Ken,
I have booked with only like 2 weeks left before the exam date. I'm pretty
sure you have to pay the lab fees up front when you are in the 90 day
period, therefore you can't be dropped due to no payment of fees.
Jeff
-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
Ken,
I have no experience in this scenario. I would just recommend if you aren't
sure whether you're ready and/or can make it, then don't book. As Earl
noted on another response and I did below, you will be charged immediately
when you book in the 90 day window. I don't know what you mean by 1
I tried to cancel within the 3 month period, and they refused. Other than
an absolute emergency I doubt they will grant you the ability to reschedule.
Jeff
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of
wilson.sam...@usc-bt.com
If they aren't specific in the lab, then it's not a requirement of the
solution. I don't believe I've ever seen or heard of this in the lab, more
of a written exam question, if anything.
Jeff
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On
Everything here -
http://www.cisco.com/cisco/web/psa/default.html?mode=prodlevel0=278875240
Also, a couple of the SRNDs will be available on the desktop. This is
common knowledge, not breaking my NDA here :)
Jeff
-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
If your Remote Destination is 4087773434, your route pattern in css-snr
would need to look like that, not \+.!. Unless you are asked for
redundancy with your GWs or to use the Application Dial Rules specifically,
the easiest way to meet the SNR requirement is a SNR partition with a Route
Here is the official list of HW and SW in the lab -
https://learningnetwork.cisco.com/docs/DOC-5292
All 7.0, particularly 7.0.1 as others have mentioned below.
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Amit Singh
Sent:
Adam,
This is correct. If you have a requirement to send the + you'll have to add
at voice-port using a translation-rule.
Jeff
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of adam compton
Sent: Wednesday, March 23, 2011 5:19
We technically arent allowed to answer your question about the lab.
Dont stress out though, if the PSTN router wont accept something or is
expecting something, its a safe bet that you will be told the information
you need.
From: ccie_voice-boun...@onlinestudylist.com
It most likely has to do with your incoming dial-peer on BR2 CME. What do
you have configured for codec there? If you have nothing, then the codec
default is g729r8.
HTH,
Jeff
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of natan
Hi Ron,
The point is that this is the expected behavior. If you don't want your
screen cluttered you can use the privacy button to toggle privacy on/off in
order to go from 2 displays to 1. To my knowledge there is no service
parameter, feature, or anything else besides the privacy setting to
Congrats Akash! Enjoy.
-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Marko Milivojevic
Sent: Friday, January 21, 2011 2:51 AM
To: akash patel
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice]
You have the whole DocCD
(http://www.cisco.com/cisco/web/psa/default.html?mode=prod
http://www.cisco.com/cisco/web/psa/default.html?mode=prodlevel0=278875240
level0=278875240) and some of the SRNDs (including the Enterprise QoS SRND)
available on the desktop. It's very much open book, but
Doc CD Voice and Unified Communications Customer Collaboration Cisco
United Contact Center Products Cisco Unified Contact Center Express
Configuration Guides Configuration Examples and Tech Notes Configure a
One Button Login for IP Phone Agents
Jeff
From:
Hi Mritunjay,
Just to clarify what I meant from my notes. In your scenario below, the
SCCP phone on CME should be able to monitor the CUCM SCCP, however the CUCM
SCCP phone will not be able to monitor the presence of the CME SCCP phone.
If you have a CUCM SIP Phone, then in 7.0.1 you
Hi Shingei,
I misread your question. Please disregard for this scenario.
Jeff
From: givemeccievoice2...@gmail.com [mailto:givemeccievoice2...@gmail.com]
Sent: Tuesday, January 18, 2011 8:30 AM
To: 'ccie_voice@onlinestudylist.com'
Subject: RE: [OSL | CCIE_Voice] Called # manupulation
What type of phone are you using? For the remote in use, this is the
correct way for it to work. Does your line turn red when you are using the
mobile phone?
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of KatGuru
Sent: Thursday,
Refer to the Administration Guide available on Cisco.com
-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of
iptuse...@hotmail.co.uk
Sent: Wednesday, January 12, 2011 8:51 AM
To: ccie_voice@onlinestudylist.com
Associate those CTI RP to the Application User and restart CUE. That should
do the trick assuming everything else is ok.
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Joli-coeur
Wouter
Sent: Wednesday, January 12, 2011 12:31 PM
To:
You add that URL under telephony-service. After that you'll have to reset
the phones to download the new config with that URL.
Jeff
-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of
iptuse...@hotmail.co.uk
Sent:
Can you ping your PC from the server? Can you ping the server from your PC?
Make sure you have connectivity first. Then look into other possible
issues.
Jeff
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Chevy
Sent: Tuesday,
Hi Francesc,
Here is what I have in my notes. I know in the past I had issues getting it
to work with a CUCM monitoring a CME phone, but the CME monitoring CUCM
worked fine. I had written notes that Vik said CUCM SCCP Phone can't
monitor CME phones in 7.0, but SIP should work fine. That
auto provision all will gather ephone and ephone-dn configuration using SNAP
and store in the running config.
auto provision dn will only gather and store the ephone-dn in running
config.
auto provision none will not store anything in the running config.
From:
There is a setting on the directory number configuration page
busy-trigger-per-button and max-calls-per-button.
The busy trigger will accomplish what you want if you set to 1. It's easier
to think of these two settings as channels.
Imagine 8 channels and there are 2 active calls on the line.
You have to create an Alternative Number for that user in CUC.
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of akash patel
Sent: Friday, January 07, 2011 2:58 PM
To: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice]
Do you have this GW configured on CUCM? Is the gateway showing registered
on CUCM? Do you have mgcp configured/enabled on the router? Have you
bounced the MGCP (no mgcp/mgcp) after configuring the pri?
You will see TEI Assigned until you have successfully configured all aspects
of the MGCP
Hi Shrini,
If you follow the Features and Services Guide as mentioned before, you will
have success. You need to configure hairpinning for MGCP to work with
MVA.
The idea is that you will accept the call using 5999, but the MVA pilot
number will be a different number. You will have to
Hi Francesc,
A payload of 20 and 10 is not correct. RSVP and LLQ calculations are two
different things. For RSVP, you calculations are correct.
Correct Payloads (20 ms)
G711 - 160
G729 - 20
For example, FRF.12, G729, with compression:
IP/UDP/RTP - 2 bytes
G729 - 20 bytes
FRF.12
Hi Francesc,
As I noted before, the RSVP bandwidth calculation is different from the LLQ
bandwidth calculation.
For the scenario of 2 RSVP calls, you will need to calculate as follows:
40 + 24 = 64 (one worst case 10ms call and one normal 20 ms)
So under the serial interfaces you will
I looked at the PG and they add in the calculation as I detailed in my most
recent email. However, I am totally with you. The RTP/LLQ is different from
the RSVP CAC and I would think that only a few extra Kbps would account for the
RSVP control traffic in the PQ.
Jeff
From:
Definitely, Im sorry I didnt understand at first J
Happy studies!
Jeff
From: Roig Borrell, Francesc Xavier [mailto:francesc.ro...@tecnocom.es]
Sent: Wednesday, January 05, 2011 12:12 PM
To: givemeccievoice2...@gmail.com; 'Shrini'
Cc: ccie_voice@onlinestudylist.com
Subject: RE: [OSL
After I just agreed with you! J
Below is not the RSVP calculation. That is the LLQ bandwidth calculations.
After I reviewed my notes and figured out the value necessary, I referred to
the PG. The PG calculates the PQ bandwidth by using 1 call at 10ms and 1 call
at 20ms. I am confused
Hi Shrini,
I believe you’re correct as well, but you were detailing the RSVP BW
calculation not the LLQ which the question was asking.
Jeff
From: Shrini [mailto:linuxbos...@gmail.com]
Sent: Wednesday, January 05, 2011 3:32 PM
To: Roig Borrell, Francesc Xavier
Cc:
Hi Shrini,
CUCM will always request the worst case scenario in bandwidth first. The
easy way to do this is to increase the max bandwidth command and use the
show ip rsvp bandwidth command.
For example:
1. Increase the ip rsvp bandwidth command to 120
2.
Hi Brian,
I would change back their pattern as they are testing you on the following
concepts.
When the call comes into the GW on CUCM, the prefix values configured will
be added to the front of the number. In order to figure out how you will
prefix this you need to look at the debug
I know this doesn't make much sense, but it is done through a VM Profile.
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of study2b ccie
Sent: Thursday, December 09, 2010 10:10 AM
To: OSL
Subject: [OSL | CCIE_Voice] CFUR display For and
Actually, I don't see a gateway command.
Try adding that next time and calls should work with CUCM.
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of study2b ccie
Sent: Tuesday, December 07, 2010 6:18 PM
To: OSL
Subject: [OSL |
Randall,
When on the Service Parameters page, either click on the actual parameter
link or go to Help This Page
That is the best and easiest way to see what a specific parameter is all
about.
Jeff
From: ccie_voice-boun...@onlinestudylist.com
Randall,
There is no central location for the phone services you may need for the exam.
You’ll have to do throughout the Cisco.com documentation.
IPMA can be found in the Help provided in CUCM, just search for IPMA and look
at the checklist for the IPMA Proxy Line mode.
IPPA can be
I believe you can find that stuff in the Release Notes. I know that's where
you find the sql statement to insert the Voicemail button functionality back
into CUCM.
HTH
From: khaled Saholy [mailto:khaled_sah...@hotmail.com]
Sent: Wednesday, December 01, 2010 12:53 PM
To:
Why do you have a different tech prefix for VIA zone? I don't believe you
need a tech prefix at all for a VIA zone / CUBE. Just have your dial-peers
configured to receive what CUCM is sending.
Also, make sure that you have your allow connections commands. Do a show
gatekeeper endpoints and
I still think your problem most likely lies in the tech prefix on the CUBE.
You don't need a tech prefix and I would make the dial-peer a little more
specific. I'm not sure completely sure that this is accurate, but I would
think that having a tech prefix on CUBE of 1# would not allow you route a
I think what Randall is getting at is the fact that you would put the
h323-gateway voip bind source ip under the voice vlan interface. The
gatekeeper source is defined under the gatekeeper with the zone local
commands.
Hope this helps,
Jeff
-Original Message-
From:
Hi Rafay,
Can you send the configuration from Site C? He was referring to the
translation profile applied incoming on the voice-port on R3.
It should look something like this:
Voice-translation-rule 1
Rule 1 /^32143/ /3/
Or (I see two different called numbers below)
Voice
Hi everyone,
It's been a long journey, but it's finally over. Thanks for the many nights
where I needed your help and you all chimed in. Thank you IPExpert for your
great study materials and Vik for the final push in the 5-day bootcamp. I
would recommend anyone who is about to attempt the
Hi Jason,
In SRST you should point to a dial-peer that goes directly to CUE. Also add
the voicemail command under telephony or voice register global that
matches this pilot number.
For example:
Dial-peer voice 1 voip
Destination-pattern 3600
Session target CUE IP address
Session
All you would need to do is re-enter the bind statements if that is the case
in the future.
-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of
cciefo...@hotmail.com
Sent: Friday, November 19, 2010 10:49 AM
To:
Try the ! wildcard instead.
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of mudassar Khalid
Sent: Monday, November 15, 2010 2:51 AM
To: roger.car...@gmail.com
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] ? wild
For the first question - You need to either record a prompt in CUC or CUE,
upload to UCCX, and then use the Play Prompt step to play to the caller.
Then use the Terminate step and go to the end of the script using a label.
For the second question - Once again record the prompt and upload.
Could you also send the output from the command debug voice ipipgw when
you attempt a call. This would also help find the problem from a CUBE
standpoint.
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cristobal
Priego
Sent: Thursday,
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