 98765001 (DID number) to 5001, therefore the caller 5002 can hear 5001's greeting instead of a general mailbox greeting.

However when 5001 press the voicemail envelop button on phone, the CLID is 98765001 therefore user 5001 doesn't able to listen to his personal greeting, as CUC doesn't recognize a subscriber has 98765001. It is not handled by the voice mail box mask.

One of the way is to enter 98765001 as alternate extension for the subscriber 5001.  If alternate extension is not a solution, you can use calling party transformation on UCM to manipulate the calling number, say for example create a new device pool for the voice mail ports on UCM and apply the calling party transformation CSS to this new device pool.

Cisco UCM + CUE integration - in SRST scenario

In previous post I have highlighted the steps and procedures that are required for a UCM + CUE JTAPI integration:


What if the CUE is located at a branch site and the connection between UCM and CUE is broken?  When the WAN link is downed, SRST kicks in and the CUE can change from a JTAPI integration into a SIP integration, so that the SRST router can talk to CUE via SIP.

0. Assume you have the JTAPI integration in place, for normal situation when the connection between hub site and branch is up:

1. Configure CUE SIP subsystem, the gateway address is your SRST router address:

ccn subsystem sip
gateway address ""
mwi sip unsolicited

2. Configure SIP trigger, the voicemail pilot number:

ccn trigger sip phonenumber 1110
application "voicemail"

3. Configure the SIP dial-peer on your router

dial-peer voice 1110 voip
destination-pattern ^1110$
session protocol sipv2
session target ipv4:
dtmf-relay sip-notify
codec g711ulaw

4. Configure the SIP MWI:

mwi-server ipv4: expires 3600 port 5060 transport udp unsolicited

5.  Sample CME as SRST configuration
srst mode auto-provision none
srst ephone template 1
srst dn template 1
srst dn line-mode octo
max-ephones 20
max-dn 20 no-reg
ip source-address port 2000
voicemail 1110
mwi relay
max-conferences 8 gain -6
transfer-system full-consult
create cnf-files version-stamp 7960 Feb 25 2012 01:03:36
ephone-dn-template  1
call-forward busy 1110
call-forward noan 1110 timeout 20
mwi sip
huntstop channel 1

When the WAN link is downed, your branch phone can still access your local CUE, leave and retrieve voicemail as usual and the end user experience, MWI, etc will be the same. 

Cisco B-ACD on CME router - Drop-through option

In Cisco CME B-ACD, drop-through option can be configured so that when the AA services receive the incoming call, it will send the call directly to the call queue without provide menu options to callers.  This is useful for some small contact center backup, when the WAN link is downed and the site can't reach the contact center express or enterprise in the main site, it can still provide basic ACD service to the callers.

My Sample Configuration:

service aa flash:app-b-acd-aa-
  paramspace english index 0
  param number-of-hunt-grps 1
  param drop-through-option 1
  param handoff-string aa
  paramspace english language en
  param max-time-vm-retry 2
  param aa-pilot 5000
  paramspace english location flash:
  param second-greeting-time 60
  param drop-through-prompt _dt_prompt.au
  param call-retry-timer 15
  param voice-mail 5555
  param max-time-call-retry 700
  param service-name callq
service callq flash:app-b-acd-
  param queue-len 10
  param aa-hunt1 5600
  param queue-manager-debugs 1
  param number-of-hunt-grps 1

dial-peer voice 5000 pots
service aa
incoming called-number 5000

ephone-hunt 1 longest-idle
pilot 5600
list 5111, 5112

When there is incoming call, on both of the phone with ephone-dn 5111 and 5112, the message "1 call in queue" will be shown on the phone screen and you will know how many callers waiting in the call queue.

For more information:

MGCP Signaling QoS marking

According to show mgcp, the default marking for MGCP signaling DSCP marking is AF31 (26):

MGCP media (RTP) dscp: ef, MGCP signaling dscp: af31

It can be override with the command:  mgcp ip qos dscp cs3 signaling
MGCP media (RTP) dscp: ef, MGCP signaling dscp: cs3

As a result the signaling is now marked as CS3 (24) instead of AF31 (26)

Difference between "set dscp" and "set ip dscp" under policy-map

They are more or less the same in the IPv4 world.  The only difference is, set dscp will work for IPv4 and IPv6 and set ip dscp is for IPv4 only.

pandasw02(config-pmap-c)#set ?
  dscp        Set DSCP in IP(v4) and IPv6 packets
  ip          Set IP specific values
  precedence  Set precedence in IP(v4) and IPv6 packets

pandasw02(config-pmap-c)#set ip ?
  dscp        Set IP DSCP (DiffServ CodePoint)
  precedence  Set IP precedence

Thursday, February 16, 2012

Off-hook alerting configuration on UCM

This is one of the common compliance clause in RFP (especially hotel, hospital) which requires off-hook alerting, which means after the phone has been off-hook for a time that expires the "Off-hook to First Digit Timer, by default 15s", a call will be placed automatically to a predefined number, usually an operator number.  You can achieve it by creating PLAR on UCM, unchecking the "Urgent Priority" in the empty translation pattern.  Here is the example on how it can be done:

1. Create a partition called PT-FOR-PLAR (for translation pattern), PT-TO-PLAR (for destination phone)
2. Create a CSS called CSS-FOR-PLAR, and include the PT-FOR-PLAR partition.  Or simply include this partition to your existing phone CSS.  
3. Apply CSS-FOR-PLAR to the phone if you haven't done so.
4. Create another CSS called CSS-TO-PLAR, which includes the partition PT-TO-PLAR
5. Create a translation pattern with empty pattern.  The partition is PT-FOR-PLAR, and the CSS is PT-TO-PLAR.  The called party transformation is configured with the number of your destinated called party number.  Say for example my operator number is 1234, the called party transformation prefix is 1234.  Remember to "uncheck" the "Urgent Priority" so that the call will not setup immediately, instead it will wait for the time that is configured in "Off-hook to First Digit Timer".
6. Your operator DN 1234 should have partition PT-TO-PLAR or other partition in CSS-TO-PLAR

The experience is when you put your phone off-hook, after 15 sec, call will be made automatically to operator 1234.

For more information about PLAR configuration, please visit the previous post:

Wednesday, February 15, 2012

Comparing N5K and N7K - a very high level key features comparison

- FC
- Unified Ports
- EvPC


Both of them support FabricPath (N7K F1 and F2 cards), vPC, L3, FEX, etc

ASR 1000 OTV supports on IOS XE 3.5

Recently OTV is introduced to the ASR 1000 platform and is supported starting from IOS XE 3.5.  Basically the OTV support on ASR1K and N7K is similar, there are only a few difference that needs to note in this release:

- Support OTV with GETVPN on ASR1K
- No Adjacency server support, multicast WAN is required
- Support fragmentation
- Support of one Joint interface and one access interface per box

Nexus 7000 - M1 modules and F2 modules quick comparison

A fetaure comparison for quick reference:

M1 F2
Max 8 x 10GE line rate ports Max 48 1/10GE line rate ports
Full L2 and L3 feature Full L2 and L3 feature
Large FIB, ACL, QoS Tables Small FIB, ACL, QoS Tables
MPLS MPLS NOT supported
LISP LISP NOT supported
FEX Support FEX Support
FabricPath NOT Supported FabricPath
FCoE NOT supported FCoE (will support in future release)

Saturday, February 11, 2012

Changing CUE license

In the lab, another common task that you will work on is to changing the CUE license.  For CME and UCM integration, different licensing file is required.

Firstly you can do a "show software license" see what license your CUE is currently using:

CUE# show software license
Installed license files:
 - voicemail_lic.sig : 25 MAILBOX LICENSE
 - ivr_lic.sig : NULL IVR LICENSE
 - port_lic.sig : 8 PORT BASE LICENSE

 - Application mode: CCM
 - Total usable system ports: 8

Voicemail/Auto Attendant:
 - Max system mailbox capacity time: 18000
 - Default # of general delivery mailboxes: 10
 - Default # of personal mailboxes: 25

 - Max # of configurable mailboxes: 35

Interactive Voice Response:
 - Max # of IVR sessions: Not Available

 - Max installed languages: 5
 - Max enabled languages: 5

According to the output, my CUE is using UCM license and I wanna change it to CME.  You can place your license file in your FTP server and install it on CUE:

CUE# software install clean url username panda password eatsbamboo

WARNING:: This command will install the necessary software to
WARNING:: complete a clean install.  It is recommended that a backup be done
WARNING:: before installing software.

Would you like to continue? [n]y

Downloading ftp cue-vm-license_25mbx_cme_7.0.1.pkg
Bytes downloaded :  5299

Validating package signature ... done
compatibility mode
Validating installed manifests ...............complete.
The system will be brought to offline state for a brief period
and will be brought back to online state automatically
No work order produced.
The system is back in online state
55296+0 records in
108+0 records out
Size of buff is: 65536
65536 bytes written
 - Total usable system ports: 8


 Important:: A Reload is required in order for the new License to take effect.

It is changed to CME licensing file successfully, reboot the CUE module for the new license to take effect.

CME cBarge - Failed to setup barge

Just labbing on the CME cBarge feature and got the error message "Failed to setup Barge" on the phone display.  The reason in my case is I have forgotten to create the ad-hoc conference ephone-dn.  Let us quickly walkthrough the steps that are needed to config cBarge on CME:

1. Create Conferencing DSP farm profile

dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 2
associate application SCCP

2. DSP farm profile SCCP configuration

sccp local GigabitEthernet0/0
sccp ccm identifier 1 priority 1 version 7.0
sccp ccm group 1
associate ccm 1 priority 1
associate profile 1 register CFB

3. sdspfarm configuration on CME under telephony-service

sdspfarm units 2
sdspfarm tag 1 CFB
no privacy
conference hardware

4. Create ephone template with cBarge softkey in remote-in-use state

ephone-template  1
softkeys remote-in-use  CBarge Newcall

5. Sample ephone configuration

ephone  1
privacy off
device-security-mode none
mac-address 0021.5555.5555
ephone-template 1
type 7965
button  1:1 2:3
ephone  2
privacy off
device-security-mode none
mac-address 0026.2222.2222
ephone-template 1
type 7975
button  1:2 2:3

6. Create shared line ephone-dn and ad-hoc conference ephone-dn
ephone-dn  3  octo-line
number 5003
ephone-dn  4  octo-line
number A123
conference ad-hoc

After I've created the ephone-dn 4 for ad-hoc conferencing, cBarge is setup successfully without the error any more!

Thursday, February 9, 2012

UCM + CUE integration example step-by-step

1. Create CUE End User on CUE

conf t

username boss create
username sec create

username boss phonenumber 1001
username sec phonenumber 1002


username boss pin 12345
username sec pin 12345

2. Config JTAPI subsystem on CUE

ccn subsystem jtapi
ctiport 1111 1112 1113 1114
ccm-manager address
ccm-manager username panda password eatsbamboo
end subsystem

3. Config JTAPI number on CUE

ccn trigger jtapi phonenumber 1110
application voicemail
maxsessions 4
end trigger

3. Create voice mailbox on CUE
voicemail mailbox owner boss

voicemail mailbox owner sec

4. Create UCM End User boss and sec on UCM

5. Create Voicemail Pilot 1110 on UCM

6. Create Voicemail Profile on UCM

7. Assign Voicemail Profile to DN of boss and sec

8. Create CTI Route Point 1110, same DN as voicemail pilot

9. Create CTI ports 1111, 1112, 1113 and 1114

10. Create application users panda and assign CTI route point and ports to it

Note: No MWI number is needed to create on UCM and CUE

UCM - Transfer On-hook

By default, when you are doing a call transfer on UCM, you will press the Transfer softkey on phone to talk to the destination callee, the press the Transfer softkey to complete the transfer.

There is a system parameter in UCM that can change this behavior:
System > Service Parameter > Transfer On-hook Enabled = True

After making the change, when you do a transfer call, after pressing the Transfer softkey, you can put the phone on hook to complete the transfer.

Quick Note about MOH for UCM

Holder (the one who press hold) - determines which audio source file to play
Holdee (the one that is being held) - determines which media resource will play that file (MRGL)

User on Hold audio source - Normal IP phone calls
Network on Hold audio source - call transfer, call park, conference - application-based hold

CLID name and number blocking on H323 gateway

You can block the CLID name or number on outbound VoIP calls from a dial-peer:

dial-peer voice 911 pots
  clid strip name ! Block CLID name

dial-peer voice 10 pots
  clid strip ! Block CLID number

Basic MGCP router configuration using ccm-manager config server command

After configuring UCM MGCP configuration, on router you can config the MGCP feature with ease.  There is a feature called "ccm-manager config server" which configs the MGCP gateway automatically, by pulling configuration file from UCM as well as automatically config the router controller, voice-port and serial interfaces.  The basic settings that you need are as follows:

mgcp bind control source-interface gi0/0
mgcp bind media source-interface gi0/0
ccm-manager config server ! UCM IP addresses, you can specify sub as primary and pub as backup
ccm-manager config

Then you can see the console messages showing the ISDN interface and other MGCP features are configured automatically.

Do a "show ccm-manager" and "show isdn status" to verify if it is configured properly.  If not, check out this post:

If you make any changes that is different from the UCM configuration (e.g. config a partial T1 instead of using all channels), remember to no ccm-manager config server otherwise your configuration will be override after reloading the router

ISDN shows TEI_ASSIGNED status on MGCP gateway

My environment:  IOS 12.4(23)T3 advanced enterprise, 2821

When I try to config the MGCP using the ccm-manager config server, in "show ccm-manager" it shows the gateway is registered to UCM successfully:

Priority        Status                   Host
Primary         Registered     
First Backup    Backup Ready   
Second Backup   None                     

However when I do "show isdn status", the Layer 2 status shows:


A router reload solved the problem, you should see


in "show isdn status" Layer 2 status.

Gatekeeper Zone Prefix and Tech Prefix

- Zone Prefix determines the routing TO a ZONE

- Tech Prefix determines the routing TO a GATEWAY within a ZONE

Preventing CME ephone-dn registering to H323 gatekeeper

If your CME also function as the H323 gateway that registers to gatekeeper, and you don't want the CME DN register to the H323 gatekeeper, you can either:

1.  Disable it globally
   max-dn 20 no-reg


2. Display per ephone-dn
ephone-dn 1
   number 1234 no-reg both

Allow sending calling name to PSTN

To display calling name:

interface serial0/0/0:15
  isdn outgoing display-ie

This display name is the "name" config in ephone-dn or voice register dn in CME

ephone-dn 1
  name panda  ! this is the calling name

ISDN B-channel selection algorithm

The default behavior is in descending order (e.g. 15 -> 14 -> 13).  To change this behavior:

interface serial0/0/0:15
   isdn bchan-number-order ascending ! i.e. 1 -> 2 -> 3

Basic Gatekeeper Security using zone subnet command

By default any gateway can register with gatekeeper, and you can specify the subnet or hosts which are allowed to register to gatekeeper
no zone subnet PANDA-GK default enable
zone subnet PANDA-GK enable

Useful Gatekeeper debug command on Cisco IOS

- debug gatekeeper main 10
- debug gatekeeper call 10

- debug ras
- debug h225 asn1

Gatekeeper CAC Example

Say for example we want to allow only 3 calls
endpoint resource-threshold ! enable the max-calls command
arj reject-resource low ! allows endpoint to reject the limit of ARQ when the endpoint reaches its max no of calls
endpoint max-calls h323id gk_ucm_2 3 ! allow up to 3 calls

UCM - Find Phone Type via CLI SQL statement

To find the phone type that is supported on UCM, one of the way is to run the following SQL statement on UCM CLI:

run sql select * from typemodel where name like '%7975%'

For newer phones that are not found in the UCM database, QED definition file is required to install in order to show up in UCM

Changing Phone Header Bar Display

To change the Phone Header Bar Display (top right hand corner of the phone):
ephone-dn 1
   number 1234
   desc +442056781234

voice register pool 1  ! Not in voice register dn
   number 1 dn 1
   description +442056781234

Change the DN External Phone Number Mask

Disable CME ephone auto-registration

The default behavior for CME is to allow phone auto registration.  When the phone with correct tftp configured, the phone will be registered to CME without a ephone-dn.

To disable this behavior:
   no auto-reg-ephone

Using SIP as UCM Auto-registration Protocol

By default, when you turn on auto-registration on UCM, SCCP is the protocol used for the phone.  You can change it to SIP by modifying the Enterprise Parameter on UCM:
System > Enterprise Parameter > Auto Registration Phone Protocol

Then enable auto-registration from System > Cisco Unified CM

Specifies the Starting and End Directory Number, as well as uncheck the auto-registration disabled checkbox.

Sunday, February 5, 2012

Quick Note on H323 and MGCP call preservation

A quick summary on H323 and MGCP gateway call preservation behavior when it works with UCM:

MGCP (primary UCM dies)
PSTN <-- MGCP GW --> UCM <--> Phone
- enabled by default
- UCM dies, call preserved
- Call continues over the gateway
- supplementary services will not work
- new calls to MGCP gateway will not work until MGCP gateway re-registers with the backup UCM

H323 (primary UCM dies)
PSTN <-- H323 GW --> UCM <--> Phone
- Default has no call preservation
- configure with "call preserve" command
- will immediately begin to use backup UCM for supplementary service, outbound and inbound call routing

Branch voice gateway (WAN link down)
- SRST kicks in
- If MGCP, then MGCP fallback - fallback to H323 / SIP processing
- MGCP - Call drop immediately at the branch router
- H323 is recommended for branch router - call preservation feature after IOS 12.4(9T)

Call preservation is configured in the gateway with the following command syntax:

Router(config)#voice service voip
Router(conf-serv-h323)#call preserve

MGCP fallback is configured with the following command syntax:

Router(config)#ccm-manager fallback-mgcp
Router(config-app-global)#service alternate Default

Automated Alternate Routing (AAR)

In the previous example, RSVP-based CAC is discussed:
In "Not Enough Bandwidth" scenario, you can enable AAR to kick in so that the call will be automatically rerouted to the PSTN instead of the congested WAN.

1.  Firstly enable the service parameter to turn on AAR.

2.  Create partition and CSS for AAR.

3.  Create AAR Group.  Add Dial Prefix if necessary.

4.  Apply AAR CSS and AAR Group to Device Pool, and assign AAR Group to DN.  If you don't input anything in the AAR destination mask, the destination phone external phone number mask will be used.

5.  Create the AAR route pattern with AAR partition to route the call

Reduce the location bandwidth and give it a try~

Friday, February 3, 2012

RSVP-based Call Admission Control

The previous blogpost discussed about the bandwidth calculation for each of the CAC method:
In this blogpost we will go a bit more details on how to get it implemented.
RSVP bandwidth calculation is based on the worst case scenario.  For a G.729 call, it is 40Kbps and for G.711, it is 96Kbps.  However it will not based on worst case scenario for every call, it is only used for the first call.  Say for example if I want to implement RSVP-based CAC to 4 x G.729 call, it should be calculated as:
40Kbps + 24Kbps + 24Kbps + 24Kbps = 112Kbps
Okay, then the next step is the router configuration.

1. Create the MTP dspfarm profile with RSVP enabled
sccp local GigabitEthernet0/0
sccp ccm identifier 1 version 7.0
sccp ccm group 1
associate ccm 1 priority 1
associate profile 2 register R1-MTP
dspfarm profile 2 mtp
codec pass-through
codec g729r8
maximum sessions software 10
associate application SCCP

interface Serial0/1/0:0
no ip address
encapsulation frame-relay
fair-queue 64 256 47
frame-relay lmi-type ansi
ip rsvp bandwidth
interface Serial0/1/0:0.2 point-to-point
bandwidth 768
ip address
ip ospf mtu-ignore
snmp trap link-status
frame-relay interface-dlci 202
ip rsvp bandwidth 112

2. Create the MTP entry on UCM, add it to MRG and add the MRG to MRGL.  Assign the MRGL to the device pool / phone.

3. Create Locations, mark the two locations relation as RSVP "Mandatory".

4. Make calls from between the 2 sites, do a "show ip rsvp reservation"

R1#sh ip rsvp reservation
To            From          Pro DPort Sport Next Hop      I/F      Fi Serv BPS UDP 18176 16670 none          none     FF LOAD 24K UDP 16670 18176  Se0/1/0: FF LOAD 24K

Since the ip rsvp bandwidth is 112Kbps, you are allowed to make up to 4 G.729 calls and when you try to make the 5th call, you will see the message "Not Enough Bandwidth" on the phone.

Using your MacBook as wireless access point

I am using a MBP and I don't want to setup a permanent wireless access point at home for some reason, so the easiest way for me to create a home wireless network occasionally is to using my MBP to share my wired connection with wifi.

To do it, first go to System Preferences, choose Sharing:

Then turn on the Internet Sharing.  Share your connection from:  "Ethernet", and To computers using "Wifi"

Your wireless icon on the top menu bar should change to:

You can optionally config security for your wireless network, however only WEP is supported  :(

My current OS Version when I write this post:  OS X Lion 10.7.3

Call Admission Control - Bandwidth Allocation

A quick summary table on how much bandwidth used in the CAC calculation for 1 call in different codecs

Gatekeeper CAC
Location CAC

Mobile Voice Access + Mobile Connect - Step by Step

To configure Mobile Voice Access + Mobile Connect:
1.  Make sure the MVA service is started on UCM

2.  Media Resource > Mobile Voice Access - to configure your MVA number, in my case my number is 3000

3. Change the Service Parameter > Cisco CallManager

4. Config your H323 gateway, where is your UCM address in my case
dial-peer voice 3000 pots
 service mva
 incoming called-number 3000
 no digit-strip
dial-peer voice 3001 voip
 destination-pattern 3000
 session target ipv4:
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad

5. Configure the CCM End User for MVA access

Also make sure that it is added to the appropriate User Group

6. Associate the user to the device, as well as the DN.  Also change the device "owner ID" to that user.

7. Create Remote Destination Profile + Remote Destination.  Make sure that the rerouting CSS can reach the route pattern for Mobile Connect.  Associate the DN on RDP to the user.

8. On the phone, add the mobile connect phone button

Test it by calling from PSTN to the MVA number.  If the PSTN number matches (partial match according to the above config) with your remote destination number, you are only required to enter the PIN without the key in the remote destination number when you call the MVA number.  Also try Mobile connect and make sure the re-routing CSS can reach the remote destination number.
Good luck!

Assigning Static IP address via Cisco IOS DHCP

For some cases that you want to assign a static IP address via a Cisco IOS DHCP server, to make sure for that particular MAC address will always get the same IP address, here is the example on how to get it done:

ip dhcp pool PandaPhone
   client-identifier 01ee.ffaa.bbcc.dd
   option 150 ip

Assuming the MAC address of the phone is EEFF.AABB.CCDD

Save energy with Energywise

I am using a 3560-8PC PoE switch in my home lab and in order to save my electric bill, energywise has been implemented on the switchport in order to turn off the PoE power in the hours that I don't need it.

Example:  To turn on the PoE power every morning at 8am and turn off at 10pm.

interface FastEthernet0/7

 energywise level 10 recurrence importance 100 at 0 8 * * *
 energywise level 0 recurrence importance 100 at 0 22 * * *

Also I have written a web page to override this behavior, by triggering the expect script on my Linux box, details can be referred by this old post:

OSX 10.7.3 - Finder and other apps keep crashing after update

After updating to 10.7.3 through the Software Updater, the Finder and other apps keep crashing and the keyboard is not responsive after the Finder is forced close.  Problem solved after download the 10.7.3 Update Client Combo, install and reboot it.  It is quite stable for 30 minutes already, hopefully it will not crash after I publish this post.  :(

Download the client combo here:

Wednesday, February 1, 2012

Provisioning Cisco Cius - which MAC address to use?

Cisco Cius uses two addresses: Ethernet MAC and Wireless LAN MAC. When adding Cisco Cius to the Cisco Unified Communications Manager, it must be provisioned using the Ethernet MAC address.


CUE - Configure Caller ID information for incoming voice mail messages

CUE supports caller ID information for incoming voice-mail messages from external caller that is not listed in the CUE subscriber directory.  Default it is disabled and the system will play "Unknown Caller" in the message envelop. Once you enable it, it will play the caller's number in the message envelop.  This is the command on how to enable this feature on CUE

CUE# config t
CUE(config)# voicemail callerid

An iPhone app for CME Config Generation:

Create Meetme conference on Cisco CME

Create a meetme conference on CME is quite similar to ad-hoc conference in terms of configuration, here is an example how to create a Meetme conference on CME:

1. Create DSP Farm Porfile
sccp local GigabitEthernet0/0
sccp ccm identifier 1 priority 1 version 7.0
sccp ccm group 1
associate ccm 1 priority 1
associate profile 1 register CFB
dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 2
associate application SCCP

2. Config telephony service to use hardware conference and associate it with the DSP Farm
sdspfarm units 1
sdspfarm tag 1 CFB
no privacy
conference hardware
no auto-reg-ephone
max-ephones 20
max-dn 20
ip source-address port 2000
cnf-file location flash:
max-conferences 8 gain -6
transfer-system full-consult

3. Create ephone template that has the meetme softkey in seized state
ephone-template  1
softkeys seized  Redial Endcall Cfwdall Pickup Gpickup Callback Meetme

4. Create meetme conference ephone-dn
ephone-dn  10  octo-line
number 1234 no-reg primary
conference meetme
no huntstop

An iPhone app for CME Config Generation:

Sample Frame Relay Switch Configuration for CCIE Voice Lab

In my CCIE voice lab, I am using a 2821 router with multiple VWIC-2MFT-G703 equipped in the PSTN router as the frame relay switch.  The other 2821 routers are also equip with VWIC-1MFT-G703 for data connection.  The advantage of this approach is you don't need to pay for the serial cable cost, it can be done using normal Cat5 cable, with the correct G.703 cross over cable pin out.  The crossover cable is 1-4, 2-5 inverse.  This is my cable pin out:

1. Orange-white - - - Blue
2. Orange - - - Blue-white
3. Green-white - - - Green-white
4. Blue - - - Orange-white
5. Blue- white - - - Orange
6. Green - - - Green
7. Brown-white - - - Brown-white
8. Brown - - - Brown

To create serial interface, channel-group command will be used under controller.  For each DS0 timeslot, the default bandwidth is 56Kbps for T1 card and 64Kbps for E1 card.

controller E1 0/0/0
 channel-group 1 timeslots 1-3

In the above configuration, a 192Kbps connection is created for the interface serial 0/0/0:1

Then the next step is to configure the interfaces that are created for frame relay switching, here is the sample configuration:

interface Serial0/0/0:1
description -- Data connection to HQ --
 no ip address
 encapsulation frame-relay
 frame-relay lmi-type ansi
 frame-relay intf-type dce
 frame-relay route 201 interface Serial0/0/1:1 101
 frame-relay route 202 interface Serial0/1/0:1 102
interface Serial0/0/1:1
 description -- Data connection to SB --
 no ip address
 encapsulation frame-relay
 no frame-relay inverse-arp
 frame-relay lmi-type ansi
 frame-relay intf-type dce
 frame-relay route 101 interface Serial0/0/0:1 201
interface Serial0/1/0:1
 description -- Data connection to SC --
 no ip address
 encapsulation frame-relay
 no frame-relay inverse-arp
 frame-relay lmi-type ansi
 frame-relay intf-type dce
 frame-relay route 102 interface Serial0/0/0:1 202

After it is created, and configuration on the frame relay routers are configured, in the "show frame-relay pvc" you should see "switched" PVC is in active state.  This is an example for interface s0/0/0:1.

PVC Statistics for interface Serial0/0/0:1 (Frame Relay DCE)

              Active     Inactive      Deleted       Static
  Local          0            0            0            0
  Switched       2            0            0            0
  Unused         0            0            0            0