I tried this for hours and there is no way (that I could find). You must
format flash to get the command
http://www.cisco.com/en/US/docs/routers/access/1800/1841/software/configurat
ion/guide/b_cflash.html#wp23144
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onli
Phone ntp reference is for SIP phones only
Sent from my iPhone
On Mar 4, 2013, at 4:42 PM, "CCIEing" wrote:
> Hello All,
>
> The following question cross my mind while doing the NTP configuration stuff..
>
> What is the difference between the Phone NTP reference configuration in the
> CCM We
Please do not take this as rudeness. I do not know anyone who messes with
this so that may be why you are not getting a reply.
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Hesham
Abdelkereem
Sent: Thursday, February 28, 2013 11:28 AM
I would rather do it on the subscriber page vs changing the template multiple
times. I think that would be faster but as always, go with whatever you
practice.
From: Chrysostomos Christofi [mailto:ch.christ...@logicom.net]
Sent: Thursday, February 28, 2013 9:07 AM
To: Cory Gray; 'Ni
Yes. I would put MOH Servers in there, Conference Bridges, and MTPs in
there. Just leave out annunciator. If required to configure any resources
you 100% know the resource you want is being used so you do not have to
worry about verifying it.
From: ccie_voice-boun...@onlinestudylist.com
[mai
I believe for that to work you have to go to the H323 Gateway configuration
page in CUCM and match up the port there. Then reset your gateway in CUCM.
I THINK. never tried it
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Brad McAllist
I had struggled with whether to match each subscriber with their correct
time zone. My GUESS is that it only matters if a Unity Connection question
involves any type of time stamp such as when the message was delivered. It
probably cannot hurt to do it as a best practice as I seriously doubt it c
n already use?
>
> Regards,
> Hugo
>
> From: ccie_voice-boun...@onlinestudylist.com
> [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cory Gray
> Sent: Sunday, January 20, 2013 5:52 PM
> To: 'sanity insanity'; ccie_voice@onlinestudylist.com
> Subject:
You probably have redirecting number outbound checked on your site a gateway.
Uncheck it, reset your gateway and let us know
Sent from my iPhone
On Feb 24, 2013, at 6:45 AM, "CISCO CCIE VOICE" wrote:
> Hi
>
> Can any one help me with Dial Plan consideration when calling from HQ Site to
> B
My guess is that your calling number was expanded from 4 digits to something
bigger because of you long distance dial peer. Choice one is to use alternate
extensions in Unity connection. If you are not allowed to do that change the
calling number mask on the hunt pilot to
Sent from my iP
Some correct if I am wrong but I believe MGCP drops it automatically. It is
transmitted to H323 but the default "dial-peer terminator" is # so that is how
it works there
Sent from my iPhone
On Feb 21, 2013, at 4:32 AM, "Jamie Parr (jamparr)" wrote:
> Hi all
>
> When configuring fast dialli
I use unsolicited for both. Of course I do not know whether it is right or
not though.
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Pixar Perfect
Sent: Tuesday, February 19, 2013 11:06 PM
To: CCIE Voice OSL
Subject: [OSL | CCIE_Voice
I
am assuming have destination patter 3001 and 3002
Hope this helps.
-Original Message-
From: Hesham Abdelkereem [mailto:heshamcentr...@gmail.com]
Sent: Sunday, February 17, 2013 6:16 PM
To: Cory Gray
Cc: 'ccie_voice'
Subject: Re: [OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUC
Translate called 1
-Original Message-
From: Hesham Abdelkereem [mailto:heshamcentr...@gmail.com]
Sent: Sunday, February 17, 2013 5:56 PM
To: Cory Gray
Cc: ccie_voice
Subject: Re: [OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUCM and CME issue
Yes i am using g729 and i configured them from
t;
wrote:
> I did that and allow connections as well
>
> On Feb 17, 2013, at 3:21 PM, Cory Gray wrote:
>
>> Not sure if this is what is breaking it but you should not have voice class
>> h323 1 on your ras dialpeer on site c
>>
>> Sent from my iPho
Not sure if this is what is breaking it but you should not have voice class
h323 1 on your ras dialpeer on site c
Sent from my iPhone
On Feb 17, 2013, at 4:59 PM, "Hesham Abdelkereem"
wrote:
> Dear All,
>
>
> I have tried to configure a gatekeeper between HQ-SC for interoperability
> betw
You have a right to worry. Default is default setting for built-in bridge.
The default is off. Barge is part of default remote in use. I would add
cbarge and not mess with anything else UNLESS explicitly told or some wording
points you in that direction.
Sent from my iPhone
On Feb 17, 2013
I have had several conversations with people on this. Everyone can easily
make SRST work but scoring points seems to be the trickiest thing in the
lab. So I do not think anyone knows for sure what should or should not be
on the "template" I have never scored any points there so I cannot give an
Neither have I
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jason Lee
Sent: Friday, February 15, 2013 9:51 AM
To: Pixar Perfect
Cc: CCIE Voice OSL
Subject: Re: [OSL | CCIE_Voice] ISDN signaling config
I have never done this. Anyo
It is permitted during the lab but I do not know of anyone who uses it.
Some use GUI for CUE but I cannot see how it would save you time for CME.
If you feel it does, go for it!
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ie ravindra
ccie_voice-boun...@onlinestudylist.com
>> [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of virajith
>>
>> Sent: Wednesday, February 13, 2013 7:29 AM
>> To: Cory Gray
>> Cc: ccie_voice@onlinestudylist.com
>>
>> Subject: Re: [OSL | CCIE_Vo
Do you have “network-clock-select 1 t1 0/0/0”?
From: vir...@rediffmail.com [mailto:vir...@rediffmail.com]
Sent: Wednesday, February 13, 2013 8:29 AM
To: Cory Gray
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CME Meet me conference not working
hi Cory ,
Here...
R3
Also
voice-card 0
dspfarm
dsp services dspfarm
Should be
voice-card 0
no dspfarm
dsp services dspfarm
From: vir...@rediffmail.com [mailto:vir...@rediffmail.com]
Sent: Wednesday, February 13, 2013 8:19 AM
To: Cory Gray
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL
I have never used “conference drop-mode local” before. What does that do?
Does your “Show sdspfarm X” show the bridge is registered? Paste it here
From: vir...@rediffmail.com [mailto:vir...@rediffmail.com]
Sent: Wednesday, February 13, 2013 8:19 AM
To: Cory Gray
Cc: ccie_voice
Did you add “meet me” to the seized layout?
1.Seize the line
2. Press “meet me” softkey
3. Dial 4321 to create the conference
4. Now anyone dialing 4321 should be in the conference
This is all assuming the CFB is registered but your config looks good.
From:
Can you install the CRS CD on a Non-MCS Server? Or is there another method
to make CRS believes your VM is a MCS Server?
From: Ravindra Lakpriya [mailto:lakpr...@gmail.com]
Sent: Monday, February 11, 2013 9:19 AM
To: Cory Gray
Cc: Todd Carswell, (tcarswel); ccie_voice@onlinestudylist.com
My method
1. Build a VM for Windows Server 2003. Because I am cheap and do not have a
license, I have to rebuild every 30 days because that is the Windows trial
period.
2. Once built, install IIS
3. Google and find the registry file that makes the VM appear as a Cisco
MCS7845 Server
4. Now that y
There has been a lot of speculation about Video. Will the next version of
CCIE Voice include Video and be called something like CCIE Collaboration or
would there be a completely new Video Track. I always thought there would
be a new video track because there is just so much on the video side, one
I can see 30 minutes if that is 30 minutes from the start of lab. I take notes
of IPS and things like that before I start so i do not get started right away.
But if it takes that long just to configure, you do need to speed up.
I do dhcp before vlans. This is a time saver for me. Because the
You need to transfer h323 to sip on SiteC
Voice services voip
Allow connections h323 to sip
Make sure your Site C Gwy register to Site A GK as H323-GWY instead of
VOIP-GWY
No gateway
Gateway
You do need a transcoder and it looks like you have it configured mostly
correct
You do not ne
I use this method to prevent having to do any additional dial peers in H323.
If only one site is using AAR
MCGP - strip as may digits as you need on the AAR Route Pattern to match the
called party requirements of the lab and use external mask to satisfy the
calling party requirements
H323 -
Ben,
If you can afford it, keep going. I took it 2 times in 2010, 3 times in 2012,
and passed last week for a total of 6. I am not ashamed of that. It is hard!
I was sinking so much time and money into it, I thought about quitting. This
was the first one where I thought I passed when I le
I noticed this behavior as well. Not sure if there a bug id on it. I still do
it on max-dn because I became very paranoid about how the exam is graded so i
put it on just in case the grading script looks for it. I doubt it but you
never know
Sent from my iPhone
On Jan 30, 2013, at 9:22 PM,
It should be pretty clear whether to use it or not
Sent from my iPhone
On Jan 30, 2013, at 6:07 AM, "Suresh Bhandari" wrote:
> Again it depends on if you are asked to do so.
>
>
> On Wed, Jan 30, 2013 at 3:56 PM, ie ravindra wrote:
>> Hi All,
>> Do we need to enable ip rsvp bandwidth comman
Jan 29, 2013 at 4:01 PM, Cory Gray wrote:
>> Here is mine
>>
>> Variable
>> 1 document variable.
>>
>> Creat a script from scratch
>> Start
>> Contact > Accept step
>> Media > recording step pointing to your document variable
&g
Here is mine
Variable
1 document variable.
Creat a script from scratch
Start
Contact > Accept step
Media > recording step pointing to your document variable
End
Recording will copy to wfavvid > temp
Sometimes it creats 2 files
Delete the 1k file
Rename the remaining file to something easy to
It will take up more resources to add g729r8 to transcoder and conference. I
do it just so I never have to think about it. I believe the only time it is
required is when you are using RSVP but unless the lab told use to restrict dsp
use, i would use it. The way you described mtp and codec pre
Congrats again Bill.
Piggybacking off of your excellent point. The difference between my passing
lab and last lab were two things.
One, I got myself ready for any question. All of my customers run CUCM and
Unity/Connection or that is at least the only thing they have questions
about and
This is how I would do what it is you are trying to do.
dial-peer voice 900 pots
translation-profile outgoing LOCAL (create a voice translation for calling
and called number)
destination-pattern 9[2-9]..$ ($ not needed but wont hurt you)
forward-digits 7 (only send the last 7 digits)
por
I would recommend all of the labs in the 5 lab handbook. You will
understand how to setup any type of gateway and meet any dial plan
requirements. Once you understand that then you will want to decide whether
it's a chart, notepad, written, etc. that works best for you to get through
it quickly.
voice lab.
I do not think the lab is passable without OWLE books. People passed
without TSHOOT class but I would not have.
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cory Gray
Sent: Thursday, January 24, 2013 4:35 PM
To
All,
I finally passed my CCIE Voice in RTP yesterday. I have an R&S so I am
still #22842. My two cents are below. Take them or leave them. Please
limit direct questions and post them to the alias so everyone can benefit.
Please do not ask me anything that would be in violation of the
Non-Di
All,
There have been some questions about the fact the when you do an access-list
on a 3750 and attach it to a class-map for QoS purposes, the "show
access-list" command does not show hits on the ACL. I did some research and
that is how the switch works.
http://www.cisco.com/en/US/docs/swi
20x212x16/logo.png
tftp-server flash:Desktops/320x212x16/logo-tn.png
telephony-service
reset all
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cory Gray
Sent: Sunday, January 20, 2013 8:47 PM
To: 'sanity insanit
Go to the phone Admin guide - Maintain and Operate > Maintain and Operate
Guides > Admin Guide for 7.0 > Customizing the Cisco Unified IP Phone
Under the Configuring a Background image it shows 216 as the middle
directory but it is 212 as listed slightly higher up
Create List.xml from the Guid
Two. One for PRI and Transcoding and one for conferencing because that takes
up entire dsp despite the number of sessions you configure
Sent from my iPhone
On Jan 14, 2013, at 8:18 PM, "Martin Lopez" wrote:
> Hi Team
>
> I'm newest on this group.
> In a kindly way, I'm asking if somebody co
Always use hardware resources and dont forget for CME phones, there is a live
record softkey
Sent from my iPhone
On Jan 14, 2013, at 1:47 PM, "CCIEing" wrote:
> Hello Guys,
>
> The following question passed my mind while I am practicing the LiveRecord
> feature under the CUE topic.
>
> What
You do not have to fast but your technique must be fast. You must have a good
template for every subject so you can just focus on the specific task in your
given lab. This will enable you to get to the end of lab with between 1.5 and
2.5 hours left. Depending on how well you think you did is
Sorry did not click the link you had to the support page...
In Unity Connection, Look under the "Maintain and Operate" docs for admin guides
Sent from my iPhone
On Jan 14, 2013, at 11:32 AM, "Cory Gray" wrote:
> The doc cd was retired from the lab many years ago.
The doc cd was retired from the lab many years ago. It is now the support
pages with the three windows for you to navigate through.
Sent from my iPhone
On Jan 14, 2013, at 11:18 AM, "CCIEing" wrote:
> Hi All,
>
> I was practicing the cisco Documentation CD that will be available during the
You never know why your score is so low when you fail. Even if you know you
did not have enough points to pass, you look at certain sections and wonder
what went wrong. There is no good explanation. When you pass you do not
get a score report at all so you never REALLY know. For Networking you
Todd. Everything is 7.0/7.0.1.
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Todd Carswell
(tcarswel)
Sent: Thursday, January 10, 2013 12:59 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Software Versions
The
I will say this. This happened to me once using MGCP and could never figure
it out. I just assumed "something was messed up" and could not find an
answer in over 2 hours of troubleshooting. If you ever figure it out,
please let me know.
-Original Message-
From: ccie_voice-boun...@online
End user on the DN pages is only needed for presence. End user phone
association and primary extension is needed to import users into CUE and CUC as
well as other features.
If DHCP is not working I doubt manual will work because both rely on two way
communication with CUCM and the voice vlans.
Without that parameter you should at least see the DN of the Unity Port as the
calling number. Are you doing a SIP integration? I am not sure what would be
the calling number in that situation.
Sent from my iPhone
On Jan 4, 2013, at 9:14 PM, "William Bell" wrote:
> Check CUCM service parame
If I hear you correctly, you have SiteC Router as the DHCP Server for SiteC
and you are using the CUCM-PUB as the DHCP Server for SiteA and SiteB. If
that is the case, you will need "ip helper-address X.X.X.X" under the voice
subnet's interface pointing to the PUB. Your scope looks good assuming
Good Catch Steffen. Rookie mistake. I had my zone prefix misconfigured.
From: Steffen Bruening [mailto:stbruen...@gmail.com]
Sent: Friday, January 04, 2013 11:45 AM
To: Cory Gray
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] GK to CUE Direct Call
Did you see the
All,
I noticed during the following setup that I cannot call DIRECTLY into CUE
but if SiteC-Phone1 is busy or does not answer, forwarding the voicemail
works. I am not even sure you would lose points for this but I always like
to make sure things like this work.
Setup is as follows.
CU
For CUC, I would use Greetings Administrator
For CUCCX I would use the recording script. It takes 2 seconds to make.
Because they are Call in and record methods, they are guaranteed to work. I
cannot imagine the lab telling you how to record your prompt. I would think
either it would be
You get points for questions in a particular section. If there is an OB
Unity Question in the Unity Section and SRST in the SRST section then no.
You could theoretically get a 100% on call routing and get a 0% on those
other two sections.
-Original Message-
From: ccie_voice-boun...@online
IF CUC and CUCCX were already integrated for you, then dial the hunt pilot on
CUC and CTI route point for CUCCX. Presence is not dial able so not sure
there. You would have to have a phone registered first though. I just cross
my fingers that they will work for me when I get there because in
Speaking strictly for RTP….
Driver’s license OR passport. Photocopy of anything would probably be
rejected. You can bring your lunch but it is provided so you are at the mercy
of whatever they bring in for the day. You are allowed to bring water, energy
drinks, etc to your desk while you
Is your authentication url in enterprise parameters pointed to the IP of the
PUB instead of the hostname? That was one issue I had.
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of SAIKAT SEN
Sent: Wednesday, December 19, 2012 2:35 PM
To
I had this 1 time before in practice and I had the reboot my CUPS server.
Nothing was working correctly on my CUPC client.
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Carlosobs _
Sent: Wednesday, December 19, 2012 5:02 AM
To: vir...
The license file is different for that module. If you have a CCO (Cisco)
account you can download those licenses at no cost
Sent from my iPhone
On Nov 27, 2012, at 12:35 PM, "Chrysostomos Christofi"
wrote:
>
> Hi folks
>
> I have the cue license for proctor labs
>
> I am wondering if I
As far as I know, you must initialize through the cli guide and after that you
can configure it through the gui or cli. Some people say learn cli because
its faster but I prefer the gui. If you find some documentation that shows the
initialization through the Gui let us know. The cli initial
Your mls qos map cos-dscp 0 8 16 24 32 46 48 56 maps cos 5 to dscp 46 aka dscp
ef. I am confused by your question/solution because you are configuring cos 5
and dscp ef but the question is asking about cos 3
Sent from my iPhone
On Nov 22, 2012, at 6:08 AM, "virajith " wrote:
> mls qos map co
The answer is to any question like this is that you can do whatever you want as
long as your method is not told to you by the lab question and your method also
does not violate the do's and dont's stated at the beginning of the lab
Sent from my iPhone
On Nov 22, 2012, at 9:26 AM, "Chrysostomos
Yes
Sent from my iPhone
On Nov 11, 2012, at 10:16 AM, "Chrysostomos Christofi"
wrote:
> Hi
>
> For IPPM the pin is from the end user configuration tab (PIN?)
>
>
> Regards
>
> ___
> For more information regarding industry leading CCIE Lab tr
Because this was said at Cisco Live it is not breaking NDA to say there are no
SIP Phones in the lab. That leaves SIP trunking on the table but I cannot
think of a situation where you would need SIP CUCME commands.
CUE can be registered to CUCME or CUCM. If CUCM and you go into SRST or
CME-SR
Pri-group timeslots 1-13 (h323)
Pri-group timeslots 1-13 service mgcp (mgcp)
I leave out the D channel in the config because IOS will automatically add
it for you. When you do a show run you will see it. The reason why I do
this is because on the controller, 1 is channel 1, but on the voice
Messing with NTP and disconnecting and reconnecting my PUB to my network has
caused similar issues in the past. Now I do not configure NTP on the PUB
until I am in the real lab. I tried all recovery methods and reboots. I
ended up just starting from scratch because I could not get my PUB and SUB
2012 9:16 AM
To: Cory Gray
Cc: vir...@rediffmail.com; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] ESW and switch config
I still need the sw access mode, because there is no "spanning-tree portfast
trunk" command for the ESW and so I would lose the marks for t
I had the same issue when I started the lab. I had never even configured a
UCCX integration in my life let alone a customer script. I did not find any
step by step guides out there that really explained the in outs. This is my
recommendation
1. Build your own from scratch. I don't hav
AN on switch access ports; voice VLAN is not
supported on trunk ports. You can only configure a voice VLAN on Layer 2 ports."
http://www.ciscosystems.com/en/US/docs/switches/metro/catalyst3750m/software/release/12.2_25_ey/configuration/guide/swvoip.html#wp1030825
2012/11/1 Cory Gray
You
You should be fine without one just know that the recommended ESW config is
Switchport mode trunk
Switch trunk native vlan X (X equals data vlan)
Switch voice vlan Y (y equals voice vlan)
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf
so that justification will be done for the right candidates.
>
> Thank you
> krishna.
>
>
> From: Cory Gray
mailto:corygray22...@hotmail.com>>
> To: 'Krishna' mailto:vinayak_...@yahoo.com>>;
'Online Study'
mailto:ccie
gmentation header?
Thanks,
Craig
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cory Gray
Sent: Wednesday, October 31, 2012 9:52 AM
To: 'Krishna'; 'Online Study'
Subject: Re: [OSL | CCIE_Voice] Cisco ri
all at the mercy of scoring and it
is what it is so you will have to decide for yourself whether to go back.
From: Krishna [mailto:vinayak_...@yahoo.com]
Sent: Wednesday, October 31, 2012 10:37 AM
To: Cory Gray; 'Online Study'
Subject: Re: [OSL | CCIE_Voice] Cisco ripped me off
C
From: Krishna [mailto:vinayak_...@yahoo.com]
Sent: Wednesday, October 31, 2012 10:14 AM
To: Cory Gray; 'Online Study'
Subject: Re: [OSL | CCIE_Voice] Cisco ripped me off
Cory,
Technically speaking, the grading has to be evaluated by taking the seating
position where we took the exam rath
Krishna,
I am sorry to hear that. I suffered something similar during my last
attempt but after much thinking I think I know what happened and maybe the
same happened to you.
Even though IPexpert recommends using switchport mode trunk on ESW
interfaces I still had been using switch mode ac
Also for 4 G711 calls you need your ip rsvp bandwidth to be 336
80k per call = 320k
1 call ringing in at a time = 16 k
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vikky Kumar
Sent: Monday, October 29, 2012 5:44 AM
To: virajith
Ramcharan,
I have never tried AVT so I will leave that one alone. For MWI I would
recommend using MWI Subscribe. If you need that config let me know. I did
test outcalling once and it took a while for me to get it to work. I
eventually did a debug ccsip messages and saw that CUE was using a di
any eq 2427 any
class-map match-any c-mgcp
match access-group name 101
policy-map p-mgcp
class c-mgcp
set dscp cs3
police 64000 8000 exceed-action drop
int fa 1/0/1 ---> trunk port to router
mls qos trust dscp
service-policy input p-mgcp
_____
From: Cory Gray
To: Kevin Spicer
Cc
I am having the same error however 1 of my 4 phones is getting the
screenshot correctly. I am going to try to run it on a different PC today
and see if it works. Did you find a fix?
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mann
Complete shot in the dark. I am pretty sure moh uses dsp resources. Remove
the moh commands and try again. If that does not, keep the commands removed
and reload the router.
Sent from my iPhone
On Oct 26, 2012, at 1:02 PM, "Nicolas MICHEL" wrote:
> Same result :(
>
> Le Friday, October 26
If you paste your config, we can be of better help
Sent from my iPhone
On Oct 24, 2012, at 12:23 PM, "Kevin Spicer" wrote:
> This is on 3750 switch? Did you enable qos globally? (mls qos)
>
> On 24 Oct 2012 17:03, "Krishna" wrote:
>> i created an acl that calls mgcp ports i.e. udp 2427 & 24
the agents ( in this case gateways) that decide BW allocation .
>
> Hence why is there a need to specify " Mandatory" in locations on the
> callmanager?
>
>
> -Vir
>
>
>
> From: Cory Gray
> Sent: Mon, 22 Oct 2012 18:49:20
> To: "'vi
Just also notices you are using a VTGO softphone and I am not familiar with
that. I would download cisco ip communicator if possible.
-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cory Gray
Sent: Monday
Are you using virtual servers? If you virtual server does not have enough
resources to run, that is a symptom I have seen before. But you definitely
have to check your PC Performance, Server performance, and I am assuming
your network connections are good. It is most likely a connectivity issue.
My responses in line.
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of virajith
Sent: Monday, October 22, 2012 2:53 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Fews questions on RSVP config in lab
Hi Guys,
If
Steffen,
Just to clarify what Krishna was saying. Here is the command reference
mls qos queue-set output qset-id threshold queue-id drop-threshold1
drop-threshold2 reserved-threshold maximum-threshold
There is no threshold 3 configuration because it is 100% and not
configurable. Where yo
Randall,
I believe the question is when going through the GUI initialization and it ask
for the IP address of CME, what IP address do you use?
Krishna,
Please correct me if I am wrong.
Sent from my iPhone
On Oct 20, 2012, at 9:06 PM, "Rrcrumm" wrote:
> The labs say to use an IP address if 1
I noticed this and I believe it is a typo further evidenced by the note next to
the command that says something to the effect that that command moves it to q1t1
Sent from my iPhone
On Oct 20, 2012, at 2:17 AM, "Pixar Perfect" wrote:
> The requirement is as follows on Lab 2 QOS section of the 5
Do you have g729r8 on the transcoder? It is not their by default so you would
have to add it to the list. I always add it to conferencing and transcoder
just to get in the habit.
Sent from my iPhone
On Oct 20, 2012, at 4:32 PM, "Steffen Bruening" wrote:
> Hi,
>
> I have 3 Sites, all of th
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