All,
I have a very strong evidence that supports my statement, and in case if i
don't get good feedback or positive response, I will approach lawyer and take
it to the court for justice so that no others would get affect in future.
thank you
krishna.
Marko,
i opened case with support, and i hope i will get some good feedback . I
appreciate your help.
thank you
krishna.
From: Marko Milivojevic
To: Krishna
Cc: Cory Gray ; Online Study
Sent: Wednesday, October 31, 2012 1:13 PM
Subject: Re: [OSL
caused 0 points for that
many tasks, otherwise we would never come to know what we did wrong.
thank you
krishna.
From: Cory Gray
To: 'Krishna' ; 'Online Study'
Sent: Wednesday, October 31, 2012 9:22 AM
Subject: RE: [OSL | CCIE_Voice]
gatekeeper twice
and check everything and works as expected. I think CCIE is not written in my
fate, and so its better to leave this journey rather holding it for none.
thank you
krishna.
From: Mohamed Hassan
To: Krishna
Cc: Online Study
Sent: Wednesday
ating the tasks thrice and
check everything line by line, and the end showing the score report as fail"...
This is completely insane. I was wondering if i can legally proceed so that
justification will be done for the right candidates.
Thank you
krishna.
the way it showed the
results... no more CCIE in my life...
i appreciate all my friends who helped me in this journey.
thank you
krishna. ___
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Are you a
unk port to router
mls qos trust dscp
service-policy input p-mgcp
From: Cory Gray
To: Kevin Spicer
Cc: Krishna ; Online Study
Sent: Wednesday, October 24, 2012 12:32 PM
Subject: Re: [OSL | CCIE_Voice] access-list not matching any packets for mgcp
ports on swi
you
krishna.___
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were filled it comes to
t3 which has 75% i.e. it is the last threshold where it will take/borrow the
memory value from reserved threshold when desired. long story short... right
way of doing it either assign it to t2 or t1 and assign threshold value of 75%
for correct approach...
thank you
krishna
Dan & Cory,
thanks for your feedback.. i thought the same but i was triggered by looking at
the ipexpert solutions guide for 5 lab handbook and so taking second thought
from expertise guys like you..
thank you
krishna.
From: Dan Quinlan (daquinla)
To:
yes cory you're correct about understanding the question exactly.
thank you
krishna.
From: Cory Gray
To: Rrcrumm
Cc: Krishna ; Online Study
Sent: Saturday, October 20, 2012 8:23 PM
Subject: Re: [OSL | CCIE_Voice] cme ip add or loopback ip i
krishna.___
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krishna.___
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it needs to make this
work..
thank you
krishna.
From: Bruno Nonogaki
To: Kevin Spicer
Cc: Online Study ; Krishna
Sent: Wednesday, October 17, 2012 4:05 PM
Subject: Re: [OSL | CCIE_Voice] Meet Me question - CME GW
Kevin,
This approach sounds good.
I
Randell,
under telephony-service define after-hours-block pattern , after-hours day
mon 0:00 24:00, repeat it for days tue,wed,thurs,friday,sat,sunday. under
ephone-dn where the phones can access meetme number apply after-hours-exempt...
and i m sure you will be good.
thank you
krishna
t;mls qos trust dscp" for the
traffic coming from router to switch isn't it?? I see some guys putting as mls
qos trust cos which i feel incorrect since layer 2 to laye3 mappings were done
using cos-dscp mapping. please correct me if i am wrong.
thank you
krishna.
___
approach is correct in dealing
this question correctly?? does it matter whether we assign cos values to t1 or
t2 or t3 in the queues???
your input is much appreciated.
thank you
krishna.___
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button is enabled with 1 . Is this is a
bug for cme as srst ??? any help is much appreciated.
thank you
krishna.
here is the config:
SiteB-RTR(config)#do sh run | s ephone
srst ephone template 1
max-ephones 10
ephone-dn-template 1
call-forward busy 2220
call-forward noan 2220 timeout 20
Amaia,
follow the blog.ipexpert.com for right information on mgcp debug events... This
blog is incorrect in showing the right output for pstn phone is ringing..
thank you
krishna.
From: Amaia Lesta
To: ccie_voice@onlinestudylist.com
Sent: Thursday
e questions and the exact
requirements is very very very unclear" After looking at my score report, i
remember kevin's wallace statement about his first lab attempt...
just let me know if you have any questions...
thank you
krishna.
Fro
1.) it depends, for example if this is a router with switch module in it i.e.
BR1, i would not use trust since the marking will be done with acls by auto qos
voip
2.) it depends, if you're told to use class based compression then you have to
use compression header ip rtp in the policy-map other
never mind guys.. after skimming the online blog, i see this is expected
behavior... the calling name does't show when the phone is ringing, it only
shows when the phone is connected..
thank you
Krishna.
From: Dan Quinlan (daquinla)
To: Krishna
Cc: O
mva is authenticated the calling name is not supported
either the call going out from the pstn gw or to internal cisco phones..
thank you
krishna.
From: Dan Quinlan (daquinla)
To: Krishna
Cc: Online Study
Sent: Sunday, September 16, 2012 1:47 PM
Subject
after authentication ... is
there a way that we can support calling name for mva set up??? Any input is
much appreciated.
thank you
Krishna.___
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Are you a
the correct url were pushed
into the telephony-service otherwise those who do cli, they have to remember
this command for reference...
thank you
krishna.
From: Kevin Spicer
To: Krishna
Cc: "Online Study (ccie_voice@onlinestudylist.com)"
; Gurp
All,
please correct that authenticate url is wrong .. the right one is url
authenticate http:// CMEipaddress/CCMCIP/authenticate.jsp, not the url
authentication http://<>/voiceview/authentication/authenticate.do.
Thank you
krishna.
From: Steven S
steven,
for srst voicemail support, just put under the hunt pilot when it
explicitly states that alternate extension cannot be used..
thank you
krishna.
From: Steven Sarrick (ssarrick)
To: "ccie_voice@onlinestudylist.com"
Sent: Monday, S
38 92 138, this takes queue
1 to 138%, and once it is over it comes back to share mode, since the priority
queue overwritten the shape, and left over is share value whatever defined at
the interface level... please correct me if i m wrong..
thank you
krishna.
r's ANI... but still tells the same prompt...
can you advice me where the crucial step i m missing here...any help on this
matter is much appreciated.
thank you
krishna.___
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ires that there shouldn't be no interdigit timeout... please advice me on
this matter...
thank you
krishna..___
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Are you a CCNP or CCIE and looking
http://docwiki.cisco.com/wiki/CUPC_ports_for_7.0_to_8.5
i see the port listed differently for the cupc versions in the ciscodoc
thank you
krishna.
From: William Bell
To: Krishna
Cc: Online Study
Sent: Thursday, August 30, 2012 11:28 AM
Subject: Re
k you
krishna.___
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no issues...
thank you
krishna.___
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jason,
from router to ccm the dnis is 4 digits whereas ANI is sent completely to cucm
whatever the pstn is delivering...
From: Jason Aarons (AM)
To: Krishna ; Online Study
Sent: Wednesday, August 29, 2012 6:36 AM
Subject: RE: [OSL | CCIE_Voice] mva showing
ed in null partition but no difference ..
thank you
krishna. ___
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hi folks,
i was wondering slrg and lrg are same in terms of functionality... can both
viewed as same ..??? please advise me on this query...
thank you
krishna.___
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bill,
the point is pstn callers are able to leave voicemail when the site is in SRST,
the real problem is retrieving the message from the CUC... i don't understand
what does it has to do with alternate extension..
thank you
krishna.
From: Bill Lak
email to this
phone while in srst... any advice on this matter is appreciated..
thank you
krishna.___
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Are you a CCNP or CCIE and looking for a job? Chec
in term of reconfiguration and web
gui access... can anyone advice what i m missing in my cli approach...
note: after doing the cue configuration using cli, i saved the config and
reloaded the cue before approaching the gui process...
any advice is much appreciated.
thank you
krishna
voicemail prompts shows up correctly but in srst its
not... is there any command that needs to be enabled in the cue for active
transfer calls??
thank you
krishna..___
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d mwi server, and on
ephone-dn mwi sipapart from this configuration do i required to create
ephone-dn for mwi as well/?
thank you
Krishna.___
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Are
any update would be much appreciated..
From: Krishna
To: Online Study
Sent: Tuesday, August 14, 2012 5:32 PM
Subject: subscribe cucm is not showing up
hi james/doug
i don't see cucm subcriber showing up on the switch , and also i cannot reach
t
CMPub Fas 1/0/4 130 H none foun eth0
SEP001BD4C6C195 Fas 1/0/2 178 H IP Phone Port 1
UCCX7-PUB. Fas 1/0/4 177 H Win2000 S Eth 2/2
can you please fix this ???
thank you
kr
ramy,
i apologize for not understanding the question properly... i overlooked at your
question..
krishna.
From: Ramy Abdelrahim
To: vinayak_...@yahoo.com; daqui...@cisco.com
Cc: ccie_voice@onlinestudylist.com; michael.se...@compucom.com
Sent: Monday
must use ntp master as well ntp source
to which the time on the hq router binds and apparently the cucm can reference
the clock from hq router as well...
thank you
krishna
From: Dan Quinlan (daquinla)
To: Ramy Abdelrahim
Cc: "ccie_voice@onlinestud
t of your questions and get you on your way.
> >
> > Thanks,
> >
> > Justin McIntyre
> >
> > This email and any files transmitted with it are confidential and are
> > intended for the sole use of the individual to whom they are addressed.
> > Black Box Corporation reserves the right to scan all e
ch is called from the
policy-map... and also another way of doing it which is police 32000 under
class-map
is it really matters when using police or bandwidth for the wan qos...??? plz
advice me on this matter...
thank you
krishna.___
For more inform
??
your help is appreciated..
thank you
krishna.___
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pilot number extensions
started ringing...
my query is do i need to have h323 voip dial peer as well for the bacd setup...
i am not sure about this and therefore would like to get some feedback from you
guys...
any help on this matter is much appreciated..
thank you
krishna
me-vm-retry 2
any advice on this matter is much appreciated..
thank you
krishna.
From: Peter Simmons
To: Krishna
Sent: Saturday, August 11, 2012 2:51 AM
Subject: Re: [OSL | CCIE_Voice] no prompts with bacd
Krishna,
You config has a couple of errors aroun
param busy
are the one created by me in the application but no use since it didnt work..
any help on this matter is much appreciated...
thank you
krishna.
___
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even get to the prompt of mva...
thank you
krishna.
From: Vipul Jindal (vipjinda)
To: ccielabrat ; "ccie_voice@onlinestudylist.com"
Sent: Tuesday, August 7, 2012 2:29 PM
Subject: Re: [OSL | CCIE_Voice] MVA -"Your call can not be completed as diale
ce or help is much much appreciated, since i spent 7 hours
troubleshooting the phone registration but couldn't crack it...
thank you
krishna..___
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Are
required if that device is
synchronized with external ntp server.. any comments on my advice is much
appreciated...
thank you
krishna.___
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Are you a CCNP or
x27;t work in srst mode??
any advice or suggestion on this query is much appreciated..
thank you
krishna.___
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www.ipexpert.com
Are you a CCNP or CCIE and looking for
using proctorlabs...
your help is much appreciated...
thank you
krishna.___
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explicitly put the srst reference under site device
pool for the sites to be in srst mode rather leaving the value to "use
default-gateway"???(provided that default-gateway is the one default-router for
the phones)
any help in making a clear understanding is much appreciated..
thank y
my bad... i assumed it is like cme service url.. thanks guys for your quick
responses...
thank you
krishna.
From: Dan Quinlan (daquinla)
To: Krishna
Cc: Online Study
Sent: Wednesday, August 1, 2012 9:49 PM
Subject: Re: [OSL | CCIE_Voice] service url in
hi folks,
can any one help me out with the service url for phoneview for ip phones in cucm
thank you
krishna.___
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Are you a CCNP or CCIE and looking for a
if it is h323 gateway, i will create a translation rule and apply at the
dial-peer... if it is mgcp gateway do it on the call manager route list detail
level...
thank you
krishna.
From: Randall Crumm
To: Online Study
Sent: Monday, July 30, 2012 11:33 AM
n i gave the command system message then it showed as your
current options.. is this is a bug in srst command under telephony service???
thank you
krishna.
From: Rrcrumm
To: Dan Quinlan (daquinla)
Cc: Online Study
Sent: Sunday, July 29, 2012 9:58 PM
Subject: R
Thanks bill dan helped me out on this perspective.. thanks once again bill
for your quick response..
From: Bill Lake
To: Krishna ; "ccie_voice@onlinestudylist.com"
Sent: Saturday, July 28, 2012 9:58 PM
Subject: Re: [OSL | CCIE_Voice] no ipcc
appreciated.
thank you
krishna.___
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check service url authentication in the service parameters to the pub ip
address...
From: Juan Carlos Anzola
To: ccie_voice@onlinestudylist.com
Sent: Thursday, July 26, 2012 4:26 PM
Subject: [OSL | CCIE_Voice] IPPM Audio and Visual Alert
Hi Guys,
I am con
Bruno & Justin,
I did enabled all these and even verified twice... what authentication url has
to be there on enterprise parameters??/
thank you
krishna.
From: Bruno Nonogaki
To: Justin McIntyre
Cc: "ccie_voice@onlinestudylist.com"
Sent: M
in on ipphone, and moreover there is no
sound notification for the chat message from other phones. Is this is expected
behavior???
I can hear the sound notifications on cupc whereas on ip phones are not... any
advice on this matter is much appreciated.
thank you
kr
lbox along with mwi functionality, though it
is redirected to voicemail .. All these sites are registered to the call
managers, where Hq, Br2 are the mgcp gateways whereas br1 is h323 gateway..
any advice or help on this issue is much appreciated....
th
rvice...
Note: TEI Assigned is nothing but D-channel is not active, and therefore it
would meet the requirement for the alert action
please correct me if i am missing something or my approach is incorrect..
thank you
krishna.
___
For more informatio
ss web gui for the cue even after providing all the
right info such as ip http server, ip http path, ip http auth local.. the web
browser sits there forever with no output...
does anyone experienced the same problem as i am??? your advice on this matter
is much appreciated.
thank y
appearing as:
cucm > 10.10.200.3
10.10.200.3 --> cme.
how could it be possible to have 10.10.200.3 acting as gateway having the
gateway config on interface lo 0.
any advice on this matter is much appreciated.
thank you
krishna.
___
For more infor
hi folks,
from my observations, i found that mlp would be used if and only if, there are
two sites connecting to a single site via the same physical interface. And, for
frf.12 we use it for ppp link. please correct me or advise me if i m wrong.
Thank you
krishna
DQ is correct with the lab set up process.
bill,
i tried both the configuration but only with trunk mode, no issues at all...
thank you all for your quick response..
appreciated.
thank you
krishna.
From: Dan Quinlan (daquinla)
To: Bill Lake
Cc: Krishna
in cucm as registered.
any advice on this matter is much appreciated.
thank you
krishna.___
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ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] transcoder not functioning in the cube section
Transcoder should be available locally for cube
On Jun 21, 2012 8:23 AM, "Krishna" wrote:
hi folks,
>
>
>I got stuck at vol2 lab1 in section 4.2, where the cube involves in routing
shutdown
HQ-RTR(config)#do sh run | s gatew
h323-gateway voip interface
h323-gateway voip id VIA ipaddr 10.10.110.1 1719
h323-gateway voip h323-id HQ-RTR
h323-gateway voip bind srcaddr 10.10.200.3
gateway
Thank you
krishna.___
For more information
Thanks to Justin , Randell for your awesome help. I will try this setting in my
next lab, and let you know with the updates. Thanks a lot once again for all of
you..
Krishna.
From: Justin McIntyre
To: "ccie_voice@onlinestudylist.com"
Sent: Thur
callstate: SEP, DEP,
Thank you
Krishna.___
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Dan,
A small correction to your statement..rfc2833 is out of band mechanism mostly,
and moreover it doesn't use audio channel, infact it uses rtp header to relay
the dtmf message with a payload identifier.
thank you
Krishna..
From: Dan Quinlan (daquinla
. And, when i called the
5002 from 1002 the call went thru, and this call put on hold and placed another
call and it works fine as well. So, from this analysis can i come to conclusion
that only location based cac, or rsvp cac can only the number of calls between
these two sites???
Thank you
Kr
d any one help me out how the calls are working in between these two. is it
because the phones are registered to cucm, but logically in a different device
pool and therefore it routes directly on cucm your help is much appreciated.
Thank you.
Kr
146
service-policy output LLQ-HQ-BR1
Thank you
Krishna.
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plz ignore my last query,... when i implemented the command frame-relay traffic
shapping on the physical interface, it showed the output
From: Krishna
To: "ccie_voice@onlinestudylist.com"
Sent: Sunday, June 10, 2012 6:45 PM
Subject: Re: no
percent 65
exceed-action set-dscp-transmit 0
fair-queue
map-class frame-relay LINKPMAP-VOIP-RTP
frame-relay cir 729600
frame-relay bc 7296
frame-relay be 0
frame-relay mincir 729600
service-policy output PVOIP-CONTROL-RTP
BR2-RTR#sh policy-map interface
BR2-RTR#
Thank you
Krishna
Ashwani,
you're right. when i uncheck the Enterprise subcription i can see it . thanks
for your help dude..
Thank you.
Regards,
krishna.
From: Ashwani
To: 'Krishna'
Sent: Saturday, June 9, 2012 6:58 PM
Subject: RE: [OSL | CCIE_Voice]
appreciated.
thank you.
Krishna.___
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i have the same issue as well... even most disgusting thing is i cannot access
web gui of the servers, and therefore i m using software vpn for accessing the
web gui for the servers.
thank you
krishna.
From: Dan Quinlan (daquinla)
To: ccie_voice
hi folks,
when i copied the standard appadmin and saved it as ucadmin and password info,
i tried to login using these credentials into ucx but no luck. isnt this is
possible the way i m doing it now???
thank you.___
For more information regarding ind
this router, so
that users can dial into this number and put the passcode for conference into
this router.
your help is much appreciated.
Thank you.
From: Dan Quinlan (daquinla)
To: Krishna
Cc: ccie_voice@onlinestudylist.com
Sent: Monday, June 4, 2012 5:
Hi folks,
i configured the transcoder and conference resources on the CME, but couldn't
make them to work. When i want to conference the line, it says cannot complete
the conference.
here is my config: Did i miss any configuration part in this below config
sccp local FastEthernet0/0
sccp
cially for UCCX and CUPS section. I am
completely lost in both these sections, and i am trying to learn these two in
order to complete the vol1 workbook completely.
thank you.
regards,
Krishna.___
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n it once and it was due to the way I configured the hunt group in CME. I used hunt-group instead of Voice hunt-group and I think that's what caused it to break going through the BACD AA application. Just a thought.> > Thanks,> > Justin> > > > Message: 4> Da
Folks,
it is so strange that when i call the hunt group number 3210 from pstn, both
sip and sccp phone rings. But with AA on cme, only cisco phone rings but not
both. even i verified with the config, and i see hunt group as the right option
when user presses the digit 2. does anyone know why th
where is the bat tool for 7961,
your help is much appreciated.
Thank you.
Regards,
Krishna.___
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hi guys,
in lab 4a section 4.2, when i tried to route the call to cucm from pstn phone,
i get busy tone. i check dialpeer, and it shows dial-peer matching as well, but
nevertheless cucm phone never rang. can anyone advice what would be causing thi
issue??? i did implement allwo connections in v
Hello...
did anyone experience the scenario which i experienced for gateway lab config.
?? basically the dial-p v 1 pots is for pots incoming dial-peer, and when i
issued the command sh dial-p voi summ, the pots dial-p 1 is showing as out
statust down, but strange thing is that calls are connec
4.404: TFTP: Looking for P003-08-6-00.sbn
thank you.
Regards,
Krishna.___
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Hello ALL,
Can anyone advice me what version of sip firmware or sccp firmware should be
given in tftp-server command in the router config.???
for instance: the version for sccp are :
PHONE/7940-7960/P003-08-6-00.bin
PHONE/7940-7960/P003-08-9-00.bin
PHONE/7940-7960/P00308000500.bin
For SIP:
Hello Friends,
I was little bit dazed about the SIP ip phone that is registered via hardware
vpn. The issue when ever i issue restart/reset, the sip phone at my desk starts
the process with upgrading the phone and then registers, whereas the vrack one
registers normally immediately. Can anyone
Note: I used the exactly the config of 'Router & switch' which is in the
proctor labs.
Thank you in advance.
regards,
Krishna Koilada.___
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Hello..
Those who are doing the vol1 workbook, there is a glitch in the lab1B: We
define Pub as dhcp server address, whereas the solutions guide refers to work
to Sub in troubleshooting the dhcp problem.
I finally realized that dhcp server is enabled on Pub, not on Sub. So, I
changed the hel
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