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. 

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:

application

service aa flash:app-b-acd-aa-3.0.0.2.tcl

  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-3.0.0.2.tcl

  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:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html

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

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:

http://pandaeatsbamboo.blogspot.com/2008/03/configuring-plar-on-cucm.html