thanks bill !!
And thats why we call it state of the art, greatest, cutting edge Catalyst :)
.. honor my sarcasm...i spent couple of hours playing with the accesslist,
policymaps etc ...
From: b...@ucguerrilla.com
Date: Wed, 17 Apr 2013 19:08:11 -0400
To: kamina.j...@yahoo.com
CC:
And where do you plan to invoke the script and vxml function?
Date: Wed, 27 Mar 2013 23:51:10 +0300
From: aboaz...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] MVA functionality
Hello Friends...
I have the following setup, I am not sure if the will be suitable to
grand welcome to the lab 7 carnival! many people get the lab7 on 2nd attempt ,
do the same things and score very high points and fail or pass (no visibility
to points) .. so with this lab 7 there is no definitive answer on many things
like CUPC, signaling, call routing.
Date: Sun, 24 Mar 2013
of the list of the commands that dont make much sense, my fav is the clock
summer-time .. lots of CCIE experts who passed have said they did the summer
time command .. does HK have summer time and do we need to put this command on
HK as well? ... in Bahrain we dont have summer, winter, monsoon
As far as i remember the requirement under call routing doesn't specify the
called party type National for long distance calls out SiteB gateway AND the
question clearly said that YOU ARE ALLOWED TO SEND 4 DIGITS in Calling Number
field to the Telco for these calls ... so why not use this
the requirement is always for SiteB calling into SiteA voicemail by hitting
Messages button. SiteA is always MGCP gateway. Calling Party Xform isnt any use
on MGCP gateway for incoming calls.
here is another way of doing it ...
Voicemail Pilot for CUC is 2200
call-manager-fallbackvoicemail
received very low marks
on this section. Can anyone help?
Thanks
Jamie Parr
Engineer - IT
jamp...@cisco.com
Phone: +44 20 8824 2641
Mobile: +44 7590622049
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com]
On Behalf Of Pixar Perfect
Sent
Based on my guesstimate these are the mandatory commands needed to make the
script happy.
dial-p v 1 voipdesc SUB-dialpeervoice-class codec 1voice-class h323 1no
vaddestination-pattern 3...$session target ipv4:10.10.210.11 --SUBip qos
dscp cs3 signalpreference 0dtmf-relay h245-alpha!!!dial-p
you still need the Phone NTP reference on the labs as CUPC client is a SIP
client ..there are no SIP phones on the Version 3 labs but we might see lot on
Version 4.
Date: Tue, 5 Mar 2013 01:05:22 +0300
From: aboaz...@gmail.com
To: corygray22...@hotmail.com; bring...@gmail.com
CC:
Gary ..you mentioned B channel busyout on service parameter. in my
understanding this was only needed when you would download the GW config from
CCM i.e., ccm-manager config. it doesn't make any sense to use this service
parameter as most of the solution guides (INE, IPXEPERT, 360) do not
hello experts, reading thru sample RSVP configs on the OSL forum i see the wide
of use codec passthru under software mtp configuration for rsvp , any reason
why codec passthru needed on the mtp. i have it working always without this
command. or is this again one of those configuration pieces
hello, any need for these commands? i saw these on IPExpert configurations and
INE guides in some of their labs.
sccp ccm gr 1ass ccm 1 pr 1ass ccm 2 pr 2ass ccm 3 pr 3 -- conf registration
for srstass pro 1 reg br2cfbass pro 2 reg br2transwitchback method
immediateswitchover method
What is the quickest way to record AU file in the lab for BACD file needs? i
tried recording script on the IPCC Express but it dumps WAV file.
CUC recording applet never works on my windows box. any better way? THANKS
Experts and wannabe experts friends,
what are the best practices for MWI in CME and SRST modes for the CUE site BR2?
i was used to using MWI ON and MWI OFF DNs on a CME but i was told by a fellow
aspirant that MWI ON/OFF are not preferred (grading wise) and that solicited
MWI is that gets you
is never discussed.
Getting the question right is the easy part!
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Pixar Perfect
Sent: Friday, February 15, 2013 5:26 PM
To: CCIE Voice OSL
Subject: [OSL | CCIE_Voice] SRST transfer system
When working on Shared DNs and cBarge question (5-lab handbook, Lab1) that
needs use of CFB on Site C, do we need to remove the Barge Softkey from the
Remote in Use state? do you think it is good idea to disable Built in Bridge
for the two phones that have a shared line and need GW CFB for
transfer-system full-consultdo we need to specify this? I thought by
default it is wnabled but I read on voiceie forum someone scored nothing on
SRST adn the only conclusion was the transfersystem consult was missing. Any
thoughts?
srst mode auto-provision all srst ephone description
Is there a need to enable(check) Setup non-ISDN Progress Indicators IE Enable
on the MGCP GW page ?___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
Are
When Site B Phone 2 (DN 3002) hits the messages button in
SRST, the SRST call control dials 912025552220 to the PSTN to ring the
Voicemail ports at Site A. Unity plays generic Welcome greeting instead of the
user greeting of user 3002. The
requirement is NOT to use Alternate
that strip the extra digits
and apply it to the vm hunt. Make the patterns only match the incoming numbers
from site B?
In IPExpert advanced labs (OWLE/5 Lab pack) that would be 408387300X so just
make the pattern 408387.300X and strip it to the dot
On Tue, Feb 5, 2013 at 2:35 AM, Pixar
CUCM registers to the GK as a VOIP-GW. How do I make it register as a H323-GW?
The requirement on a practice lab demands the CUCM register to the GK as a
H323-GW and not VOIP-GW.
On a GW,, you could just do the allow h323 to h323 to make it register as
H323-GW. But not sure abt CUCM.
Any
Hello,
is it advisable to bring down the Connection Monitor Duration under Device Pool
to a low value like 30 seconds to expedite the SRST testing in the lab? Does it
affect grading if we happen to put it back to default?
thx
On the CUC, here are the following checkboxes I typically enable (check). Any
other configuration under the phone system?
Default TRAP Switch Enable for Forwarded Message Notification Calls (by Using
DTMF) Enable for Forwarded Message Notification Calls (by Using Extension) Send
Message Counts
flexibilty in using auto qos, you can adjust all the
values as with the manual way.
Regards
Steffen
Am Samstag, 5. Januar 2013 schrieb Pixar Perfect :
What is the best practice for the real lab QOS after running the AutoQos - use
the AutoQoS nomenclature and make changes to the classmaps, policymaps
in my experience the phone had to be factory reset. make sure there is a valid
DHCP that phone can access after the factory reset sequence.
To: ch.christ...@logicom.net; b...@ucguerrilla.com
From: singh8...@in.com
Date: Sat, 5 Jan 2013 10:00:46 +0530
CC: ccie_voice@onlinestudylist.com
What is the best practice for the real lab QOS after running the AutoQos - use
the AutoQoS nomenclature and make changes to the classmaps, policymaps and
map-class OR copy the AutoQOS output on a notepad and rename policymaps,
classmaps and mapclass?
I typically follow the second route as it
that some older phone models (e.g.7960) only support external
provisioning so you might see different results depending on what hardware you
have - but the Cisco lab itself is 7965 phones (this is public published info)
On 17 Nov 2012 00:01, Pixar Perfect pixarperf...@live.com wrote:
I am working
I am working on the question from the lab experience handbook where SA phone 1
needs to see No Services Configured when it hits directory button. IPEXPERT
suggests that we delete the Service URL from EP and then setup the SA Phone 1
device configuration page Service parameter to External ? I
Guys
any suggestions as to which is the best way to remember the class selectors to
DSCP mappings? I mean CS3 to 24 and CS1 to 8 ... any easy trick?
Thanks !! ___
For more information regarding industry leading
The requirement is as follows on Lab 2 QOS section of the 5-Lab Handbook. For
traffic being sent to the Site A gateway ensure that the traffic marked with
COS 5 is dropped if the queue 1 is 75% full
The Solution guide (page 408) has the following solution.
mls qos queue-set output 2 threshold
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