>> The one thing I'm really struggling with is mapping out my dial-plan
during my read through of the lab.  I would love to hear what others are
doing.

In my previous attempts, I find it very hard too, because the questions are
verbose and I could either spend too much time reading OR not able to
encode it into the table form correctly in haste OR simply skipped and
waste too much time re-reading it.

Here is my plan for my next attempt.    I think the key is to have my
pre-fabricate table then I will create my table quickly and ONLY adjust it
while I read the question.

So, I would quickly create this template.  The list is there for easy cut &
paste.  I will only complete Site A in during the lab, then I will just
copy to SiteB & SiteC.  The rest is just modification of the table.
 (Note: I find that using "TAB" make it easier to align the columns, it
could be 3 or 4 TABs.)

In essence, focus on [1] Pre-fabrication  [2]  Quick to reproduce as a
template.   The rest is depending on how quick you can decipher verbose
question and re-adjust the table.

Get the screenshot here is the following format is bad -
https://www.dropbox.com/s/o1ftbh4katnhjcu/Quick-Dial-Plan-Template.jpg,
The TXT version is here -
https://www.dropbox.com/s/m2p2lte5ezk6q7d/Quick-Dial-Plan-Template.TXT

I think Matthew Berry youtube is good too -
http://www.youtube.com/watch?v=4mP5powuFUM.    I just think it can be
complement by the cookie-cutting approach to encode the table in Notepad
that can be reproduced quickly.

===============================
The LIST
===============================

ISDN
Unknown
Subscriber
National
International
Any

===============================
SiteA
===============================

Calling Called

Emer 7D / Unknown / ISDN 7D / Unknown / ISDN
Local 7D / Subscriber / ISDN 7D / Subscriber / ISDN
LD 10D / National / ISDN 10D / National / ISDN
Intl 1+10D / International / ISDN 011! / International / ISDN


Another variation of the table format is too cater for TEHO scenario or
BACKUP Gateway scenario.

===============================
SiteA
===============================

Calling Called

Emer  7D / Unknown / ISDN 7D / Unknown / ISDN
Local  7D / Subscriber / ISDN 7D / Subscriber / ISDN
Local(Backup via SiteB) 7D / Subscriber / ISDN 7D / Subscriber / ISDN
LD 10D / National / ISDN 10D / National / ISDN
Intl 1+10D / International / ISDN 011! / International / ISDN

Regards,
--Somphol.


On Fri, Oct 18, 2013 at 12:22 PM, VanBenschoten, Brian <
brian.vanbenscho...@corebts.com> wrote:

>  I've found the QoS questions are very specific to test a certain area of
> knowledge.  They are not looking for what we would consider a "best
> practice" system wide.  I think we could skip setting the DSCP values in
> CUCM.
>
> If you think the question calls for it you can have your class-map match
> both AF31 and CS3 for signaling.
>
>
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Bill Hatcher
> *Sent:* Thursday, October 17, 2013 3:14 PM
> *To:* probert...@gmail.com
> *Cc:* ccievoice
> *Subject:* Re: [OSL | CCIE_Voice] Base configs and Dial Plan mapping
>
>
>
> I agree to setting the service parameters to default first.  I was
> planning on doing that myself.  As to changing the DSCP values, it all
> depends on what they ask for in the QoS section of the test is all.
>
>
>
> On Thu, Oct 17, 2013 at 2:24 PM, probert...@gmail.com <
> probert...@gmail.com> wrote:
>
> Hi,
>
> I think my strategy will be to set all Service Parameters to default
> before making changes. This way I can avoid and undesirable presets.
>
> Let me know your thoughts on this.
>
> Also why are you setting DSCP for Phone Configuration and DSCP for Cisco
> CallManager to Device Interface to AF31? Default CS3 should be good, let me
> know if I'm wrong on this?
>
>
>
> Rob
>
>
>
> On Thu, Oct 17, 2013 at 1:39 PM, Bill Hatcher <wchatc...@gmail.com> wrote:
>
>   My test is just a couple of weeks away, and I've been reading different
> blogs on how to maximize your time.  The one thing I'm really struggling
> with is mapping out my dial-plan during my read through of the lab.  I
> would love to hear what others are doing.
>
> I have also been building base router configs for h323, gatekeeper, mgcp,
> srst,sip, etc so that I can practice quickly configuring those on the
> routers.
>
> One of the things I haven't really been keeping track of are some of the
> service parameters that I should adjust out of habit. Here are a few that I
> can think of off the top of my head that I plan on tweaking at the start of
> the exam.  Please feel free to add to them.
>
> Enterprise Parameters
> DSCP for Phone Configuration - Set to AF31
> DSCP for Cisco CallManager to Device Interface - Set to AF31
>
> Change the Phone URL's to IP's
>
> Organization Top Level Domain
>
> Cluster Fully Qualified Domain Name
>
>
> Service Parameters - CallManager
>
> T302 Time - Know where it is if you need ot change interdigit timeout
>
> Call Classification - Offnet
>
> Builtin Bridge Enabled - True
>
> Device Name of GK-controlled Trunk That Will Use Port 1720 - Set as needed.
> Transfer On-hook Enabled - True (Also a great thing to do in production
> when migrating users from other phone systems)
> Block offnet to offnet transfers - Know where it's at.
> Auto Call Pickup Enabled - True
> Call Back Enabled Flag - True (Verify)
> Single Button Barge/CBarge Policy - Set to Barge unless otherwise directed.
> Stop routing on Unallocated Number Flag - False - H323 redundancy
>
> Preferred G.711 Millisecond Packet Size - 20 (Verify)
> Preferred G.729 Millisecond Packet Size - 20 (Verify)
>
> G722 Codec Enabled - Disabled (Unless otherwise directed)
>
> Intraregion Audio Codec Default - G711/G722 (Verify)
>
> Interregion Audio Codec Default - G729 (Verify)
>
> Automated Alternate Routing Enabled - True (This one gets me every time on
> AAR so I turn it on by default now)
>
> Enable Mobile Voice Access - Set as required
> Mobile Voice Access Number - Set as required
>
> System Remote Access Blocked Numbers - Set as required
>
> Service Parameters -Cisco IP Voice Media Streaming App
>
> Supported MOH Codecs - G711 mulaw and G729 Annex A
>
>
>
> HTH
>
> Bill Hatcher
>
>
>
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