CCIE Voice exam passed!!

Finally....

Still remember back in 2010, my first attempt.  At that time I thought I was good enough to take this challenge, at the end I have spent 3 hours to troubleshoot the phone registration problem and I have got quite a lot of sections with zero marks.

I have put it down for 1.5 years. 2 months ago I have decided to pick it up again and took my 2nd attempt. At that time I was very confidence and I have never thought that I will fail until I see the score report.  That was close.

Finally I made it yesterday.  I can now spend more time on other stuff, instead of pressing the keypad on the phones and make calls days and nights.

3 times more joy than getting my first CCIE R&S which passed in first attempt.  :)

Ask the cloud

This is something I have created around 1.5 years ago.  Just found this video on my hard drive when I tried to free up some space of my hard drive.  This is a demonstration on how to control Cisco UCS Manager using voice recognition technology.

Cisco UCM + UCCX + VXML + DMS + UCS Manager + Nuance

At that time there is no Siri, yet, LOL

Join the Cisco Jabber Video for Telepresence Beta now and extend your Telepresence Calling Circle

Join the Cisco Jabber Video for Telepresence Beta now and extend your Telepresence Calling Circle.
https://www.ciscojabbervideo.com

I have joined and tried and it works really well on my Mac.  I have also tried to conduct a 4-party video conference with EX90 embedded Multisite and it looks great.  Sign up for yours now!

Cisco Identity Services Engine - Default web login username and password

My environment:  Cisco ISE 1.0.3.377

I have just installed ISE in my lab environment to test things out and the installation is pretty straight forward.  Mount the iso in the VMWare ESXi datastore and run it for around 45 mins.  A post-installation wizard for information like IP address, hostname, etc.  A default admin user is created during the wizard, however it is NOT the same as the web admin username and password.  To access ISE:

https:///admin

The default web admin username and password is admin/cisco.  You are prompted to change it after your first successful logon.

Haven't started configure anything yet, will post anything that found interesting.  :)

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.

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)#application

R1(config-app)#global

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:

http://pandaeatsbamboo.blogspot.com/2012/02/ucm-cue-integration-example-step-by.html

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:
http://pandaeatsbamboo.blogspot.com/2012/02/ucm-cue-integration-example-step-by.html

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

ccn subsystem sip

gateway address "192.168.166.254"

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 ipv4:192.168.166.254

dtmf-relay sip-notify

codec g711ulaw

4. Configure the SIP MWI:

sip-ua

mwi-server ipv4:142.1.66.253 expires 3600 port 5060 transport udp unsolicited

5.  Sample CME as SRST configuration

telephony-service

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 192.168.166.254 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.