[OSL | CCIE_Voice] SRST

[Scott Monasmith]

After Branch1 returns from SRST, the serial interface does not put the 'isdn
bind-l3 ccm-manager' command back on the PRI interface. Does anyone know
why?

-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] Gatekeeper Bandwith

[Scott Monasmith]

Interzone bandwidth restricts bandwidth between zones. Are you using more
than one zone in your gatekeeper config?

On Sun, May 11, 2008 at 12:32 PM, Onur Tufekci <[EMAIL PROTECTED]>
wrote:

> I am looking at the bandwidth utilizations from CCM to CME and CME to CCM.
>
> Calls that are going out from CCM are showing as using 16 K and calls that
> are originated from CME are showing up as 128 K. My interzone limit 64 K so
> I am really confused about how this might be happening. Any ideas from any
> one? I checked my regions and codec setting on the dial-peers at least 4
> times.
>
>  GATEKEEPER ZONES
>  
> GK name  Domain Name   RAS Address PORT  FLAGS
> ---  ---   --- - -
>
> HQ-RTRipexpert.com 10.X.200.3 1719  LS
>   BANDWIDTH INFORMATION (kbps) :
> Maximum total bandwidth : unlimited
> *Current total bandwidth : 128.0*
> *Maximum interzone bandwidth : 64*
> Current interzone bandwidth : 0.0
> Maximum session bandwidth : unlimited
>* Total number of concurrent calls : 1*
>   SUBNET ATTRIBUTES :
> All Other Subnets : (Enabled)
>   PROXY USAGE CONFIGURATION :
> Inbound Calls from all other zones :
>   to terminals in local zone HQ-RTR : use proxy
>   to gateways in local zone HQ-RTR  : do not use proxy
>   to MCUs in local zone HQ-RTR  : do not use proxy
> Outbound Calls to all other zones :
>   from terminals in local zone HQ-RTR : use proxy
>   from gateways in local zone HQ-RTR  : do not use proxy
>
>


-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] router BACD TCL script

[Scott Monasmith]

1. www.cisco.com/univerCD
2. click on the All Product Documentation link
3. click on the Voice Products link
4. click on the Cisco CallManager Express and Cisco IOS Telephony link
5. click on the Cisco Unified CallManager Express 4.0 link
6. click on Cisco Unified CME B-ACD and Tcl Call-Handling Application link.
7. click on Cisco Unified CME Basic Automatice Call Distribution and
Auto-Attendant Service (B-ACD) link.

On Thu, May 8, 2008 at 1:38 AM, Chuck Wow <[EMAIL PROTECTED]> wrote:

> Does anyone know the shortcut to find this info on the Cisco
> documentation website?
>
> thanks,
>
> Chuck
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

[OSL | CCIE_Voice] MLP LFI: FRTS or GTS?

[Scott Monasmith]

When configuring MLP LFI can you use either FRTS or Generic Traffic Shaping?

One of IPExpert's labs show an example of using GTS. However, the QoS SRND
and another Voice CCIE mentioned to me that you had to use FRTS with MLP
LFI. Can anyone clarify this?

Cheers,

Re: [OSL | CCIE_Voice] Question

[Scott Monasmith]

Almost :)

On Sat, Apr 19, 2008 at 2:23 PM, Gregory Jost (grjost) <[EMAIL PROTECTED]>
wrote:

>  Would anyone out there besides me rather take a claw hammer to the face
> than deal with L2 QoS?
>
>
>
>
>
> Greg Jost
>
> Network Consulting Engineer
>
> Unified Communications Practice
>
> Cisco Systems, Inc.
>
> 214-274-1922
>
>
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] UniverCD is toast...

[Scott Monasmith]

In RTP, I noticed an IPCC admin .PDF on the desktop now.

On Sun, Apr 20, 2008 at 10:02 AM, Edward French <[EMAIL PROTECTED]>
wrote:

>  My problem was the url for the phone agent
>
>  - Original Message 
> From: Jonathan Charles <[EMAIL PROTECTED]>
>  To: Edward French <[EMAIL PROTECTED]>
> Cc: CCIE Voice 
> Sent: Sunday, April 20, 2008 10:45:43 AM
> Subject: Re: [OSL | CCIE_Voice] UniverCD is toast...
>
> Well, um... there is a probability that we would need that too...
>
>
>
> Jonathan
>
> On Sat, Apr 19, 2008 at 7:36 PM, Edward French <[EMAIL PROTECTED]>
> wrote:
> >
> > Friday I had access to everything I needed except IPCCX
> >
> >
> > - Original Message 
> > From: Jonathan Charles <[EMAIL PROTECTED]>
> > To: CCIE Voice 
> > Sent: Saturday, April 19, 2008 11:03:35 AM
> > Subject: [OSL | CCIE_Voice] UniverCD is toast...
> >
> >  So, what, if anything, do we get to use on the lab?
> >
> >
> >
> > Jonathan
> >
> >
>
>


-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] CCM NTP

[Scott Monasmith]

This may not work on the IPExpert servers. I have been able to get this
config to work, but it takes about 15 minutes for the time to change/sync.

On Sat, Apr 19, 2008 at 10:02 AM, Gregory Jost (grjost) <[EMAIL PROTECTED]>
wrote:

>  This never works for me.
>
>
>
> 1.Stop Windows Time Service (W32Time)
>
> 2.Ensure NTP is started and startup automatic
>
> 3.Edit NTP.conf
>
> 4.Restart NTP
>
>
>
>
>
>
>
> Greg Jost
>
> Network Consulting Engineer
>
> Unified Communications Practice
>
> Cisco Systems, Inc.
>
> 214-274-1922
>
>
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] GK with Tech-Prefix

[Scott Monasmith]

Yes you can route via GK with just using different tech prefixes WITHOUT
using zone prefixes

On Fri, Apr 18, 2008 at 11:31 PM, Devildoc <[EMAIL PROTECTED]> wrote:

> Hello,
>
> Perhaps too much studying has scrambled my brains, but i swear that in the
> past i have gotten gk to work correctly with just tech prefix alone and no
> zone prefix.  So my question is this... can gk work correctly alone on just
> tech prefix?  Or does the gk also need a zone prefix in addition to tech
> prefix?
>
> Tonight i tried to configure the gk with only tech prefix.  Each gateway
> registered to the gk with a different tech prefix and the originating
> gateway sent the destination tech-prefix along with the destination route
> pattern to the destination gateway.  I did not configure any default
> technology prefix nor did i configured any zone prefix.  That did not work.
> I worked as soon as I added the zone prefix to the configuration.  I though
> you don't need a zone prefix when configuring a tech prefix.  And i swear i
> did that before and it worked fine but not tonight.  Anyone has any thought
> on this?  Thanks.
>
> JD
>
>
> --
> Get in touch in an instant. Get Windows Live Messenger 
> now.
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] how many ways to integrate cm with cme?

[Scott Monasmith]

Vik,

Don't you need an IPIPGW on the CME router if you are using SIP inbound to
the CME?

On Wed, Apr 16, 2008 at 7:16 PM, Vik Malhi <[EMAIL PROTECTED]> wrote:

>  SIP trunk to CME is a possibility. As is H323 gateway. As is gk
> integration.
>
> For SIP...on the CCM side you have a SIP trunk, on the CME side you have a
> SIP dial-peer pointing to the CallManager. A SIP dial-peer is a dial-peer
> with the command "session protocol sipv2"
>
>
> Vik Malhi – CCIE #13890
> Senior Technical Instructor - IPexpert, Inc.
>
> Telephone: +1.810.326.1444
> Fax: +1.810.454.0130
> Mailto: [EMAIL PROTECTED] <[EMAIL PROTECTED]>
>
> Join our free online support and peer group communities:
> http://www.IPexpert.com/communities 
>
> IPexpert - The Global Leader in Self-Study, Classroom-Based,
> Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE
> R&S Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and
> CCIE Storage Lab Certifications.
>
>
>  --
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *
> [EMAIL PROTECTED]
> *Sent:* Wednesday, April 16, 2008 4:46 PM
> *To:* Jonathan Charles
> *Cc:* ccie_voice@onlinestudylist.com
> *Subject:* Re: [OSL | CCIE_Voice] how many ways to integrate cm with cme?
>
>   Thanks Jonathan,
>
> What about sip integration?
>
> Sara
>
> *Jonathan Charles <[EMAIL PROTECTED]>* wrote:
>
> Just set up CME as an H.323 gateway.
>
> Hook up both CCME and CCM to an intervening gatekeeper.
>
> You will need to setup a transfer pattern on CCME to allow them to CCM.
>
> On the CCME side, enable H.450.12
>
>
>
> Jonathan
>
> On Wed, Apr 16, 2008 at 6:23 PM, wrote:
> > besides adding inter-cluster trunk in cm, is there any other method to
> > integrate cm with cme?
> >
> > any special things to take note regarding call-transfer?
> >
> >
> > Sara
> >
> >
> > 
> > GANBARE! NIPPON! Win your ticket to Olympic Games 2008.
> >
>
>
>  --
> GANBARE! NIPPON! Win your ticket to Olympic Games 
> 2008.
>
>


-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] Non-GK ICT from CCM to CCME?

[Scott Monasmith]

Correct. You only need to do a digit-strip when going from SIP to H323 when
using RFC2883 on the SIP side so the H323 call device doesn't get dtmf tones
both in-band and out-of-band.

Secondly, SIP to H323 is supported. The only thing I can think of that is
unsupported, relative to what we are talking about, is that transcoding
between SIP to SIP call legs is unsupported.
On Thu, Apr 17, 2008 at 10:35 AM, Jonathan Charles <[EMAIL PROTECTED]>
wrote:

> Also, silly question on 4.9:
>
> We have CCM - IPIPGW - CCME
>
> CCM to IPIPGW to CCME is H323 to SIP
> CCME to IPIPGW to CCM is H323 to SIP
>
> If I am reading it correctly
>
> Then it says to ensure when calls are coming from SIP to H323 that we
> strip RFC2833 but we aren't going from SIP to H323, just the other
> way around... I also thought SIP to H323 wasn't supported...
>
>
>
> Jonathan
>
> On Thu, Apr 17, 2008 at 9:17 AM, Gregory Jost (grjost) <[EMAIL PROTECTED]>
> wrote:
> > Oh. I guess that would be "literally" huh...  :)
> >
> >  I only know what I talking about a fraction of the time.  Maybe I
> should
> >  stay off these forums.
> >
> >
> >
> >  Greg Jost
> >  Network Consulting Engineer
> >  Unified Communications Practice
> >  Cisco Systems, Inc.
> >  214-274-1922
> >
> >
> >  -Original Message-
> >
> >
> > From: Jonathan Charles [mailto:[EMAIL PROTECTED]
> >  Sent: Thursday, April 17, 2008 9:13 AM
> >  To: Gregory Jost (grjost)
> >  Cc: Jacob Owen; CCIE Voice
> >  Subject: Re: [OSL | CCIE_Voice] Non-GK ICT from CCM to CCME?
> >
> >  They have a screenshot of the non-GK ICT being created
> >
> >
> >  Jonathan
> >
> >  On Thu, Apr 17, 2008 at 9:11 AM, Gregory Jost (grjost)
> >  <[EMAIL PROTECTED]> wrote:
> >  > I think ICT is being used figuratively here (e.g. a trunk between
> >  >  disparate systems), not literally.
> >  >
> >  >
> >  >  Greg Jost
> >  >  Network Consulting Engineer
> >  >  Unified Communications Practice
> >  >  Cisco Systems, Inc.
> >  >  214-274-1922
> >  >
> >  >
> >  >
> >  >
> >  >  -Original Message-
> >  >  From: [EMAIL PROTECTED]
> >  >  [mailto:[EMAIL PROTECTED] On Behalf Of
> Jonathan
> >  >  Charles
> >  >  Sent: Thursday, April 17, 2008 9:03 AM
> >  >  To: Jacob Owen
> >  >  Cc: CCIE Voice
> >  >  Subject: Re: [OSL | CCIE_Voice] Non-GK ICT from CCM to CCME?
> >  >
> >  >  No, I am looking at the solution thingy, and it shows a non-GK
> >  >  controlled ICT trunk... bizarre... I thought this was exclusively
> for
> >  >  connecting to another CCM cluster... in fact, the SRND says so...
> >  >
> >  >  Never tried hooking it up to an IPIPGW...
> >  >
> >  >
> >  >
> >  >  Jonathan
> >  >
> >  >  On Thu, Apr 17, 2008 at 8:59 AM, Jacob Owen <[EMAIL PROTECTED]>
> >  >  wrote:
> >  >  > Jonathan,
> >  >  >  The 172.x.10y.1 addresses are the loopbacks for the
> >  >  >  devices:
> >  >  >
> >  >  >  172.x.100.1 - HQ Router (x is pod number)
> >  >  >  172.x.101.1 - BR1 Router
> >  >  >  172.x.102.1 - BR2 Router
> >  >  >
> >  >  >  I think it's probably an ICT trunk to an H323
> >  >  >  Gatekeeper but I don't have the book handy.
> >  >  >
> >  >  >
> >  >  >
> >  >  >  --- Jonathan Charles <[EMAIL PROTECTED]> wrote:
> >  >  >
> >  >  >  > Does this work? Looking at solution for 4.9 in the
> >  >  >  > workbook, and it
> >  >  >  > shows a regular ICT but the IP address, 172.1.100.1
> >  >  >  > doesn't match up
> >  >  >  > to anything... I am curious what it is...
> >  >  >  >
> >  >  >  > BR2's loopback should be 172.X.102.1 not 100.1...
> >  >  >  > typo or am I lost?
> >  >  >  >
> >  >  >  >
> >  >  >  >
> >  >  >  >
> >  >  >  >
> >  >  >  > Jonathan
> >  >  >  >
> >  >  >
> >  >  >
> >  >  >  Jacob Owen
> >  >  >  CCIE #14063 (R&S, Service Provider), CCVP, CCDP
> >  >  >
> >  >  >
> >  >  >
> >  >  >
> >  >
> >
>  
> >  >  
> >  >  >  Be a better friend, newshound, and
> >  >  >  know-it-all with Yahoo! Mobile.  Try it now.
> >  >  http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ
> >  >  >
> >  >
> >
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] Multicast MOh

[Scott Monasmith]

I don't know why the stats are displaying under show call-manager-fallback
since you don't have 'moh music-on-hold.au' configured. I don't know if
these stats reflect multicast group announcements. I would venture to guess
that since the phone was configured to use multicast moh (via the mrgl)
using the ip of 239.1.1.1 and your router is configured to multicast moh
traffic with 239.1.1.1 that it sends out multicast packets without regard to
whether it contains music or not. BUT THAT IS JUST A GUESS.

On Fri, Apr 11, 2008 at 12:27 PM, jason sung <[EMAIL PROTECTED]> wrote:

> Thanks Scott,
>
> That is what I wanted to know. I thought so too, but your confirmation
> adds to my confidence.
>
> I do see the ccm-manager music-on-hold stats, but I have not configured
> any moh source under call-manager-fallback so why do i see the stats?
>
> Is the router acting up and picking up old configs (for which I did have
> moh configured), may be it needs a reboot???
>
>   On Fri, Apr 11, 2008 at 12:09 PM, Scott Monasmith <[EMAIL PROTECTED]>
> wrote:
>
> > Correct. I was just answering his initial question regarding the perfmon
> > counters.
> >
> > you can also run 'debug ephone moh'
> >
> >   On Fri, Apr 11, 2008 at 12:06 PM, Jonathan Charles <[EMAIL PROTECTED]>
> > wrote:
> >
> > > Right, but that does not prove it is working, the show ccm music will
> > > prove it tho...
> > >
> > >
> > > Jonathan
> > >
> > > On Fri, Apr 11, 2008 at 12:05 PM, Scott Monasmith <[EMAIL PROTECTED]>
> > > wrote:
> > > > Jason,
> > > >
> > > > If you configure your branch router to source Multicast MoH via
> > > SRST, you
> > > > should still see the MoH perfmon counters increase even though the
> > > MoH is
> > > > not traversing the WAN.
> > > >
> > > >
> > > >
> > > > On Fri, Apr 11, 2008 at 10:53 AM, jason sung <[EMAIL PROTECTED]>
> > > wrote:
> > > >
> > > > >
> > > > > Should I see my perf mon counters increase when i am using remote
> > > site
> > > > router as the moh source.
> > > > >
> > > > > Here is how I am trying to isolate.
> > > > >
> > > > > I have multicast-routing disabled on the wan interfaces (no ip pim
> > > > dense-mode)
> > > > >
> > > > > My MOH server source hop count is set to 1.
> > > > >
> > > > > Remote Site MRG is set to use Multicast MOH on callmanager
> > > configuration.
> > > > >
> > > > > I have no moh configured on the remote site router, but I do see
> > > my
> > > > Perfmon counters increase.
> > > > >
> > > > > should the end result be no MOH??
> > > >
> > > >
> > > >
> > > > --
> > > > "There are only 10 types of people in the world: Those who
> > > understand
> > > > binary, and those who don't"
> > >
> >
> >
> >
> > --
> >  "There are only 10 types of people in the world: Those who understand
> > binary, and those who don't"
> >
>
>


-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] Multicast MOh

[Scott Monasmith]

Correct. I was just answering his initial question regarding the perfmon
counters.

you can also run 'debug ephone moh'

On Fri, Apr 11, 2008 at 12:06 PM, Jonathan Charles <[EMAIL PROTECTED]>
wrote:

> Right, but that does not prove it is working, the show ccm music will
> prove it tho...
>
>
> Jonathan
>
> On Fri, Apr 11, 2008 at 12:05 PM, Scott Monasmith <[EMAIL PROTECTED]>
> wrote:
> > Jason,
> >
> > If you configure your branch router to source Multicast MoH via SRST,
> you
> > should still see the MoH perfmon counters increase even though the MoH
> is
> > not traversing the WAN.
> >
> >
> >
> > On Fri, Apr 11, 2008 at 10:53 AM, jason sung <[EMAIL PROTECTED]> wrote:
> >
> > >
> > > Should I see my perf mon counters increase when i am using remote site
> > router as the moh source.
> > >
> > > Here is how I am trying to isolate.
> > >
> > > I have multicast-routing disabled on the wan interfaces (no ip pim
> > dense-mode)
> > >
> > > My MOH server source hop count is set to 1.
> > >
> > > Remote Site MRG is set to use Multicast MOH on callmanager
> configuration.
> > >
> > > I have no moh configured on the remote site router, but I do see my
> > Perfmon counters increase.
> > >
> > > should the end result be no MOH??
> >
> >
> >
> > --
> > "There are only 10 types of people in the world: Those who understand
> > binary, and those who don't"
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] Multicast MOh

[Scott Monasmith]

Jason,

If you configure your branch router to source Multicast MoH via SRST, you
should still see the MoH perfmon counters increase even though the MoH is
not traversing the WAN.

On Fri, Apr 11, 2008 at 10:53 AM, jason sung <[EMAIL PROTECTED]> wrote:

> Should I see my perf mon counters increase when i am using remote site
> router as the moh source.
>
> Here is how I am trying to isolate.
>
> I have multicast-routing disabled on the wan interfaces (no ip pim
> dense-mode)
>
> My MOH server source hop count is set to 1.
>
> Remote Site MRG is set to use Multicast MOH on callmanager configuration.
>
> I have no moh configured on the remote site router, but I do see my
> Perfmon counters increase.
>
> should the end result be no MOH??
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] 6608 Gateway and resources

[Scott Monasmith]

Are the CCM and TFTP services running on CallManager?

On Wed, Apr 9, 2008 at 8:53 PM, Paul and Bobs <[EMAIL PROTECTED]> wrote:

> Trying to get the 6608 gateway to register.
>
> I have enable the interface in 6500 and can see that it has ip address and
> address of tftp server.
>
> I have then added it to the ccm as a 6000 t1 gateway with PRI.
>
> I can seem to get it or the resources to register.
>
> Paul
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] First Failed Attempt at CCIE LAb

[Scott Monasmith]

__
> >  Date: Mon, 7 Apr 2008 17:35:57 +0200
> > From: [EMAIL PROTECTED]
> > To: ccie_voice@onlinestudylist.com
> >
> >
> > Subject: Re: [OSL | CCIE_Voice] First Failed Attempt at CCIE LAb
> >
> >
> > You can try, but you are probably wasting time.
> > I`ve asked the same question (because i`ve had the same issues) at
> > networkers in Barcelona, but no results.
> > The statement is, we don`t care how you do it, as long as it works, so
> for
> > instance , if you have 4 questions that can only be tested if you have
> an E1
> > up, however you failed to set it up correctly.
> > And you are sure you have the 4 remaining questions correct, you will
> fail
> > all 5 questions (basterds !!!)
> >
> > It`s not fair i know, however at the moment we are stuck with this kind
> of
> > grading.
> >
> > sorry.
> >
> > erik
> >
> >
> > On Mon, Apr 7, 2008 at 5:16 PM, Devildoc <[EMAIL PROTECTED]>
> wrote:
> >
> >
> > Well, if that's the case, then i am horrified at my score.  I mean my
> score
> > for the call routing section is 0%.  How is that even possible?  I knew
> that
> > i didn't have time to complete the test, so i spent extra time on just
> one
> > section to see how it is graded.  I verified that all calls were
> successful
> > and all requirements were met.  Hell... there were many questions in
> that
> > section, and i received zero point for that section?
> >
> > Does anyone know if there is a Cisco contact where one can inquire some
> > questions about the lab exam and its grading process?
> >
> >
> > JD
> >
> >
> >
> >
> >
> >  
> >  Date: Sat, 5 Apr 2008 13:44:46 -0500
> > From: [EMAIL PROTECTED]
> > To: [EMAIL PROTECTED]
> > CC: ccie_voice@onlinestudylist.com; [EMAIL PROTECTED]
> >
> > Subject: Re: [OSL | CCIE_Voice] First Failed Attempt at CCIE LAb
> >
> >
> >
> >
> >
> > Here is an excerpt from the report card.
> >
> > "Your CCIE Certification Lab exam was scored based on grading policies
> that
> > are adhered to uniformly by our proctors worldwide. Marking was based on
> > whether the answer you provided works. Candidates are not required to
> use a
> > set methodology in achieving a correct result. The imperative is that
> the
> > solution provided produces the outcome requested. "
> >
> > I do believe that they actually grade you correctly but again there are
> > chances of proctors messing up as well.
> >
> > Like Scott mentioned, I would just focus on things I can control and
> forget
> > the rest... better luck next time buddy...
> >
> >
> >
> > On Sat, Apr 5, 2008 at 1:09 PM, Scott Monasmith <[EMAIL PROTECTED]>
> wrote:
> >
> >
> > They told me the reason why they don't hold the voice exams is because
> they
> > would take up a lot more storage (hard drives/images of 3 servers,
> router
> > configs, etc.) vs. just backing up a few config files from a
> router/switch
> > to a text file for the R&S lab.
> >
> > Sorry.
> >
> >
> >
> >
> >
> > On 4/5/08, Jonathan Charles <[EMAIL PROTECTED]> wrote:
> > What?!?!?
> >
> >
> > Jonathan
> >
> > On Sat, Apr 5, 2008 at 1:07 PM, Scott Monasmith <[EMAIL PROTECTED]>
> wrote:
> > > Correct. BUT not available for voice though, Jonathon.
> > >
> > >
> > >
> > >
> > > On 4/5/08, Jonathan Charles <[EMAIL PROTECTED]> wrote:
> > > > Yeah, as I thought:
> > > >
> > > > "Reevaluation of Results
> > > >
> > > > If you are concerned your results are in error, you may request a
> > > > "reread" until 14 days after your lab date via an email to
> > > > [EMAIL PROTECTED] Each reread costs $250.00 USD and consists of a
> > > > proctor loading your configurations into a rack to recreate the test
> > > > and re-score the entire exam. This process may take up to three
> weeks
> > > > after receipt of payment"
> > > >
> > > >
> > > >
> > > > Jonathan
> > > >
> > > > On Sat, Apr 5, 2008 at 1:04 PM, Jonathan Charles <[EMAIL PROTECTED]>
> > > wrote:
> > > > > There is a rescore option... isn't there? There was for the R&S...
> > > > >
> > > > 

Re: [OSL | CCIE_Voice] MGCP and SRST

[Scott Monasmith]

Thanks, JD.

I'm painfully figuring this out. Do you happen to know why it doesn't work
when the interface is shut down?

On Tue, Apr 8, 2008 at 7:45 AM, Devildoc <[EMAIL PROTECTED]> wrote:

> Paul,
>
> First of all, it is not recommended that you use service mgcp under a pots
> dial-peer with a PRI connection.  If you did, the ports won't get released
> properly when falling back to MGCP.  You only need to configure service mgcp
> under a pots dial-peer for CAS or FXO/FXS connections.
>
> Second, to test for SRST on BR1 router, you can do what Jonathan
> suggested.  Use a ACL to block MGCP traffics or an easier way is to
> configure an ip route for the CCM server to a null interface with this
> command statement "ip route 10.1.200.21 255.255.255.255 null0".  Shutting
> down the WAN interface on the BR1 to simulate SRST won't work in this lab.
>
> JD
>
>
> > Date: Mon, 7 Apr 2008 18:50:59 -0500
> > From: [EMAIL PROTECTED]
> > To: [EMAIL PROTECTED]
> > CC: ccie_voice@onlinestudylist.com; [EMAIL PROTECTED]
> > Subject: Re: [OSL | CCIE_Voice] MGCP and SRST
>
> >
> > Once the phones go into fallback, the H323 dial peers will take over,
> > basically because the auto-generated MGCP pots dial peers have no
> > destination-pattern, so they are ignored...
> >
> > The reverse is also true; when the GW is under MGCP control
> > (registered to CCM); the H.323 config is ignored.
> >
> > Basically you need to set up an ACL to block SCCP (to force the phones
> > into fallback) and another ACL to block MGCP (2427 and 2428) so that
> > MGCP will unregister...
> >
> > However, if you are lazy (I am); I would just shut down the serial
> > interface to the WAN...
> >
> >
> >
> > Jonathan
> >
> > On Mon, Apr 7, 2008 at 6:47 PM, Paul and Bobs <[EMAIL PROTECTED]>
> wrote:
> > > When BR1 has been configured as MGCP router adn you configure the SRST
> and
> > > want to test, how do you get the dia-peer pots with the service
> MGCPAPP not
> > > to answer the incoming calls. DO you have to remove this config or is
> there
> > > some automatic way to get srst to use the pots dial-peer when it is on
> > > control.
> > >
> > > Paul
> > >
>
>
> --
> Going green? See the top 12 foods to eat 
> organic.
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] QoS on CM port

[Scott Monasmith]

Greg, you are correct. I mis-read the original question.

On Tue, Apr 8, 2008 at 5:20 AM, Pulos, Greg <[EMAIL PROTECTED]> wrote:

> The SRND states that allowing a TRUST from a server opens up the
> possibilities that a compromised server now has priority access, among other
> issues, as taken from the SRND below:
>
> Servers-Certain servers, within the data center or otherwise, might be
> capable of correctly marking their traffic on their NICs. In such cases, the
> network administrator can choose to trust such markings. However, enforcing
> such a trust boundary requires cooperation between network administrators
> and system or server administrators, an alliance that is often fragile, at
> best, and usually involves considerable finger pointing. Additionally,
> network administrators should bear in mind that the majority of DoS/ worm
> attacks target servers. Infected servers not only might spew profuse amounts
> of traffic onto the network, but, in such cases, they might do so with
> trusted markings. There's no hard-and-fast rule that will apply to every
> situation. Some administrators prefer to trust certain servers, like Cisco
> CallManagers, due to the large number of ports that may be in use to provide
> services rather than administer complex access lists.
>
> I'd rather do the work ourselves as engineers to do our best to alleviate
> the inherent problems that can (and surely will) arise with trusting a
> server.
>
> Of course as it states, some admins would just as soon trust the server as
> it is much easier for them than doing the work up front themselves. I don't
> go for easyI go for secure and assured and error on the side of
> caution/safety.
> (of course, if the lab tells you to trust the server, then you trust the
> server...that's the only time I will)
>
> greg
>
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] On Behalf Of Jonathan Charles
> Sent: Monday, April 07, 2008 4:34 PM
> To: Pulos, Greg
> Cc: ccie_voice@onlinestudylist.com; Sven Hansen (svhansen)
> Subject: Re: [OSL | CCIE_Voice] QoS on CM port
>
> I disagree... the QoS SRND said to trust the CCM and Unity for DSCP...
>
>
>
> Jonathan
>
> On Mon, Apr 7, 2008 at 6:52 AM, Pulos, Greg <[EMAIL PROTECTED]> wrote:
> >
> >
> >
> >
> > A general rule of thumb is to never trust a server, as a server can be
> > compromised by a virus, worm or other malware thereby allowing an
> infected
> > server to generate massive amounts of traffic that would always get
> priority
> > and possibly causing everything from heavy uplink congestion to outright
> > DoS.
> >
> >
> >
> > You should mark traffic from a server port respective to the type of
> > traffic, ingress, that the server puts onto the wire; then police that
> > traffic to DSCP 8 (scavenger), if not outright drop, when exceed-burst
> > thresholds are exceeded.
> >
> >
> >
> >
> >
> > greg
> >
> >  
> >
> >
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Sven Hansen
> > (svhansen)
> >  Sent: Monday, April 07, 2008 7:38 AM
> >  To: ccie_voice@onlinestudylist.com
> >  Subject: [OSL | CCIE_Voice] QoS on CM port
> >
> >
> >
> >
> >
> >
> > Hi all
> >
> >
> > What sort of QoS is recommended for the CallManager port? Trusted server
> as
> > per the SRND?
> >
> >
> > Thanks
> >
> >
> > Sven
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > Sven Hansen
> >  Systems Engineer
> >  Sales / Channels
> >
> >  [EMAIL PROTECTED]
> >  Phone :+27 (0) 11 267 1039
> >  Mobile :+27 (0) 83 616 6216
> >  Fax :+27 (0) 11 267 1100
> >
> >
> >
> >  Sub-Saharan Africa
> >  www.cisco.com/global/ZA/
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > This e-mail may contain confidential and privileged material for the
> sole
> > use of the intended recipient. Any review, use, distribution or
> disclosure
> > by others is strictly prohibited. If you are not the intended recipient
> (or
> > authorized to receive for the recipient), please contact the sender by
> reply
> > e-mail and delete all copies of this message.
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
>
>


-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] IPCC documentation

[Scott Monasmith]

Unless it redirects you to a .PDF file, it will not be accessible in the
lab.

On Mon, Apr 7, 2008 at 11:54 PM, Paul and Bobs <[EMAIL PROTECTED]>
wrote:

> Hi Guys
>
> Whats the deal with the IPCC doco on univercd. It redirects you to another
> part of cco. Is this going to be allowed in the exam.
>
> Paul
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] CallManager Annunciator

[Scott Monasmith]

You're welcome.

Test it out and verify the steps. It has been awhile since I have done this.



On Tue, Apr 8, 2008 at 8:14 AM, Devildoc <[EMAIL PROTECTED]> wrote:

> Awsome Scott.  I was going to do some research on this problem but you
> beat me to it.  Thanks for the solution.
>
> JD
>
>  --
> Date: Mon, 7 Apr 2008 19:28:33 -0500
> From: [EMAIL PROTECTED]
> To: [EMAIL PROTECTED]
> CC: [EMAIL PROTECTED]; [EMAIL PROTECTED];
> ccie_voice@onlinestudylist.com
>
> Subject: Re: [OSL | CCIE_Voice] CallManager Annunciator
>
> Yes you can change the annuciator message for Unallocated/Unassigned
> number.
>
> Create own Call Block Message
> Record Message in unity Call Handler, then in Recorded Voice, select copyt
> o file - save to unity desktop.
> Map drive to Call Manager, and copy file to Call Manager
> Copy sound file to C:\ProgramFiles\Cisco\MOH\DropMOHaudioSourceHere
> this will create 5 new files.
> Copy all 5 files to TFTPPath\English_United_States
> Rename the original 5 Ann-VCA.* files to Ann-VCA-old.* then the newly
> created files you copy change the name to match the original Ann-VCA files.
>  DO NOT MODIFY ANY XML FILES.
>
> Route Pattern-->reject call> with Unallocated Number should play your
> file.  You may have to restart the transcoding service on both Pub and Sub,
> or reboot the box to get your file to be picked up.
>
> Enjoy
>
>
> On Mon, Apr 7, 2008 at 6:47 PM, Jonathan Charles <[EMAIL PROTECTED]>
> wrote:
>
> Well, that begs the question, can you customize the annunciator message?
>
> I think it would be cool to have a 'Doh!' sound effect whenever
> someone misdialed...
>
>
> Jonathan
>
> On Mon, Apr 7, 2008 at 6:29 PM, Erick Bergquist <[EMAIL PROTECTED]>
> wrote:
> > When I did that on 4.1, it does whatever the carrier does for
> >  Unallocated/Unassigned number. In Chicagoland, it played a message
> >  (forgot offhand the wording).
> >
> >
> >
> >  On Mon, Apr 7, 2008 at 6:14 PM, Jonathan Charles <[EMAIL PROTECTED]>
> wrote:
> >  > Will this cause CCM to play a different annunciated messages? Or will
> >  >  it just return reorder tone?
> >  >
> >  >  Besides, since the DN doesn't exist anyway, unallocated/unassigned
> >  >  number is getting sent back to the PSTN anyway...
> >  >
> >  >
> >  >
> >  >  Jonathan
> >  >
> >  >
> >  >
> >  >  On Mon, Apr 7, 2008 at 5:50 PM, Edward French <
> [EMAIL PROTECTED]> wrote:
> >  >  >
> >  >  > How about creating a translation-pattern of , select "Block
> this
> >  >  > pattern" and choose "Unallocated number" ?
> >  >  >
> >  >  > Ed
> >  >  >
> >  >  >
> >  >  >
> >  >  > - Original Message 
> >  >  > From: Scott Monasmith <[EMAIL PROTECTED]>
> >  >  > To: Jonathan Charles <[EMAIL PROTECTED]>
> >  >  > Cc: ccievoice1 <[EMAIL PROTECTED]>; "
> ccie_voice@onlinestudylist.com"
> >  >  > 
> >  >  > Sent: Monday, April 7, 2008 5:53:11 PM
> >  >  > Subject: Re: [OSL | CCIE_Voice] CallManager Annunciator
> >  >  >
> >  >  >  Unity is technically not the annuciator :)
> >  >  >
> >  >  >
> >  >  > On Mon, Apr 7, 2008 at 3:35 PM, Jonathan Charles <
> [EMAIL PROTECTED]> wrote:
> >  >  >
> >  >  > > Sure.
> >  >  > >
> >  >  > > Have a Route Pattern that is catch-all () and less-specific
> and
> >  >  > > send that to Unity to say whatever message you want, and then
> transfer
> >  >  > > them to the operator...
> >  >  > >
> >  >  > > You have to realize what is happening though...
> >  >  > >
> >  >  > > On a PRI, if it gets an unallocated/unassigned number, it is
> going to
> >  >  > > return reorder tone to the caller... same thing with a T1/CAS...
> >  >  > >
> >  >  > > So, the question is, if you have not allocated a number yet and
> >  >  > > someone calls it, what do you want to happen?
> >  >  > >
> >  >  > >
> >  >  > >
> >  >  > > Jonathan
> >  >  > >
> >  >  > >
> >  >  > >
> >  >  > >
> >  >  > > On Mon, Apr 7, 2008 at 1:58 PM, ccievoice1 <[EMAIL PROTECTED]>
> wrote:
> >

Re: [OSL | CCIE_Voice] MGCP and SRST

[Scott Monasmith]

It did happen to me on the proctorlabs last week.

On Mon, Apr 7, 2008 at 7:43 PM, Jonathan Charles <[EMAIL PROTECTED]> wrote:

> On the labs? I haven't touched the labs yet... I am still in theory
> mode...
>
>
>
> Jonathan
>
> On Mon, Apr 7, 2008 at 7:36 PM, Scott Monasmith <[EMAIL PROTECTED]>
> wrote:
> > Jonathon,
> >
> > Have you ever ran into a problem where if you shut down the serial
> interface
> > (to HQ) the PRI doesn't come up, but if you shut the callmanager service
> on
> > both servers instead, then the PRI comes up fine?
> >
> >
> >
> > On Mon, Apr 7, 2008 at 7:28 PM, Jonathan Charles <[EMAIL PROTECTED]>
> wrote:
> >
> > > Well, if you don't do auto (which I usually do when I deploy at
> > > customers, then I shut it off...); then you have to make sure you have
> > > service mgcpapp on the dial-peers... which is amusing cuz, they aren't
> > > usually automatically created (sometimes yes, sometimes no).
> > >
> > >
> > > Jonathan
> > >
> > >
> > >
> > >
> > > On Mon, Apr 7, 2008 at 7:19 PM, Paul and Bobs <[EMAIL PROTECTED]>
> > wrote:
> > > > Yip I just shut down the serial interface aswell.
> > > > do you use the auto configuration of mgcp or manually configure
> this.
> > The
> > > > auto would be quicker but are they testing the knowledge of a full
> > > > configuration
> > > >
> > > >
> > > >
> > > >  On Tue, Apr 8, 2008 at 10:50 AM, Jonathan Charles <
> [EMAIL PROTECTED]>
> > > > wrote:
> > > > > Once the phones go into fallback, the H323 dial peers will take
> over,
> > > > > basically because the auto-generated MGCP pots dial peers have no
> > > > > destination-pattern, so they are ignored...
> > > > >
> > > > > The reverse is also true; when the GW is under MGCP control
> > > > > (registered to CCM); the H.323 config is ignored.
> > > > >
> > > > > Basically you need to set up an ACL to block SCCP (to force the
> phones
> > > > > into fallback) and another ACL to block MGCP (2427 and 2428) so
> that
> > > > > MGCP will unregister...
> > > > >
> > > > > However, if you are lazy (I am); I would just shut down the serial
> > > > > interface to the WAN...
> > > > >
> > > > >
> > > > >
> > > > > Jonathan
> > > > >
> > > > >
> > > > >
> > > > >
> > > > > On Mon, Apr 7, 2008 at 6:47 PM, Paul and Bobs <
> [EMAIL PROTECTED]>
> > > > wrote:
> > > > > > When BR1 has been configured as MGCP router adn you configure
> the
> > SRST
> > > > and
> > > > > > want to test, how do you get the dia-peer pots with the service
> > MGCPAPP
> > > > not
> > > > > > to answer the incoming calls. DO you have to remove this config
> or
> > is
> > > > there
> > > > > > some automatic way to get srst to use the pots dial-peer when it
> is
> > on
> > > > > > control.
> > > > > >
> > > > > > Paul
> > > > > >
> > > > >
> > > >
> > > >
> > >
> >
> >
> >
> > --
> > "There are only 10 types of people in the world: Those who understand
> > binary, and those who don't"
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] MGCP and SRST

[Scott Monasmith]

Jonathon,

Have you ever ran into a problem where if you shut down the serial interface
(to HQ) the PRI doesn't come up, but if you shut the callmanager service on
both servers instead, then the PRI comes up fine?

On Mon, Apr 7, 2008 at 7:28 PM, Jonathan Charles <[EMAIL PROTECTED]> wrote:

> Well, if you don't do auto (which I usually do when I deploy at
> customers, then I shut it off...); then you have to make sure you have
> service mgcpapp on the dial-peers... which is amusing cuz, they aren't
> usually automatically created (sometimes yes, sometimes no).
>
>
> Jonathan
>
> On Mon, Apr 7, 2008 at 7:19 PM, Paul and Bobs <[EMAIL PROTECTED]>
> wrote:
> > Yip I just shut down the serial interface aswell.
> > do you use the auto configuration of mgcp or manually configure this.
> The
> > auto would be quicker but are they testing the knowledge of a full
> > configuration
> >
> >
> >
> >  On Tue, Apr 8, 2008 at 10:50 AM, Jonathan Charles <[EMAIL PROTECTED]>
> > wrote:
> > > Once the phones go into fallback, the H323 dial peers will take over,
> > > basically because the auto-generated MGCP pots dial peers have no
> > > destination-pattern, so they are ignored...
> > >
> > > The reverse is also true; when the GW is under MGCP control
> > > (registered to CCM); the H.323 config is ignored.
> > >
> > > Basically you need to set up an ACL to block SCCP (to force the phones
> > > into fallback) and another ACL to block MGCP (2427 and 2428) so that
> > > MGCP will unregister...
> > >
> > > However, if you are lazy (I am); I would just shut down the serial
> > > interface to the WAN...
> > >
> > >
> > >
> > > Jonathan
> > >
> > >
> > >
> > >
> > > On Mon, Apr 7, 2008 at 6:47 PM, Paul and Bobs <[EMAIL PROTECTED]>
> > wrote:
> > > > When BR1 has been configured as MGCP router adn you configure the
> SRST
> > and
> > > > want to test, how do you get the dia-peer pots with the service
> MGCPAPP
> > not
> > > > to answer the incoming calls. DO you have to remove this config or
> is
> > there
> > > > some automatic way to get srst to use the pots dial-peer when it is
> on
> > > > control.
> > > >
> > > > Paul
> > > >
> > >
> >
> >
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] CallManager Annunciator

[Scott Monasmith]

Watch the 'debug isdn q931' output when you route the call via a translation
pattern to a route pattern (where it gets blocked)

On Mon, Apr 7, 2008 at 6:14 PM, Jonathan Charles <[EMAIL PROTECTED]> wrote:

> Will this cause CCM to play a different annunciated messages? Or will
> it just return reorder tone?
>
> Besides, since the DN doesn't exist anyway, unallocated/unassigned
> number is getting sent back to the PSTN anyway...
>
>
>
> Jonathan
>
> On Mon, Apr 7, 2008 at 5:50 PM, Edward French <[EMAIL PROTECTED]>
> wrote:
> >
> > How about creating a translation-pattern of , select "Block this
> > pattern" and choose "Unallocated number" ?
> >
> > Ed
> >
> >
> >
> > - Original Message 
> > From: Scott Monasmith <[EMAIL PROTECTED]>
> > To: Jonathan Charles <[EMAIL PROTECTED]>
> > Cc: ccievoice1 <[EMAIL PROTECTED]>; "ccie_voice@onlinestudylist.com"
> > 
> > Sent: Monday, April 7, 2008 5:53:11 PM
> > Subject: Re: [OSL | CCIE_Voice] CallManager Annunciator
> >
> >  Unity is technically not the annuciator :)
> >
> >
> > On Mon, Apr 7, 2008 at 3:35 PM, Jonathan Charles <[EMAIL PROTECTED]>
> wrote:
> >
> > > Sure.
> > >
> > > Have a Route Pattern that is catch-all () and less-specific and
> > > send that to Unity to say whatever message you want, and then transfer
> > > them to the operator...
> > >
> > > You have to realize what is happening though...
> > >
> > > On a PRI, if it gets an unallocated/unassigned number, it is going to
> > > return reorder tone to the caller... same thing with a T1/CAS...
> > >
> > > So, the question is, if you have not allocated a number yet and
> > > someone calls it, what do you want to happen?
> > >
> > >
> > >
> > > Jonathan
> > >
> > >
> > >
> > >
> > > On Mon, Apr 7, 2008 at 1:58 PM, ccievoice1 <[EMAIL PROTECTED]>
> wrote:
> > > > Hi all,
> > > >
> > > > When a CallManager's ip phone dialed a unallocated DN#, the
> annunciator
> > will
> > > > play the following prompt to the calling ip phone: "Your call cannot
> be
> > > > completed as dialed. Please consult your directory and call again or
> ask
> > > > your operator for assistance. This is a recording."
> > > >
> > > > I am just wondering, can I make the annunciator to play the similar
> > prompt
> > > > when the calling party is external PSTN?
> > > >
> > > > Thanks.
> > > >
> > >
> >
> >
> >
> > --
> > "There are only 10 types of people in the world: Those who understand
> > binary, and those who don't"
> >
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] CallManager Annunciator

[Scott Monasmith]

Yes you can change the annuciator message for Unallocated/Unassigned number.

Create own Call Block Message
Record Message in unity Call Handler, then in Recorded Voice, select copyt o
file - save to unity desktop.
Map drive to Call Manager, and copy file to Call Manager
Copy sound file to C:\ProgramFiles\Cisco\MOH\DropMOHaudioSourceHere
this will create 5 new files.
Copy all 5 files to TFTPPath\English_United_States
Rename the original 5 Ann-VCA.* files to Ann-VCA-old.* then the newly
created files you copy change the name to match the original Ann-VCA files.
 DO NOT MODIFY ANY XML FILES.

Route Pattern-->reject call> with Unallocated Number should play your file.
You may have to restart the transcoding service on both Pub and Sub, or
reboot the box to get your file to be picked up.

Enjoy


On Mon, Apr 7, 2008 at 6:47 PM, Jonathan Charles <[EMAIL PROTECTED]> wrote:

> Well, that begs the question, can you customize the annunciator message?
>
> I think it would be cool to have a 'Doh!' sound effect whenever
> someone misdialed...
>
>
> Jonathan
>
> On Mon, Apr 7, 2008 at 6:29 PM, Erick Bergquist <[EMAIL PROTECTED]>
> wrote:
> > When I did that on 4.1, it does whatever the carrier does for
> >  Unallocated/Unassigned number. In Chicagoland, it played a message
> >  (forgot offhand the wording).
> >
> >
> >
> >  On Mon, Apr 7, 2008 at 6:14 PM, Jonathan Charles <[EMAIL PROTECTED]>
> wrote:
> >  > Will this cause CCM to play a different annunciated messages? Or will
> >  >  it just return reorder tone?
> >  >
> >  >  Besides, since the DN doesn't exist anyway, unallocated/unassigned
> >  >  number is getting sent back to the PSTN anyway...
> >  >
> >  >
> >  >
> >  >  Jonathan
> >  >
> >  >
> >  >
> >  >  On Mon, Apr 7, 2008 at 5:50 PM, Edward French <
> [EMAIL PROTECTED]> wrote:
> >  >  >
> >  >  > How about creating a translation-pattern of , select "Block
> this
> >  >  > pattern" and choose "Unallocated number" ?
> >  >  >
> >  >  > Ed
> >  >  >
> >  >  >
> >  >  >
> >  >  > - Original Message 
> >  >  > From: Scott Monasmith <[EMAIL PROTECTED]>
> >  >  > To: Jonathan Charles <[EMAIL PROTECTED]>
> >  >  > Cc: ccievoice1 <[EMAIL PROTECTED]>; "
> ccie_voice@onlinestudylist.com"
> >  >  > 
> >  >  > Sent: Monday, April 7, 2008 5:53:11 PM
> >  >  > Subject: Re: [OSL | CCIE_Voice] CallManager Annunciator
> >  >  >
> >  >  >  Unity is technically not the annuciator :)
> >  >  >
> >  >  >
> >  >  > On Mon, Apr 7, 2008 at 3:35 PM, Jonathan Charles <
> [EMAIL PROTECTED]> wrote:
> >  >  >
> >  >  > > Sure.
> >  >  > >
> >  >  > > Have a Route Pattern that is catch-all () and less-specific
> and
> >  >  > > send that to Unity to say whatever message you want, and then
> transfer
> >  >  > > them to the operator...
> >  >  > >
> >  >  > > You have to realize what is happening though...
> >  >  > >
> >  >  > > On a PRI, if it gets an unallocated/unassigned number, it is
> going to
> >  >  > > return reorder tone to the caller... same thing with a T1/CAS...
> >  >  > >
> >  >  > > So, the question is, if you have not allocated a number yet and
> >  >  > > someone calls it, what do you want to happen?
> >  >  > >
> >  >  > >
> >  >  > >
> >  >  > > Jonathan
> >  >  > >
> >  >  > >
> >  >  > >
> >  >  > >
> >  >  > > On Mon, Apr 7, 2008 at 1:58 PM, ccievoice1 <[EMAIL PROTECTED]>
> wrote:
> >  >  > > > Hi all,
> >  >  > > >
> >  >  > > > When a CallManager's ip phone dialed a unallocated DN#, the
> annunciator
> >  >  > will
> >  >  > > > play the following prompt to the calling ip phone: "Your call
> cannot be
> >  >  > > > completed as dialed. Please consult your directory and call
> again or ask
> >  >  > > > your operator for assistance. This is a recording."
> >  >  > > >
> >  >  > > > I am just wondering, can I make the annunciator to play the
> similar
> >  >  > prompt
> >  >  > > > when the calling party is external PSTN?
> >  >  > > >
> >  >  > > > Thanks.
> >  >  > > >
> >  >  > >
> >  >  >
> >  >  >
> >  >  >
> >  >  > --
> >  >  > "There are only 10 types of people in the world: Those who
> understand
> >  >  > binary, and those who don't"
> >  >  >
> >  >
> >
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

[OSL | CCIE_Voice] CAT6500 QoS

[Scott Monasmith]

If you are going to mark ALL SCCP traffic to CS3 on the Cat6503 switch,
would this be your config...


set qos enable
set qos acl ip MARK-SCCP dscp 24 any range 2000 2002 any (from servers)
set qos acl ip MARK-SCCP dscp 24 any any range 2000 2002 (from phones)
commit qos acl MARK-SCCP
set qos acl map MARK-SCCP 
-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] First Failed Attempt at CCIE LAb

[Scott Monasmith]

Granted, the time I scored 100% without completing the section was not on my
last attempt but on a prior attempt.

On Mon, Apr 7, 2008 at 5:05 PM, Scott Monasmith <[EMAIL PROTECTED]> wrote:

> I know this isn't going to make anyone feel any better, but I did score
> 100% on a section that I didn't even finish???
>
> I submitted a critique stating my concern about the scoring since this
> happened. I scored 100% on CRS and I didn't even do the script.
>
> This isn't intended to get everyone fired up and angry, but more so to
> express our concern to Cisco and use situations like mine as evidence that
> the scoring is flawed. Again, based on my last attempt, I felt like I nailed
> it, but my scored report said I missed it by 4-6 points.
>
>   On Mon, Apr 7, 2008 at 4:37 PM, Jonathan Charles <[EMAIL PROTECTED]>
> wrote:
>
> > So, basically an 'abandon all hope, all ye who enter here...'
> >
> > We could boycott it... but that would only hurt us...
> >
> >
> > Jonathan
> >
> > On Mon, Apr 7, 2008 at 1:40 PM, Erik Goppel <[EMAIL PROTECTED]> wrote:
> > > The thing is, that the testscripts they run (as far as i`ve heard and
> > > experienced) only test some basic but major behaviour.
> > > And if the main parts are successfull, they will start grading the
> > other
> > > parts of your lab. (as a timesaver probably)
> > > thats why they probably say, that they don`t care how you do it, as
> > long as
> > > you can provide the result.
> > >
> > > Our problem is, that we can`t defend our grading. So if we are
> > absolutely
> > > sure, that from the 5 questions we`ve answered 4 correctly. but the
> > one
> > > major question was wrong, resulting in failing the testscript, which
> > > resulting in failing all 5 questions (as Cisco sees it).
> > > We well get a 0% score on the topic. where we could have an 80% score.
> > >
> > > So if you are for instance messing up something simple as DHCP, and
> > you are
> > > not getting any phones up. but you have everything else correct
> > configured.
> > > you will fail, simply because your built solution is not working in
> > total.
> > > If you manage to do DHCP, but you suck at QOS, you will pass because
> > the QOS
> > > (depending on points, and topology) will probably allow the test
> > scripts to
> > > pass.
> > >
> > >
> > > So i think we should just take it as it comes, and aim for the
> > majority of
> > > major parts to work.
> > > There`s nothing we can do about it.
> > >
> > > sorry
> > >
> > > Erik
> > >
> > >
> > >
> > >
> > >
> > >
> > > On Mon, Apr 7, 2008 at 6:13 PM, Jonathan Charles <[EMAIL PROTECTED]>
> > wrote:
> > >
> > > > I have heard both versions... one that it is not dependent, and one
> > > > that it is... no idea any more...
> > > >
> > > >
> > > >
> > > > Jonathan
> > > >
> > > >
> > > >
> > > >
> > > > On Mon, Apr 7, 2008 at 10:54 AM, Devildoc <[EMAIL PROTECTED]>
> > wrote:
> > > > >
> > > > >  So you are saying that all of the questions in a section are
> > > inter-related
> > > > > and depends on each other for full points?  So if i failed one
> > question
> > > > > that's critical for the test success of the other questions, and
> > even
> > > though
> > > > > i get all other questions correctly, i still failed because i
> > didn't get
> > > > > first question correctly.  Now that makes sense.  And i thought
> > they
> > > award
> > > > > points based on each correct question regardless of how you answer
> > other
> > > > > questions.
> > > > >
> > > > >  JD
> > > > >
> > > > >
> > > > >
> > > > >  
> > > > >  Date: Mon, 7 Apr 2008 17:35:57 +0200
> > > > > From: [EMAIL PROTECTED]
> > > > > To: ccie_voice@onlinestudylist.com
> > > > >
> > > > >
> > > > > Subject: Re: [OSL | CCIE_Voice] First Failed Attempt at CCIE LAb
> > > > >
> > > > >
> > > > > You can try, but you are probably wasting time.
> > > > > I`ve asked the same 

Re: [OSL | CCIE_Voice] First Failed Attempt at CCIE LAb

[Scott Monasmith]

; >
> > > > erik
> > > >
> > > >
> > > > On Mon, Apr 7, 2008 at 5:16 PM, Devildoc <[EMAIL PROTECTED]>
> > wrote:
> > > >
> > > >
> > > > Well, if that's the case, then i am horrified at my score.  I mean
> my
> > score
> > > > for the call routing section is 0%.  How is that even possible?  I
> knew
> > that
> > > > i didn't have time to complete the test, so i spent extra time on
> just
> > one
> > > > section to see how it is graded.  I verified that all calls were
> > successful
> > > > and all requirements were met.  Hell... there were many questions in
> > that
> > > > section, and i received zero point for that section?
> > > >
> > > > Does anyone know if there is a Cisco contact where one can inquire
> some
> > > > questions about the lab exam and its grading process?
> > > >
> > > >
> > > > JD
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >  
> > > >  Date: Sat, 5 Apr 2008 13:44:46 -0500
> > > > From: [EMAIL PROTECTED]
> > > > To: [EMAIL PROTECTED]
> > > > CC: ccie_voice@onlinestudylist.com; [EMAIL PROTECTED]
> > > >
> > > > Subject: Re: [OSL | CCIE_Voice] First Failed Attempt at CCIE LAb
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > Here is an excerpt from the report card.
> > > >
> > > > "Your CCIE Certification Lab exam was scored based on grading
> policies
> > that
> > > > are adhered to uniformly by our proctors worldwide. Marking was
> based on
> > > > whether the answer you provided works. Candidates are not required
> to
> > use a
> > > > set methodology in achieving a correct result. The imperative is
> that
> > the
> > > > solution provided produces the outcome requested. "
> > > >
> > > > I do believe that they actually grade you correctly but again there
> are
> > > > chances of proctors messing up as well.
> > > >
> > > > Like Scott mentioned, I would just focus on things I can control and
> > forget
> > > > the rest... better luck next time buddy...
> > > >
> > > >
> > > >
> > > > On Sat, Apr 5, 2008 at 1:09 PM, Scott Monasmith <[EMAIL PROTECTED]
> >
> > wrote:
> > > >
> > > >
> > > > They told me the reason why they don't hold the voice exams is
> because
> > they
> > > > would take up a lot more storage (hard drives/images of 3 servers,
> > router
> > > > configs, etc.) vs. just backing up a few config files from a
> > router/switch
> > > > to a text file for the R&S lab.
> > > >
> > > > Sorry.
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > On 4/5/08, Jonathan Charles <[EMAIL PROTECTED]> wrote:
> > > > What?!?!?
> > > >
> > > >
> > > > Jonathan
> > > >
> > > > On Sat, Apr 5, 2008 at 1:07 PM, Scott Monasmith <[EMAIL PROTECTED]
> >
> > wrote:
> > > > > Correct. BUT not available for voice though, Jonathon.
> > > > >
> > > > >
> > > > >
> > > > >
> > > > > On 4/5/08, Jonathan Charles <[EMAIL PROTECTED]> wrote:
> > > > > > Yeah, as I thought:
> > > > > >
> > > > > > "Reevaluation of Results
> > > > > >
> > > > > > If you are concerned your results are in error, you may request
> a
> > > > > > "reread" until 14 days after your lab date via an email to
> > > > > > [EMAIL PROTECTED] Each reread costs $250.00 USD and consists
> of a
> > > > > > proctor loading your configurations into a rack to recreate the
> test
> > > > > > and re-score the entire exam. This process may take up to three
> > weeks
> > > > > > after receipt of payment"
> > > > > >
> > > > > >
> > > > > >
> > > > > > Jonathan
> > > > > >
> > > > > > On Sat, Apr 5, 2008 at 1:04 PM, Jonathan Charles <
> [EMAIL PROTECTED]>
> > > &

Re: [OSL | CCIE_Voice] QoS on CM port

[Scott Monasmith]

You can trust the DSCP on the CAT6503 but you need to do it via an ACL on
non-Gigabit ethernet linecards...

set qos enable
set qos acl ip TRUST-DSCP trust-dscp any
commit qos acl TRUST-DSCP
set qos acl map TRUST-DSCP 4/1 (server port)

On Mon, Apr 7, 2008 at 3:33 PM, Jonathan Charles <[EMAIL PROTECTED]> wrote:

> I disagree... the QoS SRND said to trust the CCM and Unity for DSCP...
>
>
>
> Jonathan
>
> On Mon, Apr 7, 2008 at 6:52 AM, Pulos, Greg <[EMAIL PROTECTED]> wrote:
> >
> >
> >
> >
> > A general rule of thumb is to never trust a server, as a server can be
> > compromised by a virus, worm or other malware thereby allowing an
> infected
> > server to generate massive amounts of traffic that would always get
> priority
> > and possibly causing everything from heavy uplink congestion to outright
> > DoS.
> >
> >
> >
> > You should mark traffic from a server port respective to the type of
> > traffic, ingress, that the server puts onto the wire; then police that
> > traffic to DSCP 8 (scavenger), if not outright drop, when exceed-burst
> > thresholds are exceeded.
> >
> >
> >
> >
> >
> > greg
> >
> >  
> >
> >
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Sven Hansen
> > (svhansen)
> >  Sent: Monday, April 07, 2008 7:38 AM
> >  To: ccie_voice@onlinestudylist.com
> >  Subject: [OSL | CCIE_Voice] QoS on CM port
> >
> >
> >
> >
> >
> >
> > Hi all
> >
> >
> > What sort of QoS is recommended for the CallManager port? Trusted server
> as
> > per the SRND?
> >
> >
> > Thanks
> >
> >
> > Sven
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > Sven Hansen
> >  Systems Engineer
> >  Sales / Channels
> >
> >  [EMAIL PROTECTED]
> >  Phone :+27 (0) 11 267 1039
> >  Mobile :+27 (0) 83 616 6216
> >  Fax :+27 (0) 11 267 1100
> >
> >
> >
> >  Sub-Saharan Africa
> >  www.cisco.com/global/ZA/
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > This e-mail may contain confidential and privileged material for the
> sole
> > use of the intended recipient. Any review, use, distribution or
> disclosure
> > by others is strictly prohibited. If you are not the intended recipient
> (or
> > authorized to receive for the recipient), please contact the sender by
> reply
> > e-mail and delete all copies of this message.
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] CallManager Annunciator

[Scott Monasmith]

Unity is technically not the annuciator :)

On Mon, Apr 7, 2008 at 3:35 PM, Jonathan Charles <[EMAIL PROTECTED]> wrote:

> Sure.
>
> Have a Route Pattern that is catch-all () and less-specific and
> send that to Unity to say whatever message you want, and then transfer
> them to the operator...
>
> You have to realize what is happening though...
>
> On a PRI, if it gets an unallocated/unassigned number, it is going to
> return reorder tone to the caller... same thing with a T1/CAS...
>
> So, the question is, if you have not allocated a number yet and
> someone calls it, what do you want to happen?
>
>
>
> Jonathan
>
> On Mon, Apr 7, 2008 at 1:58 PM, ccievoice1 <[EMAIL PROTECTED]> wrote:
> > Hi all,
> >
> > When a CallManager's ip phone dialed a unallocated DN#, the annunciator
> will
> > play the following prompt to the calling ip phone: "Your call cannot be
> > completed as dialed. Please consult your directory and call again or ask
> > your operator for assistance. This is a recording."
> >
> > I am just wondering, can I make the annunciator to play the similar
> prompt
> > when the calling party is external PSTN?
> >
> > Thanks.
> >
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] CallManager Annunciator

[Scott Monasmith]

Try this...

1. Create a translation pattern in a partition that only the inbound gateway
can access.
2. Have the translation pattern change the called number to ''
3. Configure the CSS of the translation pattern to access the '' Route
Pattern.
4. Create a route pattern of '' in a partition that the translation
pattern can access
5. On the route pattern of '', configure it to 'Block pattern...' and
then select the option of why the pattern is blocked.

This way the PSTN should see the call as answered (by the translation
pattern), then it forwards it to the route pattern which should play the
annuciator since the route pattern () is blocked.

On Mon, Apr 7, 2008 at 3:35 PM, Jonathan Charles <[EMAIL PROTECTED]> wrote:

> Sure.
>
> Have a Route Pattern that is catch-all () and less-specific and
> send that to Unity to say whatever message you want, and then transfer
> them to the operator...
>
> You have to realize what is happening though...
>
> On a PRI, if it gets an unallocated/unassigned number, it is going to
> return reorder tone to the caller... same thing with a T1/CAS...
>
> So, the question is, if you have not allocated a number yet and
> someone calls it, what do you want to happen?
>
>
>
> Jonathan
>
> On Mon, Apr 7, 2008 at 1:58 PM, ccievoice1 <[EMAIL PROTECTED]> wrote:
> > Hi all,
> >
> > When a CallManager's ip phone dialed a unallocated DN#, the annunciator
> will
> > play the following prompt to the calling ip phone: "Your call cannot be
> > completed as dialed. Please consult your directory and call again or ask
> > your operator for assistance. This is a recording."
> >
> > I am just wondering, can I make the annunciator to play the similar
> prompt
> > when the calling party is external PSTN?
> >
> > Thanks.
> >
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] dial-plan bug in IOS and workaround

[Scott Monasmith]

Zone prefix will not work independently - it still needs a tech-prefiix or a
default technology-prefix. Follow the GK call routing flow in the CCM 4.1.3
SRND.

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/4x/42dialpl.html#wp1053212

On Mon, Apr 7, 2008 at 12:27 PM, Jonathan Charles <[EMAIL PROTECTED]> wrote:

> There is one other way...
>
> Zone prefix...
>
>
> Jonathan
>
> On Mon, Apr 7, 2008 at 12:00 PM, Devildoc <[EMAIL PROTECTED]> wrote:
> >
> >  Paul,
> >
> >  There are 3 options to route calls using a GK.  Option 1 is to use a
> > tech-prefix.  Option 2 is to use default technology-prefix. And option 3
> is
> > to use the registered E164 addresses.  Option 3 only works with H323
> gateway
> > as in the case of CME and not CCM.
> >
> >  If you want to register E164 addresses, then all you have to do is
> register
> > your CME to GK as H323 gateway.  However, if you want to route calls
> from
> > CCM to CUE, then i would not recommend using option 3 since CUE is a SIP
> > endpoint and it does not register its E164 address to a GK.  So if you
> have
> > all of your phone DNs registered to a GK and not CUE, then call routing
> > would be broken.
> >
> >  JD
> >
> >
> >
> >  
> >  Date: Sun, 6 Apr 2008 17:04:17 +1100
> > From: [EMAIL PROTECTED]
> > To: ccie_voice@onlinestudylist.com
> > Subject: [OSL | CCIE_Voice] dial-plan bug in IOS and workaround
> >
> >
> > Im struggling a bit with this diaplan in CME adn gatekeeper routing. I
> am
> > trying to get CUE working without using the dialplan configuration but
> also
> > have e164 calls route through the GK to the CME from CCM. I do not want
> to
> > use tech prefix or any other method on the gatekeeper. Is there any
> other
> > way on CME to have the phones register with e164 numbers into the GK
> > dialplan automatically or am I asking to much...
> >
> >
> > Paul
> >
> > 
> > Get in touch in an instant. Get Windows Live Messenger now.
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] Frame Relay LFI

[Scott Monasmith]

Edward,

What is the 'map-class frame-relay' class for?


On Sun, Apr 6, 2008 at 5:55 PM, Edward French <[EMAIL PROTECTED]>
wrote:

>  Paul,
>
> If you are talking about MLP, enable shapping on the main interface place
> the LLQ in the SHAPE policy-map and apply the SHAPE policy-map to the
> Virtual template
>
> policy-map LLQ
> class RTP
> pri perc 33
> class SIG
> band per 5
> class class-default
> fair-q
> !
> policy-map SHAPE
> class class-default
> shape ave 729600
> service-policy LLQ
> !
> map-class frame-relay BR2
> frame-relay cir 729600
> frame-relay bc 7296
> frame-relay be 0
> frame-relay mincir 729600
> !
> int s0/1/0:0.1
> no ip add
> no ip ospf mtu-ignore
> no frame-relay interface-dlci 102 point-to-point
> frame-relay interface-dlci 102 ppp virtual-temp 1
> !
> interface virtual-template 1
> bandwidth 768
> ip add 162.8.102.2 255.255.255.0
> ip ospf mtu-ignore
> ppp multilink
> ppp multilink fragment delay 10
> ppp multilink interleave
> service-policy out SHAPE
>
>
>
>
> - Original Message 
> From: Paul and Bobs <[EMAIL PROTECTED]>
> To: ccie_voice@onlinestudylist.com
> Sent: Saturday, April 5, 2008 10:52:57 PM
> Subject: [OSL | CCIE_Voice] Frame Relay LFI
>
> HI
>
> With FRTS (not the FRF12 way) is it best to apply the LLQ config to the
> map-class FRTS then apply the service-policy FRTS to the virtual interface
> or can you apply the service-policy FRTS to the virtual-template with the
> LLQ and then on the sub-interface serial 0/0.1 apply the service-policy LLQ.
>
> Hope I have explained myself clearly here.
>
>
> Paul
>
>


-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] First Failed Attempt at CCIE LAb

[Scott Monasmith]

They told me the reason why they don't hold the voice exams is because they
would take up a lot more storage (hard drives/images of 3 servers, router
configs, etc.) vs. just backing up a few config files from a router/switch
to a text file for the R&S lab.

Sorry.


On 4/5/08, Jonathan Charles <[EMAIL PROTECTED]> wrote:
>
> What?!?!?
>
>
> Jonathan
>
> On Sat, Apr 5, 2008 at 1:07 PM, Scott Monasmith <[EMAIL PROTECTED]>
> wrote:
> > Correct. BUT not available for voice though, Jonathon.
> >
> >
> >
> >
> > On 4/5/08, Jonathan Charles <[EMAIL PROTECTED]> wrote:
> > > Yeah, as I thought:
> > >
> > > "Reevaluation of Results
> > >
> > > If you are concerned your results are in error, you may request a
> > > "reread" until 14 days after your lab date via an email to
> > > [EMAIL PROTECTED] Each reread costs $250.00 USD and consists of a
> > > proctor loading your configurations into a rack to recreate the test
> > > and re-score the entire exam. This process may take up to three weeks
> > > after receipt of payment"
> > >
> > >
> > >
> > > Jonathan
> > >
> > > On Sat, Apr 5, 2008 at 1:04 PM, Jonathan Charles <[EMAIL PROTECTED]>
> > wrote:
> > > > There is a rescore option... isn't there? There was for the R&S...
> > > >
> > > >
> > > >  Jonathan
> > > >
> > > >
> > > >
> > > >  On Sat, Apr 5, 2008 at 12:57 PM, Scott Monasmith <
> [EMAIL PROTECTED]>
> > wrote:
> > > >  > Devildoc, I do feel your frustration. I finished my last attempt
> in a
> > little
> > > >  > over 5 hours and spent the next 3 hours verifying my work. I
> walked
> > out of
> > > >  > the exam feeling like I nailed it. However, based on my score
> report,
> > it
> > > >  > reflected a score of about 74-76 points. There were 3 sections
> where
> > the
> > > >  > score came out and I was left scratching my head thinking "how
> can
> > this be?"
> > > >  > - Talk about deflating. I had plenty of time to verify everthing
> and
> > I felt
> > > >  > very good about my chances. And to this day I still have no idea
> how
> > I could
> > > >  > have missed points on those sections.
> > > >  >
> > > >  > To me, there are 2 things we can do:
> > > >  > 1. study harder
> > > >  > 2. after each failed attempt, continue to stress to cisco (via
> the
> > critique
> > > >  > in your score report) that a re-score option needs to be
> established
> > for the
> > > >  > exam.
> > > >  >
> > > >  > If I'm spending $2,000 (exam + travel) for each attempt, the
> least
> > they can
> > > >  > do is reassure us that they are doing everything possible to
> ensure
> > that
> > > >  > there are no errors in the grading.
> > > >  >
> > > >  > BTW, a proctor told me that voice is the most challenging to
> grade
> > since
> > > >  > there is more than one way to achieve the desired results
> > > >  >
> > > >  >
> > > >  >
> > > >  >
> > > >  > On 4/5/08, Jonathan Charles <[EMAIL PROTECTED]> wrote:
> > > >  > > I ran into the same problem with the R&S lab... there might be
> 3 or
> > 4
> > > >  > > ways to do something, but only one of them gets you points...
> not
> > sure
> > > >  > > if this is the same thing on the CCIE Voice lab... but I would
> bet
> > it
> > > >  > > is.
> > > >  > >
> > > >  > > No idea on how they grade the test.
> > > >  > >
> > > >  > > But I think a lot has to do with how you use the proctor... so,
> if
> > > >  > > there are 2 ways to do something, that means to go to the
> proctor
> > and
> > > >  > > say, 'hey, I have way A and way B... which one is preferred?'
> > > >  > >
> > > >  > > Now, I would bet that one of those ways doesn't meet the
> > > >  > > requirements... which is why this test is as difficult as it
> is...
> > > >  > > because you are going to have to know why 'way B' doesn't
> work...
> > > >  >

Re: [OSL | CCIE_Voice] First Failed Attempt at CCIE LAb

[Scott Monasmith]

Correct. BUT not available for voice though, Jonathon.

On 4/5/08, Jonathan Charles <[EMAIL PROTECTED]> wrote:
>
> Yeah, as I thought:
>
> "Reevaluation of Results
>
> If you are concerned your results are in error, you may request a
> "reread" until 14 days after your lab date via an email to
> [EMAIL PROTECTED] Each reread costs $250.00 USD and consists of a
> proctor loading your configurations into a rack to recreate the test
> and re-score the entire exam. This process may take up to three weeks
> after receipt of payment"
>
>
>
> Jonathan
>
> On Sat, Apr 5, 2008 at 1:04 PM, Jonathan Charles <[EMAIL PROTECTED]>
> wrote:
> > There is a rescore option... isn't there? There was for the R&S...
> >
> >
> >  Jonathan
> >
> >
> >
> >  On Sat, Apr 5, 2008 at 12:57 PM, Scott Monasmith <[EMAIL PROTECTED]>
> wrote:
> >  > Devildoc, I do feel your frustration. I finished my last attempt in a
> little
> >  > over 5 hours and spent the next 3 hours verifying my work. I walked
> out of
> >  > the exam feeling like I nailed it. However, based on my score report,
> it
> >  > reflected a score of about 74-76 points. There were 3 sections where
> the
> >  > score came out and I was left scratching my head thinking "how can
> this be?"
> >  > - Talk about deflating. I had plenty of time to verify everthing and
> I felt
> >  > very good about my chances. And to this day I still have no idea how
> I could
> >  > have missed points on those sections.
> >  >
> >  > To me, there are 2 things we can do:
> >  > 1. study harder
> >  > 2. after each failed attempt, continue to stress to cisco (via the
> critique
> >  > in your score report) that a re-score option needs to be established
> for the
> >  > exam.
> >  >
> >  > If I'm spending $2,000 (exam + travel) for each attempt, the least
> they can
> >  > do is reassure us that they are doing everything possible to ensure
> that
> >  > there are no errors in the grading.
> >  >
> >  > BTW, a proctor told me that voice is the most challenging to grade
> since
> >  > there is more than one way to achieve the desired results
> >  >
> >  >
> >  >
> >  >
> >  > On 4/5/08, Jonathan Charles <[EMAIL PROTECTED]> wrote:
> >  > > I ran into the same problem with the R&S lab... there might be 3 or
> 4
> >  > > ways to do something, but only one of them gets you points... not
> sure
> >  > > if this is the same thing on the CCIE Voice lab... but I would bet
> it
> >  > > is.
> >  > >
> >  > > No idea on how they grade the test.
> >  > >
> >  > > But I think a lot has to do with how you use the proctor... so, if
> >  > > there are 2 ways to do something, that means to go to the proctor
> and
> >  > > say, 'hey, I have way A and way B... which one is preferred?'
> >  > >
> >  > > Now, I would bet that one of those ways doesn't meet the
> >  > > requirements... which is why this test is as difficult as it is...
> >  > > because you are going to have to know why 'way B' doesn't work...
> >  > > which means a deep understanding of not just how to configure
> >  > > something, but in the way it works at a protocol level.
> >  > >
> >  > > For example... let's say I wanted you to set up CAC for a specific
> >  > > location. Now, no big, right, just set up locations-based CAC... or
> >  > > use the GK... both work... both will provide CAC... but let's say I
> >  > > added to that, 'make sure that you can adjust bandwidth on the
> fly...'
> >  > > now, we know locations-based CAC can't do that, we are looking for
> >  > > BRQs and we have to use a GK and enable BRQs in CCM.
> >  > >
> >  > > The example is probably a bad one, but it is the only one I can
> think of.
> >  > >
> >  > > I do have a question tho... I have heard from people that the CCMs
> are
> >  > > slow and nearly unresponsive... so, it can take 2 or 3 minutes for
> a
> >  > > page to load. Is this true? Whee did you take it?
> >  > >
> >  > > Also, Mark Snow has an example script on the DVDs, that looks like
> it
> >  > > would take a few hrs to configure on the lab... even if you knew
> >  > > scripting... the requireme

Re: [OSL | CCIE_Voice] First Failed Attempt at CCIE LAb

[Scott Monasmith]

Devildoc, I do feel your frustration. I finished my last attempt in a little
over 5 hours and spent the next 3 hours verifying my work. I walked out of
the exam feeling like I nailed it. However, based on my score report, it
reflected a score of about 74-76 points. There were 3 sections where the
score came out and I was left scratching my head thinking "how can this be?"
- Talk about deflating. I had plenty of time to verify everthing and I felt
very good about my chances. And to this day I still have no idea how I could
have missed points on those sections.

To me, there are 2 things we can do:
1. study harder
2. after each failed attempt, continue to stress to cisco (via the critique
in your score report) that a re-score option needs to be established for the
exam.

If I'm spending $2,000 (exam + travel) for each attempt, the least they can
do is reassure us that they are doing everything possible to ensure that
there are no errors in the grading.

BTW, a proctor told me that voice is the most challenging to grade since
there is more than one way to achieve the desired results


On 4/5/08, Jonathan Charles <[EMAIL PROTECTED]> wrote:
>
> I ran into the same problem with the R&S lab... there might be 3 or 4
> ways to do something, but only one of them gets you points... not sure
> if this is the same thing on the CCIE Voice lab... but I would bet it
> is.
>
> No idea on how they grade the test.
>
> But I think a lot has to do with how you use the proctor... so, if
> there are 2 ways to do something, that means to go to the proctor and
> say, 'hey, I have way A and way B... which one is preferred?'
>
> Now, I would bet that one of those ways doesn't meet the
> requirements... which is why this test is as difficult as it is...
> because you are going to have to know why 'way B' doesn't work...
> which means a deep understanding of not just how to configure
> something, but in the way it works at a protocol level.
>
> For example... let's say I wanted you to set up CAC for a specific
> location. Now, no big, right, just set up locations-based CAC... or
> use the GK... both work... both will provide CAC... but let's say I
> added to that, 'make sure that you can adjust bandwidth on the fly...'
> now, we know locations-based CAC can't do that, we are looking for
> BRQs and we have to use a GK and enable BRQs in CCM.
>
> The example is probably a bad one, but it is the only one I can think of.
>
> I do have a question tho... I have heard from people that the CCMs are
> slow and nearly unresponsive... so, it can take 2 or 3 minutes for a
> page to load. Is this true? Whee did you take it?
>
> Also, Mark Snow has an example script on the DVDs, that looks like it
> would take a few hrs to configure on the lab... even if you knew
> scripting... the requirements seem straightforward, but then he adds
> extra steps to the script... that seem to be from IPCC Scripting Best
> Practices... and all for probably only 4 points on the test... it
> seems like it would be almost impossible to get any points from that
> scenario...
>
>
>
>
> Jonathan
>
>
>
>
>
> On Sat, Apr 5, 2008 at 10:45 AM, Devildoc <[EMAIL PROTECTED]> wrote:
> >
> >  Hello All,
> >
> >  I just had my first attempt at the Voice CCIE lab last week, and of
> course,
> > i failed.  I knew right away after the lab that i failed.  The reason
> why I
> > failed was not due to the lack of or inadequate amount of knowledge that
> I
> > possessed but rather the lack of time. I was so nervous and stressed out
> > that I tumbled clumsily throughout the day.
> >
> >  In my opinion, the lab was not tricky or even difficult.  I actually
> think
> > that i over-studied for the lab.  The Proctor Workbook and the Bootcamp
> well
> > prepared me for the lab, so there was no problem with the knowledge
> there.
> > Having said that, i was dumbfounded when i got my scores result.
> >
> >  And here are my questions.  Does anyone know how Cisco grade these
> labs?
> > Is Cisco looking for a specific way to implement a solution or any
> method to
> > implement a solution would work as long as it satifies all of the
> > requirements asked of you in the questions?
> >
> >  The reason why i am asking these questions is because even though i did
> not
> > complete the lab, I did complete some sections.  I tested those
> completed
> > sections and verified that all requirements were met, and still i
> received
> > 0% for those completed sections.  Shouldn't I have received some points
> for
> > those sections?  I know that each question is worth ALL or NO point for
> the
> > correct answer.  However, there are many questions in a section, and if
> I
> > completed a section with all requirements met, then i would think that
> at
> > least i would get 1 or 2 questions right if not all.  But i see no point
> > awarded at all for the completed sections, so that means that i must not
> > have gotten all questions in the section right to get 0%.  But how can
> that
> > be since I tested it 

Re: [OSL | CCIE_Voice] Priority/Bandwidth percent

[Scott Monasmith]

This is fine, Mark. No reason to dig it back up. Sorry I wasn't in the
original nausium.

I understand most of the IPT concepts regarding the IE Lab. It is just that
the devil is in the details.

On Thu, Apr 3, 2008 at 1:50 PM, Mark Snow <[EMAIL PROTECTED]> wrote:

> This has been addressed ad-nausium in a very recent past thread right here
>  let me try to dig it up and forward it again - but the gist of the
> answer is:
>
> Bandwidth or Priority cmd in CBWFQ section depends on CIR value IF Traffic
> Shaping is present on that interface
> Bandwidth or Priority cmd in CBWFQ section depends on Bandwidth cmd on
> Interface value IF there IS NO Traffic Shaping present on that interface
>
>
> HTH and be back with more ...
>
>
> --
> Mark Snow
> CCIE #14073 (Voice, Security)
> CCSI #31583
>
> Senior Technical Instructor - IPexpert, Inc.
> A Cisco Learning Partner - We Accept Learning Credits!
>
> Telephone: +1.810.326.1444
> Fax: +1.309.413.4097
> Mailto: [EMAIL PROTECTED]
> --
> Join our free online support and peer group communities:
> http://www.IPexpert.com/communities <http://www.ipexpert.com/communities>
> --
> IPexpert - The Global Leader in Self-Study, Classroom-Based,
> Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE
> R&S Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and
> CCIE Storage Lab Certifications.
> --
>
>
> On Apr 3, 2008, at 10:48 AM, Scott Monasmith wrote:
>
> Does the priority/bandwidth percent command, within LLQ, calculate the
> > percentage of bandwidth based on the link speed (768k) or the shaping speed
> > (768k * .95)?
> >
> > policy-map llq
> >  class voice
> >  priority percent 33
> >  class signal
> >  bandwidth percent 5
> >  class class-default
> >  fair-queue
> >
> > policy-map shape
> >  shape average 729600 7296 0
> >  service policy llq
> >
> > Would the priority bandwidth be 254k or 241k?
> >
> > --
> > "There are only 10 types of people in the world: Those who understand
> > binary, and those who don't"
> >
>
>


-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] Inbound policing as per QOS SRND

[Scott Monasmith]

64k is only the g711ulaw codec bandwidth. Below is the reason why...

The calculation would be as follows:

BW = ([L2 overhead + IP_UDP_RTP Overhead + Sample Size] / Sample_Size) *
Codec_Speed

BW = ([32+40+160]/ 160) * 64000
BW = 92.8k

For a better understanding, read page 1-15 of the QoS SRND 3.3

HTH,
Scott


On Mon, Mar 31, 2008 at 10:16 AM, Juan Lopez Hernandez -X (jlopezhe - IBM -
INS at Cisco) <[EMAIL PROTECTED]> wrote:

>  Small Q: why does the QOS SRND polices inbound voice bearer (p.2-37) to
> 128Kbps to limit a port to 1 call max - as it's inbound, and thus one-way? I
> would think of 64K instead.
>
> cheers,
> Juan
>
>
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] Priority/Bandwidth percent

[Scott Monasmith]

If I understood it correctly, the bandwidth command would prevent you from
configuring LLQ with values above the 75% default threshold. Also, the
bandwidht command is utilized when AutoQoS is configured. Are you telling me
that when you configure LLQ and FRTS/GTS that the priority/bandwidth percent
command generates the value from the 'bandwidth' command on the interface?

On Thu, Apr 3, 2008 at 10:12 AM, Jonathan Charles <[EMAIL PROTECTED]> wrote:

> No, it is based on the bandwidth command on the interface (if I
> understand Mark correctly...) and it is limited to 75% total usage of
> that configured bandwidth...
>
> However, this can be changed with a max-reserved-bandwidth 90
>
> The reason it is is limited so you don't choke out your routing
> protocols...
>
>
> Jonathan
>
> On Thu, Apr 3, 2008 at 9:48 AM, Scott Monasmith <[EMAIL PROTECTED]>
> wrote:
> > Does the priority/bandwidth percent command, within LLQ, calculate the
> > percentage of bandwidth based on the link speed (768k) or the shaping
> speed
> > (768k * .95)?
> >
> > policy-map llq
> >  class voice
> >   priority percent 33
> >  class signal
> >   bandwidth percent 5
> >  class class-default
> >   fair-queue
> >
> > policy-map shape
> >  shape average 729600 7296 0
> >  service policy llq
> >
> > Would the priority bandwidth be 254k or 241k?
> >
> > --
> > "There are only 10 types of people in the world: Those who understand
> > binary, and those who don't"
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

[OSL | CCIE_Voice] Priority/Bandwidth percent

[Scott Monasmith]

Does the priority/bandwidth percent command, within LLQ, calculate the
percentage of bandwidth based on the link speed (768k) or the shaping speed
(768k * .95)?

policy-map llq
 class voice
  priority percent 33
 class signal
  bandwidth percent 5
 class class-default
  fair-queue

policy-map shape
 shape average 729600 7296 0
 service policy llq

Would the priority bandwidth be 254k or 241k?

-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] Shaping for FRF.12

[Scott Monasmith]

Understood.

Next question is - Do you have to configure a form of FRTS with FRF.12? Can
you just configure the following...

map-class frame-relay frf12
 frame-relay fragment 960




On Thu, Apr 3, 2008 at 9:29 AM, Mark Snow <[EMAIL PROTECTED]> wrote:

> Exactly correct - both methods described below configure LFI (strictly
> speaking about LFI here) in the exact same way. 1st method combines FRF.12
> LFI with GTS (Generic Traffic Shaping).
> 2nd method combines FRF.12 LFI with FRTS proper.
>
> So - I will speak to some caveats - and then let you be the judge which to
> use (but which to use for the CCIE V Lab really comes down to what it always
> comes down to which is "how was the task worded?").
> If you use FRTS - then you MUST use the command "frame-relay
> traffic-shape" on the PHYSICAL interface on which the Sub ints live. This
> does 2 things - allows the FRTS that you have applied to any given Sub
> interface to actually work (hint: always do from enable mode a 'sh
> traffic-shape' to ensure that it is on and working)
> But from a possibly negative perspective - it ALSO AUTOMATICALLY SHAPES
> EVERY OTHER SUB-INTERFACE TO 56K!!! - so watch out - if you apply FRTS to
> one sub interface - you MUST apply some variation to every other sub
> interface - lest they all become 56K CIR links!! :)
>
> With GTS and FRF.12 LFI - you don't have to use the Phy interface cmd
> 'frame-relay traffic-shape' and thus don't have to worry about the above
> caveat.
>
>
> Cheers,
>
>
>   --
> Mark Snow
> CCIE #14073 (Voice, Security)
> CCSI #31583
>
> Senior Technical Instructor - IPexpert, Inc.
> A Cisco Learning Partner - We Accept Learning Credits!
>
> Telephone: +1.810.326.1444
> Fax: +1.309.413.4097
> Mailto: [EMAIL PROTECTED]
> --
> Join our free online support and peer group communities:
> http://www.IPexpert.com/communities <http://www.ipexpert.com/communities>
> --
> IPexpert - The Global Leader in Self-Study, Classroom-Based,
> Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE
> R&S Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and
> CCIE Storage Lab Certifications.
> --
>
> On Apr 2, 2008, at 9:37 PM, Scott Monasmith wrote:
>
> ??? - I don't understand. You can configure LFI either can't you?
>
> What is the difference below?
>
> 1)
> policy-map shape
>  class class-default
>   shape average 729600 7296 0
>
> map-class frame-relay frf12
>  frame-relay fragment 960
>  service-policy out shape
>
> 2)
> map-class frame-relay frf12
>  frame-relay fragment 960
>  frame-relay cir 729600
>  frame-relay bc 7296
>  frame-relay be 0
>  frame-relay mincir 729600
>
> On Wed, Apr 2, 2008 at 8:31 PM, jason sung <[EMAIL PROTECTED]> wrote:
>
> > One reason you might want to use FRF.12 over legacy FRTS is because you
> > can fragment the packets.
> >
> >
> >
> > On Wed, Apr 2, 2008 at 8:28 PM, Scott Monasmith <[EMAIL PROTECTED]>
> > wrote:
> >
> > > Does it matter if you use FRTS or legacy FRTS with FRF.12? Is there
> > > really any difference? Why would you use one over the other?
> > >
> > > --
> > > "There are only 10 types of people in the world: Those who understand
> > > binary, and those who don't"
> >
> >
> >
>
>
> --
> "There are only 10 types of people in the world: Those who understand
> binary, and those who don't"
>
>
>


-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] Lab restrictions

[Scott Monasmith]

Actually, there are a few 'minor' commands you are not allowed to
change/configure and they are listed on the first page of the exam.
Otherwise, the lab questions will let you know whether or not you can do
certain things.

Read the entire exam carefully!

On Wed, Apr 2, 2008 at 12:25 AM, Jonathan Charles <[EMAIL PROTECTED]> wrote:

> Actually, Jason answered my questions.
>
> On the R&S lab they were very specific, from long before I took the
> lab that you cannot use static routes, you cannot change the ospf
> interface type, etc... there was a public list of commands you would
> not be allowed to use on the lab.
>
> There is no such list for voice, which is the answer I was looking for...
>
>
>
> Jonathan
>
> On Tue, Apr 1, 2008 at 9:31 PM, Jacob Owen <[EMAIL PROTECTED]> wrote:
> > Jonathan,
> >  That is one of the big challenges of any CCIE lab, you
> >  will need to know how to do 1 thing 2-3 different ways
> >  as you won't know what you'll be allowed to do and not
> >  in the real thing.  Remember, it isn't the hardest
> >  test around because it's easy, and when you pass
> >  you'll feel so much better knowing it was hard.  Wish
> >  I had a better answer, but I think Jason was right on
> >  with his response.
> >
> >
> >
> >  --- Jonathan Charles <[EMAIL PROTECTED]> wrote:
> >
> >  > Right
> >  >
> >  > But I am currently studying different ways of doing
> >  > things... if they
> >  > aren't going to let me do things, I would like to
> >  > know...
> >  >
> >  >
> >  >
> >  > Jonathan
> >  >
> >  > On Tue, Apr 1, 2008 at 9:02 PM, jason sung
> >  > <[EMAIL PROTECTED]> wrote:
> >  > > Jonathan,
> >  > >
> >  > > From my one experience, I can say that there are
> >  > no restrictions
> >  > > universally, if there are any than it will be
> >  > noted in the question.
> >  > >
> >  > >
> >  > >
> >  > > On Tue, Apr 1, 2008 at 8:24 PM, Jonathan Charles
> >  > <[EMAIL PROTECTED]> wrote:
> >  > >
> >  > > > On the R&S lab you can't use static routes, etc.
> >  > > >
> >  > > > What restrictions are on the CCIE Voice lab? Can
> >  > you not use the web
> >  > > > interface to configure CallManager?
> >  > > >
> >  > > > No, seriously.
> >  > > >
> >  > > > Can you use ccm-manager config?
> >  > > >
> >  > > > That kinda stuff?
> >  > > >
> >  > > >
> >  > > >
> >  > > >
> >  > > > Jonathan
> >  > > >
> >  > >
> >  > >
> >  >
> >
> >
> >  Jacob Owen
> >  CCIE #14063 (R&S, Service Provider), CCVP, CCDP
> >
> >
> >
> >
> 
> >  You rock. That's why Blockbuster's offering you one month of
> Blockbuster Total Access, No Cost.
> >  http://tc.deals.yahoo.com/tc/blockbuster/text5.com
> >
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] Shaping for FRF.12

[Scott Monasmith]

??? - I don't understand. You can configure LFI either can't you?

What is the difference below?

1)
policy-map shape
 class class-default
  shape average 729600 7296 0

map-class frame-relay frf12
 frame-relay fragment 960
 service-policy out shape

2)
map-class frame-relay frf12
 frame-relay fragment 960
 frame-relay cir 729600
 frame-relay bc 7296
 frame-relay be 0
 frame-relay mincir 729600

On Wed, Apr 2, 2008 at 8:31 PM, jason sung <[EMAIL PROTECTED]> wrote:

> One reason you might want to use FRF.12 over legacy FRTS is because you
> can fragment the packets.
>
>
>
> On Wed, Apr 2, 2008 at 8:28 PM, Scott Monasmith <[EMAIL PROTECTED]>
> wrote:
>
> > Does it matter if you use FRTS or legacy FRTS with FRF.12? Is there
> > really any difference? Why would you use one over the other?
> >
> > --
> > "There are only 10 types of people in the world: Those who understand
> > binary, and those who don't"
>
>
>


-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

[OSL | CCIE_Voice] Shaping for FRF.12

[Scott Monasmith]

Does it matter if you use FRTS or legacy FRTS with FRF.12? Is there really
any difference? Why would you use one over the other?

-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

[OSL | CCIE_Voice] SRST Fallback

[Scott Monasmith]

What would cause a fractional PRI to fail to come up during SRST failover?
The original config is using a MGCP gateway. During the normal operation,
the MGCP gateway works just fine. However, during SRST, the PRI never comes
up. I keep seeing TEI_ASSIGNED.

isdn switch-type primary-ni

controller t1 0/0/0
 pri-group timeslots 1-3 service mgcp

int serial 0/0/0:23
 isdn bind-l3 ccm-manager

mgcp call-agent 10.1.200.22
mgcp

ccm-manager mgcp
ccm-manager fallback-mgcp
ccm-manager music-on-hold
ccm-manager switch immedate
ccm-manager redundant 10.1.200.21

application
 global
  service alternate default

call-manager-fallback
 max-ephone 10
 max-dn 20
 source ip-address 172.2.100.1

Re: [OSL | CCIE_Voice] QoS SRND in lab

[Scott Monasmith]

Jonathan, the depth and breadth of IP Telephony is deep and wide. I find it
virtually impossible to memorize every facet of IPT or have ALL the answers.
Therefore, if you see Cat6500's running CatOS in the field as much as I do
(which is never), then why concern yourself with memorizing it? Read it.
Understand it (as much as possible) for the lab. Then refer to the SRND if
you ever run into it in the field or the lab. The CCIE lab is a tough exam.
However, in my opinion, being a CCIE is more about how resourceful you can
be. In the end, the CCIE lab is a great accomplishment, but it only
scratches the surface of IP Telephony.

In any event, don't get discouraged, you'll figure it out.


On Thu, Mar 27, 2008 at 1:18 PM, Jonathan Charles <[EMAIL PROTECTED]> wrote:

> I am going to proceed from the idea that I need to read the QoS SRND
> about six times and memorize the queue layout of the 6500... it seems
> to be a bit of a bear... Mark is right; at the end of the day, if your
> only ability is to copy and paste from Cisco's website, you are
> probably going to be a piss-poor engineer.
>
> That being said, I have a ways to go here; the QoS for the 6500 is
> completely alien to me I barely understand the queue layout
> (1p2q2t is a bit of a mystery...); but it is just another hurdle...
>
>
>
> Jonathan
> This too shall pass
>
> On Thu, Mar 27, 2008 at 1:01 PM, Jane Ryer (jryer) <[EMAIL PROTECTED]>
> wrote:
> > Hello, Mark,
> >
> >  I'll forward below the previous post that describes the ACL that we're
> >  concerned about.  I can't find it in the QoS SRND v3.3.  That version
> of
> >  the SRND instead lists the first line (tcp any range 2000 2002 any) and
> >  then makes a comment about possibly needing other TCP and UDP ports.
> >
> >  Unless I can find this in the documentation that is available in the
> >  lab, I intend to memorize the list of TCP and UDP ports before my next
> >  lab attempt.
> >
> >  Jane
> >
> >  
> >  From: [EMAIL PROTECTED]
> >  [mailto:[EMAIL PROTECTED] On Behalf Of Edward
> >  French
> >  Sent: Sunday, March 16, 2008 5:01 PM
> >  To: CCIE Maillist
> >  Subject: [OSL | CCIE_Voice] QOS 6500
> >
> >  Lab 25 task 46 says to
> >  Configure the Catalyst 6500 to mark all VOIP control traffic from the
> >  CallManager as AF31.
> >
> >  Which would be:
> >  set port qos 3/23 port-based
> >  set qos acl ip SERVER dscp 26 tcp any range 2000 2002 any
> >  set qos acl ip SERVER dscp 26 tcp any any range 11000 11999
> >  set qos acl ip SERVER dscp 26 tcp any any range 1024 4999
> >  set qos acl ip SERVER dscp 26 tcp any any range 1719 1720
> >  set qos acl ip SERVER dscp 26 udp any eq 2427 any
> >  set qos acl ip SERVER dscp 26 tcp any eq 2428 any
> >  commit qos acl SERVER
> >  set qos acl map SERVER 3/23
> >
> >
> >  And in the DVD and Audio it says the best thing to do is copy the QOS
> >  commands from the SRND. However the above commands are not in the SRND.
> >  The SRND say do the following
> >
> >
> >  set port qos 3/23 trust trust-dscp
> >
> >  or on a 2Q2T port
> >
> >  set qos acl ip TRUST-DSCP trust-dscp any
> >  commit qos acl TRUST-DSCP
> >  set qos acl map TRUST-DSCP 3/23
> >
> >  My question is do the commands from the SRND meet the requirements of
> >  the task? And if not is there an easier way than the first solution,
> >  since this ACL is not in the SRND? Also given that the first solution
> is
> >  an older solution is this what I should expect on the lab?
> >
> >  Basically what is the easiest way to find and copy the QOS statements
> >  for the 6500? I do not have access to one outside of the lab so I would
> >  appreciate any tricks or tips rather than memorization.
> >
> >  Thanks
> >
> >  Ed
> >
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] More DocCD changes!!!

[Scott Monasmith]

Scott,

The DocCD is a website with more info than our brains can withstand

www.cisco.com/univercd/



On Wed, Mar 19, 2008 at 10:30 AM, Scott Voll <[EMAIL PROTECTED]> wrote:

> how do you get a copy of the docCD?  I see the web site but if I wanted a
> copy to start messing around with, how do I get a coppy?
>
> TIA
>
> Scott
>
>   On Tue, Mar 18, 2008 at 8:03 AM, Scott Monasmith <[EMAIL PROTECTED]>
> wrote:
>
> > Those of you that plan on utilizing the DocCD during the lab exam as
> > part of your test taking strategy, I can not stress enough that you need to
> > continually monitor the status of the links that you plan on using for the
> > lab. The ENTIRE Contact Center portion of the DocCD has now been moved off
> > of www.cisco.com/univerCD. Granted, redirects from DocCD to a particular
> > page should work during the exam, but I don't believe this particular move
> > regarding Contact Center will work since you are not being redirected to a
> > particular .PDF or web page but rather the Cisco.com 
> > <http://cisco.com/>support home page.
> >
> > Cheers,
> > Scott
> >
> >
>


-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] More DocCD changes!!!

[Scott Monasmith]

Great info, Jane.

Thanks!

On Tue, Mar 18, 2008 at 6:01 PM, Jane Ryer (jryer) <[EMAIL PROTECTED]> wrote:

>  Hi, Scott,
>
>
>
> I took the lab in RTP last Tuesday (March 11th), and there was a PDF of
> the Cisco CAD Installation Guide (CAD 6.1 for IP Contact Center Express
> Edition Release 4.0) on the desktop, along with the QoS SRND and the Call
> Manager SRND.  They have presumably added this since the links are broken if
> you try to navigate to it from DocCD.
>
>
>
> But, I agree that part of lab preparation should be knowing how to get to
> navigate to various topics from the main level of
> http://www.cisco.com/univercd .There is no extra time during a lab
> attempt to be figuring it out.
>
>
>
> Jane Ryer
>
>
>  --
>
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *Scott Monasmith
> *Sent:* Tuesday, March 18, 2008 9:04 AM
> *To:* CCIE Maillist
> *Subject:* [OSL | CCIE_Voice] More DocCD changes!!!
>
>
>
> Those of you that plan on utilizing the DocCD during the lab exam as part
> of your test taking strategy, I can not stress enough that you need to
> continually monitor the status of the links that you plan on using for the
> lab. The ENTIRE Contact Center portion of the DocCD has now been moved off
> of www.cisco.com/univerCD. Granted, redirects from DocCD to a particular
> page should work during the exam, but I don't believe this particular move
> regarding Contact Center will work since you are not being redirected to a
> particular .PDF or web page but rather the Cisco.com 
> <http://cisco.com/>support home page.
>
>
>
> Cheers,
>
> Scott
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

[OSL | CCIE_Voice] More DocCD changes!!!

[Scott Monasmith]

Those of you that plan on utilizing the DocCD during the lab exam as part of
your test taking strategy, I can not stress enough that you need to
continually monitor the status of the links that you plan on using for the
lab. The ENTIRE Contact Center portion of the DocCD has now been moved off
of www.cisco.com/univerCD. Granted, redirects from DocCD to a particular
page should work during the exam, but I don't believe this particular move
regarding Contact Center will work since you are not being redirected to a
particular .PDF or web page but rather the Cisco.com support home page.

Cheers,
Scott

Re: [OSL | CCIE_Voice] Gatekeeper h225 trunks and making calls

[Scott Monasmith]

Paul,

Are you trying to route a call via the GK out the BR2 gateway?




On Mon, Mar 17, 2008 at 9:15 PM, Paul and Bobs <[EMAIL PROTECTED]>
wrote:

> Hi
>
> just working through the labs at the moment and am getting stuck on the
> gatekeepers section. I have setup the gatekeeper in IOS and in CCM with the
> h225 trunks and they are registered in the gateway. I have a route pattern
> pointing to the trunks in CCM and a a dial peer pots pointing to my E1
> interface. IS there something I am missing to make the call through the
> gatekeeper. Am I mistaken in thinking that you can route calls through a
> gatekeeper the same as you can through a h323 gateway.
>
> Any ideas would be great
>
> Thanks
>
> Paul
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] Lab running on VMWare

[Scott Monasmith]

The challenge I'm having, is figuring out how to get it on the VM. Do you
install it from scratch? If so, how do you get by the hardware check?

Or, do you perform a physical to virtual from an existing server? If so, how
do you get by the licensing verification on the Windows box?




On Mon, Mar 17, 2008 at 9:08 AM, Mark Cardwell <[EMAIL PROTECTED]>
wrote:

> Cisco does
> --
> Sent from Blackberry
> Mark C Cardwell
> Network Engineer
> Presidio Corp
> 571.225.0132
>
> - Original Message -
> From: [EMAIL PROTECTED] <
> [EMAIL PROTECTED]>
> To: CCIE Maillist 
> Sent: Mon Mar 17 10:04:35 2008
> Subject: [OSL | CCIE_Voice] Lab running on VMWare
>
> Is anyone running their lab equipment in VMWare?
>
>
>


-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

[OSL | CCIE_Voice] Lab running on VMWare

[Scott Monasmith]

Is anyone running their lab equipment in VMWare?

[OSL | CCIE_Voice] FRF and FRTS

[Scott Monasmith]

For frame-relay, is it necessary to configure a form of traffic-shaping
(either FRTS or class-based FRTS) when configuring FRF.12? If not, is the
bandwidth command required on the interface/sub-interface that the
FRF.12map-class is tied to?

[OSL | CCIE_Voice] ATA questions

[Scott Monasmith]

I have a couple questions regarding ATAs.

1. If an ATA is configured for DHCP and has and IP Addressed assigned to it
via DHCP. How do you get the ATA to release/renew its IP address?

2. Can you enable/disable SRST for ATAs on the ATA itself? I'm not referring
to a separate device pool (w/ srst reference disabled) for the ATA.

Re: [OSL | CCIE_Voice] IPCC Phone Agent Service URL - moved from Univercd ??

[Scott Monasmith]

This is a common misconception, my friend. It does redirect you outside of
UniverCD, however, on the redirect page it will prompt you to either click
on the redirected link or wait 10 seconds for your browser to do it
automatically.

If you wait 10 seconds, it will redirect you just fine. However, if you
click on the link instead it will fail. Trust me.

Cheers,
Scott

On Fri, Feb 29, 2008 at 3:25 PM, Mike Prestidge <[EMAIL PROTECTED]>
wrote:

> Hi Scott,
>
> the path below is what I had originally used to find the IP Phone Agent
> URL, however the fact that it now redirects to a link outside of
> cisco.com/univercd means that it is not reachable during the lab exam.
>
> Mike
>
> ________
>
> From: Scott Monasmith [mailto:[EMAIL PROTECTED]
> Sent: Sat 1/03/2008 3:56 a.m.
> To: Mike Prestidge
> Cc: ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] IPCC Phone Agent Service URL - moved from
> Univercd ??
>
>
> Here is the path for IP Phone agent URL
>
> - DocCD
> - Customer Contact Center
> - Cisco IPCC Express and IP IVR
> - Cisco Customer Response Solution 5.0(x)
>
> - English
> - Documentation for Cisco IP Agents
>
> - Cisco CAD Installation Guide CAD 6.4 for Unified
> -- This will redirect you to a .pdf file. Do a search for http:// inside
> this .PDF and you'll eventually find the URL. However, this document does
> not tell you what variables you need to setup (Pwd, Ext, ID) for the IP
> Phone service.
>
> here is the direct link.
>
> http://www.cisco.com/univercd/cc/td/doc/product/voice/sw_ap_to/apps_5_0/english/agents/cad64ig.pdf
>
> Cheers,
> Scott
>
>
> On Tue, Feb 26, 2008 at 4:22 PM, Mike Prestidge <
> [EMAIL PROTECTED]> wrote:
>
>
>It seems that the (annoying) people moving the documents off
> Univercd have now also moved the documents with the URL for IP Phone
> agents!!
>
>I used to be able to find this by browsing via the following:
>
>Univercd > Customer Contact Software > IPCC Express and IP IVR >
> CRS 5.0(x) > English > Documentation for Cisco IP Agents > Cisco CAD
> Installation Guide 6.4
>
>Now this documentation has also been moved to a link that is not
> available in the lab.  Does anyone know an alternative location to find the
> URL within Univercd?
>
>Mike
>
>This communication, including any attachments, is confidential. If
> you are not the intended recipient, you should not read it - please contact
> me immediately, destroy it, and do not copy or use any part of this
> communication or disclose anything about it. Thank you. Please note that
> this communication does not designate an information system for the purposes
> of the Electronic Transactions Act 2002.
>
>
>
>
>
>
> --
> "There are only 10 types of people in the world: Those who understand
> binary, and those who don't"
>
>


-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] IPCC Phone Agent Service URL - moved from Univercd ??

[Scott Monasmith]

Here is the path for IP Phone agent URL

- DocCD
- Customer Contact Center
- Cisco IPCC Express and IP IVR
- Cisco Customer Response Solution 5.0(x)
- English
- Documentation for Cisco IP Agents
- Cisco CAD Installation Guide CAD 6.4 for Unified
-- This will redirect you to a .pdf file. Do a search for http:// inside
this .PDF and you'll eventually find the URL. However, this document does
not tell you what variables you need to setup (Pwd, Ext, ID) for the IP
Phone service.

here is the direct link.
http://www.cisco.com/univercd/cc/td/doc/product/voice/sw_ap_to/apps_5_0/english/agents/cad64ig.pdf

Cheers,
Scott

On Tue, Feb 26, 2008 at 4:22 PM, Mike Prestidge <[EMAIL PROTECTED]>
wrote:

> It seems that the (annoying) people moving the documents off Univercd have
> now also moved the documents with the URL for IP Phone agents!!
>
> I used to be able to find this by browsing via the following:
>
> Univercd > Customer Contact Software > IPCC Express and IP IVR > CRS 5.0(x)
> > English > Documentation for Cisco IP Agents > Cisco CAD Installation Guide
> 6.4
>
> Now this documentation has also been moved to a link that is not available
> in the lab.  Does anyone know an alternative location to find the URL within
> Univercd?
>
> Mike
>
> This communication, including any attachments, is confidential. If you are
> not the intended recipient, you should not read it - please contact me
> immediately, destroy it, and do not copy or use any part of this
> communication or disclose anything about it. Thank you. Please note that
> this communication does not designate an information system for the purposes
> of the Electronic Transactions Act 2002.
>
>


-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] CME: BACD Issue

[Scott Monasmith]

Also, verify the syntax of your .tcl and .au files configured for the aa and
acd applications to make sure that dashes should be dashes and underscores
should be underscores. If the filenames are incorrect, you could this
behavior as well.

Cheers,
Scott

On Fri, Feb 29, 2008 at 6:43 AM, Kumar, Narinder <
[EMAIL PROTECTED]> wrote:

>  Hi,
> I have the below config but when I dial the pilot number 88373000, I hear
> a dead silence for 30 seconds than fast busy tone. Can anyone suggest what's
> wrong??
>
> dial-peer voice 20 pots
>  service aa
>  incoming called-number 88373000
>  direct-inward-dial
>  port 0/3/0:0
> !
>
> application
>  service queue flash:app-b-acd-2.1.2.2.tcl
>   param queue-len 20
>   param number-of-hunt-grps 1
>   param aa-hunt2 3020
>  !
>  service aa flash:app-b-acd-aa-2.1.2.2.tcl
>   paramspace english index 1
>   param number-of-hunt-grps 1
>   param handoff-string aa
>   param dial-by-extension-option 3
>   paramspace english language en
>   param max-time-vm-retry 2
>   param aa-pilot 88373000
>   param max-extension-length 4
>   paramspace english location flash:
>   param second-greeting-time 30
>   param welcome-prompt _bacd_welcome.au
>   param call-retry-timer 15
>   param voice-mail 5000
>   param max-time-call-retry 600
>   param service-name queue
>
> Regards
> NK
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] CCIE Voice Lab Passed on first attempt

[Scott Monasmith]

Good response, David.

I feel compelled to respond to the original email as well. I have yet to
pass the lab exam, but I know that IPExpert helped me tremendously with
bridging the gap in my VoIP knowledge and with my career in general. Even if
I never pass the lab, it will not define my career as an engineer. I've been
able to help my company and fellow engineers with the knowledge and
experience I've obtained with IPExpert.

I would like to say thanks to Vik and Mark for the countless hours they have
spent developing the curriculum, facilitating classes, and monitoring this
forum every night AND on the weekends. I think this alone says enough about
their support and committment.

Eventhough I didn't pass my lab on the first attempt has nothing to do with
IPExpert. However, when I do pass, it will because IPExpert helped give me
the necessary momentum.

Cheers,
Scott

On Feb 12, 2008 9:44 PM, David Blair <[EMAIL PROTECTED]> wrote:

> Disclaimer: I have known Wayne Lawson for about 10 years when he was still
> in the Marine Corp. I have also made friends with Scott Morris and Vik
> Malik. I know several other IPexpert employees including Matt Brooks.
> Finally, I have been in the IT business for over 21 years and I currently
> work for Cisco Systems.
>
> AS a previous CCIE you know how difficult the CCIE Lab is. From all the
> information I have been able to gather the CCIE Voice track is one of the
> tougher CCIE Labs to pass. The CCIE Voice Lab has one of the lowest passing
> rate of any CCIE Lab. Each person is different. I know some very talented
> engineers who have taken 2 or more tries to pass the CCIE Voice Lab. Some
> like yourself this was there second or third CCIE.
>
> I have a few questions:
>
> 1) Would you prefer IPexpert say that everyone can pass on the first
> attempt?
>
> 2) If everyone passed on the first attempt would it still be a worthwhile
> certification to get?
>
> 3) I personally have known only one person to pass the CCIE Voice Lab on
> his first attempt. The engineers I know who have passed the CCIE Voice Lab
> the average passing is 3 tries. I have one engineer who took 8 times to
> pass.
>
> I find your posting insulting and counter productive. When I took the
> IPexpert 5 day CCIE Voice bootcamp a year ago. I felt Vik set my
> expectations correctly saying anyone can pass the first time just the
> chances of that are quite low.
>
> I know I am far from perfect. I also know every CCIE lab preparation
> system has errors. I have shared with the IPexpert team when I have issues
> with their products. I know IPexpert wants your feedback whether it is bad
> or good. How can anyone improve if they do not know what the problem(s)
> is/are?
>
>
> David L. Blair
>
>
>  On 2/12/08, *jatinder khalsa* <[EMAIL PROTECTED]> wrote:
>
> Dear All,
>
> Today, I passed my ccie voice lab at brussels on my first attempt. Firstly
> I would like to thank all who helped me towards achieving this goal. Special
> thanks to Tarun pahuja, and the CCBootcamp team (Brad) who gave me personal
> support. Even though I did not buy any of their products they gave their
> personal contact details and helped me when I needed much help.
>
> I started this journey in early 2007, I cleared the written after reading
> all the recommended cisco press books. Then my company supported me in
> buying all the voice lab equipment which I setup in my office. They also
> approved for me to go on a bootcamp, and I choose to go to the IPexpert  for
> 12 days.
>
> I bought all IPexpert products including COD and spent  more then $10,000.
> However I found their bootcamp demoralizing and  disappointing .  I was
> discouraged and told that no one could pass  the lab in the first
> attempt.  I felt they were hiding things from me so as to prolong my studies
> so as to keep me lingering on.. Their support was terrible, their workbook
> had errors, little explanations and I found myself  spending hours being
> frustrated when things would not work due to ommisions or things not
> explained properly. As a customer I expect a teaching product to  be
> accurate and explain things so as to reinforce my technical abilities,  but
> soon I discovered that their strategy was to hide things, to make me it
> difficult instead  and to mentally discourage me so that I was guaranteed to
> fail on first attempt.
>
> However I kept positive and determined to pass this exam and made a
> challenge to pass it on my first attempt. I spent the next two months off
> work and studying 15-18 hours per day and working on my weaknesses. I
> consulted the cisco press books several times until I was comfortable with
> every technology and spent hours on-hands practising on all the
> technologies.
>
> From my experience the lab was not as hard as I thought it would be. I got
> back from Brussels today and saw the good news.. now its time for a break
> and some celebration.
>
> Thanks to you all…..
>
> J P Singh
>
> CCIE (R & S , Voice)
>
>
>
>   
> _

Re: [OSL | CCIE_Voice] Configuring QoS on the ATA

[Scott Monasmith]

Nevermind, I got it.

On Feb 11, 2008 4:00 PM, Scott Monasmith <[EMAIL PROTECTED]> wrote:

> Can anyone explain to me how to decipher the ToS paramenter on the ATA
> configuration page? I wan't to be able to set the COS and/or DSCP of the ATA
> traffic but I can't make sense of the following...
>
>  TOS
>
> Description
>
> This parameter allows you to configure Type of Service (ToS) bits by
> specifying the precedence and delay of audio and signaling IP packets, as
> follows:
>
> •Bits 0-7—These bits are for the ToS value for voice data packets.
>
> –Range: 0-255
>
> –Default: 184
>
> •Bits 8-15—These bits are for the ToS value for signaling-data packets
>
> –Range: 0-255
>
> –Default: 168
>
> •Bits 16-31—Reserved.
>
> Value Type
>
> Bitmap
>
> Default
>
> 0x68B8
>
> Voice Configuration Menu Access Code
>
> 255
>
> --
> "There are only 10 types of people in the world: Those who understand
> binary, and those who don't"
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

[OSL | CCIE_Voice] Configuring QoS on the ATA

[Scott Monasmith]

Can anyone explain to me how to decipher the ToS paramenter on the ATA
configuration page? I wan't to be able to set the COS and/or DSCP of the ATA
traffic but I can't make sense of the following...

 TOS

Description

This parameter allows you to configure Type of Service (ToS) bits by
specifying the precedence and delay of audio and signaling IP packets, as
follows:

•Bits 0-7—These bits are for the ToS value for voice data packets.

–Range: 0-255

–Default: 184

•Bits 8-15—These bits are for the ToS value for signaling-data packets

–Range: 0-255

–Default: 168

•Bits 16-31—Reserved.

Value Type

Bitmap

Default

0x68B8

Voice Configuration Menu Access Code

255

-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

[OSL | CCIE_Voice] Nesting Map-class(es)

[Scott Monasmith]

Whenever I try to add the following command I receive the following error:

P20-BR2-RTR(config-subif)#frame-relay interface-dlci 102
P20-BR2-RTR(config-fr-dlci)#class FRF
Policy FRTS does not exist

**Can you nest map-class(es)?

class-map voice
 match ip dscp ef
class-map match-any signal
 match ip dscp 24
 match ip dscp 26

policy-map LLQ
 class voice
  pri 108
 class signal
  band 32
 class class-default
  fair-queue

map-class frame-relay FRTS
 frame-relay cir 729600
 frame-relay bc 7296
 frame-relay be 0
 frame-relay mincir 729600
 service-policy out LLQ

map-class frame-relay FRF
 frame-relay fragme 960
 service-policy out FRTS

-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] QOS Confirmation

[Scott Monasmith]

When it states that the destination port for SCCP on CCM is 2000 - it is
also saying that CCM is 'listening' on port 2000 for any SCCP traffic
destined to it.

Whenever a device is listening on a specific port for a certain protocol, it
also means that this port is the source port for communicating with
this same protocol from this device. So in the example ACL below, we are
trying to police all signaling traffic FROM callmanager:

CallManager listens on ports - 2000, 2001, 2002, 2427, 2428
Gatekeeper listens on port - 1718, 1720
1719 is the source in this example because Vik said so - :)

On Feb 10, 2008 8:55 AM, Devildoc <[EMAIL PROTECTED]> wrote:

> One thing i don't understand is that according to the Cisco documentation
> "CISCO UNIFIED CALLMANAGER 4.1 TCP AND UDP PORT USAGE", those ports are
> all destination ports. So why is it that i keep seeing some of these ports
> referenced as source ports in the ACL?  Can someone explain this to me.  Is
> it because the remote endpoints communicate using these destination ports to
> the CallManager server, and the CM server in turn reply to these remote
> endpoints using these ports as the source ports?   Here is the active link
> to the port documentation.
>
> http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/port/4_1/41plrev2.pdf
>
> JD
>
> --
> Date: Sat, 9 Feb 2008 13:50:21 -0800
> From: [EMAIL PROTECTED]
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] QOS Confirmation
>
>
> all,
>
>
>
>sadly in the last week they have migrated all of the Call manager docs
> off of univercd and including the nice ports doc.  from a prior post from
> vik
>
>
>  Couple of issues. For a full list of port numbers see link below. I've
> modifiedthe ACL with the SIP port #'s and removed the unused H323 port
> numbers. Also each line within the ACL will need the policer applied if you
> are policing all signaling traffic.
>
>
> http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/sec_vir/udp_tcp/41plrev2.pdf
>
>
> set qos policed-dscp-map 24,26:10
> set qos policer aggregate POLICE-CCM rate 32 burst 8000 policed-dscp
> set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM  tcp any range 2000
> 2002 any
> set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM  tcp any eq 2428
> any
> set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM  udp any eq 2427
> any
> set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM  tcp any any eq
> 1720
> set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM  tcp any any eq
> 1718
> set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM  udp any eq 1719any
> set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM  tcp any any eq
> 5060
> set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM  udp any any eq
> 5060
> commit qos acl CCM-SIGNAL
> set qos acl map CCM-SIGNAL 3/3
>
>
>
> I was wondering i think that 1719 is actually a destination port not a
> source port and it is wrong in this?  Can anyone confirm or deny?
>
> Thanks,
>
> Chad
>
>
> --
> Helping your favorite cause is as easy as instant messaging. You IM, we
> give. Learn 
> more.
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] QOS Confirmation

[Scott Monasmith]

Chad,

I actually downloaded a copy of this PDF many moons ago. I'll send you a
copy.

If anyone else would like a copy, just email me directly.

Cheers,
Scott



On Feb 9, 2008 3:50 PM, Chad Stachowicz <[EMAIL PROTECTED]> wrote:

> all,
>
>
>
>sadly in the last week they have migrated all of the Call manager docs
> off of univercd and including the nice ports doc.  from a prior post from
> vik
>
>
>  Couple of issues. For a full list of port numbers see link below. I've
> modifiedthe ACL with the SIP port #'s and removed the unused H323 port
> numbers. Also each line within the ACL will need the policer applied if you
> are policing all signaling traffic.
>
>
> http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/sec_vir/udp_tcp/41plrev2.pdf
>
>
> set qos policed-dscp-map 24,26:10
> set qos policer aggregate POLICE-CCM rate 32 burst 8000 policed-dscp
> set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM  tcp any range 2000
> 2002 any
> set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM  tcp any eq 2428
> any
> set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM  udp any eq 2427
> any
> set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM  tcp any any eq
> 1720
> set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM  tcp any any eq
> 1718
> set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM  udp any eq 1719any
> set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM  tcp any any eq
> 5060
> set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM  udp any any eq
> 5060
> commit qos acl CCM-SIGNAL
> set qos acl map CCM-SIGNAL 3/3
>
>
>
> I was wondering i think that 1719 is actually a destination port not a
> source port and it is wrong in this?  Can anyone confirm or deny?
>
> Thanks,
>
> Chad
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] Where to configure LLQ

[Scott Monasmith]

Thanks, Mark. Your answer actually seems logical to me. However, the QoS
SRND seems to say otherwise on page 3-43 (or page 221 on the PDF). Could you
verify?

Thanks always,
Scott

On Feb 6, 2008 8:03 PM, Mark Snow <[EMAIL PROTECTED]> wrote:

> under your 'policy-map SHAPE' statement.
>  Mark Snow
> CCIE #14073 (Voice, Security)
> CCSI #31583
> Senior Technical Instructor - IPexpert, Inc.
> A Cisco Learning Partner - We Accept Learning Credits!
> Telephone: +1.810.326.1444
> Fax: +1.309.413.4097
> Mailto: [EMAIL PROTECTED]
>
> IPexpert - The Global Leader in Self-Study, Classroom-Based, Video On
> Demand and Audio Certification Training Tools for the Cisco CCIE R&S Lab,
> CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE
> Storage Lab Certifications.
>
>
>  On Feb 6, 2008, at 3:47 PM, Scott Monasmith wrote:
>
> policy-map SHAPE
>
>
>


-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

[OSL | CCIE_Voice] Where to configure LLQ

[Scott Monasmith]

With the config below, where is the best place to put 'service-policy LLQ'?


class-map voice

 match ip dscp ef

class-map match-any signal

 match ip dscp 24

 match ip dscp 26



policy-map LLQ

 class voice

  priority percent 33

 class signal

  band percent 2

 class class-default

  fair-queue



policy-map SHAPE

 class class-default

  shape average 729600 3648 0

  shape adaptive 364800

  fr-voice-adapt

  --> service policy LLQ (Option 1)

-

int s0/0

 frame-relay traffic-shaping

 frame-relay fragmentation voice-adaptive

-

map-class frame-relay FRTS

 service-policy output SHAPE

-

int s0/0.1

 no ip address

 bandwidth 768

 frame-relay interface-dlci 100 ppp Virtual-Template10

  class FRTS



interface Virtual-Template10

 bandwidth 768

 ip add 127.0.0.1 255.255.255.0

 --> service-policy out LLQ (Option 2)

 ppp multilink

 ppp multilink fragment-delay 10

 ppp multilink interleave


-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] Documentation CD

[Scott Monasmith]

Case in point...

http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/sec_vir/udp_tcp/index.htm

All 4 links broken. Anyone know of another location within
www.cisco.com/univercd/ that shows TCP/UDP port utilization for IPT
protocols?

On Jan 25, 2008 3:16 PM, Scott Monasmith <[EMAIL PROTECTED]> wrote:

> The DocCD is still there HOWEVER several links are now broken as they said
> they are in the process of moving some stuff around. Believe me, it was a
> painful, and expensive, shock when I took my lab 2 weeks ago and a few items
> that I decided not to memorize (because I knew where to find them in the
> DocCD) were not there. I missed the lab by about 10-11 points.
>
> I'm a tad discouraged. Start navigating the DocCD now to make sure the
> documentation you may need during the lab is available. If not, start
> memorizing.
>
> Cheers,
>
>
>   On Jan 24, 2008 11:52 AM, Ahmed Nawar <[EMAIL PROTECTED]> wrote:
>
> >
> > Any body has taken the lab early this year on brussels. I just wanna
> > make
> > sure that we have an
> > access to the documentation CD or not as i heard it is not there any
> > more.
> >
> >
>
>
> --
> "There are only 10 types of people in the world: Those who understand
> binary, and those who don't"




-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

[OSL | CCIE_Voice] Blocking outbound callerID

[Scott Monasmith]

What is the best way to block outbound callerID for a particular pattern? I
have set both the Calling/Connected Line ID/Number Presentation on the
pattern to 'Restricted'. When I make an outbound call this is what I see...

Channel ID i = 0xA98383
Exclusive, Channel 3
Calling Party Number i = 0x0081, '1003'
Plan:Unknown, Type:Unknown
Called Party Number i = 0x80, '0113313203003'

On the destination gateway I see this...

Channel ID i = 0xA98381
Exclusive, Channel 1
Calling Party Number i = 0x0081, '1003'
Plan:Unknown, Type:Unknown
Called Party Number i = 0x80, '3313203003'

The originating phone display says 'Private'
The destination phone displays '1003'

What do I need to do to get the destination phone display to NOT show the
callerID? I have also sent this through a translation pattern before sending
it out the originating gateway but same result.

Cheers,
Scott

Re: [OSL | CCIE_Voice] Unity Telephony Integration Manager question...

[Scott Monasmith]

Verify that the prefix on UTIM is set to CiscoUM1-VI and not CiscoUM1-VI1

On Feb 3, 2008 10:29 PM, Alex Hannah <[EMAIL PROTECTED]> wrote:

>  So last night I was running the TIM on Unity.  When I got to the "Set
> number of voice messaging ports" screen I the wizard there is a textbox for
> Call Manager Device name prefix.
>
>
>
> No matter what I put into this box last night I could not get the verify
> button to complete successfully.  The steps I tried were as follows:
>
>
>
> 1.I verified what CM has set up as the name prefix under the VM
> Port Configuration Screen
>
> a.   CM had CiscoUM1-VI1 configured as it's prefix.
>
> 2.   I added CiscoUM1-VI to the lmhosts file on Unity b/c I read this
> is related to the NetBios settings.  ( My feeble attempt to resolve the
> issue )
>
>
>
> When the verification step failed I pressed on, and completed the TIM
> process, I was able to press the VM button on the phone and get into Unity
> after the services restarted.
>
>
>
> How do you verify and troubleshoot the Device Name prefix?  Has anyone run
> into this problem before?
>
>
>
> Thanks,
>
>
>
> Alex
>
>
>
>
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] MGCP PRI L2 issue

[Scott Monasmith]

Juan,

Remove dial-peer v 999 pots - You don't need this for the PRI.
Then do a 'no mgcp' then 'mgcp' - Maybe reset the gateway on CCM as well and
you should be good to go.

Cheers,
Scott

On Feb 2, 2008 3:30 PM, Juan Lopez Hernandez -X (jlopezhe - IBM - INS at
Cisco) <[EMAIL PROTECTED]> wrote:

>  Hi all,
>
> today, I swapped some HW in my lab, replacing the NM-HDV by
> NM-HDV2-1T1/E1. I had a working MGCP GW, connected via PRI to the PSTN. When
> using the NM-HDV2, I expercience a L2 issue (see debug below), causing all
> calls over that PRI to fail. I doubt that the HW is broken in this case, and
> it's a matter of a missing command... Does anybody have a clue or
> experienced the same - please let me know. I attached the config used on the
> GW - it's functionally not different from the working one when using the
> NM-HDV.
>
> ps: I have a PVDM2 sitting on the NM-HDV2, from which a part will be used
> as dspfarm ('voice-card 1')
>
> Kind regards,
> Juan
>
>
>
>
> isdn switch-type primary-ni
> voice-card 1
>  dspfarm
>  dsp services dspfarm
>
> controller T1 1/1
>  framing esf
>  linecode b8zs
>  cablelength short 133
>  pri-group timeslots 1-3,24 service mgcp
>
> interface Serial1/1:23
>  no ip address
>  isdn switch-type primary-ni
>  isdn incoming-voice voice
>  isdn bind-l3 ccm-manager
>  no cdp enable
>
> voice-port 1/1:23
>
> dial-peer voice 999 pots
>  application mgcpapp
>  incoming called-number .
>  port 1/1:23
>
> BR1-RTR#sh isdn status
> Global ISDN Switchtype = primary-ni
>
> %Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output may
> not apply
>
> ISDN Serial1/1:23 interface
> dsl 0, interface ISDN Switchtype = primary-ni
> L2 Protocol = Q.921 0x  L3 Protocol(s) = CCM MANAGER 0x0003
> Layer 1 Status:
> ACTIVE
> Layer 2 Status:
> TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
> Layer 3 Status:
> 0 Active Layer 3 Call(s)
>
> BR1-RTR#debug isdn q921
> debug isdn q921 is  ON.
> BR1-RTR#term mon
> BR1-RTR#
> Feb  2 21:21:20.940: ISDN Se1/1:23 Q921: S7_T203_EXPIRY: Lost
> communication with remote.
> Feb  2 21:21:20.940: ISDN Se1/1:23 Q921: User TX -> RRp sapi=0 tei=0 nr=0
> Feb  2 21:21:20.944: ISDN Se1/1:23 Q921: User RX <- RRp sapi=0 tei=0 nr=0
> Feb  2 21:21:20.944: ISDN Se1/1:23 Q921: User TX -> RRf sapi=0 tei=0 nr=0
> Feb  2 21:21:20.944: ISDN Se1/1:23 Q921: User RX <- RRf sapi=0 tei=0 nr=0
>
>
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] CCM to CME/CUE issues

[Scott Monasmith]

For Scenario 1: Transcoding codecs between inbound and outbound SIP calls is
not supported.
For Scenarios 2 & 3: What inbound and outbound DTMF types are you using?
On Feb 1, 2008 3:30 PM, Earnieball78 <[EMAIL PROTECTED]> wrote:

> Hello,
> I am refering to task 4.9 in the workbook. I have a few issues I'd like
> some help straighting out.
>
> Scenario 1: CCM --> H323 G711 to HQGW --> SIP G729 to CME
> Calling from CCM, calls works fine, can pick up CME phone and talk. I see
> transcoder on HQRTR being invoked so all is good. Problems start to pile up
> when trying to get to the CUE.
> When call is forwarded to VM, it get fast busy. When calling the VM pilot
> itself it never picks up. It doesn't seem to want to invoke the CME
> transcoder.
>
> Scenario 2: CCM --> H323 G711 to HQGW --> SIP G711 to CME
> Now calls go to VM just fine BUT my DTMF is not being recognized.
>
> Scenario 3: CCM --> H323 G711 to HQGW --> H323 G729 to CME
> Calls work. I can get to VM and I see both the HQ and CME transcoders
> being invoked.
> Problem is DTMF here again. Keep in mind that I change my DTMF settings on
> the dialpeers when going from SIP to H323.
>
> On a side note, when using a gatekeeper, CCM side being G729 to CME,
> transcoder is being invoked on the CME and calls to CUE work just fine,
> including DTMF.
>
> So, I'm not quite sure where to go from here. Any suggestions or pointers
> would be greatly appreciated.
>
> Cheers,
> Christian
>
> --
> Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it
> now.
>
>


-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] Cat6k QoS link

[Scott Monasmith]

Strong, here's how you get to it...

1. goto www.cisco.com/univercd
2. Under the 'Cisco IOS Software' drop-down - select 'Integrated Networking
Solutions'
3. Click on 'ESM Solution Reference Network Design Guides'
4. Click on 'Enterprise QoS Solution Reference Network Design Guide Version
3.3'
Voila.

For the Lab Exam, this PDF is on your desktop


On Feb 1, 2008 9:20 AM, ovais Iqbal <[EMAIL PROTECTED]> wrote:

>
>
>
> http://www.cisco.com/application/pdf/en/us/guest/netsol/ns432/c649/ccmigration_09186a008049b062.pdf
>
>
>   On 2/1/08, Strong Frog <[EMAIL PROTECTED]> wrote:
> >
> > Can anyone help me to find a nice QoS link for Cat6k on the UniverCD?
> >
> >
> > --
> > Smile, you'll save someone else's day!
> > Frog
>
>
>
>
> --
> Ovais Iqbal
> 416-294-7869




-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] Documentation CD

[Scott Monasmith]

The DocCD is still there HOWEVER several links are now broken as they said
they are in the process of moving some stuff around. Believe me, it was a
painful, and expensive, shock when I took my lab 2 weeks ago and a few items
that I decided not to memorize (because I knew where to find them in the
DocCD) were not there. I missed the lab by about 10-11 points.

I'm a tad discouraged. Start navigating the DocCD now to make sure the
documentation you may need during the lab is available. If not, start
memorizing.

Cheers,


On Jan 24, 2008 11:52 AM, Ahmed Nawar <[EMAIL PROTECTED]> wrote:

>
> Any body has taken the lab early this year on brussels. I just wanna make
> sure that we have an
> access to the documentation CD or not as i heard it is not there any more.
>
>


-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] Lost connection to host router

[Scott Monasmith]

Nevermind. I was able to resolve it.

On Jan 17, 2008 2:50 PM, Scott Monasmith <[EMAIL PROTECTED]> wrote:

> Has anyone ever logged into the CUE GUI and receive "Lost connection to
> host router" If so, how can I recover from this?
>

[OSL | CCIE_Voice] Lost connection to host router

[Scott Monasmith]

Has anyone ever logged into the CUE GUI and receive "Lost connection to host
router" If so, how can I recover from this?

[OSL | CCIE_Voice] Policing on ESW

[Scott Monasmith]

Is there a way to police traffic on an ESW (or ESM)  to where the the
exceeding traffic is marked down from say dscp 24 to dscp 10 like on a 3550?

Re: [OSL | CCIE_Voice] VM integration with SRST

[Scott Monasmith]

I'm sorry, it should be 'call-manager-fallback' not 'telephony-setup'

On Jan 9, 2008 11:04 AM, Scott Monasmith <[EMAIL PROTECTED]> wrote:

> For myself, VM-integration is the only way I have been able to get VM to
> work during SRST. You may be able to get creative with translation patterns
> as well.
>
> CONFIGURATION
> telephony-setup
>  voicemail 913335551234
>  call-forward busy 913335551234
>  call-forward noan 913335551234 timeout 12
>  timeout interdigit 7
>
> vm-integration
>  pattern direct # CGN (setup alternate extension in Unity that matches
> full 10-digit number)
>  pattern trunk-to-ext busy # FDN
>  pattern trunk-to-ext noan # FDN
>  pattern ext-to-ext busy # FDN
>  pattern ext-to-ext noan # FDN
>
> dial-peer voice 86 pots
>  destination-pattern 913335551234T (make sure you add the T so it will
> append the # FDN)
>  port 0/0/0:23
>  prefix 13335551234
>
> What this does during a busy condition on the phone (x2001) in the SRST
> location is send a call to the HQ gateway, via the PSTN, with a called
> number of 13335551234#6175212001.
>
> When the HQ gateway receives this call, the significant digits field is
> set to 4 for inbound calls on the HQ. Therefore, 13335551234#6175212001 is
> stripped down to 2001. When CallManager tries to contact the phone @ x2001
> and sees that it is not registered (because of the SRST condition) it
> forwards the call based on the Busy condition of x2001 (which should be
> voicemail).
>
> Voila! This should work.
>
>   On Jan 9, 2008 10:44 AM, Ahmed Nawar <[EMAIL PROTECTED]> wrote:
>
> > I really wonder do we have to do VM-Integration in order for the VM to
> > work
> > probably in the SRST mode?
> >
> > And if so what is the needed configuration to do so? and what is the
> > working pattern ?
> >
> >
>
>
>


-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] VM integration with SRST

[Scott Monasmith]

For myself, VM-integration is the only way I have been able to get VM to
work during SRST. You may be able to get creative with translation patterns
as well.

CONFIGURATION
telephony-setup
 voicemail 913335551234
 call-forward busy 913335551234
 call-forward noan 913335551234 timeout 12
 timeout interdigit 7

vm-integration
 pattern direct # CGN (setup alternate extension in Unity that matches full
10-digit number)
 pattern trunk-to-ext busy # FDN
 pattern trunk-to-ext noan # FDN
 pattern ext-to-ext busy # FDN
 pattern ext-to-ext noan # FDN

dial-peer voice 86 pots
 destination-pattern 913335551234T (make sure you add the T so it will
append the # FDN)
 port 0/0/0:23
 prefix 13335551234

What this does during a busy condition on the phone (x2001) in the SRST
location is send a call to the HQ gateway, via the PSTN, with a called
number of 13335551234#6175212001.

When the HQ gateway receives this call, the significant digits field is set
to 4 for inbound calls on the HQ. Therefore, 13335551234#6175212001 is
stripped down to 2001. When CallManager tries to contact the phone @ x2001
and sees that it is not registered (because of the SRST condition) it
forwards the call based on the Busy condition of x2001 (which should be
voicemail).

Voila! This should work.

On Jan 9, 2008 10:44 AM, Ahmed Nawar <[EMAIL PROTECTED]> wrote:

> I really wonder do we have to do VM-Integration in order for the VM to
> work
> probably in the SRST mode?
>
> And if so what is the needed configuration to do so? and what is the
> working pattern ?
>
>

[OSL | CCIE_Voice] aliases on SRST

[Scott Monasmith]

Below is my config for SRST:


call-manager-fallback
 max-conferences 1 gain -6
 timeouts interdigit 5
 ip source-address 10.2.201.1 port 2000
 max-ephones 10
 max-dn 20 dual-line preference 2
 dialplan-pattern 1 6175222... extension-length 4
 transfer-pattern .T
 alias 1 2001 to 2001 preference 1 cfw 91211003 timeout 4
 translation-profile incoming xlate
 call-forward pattern .T

dial-peer voice 1 pots
 translation-profile incoming xlate
 incoming called-number .
 direct-inward-dial
!
dial-peer voice 11 pots
 destination-pattern 91[2-9]..[2-9]..
 port 0/0/0:23
 forward-digits 11

1) During SRST, if I call 1.617.522.2001 from HQ, it will ring extension
2001, but it will give me a fast busy after ringing for 4 seconds
2) I ran 'debug voip dialpeer' and 'debug isdn q931' and I do not get any
output when troubleshooting step 1
3) If I remove the alias command and enter 'call-forward noan 91211003
timeout 4' the call forwards from extension 2001 just fine and the debugs
regurgitate the dialpeer and isdn info.

I have been able to do this before several times. Do you know why I can't
get the alias to work?

Re: [OSL | CCIE_Voice] Conference bridge on CME

[Scott Monasmith]

Therefore, you can only setup 3-way conferencing via Telephony-Setup using
the 'max-conferences' command?

On Jan 8, 2008 9:10 AM, Voice Noob <[EMAIL PROTECTED]> wrote:

> I don't think CME 3.3 supports conferencing using DSP's. ONly
> transconding.
>
>
> On 1/8/08, Scott Monasmith <[EMAIL PROTECTED]> wrote:
> >
> > When you configure the DSPs on CME for conferencing, do you need to
> > register them under Telephone-setup like you do for the transcoder?
> >
> >
> >
> >
>
>


-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

[OSL | CCIE_Voice] Conference bridge on CME

[Scott Monasmith]

When you configure the DSPs on CME for conferencing, do you need to register
them under Telephone-setup like you do for the transcoder?

Re: [OSL | CCIE_Voice] Cat6500 QoS ACLs

[Scott Monasmith]

Thanks, Vik.

I assume the TRUST-DSCP acl can run concurrently with the CCM-SIGNAL acl on
the same port then?

In one of IPExpert's workbooks, the solutions mentioned the port ranges of
1024-4999 and 11000-11999. What were these ports referring to?

Thanks,
Scott

On Jan 8, 2008 2:46 AM, Vik Malhi <[EMAIL PROTECTED]> wrote:

>  Couple of issues. For a full list of port numbers see link below. I've
> modifiedthe ACL with the SIP port #'s and removed the unused H323 port
> numbers. Also each line within the ACL will need the policer applied if you
> are policing all signaling traffic.
>
>
> http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/sec_vir/udp_tcp/41plrev2.pdf
>
>
> set qos policed-dscp-map 24,26:10
> set qos policer aggregate POLICE-CCM rate 32 burst 8000 policed-dscp
> set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM  tcp any range 2000
> 2002 any
> set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM  tcp any eq 2428
> any
> set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM  udp any eq 2427
> any
> set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM  tcp any any eq
> 1720
> set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM  tcp any any eq
> 1718
> set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM  udp any eq 1719any
> set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM  tcp any any eq
> 5060
> set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM  udp any any eq
> 5060
> commit qos acl CCM-SIGNAL
> set qos acl map CCM-SIGNAL 3/3
>
> Vik Malhi
> CCIE Voice Instructor / Developer - IPexpert, Inc.
> CCIE Voice #13890 CCSI #31584
> URL: http://www.IPexpert.com <http://www.ipexpert.com/>
> Toll Free: +1.866.225.8064
> International: +1.810.326.1444
>
>
>  --
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *Scott Monasmith
> *Sent:* Monday, January 07, 2008 2:36 PM
> *To:* ccie_voice@onlinestudylist.com
> *Subject:* [OSL | CCIE_Voice] Cat6500 QoS ACLs
>
>   Would the following configuration accomplish following
> 3 items correctly:
> 1) trust DSCP from CCM server
> 2) mark signaling traffic from CCM to cs3
> 3) police signaling from CCM to 32k
>
>
> set port qos 3/3 port-based
> set port qos 3/3 trust trust-dscp
> set qos acl ip TRUST-DSCP trust-dscp any
> commit qos acl TRUST-DSCP
> set qos acl map TRUST-DSCP 3/3
> set qos cos-dscp-map 0 8 16 24 32 46 48 56
> set qos acl ip CCM-SIGNAL dscp 24 tcp any range 2000 2002 any
> set qos acl ip CCM-SIGNAL dscp 24 tcp any eq 2428 any
> set qos acl ip CCM-SIGNAL dscp 24 udp any eq 2427 any
> set qos acl ip CCM-SIGNAL dscp 24 tcp any any eq 1720
> set qos acl ip CCM-SIGNAL dscp 24 udp any any eq 1719
> set qos acl ip CCM-SIGNAL dscp 24 tcp any any range 1024 4999
> set qos acl ip CCM-SIGNAL dscp 24 tcp any any range 11000 11999
> commit qos acl CCM-SIGNAL
> set qos acl map CCM-SIGNAL 3/3
> set qos policed-dscp-map 24,26:10
> set qos policer aggregate POLICE-CCM rate 32 burst 8000 policed-dscp
> set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM  any
> commit qos acl CCM-SIGNAL
> set qos acl map CCM-SIGNAL 3/3
>
>
>
>
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

[OSL | CCIE_Voice] Cat6500 QoS ACLs

[Scott Monasmith]

Would the following configuration accomplish following 3 items correctly:
1) trust DSCP from CCM server
2) mark signaling traffic from CCM to cs3
3) police signaling from CCM to 32k


set port qos 3/3 port-based
set port qos 3/3 trust trust-dscp
set qos acl ip TRUST-DSCP trust-dscp any
commit qos acl TRUST-DSCP
set qos acl map TRUST-DSCP 3/3
set qos cos-dscp-map 0 8 16 24 32 46 48 56
set qos acl ip CCM-SIGNAL dscp 24 tcp any range 2000 2002 any
set qos acl ip CCM-SIGNAL dscp 24 tcp any eq 2428 any
set qos acl ip CCM-SIGNAL dscp 24 udp any eq 2427 any
set qos acl ip CCM-SIGNAL dscp 24 tcp any any eq 1720
set qos acl ip CCM-SIGNAL dscp 24 udp any any eq 1719
set qos acl ip CCM-SIGNAL dscp 24 tcp any any range 1024 4999
set qos acl ip CCM-SIGNAL dscp 24 tcp any any range 11000 11999
commit qos acl CCM-SIGNAL
set qos acl map CCM-SIGNAL 3/3
set qos policed-dscp-map 24,26:10
set qos policer aggregate POLICE-CCM rate 32 burst 8000 policed-dscp
set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM  any
commit qos acl CCM-SIGNAL
set qos acl map CCM-SIGNAL 3/3

Re: [OSL | CCIE_Voice] FRTS in e-book 4

[Scott Monasmith]

If you are asked in the lab to configure FRTS, does that specifically mean
legacy FRTS (which I assume is done via a map-class)? Or, does it leave it
up to the candidate to decide whether to use class-based or use a map-class?

Basically, does 'FRTS' mean to use the map-class only method. Or, will the
'map-class' method only be described as 'legacy' FRTS?

On Jan 4, 2008 7:56 AM, Vik Malhi <[EMAIL PROTECTED]> wrote:

>
> To put "other" traffic into the WFQ use the following command in the
> policy-map:
>
> Class class-default
>  fair-queue
>
> When to use class-based shaping and when to use legacy FRTS? Answer- you
> have a choice unless specified. It doesn't make a difference.
>
> Something to be aware of though- when using legacy FRTS you have to enable
> frame-relay traffic-shaping on the PHYSICAL interface so all PVC's would
> be
> subject to legacy FRTS. So in summary, all PVC's should use the same
> method
> of shaping. [Note: the class-based shaping method does not require
> frame-relay traffic-shaping to be enabled on the interface].
>
>
>
> Vik Malhi
> CCIE Voice Instructor / Developer - IPexpert, Inc.
> CCIE Voice #13890 CCSI #31584
> URL: http://www.IPexpert.com 
> Toll Free: +1.866.225.8064
> International: +1.810.326.1444
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Ahmed Nawar
> Sent: Wednesday, January 02, 2008 5:18 AM
> To: ccie_voice@onlinestudylist.com
> Cc: Mark Snow
> Subject: [OSL | CCIE_Voice] FRTS in e-book 4
> Importance: High
>
>
> Gents,
>  I have a confusion regarding the Qos question in the e-book 4 which
> states
> that we should configure llq with MLP and to place the voice traffic in
> the
> priority queue, the control traffic must be placed into CBWFQ and all
> other
> traffic must be placed into a WFQ
>
> I realy need to know what is the configuration needed for the last part in
> red.
> Also when to use the Clas based traffic shaping and when to use the normal
> way of traffic shaping ( map-class frame relay  method)
>
> With Regards
>
>
> 
> Ahmed Mohammed Nawar
> Networking Specialist
> IBM Integrated Communications Services
> Cisco IP Communications Support Specialist Cisco IP Telephony Operations
> Specialist CCNA , CCNP,  CCVP , CCIE Voice Written
>
> IBM-WTC  ,  Egypt Branch
> Building  C - 10
> Pyramids Heights Office Park,
> KM.22 -Cairo - Alex. Desert Road,
> P.O. Box 166 El-Ahram
> Giza, Egypt.
>
> Mobile:(20-10) 1552657
> Tel.:  (202) 3536 2 536 Ext. 1120
> DID:  (202) 3536 1120
> Fax:  (202) 3536 2 505
> Email: [EMAIL PROTECTED]
>
>


-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

Re: [OSL | CCIE_Voice] CME on DocCD

[Scott Monasmith]

Thanks for replying, Matthew, and I understand what you are referring to.
However, this isn't a redirect this is an exodus that requires you to move
outside of univerCD manually. Try the link below...

http://www.cisco.com/univercd/cc/td/doc/product/voice/its/cme33/cme33sa/sa33_mov.htm

On Jan 7, 2008 9:05 AM, Matthew Cody <[EMAIL PROTECTED]> wrote:

>  I forget if it was Mark or Vik, but one of the instructors confirmed that
> any 'redirects' will still work in the lab.  In other words, if you can get
> to it via a UniverCD link today you will be OK in the lab.
>
>
>
>
>
>
>
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *Scott Monasmith
> *Sent:* Monday, January 07, 2008 9:39 AM
> *To:* ccie_voice@onlinestudylist.com
> *Subject:* [OSL | CCIE_Voice] CME on DocCD
>
>
>
> I just noticed that the CallManager Express System Admin guide has moved
> away from the DocCD (www.cisco.com/univercd). Does anyone know of another
> usuable rendition of this in case it needs to be referred to during the
> exam?
>
>
>
> Scott
>



-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

[OSL | CCIE_Voice] CME on DocCD

[Scott Monasmith]

I just noticed that the CallManager Express System Admin guide has moved
away from the DocCD (www.cisco.com/univercd). Does anyone know of another
usuable rendition of this in case it needs to be referred to during the
exam?

Scott

[OSL | CCIE_Voice] frame-relay traffic shaping

[Scott Monasmith]

Is there a difference/preference to configure shaping on Frame-Relay with
Class-based vs. Map-Class Frame-Relay?

policy-map FRTS
 class class-default
  shape average 729600 72960 0

VS.

map-class frame-relay FRTS
 frame-relay cir 729600
 frame-relay bc 7296
 frame-relay be -
 frame-relay mincir 729600

Secondly, when do you have to configure the "frame-relay traffic-shaping"
command on the physical interface?

Re: [OSL | CCIE_Voice] Proctor Labs - PSTN phone

[Scott Monasmith]

Thanks, Mark.

What are the numbers on the PSTN phone, so I can test calling to the PSTN
phone from CCM via GK via CME and vice versa.

Cheers,
Scott

On Dec 5, 2007 2:08 PM, Mark Snow <[EMAIL PROTECTED]> wrote:

> Scott,
>
> The best way to verify that calls are coming in from the PSTN properly
> - is to first get outbound call routing working - say from HQ site -
> you can dial 911 and 1) you hear ringback  and 2) it auto-answers.
> Then you can make a call from HQ Phone out to the PSTN and into BR1
> site - so have your HQ IP blue (or hardware IP Phone if you are using
> our EasyVPN config) call out to 9-1-617-52x-2003 and see if you see
> the call come in properly on the BR1 router using a "debug isdn q931"
> on that router - and then finally your BR1 Phone 3 should ring.
>
> We are working on getting the PSTN Phone to be user-controllable - and
> we have a working solution - but it is not fully rolled out just yet.
> Will certainly keep you updated once it is fully supported.
>
> HTH,
>
> Mark Snow
> CCIE #14073 (Voice, Security)
> CCSI #31583
> Senior Technical Instructor - IPexpert, Inc.
> A Cisco Learning Partner - We Accept Learning Credits!
> Telephone: +1.810.326.1444
> Fax: +1.309.413.4097
> Mailto: [EMAIL PROTECTED]
>
> IPexpert - The Global Leader in Self-Study, Classroom-Based, Video On
> Demand and Audio Certification Training Tools for the Cisco CCIE R&S
> Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and
> CCIE Storage Lab Certifications.
>
>
> On Dec 5, 2007, at 12:52 PM, Scott Monasmith wrote:
>
> > What is the best way to utilize the PSTN phone when using a Proctor
> > Labs session? I see the VTGO PSTN (ip blue) phone icon on one of the
> > servers, but when you launch it, it fails because it cannot find a
> > sound card/device. Is there a another way to verify inbound calls
> > from the PSTN when using a Proctor Lab session?
> >
> > Cheers,
> > Scott
> >
> > --
> > "There are only 10 types of people in the world: Those who
> > understand binary, and those who don't"
>
>


-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

[OSL | CCIE_Voice] Proctor Labs - PSTN phone

[Scott Monasmith]

What is the best way to utilize the PSTN phone when using a Proctor Labs
session? I see the VTGO PSTN (ip blue) phone icon on one of the servers, but
when you launch it, it fails because it cannot find a sound card/device. Is
there a another way to verify inbound calls from the PSTN when using a
Proctor Lab session?

Cheers,
Scott

-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"

[OSL | CCIE_Voice] CUE Recovery/Setup

[Scott Monasmith]

When you lose access to the CUE CLI, what is the best way to restore it?

I have setup CUE in the following method:

service-engine 0/0
no shut
ip address unnumbered f0/0.200
service-module ip add 10.10.10.2 255.255.255.0
service-module ip default 10.10.10.1 255.255.255.0

ip route 10.10.10.2 255.255.255.255 service-engine 0/0

However, when I try the following command - "ser ser 0/0 session" - I am
unable to access CUE and receive the following message scroll across the
screen:

out of memory for event sources.

Any info on this would be greatly appreciated.