Re: [OSL | CCIE_Voice] Directory folder on Router

2013-03-12 Thread Cory Gray
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...@onlinestudylist.com] On Behalf Of CISCO CCIE
VOICE
Sent: Tuesday, March 12, 2013 10:51 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Directory folder on Router

 

Hi Experts,

 

can any one help me I want to create directory folder on router without
formatting flash  when i use mkdir command its saying that invalid input 

 

thnks

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Re: [OSL | CCIE_Voice] Phone NTP Reference Vs NTP Reference

2013-03-04 Thread Cory Gray
Phone ntp reference is for SIP phones only

Sent from my iPhone

On Mar 4, 2013, at 4:42 PM, CCIEing aboaz...@gmail.com 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 Web administration page 
 and
 The NTP reference on the OS Administration page??
 
 does the 1st one for the endpoints where the 2nd one is for the CUCM itself?
 
 Thanks 
 
 
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Re: [OSL | CCIE_Voice] Unity Connection user template time format

2013-02-28 Thread Cory Gray
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 can
hurt your scoring but you never know so you have to decide what is best.

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Nicolas MICHEL
Sent: Thursday, February 28, 2013 6:45 AM
To: Jamie Parr (jamparr)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Unity Connection user template time format

 

Does CUCN has something to do with the display of the phone ? :=)




Le 2/28/2013 12:07 PM, Jamie Parr (jamparr) a écrit :

Hi all

 

If we are instructed to display the phones time in 24 hour format, should we
reflect this in the user templates for Cisco Unity?

 

Thanks

 


http://www.cisco.com/web/europe/images/email/signature/logo06.jpg


Jamie Parr
Engineer - IT
 mailto:jamp...@cisco.com jamp...@cisco.com
Phone: +44 20 8824 2641
Mobile: +44 7590622049




Cisco Systems
9-11 New Square
Bedfont Lakes
Feltham
Middlesex
TW14 8HA
United Kingdom
 http://cisco.com www.cisco.com



 



 

 






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Re: [OSL | CCIE_Voice] BACD w/ custom h225 port

2013-02-28 Thread Cory Gray
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 McAllister
Sent: Thursday, February 28, 2013 8:00 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] BACD w/ custom h225 port

 

All,

I recently completed the tasks for WB1 Lab 5a. Afterwards I decided to
configure BACD on BR2. After spending hours trying to figure out why I
received a fast busy immediately after dialing into my aa script, I found
the h225 setting from lab5 was the culprit! See below:

voice service voip 
 h323
  h225 listen-port 1820

As soon as I remove the h225 listen-port 1820, BACD works fine.

Is this expected behavior? Is there a way to make BACD work when the h225
listen-port is modified?

Thanks,

- Brad

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Re: [OSL | CCIE_Voice] MRG and MRGL

2013-02-28 Thread Cory Gray
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
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ben John
Sent: Thursday, February 28, 2013 8:35 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] MRG and MRGL

 

When i configure MRG and MRGL for a site ( HQ or Site B or Site C) i
configure another MRG and MRGL i name it something like DEFAULT and put the
unused Media Resources in there so no one can access them. Is this a good
practice ?

Ben

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Re: [OSL | CCIE_Voice] Unity Connection user template time format

2013-02-28 Thread Cory Gray
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; 'Nicolas MICHEL'; 'Jamie Parr (jamparr)'
Cc: ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] Unity Connection user template time format

 

Guys

 

Take it logically

 

If HQ site has different time zone with Site B then for sure the users in CUC 
must have the correct time zone for each branch

 

1)  User template in CUC (modify there anything you want include time 
zone),Import HQ users

2)   Then modify again the user template to the correct time zone for users 
in site B and then import the users for site B

 

 

Regards

 

 

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cory Gray
Sent: Πέμπτη, 28 Φεβρουαρίου 2013 2:53 μμ
To: 'Nicolas MICHEL'; 'Jamie Parr (jamparr)'
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Unity Connection user template time format

 

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 can hurt your 
scoring but you never know so you have to decide what is best.

 

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Nicolas MICHEL
Sent: Thursday, February 28, 2013 6:45 AM
To: Jamie Parr (jamparr)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Unity Connection user template time format

 

Does CUCN has something to do with the display of the phone ? :=)




Le 2/28/2013 12:07 PM, Jamie Parr (jamparr) a écrit :

Hi all

 

If we are instructed to display the phones time in 24 hour format, should we 
reflect this in the user templates for Cisco Unity?

 

Thanks

 


http://www.cisco.com/web/europe/images/email/signature/logo06.jpg


Jamie Parr
Engineer - IT
 mailto:jamp...@cisco.com jamp...@cisco.com
Phone: +44 20 8824 2641
Mobile: +44 7590622049




Cisco Systems
9-11 New Square
Bedfont Lakes
Feltham
Middlesex
TW14 8HA
United Kingdom
 http://cisco.com www.cisco.com



 



 

 






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Re: [OSL | CCIE_Voice] How to edit and overwrite .cnf.xml file on CUCM? (3rd Time)

2013-02-28 Thread Cory Gray
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
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] How to edit and overwrite .cnf.xml file on CUCM?
(3rd Time)

 

Dear All,

I would like to download .cnf.xml for a specific phone so that I can edit
it's Directories button for that particular phone only.
However , I can do the following http://cucmip:6970/phonemac.cnf.xml
I click on it --- refresh save as and I can get it and edit it fine.
But there is a big problem when i edit it and upload it back to CUCM nothing
happens
I did the following
Service Parameters  CISCO TFTP ---Advanced -- Build CNF Files (Build
All) then Enable Caching of constant and bin (false).

Then I have went to OS Admin --- TFTP upload then i uploaded with /
directory
Then i restarted TFTP and restarted the phone then nothing happened.

Please give me some advice this is very important for you to beat the CCIE
lab phone customization so quickly and efficiently

 

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Re: [OSL | CCIE_Voice] Dail Plan Consideration in SRST Mode

2013-02-24 Thread Cory Gray
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 ccievoic...@gmail.com wrote:

 Hi 
 
 Can any one help me with Dial Plan consideration when calling from HQ Site to 
 Branch 1 Site,following what i have configure.But the problem is that on 
 B1PH1 screen  its showing  as below 
 
 From +14082021001
 Forward by:2001
 
 
 HQ SITE:
 
 Extension Range:1XXX
 
 Partition:Branch_1_SRST_PT
 CSS :Branch_1_SRST_CSS--contains-Branch_1_SRST_PT
 
 B1PH1:i have assign CFUR as 2001 with CSS:Branch_1_SRST_CSS
 
 
 Route Pattern: 2XXX--Branch_1_SRST_PT
 Route List  : Standard Local Route Group
 Prefix :91972303
 
 
 
 BRANCH 1 SITE:
 
 Extension Range :2XXX
 
 dial-peer voice 10 pots
 destination-pattern 1...
 perfix 14082021
 port 0/0/0:23
 
 thanks
 
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Re: [OSL | CCIE_Voice] Custom Background image for phones registered to cme

2013-02-24 Thread Cory Gray
Not sure.  I created my own in mspaint.  Mspaint on windows 7 allows to specify 
pixel height and width explicitly so it is very easy to do.

Sent from my iPhone

On Feb 24, 2013, at 12:20 PM, Barrera, Hugo hugo.barr...@nexusis.com wrote:

 Does proctor labs have any background images that we can 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: Re: [OSL | CCIE_Voice] Custom Background image for phones registered 
 to cme
  
 A little more detail
  
 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 your Files
 Create List.xml from the Guide
 Create Full size image—320 pixels (width) X 212 pixels (height)
 Create Thumbnail image—80 pixels (width) X 53 pixels (height)
  
 Upload the Files to CUCM
 Upload them to CUCM using TFTP File Management
 Restart TFTP Service
  
 Create Directories on the Router
 mkdir flash:Desktops
 mkdir flash:Desktops/320x212x16/
  
 Download the files from CUCM to the Router
 copy tftp flash:Desktops/320x212x16/ (for all three files)
  
 Serve all three files on the Router
 tftp-server flash:Desktops/320x212x16/List.xml
 tftp-server flash:Desktops/320x212x16/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 insanity'; ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] Custom Background image for phones registered 
 to cme
  
 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 Guide
 Create Full size image—320 pixels (width) X 212 pixels (height)
 Create Thumbnail image—80 pixels (width) X 53 pixels (height)
  
 Upload all three to the router
 mkdir flash:Desktops
 mkdir flash:Desktops/320x212x16/
 copy tftp flash:Desktops/320x212x16/ (for all three files)
  
 Serve all three files
 tftp-server flash:Desktops/320x212x16/List.xml
 tftp-server flash:Desktops/320x212x16/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 sanity insanity
 Sent: Sunday, January 20, 2013 8:22 PM
 To: ccie_voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice] Custom Background image for phones registered to 
 cme
  
 Guys,
 
 What is the easiest way of getting a customized background image and 
 configuring it for cme  ip phones . Given that there is no TFTP server , we 
 just need to use the callmanger . Please illustrate the steps...
 
 Regards,
 Vir
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Re: [OSL | CCIE_Voice] VM in SRST

2013-02-23 Thread Cory Gray
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 iPhone

On Feb 23, 2013, at 2:09 PM, Ben John benjoh...@hotmail.com wrote:

 I configured my VM when in SRST when i press the message button it goes to 
 the tutorial instead of asking for the pin. Bellow is my config. Any feedback 
 why it is going to the tutorial ?
 
 
 call-manager-fallback
  secondary-dialtone 9
  max-conferences 8 gain -6
  transfer-system full-consult
  timeouts interdigit 3
  ip source-address 10.10.65.254 port 2000
  max-ephones 9
  max-dn 9 octo-line preference 9 no-reg
  transfer-pattern .T
  voicemail 2220
  huntstop channel 1
  call-forward pattern .T
  call-forward busy 2220
  call-forward noan 2220 timeout 20
  moh music-on-hold.au
  multicast moh 239.1.1.1 port 16384 route 10.10.65.254 10.10.110.2 
 142.102.65.254 142.1.65.254
  time-zone 8
  time-format 24
 
 dial-peer hunt 2
 num-exp 2220 912025552220
 
 
 Ben
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Re: [OSL | CCIE_Voice] Fast dialling

2013-02-21 Thread Cory Gray
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) jamp...@cisco.com wrote:

 Hi all
  
 When configuring fast dialling, do we need to configure the patterns to drop 
 the trailing #
  
 The calls seem to leave the gateway looking the same either way?
  
 Thanks
  
 image001.jpg
 Jamie Parr
 Engineer - IT
 jamp...@cisco.com
 Phone: +44 20 8824 2641
 Mobile: +44 7590622049
 
 
 Cisco Systems
 9-11 New Square
 Bedfont Lakes
 Feltham
 Middlesex
 TW14 8HA
 United Kingdom
 www.cisco.com
 
  
  
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inline: image001.jpg___
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Re: [OSL | CCIE_Voice] MWI Best Practice

2013-02-20 Thread Cory Gray
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] MWI Best Practice

 

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 to the needed points. 

 

however i have seen solicited and unsolicited to be verify unreliable on
7965 phones .. you have to do no mwi sip and mwi sip to get solicited to
work and sometimes reboot CUE or router to get both solicited and
unsolicited to work. I am 1 month away from exam date and dont want to waste
time exploring instead adopt best common practice that works flawlessly
..and so far it has been ON/OFF DN

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Re: [OSL | CCIE_Voice] cBarge Barge softkey

2013-02-17 Thread Cory Gray
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, at 6:01 AM, Pixar Perfect pixarperf...@live.com wrote:

 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 conferencing.?
 
 the solution guide has an example that has the Barge softkey left there in 
 Remote In Use. Per IPEXPERT's bootcamp, the recommendation was not to tamper 
 with the existing Softkey layout and keep adding softkeys. It makes sense 
 however, this particular Barge vs cBarge is tricky thing ... i would be least 
 worried abt these things but it will be unfortunate if the script is looking 
 for Barge softkey as well :) ... the notorious grading script process 
 worries me as it is the deal breaker :)
 
 
 thx...pixar
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Re: [OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUCM and CME issue

2013-02-17 Thread Cory Gray
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 heshamcentr...@gmail.com 
wrote:

 Dear All,
 
 
 I have tried to configure a  gatekeeper between HQ-SC for interoperability 
 between CME and HQ
 The issue is I am just able to call from CME to CUCM but Unable to call from 
 CUCM to CME.
 Knowing that I have created a Device Pool , Route Pattern , Gatekeeper info , 
 Gatekeeper controlled trunk for Gatekeeper to call CME from CUCM Side
 when I debug i always get ARJ Admission Rejection.
 I don't want to change anything in the technology prefix or anything.
 I don't want to use default technology prefix.
 I want show gatekeeper endpoints and show gatekeeper gw-type-prefix to be the 
 same exactly.
 I just want to troubleshoot the issue of calling from CUCM to CME.
 Thank you so much for all your efforts
 
 
 However, here you are my configs
 
 GATEKEEPER HQ Router - SIDE
 
 voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 
 interface Loopback0
 ip address 177.1.254.1 255.255.255.255
 h323-gateway voip bind srcaddr 177.1.254.1
 
 gatekeeper
 zone local CUCM cisco.com 177.1.254.1
 zone local CUCME cisco.com
 zone prefix CUCM 1...
 zone prefix CUCM 2...
 zone prefix CUCME 3...
 gw-type-prefix 1*
 no shutdown
 
 
 
 
 SC Side
 
 interface Loopback0
 ip address 177.1.254.3 255.255.255.255
 h323-gateway voip interface
 h323-gateway voip id CUCM ipaddr 177.1.254.1 1719
 h323-gateway voip h323-id CUCME
 h323-gateway voip tech-prefix 31
 h323-gateway voip bind srcaddr 177.1.254.3
 
 
 dial-peer voice 85 voip
 destination-pattern [12]...$
 voice-class h323 1
 session target ras
 dtmf-relay h245-alphanumeric
 
 
 CorpHQ(config-dial-peer)#do show gatekeeper end
GATEKEEPER ENDPOINT REGISTRATION

 CallSignalAddr  Port  RASSignalAddr   Port  Zone Name TypeFlags
 --- - --- - - -
 177.1.10.10 1720  177.1.10.10 32811 CUCM  VOIP-GW
H323-ID: CUCM_TRUNK_1
Voice Capacity Max.=  Avail.=  Current.= 0
 177.1.10.20 1720  177.1.10.20 32788 CUCM  VOIP-GW
H323-ID: CUCM_TRUNK_2
Voice Capacity Max.=  Avail.=  Current.= 0
 177.1.254.3 1720  177.1.254.3 63360 CUCM  H323-GW
H323-ID: CUCME
Voice Capacity Max.=  Avail.=  Current.= 0
 Total number of active registrations = 3
 
 CorpHQ(config-dial-peer)#
 
 
 CorpHQ(config-dial-peer)#do show gatekeeper gw-type-prefix
 GATEWAY TYPE PREFIX TABLE
 =
 Prefix: 31*
  Zone CUCM master gateway list:
177.1.254.3:1720 CUCME
 
 Prefix: 1*
  Zone CUCM master gateway list:
177.1.10.10:1720 CUCM_TRUNK_1
177.1.10.20:1720 CUCM_TRUNK_2
 
 
 CorpHQ(config-dial-peer)#
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Re: [OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUCM and CME issue

2013-02-17 Thread Cory Gray
Should not have allow connections either unless you are doing cube but that 
should not break it.  Debug h22r ans1 and look to see if there is detail on why 
the call is failing.  Make sure you are using g729 as well

Sent from my iPhone

On Feb 17, 2013, at 5:39 PM, Hesham Abdelkereem heshamcentr...@gmail.com 
wrote:

 I did that and allow connections as well
 
 On Feb 17, 2013, at 3:21 PM, Cory Gray corygray22...@hotmail.com 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 iPhone
 
 On Feb 17, 2013, at 4:59 PM, Hesham Abdelkereem heshamcentr...@gmail.com 
 wrote:
 
 Dear All,
 
 
 I have tried to configure a  gatekeeper between HQ-SC for interoperability 
 between CME and HQ
 The issue is I am just able to call from CME to CUCM but Unable to call 
 from CUCM to CME.
 Knowing that I have created a Device Pool , Route Pattern , Gatekeeper info 
 , Gatekeeper controlled trunk for Gatekeeper to call CME from CUCM Side
 when I debug i always get ARJ Admission Rejection.
 I don't want to change anything in the technology prefix or anything.
 I don't want to use default technology prefix.
 I want show gatekeeper endpoints and show gatekeeper gw-type-prefix to be 
 the same exactly.
 I just want to troubleshoot the issue of calling from CUCM to CME.
 Thank you so much for all your efforts
 
 
 However, here you are my configs
 
 GATEKEEPER HQ Router - SIDE
 
 voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 
 interface Loopback0
 ip address 177.1.254.1 255.255.255.255
 h323-gateway voip bind srcaddr 177.1.254.1
 
 gatekeeper
 zone local CUCM cisco.com 177.1.254.1
 zone local CUCME cisco.com
 zone prefix CUCM 1...
 zone prefix CUCM 2...
 zone prefix CUCME 3...
 gw-type-prefix 1*
 no shutdown
 
 
 
 
 SC Side
 
 interface Loopback0
 ip address 177.1.254.3 255.255.255.255
 h323-gateway voip interface
 h323-gateway voip id CUCM ipaddr 177.1.254.1 1719
 h323-gateway voip h323-id CUCME
 h323-gateway voip tech-prefix 31
 h323-gateway voip bind srcaddr 177.1.254.3
 
 
 dial-peer voice 85 voip
 destination-pattern [12]...$
 voice-class h323 1
 session target ras
 dtmf-relay h245-alphanumeric
 
 
 CorpHQ(config-dial-peer)#do show gatekeeper end
  GATEKEEPER ENDPOINT REGISTRATION
  
 CallSignalAddr  Port  RASSignalAddr   Port  Zone Name TypeFlags
 --- - --- - - -
 177.1.10.10 1720  177.1.10.10 32811 CUCM  VOIP-GW
  H323-ID: CUCM_TRUNK_1
  Voice Capacity Max.=  Avail.=  Current.= 0
 177.1.10.20 1720  177.1.10.20 32788 CUCM  VOIP-GW
  H323-ID: CUCM_TRUNK_2
  Voice Capacity Max.=  Avail.=  Current.= 0
 177.1.254.3 1720  177.1.254.3 63360 CUCM  H323-GW
  H323-ID: CUCME
  Voice Capacity Max.=  Avail.=  Current.= 0
 Total number of active registrations = 3
 
 CorpHQ(config-dial-peer)#
 
 
 CorpHQ(config-dial-peer)#do show gatekeeper gw-type-prefix
 GATEWAY TYPE PREFIX TABLE
 =
 Prefix: 31*
 Zone CUCM master gateway list:
  177.1.254.3:1720 CUCME
 
 Prefix: 1*
 Zone CUCM master gateway list:
  177.1.10.10:1720 CUCM_TRUNK_1
  177.1.10.20:1720 CUCM_TRUNK_2
 
 
 CorpHQ(config-dial-peer)#
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Re: [OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUCM and CME issue

2013-02-17 Thread Cory Gray
I am sorry.  I had it backwards.  I thought you had an issue routing to
CUCM.  For call into CUCME, you need this
Dial peer voice 3000 voip
Incoming called ^3...$
Dtmf-r h245a
No vad
Translation-profile in STRIP
!
Voice translation-rule 1
Rule 1 /.+\(\)$/ /\1/
!
Voice translation-profile STRIP
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 both sides CUCM side as
region and location /devicepool and voice class codec as cme side.
I am able to send calls from CME to CUCM but cucm unable to place calls to
CME

On Feb 17, 2013, at 3:51 PM, Cory Gray corygray22...@hotmail.com wrote:

 Should not have allow connections either unless you are doing cube but 
 that should not break it.  Debug h22r ans1 and look to see if there is 
 detail on why the call is failing.  Make sure you are using g729 as 
 well
 
 Sent from my iPhone
 
 On Feb 17, 2013, at 5:39 PM, Hesham Abdelkereem
heshamcentr...@gmail.com wrote:
 
 I did that and allow connections as well
 
 On Feb 17, 2013, at 3:21 PM, Cory Gray corygray22...@hotmail.com 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 iPhone
 
 On Feb 17, 2013, at 4:59 PM, Hesham Abdelkereem
heshamcentr...@gmail.com wrote:
 
 Dear All,
 
 
 I have tried to configure a  gatekeeper between HQ-SC for 
 interoperability between CME and HQ The issue is I am just able to call
from CME to CUCM but Unable to call from CUCM to CME.
 Knowing that I have created a Device Pool , Route Pattern , 
 Gatekeeper info , Gatekeeper controlled trunk for Gatekeeper to call
CME from CUCM Side when I debug i always get ARJ Admission Rejection.
 I don't want to change anything in the technology prefix or anything.
 I don't want to use default technology prefix.
 I want show gatekeeper endpoints and show gatekeeper gw-type-prefix to
be the same exactly.
 I just want to troubleshoot the issue of calling from CUCM to CME.
 Thank you so much for all your efforts
 
 
 However, here you are my configs
 
 GATEKEEPER HQ Router - SIDE
 
 voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 
 interface Loopback0
 ip address 177.1.254.1 255.255.255.255 h323-gateway voip bind 
 srcaddr 177.1.254.1
 
 gatekeeper
 zone local CUCM cisco.com 177.1.254.1 zone local CUCME cisco.com 
 zone prefix CUCM 1...
 zone prefix CUCM 2...
 zone prefix CUCME 3...
 gw-type-prefix 1*
 no shutdown
 
 
 
 
 SC Side
 
 interface Loopback0
 ip address 177.1.254.3 255.255.255.255 h323-gateway voip interface 
 h323-gateway voip id CUCM ipaddr 177.1.254.1 1719 h323-gateway voip 
 h323-id CUCME h323-gateway voip tech-prefix 31 h323-gateway voip 
 bind srcaddr 177.1.254.3
 
 
 dial-peer voice 85 voip
 destination-pattern [12]...$
 voice-class h323 1
 session target ras
 dtmf-relay h245-alphanumeric
 
 
 CorpHQ(config-dial-peer)#do show gatekeeper end
 GATEKEEPER ENDPOINT REGISTRATION
 
 CallSignalAddr  Port  RASSignalAddr   Port  Zone Name Type
Flags
 --- - --- - - 
-
 177.1.10.10 1720  177.1.10.10 32811 CUCM  VOIP-GW
 H323-ID: CUCM_TRUNK_1
 Voice Capacity Max.=  Avail.=  Current.= 0
 177.1.10.20 1720  177.1.10.20 32788 CUCM  VOIP-GW
 H323-ID: CUCM_TRUNK_2
 Voice Capacity Max.=  Avail.=  Current.= 0
 177.1.254.3 1720  177.1.254.3 63360 CUCM  H323-GW
 H323-ID: CUCME
 Voice Capacity Max.=  Avail.=  Current.= 0 Total number of active 
 registrations = 3
 
 CorpHQ(config-dial-peer)#
 
 
 CorpHQ(config-dial-peer)#do show gatekeeper gw-type-prefix GATEWAY 
 TYPE PREFIX TABLE =
 Prefix: 31*
 Zone CUCM master gateway list:
 177.1.254.3:1720 CUCME
 
 Prefix: 1*
 Zone CUCM master gateway list:
 177.1.10.10:1720 CUCM_TRUNK_1
 177.1.10.20:1720 CUCM_TRUNK_2
 
 
 CorpHQ(config-dial-peer)#
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Re: [OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUCM and CME issue

2013-02-17 Thread Cory Gray
With CUBE, there is no tech prefix so that is why you don't need it here.
Based on your config, I am assuming your CUCME phones are 3XXX.  That strip
pattern (taught by IPexpert) will take the last 4 digits of any inbound
call.  
H323 has two legs.
1.  Inbound Call - which reminds me... needs to be ^313...$ because Site A
GK will send the tech-prefix to Site C Gateway (your output shows 31 as the
tech prefix for Site C)
2.  Outbound Call - now that you have accepted the call on dial peer 3000
(or whatever you decided to use) Site C Gateway will look to make another
call out based on destination-pattern.  Normally the call would be made to
313 but we will use the stip translation rule to make it 3XXX before
trying to make the call

Where is destination pattern 3XXX?
You hidden CUCME dial-peers is where.
Show voice dial-peer summary will show your hidden CUCME dial-peer which 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 CUCM and CME issue

Thank you so much for your efforts.
I believe it may need a strip but i don't know exactly what or how to strip
the prefix as with CUBE it works without need for translation rule.

Thanks for info i will try and feed you back.

Thanks,
Hesham
On Feb 17, 2013, at 4:08 PM, Cory Gray corygray22...@hotmail.com wrote:

 I am sorry.  I had it backwards.  I thought you had an issue routing 
 to CUCM.  For call into CUCME, you need this Dial peer voice 3000 voip 
 Incoming called ^3...$ Dtmf-r h245a No vad Translation-profile in 
 STRIP !
 Voice translation-rule 1
 Rule 1 /.+\(\)$/ /\1/
 !
 Voice translation-profile STRIP
 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 both sides CUCM side as 
 region and location /devicepool and voice class codec as cme side.
 I am able to send calls from CME to CUCM but cucm unable to place 
 calls to CME
 
 On Feb 17, 2013, at 3:51 PM, Cory Gray corygray22...@hotmail.com wrote:
 
 Should not have allow connections either unless you are doing cube 
 but that should not break it.  Debug h22r ans1 and look to see if 
 there is detail on why the call is failing.  Make sure you are using 
 g729 as well
 
 Sent from my iPhone
 
 On Feb 17, 2013, at 5:39 PM, Hesham Abdelkereem
 heshamcentr...@gmail.com wrote:
 
 I did that and allow connections as well
 
 On Feb 17, 2013, at 3:21 PM, Cory Gray corygray22...@hotmail.com
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 iPhone
 
 On Feb 17, 2013, at 4:59 PM, Hesham Abdelkereem
 heshamcentr...@gmail.com wrote:
 
 Dear All,
 
 
 I have tried to configure a  gatekeeper between HQ-SC for 
 interoperability between CME and HQ The issue is I am just able to 
 call
 from CME to CUCM but Unable to call from CUCM to CME.
 Knowing that I have created a Device Pool , Route Pattern , 
 Gatekeeper info , Gatekeeper controlled trunk for Gatekeeper to 
 call
 CME from CUCM Side when I debug i always get ARJ Admission Rejection.
 I don't want to change anything in the technology prefix or anything.
 I don't want to use default technology prefix.
 I want show gatekeeper endpoints and show gatekeeper 
 gw-type-prefix to
 be the same exactly.
 I just want to troubleshoot the issue of calling from CUCM to CME.
 Thank you so much for all your efforts
 
 
 However, here you are my configs
 
 GATEKEEPER HQ Router - SIDE
 
 voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 
 interface Loopback0
 ip address 177.1.254.1 255.255.255.255 h323-gateway voip bind 
 srcaddr 177.1.254.1
 
 gatekeeper
 zone local CUCM cisco.com 177.1.254.1 zone local CUCME cisco.com 
 zone prefix CUCM 1...
 zone prefix CUCM 2...
 zone prefix CUCME 3...
 gw-type-prefix 1*
 no shutdown
 
 
 
 
 SC Side
 
 interface Loopback0
 ip address 177.1.254.3 255.255.255.255 h323-gateway voip interface 
 h323-gateway voip id CUCM ipaddr 177.1.254.1 1719 h323-gateway 
 voip h323-id CUCME h323-gateway voip tech-prefix 31 h323-gateway 
 voip bind srcaddr 177.1.254.3
 
 
 dial-peer voice 85 voip
 destination-pattern [12]...$
 voice-class h323 1
 session target ras
 dtmf-relay h245-alphanumeric
 
 
 CorpHQ(config-dial-peer)#do show gatekeeper end
GATEKEEPER ENDPOINT REGISTRATION

 CallSignalAddr  Port  RASSignalAddr   Port  Zone Name Type
 Flags

Re: [OSL | CCIE_Voice] ISDN signaling config

2013-02-15 Thread Cory Gray
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.  Anyone else?

 

On Wed, Feb 13, 2013 at 9:11 PM, Pixar Perfect pixarperf...@live.com
wrote:

Is there a need to enable(check) Setup non-ISDN Progress Indicators IE
Enable  on the MGCP GW page ?  


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visit www.ipexpert.com

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For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] SRST transfer system and pattern

2013-02-15 Thread Cory Gray
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
OPINION on what should or should not be there.  People say they score points
and then go with the same template on the next lab and get 0 so it is a
mystery.  People can share templates without breaking NDA since the question
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 and pattern

 

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 SRST-EPHONES-CME  

 srst dn template 1

 srst dn line-mode octo

 max-ephones 10

 max-dn 10 preference 2 no-reg both

 ip source-address 10.10.1.13 SiteC Loopback  port 2000

 time-zone 42

 max-conferences 8 gain -6

 call-forward pattern .T

 time-webedit 

 transfer-system full-consult

 transfer-pattern .T

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Re: [OSL | CCIE_Voice] CME Meet me conference not working

2013-02-14 Thread Cory Gray
No but we need max conferences under Cme.  I forgot to ask about that

Sent from my iPhone

On Feb 14, 2013, at 5:14 AM, ie ravindra ieravin...@gmail.com wrote:

 Do not we need to set max trans-coding sessions in sdsp farm category under 
 telephony-service ??
 
 
 On Wed, Feb 13, 2013 at 8:19 PM, Craig Hill (crahill) crah...@cisco.com 
 wrote:
 It would be helpful to see your complete config. There are a few items that 
 need to be checked such as your voice class custom-cptone configuration, 
 complete sccp configuration, and your complete telephony service 
 configuration.  I found meetme to be very sensitive and often times it 
 boiled down to a typo, missed configuration line, or wrong sccp version 
 (needs to be 4.0 or higher). My 2 cents.
 
  
 
 Craig
 
  
 
 From: 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_Voice] CME Meet me conference not working
  
 
 hi Cory ,
 
 
 
 Here...
 
 
 R3#show sdspfarm units ?
   namename of the unit
   registerregistered units
   summary summary of units
   tag tag of unit
   unregister  unregistered units
   |   Output modifiers
   cr
 
 R3#show sdspfarm units reg
 R3#show sdspfarm units register
 
 conf-1 Device:sc-cfb TCP socket:[6]  REGISTERED in SCCP ver 17/10
 actual_stream:24 max_stream 24 IP:14.160.120.98  60315  Conference Dixieland 
 kee
 palive 193
 Supported codec:
  G711Ulaw
  G711Alaw
  G729
  G729a
  G729b
  G729ab
 R3#show sdspfarm units tag 1
 
 conf-1 Device:sc-cfb TCP socket:[6]  REGISTERED in SCCP ver 17/10
 actual_stream:24 max_stream 24 IP:14.160.120.98  60315  Conference Dixieland 
 kee
 palive 194
 Supported codec:
  G711Ulaw
  G711Alaw
  G729
  G729a
  G729b
  G729ab
 
 -conf bridge is registered.
 
 - I remove conference drop-mode local  -  I got it from the srnd
 
 - also did voice-card 0
  no dspfarm
  dsp services dspfarm
 
 - still no good
 
 -Vir
 
 
 From: Cory Gray corygray22...@hotmail.com
 
 Sent: Wed, 13 Feb 2013 18:52:56 
 To: vir...@rediffmail.com
 Cc: ccie_voice@onlinestudylist.com
 
 Subject: Re: [OSL | CCIE_Voice] CME Meet me conference not working
 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 | CCIE_Voice] CME Meet me conference not working
  
 
 Yes check this...
 
 ephone-template  1
  conference drop-mode local
  softkeys idle  ConfList Dnd Mobility Newcall Pickup Redial
  softkeys seized  Callback Cfwdall Gpickup Meetme Pickup Redial
  softkeys connected  Acct ConfList Endcall Hold Join Mobility Park TrnsfVM 
 Trnsfer
 
 
 
 From: Cory Gray corygray22...@hotmail.com
 Sent: Wed, 13 Feb 2013 18:46:23 
 To: 'virajith ' vir...@rediffmail.com, ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] CME Meet me conference not working
 
 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: ccie_voice-boun...@onlinestudylist.com 
 [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of virajith 
 Sent: Wednesday, February 13, 2013 8:03 AM
 To: ccie_voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice] CME Meet me conference not working
 
  
 
 hi  Guys,
 
 My cme meet me conference does not work. I dial 4321 ...it says Cannot 
 Complete Conference
 
 Here is my config ...
 
 voice-card 0
  dspfarm
  dsp services dspfarm
 !
 ephone-dn  7  octo-line
  number 4321 no-reg primary
  conference meetme
 !
 telephony-service
  sdspfarm units 1
  sdspfarm tag 1 sc-cfb
  conference hardware
 !
 sccp ccm group 1
  bind interface FastEthernet0/1
  associate ccm 1 priority 1
  associate profile 1 register sc-cfb
 !
 dspfarm profile 1 conference
  codec g711ulaw
  codec g711alaw
  codec g729ar8
  codec g729abr8
  codec g729r8
  codec g729br8
  maximum sessions 3
  conference-join custom-cptone Join
  conference-leave custom-cptone leave
  associate application SCCP
 !
 
 What next do I check ?
 
 -Vir
 
 
 
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Re: [OSL | CCIE_Voice] [CME WEB ADMIN]

2013-02-14 Thread Cory Gray
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
Sent: Thursday, February 14, 2013 9:42 AM
To: CCIE Study
Subject: [OSL | CCIE_Voice] [CME WEB ADMIN]

 

Hi folks,

As I know there are certain control in CUCME using web administration. I
believe if we used web administration page over CME configuration part might
easier. But is it permitted during the lab time. How we can aproach ? 

Thanks, 

Ravi.

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Re: [OSL | CCIE_Voice] CME Meet me conference not working

2013-02-13 Thread Cory Gray
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: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of virajith 
Sent: Wednesday, February 13, 2013 8:03 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] CME Meet me conference not working

 

hi  Guys,

My cme meet me conference does not work. I dial 4321 ...it says Cannot 
Complete Conference

Here is my config ...

voice-card 0
 dspfarm
 dsp services dspfarm
!
ephone-dn  7  octo-line
 number 4321 no-reg primary
 conference meetme
!
telephony-service
 sdspfarm units 1
 sdspfarm tag 1 sc-cfb
 conference hardware
!
sccp ccm group 1
 bind interface FastEthernet0/1
 associate ccm 1 priority 1
 associate profile 1 register sc-cfb
!
dspfarm profile 1 conference
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 codec g729r8
 codec g729br8
 maximum sessions 3
 conference-join custom-cptone Join
 conference-leave custom-cptone leave
 associate application SCCP
!

What next do I check ?

-Vir

 
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Re: [OSL | CCIE_Voice] CME Meet me conference not working

2013-02-13 Thread Cory Gray
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@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CME Meet me conference not working

 

Yes check this...

ephone-template  1
 conference drop-mode local
 softkeys idle  ConfList Dnd Mobility Newcall Pickup Redial
 softkeys seized  Callback Cfwdall Gpickup Meetme Pickup Redial
 softkeys connected  Acct ConfList Endcall Hold Join Mobility Park TrnsfVM 
Trnsfer



From: Cory Gray corygray22...@hotmail.com
Sent: Wed, 13 Feb 2013 18:46:23 
To: 'virajith ' vir...@rediffmail.com, ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CME Meet me conference not working

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: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of virajith 
Sent: Wednesday, February 13, 2013 8:03 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] CME Meet me conference not working

 

hi  Guys,

My cme meet me conference does not work. I dial 4321 ...it says Cannot 
Complete Conference

Here is my config ...

voice-card 0
 dspfarm
 dsp services dspfarm
!
ephone-dn  7  octo-line
 number 4321 no-reg primary
 conference meetme
!
telephony-service
 sdspfarm units 1
 sdspfarm tag 1 sc-cfb
 conference hardware
!
sccp ccm group 1
 bind interface FastEthernet0/1
 associate ccm 1 priority 1
 associate profile 1 register sc-cfb
!
dspfarm profile 1 conference
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 codec g729r8
 codec g729br8
 maximum sessions 3
 conference-join custom-cptone Join
 conference-leave custom-cptone leave
 associate application SCCP
!

What next do I check ?

-Vir

 
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Re: [OSL | CCIE_Voice] CME Meet me conference not working

2013-02-13 Thread Cory Gray
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 | CCIE_Voice] CME Meet me conference not working

 

Yes check this...

ephone-template  1
 conference drop-mode local
 softkeys idle  ConfList Dnd Mobility Newcall Pickup Redial
 softkeys seized  Callback Cfwdall Gpickup Meetme Pickup Redial
 softkeys connected  Acct ConfList Endcall Hold Join Mobility Park TrnsfVM 
Trnsfer



From: Cory Gray corygray22...@hotmail.com
Sent: Wed, 13 Feb 2013 18:46:23 
To: 'virajith ' vir...@rediffmail.com, ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CME Meet me conference not working

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: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of virajith 
Sent: Wednesday, February 13, 2013 8:03 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] CME Meet me conference not working

 

hi  Guys,

My cme meet me conference does not work. I dial 4321 ...it says Cannot 
Complete Conference

Here is my config ...

voice-card 0
 dspfarm
 dsp services dspfarm
!
ephone-dn  7  octo-line
 number 4321 no-reg primary
 conference meetme
!
telephony-service
 sdspfarm units 1
 sdspfarm tag 1 sc-cfb
 conference hardware
!
sccp ccm group 1
 bind interface FastEthernet0/1
 associate ccm 1 priority 1
 associate profile 1 register sc-cfb
!
dspfarm profile 1 conference
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 codec g729r8
 codec g729br8
 maximum sessions 3
 conference-join custom-cptone Join
 conference-leave custom-cptone leave
 associate application SCCP
!

What next do I check ?

-Vir

 
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Re: [OSL | CCIE_Voice] CME Meet me conference not working

2013-02-13 Thread Cory Gray
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#show sdspfarm units ?
  namename of the unit
  registerregistered units
  summary summary of units
  tag tag of unit
  unregister  unregistered units
  |   Output modifiers
  cr

R3#show sdspfarm units reg
R3#show sdspfarm units register

conf-1 Device:sc-cfb TCP socket:[6]  REGISTERED in SCCP ver 17/10
actual_stream:24 max_stream 24 IP:14.160.120.98  60315  Conference Dixieland kee
palive 193
Supported codec:
 G711Ulaw
 G711Alaw
 G729
 G729a
 G729b
 G729ab
R3#show sdspfarm units tag 1

conf-1 Device:sc-cfb TCP socket:[6]  REGISTERED in SCCP ver 17/10
actual_stream:24 max_stream 24 IP:14.160.120.98  60315  Conference Dixieland kee
palive 194
Supported codec:
 G711Ulaw
 G711Alaw
 G729
 G729a
 G729b
 G729ab

-conf bridge is registered.

- I remove conference drop-mode local  -  I got it from the srnd

- also did voice-card 0
 no dspfarm
 dsp services dspfarm

- still no good

-Vir


From: Cory Gray corygray22...@hotmail.com
Sent: Wed, 13 Feb 2013 18:52:56 
To: vir...@rediffmail.com
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CME Meet me conference not working

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 | CCIE_Voice] CME Meet me conference not working

 

Yes check this...

ephone-template  1
 conference drop-mode local
 softkeys idle  ConfList Dnd Mobility Newcall Pickup Redial
 softkeys seized  Callback Cfwdall Gpickup Meetme Pickup Redial
 softkeys connected  Acct ConfList Endcall Hold Join Mobility Park TrnsfVM 
Trnsfer



From: Cory Gray corygray22...@hotmail.com
Sent: Wed, 13 Feb 2013 18:46:23 
To: 'virajith ' vir...@rediffmail.com, ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CME Meet me conference not working

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: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of virajith 
Sent: Wednesday, February 13, 2013 8:03 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] CME Meet me conference not working

 

hi  Guys,

My cme meet me conference does not work. I dial 4321 ...it says Cannot 
Complete Conference

Here is my config ...

voice-card 0
 dspfarm
 dsp services dspfarm
!
ephone-dn  7  octo-line
 number 4321 no-reg primary
 conference meetme
!
telephony-service
 sdspfarm units 1
 sdspfarm tag 1 sc-cfb
 conference hardware
!
sccp ccm group 1
 bind interface FastEthernet0/1
 associate ccm 1 priority 1
 associate profile 1 register sc-cfb
!
dspfarm profile 1 conference
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 codec g729r8
 codec g729br8
 maximum sessions 3
 conference-join custom-cptone Join
 conference-leave custom-cptone leave
 associate application SCCP
!

What next do I check ?

-Vir

 
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Catch India as it happens with the Rediff News App. To download it for FREE, 
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Re: [OSL | CCIE_Voice] Virtualized UCCX

2013-02-11 Thread Cory Gray
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
Subject: Re: [OSL | CCIE_Voice] Virtualized UCCX

 

Yes Corry that's why I went ahead with my method. No jtapi errors :-) 

Sent from hand held

On Feb 11, 2013 6:15 PM, Cory Gray corygray22...@hotmail.com wrote:

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 your server is a MCS7845, install CUCCX
5. After CUCCX is installed take a snapshot of the VM BEFORE you do the CUCM
integration.  This is so when you start new labs, no need to recreate the
VM... just redo the integration.

PS.  I always get JTAPI inconsistent versions between CUCM and CUCCX but
that has never impacted functionality.

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Todd Carswell
(tcarswel)
Sent: Monday, February 11, 2013 9:00 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Virtualized UCCX

Does anyone have their lab UCCX on VMWare ESXi 5.0?  If yes, what method did
you use to get it installed on a VM?  I've heard of folks installing on an
MCS server and using VMWare Converter to migrate over to a virtual machine.
Any thoughts?

Thanks!

--Todd
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Re: [OSL | CCIE_Voice] [OSL || CCIE VOICE ]

2013-02-03 Thread Cory Gray
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 phones are 
not actually connected to your switch the only way to power cycle them before 
they register is to pull the power card.  I used to do this to speed up the 
phones requesting an IP.

Now i do DHCP first so if I do vlans in order... Sitea, b, the c By the 
time it get through site C, site a and site b have ips without me having to 
touch the phone.  This is all assuming nothing in your config will stop dhcp 
from working correctly



Sent from my iPhone

On Feb 3, 2013, at 2:21 AM, Gurpreet Singh Kukreja 
tycoononway1...@gmail.com wrote:

 Ravi,
 
 You need to speed up. 30 minutes for these sections is too much.. What 
 exactly in these two sections is taking too long for you?
 
 usually DHCP and VLANs should not take more than 10 to 15 minutes as far as 
 my practice goes. Practice them over and over to build speed and accuracy.
 
 Regards
 G
 
 On Sat, Feb 2, 2013 at 11:11 PM, ie ravindra ieravin...@gmail.com wrote:
 Hi Mates , 
 
 I have few questions on the real lab just to make up my speed. Thanks for 
 all you guys I believe I have made my hardest part DialPlan two easier. my 
 only clue is now AAR. but thanks for Jsutin and Cory I am evaluate those 
 today. 
 Now I got another problem. I am still following technology based lab 
 approach since I am not too practiced device based aproach yet. But still I 
 am getting 30 Mins to complete DHCP and VLAN section except NTP. will that 
 be a problem in starting minutes in the real lab. Do you guys have any 
 advices on that as well as the whole lab exam what ever the experiences you 
 got in the real lab, 
 
 Thanks  , 
 Ravi.
 
 ___
 For more information regarding industry leading CCIE Lab training, please 
 visit www.ipexpert.com
 
 Are you a CCNP or CCIE and looking for a job? Check out 
 www.PlatinumPlacement.com
 
 ___
 For more information regarding industry leading CCIE Lab training, please 
 visit www.ipexpert.com
 
 Are you a CCNP or CCIE and looking for a job? Check out 
 www.PlatinumPlacement.com
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com___
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Re: [OSL | CCIE_Voice] [OSL | Automated Alternative Routing]

2013-02-02 Thread Cory Gray
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 - strip and prefix digits on the called number to match an existing
dialpeer on the gateway.  Do not do any calling number manipulations as that
is already taken care of by the dial peer

 

If both sites are using AAR

Attach calling called number transformations to the gateway to achieve
everything from above

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Justin Carney
Sent: Friday, February 01, 2013 11:56 PM
To: ie ravindra
Cc: CCIE Study
Subject: Re: [OSL | CCIE_Voice] [OSL | Automated Alternative Routing]

 

I noticed a typo in that last email (sorry, I clicked send too soon) - in
the 3 options for H323 digit manipulation, I said num-exp will be between
the inbound voip dial peer and outbound VOIP dial peer...the outbound dial
peer is POTS in this case, not voip.  (using two voip dial-peers on both
inbound and outbound is CUBE, which is not relevant to AAR configuration).

 

-Justin

 

On Fri, Feb 1, 2013 at 11:46 PM, Justin Carney justin.s.car...@gmail.com
wrote:

Yes, AAR is triggered on CAC reporting out of bandwidth.  (Side note - the
phone will display Network Congestion. Rerouting and this is a service
parameter that can be customized, in case that is part of the question
requirement.)  You are also correct that both phones must be registered to
the same CUCM cluster.  I don't understand if your last sentence is a
question - for some reason that call fails to call by extension - if there
is a CSS/PT issue where phone A can't see the DN of phone B, AAR will not
kick in.  Under normal conditions phone A must be able to call phone B, then
when there is no more bandwidth (per CAC) AAR will reroute via PSTN.  If the
phone B were in SRST mode and the WAN was down, not congested, this would
instead use CFUR to reroute.

 

I'll answer question 2 first.  A common way to achieve AAR is to use a
separate CSS/PT just for AAR, along with an AAR Group assigned to both lines
(you can assign AAR group to phones for other reasons, but you *must* put
the AAR group on the line/DN).  When AAR is triggered (CAC), the called
phone B's external number mask will be the new   DNIS which should be in
E.164 format already, and the calling phone A's AAR-CSS will be used to
lookup a route for that DNIS.  Simply put a \+.! route pattern in your
AAR-PT that routes to the LRG, and the AAR-CSS should contain this AAR-PT.
This gets the call to the gateway.  If the gateway is MGCP, you may need to
manipulate the plan/type to match what the PSTN expects (You may
also/instead need to have a \+1.! pattern in AAR-PT in the event your MGCP
router's PRI expect a 10 digit DNIS.)  For H323 don't do any digit
manipulation here, used the gateway to perform all manipulations.

 

Question 1, dial peers needed.  If using the strategy above, you might not
need any new dial peers.  For the MGCP sites there are no dial peers on the
router so you are done after CUCM routes the call to the gateway in the
proper format.

 

For the H323 sites that need to route the AAR call, the DNIS will be the
E.164 number when the call gets to the inbound voip dial peer.  If you have
an existing outbound pots dial peer that will match this E164 number there
is nothing extra to do, your AAR call should be working.  (make sure you
have the appropriate number of digits and type/plan sent to the PSTN for
both ANI and DNIS).  If your existing dial peers do not match, you have a
few options:

1.  you could use a translation-profile on the inbound voip dial peer to
manipulate the DNIS into something that matches an existing outbound POTS
dial peer

1.  for example if your DNIS is +1 408 555 1234
tel:%2B1%20408%20555%201234 , you would change the +1 to 91 and you would
match the existing long distance outbound dial peer

2.  you could add a new outbound dial peer that will match this DNIS
(optionally putting a translation-profile on this dial peer if you need a
specific plan/type)

1.  for example if your DNIS is +1 408 555 1234
tel:%2B1%20408%20555%201234 , you can copy your existing long distance
dial peer (9+11 digits) and just remove the 9 (leaving 11 digits)

3.  A third option (I would recommend you do NOT use this option) would
be to use number expansion to manipulate the DNIS between the inbound voip
dial peer match and the outbound voip dial peer match - the reason I don't
recommend this is because number expansion ALWAYS takes place between the
inbound and outbound dial peers even if you don't want it to.  This means if
you're not careful it could break something else that was already working
correctly.

For question 3, TEHO - if you use the method above, your TEHO patterns will

Re: [OSL | CCIE_Voice] Site A to Site C can't leave vm on CUE

2013-02-02 Thread Cory Gray
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 need ephone 3

Also it is a good practice to use version 6 or higher for sccp ccm
configuration or higher

Also use bind interface X under sccp ccm group

You should also not need mailbox-selection last-redirect-num

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Joe Fearday
Sent: Saturday, February 02, 2013 11:28 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Site A to Site C can't leave vm on CUE

 

Site A to Site C can't leave vm on CUE 

 

Site A to Site C configured for g729
Site A routes to Site C using gk in region C
PSTN can leave vm on CUE as can other site C phones
I am thinking this is a site C transcoding problem
CUE only supports g711; right?
Site C transcoding detailed below along with ephone:

 

sccp local Vlan102
sccp ccm 10.196.102.1 identifier 1 version 3.1 
sccp
!
sccp ccm group 1
 associate ccm 1 priority 1
 associate profile 1 register sctrans
!
dspfarm profile 1 transcode  
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 codec g729r8
 maximum sessions 12
 associate application SCCP
 
 telephony-service
 sdspfarm units 1
 sdspfarm transcode sessions 20
 sdspfarm tag 1 sctrans

 

ephone  3
codec g729r8 dspfarm-assist

 

dial-peer voice 9000 voip
 mailbox-selection last-redirect-num
 destination-pattern 3180  (CUE)
 session protocol sipv2
 session target ipv4:10.196.102.2
 dtmf-relay sip-notify
 codec g711ulaw
 no vad
 
Any assistance is greatly apprecaited.
Joe Fearday

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Re: [OSL | CCIE_Voice] Failed Voice Lab

2013-01-31 Thread Cory Gray

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 left the exam.  Others on 
this list have ran into the issue you did of thinking they passed and then 
ended up failing.  Then they are successful on the next one.  Keep pressing on. 
 If there is a certain section you always score low on, you may need to adjust 
your methods.  I never got points on SRST so I looked to people here for help 
on what they do every time they configure SRST to see if my whole approach was 
wrong.  We are all here for things like that so let us know if we can help.

Cory

Date: Thu, 31 Jan 2013 14:31:26 -0600
From: ramcharan.a...@gmail.com
To: benjoh...@hotmail.com
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Failed Voice Lab

Hi Ben,
I feel bad about this news  you did not score passing marks in lab exam. Can 
you please share your experience  without breaking NDA.
At least this will be useful for others who are preparing for lab.

Thanks  Regards,
Ramcharan AryaCCIE # 28926

On Thu, Jan 31, 2013 at 1:39 PM, Ben John benjoh...@hotmail.com wrote:






Guys,
i took my lab last week in SJ and i failed again that was my fourth attempt. i 
got everything working tested and i was confident i will get it this time but 
when i got my result i was disappointed i don't feel like going back again i 
feel like i am not spending enough times with my family and it cost me almost 
2K for each attempt. i have seen all the labs as far as i know i don't if i do 
go back which one i will get and there is no guaranty i will pass it looks like 
i am doing the way cisco wants it. i am really down right the good thing is i 
can do Voice. 


Thanks,

Ben
  

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___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com ___
For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] [OSL | CCIE_VOICE] Location Based Call Admission Control

2013-01-30 Thread Cory Gray
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 bring...@gmail.com wrote:

 Again it depends on if you are asked to do so.
 
 
 On Wed, Jan 30, 2013 at 3:56 PM, ie ravindra ieravin...@gmail.com wrote:
 Hi All, 
 Do we need to enable ip rsvp bandwidth command when we configure location 
 based CAC. 
 
 Thanks, 
 Ravi.
 
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 -- 
 Suresh Bhandari
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Re: [OSL | CCIE_Voice] CME 12.4T and max-dn X no-reg

2013-01-30 Thread Cory Gray
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, Jason Aarons scubajas...@gmail.com wrote:

 Is there a known bug with max-dn 30 no-reg in that ephone-dns still register 
 to gatekeeper? Anyone got a bug id? 
  
 Pretty clear it don’t work, you have to put no-reg in the ephone-dn itself.
  
 -jason
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Re: [OSL | CCIE_Voice] Correct use of G729r8

2013-01-29 Thread Cory Gray
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 preference os 100% 
correct

Sent from my iPhone

On Jan 29, 2013, at 3:41 PM, Alan Parr ap...@abbmail.com wrote:

 I am in the habit when creating IOS MTP, Conference and Transcoder resources 
 to use G729r8. In the case of MTP, a “no codec g711u” and then “codec 
 g729r8”. In the case of the other two, I just add it to the existing list. I 
 understand that it consumes more resources, but I was under the impression 
 that it was the correct thing to do. Maybe a Cisco salesperson with a quota 
 of DSP’s to sell suggested that J
  
 Likewise, in my “voice class voice” I use g729r8 as the second preference. I 
 see however in reading the solution guides that the addition of g729r8 to 
 transcoding and conferencing resources is not typical. The use of g729r8 in 
 the voice class codec is typical. Maybe I am missing something, but if it’s 
 the preference in the voice class codec, shouldn’t the transcoder also 
 support it in case a transcoder is needed?
  
 What is your experience, both in your studies for the lab and in real 
 deployments, on the use of g729r8 ?
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Re: [OSL | CCIE_Voice] UCCX Recordings

2013-01-29 Thread Cory Gray
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 remember

Upload this to uccx througj prompt upload

This method works 100% of the time if you can call into cuccx.

Sent from my iPhone

On Jan 29, 2013, at 3:58 PM, Tanner Ezell tanner.ez...@gmail.com wrote:

 User uAuthUser
 Document dRecording
 
 uAuthUser = Get User (myuser, by User Id)
 Authenticate (uAuthUser, myusepassword)
  - Success:
- dRecording = Recording()
  - Success:
- Upload Document(myPrompt.wav, dRecording)
 
 I'd throw in a get digits step to collect the prompt number simplify the 
 process a bit.
 
 Thats the bare minimum to get recording going.
 
 Tanner.
 
 On Tue, Jan 29, 2013 at 11:10 AM, Barrera, Hugo hugo.barr...@nexusis.com 
 wrote:
 Hi,
 
  
 
 I just want to hear some different suggestions on how you have done your 
 recordings for uccx in the lab? I know you can do them on CUC through 
 greetings administrator but what about from uccx? Is there a quick script to 
 throw together that will allow me to directly record greetings on uccx I 
 think there is??
 
  
 
  
 
 Hugo
 
  
 
 
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 ___
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 visit www.ipexpert.com
 
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Re: [OSL | CCIE_Voice] UCCX Recordings

2013-01-29 Thread Cory Gray
It is quick and works on version 7.  Is that not the goal?  If it messes you up 
in the CCIE lab, please say way.

Sent from my iPhone

On Jan 29, 2013, at 6:04 PM, Tanner Ezell tanner.ez...@gmail.com wrote:

 Not only is this bad practice, it won't be supported on 8.x or later.
 
 On Tue, Jan 29, 2013 at 4:01 PM, Cory Gray corygray22...@hotmail.com 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
 End
 
 Recording will copy to wfavvid  temp
 
 Sometimes it creats 2 files
 
 Delete the 1k file
 
 Rename the remaining file to something easy to remember
 
 Upload this to uccx througj prompt upload
 
 This method works 100% of the time if you can call into cuccx.
 
 Sent from my iPhone
 
 On Jan 29, 2013, at 3:58 PM, Tanner Ezell tanner.ez...@gmail.com wrote:
 
 User uAuthUser
 Document dRecording
 
 uAuthUser = Get User (myuser, by User Id)
 Authenticate (uAuthUser, myusepassword)
  - Success:
- dRecording = Recording()
  - Success:
- Upload Document(myPrompt.wav, dRecording)
 
 I'd throw in a get digits step to collect the prompt number simplify the 
 process a bit.
 
 Thats the bare minimum to get recording going.
 
 Tanner.
 
 On Tue, Jan 29, 2013 at 11:10 AM, Barrera, Hugo hugo.barr...@nexusis.com 
 wrote:
 Hi,
 
  
 
 I just want to hear some different suggestions on how you have done your 
 recordings for uccx in the lab? I know you can do them on CUC through 
 greetings administrator but what about from uccx? Is there a quick script 
 to throw together that will allow me to directly record greetings on uccx 
 I think there is??
 
  
 
  
 
 Hugo
 
  
 
 
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 visit www.ipexpert.com
 
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 www.PlatinumPlacement.com
 
 ___
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 visit www.ipexpert.com
 
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 www.PlatinumPlacement.com
 
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Re: [OSL | CCIE_Voice] [CCIE VOICE - Answer the Dial-Plan questions.

2013-01-25 Thread Cory Gray
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.

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ramcharan Arya
Sent: Friday, January 25, 2013 10:02 AM
To: ie ravindra
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] [CCIE VOICE - Answer the Dial-Plan
questions.

 

Hi ,

I suggest you work on VOL 1 Lab 5A  5C this will help you understanding
dialplan.Both labs are very good for call routing section.


Regards,
Ramcharan Arya
CCIE # 28926

On Fri, Jan 25, 2013 at 8:58 AM, ie ravindra ieravin...@gmail.com wrote:

.  

 

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Re: [OSL | CCIE_Voice] [ Dial-Peer Question ]

2013-01-25 Thread Cory Gray
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)

port 0/0/0:23

!

dial-peer voice 901 pots

translation-profile outgoing LD (create a voice translation for calling and
called number)
destination-pattern 91[2-9]..[2-9]..$ ($ not needed but wont hurt you)
forward-digits 11

port 0/0/0:23 

3] dial-peer voice 902 pots 

translation-profile outgoing INTL (create a voice translation for calling
and called number)
destination-pattern 9011T (Need the T to wait for extra digits and allow
the user to keep dialing. # is a default terminator so a user can hit #
when they are finished dialing)
prefix 011 (You cannot use forward-digits because international is a
variable amount of digits. 9011 is stripped by default so if you want to add
011 prefix it

port 0/0/0:23 



 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Suresh Bhandari
Sent: Friday, January 25, 2013 11:38 AM
To: ie ravindra
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] [ Dial-Peer Question ]

 

In your dial peers, the exact matches 9 in your dial peer [1], 91 in your
second dialpeer and 9011 in your third dialpeer are stripped off of the
destination pattern.

For your second question, 91, the exact match from your destination pattern
is removed, and 1 is prefixed to whatever 10 digits are matched by the
wildcard.

Hope this helps.

 

On Fri, Jan 25, 2013 at 10:05 PM, ie ravindra ieravin...@gmail.com wrote:


Dear All , 

Thanks for the answers you given for my last questions. for the above what
are the digits will forwarded to PSTN. How we can use prefix command. any
way sorry for my wrong english.

1] dial-peer voice 900 pots 
destination-pattern 9[2-9]..$
port 0/0/0:23 

2] dial-peer voice 901 pots 
destination-pattern 91[2-9]..[2-9]..$ 
port 0/0/0:23 
prefix 1

3] dial-peer voice 902 pots 
destination-pattern 9011
port 0/0/0:23 
prefix 011 

[2-9]XX needed to forwarded to PSTN
1[2-9]XX[2-9]XX needed to forward PSTN 
011 should forwarded to PSTN. 

Qusetion is 
1. as per the dial-peer [1] how should I remove 9 from the dial peer. 
2. in Dial-peer 2 [901 pots] / prefix is 1. is it means 901 dial-peer routes
from starting digit as  1[2-9]..[2-9]..$ or is it route as
191[2-9]..[2-9]..$ to the PSTN. 

Thanks,
Ravi.


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-- 
Suresh Bhandari

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Re: [OSL | CCIE_Voice] CCIE

2013-01-25 Thread Cory Gray
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 engage me one.  So coming into this, I never setup CUE, CUCME
(back when it was still ITS), GK, CUBE, CUCCX, CUP, QoS(CCIE Lab QoS J ), or
read sdi traces.  I finally took the time to not accept any weaknesses.
CUCCX was the hardest to learn from scratch.  You cannot go in there
PLANNING to leave points on the table.

 

Second.  I had a study partner.  Like Bill said, there are so many ways to
do the most basic tasks.  Having a study partner will expose you to
different methods of doing things.  These methods you have not thought of
can speed things up or maybe even show you that you were completely wrong.

 

The same thing that makes questions vague makes the lab strategy either.
They rarely give you restrictions so it is up to you how easy or how hard
you make fulfilling a task.

 

Congrats again Bill!

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Bill Lake
Sent: Friday, January 25, 2013 3:38 PM
To: CCIE Study
Subject: [OSL | CCIE_Voice] CCIE

 

Hello All,

It has been a tough, long and challenging journey but one that has made me a
much better at my job.  Today I got the great news of passing my lab in RTP.


My feedback is as follows and I do hope that it will help someone turn the
corner and pass their lab.

Cory earlier said to create note pads, I agree that is very helpful.  While
I don't do everything in notepad it makes it much easier if you find a
mistake to fix, first no the bad command and then add the good one.  I
actually do it basically like he said, one for each site, one for the switch
and I add one for services and a temp pad for cut and paste items that need
to be adjusted.

I would also say that one of the biggest things that I did to help is
practice as much as possible on live lab equipment.  This helps you
recognized where the issue is, if you can't set up a full lab then surely
get one router and as many phones as you can and do hardware VPN

Next practice your base configs until you see them in your sleep, know where
you tend to make mistakes and find any copy and paste resources you can to
help.  Never shy away from a subject because you are weak at it, focus on
getting better all the time.  I used to dread complex dial plans but now I
find them pretty easy, I tried to find as many creative ways to solve every
problem and requested design.  This give you flexibility you will need on
the lab. Trust me, you could easily have a freeze moment where you know how
to do something but just can't remember it.  But if you have a second way to
do the task, you will more than likely remember one of them.  Even shortcut
items might not get you the points but if your out of time and have to try
to get something to work, give it a try it can't be any worse than not doing
it.

Now that you are got your base in place, practice adding everything you can
think of to them, so if someone says can you do this, say sure but I can
also do xyz, a perfect example of this would be something like setting up a
meetme conference, can you have it announce the name of the caller, can you
make them dial a different number then be rerouted to the conference when
they enter their pin.  Things like that, again you learn where you make your
mistakes (everyone does) and be able to fix them quickly and effectively.

Last recommendation is prepared for the unexpected, so what do you do if
everything looks like it should work but doesn't, have a system for solving
that. 

 Everyone will have a different path but having the knowledge, ability to
apply that knowledge and a little luck will help you be a CCIE.

I would especially like to thank Vik, Kevin and the crew at IPexpert.  I
would like to thank Randall and Michael as they were both very helpful in me
bouncing ideas and thoughts off.  Finally and most importantly I want to
thank my wife and daughter who sacrificed their time with me to have me
study endlessly (it seems) to pass this test. 

Bill

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[OSL | CCIE_Voice] Passed!!!!!

2013-01-24 Thread Cory Gray
All,

 

I finally passed my CCIE Voice in RTP yesterday.  I have an RS 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-Disclosure Agreement (NDA).  I am a Cisco Employee so that increases
your chances of being reported J

 

Observations

It is not obvious which task depend on each so you have to think about how a
lot of task may affect others.  The Voice lab is very wordy.  IPexpert's
practice labs are very clear and concise in their requirements.  It is very
easy to get lost in the words due to time pressure and how we read.  My
guess is if English is your first language you will skip words and if
English is not, it will be harder to interpret than it is for native
speakers.  That is just a case.  Yesterday, I read a task that said should
and I thought it said should not.  If it was not for a later task that
conflicted with should not,  I would have never noticed and would have
lost the points (assuming I still did not lose the pointsJ ).

 

So this lab I focused on attention to detail.  I read slower and took more
time so I was only finished with about 1hour and 15 minutes left.  Some
requirements may not be clear to you so making an assumption is the worst
thing you can do.  Seek clarity from the proctor.  We all know it is hard to
ask the question in the right way so they can answer but it is better than
assuming.  My strategy was to mark down the questions I have and then bring
the proctor over to my desk and ask him every question in one sitting to
save time.  That worked.  We had our debates about clarifying questions
but this saved me a huge amount of time doing all of my questions at the
same time.  One of my questions was even answered further in the lab based
on what another task asked.

 

For the rest of the time I went back through my lab.  The only thing I do
not verify first time through is the dial plan.  This is my first time
making it to the end without leaving any points on the table so I had enough
time to verify my dial plan.  I figured there is no way every test call
could work 100% without me testing anything but 911.  I was right.  I fixed
a lot of silly mistakes.

 

IPexpert Training Notes

You need the 5-lab handbook and the 4 labs you get from the One Week Lab
Experience.  I have 3 more labs from the previous round of OWLE and those
helped as well.  I hope IPexpert finds a way to include those questions that
require you to think outside the box or features that no one uses in the
real world in their materials.  The proctor told us during one lab there are
several tracks per CCIE Lab.  There is a 30% change between each lab in the
same track.  So let's say there are 4 voice tracks and 4 labs per track
(just using round numbers).  Even if you get Track 3 Lab 1 and then get
Track 3 Lab2, you can still easily fail.  That can be a big 30%.  The point
is every practice question is important so the more questions you have the
better.

 

I have had scenarios that I could not get working during past labs but there
was never one scenario where I said, I have no idea how to do this.
IPexpert will give you all of the technical skills to pass but unfortunately
that may not enough.  With all of the words and all of the assumptions they
expect you to make, there is no good way I can think of to prepare for that.
Especially with integrated troubleshooting.  You may think of things they do
that make sure phones not get DHCP but there are things that will not break
your configuration that you still have to spot and fix.  After taking the
lab many times, you start to understand what are some of these things to
look for and you making check those things part of your routine.  

 

The new Troubleshooting course is essential.  Seeing the debugs in the
IPexpert practice labs is nowhere as good as having Vik walk you through it.
I can now easily read h323, SIP, and mgcp traces thanks to that class.

 

Other Notes

I heard in the past people could skip sections like CUCCX, QoS, and/or
Troubleshooting and still pass.  I do not believe that is possible anymore
if it ever was in the past.  Assume the worst that every questions is worth
5 points.  That means you can only miss 4 questions.  There is always stuff
that you think you configured right that you really did not so you cannot
leave ANY points on the table.  You have to leave thinking you got 90+
points to have a chance in passing.

 

I am done ranting.

 

Let me (and the list) know if you have any questions,

 

Cory

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Re: [OSL | CCIE_Voice] Passed!!!!!

2013-01-24 Thread Cory Gray
Some Questions I just got

 

1.   Any tips on using configurations in notepad or direct typing into
router without using tab.

 

I create 4 notepads for every exam

Info.txt - Any information such as IPs, VLAN numbers, login information etc
for each site.

SiteA.txt - Dialplan for Site A

SiteB.txt - Dialplan for Site B

SiteC.txt - Dialplan for Site C

If a site is H323 I do the entire dialplan in the text file and then copy
and paste it to the router.  I can then just edit my notepad as needed.  If
I have 2 H323 sites, I copy one dial plan configuration to the other and
change what I need to change.

I always use notepad for my SCCP media resources configuration.  I grab the
template from the documentation website, edit it, apply it to a site, change
what needs to be changes for next site, apply it, and do it for the third.
This is my first lab that I registered Conference, Transcoder, and MTP all
at the same time before I even had gateways configured.  This is just in
case I needed one and did not realize it and I could do it fairly quickly
and never have to worry about it for the rest of the day.  You SHOULD not
loose point for extra configuration as long as it does not affect something
else (be careful!).  I do a lot of things in every lab that I have never
used before so I can get into a routine that was bullet proof against ANY
future scenario.  For me, it is a waste of time to read through the lab.  I
have templates but I never read ahead I just always prepare for the unknown.
I even configure things in CUCM out of habit that I never used because they
are not on my lab.  I just would not apply them.

Does routing dial plan table is right strategy for lab.

This is a VERY personal question on how to do it but I do not believe you
cannot get through the dial plan quick enough without notes.  In one of my
earlier attempts when I was not ready, it took me 2 hours!  I did not take
notes!  This is what I do.

Pretend these are your requirements for a fictional US dialplan

9 outside code

Plan and type whatever

Emergency - 911 with 4 digit caller id

Local - 7 digits with 7 digit caller id

LD - 11 digits with 10 digit caller id

International - any digits with + caller id

 

This is what my text file would look like

 

EMER Calling - 4

EMER Called - 911 unk unk

 

LOCAL calling - 7

LOCAL called - 9.7 sub isdn

 

LD Calling - 10

LD Called - 9.1+10

 

INTL calling - E164

INTL called - T

What was the monitor size in RTP.?

 

Not sure.  Big enough that you can see your GUI pages FULL SCREEN and still
see your notepad right beside it

Can you share how useful OWLE books and Troubleshooting class might help me
in preperation of my 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: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Passed!

 

All,

 

I finally passed my CCIE Voice in RTP yesterday.  I have an RS 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-Disclosure Agreement (NDA).  I am a Cisco Employee so that increases
your chances of being reported J

 

Observations

It is not obvious which task depend on each so you have to think about how a
lot of task may affect others.  The Voice lab is very wordy.  IPexpert's
practice labs are very clear and concise in their requirements.  It is very
easy to get lost in the words due to time pressure and how we read.  My
guess is if English is your first language you will skip words and if
English is not, it will be harder to interpret than it is for native
speakers.  That is just a case.  Yesterday, I read a task that said should
and I thought it said should not.  If it was not for a later task that
conflicted with should not,  I would have never noticed and would have
lost the points (assuming I still did not lose the pointsJ ).

 

So this lab I focused on attention to detail.  I read slower and took more
time so I was only finished with about 1hour and 15 minutes left.  Some
requirements may not be clear to you so making an assumption is the worst
thing you can do.  Seek clarity from the proctor.  We all know it is hard to
ask the question in the right way so they can answer but it is better than
assuming.  My strategy was to mark down the questions I have and then bring
the proctor over to my desk and ask him every question in one sitting to
save time.  That worked.  We had our debates about clarifying questions
but this saved me a huge amount of time doing all of my questions at the
same time.  One of my questions was even answered further in the lab based
on what another task asked.

 

For the rest

[OSL | CCIE_Voice] QoS Access List

2013-01-21 Thread Cory Gray
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/switches/lan/catalyst3750/software/release/1
2.2_44_se/configuration/guide/swacl.html

 

The switch does not support these Cisco IOS router ACL-related features:

ACL logging for port ACLs and VLAN maps

 

If you read further down it says you can put Log on your entries so you
can see hits via syslog messages.

 

But if you try to do it for QoS, you get this error message

 

class-map CLASS_MGCP : access-list with 'log' not supported, pls remove
'log' from access-list otherwise class-map CLASS_MGCP will not work properly

 

I tried to put the ACL directly on the router for giggles but it caused the
switch to stop passing traffic.  I saw the document advising this could
happen but my ACL only covered MGCP so I thought I would be ok.

 

The moral of the story is, you just have to trust your config and use qos
related show commands to get a feel for what is going on.

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Re: [OSL | CCIE_Voice] Custom Background image for phones registered to cme

2013-01-20 Thread Cory Gray
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 Guide

Create Full size image-320 pixels (width) X 212 pixels (height)

Create Thumbnail image-80 pixels (width) X 53 pixels (height)

 

Upload all three to the router

mkdir flash:Desktops

mkdir flash:Desktops/320x212x16/

copy tftp flash:Desktops/320x212x16/ (for all three files)

 

Serve all three files

tftp-server flash:Desktops/320x212x16/List.xml

tftp-server flash:Desktops/320x212x16/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 sanity insanity
Sent: Sunday, January 20, 2013 8:22 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Custom Background image for phones registered to
cme

 

Guys,

What is the easiest way of getting a customized background image and
configuring it for cme  ip phones . Given that there is no TFTP server , we
just need to use the callmanger . Please illustrate the steps...

Regards,
Vir

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Re: [OSL | CCIE_Voice] Custom Background image for phones registered to cme

2013-01-20 Thread Cory Gray
A little more detail

 

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 your Files

Create List.xml from the Guide

Create Full size image-320 pixels (width) X 212 pixels (height)

Create Thumbnail image-80 pixels (width) X 53 pixels (height)

 

Upload the Files to CUCM

Upload them to CUCM using TFTP File Management

Restart TFTP Service

 

Create Directories on the Router

mkdir flash:Desktops

mkdir flash:Desktops/320x212x16/

 

Download the files from CUCM to the Router

copy tftp flash:Desktops/320x212x16/ (for all three files)

 

Serve all three files on the Router

tftp-server flash:Desktops/320x212x16/List.xml

tftp-server flash:Desktops/320x212x16/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 insanity'; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Custom Background image for phones
registered to cme

 

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 Guide

Create Full size image-320 pixels (width) X 212 pixels (height)

Create Thumbnail image-80 pixels (width) X 53 pixels (height)

 

Upload all three to the router

mkdir flash:Desktops

mkdir flash:Desktops/320x212x16/

copy tftp flash:Desktops/320x212x16/ (for all three files)

 

Serve all three files

tftp-server flash:Desktops/320x212x16/List.xml

tftp-server flash:Desktops/320x212x16/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 sanity insanity
Sent: Sunday, January 20, 2013 8:22 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Custom Background image for phones registered to
cme

 

Guys,

What is the easiest way of getting a customized background image and
configuring it for cme  ip phones . Given that there is no TFTP server , we
just need to use the callmanger . Please illustrate the steps...

Regards,
Vir

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Re: [OSL | CCIE_Voice] cisco DocCD

2013-01-14 Thread Cory Gray
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 aboaz...@gmail.com wrote:

 Hi All, 
 
 I was practicing the cisco Documentation CD that will be available during the 
 lab http://www.cisco.com/cisco/web/psa/default.html 
 to get used on it, I was searching for the topic in unity connection How to 
 set up the Phone View ,this topic is mentioned in the System Administration 
 for Cisco Unity Connection Release 7.X 
 (http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/administration/guide/7xcucsag245.html)
 I was not able to find the path in the Doc CD, I used Google to fine this 
 guide from cisco site.
 
 Do you guys practice the Doc CD - If not, then you have to, as this is the 
 only documentation resource during the exam..-, 
 
 do you know the Path to the system administration guide for the products.
 
 Awaiting your response, Appreciate that.
 
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Re: [OSL | CCIE_Voice] cisco DocCD

2013-01-14 Thread Cory Gray
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 corygray22...@hotmail.com wrote:

 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 aboaz...@gmail.com wrote:
 
 Hi All, 
 
 I was practicing the cisco Documentation CD that will be available during 
 the lab http://www.cisco.com/cisco/web/psa/default.html 
 to get used on it, I was searching for the topic in unity connection How to 
 set up the Phone View ,this topic is mentioned in the System Administration 
 for Cisco Unity Connection Release 7.X 
 (http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/administration/guide/7xcucsag245.html)
 I was not able to find the path in the Doc CD, I used Google to fine this 
 guide from cisco site.
 
 Do you guys practice the Doc CD - If not, then you have to, as this is the 
 only documentation resource during the exam..-, 
 
 do you know the Path to the system administration guide for the products.
 
 Awaiting your response, Appreciate that.
 
 ___
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 visit www.ipexpert.com
 
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 www.PlatinumPlacement.com
 ___
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Re: [OSL | CCIE_Voice] Rumours or True

2013-01-14 Thread Cory Gray
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 going to determine 
if you are spending the rest of the time trying to pick up points or double 
check.  I have been looking for points in my last 2 hours but am confident I 
will be double checking next attempt :) The only thing I do not verify on he 
first time through is every single dialplan call.  I configure it completely 
but it is easy to miss one 4 to 10 conversion or something else simple if you 
do not make the call.  The would be caught on the doublecheck.

Sent from my iPhone

On Jan 14, 2013, at 1:01 PM, Chrysostomos Christofi 
ch.christ...@logicom.net wrote:

 Guys
  
 they say that you have to be very fast in the lab
 what actual that means?
  
 do you have time for check every task?
  
 you can check every task when you finish it ,or if you check every task then 
 you will not have a time to finish the remaining lab
  
 can you pls send your feedback?
  
  
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Re: [OSL | CCIE_Voice] Configuriong Conferance that used with LiveRecord in CUE

2013-01-14 Thread Cory Gray
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 aboaz...@gmail.com wrote:

 Hello Guys,
 
 The following question passed my mind while I am practicing the LiveRecord 
 feature under the CUE topic.
 
 What type of conference should we used for the purpose  of configuring the 
 Liverecord under CUE task, is it the 
 Ad-hoc conference  , just like below configuration example :
 Ephone-dn 1 dual-line
 Number A101
 Conference Ad-hoc
 !
 Ephone-dn 2 dual-line
 Number A101
 Conference Ad-hoc
 !
 Ephone-dn 3 dual-line
 Number A101
 Conference Ad-hoc
 !
 Ephone-dn 4 dual-line
 Number A101
 Conference Ad-hoc
 OR
 Use the 
 Hardware conference bridge
 and based on which conditions shall we chose between the above two options .
 
 Thanks 
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Re: [OSL | CCIE_Voice] DSPs resources for CCIE-V Lab

2013-01-14 Thread Cory Gray
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 ma7...@gmail.com wrote:

 Hi Team
 
 I'm newest on this group.
 In a kindly way, I'm asking if somebody could tell me the DSPs
 resources for CCIE Lab (HQ, SA, SB and PSTN GWs).
 How many DSPs do I need for every router?
 I'm building my own lab
 
 Thank in advance
 Martin Lopez
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Re: [OSL | CCIE_Voice] voice lab scores

2013-01-12 Thread Cory Gray
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 are
either going to get 100% or around 70% if you feel you nailed everything.
Go figure where you are losing points.  I wish I had a better answer for you
other than double check everything at the end of your lab matches up to the
task AND the preconfigured stuff matches to what is in all of the diagrams.

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Barrera, Hugo
Sent: Saturday, January 12, 2013 1:48 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] voice lab scores

 

Hi,

 

So I received my results from taking my Lab for the first time last week, I
did NOT pass, and thought I had at least nailed the network piece. I  had
everything working I banged out the origin file in a few minutes, I had all
my phones up and pulling from their proper dhcp scopes, I'm trying to figure
out why they scored me so low, it was really low.

 

Any thoughts on this would be much appreciated!

 

- Hugo 

 

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Re: [OSL | CCIE_Voice] Software Versions

2013-01-10 Thread Cory Gray
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 Cisco web site says that any version of software that has been available
for 6 months is open game for the CCIE Voice Lab.  Just wondering what
version most people use for their lab studies?  Some I've talked to are
using 8.6 while others are sticking with 7.1(5). 

 

What do you recommend?

 

--

Todd 

 

 

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Re: [OSL | CCIE_Voice] ISDN Channel not available :D

2013-01-09 Thread Cory Gray
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...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Nicolas MICHEL
Sent: Wednesday, January 09, 2013 12:49 PM
To: Bill
Cc: OSL Voice
Subject: Re: [OSL | CCIE_Voice] ISDN Channel not available :D

Hi Guys,


no the HQ is H323 otherwise I would not change the ISDN bchan order in the
IOS but rather in the CUCM interface.

the ISDN status shows Multi Frame Establish, all is all right. I can have
incoming calls to the HQ but not outgoing calls to the Local and LD route
pattern.
 From what I remember, International calls are working great :(


Next time I'll debug Q931 with detail enabled


Thanks for the help :)


Nic







Le 1/9/2013 11:41 AM, Bill a écrit :
 So is you hq a mgcp? If so it looks like you are missing your isdn 
 bind-l3 ccm command

 Is it h323 or maybe stand alone cme with just a plan pri?

 If h323 the does it show status in gateways? Does it show unknown unknown
instead of unknown and ip?

 Try to give us more and clearer information because right now all we know
is pri, but most likely hq is a gateway for CUCM and we don't know what
type.  We don't know if this is your lab, rack rental or real world.  It is
very helpful to have this information to be able to give you help.



 Bill


 On Jan 9, 2013, at 3:38 AM, Heath Williams heath...@gmail.com wrote:

 Hi,

 I had a similar issue on a production network from a cm to cme on net.

 Look at your dial-peers and also your class codec applied to them.

 Also try using the ccapi inout and the asn 245 debugs to fault find.

 Hope this helps.

 Sent from my iPhone

 On 09/01/2013, at 8:03 PM, Nicolas MICHEL mcl.nico...@gmail.com wrote:

 Hey Guys !

 I feel like I am doomed :D
 When I am calling 911/9911 from HQ, the calls fails.
 It makes the PSTN phones ring and then it automagically disconnect the
call.

 ISDN Q931 debugs says that the requested circuit is not available, I 
 have tried isdn bchan ascending and descending order but with NO 
 luck :(

 The problem is that the incoming calls from the PSTN on both channel 
 (1 and 3 = Fractionnal PRI) are working properly :D


 If anyone has an idea I would be glad to hear it :D


 Thanks !!

 Nic !!




 HQ-RTR#
 HQ-RTR#
 HQ-RTR#
 HQ-RTR#
 *Jan  9 08:58:55.739: ISDN Se0/0/0:23 Q931: pak_private_number: 
 Invalid type/plan 0x0 0x0 may be overriden; sw-type 13 *Jan  9 
 08:58:55.743: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type 
 0xD is 0x2 0x1, Calling num 2123945002 *Jan  9 08:58:55.743: ISDN
Se0/0/0:23 Q931: Applying typeplan for sw-type 0xD is 0x0 0x0, Called num
911 *Jan  9 08:58:55.743: ISDN Se0/0/0:23 Q931: TX - SETUP pd = 8 callref =
0x0081
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98383
Exclusive, Channel 3
Calling Party Number i = 0x2181, '2123945002'
Plan:ISDN, Type:National
Called Party Number i = 0x80, '911'
Plan:Unknown, Type:Unknown *Jan  9 08:58:55.775: ISDN 
 Se0/0/0:23 Q931: RX - CALL_PROC pd = 8  callref = 0x8081
Channel ID i = 0xA98383
Exclusive, Channel 3
 *Jan  9 08:58:55.787: ISDN Se0/0/0:23 Q931: RX - ALERTING pd = 8
callref = 0x8081
Pro
 HQ-RTR#gress Ind i = 0x8188 - In-band info or appropriate now 
 available *Jan  9 08:58:55.831: ISDN Se0/0/0:23 Q931: TX - DISCONNECT
pd = 8  callref = 0x0081
Cause i = 0x80AC - Requested circuit/channel not available 
 *Jan  9 08:58:55.843: ISDN Se0/0/0:23 Q931: RX - RELEASE pd = 8 
 callref = 0x8081 *Jan  9 08:58:55.847: ISDN Se0/0/0:23 Q931: TX - 
 RELEASE_COMP pd = 8  callref = 0x0081 HQ-RTR# HQ-RTR# HQ-RTR# 
 HQ-RTR#config t Enter configuration commands, one per line.  End 
 with CNTL/Z.
 HQ-RTR(config)#int s0/0/0:23
 HQ-RTR(config-if)#isdn bchan as
 HQ-RTR(config-if)#
 *Jan  9 08:59:26.131: ISDN Se0/0/0:23 Q931: pak_private_number: 
 Invalid type/plan 0x0 0x0 may be overriden; sw-type 13 *Jan  9 
 08:59:26.131: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type 
 0xD is 0x2 0x1, Calling num 2123945002 *Jan  9 08:59:26.131: ISDN
Se0/0/0:23 Q931: Applying typeplan for sw-type 0xD is 0x0 0x0, Called num
911 *Jan  9 08:59:26.135: ISDN Se0/0/0:23 Q931: TX - SETUP pd = 8 callref =
0x0082
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98381
Exclusive, Channel 1
Calling Party Number i = 0x2181, '2123945002'
Plan:ISDN, 

Re: [OSL | CCIE_Voice] General questions on Lab

2013-01-06 Thread Cory Gray
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.  But setting up DHCP correctly is 
only worth a few points so you could easily still pass if everything else with 
your phones is correct in theory.

Sent from my iPhone

On Jan 6, 2013, at 8:21 PM, sanity insanity 
networksanitytoinsan...@gmail.com wrote:

 hi Guys,
 
 Just need to know the following
 
 1) Is it required for every END USER to be associated with their 
 corresponding username  on  Directory number level of the phone   or  is it 
 enough if we can  just  do this on the END USER page of the callmanger by 
 associating  the MAC address of the phone  to  the end user and specifying 
 his primary extension , IPCC extension and roles there ?
 
 2) Is DHCP assignment and auto registration does not work for any reason can 
 we just use manual registration to save on time?
 
 
 -MJ
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Re: [OSL | CCIE_Voice] Unity connection recording for UCCX prompts

2013-01-04 Thread Cory Gray
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 provided for you or you would have to record it for added
complexity.  I seriously doubt they would go further and tell you what
method to use.

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Derek Wyss
Sent: Friday, January 04, 2013 10:10 AM
To: William Bell
Cc: ccie_voice@onlinestudylist.com Voice; singh
Subject: Re: [OSL | CCIE_Voice] Unity connection recording for UCCX prompts

 

Bill,

I haven't personally seen a scenario with the recording script not working.
Unless they specifically ask for 1 way or the other.

Derek

On Fri, Jan 4, 2013 at 9:06 AM, William Bell b...@ucguerrilla.com wrote:

Derek,

 

Is it possible to expand on your statement without violating NDA? I ask
because I struggle trying to imagine a scenario where I could get to UCCX to
run the script that plays the prompts but I would be unable to create a
script that records the prompts (thus forcing me to use CUC or some other
method). 

 

-Bill

--

William Bell

blog: http://ucguerrilla.com

twitter: @ucguerrilla

 

 

 

On Jan 4, 2013, at 7:58 AM, Derek Wyss wrote:





I would recommend knowing how to do it both ways as certain circumstances
might require it.

Derek

On Thu, Jan 3, 2013 at 11:21 PM, William Bell b...@ucguerrilla.com wrote:

I assume everyone has their own approach here. I do the following:

 

1. For Unity Connection recordings (call handlers) I use CUGA

 

2. For UCCX prompts, I write a script in UCCX and record/upload the prompts
from the UCCX server

 

3. For BACD prompts, I use the UCCX to record the prompt / upload to
UCM-TFTP / TFTP copy the file to flash

 

4. For CUE prompts, I use the CUE prompt management app

 

-Bill

--

William Bell

blog: http://ucguerrilla.com http://ucguerrilla.com/ 

twitter: @ucguerrilla

 

 

 

On Jan 4, 2013, at 12:02 AM, singh wrote:

 


HI Guys,

I am planning to use Unity connection to record and download prompts for the
UCCX scripts . I am just wondering if this is the best approach or a
recording script needs to be written on UCCX.


Also from machine on which UCCX is installed can the Unity connection web
interface be accessed directly ?


-singh



 
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[OSL | CCIE_Voice] GK to CUE Direct Call

2013-01-04 Thread Cory Gray
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.

 

CUCM  Site A GK  Site C Gateway  CUE

 

Does anyone know what extra configuration is need to accomplish this?

 

Again, this Setup works fine

 

CUCM  Site A GK  Site C Gateway  SiteC Phone 1 CUE

 

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Re: [OSL | CCIE_Voice] GK to CUE Direct Call

2013-01-04 Thread Cory Gray
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 call incoming on the cme? debug voip dial-peer?

Am Freitag, 4. Januar 2013 schrieb Cory Gray :

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.

 

CUCM  Site A GK  Site C Gateway  CUE

 

Does anyone know what extra configuration is need to accomplish this?

 

Again, this Setup works fine

 

CUCM  Site A GK  Site C Gateway  SiteC Phone 1 CUE

 

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Re: [OSL | CCIE_Voice] PHONES REGISTRATION ISSUE WITH SITES

2013-01-04 Thread Cory Gray
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 your
DHCP Server configuration has the TFTP Servers on it.  If you have all of
that then you need to make sure you have bi-directional communication
between your Server Subnet and phone subnets.  Do a debug ip dhcp events
to see the DHCP messages.

 

I cannot speak to phoneview.  I stopped using it because it was not working
as I expected and use local phones.

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of SAIKAT SEN
Sent: Friday, January 04, 2013 4:11 PM
To: Derek Wyss
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] PHONES REGISTRATION ISSUE WITH SITES

 

Thanks for your reply !!! Actually I had created DHCP Server on CUCM_PUB. I
was trying to register those phones manually. Am I missing something here ??
What is the right  way to do that ?? Waiting for reply !!

  _  

Date: Fri, 4 Jan 2013 07:00:22 -0600
Subject: Re: [OSL | CCIE_Voice] PHONES REGISTRATION ISSUE WITH SITES
From: wys...@gmail.com
To: saikat...@hotmail.com
CC: ccie_voice@onlinestudylist.com

Your screenshots don't really help much.  Show us your dhcp config for site
B and HQ.  Are you using dhcp helper or local dhcp servers?  Please
elaborate...

Thanks,

Derek

On Fri, Jan 4, 2013 at 1:48 AM, SAIKAT SEN saikat...@hotmail.com wrote:

Hello Friends !!

I am practicing my own lab. I had created three
sites, HQ , SiteB, SiteC and I was trying to register local phones with
those sites. SiteC local phone register with CUCM-SUB easily, but having
trouble to register SiteB and HQ local phones  with CUCM-SUB. I am using
PhoenViewer to access phones remotely. I was getting phone display and it
was showing DN number which one I configure in SiteC, but when I was trying
to refresh the phone. IT showing error and could not able to dail any
number. Please !!! help friends !!! your help will be much appreciated !!


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Re: [OSL | CCIE_Voice] OWLE Lab 4 : Task 6.3 :Voicemail Forward to Alternate extension

2013-01-04 Thread Cory Gray
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 b...@ucguerrilla.com wrote:

 Check CUCM service parameters Display Original Calling Number on Transfer 
 from Unity. Set this to true.
 
 HTH.
 
 -Bill
 --
 William Bell
 blog: http://ucguerrilla.com
 twitter: @ucguerrilla
 
 
 
 On Jan 4, 2013, at 6:22 PM, Ramcharan Arya wrote:
 
 Hello,
 
 Have issue with particular task:
 
 When someone call ext 2002 ( Gwashington) during voicemail greeting when 
 press 9 then call should forward to external number 4678124 ( PSTN Line).
 
 This is working as expected.
 
 Below is debug isqn q931 output:
 
 HQ-RTR(config)#
 Jan  4 23:09:57.642: ISDN Se0/0/0:23 Q931: TX - DISCONNECT pd = 8  callref 
 = 0x0004 
 Cause i = 0x8290 - Normal call clearing
 Jan  4 23:09:57.654: ISDN Se0/0/0:23 Q931: RX - RELEASE pd = 8  callref = 
 0x8004
 Jan  4 23:09:57.686: ISDN Se0/0/0:23 Q931: TX - RELEASE_COMP pd = 8  
 callref = 0x0004
 Jan  4 23:12:44.994: ISDN Se0/0/0:23 Q931: TX - SETUP pd = 8  callref = 
 0x0005 
 Bearer Capability i = 0x8090A2 
 Standard = CCITT 
 Transfer Capability = Speech  
 Transfer Mode = Circuit 
 Transfer Rate = 64 kbit/s 
 Channel ID i = 0xA98388 
 Exclusive, Channel 8 
 Display i = 'VoiceMail' 
 Called Party Number i = 0xC1, '4678124' 
 Plan:ISDN, Type:Subscriber(local)
 Jan  4 23:12:45.042: ISDN Se0/0/0:23 Q931: RX - CALL_PROC pd = 8  callref = 
 0x8005 
 Channel ID i = 0xA98388 
 Exclusive, Channel 8
 Jan  4 23:12:45.058: ISDN Se0/0/0:23 Q931: RX - ALERTING pd = 8  callref = 
 0x8005 
 Progress Ind i = 0x8188 - In-band info or appropriate now available 
 Jan  4 23:12:55.886: ISDN Se0/0/0:23 Q931: TX - DISCONNECT pd = 8  callref 
 = 0x0005 
 Cause i = 0x8290 - Normal call clearing
 
 This is no sending any calling number when Unity make call to external 
 phone. This does not meet Telco requirements.
 
 Is there any way to fix this problem when call goes out to external number 
 from PSTN it should show calling number in isdn q931. Any thought.?
 
 Thanks  Regards,
 Ramcharan Arya
 
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Re: [OSL | CCIE_Voice] Lab question call routing

2012-12-22 Thread Cory Gray
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...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Rrcrumm
Sent: Saturday, December 22, 2012 10:15 PM
To: Ccie
Subject: [OSL | CCIE_Voice] Lab question call routing

Hello,
Do you lose points in the call routing section if your SRST call routing is
not sending the correct ANI and/or DNIS  or if OB calls from Unity
Connections are not sent with the correct ANI and/or DNIS? 
 
Or do you just lose points for that particular question.
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Re: [OSL | CCIE_Voice] Documents and other details

2012-12-20 Thread Cory Gray
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 work.  I forget if food is allowed at the 
desk.  

 

No documentation is allowed.  Any documentation allowed is through the Cisco 
Support Pages accessed through Internet Explorer.  You will also have CUCM, 
CUCME, CUCCX, and QoS SRND on your desktop.  You will have a desktop with a 
monitor that is so big is like having two monitors.  The keyboard is a standard 
keyboard so if you practice on your laptop, it may take you a few minutes to 
adjust to minor physical differences but of course it is still a QWERTY 
keyboard.  

 

No electronics including car keys that have remotes built in or separate fobs…. 
even wrist watches are allowed at your desk.  This is stop cheaters from 
stealing the lab.  Cell Phones must be completely off.  All electronics will go 
in the locker in the lab so they are safe.  If a cell phone vibrates, the 
proctor will make everyone stop the lab, double check their phones, and the 
time will not be given back.  I just leave mine in the car.

 

 

 

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of virajith 
Sent: Wednesday, December 19, 2012 10:19 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Documents and other details

 

hi Guys,

I have a question regarding if we need to carry any identity documents when we 
go to the  exam center such as
- driver's license or passport photocopy .
- Also do we need to carry lunch and water or is  it provided?
- any other details  or documents we need to carry with us when we go to the 
exam center for taking the exam?
- Is it a desktop they give us to operate? Does it have 2 monitors?

Thanks,
Vir



 
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Re: [OSL | CCIE_Voice] Procedure to check if Unity Connection , UCCX and Presence server ( CUPS) are working with CUCM or not?

2012-12-20 Thread Cory Gray
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 my opinion it 
would be too much out of the way to test up front.  Even if you found 
something, good luck convincing the proctor that there is really something wrong

Sent from my iPhone

On Dec 20, 2012, at 11:07 PM, Symon Phares symonpha...@gmail.com wrote:

 you will need to do the tests from cup n unity connection side. On unity 
 connection, run check configuration from phone system menu and on cups on 
 system troubleshooter dasshboard.
 
 Regards
 Symon Phares
 
 From: sanity insanity networksanitytoinsan...@gmail.com
 Sent: Friday, December 21, 2012 06:45
 To: ccie_voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice] Procedure to check if Unity Connection , UCCX and 
 Presence server ( CUPS) are working with CUCM or not?
 
 hi Guys,
 
 I was just wondering if there is a short procedure or check we can use before 
 starting any configurations  just to see if the integration of CUCM with the 
 following servers is working fine or not...
 
 1) Cisco Unity Connection
 2) UCCX
 3) CUPS
 
 
 -I know if can to ping tests.  Is there any other way to confirm if the 
 integration is working fine?
 
 Thanks,
 MJ
 
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Re: [OSL | CCIE_Voice] CUPS- Presence - deskphone mode not active

2012-12-19 Thread Cory Gray
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...@rediffmail.com; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CUPS- Presence - deskphone mode not active

 

Hi Vir,

Please ensure following is correct in your setup.

* Verify that user devices are registered to the CUCM.
* Verify that the end user is associated with the IP phones in end user
configuration.
* Ensure that the CCMCIP profile in CUPsever is applied to the end user.
* Verify in CUCM that the device and directory number can be controlled by
CTI.
 
Carlos

  _  

Date: Wed, 19 Dec 2012 05:56:57 +
To: ccie_voice@onlinestudylist.com
From: vir...@rediffmail.com
Subject: [OSL | CCIE_Voice] CUPS- Presence - deskphone mode not active

hi Guys,

In the client I get the option of deskphone mode in CUPC but when I select
it  the mode does not get applied.

In show server health I see the Desk phone mode says  Partially
Connected(Primary) - Cannot Connect to Phone.

I have checked the cti server / gateway profile and it is fine.

What else to check?

-Vir

 
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Re: [OSL | CCIE_Voice] unable to get phones on phoneviewer

2012-12-19 Thread Cory Gray
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: .
Subject: [OSL | CCIE_Voice] unable to get phones on phoneviewer

 

Hello Guys !!

 I am finding difficulty to access phones remotely on
phoneviewer. I followed all the steps from the config ebook. but its not
showing any phones.  I am practicing my own labs. Please !! help guys  

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Re: [OSL | CCIE_Voice] CUE LIC FILE

2012-11-27 Thread Cory Gray
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 
ch.christ...@logicom.net wrote:

  
 Hi folks
  
 I have the cue license for proctor labs
  
 I am wondering if I can use it for my cisco router (for lab) which not has a 
 AIM cue module but ISM
 Is that possible?
  
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Re: [OSL | CCIE_Voice] QUESTION FOR THE LAB

2012-11-22 Thread Cory Gray
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 Christofi 
ch.christ...@logicom.net wrote:

 Hi
  
 If we have cue in the lab then we CAN proceed with the initialization through 
 the web interface?
 Or we have to proceed through cli
  
 Or you can go with both ways
  
 I hope that my question is not violate  any policies !
  
 Regards
 cc
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Re: [OSL | CCIE_Voice] QOS confusion..

2012-11-22 Thread Cory Gray
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  vir...@rediffmail.com wrote:

 mls qos map cos-dscp 0 8 16 24 32 46 48 56
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Re: [OSL | CCIE_Voice] QUESTION FOR THE LAB

2012-11-22 Thread Cory Gray
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 initialization is just 
answering questions so I am not sure why you would want to do this through GUI 
anyway.

Sent from my iPhone

On Nov 22, 2012, at 9:41 AM, Chrysostomos Christofi 
ch.christ...@logicom.net wrote:

 Guys
  
 I am wondering if we can’t get in the web qui
  
 Is that possible?
  
  
  
  
 From: Edgar Feliz [mailto:ejzi...@gmail.com] 
 Sent: Πέμπτη, 22 Νοεμβρίου 2012 4:36 μμ
 To: Chrysostomos Christofi
 Cc: Online Study (ccie_voice@onlinestudylist.com)
 Subject: Re: [OSL | CCIE_Voice] QUESTION FOR THE LAB
  
 Unless it is specifically stated to use one or the other, use whatever you 
 feel comfortable with as long as the tasks are completed correctly.
  
 Edgar
  
 On Thu, Nov 22, 2012 at 9:05 AM, Chrysostomos Christofi 
 ch.christ...@logicom.net wrote:
 Hi
  
 If we have cue in the lab then we CAN proceed with the initialization through 
 the web interface?
 Or we have to proceed through cli
  
 Or you can go with both ways
  
 I hope that my question is not violate  any policies !
  
 Regards
 cc
 
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Re: [OSL | CCIE_Voice] IPPM

2012-11-11 Thread Cory Gray
Yes

Sent from my iPhone

On Nov 11, 2012, at 10:16 AM, Chrysostomos Christofi 
ch.christ...@logicom.net wrote:

 Hi
  
 For IPPM the pin is from the end user  configuration tab (PIN?)
  
  
 Regards
  
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Re: [OSL | CCIE_Voice] EI PRI

2012-11-03 Thread Cory Gray
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-port, 0
is channel 1.  This way you do not get confused but do whatever is most
comfortable for you. 1-13 is 13 channels either way.

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Chrysostomos
Christofi
Sent: Saturday, November 03, 2012 12:57 PM
To: Online Study (ccie_voice@onlinestudylist.com)
Subject: [OSL | CCIE_Voice] EI PRI

 

Hi Folks

 

I got a little confused

 

If I need 13  channels only ,  then the below is correct?

 

Controller e1 0/1/1

Pri-group times 1-12,16 

 

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Re: [OSL | CCIE_Voice] SRST CUE MGCP - Keeping it All Straight!

2012-11-03 Thread Cory Gray
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-SRST that is also a valid configuration.  So you do not always need 
telephony-service or callmanger-fallback. 

You can source moh from CME or SRST.  What I do is read ahead in my lab and see 
if the site needing moh uses SRST and that determines whether I use CME or 
SRST.  If that site is CME-SRST, then configure moh in telephony service.  If 
traditional SRST configure it under callmanager-fallback.

I hope that helps.  The more specific examples you give, the more we can help.

Sent from my iPhone

On Nov 3, 2012, at 6:40 PM, Brandon brana...@yahoo.com wrote:

 I'm having trouble keeping straight all of the various voice router configs 
 and how they interact and relate to each other and how even if you are 
 running MGCP or SRST or CUE or SIP you still need some commands present from 
 the other sections.
 
 ccm-manager fallback-mgcp
 ccm-manager mgcp
 mgcp
 call-manager-fallback
 telephony-service
 voice register global
 
 I think I feel comfortable in knowing you use either call-manager-fallback or 
 telephony-service to specify SRST or CUE respectively.
 
 But where I get lost is then how MGCP and ccm-manager come into the picture 
 for each type of setup (SRST vs. CUE)
 
 For example, when I'm running MGCP and my GW is registered to CUCM without 
 problems, it seems the call-manager-fallback section is needed to source MOH 
 from the router if I don't want multicast across my WAN.  But if it isn't in 
 fallback mode the moh commands are still relevant for MOH to be served from 
 the router.  And then you still need ccm-manager music-on-hold to make it 
 work on top of that.
 
 Then if SIP is thrown into the picture it just gets worse from there!
 
 I need to figure out a good way to know this with confidence.
 I don't think I could sit down and type from scratch a config I'm comfortable 
 knowing would work for the various scenarios.  I'm not even sure at this 
 point how many scenarios you can have setup.
 
 Do you have any info that can help us keep it all straight?
 
  - Brandon
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Re: [OSL | CCIE_Voice] ESW and switch config

2012-11-01 Thread Cory Gray

IPexpert and myself have run into problems doing it the traditional way.  But 
for the authoritative source you should be looking at the LAN Switching Guide 
for IOS 12.4T which confirms that recommendation.  The guide you point to is 
for Metro 3750 switches.  The ISR routers in the lab run 12.4T.

http://www.cisco.com/en/US/docs/ios-xml/ios/lanswitch/configuration/12-4t/lsw-hwic-ethsw-ic.html#GUID-379450C0-5434-4AC3-9BED-396CB7D162C1



Date: Thu, 1 Nov 2012 08:08:36 +
Subject: Re: [OSL | CCIE_Voice] ESW and switch config
From: stbruen...@gmail.com
To: corygray22...@hotmail.com
CC: vir...@rediffmail.com; ccie_voice@onlinestudylist.com

Recommed by whom? Ipexpert, I know, but the problem is that the 
Proctors/Scripts which marks your lab are not from IPExpert they are from 
Cisco. Therefore it should be better to follow the cisco guidelines:

You should configure voice VLAN 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 corygray22...@hotmail.com

You should be fine without one just know that the recommended ESW config is
Switchport mode trunkSwitch 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 Of virajith 

Sent: Wednesday, October 31, 2012 11:22 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] ESW and switch config
 hi Guys,

I am just wondering if one can practice the labs without a ESW module .

I have a setup in with I am using a switch (3750) for the vlan config.


Is an ESW module necessary for the lab practice? 

How is the above config on 3750 different from using an ESW module?


-Vir


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www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] Uccx scripting making me mad!!

2012-11-01 Thread Cory Gray
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 have a Windows Server license
so I just rebuild it every 30 days when the server evaluation expires

2.   Build your integration.  I had to learn the basics like CUCCX
configures CUCM accounts, CTI ports, and CTI route points for you.  Some
that has the smallest amount of experience knows this.

3.   Take time to really understand the integration in case you have to
troubleshoot it in your lab

4.   Start off testing basic skills based routing and resource group
routing using the default icd.aef script

5.   Now that you understand how UCCX works it is time to working on
custom scripting

6.   Open up the script writing application, find and choose icd.aef,
and before doing anything, do a save as to your desktop with a custom name
of your choosing.  You do not want to edit the default icd.aef script.

7.   This is where the IPexpert workbooks come in.  I would recommend
the 5 lab handbook and One-Week Lab experience labs for custom scripting.  I
have not tried the Workbook 2 ones so maybe someone else can comment on
those.  Copy all of the lab scripts onto your CUCCX server.  Have IPexperts
script side by side with your new based on icd.aef.  Double Clicking on
IPexpert's script while you are already in the script building application
will automatically bring up a second window and you will be able to see both
sripts.  When you are building your script to match IPexpert's script, you
will start to understand how it works.  You will understand how to create
variables, do if then analysis, create prompts, etc.

 

I cannot do all of their scripts without help from looking at the final
script but am confident that I can troubleshoot an existing script in the
lab or build one of medium complexity from scratch.  I may use this next 30
days to master the scripts but from what I have seen in the lab, I think I
am prepared enough but you never know!

 

HTH

 

Cory

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of sanity insanity
Sent: Thursday, November 01, 2012 12:41 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Uccx scripting making me mad!!

 

hi Guys,

I really need your help to understand UCCX scripts ...How they are made? and
how they work?

Please help guys!

-Mark

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Re: [OSL | CCIE_Voice] ESW and switch config

2012-11-01 Thread Cory Gray
I have not seen bypass all of the spanning tree states in IPexpert's labs
and I cannot comment on what I have seen in the real lab.  If portfast is a
requirement I guess you have to do what you have to do but I will say this
again.. I believe I lost all points on my CUCME section because my phones
were down when my lab was graded.  Using switchport mode access and switch
access vlan X on the ESW has been known to randomly stop working.  Use those
commands at your own risk as I mistakenly did.

 

Cory

 

From: Steffen Bruening [mailto:stbruen...@gmail.com] 
Sent: Thursday, November 01, 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 the vlan
section because in most cases it is required to bypass all the spanning-tree
states or just that the phones should boot up as fast as possible.

 

Regards

Steffen

2012/11/1 Steffen Bruening stbruen...@gmail.com

Oh okay. I apologize for the confusion I made.

 

2012/11/1 Cory Gray corygray22...@hotmail.com

IPexpert and myself have run into problems doing it the traditional way.
But for the authoritative source you should be looking at the LAN Switching
Guide for IOS 12.4T which confirms that recommendation.  The guide you point
to is for Metro 3750 switches.  The ISR routers in the lab run 12.4T.

http://www.cisco.com/en/US/docs/ios-xml/ios/lanswitch/configuration/12-4t/ls
w-hwic-ethsw-ic.html#GUID-379450C0-5434-4AC3-9BED-396CB7D162C1





  _  

Date: Thu, 1 Nov 2012 08:08:36 +
Subject: Re: [OSL | CCIE_Voice] ESW and switch config
From: stbruen...@gmail.com
To: corygray22...@hotmail.com
CC: vir...@rediffmail.com; ccie_voice@onlinestudylist.com



Recommed by whom? Ipexpert, I know, but the problem is that the
Proctors/Scripts which marks your lab are not from IPExpert they are from
Cisco. Therefore it should be better to follow the cisco guidelines:

 

You should configure voice VLAN 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 corygray22...@hotmail.com

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 Of virajith 
Sent: Wednesday, October 31, 2012 11:22 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] ESW and switch config

 

hi Guys,

I am just wondering if one can practice the labs without a ESW module .

I have a setup in with I am using a switch (3750) for the vlan config.

Is an ESW module necessary for the lab practice? 

How is the above config on 3750 different from using an ESW module?


-Vir

 
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Re: [OSL | CCIE_Voice] Re-Built PUB

2012-11-01 Thread Cory Gray
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
to work correctly even though my database replication showed good.  I would
just rebuild or reload fresh VMs if you have them

 

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Chrysostomos
Christofi
Sent: Thursday, November 01, 2012 9:22 AM
To: Craig Hill (crahill)
Cc: Online Study (ccie_voice@onlinestudylist.com)
Subject: Re: [OSL | CCIE_Voice] Re-Built PUB

 

Hi

 

I have followed the proposed solution but

 

On PUB

admin:utils dbreplication reset all

 


**

Replication Reset cannot be performed on a cluster with a single active node

Aborting replication reset operation

***

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Chrysostomos
Christofi
Sent: Πέμπτη, 1 Νοεμβρίου 2012 3:08 μμ
To: Craig Hill (crahill); Online Study (ccie_voice@onlinestudylist.com)
Subject: Re: [OSL | CCIE_Voice] Re-Built PUB

 

Hi craiq

 

Thank you very much

 

 

I will try it and will let you know

Do you know why in the admin page of PUB everything is unknown?Is because
of the replication issue?

 

Regards

 

 

 

 

From: Craig Hill (crahill) [mailto:crah...@cisco.com] 
Sent: Πέμπτη, 1 Νοεμβρίου 2012 3:06 μμ
To: Chrysostomos Christofi; Online Study (ccie_voice@onlinestudylist.com)
Subject: RE: Re-Built PUB

 

Try the following:

 

 

1. Sub: utils dbreplication stop (wait for it to complete before going to
step 2.)

2. Pub: utils dbreplication stop (wait for it to complete before going to
step 3.)

3. Pub: utils dbreplication reset all (monitor the dbstatus in RTMT for all
2's.)

4. Reboot the cluster and re-verify in RTMT.

 

This resolves most issues, and I do this after each reset of my lab in ESXi.
If this doesn't work, you may need to rebuild your subscriber

 

Craig




 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Chrysostomos
Christofi
Sent: Thursday, November 01, 2012 7:40 AM
To: Online Study (ccie_voice@onlinestudylist.com)
Subject: [OSL | CCIE_Voice] Re-Built PUB

 

 

Hi

 

If you can help with that it will be appreciated

I have re-installed pub and then the replication is not working any more

Do you have something to propose?

 

admin:utils dbreplication status 

 

  utils dbreplication status  

 

 

Status cannot be performed when replication is down on the publisher, or on
a cluster with a single active node or a cores node; aborting replication
status check operation

 

From the SUB

 

 

SERVER ID STATESTATUS QUEUE  CONNECTION CHANGED

---

g_ccm1_ccm7_1_5_31900_32 Active   Dropped861678 Nov  1 14:20:15
(PUB)

g_cmprimary_ccm7_1_5_31900_35 Active   Local   0
(SUB)

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Re: [OSL | CCIE_Voice] Cisco ripped me off

2012-10-31 Thread Cory Gray
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 access because it never
failed. I also did this because using switchport mode trunk would show
nothing in the show vlan-switch command so I was scared this was how it was
being graded and would miss the points.  IPexpert recommends this because
they say the other way has been known to stop working for no reason.

 

When I got my score report the next day, I could see several sections wrong
that I knew I configured right.  Doing the math I believe when they went to
grade my exam the next day that my CUCME phones were no longer registered.
I will use switchport mode trunk for now on.

 

What did you do?  That is my only theory.  Maybe you have one different that
can help others if you choose not to take it again.

 

I will be back 11/30 and am hoping to do as well as I did last time but pass
J

 

Thanks,

 

Cory 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Krishna
Sent: Wednesday, October 31, 2012 8:08 AM
To: Online Study
Subject: Re: [OSL | CCIE_Voice] Cisco ripped me off

 

all,

 

yesterday i took my second attempt in rtp, and i am 200 % sure that i pass
the exam. I got 1 hour left even after testing it thrice, but looking at the
score report i was shocked, and i completely disagree with my score report.
F... CCIE lab script evaluation.. i am completely pissed off 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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Re: [OSL | CCIE_Voice] Cisco ripped me off

2012-10-31 Thread Cory Gray
Krishna,

 

I had some funny things going on with my rack but cannot get into it because
of NDA.  I am extremely frustrated.  I would tell you this.  I am already a
CCIE in RS and work for Cisco.  I am not sure if you have CCIE yet.  I know
it is frustrating, expensive, and time consuming but as you can see from the
last few weeks, several people on this list have passed.  Me and you are so
far along (I was done an hour early also) that the worst thing you can do is
give up now.  All of the effort you put in to get this far will be wasted if
you do not complete your journey.  It took me a few days to get over it.
Get back in there as soon as possible and knock it out!  Especially if you
do not have any CCIE's, passing this exam will be a career defining moment
that will help you more than any project or customer experience you can
think of.

 

Don't quit!

 

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 rather doing it remotely for their
convenience. i used switchport mode trunk, switchport trunk native vlan data
on sb and sc. 

 

 Can anyone expect fail in the exam after evaluating 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.

 

  _  

From: Cory Gray corygray22...@hotmail.com
To: 'Krishna' vinayak_...@yahoo.com; 'Online Study'
ccie_voice@onlinestudylist.com 
Sent: Wednesday, October 31, 2012 7:41 AM
Subject: RE: [OSL | CCIE_Voice] Cisco ripped me off

 

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 access because it never
failed. I also did this because using switchport mode trunk would show
nothing in the show vlan-switch command so I was scared this was how it was
being graded and would miss the points.  IPexpert recommends this because
they say the other way has been known to stop working for no reason.

 

When I got my score report the next day, I could see several sections wrong
that I knew I configured right.  Doing the math I believe when they went to
grade my exam the next day that my CUCME phones were no longer registered.
I will use switchport mode trunk for now on.

 

What did you do?  That is my only theory.  Maybe you have one different that
can help others if you choose not to take it again.

 

I will be back 11/30 and am hoping to do as well as I did last time but pass
J

 

Thanks,

 

Cory 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Krishna
Sent: Wednesday, October 31, 2012 8:08 AM
To: Online Study
Subject: Re: [OSL | CCIE_Voice] Cisco ripped me off

 

all,

 

yesterday i took my second attempt in rtp, and i am 200 % sure that i pass
the exam. I got 1 hour left even after testing it thrice, but looking at the
score report i was shocked, and i completely disagree with my score report.
F... CCIE lab script evaluation.. i am completely pissed off 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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Re: [OSL | CCIE_Voice] Cisco ripped me off

2012-10-31 Thread Cory Gray
Krishna,

 

I understand, I have had 100% on call routing in previous labs at the same time 
not having 100% on the gateway section.  That does not make sense.  Cisco does 
not tell how they grade each section but the proctor will tell you it is part 
automatic and part manual for voice.  We are 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

 

Cory,

 

i really appreciated your motivation skills, but looking at the score report i 
am unable to understand where i did wrong and so how can i comprehend myself by 
looking at the lab result, and taking exam one more time. i would say that 
cisco should give us a statement feedback for what had caused 0 points for that 
many tasks, otherwise we would never come to know what we did wrong.

 

thank you

krishna.

 

  _  

From: Cory Gray corygray22...@hotmail.com
To: 'Krishna' vinayak_...@yahoo.com; 'Online Study' 
ccie_voice@onlinestudylist.com 
Sent: Wednesday, October 31, 2012 9:22 AM
Subject: RE: [OSL | CCIE_Voice] Cisco ripped me off

 

Krishna,

 

I had some funny things going on with my rack but cannot get into it because of 
NDA.  I am extremely frustrated.  I would tell you this.  I am already a CCIE 
in RS and work for Cisco.  I am not sure if you have CCIE yet.  I know it is 
frustrating, expensive, and time consuming but as you can see from the last few 
weeks, several people on this list have passed.  Me and you are so far along (I 
was done an hour early also) that the worst thing you can do is give up now.  
All of the effort you put in to get this far will be wasted if you do not 
complete your journey.  It took me a few days to get over it.  Get back in 
there as soon as possible and knock it out!  Especially if you do not have any 
CCIE’s, passing this exam will be a career defining moment that will help you 
more than any project or customer experience you can think of.

 

Don’t quit!

 

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 rather doing it remotely for their convenience. 
i used switchport mode trunk, switchport trunk native vlan data on sb and sc. 

 

 Can anyone expect fail in the exam after evaluating 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.

 

  _  

From: Cory Gray corygray22...@hotmail.com
To: 'Krishna' vinayak_...@yahoo.com; 'Online Study' 
ccie_voice@onlinestudylist.com 
Sent: Wednesday, October 31, 2012 7:41 AM
Subject: RE: [OSL | CCIE_Voice] Cisco ripped me off

 

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 access because it never failed. I also did 
this because using switchport mode trunk would show nothing in the show 
vlan-switch command so I was scared this was how it was being graded and would 
miss the points.  IPexpert recommends this because they say the other way has 
been known to stop working for no reason.

 

When I got my score report the next day, I could see several sections wrong 
that I knew I configured right.  Doing the math I believe when they went to 
grade my exam the next day that my CUCME phones were no longer registered.  I 
will use switchport mode trunk for now on.

 

What did you do?  That is my only theory.  Maybe you have one different that 
can help others if you choose not to take it again.

 

I will be back 11/30 and am hoping to do as well as I did last time but pass J

 

Thanks,

 

Cory 

 

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Krishna
Sent: Wednesday, October 31, 2012 8:08 AM
To: Online Study
Subject: Re: [OSL | CCIE_Voice] Cisco ripped me off

 

all,

 

yesterday i took my second attempt in rtp, and i am 200 % sure that i pass the 
exam. I got 1 hour left even after testing it thrice, but looking at the score 
report i was shocked, and i completely disagree with my score report. F... CCIE 
lab script evaluation.. i am completely pissed off 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. 

 

 

___
For more

Re: [OSL | CCIE_Voice] QoS Question

2012-10-31 Thread Cory Gray
Use 2 bytes for cRTP.  

 

1 FRF.12 G729 call should be as follows:

G729 Sample in bytes – 20

IP/UDP/RTP in bytes – 2 (cRTP)

Layer 2 overhead in bytes – 8 (FRF.12)

Packeterization Rate – 50

Convert bytes to bit (20 + 2 + 8) * 8 = 240

Multiply bits by packet rate 240 * 50 = 12,000 bits = 12k

 

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Craig Hill 
(crahill)
Sent: Wednesday, October 31, 2012 11:21 AM
To: 'Online Study'
Subject: [OSL | CCIE_Voice] QoS Question

 

Hey,

 

I was hoping to get some input on my QoS calculations for header compression. I 
have been using the Cisco 360 program, and in their exercise workbooks it takes 
into account the 4 byte checksum for UDP, but I ran Wireshark in my lab to 
confirm the values. From what I see, the UDP Checksum is disabled on RTP 
traffic which really means the value is 2 bytes for compression. Which value 
should be used? Also, the SRND states 8 bytes for FRF.12 and I have always used 
this, which is contrary to other values I have seen. However, just for 
clarification, is the 8 bytes a factor of the Frame Relay header of 2 bytes + 
the 6 byte fragmentation 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 ripped me off

 

Krishna,

 

I understand, I have had 100% on call routing in previous labs at the same time 
not having 100% on the gateway section.  That does not make sense.  Cisco does 
not tell how they grade each section but the proctor will tell you it is part 
automatic and part manual for voice.  We are 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

 

Cory,

 

i really appreciated your motivation skills, but looking at the score report i 
am unable to understand where i did wrong and so how can i comprehend myself by 
looking at the lab result, and taking exam one more time. i would say that 
cisco should give us a statement feedback for what had caused 0 points for that 
many tasks, otherwise we would never come to know what we did wrong.

 

thank you

krishna.

 

  _  

From: Cory Gray corygray22...@hotmail.com
To: 'Krishna' vinayak_...@yahoo.com; 'Online Study' 
ccie_voice@onlinestudylist.com 
Sent: Wednesday, October 31, 2012 9:22 AM
Subject: RE: [OSL | CCIE_Voice] Cisco ripped me off

 

Krishna,

 

I had some funny things going on with my rack but cannot get into it because of 
NDA.  I am extremely frustrated.  I would tell you this.  I am already a CCIE 
in RS and work for Cisco.  I am not sure if you have CCIE yet.  I know it is 
frustrating, expensive, and time consuming but as you can see from the last few 
weeks, several people on this list have passed.  Me and you are so far along (I 
was done an hour early also) that the worst thing you can do is give up now.  
All of the effort you put in to get this far will be wasted if you do not 
complete your journey.  It took me a few days to get over it.  Get back in 
there as soon as possible and knock it out!  Especially if you do not have any 
CCIE’s, passing this exam will be a career defining moment that will help you 
more than any project or customer experience you can think of.

 

Don’t quit!

 

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 rather doing it remotely for their convenience. 
i used switchport mode trunk, switchport trunk native vlan data on sb and sc. 

 

 Can anyone expect fail in the exam after evaluating 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.

 

  _  

From: Cory Gray corygray22...@hotmail.com
To: 'Krishna' vinayak_...@yahoo.com; 'Online Study' 
ccie_voice@onlinestudylist.com 
Sent: Wednesday, October 31, 2012 7:41 AM
Subject: RE: [OSL | CCIE_Voice] Cisco ripped me off

 

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 access because it never failed. I also did 
this because using switchport mode trunk would show nothing

Re: [OSL | CCIE_Voice] Cisco ripped me off

2012-10-31 Thread Cory Gray
You are correct in that the proctor will tell you all equipment is back in
San Jose.  I have experienced the slowness you speak about in RTP but live
12 minutes from there so I will not be switching locations :)


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Holloway
Sent: Wednesday, October 31, 2012 11:35 AM
To: Leslie Meade
Cc: Online Study
Subject: Re: [OSL | CCIE_Voice] Cisco ripped me off

If memory serves me correctly, if they reach a point in the lab grading
where you've already reached the failing mark, they don't proceed to grade
the rest of the exam and you just get Fail for remaining sections. I cannot
verify that though and it has been two years since I passed.

I took my first CCIE Voice attempt in RTP and additional attempts in San
Jose. I felt the network in San Jose was much faster. It feels like the RTP
network must be connecting back to San Jose, because it was slow in
comparison. There were some odd behaviors in RTP that made my palms sweaty
because I thought things were tanking on the back end of the network, but
everything eventually proceeded normally. I never felt that way in San Jose.



On Oct 31, 2012, at 10:40 AM, Leslie Meade wrote:

 Plus,
 I still think that they will stop marking as soon as they know you won't
pass.
 Sp parts you know you nailed, you will see a zero
 
 
 
 Sent from my iPad
 
 On Oct 31, 2012, at 7:35 AM, Krishna
vinayak_...@yahoo.commailto:vinayak_...@yahoo.com wrote:
 
 Cory,
 
 Technically speaking, the grading has to be evaluated by taking the
seating position where we took the exam rather doing it remotely for their
convenience. i used switchport mode trunk, switchport trunk native vlan data
on sb and sc.
 
  Can anyone expect fail in the exam after evaluating 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.
 
 
 From: Cory Gray
corygray22...@hotmail.commailto:corygray22...@hotmail.com
 To: 'Krishna' vinayak_...@yahoo.commailto:vinayak_...@yahoo.com;
'Online Study'
ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com
 Sent: Wednesday, October 31, 2012 7:41 AM
 Subject: RE: [OSL | CCIE_Voice] Cisco ripped me off
 
 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 access because it never
failed. I also did this because using switchport mode trunk would show
nothing in the show vlan-switch command so I was scared this was how it was
being graded and would miss the points.  IPexpert recommends this because
they say the other way has been known to stop working for no reason.
 
 When I got my score report the next day, I could see several sections
wrong that I knew I configured right.  Doing the math I believe when they
went to grade my exam the next day that my CUCME phones were no longer
registered.  I will use switchport mode trunk for now on.
 
 What did you do?  That is my only theory.  Maybe you have one different
that can help others if you choose not to take it again.
 
 I will be back 11/30 and am hoping to do as well as I did last time but
pass :)
 
 Thanks,
 
 Cory
 
 From:
ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-bounces@onlinestudy
list.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of
Krishna
 Sent: Wednesday, October 31, 2012 8:08 AM
 To: Online Study
 Subject: Re: [OSL | CCIE_Voice] Cisco ripped me off
 
 all,
 
 yesterday i took my second attempt in rtp, and i am 200 % sure that i pass
the exam. I got 1 hour left even after testing it thrice, but looking at the
score report i was shocked, and i completely disagree with my score report.
F... CCIE lab script evaluation.. i am completely pissed off 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 CCNP or CCIE and looking for a job? Check out
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Re: [OSL | CCIE_Voice] ESW and switch config

2012-10-31 Thread Cory Gray
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 Of virajith 
Sent: Wednesday, October 31, 2012 11:22 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] ESW and switch config

 

hi Guys,

I am just wondering if one can practice the labs without a ESW module .

I have a setup in with I am using a switch (3750) for the vlan config.

Is an ESW module necessary for the lab practice? 

How is the above config on 3750 different from using an ESW module?


-Vir

 
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Re: [OSL | CCIE_Voice] Help with Unified FX

2012-10-30 Thread Cory Gray
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 Chaddha
Sent: Wednesday, October 24, 2012 6:49 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Help with Unified FX

 

Hi Guys

I am facing some issues with Unified FX application to get Screenshots of
the phones. I can remote manage the phones alright but I am simply not able
to view their screens.

Here is what I have done:
1. Created an app userpvadmin with Server Monitoring, EM Authentication 
Tab Sync User Groups.
2. Created an end user pview with Standard CTI Enabled.
3. Associated all phones with pview end user.

Here is the error that I receive when I try to see screenshot:
Command (Cmd:Screenshot) sent to device (MAC) using thread (0) with response
(XML Error response from phone)

I am using the Version 2.1.37 which is the latest one. 

Please let me know if anyone has faced similar issues.

Thanks
Mann

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Re: [OSL | CCIE_Voice] access-list not matching any packets for mgcp ports on switch

2012-10-30 Thread Cory Gray
Krishna,

 

Did you get this to work?  If not I will lab it up today.

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Sergey Heyphets
Sent: Wednesday, October 24, 2012 3:43 PM
To: Online Study
Subject: Re: [OSL | CCIE_Voice] access-list not matching any packets for
mgcp ports on switch

 

So you expect MGCP traffic to come from the router to the switch, and where
do you expect it to go? Do you have CUCM connected to that switch or MGCP
phones? 

 

You may want to check if the router actually sends any MGCP traffic over
that port by applying permit ACL with log action on the router. Don't
forget permit any any at the end. 

 

On Wed, Oct 24, 2012 at 2:08 PM, Krishna vinayak_...@yahoo.com wrote:

ip access-list extended 101

permit tcp any any eq 2428

permit udp any any eq2427

permit tcp any eq 2428 any

permit udp 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 corygray22...@hotmail.com
To: Kevin Spicer ke...@kevinspicer.co.uk 
Cc: Krishna vinayak_...@yahoo.com; Online Study
ccie_voice@onlinestudylist.com 
Sent: Wednesday, October 24, 2012 12:32 PM
Subject: Re: [OSL | CCIE_Voice] access-list not matching any packets for
mgcp ports on switch

 

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 ke...@kevinspicer.co.uk
wrote:

This is on 3750 switch?  Did you enable qos globally?  (mls qos)

On 24 Oct 2012 17:03, Krishna vinayak_...@yahoo.com wrote:

i created an acl that calls mgcp ports i.e. udp 2427  2428 with extended
acl permit tcp any any eq 2428, permit tcp any eq 2428 any , permit udp any
any eq 2427, permit udp any eq 2427 any. 

 

i called the acl in the class map, where the class map is referenced in the
policy map with appropriate bandwidth and qos configuration. i applied acl
on the trunk port that connects to router. 

 

when i issued show access-lists, i am not seeing any matches on the acl and
so i was wondering how could i verify that whether i am doing it right way
or not.. any help is much appreciated.

 

 

thank you

krishna.


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Re: [OSL | CCIE_Voice] Unable to call AVT from PSTN 911

2012-10-30 Thread Cory Gray
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 different
number for outcalling.  You have 4010 and 4011 for mwi on and off.
Turn on your debug and leave a message and I am pretty sure CUE is not
calling 4011 and you will have to replace your ephone-dn with something
different to match.  I am not sure if the first 4 digits are configurable in
CUE or not but it did not honor my first 4 digits... it picked its own.
Also get rid of your MWI dial-peers.  They are not needed.  Just the ephone
dns.

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ramcharan Arya
Sent: Thursday, October 25, 2012 5:18 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Unable to call AVT from PSTN 911

Hello,

I am working on a task trying to call AVT extension  4005 from PSTN 911. I
am able to call voice mail and auto-attendent but when I call AVT number I
get re-order tone.

Has anyone try AVT setup on CUE. Please let me know.

I got another problem with MWI lamp is not working when use left voice mail
to extension.


telephony-service
no auto-reg-ephone
  max-ephones 52
 max-dn 192
 ip source-address 10.10.128.1 port 2000  voicemail 4004  max-conferences 8
gain -6  moh music-on-hold.au  multicast moh 239.23.4.10 port 2000  web
admin system name Admin password admin  dn-webedit  time-webedit
transfer-system full-consult !
ephone-dn  98
 number 4010
 mwi off
!
!
ephone-dn  99
 number 4011
 mwi on

!
dial-peer voice 98 voip
 description MWI OFF
 incoming called-number 4010
 codec g711ulaw
 no vad
!
dial-peer voice 99 voip
 description MWI ON
 incoming called-number 4011
 codec g711ulaw
 no vad
!
dial-peer voice 4004 voip
 destination-pattern 4004
 session protocol sipv2
 session target ipv4:10.10.128.2
 dtmf-relay sip-notify
 codec g711ulaw
 no vad
!
dial-peer voice 4005 voip
   ** Description AVT ***
 destination-pattern 4005
 session protocol sipv2
 session target ipv4:10.10.128.2
 dtmf-relay sip-notify
 codec g711ulaw
 no vad

Can you please  suggest is something is missing in my configuration.

Thanks  Regards,
Ramcharan Arya
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Re: [OSL | CCIE_Voice] Fews questions on RSVP config in lab

2012-10-30 Thread Cory Gray
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
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Fews questions on RSVP config in lab

 

Dear Vir,

As per my understanding:
1) right , you can remove passthru probably can test same
2) yes, it works same way: you need to specify ip rsvp bandwidth b/w# on 
the egress interface
3) between 2 or more sites, location RSVP mandatory is important to trigger RSVP

All, Please correct me if wrong anywhere.

Regards,

Vikky



On Mon, Oct 22, 2012 at 9:53 AM, virajith vir...@rediffmail.com wrote:

Hi Guys,

If we need to configure RSVP between 2 sites and There can be 4 concurrent 
calls. G711 CODEC to be used for
multi-directional audio.


this mean that we set things up as this : -


dspfarm profile 1 mtp
codec g711u
codec pass‐through
rsvp
maximum sessions software 4
associate application SCCP


Questions:
=

1) However the above would mean that when code pass-thru fails the call will be 
sent over
wan as g711ulaw . Am I correct?


2)Does RSVP over fastethernet and gigainterface   work in the same fashion as 
point-to-point 
multilink over serial cables?


3) Do we need to make the location setting on callmanger as mandatory or can  
we  just leave it as system default in this scenario?


-Vir

 
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Re: [OSL | CCIE_Voice] Probleme with conference Bridge BR1

2012-10-26 Thread Cory Gray
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 mcl.nico...@gmail.com wrote:

 Same result :(
 
 Le Friday, October 26, 2012 6:57:30 PM, Abel ... a écrit :
 Try it with voice-card 0 with dspfarm also. Then the sessions under
 the media resource.
 
 On Fri, Oct 26, 2012 at 12:54 PM, Nicolas MICHEL
 mcl.nico...@gmail.com mailto:mcl.nico...@gmail.com wrote:
 
Hi Guys,
 
Looks like I cannot add a Conference bridge on BR1-RTR (I'm using PL).
 
I have configured the voice-card 0 to share the DSP like this
 
voice-card 0
 no dspfarm
 dsp services dspfarm
 
But when it comes to CFB configuration, the max number of session
is 0-0 :D
It happened to me before on HQ-RTR but this was because i forgot
the voice-card configuration :()
 
 
BR1-RTR(config)#dspfarm profile 3 conf
BR1-RTR(config-dspfarm-profile)#max sess 1
 ^
% Invalid input detected at '^' marker.
 
BR1-RTR(config-dspfarm-profile)#max sess ?
  0-0  Number of sessions assigned to this profile
 
 
 
Here is the full configuration :
 
 
BR1-RTR#sh
Oct 26 20:53:05.391: %SYS-5-CONFIG_I: Configured from console by
console
BR1-RTR#sh run
Building configuration...
 
Current configuration : 6051 bytes
!
! Last configuration change at 20:53:05 UTC Fri Oct 26 2012
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
!
hostname BR1-RTR
!
boot-start-marker
warm-reboot
boot-end-marker
!
logging message-counter syslog
!
no aaa new-model
network-clock-participate wic 0
network-clock-select 1 T1 0/0/0
!
dot11 syslog
ip source-route
!
!
ip cef
!
!
ip domain name proctorlabs.com http://proctorlabs.com
no ipv6 cef
!
multilink bundle-name authenticated
!
!
isdn switch-type primary-ni
!
!
!
voice service voip
 allow-connections sip to h323
 allow-connections sip to sip
 sip
  registrar server
!
!
!
!
!
!
!
!
!
!
!
!
!
!
voice register global
 max-dn 1
 max-pool 1
!
voice register pool  1
 id network 10.10.201.0 mask 255.255.255.0
 application sip.app
 preference 2
 incoming called-number
 codec g711ulaw
!
!
voice translation-rule 1
 rule 1 /617863\(1...\)$/ /\1/
!
voice translation-rule 7
 rule 1 /^1...$/ /863\0/
!
voice translation-rule 10
 rule 1 /^1...$/ /617863\0/
!
voice translation-rule 1001
 rule 1 /^1001$/ /1002/
!
!
voice translation-profile 1001
 translate redirect-called 1001
!
voice translation-profile 10digit
 translate calling 10
!
voice translation-profile 7digit
 translate calling 7
!
voice translation-profile strip-dnis
 translate called 1
!
!
voice-card 0
 no dspfarm
 dsp services dspfarm
!
!
!
!
!
archive
 log config
  hidekeys
!
!
!
!
!
controller T1 0/0/0
 framing esf
 linecode b8zs
 pri-group timeslots 1-3,24 service mgcp
!
controller T1 0/0/1
 framing esf
 linecode b8zs
 channel-group 0 timeslots 1-24
!
!
!
!
!
interface Loopback0
 ip address 10.10.110.2 255.255.255.255
 ip ospf network point-to-point
!
interface FastEthernet0/0
 no ip address
 shutdown
 duplex auto
 speed auto
!
interface FastEthernet0/1
 no ip address
 shutdown
 duplex auto
 speed auto
!
interface Serial0/0/0:23
 no ip address
 encapsulation hdlc
 isdn switch-type primary-ni
 isdn incoming-voice voice
 isdn bind-l3 ccm-manager
 no cdp enable
!
interface Serial0/0/1:0
 no ip address
 encapsulation frame-relay IETF
 no fair-queue
 frame-relay lmi-type ansi
!
interface Serial0/0/1:0.1 point-to-point
 bandwidth 1536
 ip address 10.10.111.2 255.255.255.0
 ip ospf mtu-ignore
 snmp trap link-status
 frame-relay interface-dlci 101
!
interface FastEthernet1/0
 switchport trunk native vlan 130
 switchport mode trunk
 switchport voice vlan 240
!
interface FastEthernet1/1
!
interface FastEthernet1/2
!
interface FastEthernet1/3
!
interface FastEthernet1/4
!
interface FastEthernet1/5
!
interface FastEthernet1/6
!
interface FastEthernet1/7
!
interface FastEthernet1/8
 switchport trunk native vlan 130
 switchport mode trunk
 switchport voice vlan 240
!
interface FastEthernet1/9
!
  

Re: [OSL | CCIE_Voice] Fews questions on RSVP config in lab

2012-10-22 Thread Cory Gray
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 we need to configure RSVP between 2 sites and There can be 4 concurrent 
calls. G711 CODEC to be used for
multi-directional audio.


this mean that we set things up as this : -


dspfarm profile 1 mtp
codec g711u
codec pass‐through
rsvp
maximum sessions software 4
associate application SCCP


Questions:
=

1) However the above would mean that when code pass-thru fails the call will be 
sent over
wan as g711ulaw . Am I correct?



Codec pass-through does not fail.  It allows the two endpoints to negotiate a 
codec versus relying on the MTP.  I typically do not use it but ran across this 
“The pass-through keyword is supported for transcoding and MTP profiles only; 
the keyword is not supported for conferencing profiles. To support the Resource 
Reservation Protocol (RSVP) agent on a Skinny Client Control Protocol (SCCP) 
device, you must use the codecpass-through command. In the pass-through mode, 
the SCCP device processes the media stream by using a pure software MTP, 
regardless of the nature of the stream, which enables video and data streams to 
be processed in addition to audio streams.”  So you can test on your lab and 
decide whether or not to use it if you see it in your lab.

 



2)Does RSVP over fastethernet and gigainterface   work in the same fashion as 
point-to-point 
multilink over serial cables?
Command reference does not list interface types in its caveats or restrictions 
so test them both in your lab and let us know.  I practice doing it over serial 
interfaces.  I could not see the point of using it on Ethernet interfaces as 
the serial interface is how one site connects to the other.

3) Do we need to make the location setting on callmanger as mandatory or can  
we  just leave it as system default in this scenario?
System default unless changed in service parameters is no reservation.  
Mandatory uses RSVP for every call.  Optional will allow the call to proceed 
and change the dscp value to a marking that is configurable in service 
parameters.

-Vir

 
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Re: [OSL | CCIE_Voice] Issue using VTGO softphone as pstn phone

2012-10-22 Thread Cory Gray
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.

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of
otunola.aker...@gmail.com
Sent: Monday, October 22, 2012 7:06 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Issue using VTGO softphone as pstn phone

Hi people,
 The softphone connect and disconnect like every 10 seconds please what
do I do?

Ola
Sent from my BlackBerryR Smartphone, from Etisalat. Enjoy high speed
internet service with Etisalat easy net, available at all our experience
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Re: [OSL | CCIE_Voice] Issue using VTGO softphone as pstn phone

2012-10-22 Thread Cory Gray
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, October 22, 2012 10:38 AM
To: otunola.aker...@gmail.com; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Issue using VTGO softphone as pstn phone

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.

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of
otunola.aker...@gmail.com
Sent: Monday, October 22, 2012 7:06 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Issue using VTGO softphone as pstn phone

Hi people,
 The softphone connect and disconnect like every 10 seconds please what
do I do?

Ola
Sent from my BlackBerryR Smartphone, from Etisalat. Enjoy high speed
internet service with Etisalat easy net, available at all our experience
centres ___
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Re: [OSL | CCIE_Voice] Fews questions on RSVP config in lab

2012-10-22 Thread Cory Gray
You must set the location bandwidth to unlimited so that CUCM uses RSVP 
determines if there is enough bandwidth.  It sounds like you are using 
locations CAC as well.  You must do one or the other.

Sent from my iPhone

On Oct 22, 2012, at 11:34 PM, virajith  vir...@rediffmail.com wrote:

 
 Hi Cory,
 
 Thanks for  your inputs
 
 Whenever I use Mandatory in the location settings the call fails and I see a 
 message on the phone saying Not enough Bandwidth
 
 I have read that the callmanger does not directly participate in RSVP as it 
 is 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 corygray22...@hotmail.com
 Sent: Mon, 22 Oct 2012 18:49:20 
 To: 'virajith ' vir...@rediffmail.com, ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] Fews questions on RSVP config in lab
 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 we need to configure RSVP between 2 sites and There can be 4 concurrent 
 calls. G711 CODEC to be used for
 multi-directional audio.
 
 
 this mean that we set things up as this : -
 
 
 dspfarm profile 1 mtp
 codec g711u
 codec pass‐through
 rsvp
 maximum sessions software 4
 associate application SCCP
 
 
 Questions:
 =
 
 1) However the above would mean that when code pass-thru fails the call will 
 be sent over
 wan as g711ulaw . Am I correct?
 
 
 Codec pass-through does not fail.  It allows the two endpoints to negotiate a 
 codec versus relying on the MTP.  I typically do not use it but ran across 
 this “The pass-through keyword is supported for transcoding and MTP profiles 
 only; the keyword is not supported for conferencing profiles. To support the 
 Resource Reservation Protocol (RSVP) agent on a Skinny Client Control 
 Protocol (SCCP) device, you must use the codecpass-through command. In the 
 pass-through mode, the SCCP device processes the media stream by using a pure 
 software MTP, regardless of the nature of the stream, which enables video and 
 data streams to be processed in addition to audio streams.”  So you can test 
 on your lab and decide whether or not to use it if you see it in your lab.
 
  
 
 
 
 2)Does RSVP over fastethernet and gigainterface   work in the same fashion as 
 point-to-point 
 multilink over serial cables?
 Command reference does not list interface types in its caveats or 
 restrictions so test them both in your lab and let us know.  I practice doing 
 it over serial interfaces.  I could not see the point of using it on Ethernet 
 interfaces as the serial interface is how one site connects to the other.
 
 3) Do we need to make the location setting on callmanger as mandatory or can  
 we  just leave it as system default in this scenario?
 System default unless changed in service parameters is no reservation.  
 Mandatory uses RSVP for every call.  Optional will allow the call to proceed 
 and change the dscp value to a marking that is configurable in service 
 parameters.
 
 -Vir
 
 
 
 Catch India as it happens with the Rediff News App. To download it for FREE, 
 click here
 
 
 
 Catch India as it happens with the Rediff News App. To download it for FREE, 
 click here
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Re: [OSL | CCIE_Voice] LAN Qos questions

2012-10-21 Thread Cory Gray
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 you have 75% is the reserved-threshold not threshold 3.
Moving cos and dscp values to threshold 1 and changing threshold 1 while
keeping threshold 2, reserved-threshold, and maximum-threshold at 100 is
just the easiest way to do it in the lab.  If you use auto qos on an unused
port like I do it will change the default mappings so everything is not in
threshold 1 allowing you to move just values you want into threshold 1.

 

HTH

 

Cory

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Steffen
Bruening
Sent: Sunday, October 21, 2012 1:45 AM
To: Krishna
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] LAN Qos questions

 

Hi Krishna,

 

sounds good so far. But as I said when COS 5 is the only COS in Queue 1,
which Traffic/COS could fill T1 and T2 up to 100% when COS 5 is only mapped
to T3?

 

Regards

 

Steffen

2012/10/21 Krishna vinayak_...@yahoo.com

steffen,

 

your approach is not right way of doing it because when u look the threshold
values of the queues you have allocated max threshold is 100 and reserved
threshold is 100, guess what both threshold i.e. t1 and t2 takes up to 100%
value when desired and that being said after t1 and t2 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.

  _  

From: Steffen Bruening stbruen...@gmail.com
To: Pixar Perfect pixarperf...@live.com 
Cc: ccie_voice@onlinestudylist.com 
Sent: Saturday, October 20, 2012 6:38 PM
Subject: Re: [OSL | CCIE_Voice] LAN Qos questions

 

I have this seen this also, to be honest I think it shouldn't matter whether
it is in threshold 1 or 3 as long as no other COS is in same Threshold of
queue 1 of queset 2. When you leave in in threshold 3 I think you should be
fine with:

 

mls qos queue-set output 2 threshold 1  100 100 75 100.

 

Maybe I am completly wrong but thats they way I understood this.

 

Regards

 

Steffen

2012/10/20 Pixar Perfect pixarperf...@live.com

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 1  75 100 100 100   -- queset is
preconfigured on the port to 2

mls qos srr-queue output cos-map queue 1 threshold 3   5

 

..

My interpretation was to move the Cos 5 into Q1t1 but the command says
threshold 3 .. is this just a typo or am I missing something obvious. 

 

 

Thanks! 


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Re: [OSL | CCIE_Voice] RSVP brokes transcoding

2012-10-20 Thread Cory Gray
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 stbruen...@gmail.com wrote:

 Hi,
 
 I have 3 Sites, all of them configured on the CUCM, Site C has Voicemail with 
 local CUE. When I am dialing from Site B to C codec g279 will be used and I 
 can reach the voicemail, so I know When I am dialing from Site A to C through 
 an RSVP CAC Location I get a fast busy when reaching the VM Pilot of Site C. 
 When I take of the RSVP I can also reach VM Pilot of Site C with G729. 
 
 Maybe somebody can explain to me why these RSVP Calls to VM are failing.
 
 My workaround for this to create a new location for VM Ports and Route Points 
 which does not use RSVP. But this will brake the requirement of the question 
 to allow only 4 calls between HQ und SC over the WAN.
 
 Regards
 
 Steffen
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Re: [OSL | CCIE_Voice] LAN Qos questions

2012-10-20 Thread Cory Gray
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 pixarperf...@live.com wrote:

 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 1  75 100 100 100   -- queset is 
 preconfigured on the port to 2
 mls qos srr-queue output cos-map queue 1 threshold 3   5
 
 ..
 My interpretation was to move the Cos 5 into Q1t1 but the command says 
 threshold 3 .. is this just a typo or am I missing something obvious. 
 
 
 Thanks! 
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Re: [OSL | CCIE_Voice] cme ip add or loopback ip in CUE??

2012-10-20 Thread Cory Gray
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 rrcr...@yahoo.com wrote:

 The labs say to use an IP address if 10.10.115.2. So under the lo1 you need 
 to add ip unnumbered lo1(also make sure to add the OSPG statement if needed 
 and clear the ip OSPG process$
 
 Then add the IP address and default gateway.
 
 Then make sure to add the static route
 
 Hth
 Randall
 
 On Oct 20, 2012, at 2:25 PM, Krishna vinayak_...@yahoo.com wrote:
 
 
 when using loopback address for CUE setup, does it matter whether what ip 
 address we put it for cme in cue  .i.e. for example loopback 10.10.115.1 or 
 10.10.202.1(cme ip address)... it works in both the cases but just want to 
 make sure which one is the right way of doing it..
 
 thank you
 krishna.
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