Tomslin's Blog

Social networks and customer care

2011-11-25T18:28:00.001+03:00

I love logging into facebook now & then to check what my pals are upto, the photos and the statuses always catch my attention. But apart from all the relationship and copy-paste-quotes statuses I see, people tend to express their disappointments,anger and opinions about the services rendered by different service providers/companies out there. The problem about this is,if these companies do not get to these unhappy customers fast enough to address their concerns, they risk losing a customer or worst still, get some bad publicity on these social sites.

Some companies like Cisco must have thought exactly this way to put together projects to develop products that will address such a problem. I must say I was too busy to realize that many products had already been released that will browse social websites on behalf of a company for any tweet/post about the company and immediately, intelligently route it to the right customer-care representative to respond and deal with customers' posts in real-time. I thought this was a brillant idea. As there will be more happy customers out there and service providers will have a new opportunity to proactively upsell as well.

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It's done!

2011-11-25T17:58:00.001+03:00

Well, you remember my last post was about how I didn't have psyk to study for my GVP exam, I did manage to pull together some psyk amidst lots of parties to study and sit for the exam. Am actually glad its over now because I had to postpone that exam thrice...had cold feet...don't ask!

While I was traveling in a bus today,the question 'what next' kept popping up in my head. That's when I realized I have an unfinished business with Cisco, (as much as I am done with Genesys. ) Since I believe in the saying "you must finish what you started",I am resuming studying for Cisco's CCIE Voice after my wedding in December this year.

Talking about weddings, please wish me luck in this one....I mean it! Never thouht nervousness was part of getting maried, now I know better.

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Why do I feel this way?

2011-09-02T10:02:00.002+03:00

Interesting title huh?

Here is the thing, I am trying to study for my Genesys Voice Platform GVP 8.1 exam, but I keep falling asleep on my books. I don't know if it is because I am tired of taking these certification exams or because the subject is boring. But I have promised to try as hard as I can to keep studying and sit for the exam.

What is worse is that I haven't yet booked a date with Pometric for the exam, could this also be part of the reason why am dozing while studying?

My target is to sit for it end of this month of September, so i think i must just go ahead and book a date so i can get serious. off to pometric's website now!

Cost containment v/s Customer-centric contact centers

2010-08-20T12:39:00.000+03:00

Looking back at the revolution of contact centers, we'll notice that it was with cost in mind that the first call center was born. It all began in the US aviation industry in the late 1960s,where reservations were being handled, it wasn't long before businesses in other sectors had these 'bright' ideas for call centers: "these could reduce the need for equipment in every branch and allow us to strip out redundant people,processes and technology".

The first generation of customer service automation was the switchboard and later the PBX. These enabled customers to contact companies by phone and be routed to a particular person or department to reslove their issue. These technologies were breakthrough advancements, but could not scale to meet the pace and needs of modern customer service communications.

To solve these challenges, technologies such as Automatic Call Distributors (ACDs) and IVRs were introduced, with features such as pre-routing, music on hold, and even self-service. IVRs are typically deployed with a focus on cost containment — a way to reduce the number of calls requiring agent assistance by forcing callers through complex menu trees, options, and questions. IVRs helped reduce call volumes, but provided no assurance that customers solved their problem or reached an agent. Within this paradigm, there was no telling how many hung up in frustration.

While these solutions alleviated many drawbacks of early phone system technology, they

did not improve the customer experience and, in fact, often served to exasperate customers.

In contrast, customer-centric contact centers thrive to determine the most effective way to interact with customers and resolve their requests. Rather than force customers into frustrating IVR menus and 'voice mail jail," they leverage data and systems to efficiently interpret each interaction.

This approach not only drives customer satisfaction, but delivers efficiency gains and improved selling opportunities that translate into real bottom-line benefits.

By utilizing media resources such as speech recognition and text-to-speech, customer-centric contact centers deliver state-of-the-art capabilities to guide the machine-to-human interaction in the most compelling, effective way for the caller. Tight integration with customer data, corporate data, and content management systems deliver the detailed, up-to-date information that the solution needs to guide the call flow. Advanced business rules take this data and determine how to anticipate the caller's needs and handle them proactively.

In today's businesses where the question to answer is "how do we provide the best customer experience while managing the cost-to-serve?", it is important that businesses take on the new customer-centric approach to achieve their aim :customer retention, best customer experience and low cost-to-serve.

A new generation of contact center solution

2010-08-20T10:49:00.003+03:00

Ever wanted your IVR to respond with a custom,personalized greeting when your customers call your contact center? To automate up-selling/cross-selling at the right time and opportunity? Ever wanted a dynamic IVR menu option to your customers, discerning their identities and intent of the caller and determining how to treat different callers based on preferences, caller history and business rules?

If the answer is YES to these questions then Genesys intelligent Customer Front Door (iCFD) is the product you've been waiting for all these years.

Genesys iCFD consists of a unique combination of Dynamic Contact Center capabilities and advanced self-service applications. The result is a transformative intelligent routing solution that leverages a business rules engine to apply context to self-service calls to determine the most appropriate next step — self-service or assisted service — to deliver faster call resolution and superior service.

The iCFD delivers a branded, personalized, and effective customer experience that:

Discerns the identity and intent of callers in the fewest steps
Gathers relevant information from back-end data or workflow to understand the context of their calls
Determines how to treat callers based on preferences, resource availability, and business rules
Matches the most relevant and available resource, including self-service, proactive notification, automatic callback, or assisted service

The iCFD can co-exist within multi-vendor environments, operate in multiple languages

with any speech recognition software, and include business processes tailored to the unique needs of any customer service organization.

The core principle behind the iCFD is to deliver a superior customer experience by

automatically responding in a personalized way to the caller’s needs.

From the Customer's perspective,

When callers contact a company, the iCFD answers the call and assesses existing customer information. For each caller, the iCFD creates a unique interaction based on the contextual view of why the caller is making contact. By using all the information resources available about the customer, the iCFD can proactively guide them to the most effective and expedient solution, provide self-service, or route the call directly to the best available agent and offer the most relevant services. Further, the caller experiences a common voice for the company — one that reinforces the company’s brand personality and promise.

Customers can also experience the iCFD in other innovative ways — through call backs,

notifications and alerts, surveys, and other services that help further personalize and improve the experience. For instance, the iCFD can offer bill payment options when bills are overdue, route the caller to a ‘collections expert’, offer cross-selling suggestions after recent purchases, or suggest other highly relevant offers in anticipation of the individual caller’s needs.

How cool can a contact center be? There's no more leaving customers frustrated after an IVR session.

Watch how it works on this video below:

Genesys Forensics tool

2010-07-21T15:01:00.003+03:00

Wow! Another cool tool I saw today for the Genesys consultants and engineers.

It is called Audit Trail for Genesys.

Many Genesys engineers will usually worry about how to monitor the Genesys environment and track changes made in CME. If you are one of those then you got yourself a saviour!
This tool actually details WHAT, WHEN and WHO made the change to WHICH Genesys application. How cool is that?

And that's not all, it can roll-back to your working config if the changes made have broken anything.Forgive me am just too excited about this tool, it is really cool.

Other features include:

Capability to manage and implement scheduled lock-downs for CME access. This feature is intended to prevent changes being made during pre-defined critical operational hours.
Email notifications to users and system administrators of changes made by users as they occur throughout the day. This feature is intended to prevent changes being made during pre-defined critical operational hours.
Execution of external triggers such as shell scripts when a particular object type is changed.

Audit Trail for Genesys is awesome. If you are a Genesys consultant or engineer, you need this one!

Mobile IVR

2010-07-20T20:54:00.002+03:00

As I sit here waiting for Pizza,my mind drifts to a solution I was shown a demo of today. MOBILE IVR.

I was very impressed, not just with the technology, but the big savings that Service Providers will see with this solution. It already claims to give SPs about 70% of CapEX savings on network resources and 50% savings on OpEX.The question: how does this exactly work?

The way it works is that an app will be installed and always be running on the phone, this app will be able to detect when a customer dials an IVR number.If its the first time they are dialing, it downloads the menu options and a visual menu on their handset,via a data connection to the IVR server,then it simulates a call to the network,playing a ringback to the customer,it then plays the menu voice prompts locally on the users device using TTS. When the user selects a menu option, messages are passed between the device and the IVR server at the SP's premises over a data connection again.What's imporant here is that a voice channel actually gets used only when a user selects the option to talk to an agent,then the mobile IVR app actually makes a call to the network, otherwise, no voice channel on the SP's network is used.

This gives the caller a much better experience, no more 'network busy' errors, improved customer handling and network resource savings by the SP.

I can't wait to see this in action in Africa

Unified CM Classes of Service with Traditional approach vs Line/Device approach

2009-09-22T19:11:00.002+03:00

Hhmm...if you've been in this world before Unified CM 6.x, then you'd understand the pain of all the partitions and Calling Search Spaces(CSS) you have to create when deploying a multisite centralized call processing project. Usually you have to create partitions as many as x number of classes of service multiplied by y number of sites. Now this could be very many partitions if you had only 5 sites. Same with the CCSs. This is the traditional approach but the line/device approach really makes things easy for us now. The number of partitions you need and CSSs is just equal to

Total Partitions = (Number of classes of service) + (Number of sites) + (1 Partition for all IP phone
DNs)

Total Calling Search Spaces = (Number of classes of service) + (Number of sites)

The number comes to a very small amount compared to the traditional approach. You can really get to see the impact as you increase the number of sites.

How does this Line/Device approach work?
This is possible because unlike the traditional approach where the device calling search space is used to determine both the path selection and the classes of service, in the device/line approach the device CSS is used only to determine the path selection and the line CSS is used to determine the classes of service. To understand this, you need to remember how the resulting calling search space looks like when the line CSS and the device CSS are combined together for each IP phone.
The Line CSSs appear ontop of the device CSS and are used for call routing. So with the line/device approach, the device CSSs are given site specific unlimited access CSSs with partitions having route patterns that point to site specific route lists/route groups. While the line or lines are assigned CSSs with translation patterns that block specific patterns as per the class of service of the line.

Cisco recommends the following guidelines when implementing the line/device approach in a multisite with centralized call processing:

Create an unrestricted calling search space for each site and assign it to the phone's device callingsearch space. This calling search space should contain a partition featuring route patterns that route the calls to the appropriate gateway for the phone's location (for example, a co-located branch gateway for emergency services and a centralized gateway for long-distance calls).
Create calling search spaces containing partitions featuring blocked translation/route patterns for those types of calls not part of the user's dialing privileges, and assign them to the user's lines. For instance, if a user has access to all types of calls except international, that user's line (or lines) should be configured with a calling search space that blocks the 9.011! route pattern.

Cisco Smartports Macros

2009-07-31T10:52:00.002+03:00

Cisco Smartports macros can speed up deployment while ensuring accuracy and consistency on your switch ports on Cisco Routers and Switches. This is a cool way to quickly build configs with best practice templates/scripts pre-built on these devices.

These are not new commands but a group of already existing commands in the IOS. You'll find these smartports on most of the recently deployed switches and routers.
You can access these macros either using the Cisco network Assistant (CNA), the switch web interface(SDM) or the CLI.

GUI applications can usually tell you the type of devices connected to the port, and they can recommend specific Smartports macros for the proper kind of device.

On the CLI, you can view all available macros with the show parser macro command. Below are the macros and what they do with the actual commands that the scripts enter for you.

Switch#show parser macro
Total number of macros = 6
--------------------------------------------------------------
Macro name : cisco-global
Macro type : default global
# Enable dynamic port error recovery for link state
# failures
errdisable recovery cause link-flap
errdisable recovery interval 60

# Enable aggressive mode UDLD on all fiber uplinks
udld aggressive

# Enable Rapid PVST+ and Loopguard
spanning-tree mode rapid-pvst
spanning-tree loopguard default
spanning-tree extend system-id
--------------------------------------------------------------
Macro name : cisco-desktop
Macro type : default interface
# macro keywords $access_vlan
# Basic interface - Enable data VLAN only
# Recommended value for access vlan should not be 1
switchport access vlan $access_vlan
switchport mode access

# Enable port security limiting port to a single
# MAC address -- that of desktop
switchport port-security
switchport port-security maximum 1

# Ensure port-security age is greater than one minute
# and use inactivity timer
switchport port-security violation restrict
switchport port-security aging time 2
switchport port-security aging type inactivity

# Configure port as an edge network port
spanning-tree portfast
spanning-tree bpduguard enable

# Remark all inbound data packets with COS=0 & DSCP =0
mls qos cos override
--------------------------------------------------------------
Macro name : cisco-phone
Macro type : default interface
# Cisco IP phone + desktop template

# macro keywords $access_vlan $voice_vlan         

# VoIP enabled interface - Enable data VLAN
# and voice VLAN
# Recommended value for access vlan should not be 1
switchport access vlan $access_vlan
switchport mode access

# Update the Voice VLAN value which should be
# different from data VLAN
# Recommended value for voice vlan should not be 1
switchport voice vlan $voice_vlan

# Enable port security limiting port to 3 MAC
# addressess -- One for desktop and two for phone
switchport port-security
switchport port-security maximum 3

# Ensure port-security age is greater than one minute
# and use inactivity timer
switchport port-security violation restrict
switchport port-security aging time 2
switchport port-security aging type inactivity

# Enable qos to extend trust to attached Cisco phone
mls qos trust device cisco-phone

# Configure port as an edge network port
spanning-tree portfast
spanning-tree bpduguard enable
--------------------------------------------------------------
Macro name : cisco-switch
Macro type : default interface
# macro keywords $native_vlan
# Access Uplink to Distribution
# Do not apply to EtherChannel/Port Group

# Define unique Native VLAN on trunk ports
# Recommended value for native vlan should not be 1
switchport trunk native vlan $native_vlan

# Update the allowed VLAN range (ALL) such that it
# includes data, voice and native VLANs
switchport trunk allowed vlan ALL

# Hardcode trunk and disable negotiation to
# speed up convergence
switchport mode trunk
switchport nonegotiate

# Configure qos to trust this interface
mls qos trust cos

# 802.1w defines the link as pt-pt for rapid convergence
spanning-tree link-type point-to-point
--------------------------------------------------------------
Macro name : cisco-router
Macro type : default interface
# macro keywords $native_vlan
# Access Uplink to Distribution

# Define unique Native VLAN on trunk ports
# Recommended value for native vlan should not be 1
switchport trunk native vlan $native_vlan

# Update the allowed VLAN range (ALL) such that it
# includes data, voice and native VLANs
switchport trunk allowed vlan ALL

# Hardcode trunk and disable negotiation to
# speed up convergence
switchport mode trunk
switchport nonegotiate

# Configure qos to trust this interface
mls qos trust cos

# Ensure fast access to the network when enabling the interface.
# Ensure that switch devices cannot become active on the interface.
spanning-tree portfast trunk
spanning-tree bpduguard enable
--------------------------------------------------------------
Macro name : cisco-wireless
Macro type : default interface
# macro keywords $native_vlan
# Access Uplink to Distribution

# Define unique Native VLAN on trunk ports
# Recommended native vlan should NOT be 1
switchport trunk native vlan $native_vlan

# Update the allowed VLAN range such that it
# includes data, voice and native VLANs
switchport trunk allowed vlan ALL
        
# Hardcode trunk and disable negotiation to speed up convergence
switchport mode trunk
switchport nonegotiate

# Configure qos to trust this interface
mls qos trust cos

# Ensure that switch devices cannot become active on the interface.
spanning-tree bpduguard enable
--------------------------------------------------------------
Switch#

Gatekeeper static registration

2009-05-03T23:07:00.002+03:00

Statically configure nodes that are unable to register like Unified Communications Manager:

(gatekeeper) alias static ip-signaling-addr [port] gkid gatekeeper-name
 [ras ip-ras-addr port] [terminal | mcu | gateway {h320 | h323-proxy |
 voip}] [e164 e164-address] [h323id h323-id]

If an endpoint or node cannot register with a gatekeeper for some reason, it can be statically defined with an alias in the gatekeeper.

The node's IP address and port are given as ip-signaling-addr [port].
The gatekeeper name for the node's zone is given as gkid gatekeeper-name.
You can specify the RAS address and port used by the node with ras ip-ras-addr port.
The type of H.323 endpoint is given by terminal (H.323 terminal),mcu (multiple control unit), or gateway.
Gateway types are given by h320 (H.320), h323-proxy (H.323 proxy), or voip (VoIP).
One or more E.164 numbers for the node can be given as e164 e164-address (up to 128 characters total).
One or more H.323 identification strings can be assigned to the node with h323id h323-id (up to 256 characters total).

Restrictions for Conferencing and Transcoding for Voice Gateway Routers

2009-04-17T10:59:00.002+03:00

DSP resources communicate with Cisco Unified CallManager using Skinny Client Control Protocol (SCCP). Other protocols are not supported.
DSP resources cannot be enabled for a slot on the Cisco 1700 Series so the dsp services dspfarm command is not supported and cannot be configured for a voice card on the Cisco 1700 Series.
Conferencing is not supported on a Cisco 3640 using the NM-HD-1V, NM-HD-2V, or NM-HD-2VE.
Hardware MTPs support only G.711 a-law and G.711 u-law. If you configure a profile as a hardware MTP, and you want to change the codec to something other than G.711, you must first remove the hardware MTP by using the no maximum sessions hardware command.
Software MTPs are supported on the router only if the dsp services dspfarm command is not enabled on the voice card.
Only one codec is supported for each MTP profile. In order to support multiple codecs, you must define a separate MTP profile for each codec.
If an MTP call is received but MTP is not configured, transcoding is used if resources are available.
Dynamic conference and transcoding resource allocation is not supported.
Fax is not supported for transcoding.

Configuring QoS on Non-Gigabitethernet 6500 modules like WS-X6248-RJ-xx and WS-X6348-RJ-xx linecards

2009-02-02T10:19:00.003+03:00

On such modules,there's a hardware limitation that prevents the proper functioning of port-based trust (which affects trust-cos,trust-ipprec and trust-dscp).
On such modules, a work-around ACL can be used to achieve trust functionality for trust-cos,trust-ipprec, and trust-dscp.
The workaround ACL for trust-cos for such linecards is shown below:

6500-CATOS> set port qos 3/1 vlan-based
6500-CATOS> set qos acl ip TRUST-CoS trust-cos any
6500-CATOS> commit qos acl TRUST-CoS
6500-CATOS> set qos acl map TRUST-CoS 10

QoS policies can be applied to either vlans or ports. In our example above,we apply it to a vlan (10) since typically all phones will have same QoS settings. Port-based QoS policies are normally used to apply ACLs to ports to which servers are connected.

NOTE: that to apply the ACL above, it must first be committed to hardware

Routing calls to UCM in a Gatekeeper controlled network without a technology prefix and/or default technology prefix on Gatekeeper

2009-01-12T15:57:00.005+03:00

If we refer to the Gatekeeper call routing logic,we'll determine that the solution is to register the E164 numbers of UCM with Gatekeeper.

But while on terminals such as ATA or IOS gateway such as CME where this behavior is the default,we have a little bit more work to get CallManager to register its numbers.

The trick lies with the alias static command on the Gatekeeper. The syntax of this command is:

alias static ip-signaling-addr [port] gkid Gatekeeper-name [ras ip-ras-addr port] [terminal | mcu | gateway {h320 | h323-proxy | voip}] [e164 e164-address] [h323id h323-id]

In effect we want to add a static route to CallManager for the relevant E164 numbers so we have to specify the following:

the signalling address (of the remote gateway/terminal)
the port that H225 messages will be sent over (default is TCP 1720)
the zone name the numbers register with
the RAS ip address (of the remote gateway/terminal)
the RAS port number (default is UDP 1719)
the endpoint type- Terminal or gateway
the E164 numbers (enter each number explicitly)

On the CallManager when configuring the H225 trunk,the technology prefix is not required. Be careful though to specify the type of endpoint when issuing the alias static command.- The default is “terminal” which will still work as long as you have defined the correct “Terminal Type” in the trunk page on UCM.

IMPORTANT: Note that UCM by default won't be expecting H225 and RAS messages over TCP 1720 and UDP 1719 respectively. Therefore using these ports in your alias static command may cause your calls to fail even though the UCM E164 numbers are registered with Gatekeeper. Now, if you were to find out on which ports UCM is expecting these messages (by using the command sh gatekeeper endpoint) and use them with your alias static command, the calls will succeed BUT in the event that someone restarts UCM, resets the trunk or even restarting the Gatekeeper(this is bound to be done by you at some point),the expecting H225 and RAS ports on UCM will change and you'll be back to the scenario of mismatch in the ports used by your Gatekeeper and the ports UCM is expecting H225 and RAS messages over.

Tricky right? Don't give up yet!!

The solution lies in configuring the UCM trunk to not dynamically chose RAS/H225 ports each time it registers with Gatekeeper and the answer lies in the CallManager Service Parameters of “Controlled Trunk That Will Use Port 1720*” and “Host Name/IP Address of GK That will Use RAS UDP Port 1719*” on the CallManager Service Parameters page. Both these parameters have a default value of NONE.

Enter the name of your trunk that will use port 1720 for H225 signaling and the IP address of the Gatekeeper that will use port 1719 for RAS- you can only have one trunk/Gatekeeper that uses these ports. You will need to reset your trunk and Gatekeeper after this change.

IP-to-IP Gateway(IPIPGW) TIPs:slow-start to fast-start interworking

2009-01-09T09:26:00.004+03:00

Slow-start to fast-start interworking prevents the IPIPGW from dropping calls down to slow-start when it detects different call signaling on the incoming and outgoing call legs of H.323 to H.323 calls.Slow-start to fast-start interworking supports voice,DTMF relay and T.38 fax.
Configurations can either be done on the dial peer level or global config level.
The following config shows slow-start to fast-start interworkin on an IPIPGW at the global level:

voice service voip
h323
call start interwork

The call start interwork command only supports interwork between fast-start to slow-start.It shouldn't be used in situations where fast-start t0 fast-start or slow-start to slow-start calls are possible

Why you should have a specific inbound dial peer configured on an H323 IOS gateway

2009-01-08T16:42:00.004+03:00

If no incoming dial peer is matched by the router or gateway, the inbound call leg is automatically routed to a default dial peer (POTS or Voice-Network). This default dial peer is referred to as dial-peer 0 or pid:0.

Note: There is an exception to this statement. Cisco voice and dial platforms, such as the AS53xx and AS5800, require that a configured inbound dial peer is matched for incoming POTS calls to be accepted as voice calls. If there is no inbound dial peer match, the call is treated and processed as a dial-up (modem) call.

Dial-peer 0 has a default configuration that cannot be changed. The default dial-peer 0 fails to negotiate non-default capabilities, services, and applications such as:

Non-default Voice-Network capabilities: dtmf-relay, no vad, and so forth.
Direct Inward Dial (DID)
TCL Applications

Dial-peer 0 for inbound VoIP peers has this configuration:
any codec
vad enabled
no rsvp support
fax-rate voice
no dtmf relay
IP precedence 0

Note: The default DSCP for voice is EF codepoint 101110 (RFC 2598), and the default DSCP for signaling is AF31 codepoint 011010 (RFC 2597). The default dial peer does not mark packets to DSCP 0. All voice packets on the routers are marked by default (this can be overridden by the dial peer), signaling with AF31 and media with EF. Calls that match the default dial peer 0 should also have this behavior.

Dial-peer 0 for inbound POTS peers has this configuration:
no ivr application

I would imagine then that you wouldn't want a situation where you have no control of all incoming call legs, plus if there is any dtmf-relay mismatch, calls might not be routed properly or fail.It is therefore recommended that you configure an inbound dial peer for all incoming call legs.
Before I tell you how to do this,lets take a few seconds to look at how the router selects an inbound POTs dial peer based on the information elements in the setup message of the call with the dial peer attributes.
The router attempts to match these items in the following order:
1. Called number(DNIS) with the incoming called-number command
2. Calling number (ANI) with the answer-address commend
3. Calling number(ANI) with the destination-pattern command
4. Incoming voice port with configured voice port

The following configuration is recommended for incoming POTs dial peers:

dial-peer voice tag pots
incoming called-number .
port X/X/X

Note: that (.) is the only wildcard for incoming called-number. Also ensure to add the direct-inward-dial command if DID is supported.
For VOIP dial peers,the same routing matches as with the POTs dial-peer(except the direct-inward-dial cannot be used).If there are no matches with the incoming called-number,answer-address and destination-pattern (matching on the voice port is not an option for voip dial-peers),then the default dial-peer is used.It is recommended to always use a dial-peer with the incoming called-number command to ensure we always have a match.
Don't forget to change properties like CODECS,VAD(Voice Activity Detection) and DTMF-RELAY to match what you desire.

CME Tips (Changing a MOH file)

2009-01-06T12:26:00.004+03:00

To change a MOH audio file,you must first remove the existing file with the no moh file-name command before specifying the new file, as shown in the example below.
If you configure a second file without removing the first,the MOH mechanism stops working and may require a router reboot to clear the problem.

(config-telephony)#no moh music-file-old.au
(config-telephony)#moh music-file-new.au

Also note that,when MOH from an audio file and MOH from a live feed are both configured on the Cisco CME router,the router seeks the live feed first.If a live feed is found,it displaces the audio file source.If the live feed is not found or fails at any time,it falls back to the audio file source.

CME Tips (Non-Standard ephone-DN assignment options to ephones)

2009-01-06T09:01:00.001+03:00

When assigning ephone-DNs to the ephones, there are options beyond the standard assignment.Below are some of them:

1. Normal phone lines – button 1:2 2:5
2. Silent ring,ringer muted,message waiting beep muted – button 1s2 2s5
3. Silent ring,ringer muted,message waiting beep not muted – button 1b2 2b5
4. Feature ring – button 1f2 2f5
5. Monitor line,silent ring,call waiting display suppressed – button 1m2 2m5
6. overlay lines,combine multiple lines per physical button – button 1o2,3,5
7. overlay call-waiting,combine multiple lines per physical button – button 1c2,3,5

CME Tips (Channel Hunting with dual-line DNs)

2009-01-06T08:42:00.002+03:00

When configuring dual-line ephone-DNs with CME, you can prevent calls from hunting from one voice channel to the second when the first voice channel is busy with then huntstop channel command. The default behavior is no huntstop channel.

This also prevents the scenario where a call comes to a line,rings the first channel for 30sec then rings the second channel for another 30sec before rolling over to another line.

The following example shows dual-line ephone-DN configuration in which calls do not hunt to the second channel of any ephone-DN,but they do hunt through each ephone-DN's channel 1 in the order:

ephone-DN 10 → ephone-DN 11 → ephone-DN 12

ephone-dn 10 dual-line
number 1001
no huntstop
huntstop channel

ephone-dn 11 dual-line
number 1001
no huntstop
huntstop channel
preference 1

ephone-dn 12 dual-line
number 1001
no huntstop
huntstop channel
preference 2

TIPS:Recovering a Cisco Unity Express Module from an Errant State

2008-12-17T16:41:00.002+03:00

When candidates who are taking the Cisco CCIE® Voice lab exam initiate the Cisco Unity Express module,
they occasionally find it stuck in an errant state, where it continuously displays this message across the console:
EXCEPTION: /sw/info/ps: -5: out of memory for event sources
EXCEPTION: /sw/info/ps: -5: out of memory for event sources
Waiting for IOS to register IP address.
- waited 12490 seconds...
WARNING:: IOS communication appears delayed!
WARNING::
WARNING:: Please verify the Service Engine IP Address
WARNING:: and Default Gateway are configured correctly
WARNING:: on the service engine interface in IOS.
EXCEPTION: /sw/info/ps: -5: out of memory for event sources
EXCEPTION: /sw/info/ps: -5: out of memory for event sources
EXCEPTION: /sw/info/ps: -5: out of memory for event sources
This “out of memory” message is caused by the defect CSCeg66388. In this scenario, the Cisco Unity Express module is stuck after repeated failed attempts to establish IP connectivity with the Cisco IOS router.
The solution is to reset the Cisco Unity Express module to recover it from this faulty state. Any of these three methods will solve the problem.
1. Use “--6 x” to return to the Cisco CallManager Express enable mode. Then
reset the Cisco Unity Express module using this command:
CCME# service-module service-Engine 1/0 reset
2. Launch another Telnet session to Cisco CallManager Express and execute the command-
line interface (CLI) above to reset the Cisco Unity Express module.
3. “wr mem” and reload the router. This method is a little dramatic, but it does work.
Here are some additional commands that can be used to manage or retrieve information about the Cisco Unity Express module from the router CLI:
CCME# service-module service-Engine 1/0 ?
password-reset Password reset of Service Module
reload Reload service module
reset Hardware reset of Service Module
session Service module session
shutdown Shutdown service module
status Service Module Information
When encountering unexpected problems, such as this errant state defect, it is important to stay calm and to use common sense and experience to resolve them. Good luck pursuing the CCIE Voice Certification.

CCIE Voice Lab exam overview(both Versions)

2008-12-17T15:27:00.002+03:00

Ok guys and gals, I mentioned in my last post the blueprint for the lab exam has been updated from version 2.0 to 3.0. There's good news for those who'll be sitting for the lab before mid-July 2009.

Lab Exam Refresh: Effective Mid-July 2009, important content changes will be implemented in the CCIE Voice Lab Exam.

Candidates for lab exams scheduled Mid-July 2009 or later should prepare using the v3.0 Lab Equipment and Software Versions. Candidates prior to Mid-July 2009 should continue using the v2.0 Lab Equipment and Software Versions.

. Configuring a network to given specifications
The CCIE lab exam is an eight-hour, hands-on exam which requires you to configure a Cisco enterprise voice solution over an IP network. Although basic network connectivity is provided, you will be responsible for configuring the pre-installed applications to satisfy the requirements of the lab, and to troubleshoot important parameters of a voice network, such as quality of service, VLANs, gateways and gatekeepers.

CCIE Voice Lab blueprint UPDATED!

2008-12-17T14:14:00.003+03:00

If you've been preparing for the lab or have invested on lab equipment based on version 2.0 of the lab blueprint, then you want to tie some loose ends, because version 3.0 of the voice lab blueprint is out.Some things have changed,some have been added and some removed. Let's take a few paragraphs to look at the changes.

First, it is worth noting thatthis blueprint introduces pre-configurations of basic tasks (such as phone registration, basic application integration, basic dial plan, etc.), in order to devote additional focus on expert level skills (advanced configuration and troubleshooting) assessments.
Cat 6500 is out of way now,replaced with Cat 3750 series and no CAT-OS, no VG248, NO ATA,NO FXO and FXS. All old stuff gone.
CallManager 4.1 has been replaced with Unified Communications Manager 7.0
Unity has been replaced with Unity Connection 7.0
Cisco Unified Communications Manager Express 7.0,Cisco Unified Contact Center Express 7.0,Cisco Unified Presence 7.0 are some of the new versions included.

If you've been on version 2.0,don't worry as the underlying concepts and technology of these products are the same.It is just a change in product name and (moderate but not tremendous) changes in the product interfaces.
Below is Blueprint v3.0

CCIE Voice Lab Exam v3.0 Blueprint

The blueprint is a detailed outline of the topics likely to appear on the lab exam. This blueprint introduces pre-configurations of basic tasks (such as phone registration, basic application integration, basic dial plan, etc.), in order to devote additional focus on expert level skills (advanced configuration and troubleshooting) assessments. As usual, knowledge of troubleshooting is an important skill and candidates are expected to diagnose and solve issues as part of the CCIE lab exam. The topics listed are guidelines and other relevant or related topics may also appear.

1.00	Implement and Troubleshoot Campus Infrastructure and Services
1.01	VLAN
1.02	DHCP
1.03	TFTP
1.04	NTP

2.00	Implement and Troubleshoot CUCM Endpoints
2.01	CUCM SCCP Endpoints
2.02	CUCM SIP Endpoints

3.00	Implement and Troubleshoot CUCME Endpoints
3.01	CUCME SCCP Endpoints
3.02	CUCME SIP Endpoints

4.00	Implement and Troubleshoot Voice Gateways
4.01	T1/E1 PRI
4.02	T1/E1 CAS
4.03	H.323
4.04	MGCP
4.05	SIP
4.06	H.323 RAS
4.07	IP-IP Gateway/CUBE

5.00	Implement and Troubleshoot Call Routing Policies
5.01	Route Patterns and Dial-peers
5.02	Digit Manipulations and Translations
5.03	Class of Services
5.04	Route Selection Preference and Redundancy
5.05	Mobility and Single Number Reach

6.00	Implement and Troubleshoot High Availability Features
6.01	SRST
6.02	AAR

7.00	Implement and Troubleshoot Media Resources
7.01	CODEC Selection and Flexibility
7.02	Conference Bridges
7.03	Transcoder
7.04	Music-on-hold
7.05	Media Resources Preference and Redundancy
7.06	Other CUCM Media Resources

8.00	Implement and Troubleshoot Supplementary Services
8.01	Call Park
8.02	Call Pickup
8.03	Barge
8.04	Callback
8.05	Other Supplementary Services

9.00	Implement and Troubleshoot Other CUCM Voice Applications
9.01	Extension Mobility
9.02	IPMA
9.03	Other CUCM Voice Applications

10.00	Implement and Troubleshoot QoS and CAC
10.01	L2/L3 Traffic Classifications and Policing
10.02	L2/L3 Queuing Mechanisms
10.03	L2 LFI
10.04	RSVP
10.05	Call Admission Control

11.00	Implement and Troubleshoot Messaging
11.01	Cisco Unity Connection
11.02	Cisco Unity Express
11.03	Call Handling and Routing

12.00	Implement and Troubleshoot Cisco Unified Contact Center Express
12.01	Advanced Configuration
12.02	Script Customization
12.03	Redundancy

13.00	Implement and Troubleshoot Cisco Unified Presence
13.01	CUCM Presence
13.02	Cisco Unified Presence Server Integration

Tip from a proctor

2008-11-23T15:14:00.002+03:00

The Voice lab Configuration Flow

Through the years of proctoring and spending time in CCIE Voice lab, I have noticed that many candidates use special configuration techniques or “flows” to save time. The CCIE Voice lab exam, as many of you already aware, is configuration intensive, where candidates could easily spend more than half of their allotted lab time entering the configurations.

Naturally, different people have different ways of doing things. However, the goal of the “flow” is the same, that is, to save precious lab time by reducing the frequencies of visiting and updating the devices configuration pages. For example, imagine how many times you will have to visit the same IP phone configuration page to update and reset, if you only update one IP phone configuration setting at a time. Think about location, device pool, media resources group list, AAR group, and the list goes on. Also remember there are seven IP phones in the lab, so time quickly adds up.

Here are some of the configuration tips and flows shared with me by a few Voice CCIEs, I am sure they do not mind me sharing with the engineers on the Cisco Learning Network. Before I proceed, I must mention that these are merely examples of tips I summarized after speaking to past candidates, to give you ideas on some of the popular time-saving techniques used in the lab. These are not mandatory configuration rules so there is no right or wrong.

1. Before starting with any configuration flow, read the entire lab book to gain a high-level understanding of the exam requirements.
2. Configure Cisco CallManager Groups and Date/Time Groups
3. Define Regions and SRST References
4. Define Locations
5. Define and configure Device Pools
6. Define AAR Groups
7. Configure Partitions and Calling Search Spaces
8. Configure Gateways, Route Groups, Route Lists, and Route Patterns
9. Configure Phones

After these steps you should be able to test and verify most of your call routing requirements. Some people also configure media resources media resources, MRG, and MRGL before configuring gateways and phones.

I must emphasis again that these tips are not absolute rules, just because it worked for someone does not mean that it will work for you. I must say that from my observation, candidates who has their own configuration “flow” performs better in the lab exam than those who don’t.

CCIE Voice lab exam overview

2008-11-23T14:55:00.002+03:00

Lab Exam Overview

Configuring a network to given specifications
The CCIE lab exam is an eight-hour, hands-on exam which requires you to configure a Cisco enterprise voice solution over an IP network. Although basic network connectivity is provided, you will be responsible for configuring the pre-installed applications to satisfy the requirements of the lab, and to troubleshoot important parameters of a voice network, such as quality of service, VLANs, gateways and gatekeepers. Point values and testing criteria are provided. More detail is found on the Voice Lab Exam Blueprint and list of Lab Equipment and IOS Versions.

Cost
Lab exams cost US$1400 per attempt, not including travel and lodging expenses. Costs may vary due to exchange rates and local taxes (VAT, GST). You are responsible for any fees your financial institution charges to complete the payment transaction.

Price not confirmed and is subject to change until full payment is made. Please reference the CCIE Policies for more information.

Lab Environment
The Cisco documentation CD is available in the lab room, but the exam assumes knowledge of the more common protocols and technologies. As of March 2006, the documentation can only be navigated using the index; the search function has been disabled. No outside reference materials are permitted in the lab room. You must report any suspected equipment issues to the proctor during the exam; adjustments cannot be made once the exam is over.

Lab Exam Grading
Each question on the lab has specific criterion. The labs are graded by proctors who ensure all the criterion are met and points are awarded accordingly. The proctors use automatic tools to gather information from the routers to perform some preliminary evaluations, but the final determination of a correct or incorrect configuration is done by a trained proctor.

Results
You can review your lab exam results online (login required), usually within 48 hours. Results are Pass/Fail and failing score reports indicate major topic areas where additional study and preparation may be useful.

Reevaluation of Lab Results
You may request a reevaluation of results for Routing and Switching, Security and Service Provider labs for up to 14 days following your exam date. Use the link next to your lab record called "Request for Reread". Due to the equipment used, rereads are not available for the Voice and Storage Networking exams. Each reread costs US$250 plus any applicable local taxes. Payment is made online via credit card and your card will be charged upon receipt of the request. You may not cancel the reread request once the process has been initiated and refunds are only given when the results change from Fail to Pass.

A reread consists of a second proctor loading your configurations onto a rack to recreate the test and rescore the entire exam. This process may take up to three weeks after receipt of payment. Only one reread per lab attempt is permitted. The result of the reread is an updated score report with success rates for each major section. Be aware that scores may decrease. Exams receive a Pass mark only when the total exam score exceeds 80%. Before requesting a reread, consider that, historically, only 0.3% of exams have been changed from Fail to Pass.

Voice Lab Locations
Security exams are offered at the Cisco locations below. Additional information, such as start times, directions and hotel recommendations can be found on the following links.

CCIE Voice Prep

2008-08-01T15:07:00.002+03:00

As I said in my last post,I've been preparing for the CCIE Voice lab since May but things aren't easy.There's just too much to read. I don't know why Cisco decides to make us study all these but the blueprint for that exam carries a lot.
While I was browsing the web, I came across some important article at Network World. It said 'How to pass the CCIE Voice written exam' by Mark Lewis. I thought this is worth sharing on my blog.I'm putting an excerpt of it up here.

The CCIE Voice written exam consists of 100 multiple choice questions, covering a wide range of topics. Preparation for the exam will consist, to a very large extent, of reading.

The amount of reading will depend, of course, on your level of knowledge and expertise in each of areas covered by the exam. In order to maximize the efficacy of your study and minimize the amount of reading that you have to do, it’s very important to select the correct preparation resources (books and documents).

But before taking a closer look at preparation resources, first take a look at the exam blueprint. It’s a good idea to print it out, and take a few minutes to carefully consider how strong you are in relation to each of the objectives. Once you have assessed yourself against the main objectives, you can decide in which order you want to tackle them when preparing for the exam.

You may decide to approach the blueprint in a top-down fashion – just start with CallManager, and run through the objectives one-by-one. Or you may decide to fill in the gaps with subjects where are stronger before tackling those subjects where you are weaker. Or you may even decide to tackle the toughest subjects first, while enthusiasm is highest!

Don’t be under any illusions when tackling the CCIE Voice written – it may, depending on your current level of knowledge and experience, take some substantial effort. But console yourself with the fact that reading through all of the resources necessary to pass the written exam will give you a great platform to go for the lab exam.
One of the particular difficulties of the CCIE Voice written exam is that there is relatively little overlap between the CCIE Voice track and the other tracks. So, if you are coming from a routing/switching background then the learning curve for this exam may be pretty steep. Having said that, it is definitely possible to pass the CCIE Voice written exam with a little study.

It’s probably best to set aside a certain amount of time every day to study for the written exam, and stick to that schedule. You may find it useful to (briefly) record in a journal what exam preparation you do every day.

So, let’s have a look at study resources that you can use in preparing for the exam.

Exam Objective 1: CallManager

The first item on the blueprint is CallManager (CCM). If you know anything at all about Cisco’s enterprise IP telephony products and solutions, then you’ll know that their solution is centred on CCM. Given this fact, it’s a pretty good bet that you’ll have to have a pretty firm grasp of CCM in order to pass the written exam.

If you have passed the CCVP 642-444 CIPT-4.x (Cisco IP Telephony for Release 4.x) exam then you’ll have a good base for CCM related questions on the exam, but you will probably need to add to your knowledge.

So, what are the best resources for someone preparing for the CCM portion of the CCIE Voice written? Well, the ‘Cisco CallManager Fundamentals, Second Edition’ (Cisco Press) is an excellent book for preparing for the exam. In addition, ‘Cisco Unified Communications SRND Based on Cisco Unified CallManager 4.x’ is also essential reading.

You can find ‘Cisco Unified Communications SRND Based on Cisco Unified CallManager 4.x’, together with other Solution Reference Network Design documents (SRNDs) here. Other documents referenced in this blog post can easily be found by searching for them on the Cisco website. Another resource that you may want to read is ‘Cisco IP Video Telephony Solution Reference Network Design (SRND) Cisco CallManager Release 4.0’.

Having read the book and documents (and understood and remembered them!), you should be in fairly good shape for the CCM portion of the exam.

Exam Objective 2: QoS

Next on the blueprint is Quality of Service (QoS). If you come from a routing and switching background, this one of the few areas where you may be able to leverage existing knowledge.

Two resources that are very useful in preparation for the QoS related aspects of the written exam are ‘End-to-End QoS Network Design: Quality of Service in LANs, WANs, and VPNs’ (Cisco Press) and ‘Enterprise QoS Solution Reference Network Design Guide’.

Exam Objective 3: Telephony Protocols

Third on the list of possible exam topics listed in the blueprint is ‘Telephony Protocols’. These telephony protocols include MGCP, SIP, H.323, SCCP, RTP/cRTP, and analog/TDM signalling.

Pretty much all IP telephony related books will cover these protocols to some extent, but two books that are perhaps more focused on them than other books are ‘Voice Over IP Fundamentals’ (Cisco Press) and ‘Integrating Voice and Data Networks’ (Cisco Press). ‘Integrating Voice and Data Networks’ is now out-of-print, so you may have to resort to ebay in order to get hold of a copy. Alternatively, you can read it online at the Safari website.

‘Voice Over IP Fundamentals’ and ‘Integrating Voice and Data Networks’ also cover areas not required for the written exam. For example, based on a quick read of the blueprint, it’s probably a fair bet that SS7 is not going to be a major focus of the exam. So, you can probably save some valuable time by reading only the chapters of these two books that cover subjects shown in the blueprint.

Exam Objective 4: Unity

Next is Unity. A good start will be ‘Cisco Unity Fundamentals’ (Cisco Press). Once you have read that, a good book to read in order to bring your level of knowledge up to a more advanced level is ‘Cisco Unity Deployment and Solutions Guide’ (Cisco Press).

Exam Objective 5: IOS IP telephony skills

IOS IP Telephony skills include SRST, CME, and CUE.

One very useful book to read when preparing for the exam is ‘Cisco IP Communications Express: CallManager Express with Cisco Unity Express’.

Useful documents include ‘Cisco Unified CallManager ExpressSolution Reference Network Design Guide’ and ‘Cisco Unity Express Design Guide’. Useful information on SRST can be found here.

If exam day is approaching and you need to get up to speed quickly, it’s probably a good idea to read the documents first followed by the book (if you have time).

Exam Objective 6: IP IVR/IPCC Express

Okay, so much for CME, SRST, and CUE. But how about the next exam objective, IP IVR/IPCC Express?

It is a little more difficult to find good resources for IP IVR/IPCC simply due to the fact that there haven’t been any books published on the subject. So, a little more digging around for good information is required.

A good place to start is the ‘Cisco IPCC Express Edition Solution Reference Network Design’. This is good, but it will almost certainly not be enough to bring you up to the level of knowledge required for the exam.

Supplement the SRND with other resources including Cisco Customer Response Solutions Administration Guide, Cisco Customer Response Solutions Editor Step Reference Guide (Cisco CRS Scripting and Development Series: Volume 2), and Cisco Customer Response Solutions Getting Started with Scripts (Cisco CRS Scripting and Development Series: Volume 1).

Exam Objective 7, 8, 9, and 10 : Security, Infrastructure protocols, Application Protocols, Operation and Network Management

Information on the last four exam objectives (security, infrastructure protocols, application protocols, operations and network management) cannot be found in one particular book or document.

One excellent book to read is, ‘Troubleshooting Cisco IP Telephony’. Other than that book, you’ll find information about these exam objectives sprinkled throughout a number of the books and documents that I have already mentioned.

The approach that I recommend with regard to these exam objectives is to leave them until last, read ‘Troubleshooting Cisco IP Telephony’, and then fill in any gaps in your knowledge.

Final preparation:

So, you have stepped through all the study resources mentioned above, and you are almost ready to book the exam. One last resource that will be very useful is my ‘CCIE Voice Exam Quick Reference Sheets (Digital Short Cut)’. This will help you to quickly fill in any gaps in your knowledge, and also give you a good idea of the level of knowledge that you are going to need in order to pass the exam.

At this point, you’ll be ready to book your exam. Before finishing it is worth mentioning time management for the exam. Don’t forget that there are 100 questions to answer in 120 minutes. So, that’s just a little over a minute per question. Don’t get sucked into spending too much time on one question. If you have a problem with a question, mark it and go back to it at the end.

Good luck on the exam!