Re: [OSL | CCIE_Voice] Lab exam date availability at SJC and RTP

[ccie2k12]

cuz they have increased one seat (making it something like three ) for American 
sites as well as Brussels.

and feb for brussels is already almost gone :)

hope they open in dubai as well :) 

 

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of OSL StudyList
Sent: Wednesday, October 02, 2013 4:25 PM
To: Martin Sloan
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Lab exam date availability at SJC and RTP

 

FYI

 

There are a BUNCH of open lab dates in RTP and SJC.  No idea why but, there are 
open labs all of Oct through February.  

:)  

 

On Fri, Sep 27, 2013 at 11:45 AM, Martin Sloan  wrote:

Hey Alex,


I hear ya.  I went through all locations and checked availability and Tokyo is 
the closest for me, which is about an 18 hour flight.  I've priced it all out.  
I'm on the fence a bit about traveling there but at this point I'm leaning 
toward not.  On top of the additional expense for travel and time away from 
family, I'm paid by the hour so it would be at least 4 days unpaid for me.  It 
starts to add up.  Like everyone else I've invested a lot of money into this 
and I'm starting to get a little gun shy on putting up another couple thousand 
dollars for something that's not guaranteed.  It could be money straight down 
the toilet.  At a certain point, enough is enough.

Good luck on your lab in RTP in Feb!

Marty

 

On Fri, Sep 27, 2013 at 12:17 PM, Alex Mendoza  wrote:

As Dave says, you can book at Tokyo or other location.

 

I'm from Mexico and can book at RTP in february just one week ago.

 

More pressure because will be my 2nd and last attempt.

 

If you are so close to get your CCIE, look for a seat at other location even if 
you must pay for travel expenses.

 

All my best for the last candidates.

 

best regards

Alex

On Sep 27, 2013, at 10:57 AM, Martin Sloan  wrote:

 

I'm really disappointed as well.  I just failed my second attempt on Wed and 
was worried about getting a 4th try in when I logged on to see no seats left 
for a 3rd!  I figured it would get tight but this is nuts.  I made a big 
improvement on my score from the first try and feel like the third time could 
have been the charm.  Oh well.

 

On Fri, Sep 27, 2013 at 11:35 AM, Dane Warner  
wrote:

There are no open dates in either San Jose or RTP anymore, period.

 

Looks like if we want to take the Voice exam, which I’m sure Cisco doesn’t want 
us to do anymore, then it’s either Tokyo or we’re SOL.

 

Very disappointing.

 

Dane Warner, CCVP

Sr. Network Engineer

Epoch Universal, Inc.

(909)226-0755  

  dwar...@epochuniversal.com  



 

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of OSL StudyList
Sent: Friday, September 27, 2013 3:19 AM
To: Josh Petro
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Lab exam date availability at SJC and RTP

 

Do you know what times the lab dates are released for those who have not paid?  
 I thought it was at midnight SJC time but, I am not sure.  

 

On Fri, Sep 27, 2013 at 5:17 AM, Josh Petro  wrote:

If you mean Voice availability, then you are correct in that RTP is filled. San 
Jose had a few open spots in Jan Feb last week. 
I don't believe Collaboration dates are open yet for scheduling.
Josh

On Sep 27, 2013 5:58 AM, "OSL StudyList"  wrote:

Is anyone having any luck scheduling exams at RTP or SJC?   When I try to find 
an available date, I am seeing NOTHING available.   


___
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 

 

 

 

___
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

Re: [OSL | CCIE_Voice] tfr to vm....

[ccie2k12]

transfer system full-consult

 

its enable in you srst and not in CUCM.

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Amit Sharma
Sent: Sunday, August 04, 2013 10:58 AM
To: CCIE Study
Subject: [OSL | CCIE_Voice] tfr to vm

 

guys, 

when i accept a call from pstn phone and then transfer to vm by press shift
and speed dail that is configure for VMit still showing to press
transfer again

but as i apply same process in SRST case...for 4001it works and call
moves goes away from 40001 and it shows on pstn phone to play vm...




what could be issue due to that it is not working with cucm but working with
SiteC srst case?



anyone help would be more appreciated!



-- 

Thanks & Regard's

Amit Sharma

___
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

Re: [OSL | CCIE_Voice] Lab 4A BR1 MGCP Issue

[ccie2k12]

try removing

 

ccm-manager config server 10.10.210.11  
ccm-manager config



and remove

 isdn bind-l3 ccm-manager

and put it back again.

 

also

BR1

 pri-group timeslots 1-3,24

PSTN

 pri-group timeslots 1-3,23-24

 

should be same.

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Alex Pishko
Sent: Saturday, July 27, 2013 11:39 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Lab 4A BR1 MGCP Issue

 

All,

 

Struggling with the MGCG PRI circuit in this lab.  I've been through it a
few times and never had issues, however for this go around I cannot seem to
get L2 up when I issue the show isdn status command.  Also, when I look at
the serial inteface is shows up/up, however it's up/up (spoofing).

 

The MGCP portion appears to be workign correctly as when I issue a show
ccm-manager command I see it registered to the primary and within CUCM I see
the PRI as registered.  Below is my relevant configuration from the BR1
Router as well as the PSTN router.  Also, on the PSTN I've tried to change
the pri-group to just 1-3,24 both had the same result.  Also, when I issue
debug isdn q921 command I do not see any response for the SAMBE packet.  Any
help on this would be much appreciated.  I'm starting to think I may have a
hardware problem, but need a second or multiple sets of eyes on this before
I head down that router.

 

Thank you all.

 

Alex

 

BR1 Router

 

network-clock-participate wic 1 
network-clock-select 1 T1 0/1/0

 

controller T1 0/1/0
 framing esf
 linecode b8zs
 cablelength long 0db
 pri-group timeslots 1-3,24 service mgcp
 description **T1 Voice Connection to PSTN 0/1/1**

 

interface Serial0/1/0:23
 no ip address
 encapsulation hdlc
 isdn switch-type primary-ni
 isdn incoming-voice voice
 isdn bind-l3 ccm-manager
 no cdp enable

 

voice-port 0/1/0:23
!
ccm-manager switchback immediate
ccm-manager redundant-host 10.10.210.10
ccm-manager mgcp
ccm-manager music-on-hold
ccm-manager config server 10.10.210.11  
ccm-manager config
!
mgcp
mgcp call-agent 10.10.210.11 2427 service-type mgcp version 0.1
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp package-capability sst-package
mgcp package-capability pre-package
no mgcp package-capability res-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp rtp payload-type g726r16 static
mgcp bind control source-interface Loopback0
mgcp bind media source-interface Loopback0

 

 

PSTN Router 

 

network-clock-participate wic 1 

 

controller T1 0/1/1
 framing esf
 clock source internal
 linecode b8zs
 cablelength long 0db
 pri-group timeslots 1-3,23-24
 description **T1 VOICE CONNECTION TO BR1-RTR 0/1/0**

 

interface Serial0/1/1:23
 no ip address
 encapsulation hdlc
 isdn switch-type primary-ni
 isdn protocol-emulate network
 isdn incoming-voice voice
 no cdp enable

 

voice-port 0/1/1:23
 translation-profile incoming block-call-into-BR1
 translation-profile outgoing display-proper-ani-into-BR1

 

 

___
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

Re: [OSL | CCIE_Voice] Conf Meetme did not disconnect.

[ccie2k12]

have u tried it?

"Drop Ad Hoc Conference" doesn't work for meet-me.

 

Regards,

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Karen Johnson
Sent: Monday, July 15, 2013 8:46 PM
To: michael.se...@compucom.com; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Conf Meetme did not disconnect.

 

hi Mike and Tom,

 

Thanks, I will, just prepare if they ask me to drop.

 

K

 

From: "michael.se...@compucom.com" 
To: ccie_voice@onlinestudylist.com 
Cc: karen.johnson...@yahoo.ca 
Sent: Monday, July 15, 2013 7:10:03 AM
Subject: Re: Conf Meetme did not disconnect.


Karen,

Tom is right on.  Go to CUCM service parameter "Drop Ad Hoc Conference" and
change from default setting "Never" to "When Conference Controller Leaves".

But does the question explicitly state that all callers should be dropped
from the conference in the event the leader leaves  the conference?  In some
cases they may want the bridge to stay open if the organizer leaves the
bridge.  This is a case where you have to read the question very carefully.
It may not matter if the bridge stays open.

Michael Sears, CCIE(V)#38404
"Designing and Implementing Cisco Unified Communications on Unified
Computing Systems"





___
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

Re: [OSL | CCIE_Voice] Conf Meetme did not disconnect

[ccie2k12]

don't think thats possible, at least not in cucm 7.x

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Karen Johnson
Sent: Monday, July 15, 2013 4:25 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Conf Meetme did not disconnect

 

folks,

 

when the Host disconnect from Meetme 2500, other phones who joined did not
disconnect.

is there any service parameter/config need to be done ,so it can also
disconnect?

 

K

___
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

Re: [OSL | CCIE_Voice] cups best practice

[ccie2k12]

and exactly how else do you suggest to configure CUPS without a SIP trunk
between CUCM and CUPS??

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Karen Johnson
Sent: Sunday, July 14, 2013 11:17 AM
To: ikizoo hello; William Bell
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] cups best practice

 

hi ikizo,

 

i heard there is variation that asked not to use SIP trunk for Cups, maybe
you missed that.

 

Can someone confirm ?

 

K

 

From: ikizoo hello 
To: William Bell ; Karen Johnson
 
Cc: "ccie_voice@onlinestudylist.com"  
Sent: Thursday, July 11, 2013 12:10:28 PM
Subject: RE: [OSL | CCIE_Voice] cups best practice

 

hi Bill,
after verification VM, IM , Presence, all working fine, but still got 0
score.
only one thing, in the screenshot the user id was h...@ccievoice.com [not
h...@ccievoice.com, in CUCM it is HQ2].
do you have any idea how make this happen?

ikizoo

From: b...@ucguerrilla.com
Date: Fri, 24 May 2013 02:48:17 -0400
To: karen.johnson...@yahoo.ca
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] cups best practice

Well, I can't tell you why you got 0 points. When I provision CUPS, I do the
following: 

 

*   CUCM

*   Check to ensure Application Server exists (it should, but check)
*   Add SIP Security Profile for CUPS, with appropriate options
*   Add SIP trunk for CUPS
*   License capabilities for end users
*   Associate end users to DNs (including all shared lines)
*   Associate devices to end users
*   Add end users to End Users and Standard CTI user groups
*   If doing IPPM

*   Add service URL
*   associate with phone(s)
*   Add IPPM user in CUCM

*   Add CUPC if required

*   CUPS

*   Update Cluster Topology so CUPS node uses IP address and not name
*   Update service parameters (UC Proxy, domain setting)
*   Check service parameters (look for anything that isn't default)
*   Check CUCM Publisher status (ensure all green)
*   Add CUCM gateway (technically, not required since we are using
Publish mode but I did it anyway)
*   Add ACLs for incoming (ALL,ALL)
*   Activate/Start services
*   IPPM - customize service account user ID and password to match CUCM
*   CTI - associate users
*   General settings: TFTP servers (make sure they match what you are
using for phones)
*   VM Profile

*   VM Server - CUC
*   Mailstore is NOT necessary with this version and CUC
*   VM Pilot number
*   Unity Connection: Make sure you mod the CoS to allow UPC to do its
thing

*   Log into CUPS as user and add contacts manually

 

 

When testing:

*   Ensure contact list is built correctly
*   Ensure you have presence status updates for onhook/offhook events
*   Ensure you receive presence status changes when client toggles
status
*   If soft phone, place and receive calls
*   If RPC, place and receive calls
*   If VM, then leave a message, ensure notification is received,
playback the message (though you won't have audio)
*   IM between clients, both ways
*   If using IPPM, log in, test, and remain logged in
*   If CUPC, log in, test, and remain logged in
*   If you have a requirement to send messages then do so and leave the
messages on-screen (all clients)
*   If lab guide gives you screen shots, pay attention to every little
detail of the screen shot and make sure your screen looks identical

 

I never had issues getting points on CUPS and I always followed the same
procedure. I have spoken to several people who have said that they
configured everything, everything worked, but they still got 0 points. I
trust that they are right, everything was configured and working. So, that
leave two things. One, if they give you a screen shot of what they want,
check everything and make sure you mimic the screenshot. Two, pay attention
to any clues as to how the clients should be left when you are done with the
exam. For instance, shutting down the clients when not told to do so is
probably a bad idea. If they say send a broadcast then I would do so and
leave it on the screen. Things like that.

 

HTH.

 

-Bill

 

--

William Bell, CCIE #38914

blog: http://ucguerrilla.com/

Follow me on twitter @ucguerrilla

 





 

On May 23, 2013, at 11:39 PM, Karen Johnson 
wrote:

 

tks Ccieing, i have included steps u mention below and also some service
parameter.

Also soft and desk phone was login and send message each other, but i still
got 0 mark.

 

any advice from people who got full mark in this section please ?

 

From: CCIEing 
To: Karen Johnson  
Cc: "ccie_voice@onlinestudylist.com"  
Sent: Thursday, May 23, 2013 5:15:54 PM
Subject: Re: [OSL | CCIE_Voice] cups best practice

 

I do not set to the exam yet.. but here are below the steps I use to
configure my CUPS: 

CUPC Configuration (Softphone)

Form CUPS Side : 

. Application --> 

Re: [OSL | CCIE_Voice] CUPC client to play voicemail

[ccie2k12]

check server health in CUPC, to make voice mail is showing green

also have you entered the voice mail user id and password in preferences

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Drake J
Sent: Monday, July 08, 2013 9:46 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] CUPC client to play voicemail

 

hi Guys,

How do we get our CUPC client to play voicemail and display it on the client
as a red Icon?

I tried many times but not see this icon.

-Drake

___
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

Re: [OSL | CCIE_Voice] Changing CUCM IP address on Cisco Presence

[ccie2k12]

on CUPS goto 
system > cucm publisher
and change the publisher ip address
for sure it will reset all the services,  but will change the ip address.
hope thats what you were looking for.

Regards,

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Daniel Gomez
Sent: Sunday, July 07, 2013 9:38 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Changing CUCM IP address on Cisco Presence

Hello,

Does anybody know if there's any guide showing the procedure to change CUCM IP 
address on a presence server.

CUCM IP address has to change and therefore we need to change it on presence 
server.

Regards,

Daniel Gómez Quijada
danie...@gmail.com
Sent from my iPhone
___
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

Re: [OSL | CCIE_Voice] need help

[ccie2k12]

ccm-manager config server 10.10.210.11  

ccm-manager config

 

I think the above config on hq-rtr will prevent it from 

using the below 

 

pri-group timeslots 1-12,24 service mgcp

 

hence not using only first 12 channels,

try to remove the ccm-manage config and config server.

 

regards,

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Amit Sharma
Sent: Wednesday, June 12, 2013 8:55 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] need help

 

when i try to call from hq phone to 911 or locla number...showing busy...!

please see the attached log and running config...!

 

 

i also try to call with hq mgcp fail then it should go with br1 h.323
gateway...

that also not moving...

no logs coming on any router...!

 

can anyone help me?




 

-- 

Thanks & Regard's

Amit Sharma

___
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

[OSL | CCIE_Voice] CUE QOS

[ccie2k12]

while doing QOS marking for cue 

shall we use specefic IPs from PUB & SUB 

or just putting the service policy on interface connected to vmware 

can have any any eq 2748

 

like below?

 

 

ip access-list extended cue
permit tcp any any eq 2748
permit tcp any eq 2748 any 

or 

ip access-list extended cue
permit tcp CUCMIP cueIP eq 2748
permit tcp CUCMIP eq 2748 cueIP

ofcourse CUCMIP and CUEIP would be real IPs of CUE and CUCM.

___
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

Re: [OSL | CCIE_Voice] CUE AFTER SRST

[ccie2k12]

sorry guys wrong jtapi trigger number :(

its working after correcting the issue.

 

From: ccie2k12 [mailto:ccie2...@gmail.com] 
Sent: Friday, June 07, 2013 10:22 PM
To: 'Martin Sloan'; 'Ivan Darío Sánchez Calderón'
Cc: 'ccie_voice@onlinestudylist.com'
Subject: RE: [OSL | CCIE_Voice] CUE AFTER SRST

 

hi, i have problem with my cue to cucm 

it shows jtapi registered with cucm 

the ctiports are registered to the siteC

but even from sitec i'm getting a fast busy

any ideas what might i be missing?

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Martin Sloan
Sent: Friday, June 07, 2013 9:31 PM
To: Ivan Darío Sánchez Calderón
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CUE AFTER SRST

 

Hi Ivan,

 

Check your CUCM configs in the CUE web admin page.  Go to 'Configure->Call
Manager' and make sure you have all of the required information populated.
The CUE setup wizard will let you walk through without specifying a JTAPI
user/pass but if you look into the menu I mentioned, it's listed as required
information.  Verify your CUCM app user information there and reload the CUE
module.  I had the same problem, with no JTAPI info filled out.  The RP's
should then register.

 

HTH

Marty

 

On Thu, Jun 6, 2013 at 6:49 PM, Ivan Darío Sánchez Calderón
 wrote:

Hi everyone, 

When I have cue integrated with CUCM and I make  some voicemail test when
the gateway is in srst everything works fine, but when I come back from srst
the CTI RP and the CTI Ports don't register. I restart the cue, the CTI
service on CUCM but still shows unregistered, someone knows how to fix this
issue.? 


___
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

Re: [OSL | CCIE_Voice] CUE AFTER SRST

[ccie2k12]

hi, i have problem with my cue to cucm 

it shows jtapi registered with cucm 

the ctiports are registered to the siteC

but even from sitec i'm getting a fast busy

any ideas what might i be missing?

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Martin Sloan
Sent: Friday, June 07, 2013 9:31 PM
To: Ivan Darío Sánchez Calderón
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CUE AFTER SRST

 

Hi Ivan,

 

Check your CUCM configs in the CUE web admin page.  Go to 'Configure->Call
Manager' and make sure you have all of the required information populated.
The CUE setup wizard will let you walk through without specifying a JTAPI
user/pass but if you look into the menu I mentioned, it's listed as required
information.  Verify your CUCM app user information there and reload the CUE
module.  I had the same problem, with no JTAPI info filled out.  The RP's
should then register.

 

HTH

Marty

 

On Thu, Jun 6, 2013 at 6:49 PM, Ivan Darío Sánchez Calderón
 wrote:

Hi everyone, 

When I have cue integrated with CUCM and I make  some voicemail test when
the gateway is in srst everything works fine, but when I come back from srst
the CTI RP and the CTI Ports don't register. I restart the cue, the CTI
service on CUCM but still shows unregistered, someone knows how to fix this
issue.? 




___
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

[OSL | CCIE_Voice] Sip Troubleshooting

[ccie2k12]

I know the below message is clearly telling us that calling side is doing
SIP Early offer and using TCP as the transport protocol

but from the below message where do you think we should mark  for TCP
and Early Offer

 

TCP 

 

1) The first line "//SIP/SIPTcp...Outgoing SIP TCP"

2) ""

 

For Early Offer

 

1)Content-Type: application/sdp

2)v=0

 

CCM|//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 151.1.2.2 on
port 5060 index 6 

INVITE sip:911@151.1.2.2:5060 SIP/2.0

Date: Fri, 07 Jun 2013 04:09:56 GMT

Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
SUBSCRIBE, NOTIFY

From: "HQPH1"
;tag=06bb5c4b-f416-48f6-9225-822962b6cef1-29652495

Allow-Events: presence, kpml

P-Asserted-Identity: "HQPH1" 

Supported: timer,resource-priority,replaces

Min-SE:  1800

Remote-Party-ID: "HQPH1"
;party=calling;screen=yes;privacy=off

Content-Length: 210

User-Agent: Cisco-CUCM7.0

To: 

Contact: 

Expires: 180

Content-Type: application/sdp

Call-ID: 1b618780-1b115d14-5-a0a01b1@151.1.10.10

Via: SIP/2.0/TCP 151.1.10.10:5060;branch=z9hG4bK77edd7cbb

CSeq: 101 INVITE

Session-Expires:  1800

Max-Forwards: 70

 

v=0

o=CiscoSystemsCCM-SIP 2000 1 IN IP4 151.1.10.10

s=SIP Call

c=IN IP4 151.1.254.1

t=0 0

m=audio 16416 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=ptime:20

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

___
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

Re: [OSL | CCIE_Voice] Region Codec matrix really needed

[ccie2k12]

ok found something a little usefull

in BAT there is Populate/DePopulate Region Matrix

that does the job.

or may be its also same number of clicks :)

 

From: Josh Petro [mailto:josh.pe...@gmail.com] 
Sent: Wednesday, June 05, 2013 3:00 PM
To: ccie2k12
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Region Codec matrix really needed

 

You are correct. However, I have found its a short config process to setup
the region codec as it should be per the question simply for piece of mind
and to rule out any potential bugs in the software.
Josh

On Jun 5, 2013 5:52 AM, "ccie2k12"  wrote:

Hi,

If the question ask calls in same region should use g711

and calls in different regions use g729

 

Do we really have to define region wise codecs in region setup page or just
creating the region is enough ?

because if we have 

InterAregion Codec = g711/g722

and 

interEregion Codec = g279 

in service parameter then we would not have a need to set these again in
region setup page

right??


___
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

[OSL | CCIE_Voice] Region Codec matrix really needed

[ccie2k12]

Hi,

If the question ask calls in same region should use g711

and calls in different regions use g729

 

Do we really have to define region wise codecs in region setup page or just
creating the region is enough ?

because if we have 

InterAregion Codec = g711/g722

and 

interEregion Codec = g279 

in service parameter then we would not have a need to set these again in
region setup page

right??

___
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

Re: [OSL | CCIE_Voice] TEHO BEST PRACTICE

[ccie2k12]

weather national or local, should be decided on gateway level.

from HQ it should be national and from SB it should be local.

 

regards,

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Hesham
Abdelkereem
Sent: Thursday, May 30, 2013 1:30 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] TEHO BEST PRACTICE

 

Dear Experts,

 

Guys have a very tricky question for you.

Suppose you are asked to call from HQ (408) to 972 TEHO

1ST you will use remote gateway SB (972) and Second you will use SLRG

Ok my question here

If I will use the Remote gateway siteb

 

What should I do my pattern , ANI AND DNIS manipulation?

If I call 972 numbers from HQ via SB 972 Gateway

Should I make my pattern 91972.XXX

make my ANI 408XXX NATIONAL

DNIS 7 Digits Subscriber or 10 DIGIT NATIONAL

Please let me know the best practice 

Which one makes more sense

to make ANI 10 DIGIT 408XX NATIONAL

DNIS 7 DIGIT LOCAL

or ANI 10 DIGIT 408XXX NATIONAL

DNIS 10 DIGIT NATIONAL and prefix 1972

 

Thank you so much in advance.

 

Hesham

___
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

Re: [OSL | CCIE_Voice] xfer to VM in 2key stroke

[ccie2k12]

 

 

As I mentioned before in addition to the transfer soft key and VM button
speed dial.

we need to set "TransferOnHookEnabled" service parameter to True.

 

regards,

 

 

From: ikizoo hello [mailto:ikiz...@hotmail.com] 
Sent: Saturday, May 25, 2013 11:49 AM
To: ccie2k12; ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] xfer to VM in 2key stroke

 

still need 3 keys to complete the transfer, 

 

1. transfer[soft key]

2. transfer-VM[button]: speed dial

3. transfer[soft key]

 

any tricks?

 

 

  _  

From: ccie2...@gmail.com
To: ccie_voice@onlinestudylist.com
Date: Fri, 24 May 2013 12:52:37 +0500
Subject: Re: [OSL | CCIE_Voice] xfer to VM in 2key stroke

if it's just one phone button then,

all you need to do is to create a speed dial to vm

and set service parameter transfer on-hook to true

as transfer softkey is already there in connected state.

 

regards, 

 

From: ikizoo hello [mailto:ikiz...@hotmail.com] 
Sent: Friday, May 24, 2013 12:00 PM
To: ali raza; ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] xfer to VM in 2key stroke

 

in the question, shows the phone screen clearly that line button and
xsfer-VM[button]: speed dial only. no Transfer button.

thanks

-ikizoo

  _  

From: ccie2...@gmail.com
To: ccie_voice@onlinestudylist.com
Date: Fri, 24 May 2013 11:26:54 +0500
Subject: Re: [OSL | CCIE_Voice] xfer to VM in 2key stroke

I believe there are two ways of doing this

 

 

First way (must be the correct way)

 

Assign "transfer" key to a phone button

Assign speed dial to VM-Box  on another button

Set transfer on-hook service parameter to true.

 

call comes in pickup the call

hit transfer

hit VM speed dial

hang-up

 

Second Way (I would call it stupid way)

 

create a soft-key template for "Ring In" state

add iDivert soft-key 

add two or three (or as many are needed) "Undefined" between iDivert key and
other keys( i.e. Answer, DND keys),

so that the iDivert key would come after hitting "more" key.

 

now when the call rings

hit "more" (key stroke 1)

hit "iDivert" ( key stroke 2)

stupid I believe but its two key strokes, I call it stupid cuz actually that
was one key solution, which we made two key solution.

 

Regards,

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Avrille Alvaris
Sent: Friday, May 24, 2013 10:44 AM
To: ikizoo hello; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] xfer to VM in 2key stroke

 

No you are wrong!

 

Yes transfer on hook is right

 

thanks

 

  _  

From: ikizoo hello 
To: "ccie_voice@onlinestudylist.com"  
Sent: Friday, May 24, 2013 8:01 AM
Subject: [OSL | CCIE_Voice] xfer to VM in 2key stroke

 

question says 2 key stroke but the solution using vm pilot mask need 3 key
stroke to complete transfer caller to VM

1. transfer[soft key]

2. transfer-VM[button]: speed dial

3. transfer[soft key]

 

transfer on-hook is only solution?

 

- ikizoo


___
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

___
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

Re: [OSL | CCIE_Voice] cups best practice

[ccie2k12]

there is a service parameter in cucm

"SipPublishTrunk" we need to select the Sip trunk we use for cucm to cups
connection.

 

regards,  

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Piyush Jain
Sent: Friday, May 24, 2013 10:07 PM
To: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] cups best practice

 

Hello ccie2k12,

I don't understand what do you mean by SipPublishTrunk in CUCM service
parameter?
Could you please clarify?

Thanks,
Piyush

 

  _  



Message: 1
Date: Fri, 24 May 2013 19:31:59 +0500
From: "ccie2k12" 
To: "'William Bell'" 
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] cups best practice
Message-ID: <018a01ce588b$74decce0$5e9c66a0$@com>
Content-Type: text/plain; charset="us-ascii"

one little thing I think is missing from that otherwise perfect list.

CUCM service parameter for SipPublishTrunk

or is it not necessary to publish the trunk?



From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of William Bell
Sent: Friday, May 24, 2013 11:48 AM
To: Karen Johnson
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] cups best practice



Well, I can't tell you why you got 0 points. When I provision CUPS, I do the
following:



*CUCM

*Check to ensure Application Server exists (it should, but check)
*Add SIP Security Profile for CUPS, with appropriate options
*Add SIP trunk for CUPS
*License capabilities for end users
*Associate end users to DNs (including all shared lines)
*Associate devices to end users
*Add end users to End Users and Standard CTI user groups
*If doing IPPM

*Add service URL
*associate with phone(s)
*Add IPPM user in CUCM

*Add CUPC if required

*CUPS

*Update Cluster Topology so CUPS node uses IP address and not name
*Update service parameters (UC Proxy, domain setting)
*Check service parameters (look for anything that isn't default)
*Check CUCM Publisher status (ensure all green)
*Add CUCM gateway (technically, not required since we are using
Publish mode but I did it anyway)
*Add ACLs for incoming (ALL,ALL)
*Activate/Start services
*IPPM - customize service account user ID and password to match CUCM
*CTI - associate users
*General settings: TFTP servers (make sure they match what you are
using for phones)
*VM Profile

*VM Server - CUC
*Mailstore is NOT necessary with this version and CUC
*VM Pilot number
*Unity Connection: Make sure you mod the CoS to allow UPC to do its
thing

*Log into CUPS as user and add contacts manually





When testing:

*Ensure contact list is built correctly
*Ensure you have presence status updates for onhook/offhook events
*Ensure you receive presence status changes when client toggles
status
*If soft phone, place and receive calls
*If RPC, place and receive calls
*If VM, then leave a message, ensure notification is received,
playback the message (though you won't have audio)
*IM between clients, both ways
*If using IPPM, log in, test, and remain logged in
*If CUPC, log in, test, and remain logged in
*If you have a requirement to send messages then do so and leave the
messages on-screen (all clients)
*If lab guide gives you screen shots, pay attention to every little
detail of the screen shot and make sure your screen looks identical



I never had issues getting points on CUPS and I always followed the same
procedure. I have spoken to several people who have said that they
configured everything, everything worked, but they still got 0 points. I
trust that they are right, everything was configured and working. So, that
leave two things. One, if they give you a screen shot of what they want,
check everything and make sure you mimic the screenshot. Two, pay attention
to any clues as to how the clients should be left when you are done with the
exam. For instance, shutting down the clients when not told to do so is
probably a bad idea. If they say send a broadcast then I would do so and
leave it on the screen. Things like that.



HTH.



-Bill



--

William Bell, CCIE #38914

blog: http://ucguerrilla.com <http://ucguerrilla.com/> 

Follow me on twitter @ucguerrilla









On May 23, 2013, at 11:39 PM, Karen Johnson 
wrote:





tks Ccieing, i have included steps u mention below and also some service
parameter.

Also soft and desk phone was login and send message each other, but i still
got 0 mark.



any advice from people who got full mark in this section please ?



From: CCIEing 
To: Karen Johnson  
Cc: "ccie_voice@onlinestudylist.com"  
Sent: Thursday, May 23, 2013 5:15:54 PM
Subject: Re: [OSL | CCIE_Voice] cups best practice



I do not set to the exam yet.. but here are below the steps I use to
configure my 

Re: [OSL | CCIE_Voice] cups best practice

[ccie2k12]

one little thing I think is missing from that otherwise perfect list.

CUCM service parameter for SipPublishTrunk

or is it not necessary to publish the trunk?

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of William Bell
Sent: Friday, May 24, 2013 11:48 AM
To: Karen Johnson
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] cups best practice

 

Well, I can't tell you why you got 0 points. When I provision CUPS, I do the
following:

 

*   CUCM

*   Check to ensure Application Server exists (it should, but check)
*   Add SIP Security Profile for CUPS, with appropriate options
*   Add SIP trunk for CUPS
*   License capabilities for end users
*   Associate end users to DNs (including all shared lines)
*   Associate devices to end users
*   Add end users to End Users and Standard CTI user groups
*   If doing IPPM

*   Add service URL
*   associate with phone(s)
*   Add IPPM user in CUCM

*   Add CUPC if required

*   CUPS

*   Update Cluster Topology so CUPS node uses IP address and not name
*   Update service parameters (UC Proxy, domain setting)
*   Check service parameters (look for anything that isn't default)
*   Check CUCM Publisher status (ensure all green)
*   Add CUCM gateway (technically, not required since we are using
Publish mode but I did it anyway)
*   Add ACLs for incoming (ALL,ALL)
*   Activate/Start services
*   IPPM - customize service account user ID and password to match CUCM
*   CTI - associate users
*   General settings: TFTP servers (make sure they match what you are
using for phones)
*   VM Profile

*   VM Server - CUC
*   Mailstore is NOT necessary with this version and CUC
*   VM Pilot number
*   Unity Connection: Make sure you mod the CoS to allow UPC to do its
thing

*   Log into CUPS as user and add contacts manually

 

 

When testing:

*   Ensure contact list is built correctly
*   Ensure you have presence status updates for onhook/offhook events
*   Ensure you receive presence status changes when client toggles
status
*   If soft phone, place and receive calls
*   If RPC, place and receive calls
*   If VM, then leave a message, ensure notification is received,
playback the message (though you won't have audio)
*   IM between clients, both ways
*   If using IPPM, log in, test, and remain logged in
*   If CUPC, log in, test, and remain logged in
*   If you have a requirement to send messages then do so and leave the
messages on-screen (all clients)
*   If lab guide gives you screen shots, pay attention to every little
detail of the screen shot and make sure your screen looks identical

 

I never had issues getting points on CUPS and I always followed the same
procedure. I have spoken to several people who have said that they
configured everything, everything worked, but they still got 0 points. I
trust that they are right, everything was configured and working. So, that
leave two things. One, if they give you a screen shot of what they want,
check everything and make sure you mimic the screenshot. Two, pay attention
to any clues as to how the clients should be left when you are done with the
exam. For instance, shutting down the clients when not told to do so is
probably a bad idea. If they say send a broadcast then I would do so and
leave it on the screen. Things like that.

 

HTH.

 

-Bill

 

--

William Bell, CCIE #38914

blog: http://ucguerrilla.com

Follow me on twitter @ucguerrilla

 





 

On May 23, 2013, at 11:39 PM, Karen Johnson 
wrote:





tks Ccieing, i have included steps u mention below and also some service
parameter.

Also soft and desk phone was login and send message each other, but i still
got 0 mark.

 

any advice from people who got full mark in this section please ?

 

From: CCIEing 
To: Karen Johnson  
Cc: "ccie_voice@onlinestudylist.com"  
Sent: Thursday, May 23, 2013 5:15:54 PM
Subject: Re: [OSL | CCIE_Voice] cups best practice

 

I do not set to the exam yet.. but here are below the steps I use to
configure my CUPS: 

CUPC Configuration (Softphone)

Form CUPS Side : 

. Application --> Settings 

. Application --> CUCP --> add Voice Mail Server

. Application --> CUCP --> add Mail Store (default 143)

.  Application --> CUPC --> add Voicemail Profile , then add users
to this profile 

. Application --> CUCP --> add CTI Gateway (should be created
automatic when start services )

. Application --> CUCP --> add CTI Gateway Profile  

Form CUCM Side : 

. Add CUPC soft phone: Device --> add Cisco Unified Personal
Communicator with the name UPCUSERNAME

Desk phone Configuration

. CUCM --> Associate device to end user (End user configuration
page--> add the device in the Device association list )

. CUCM --> Specify Primary extension for

Re: [OSL | CCIE_Voice] xfer to VM in 2key stroke

[ccie2k12]

if it's just one phone button then,

all you need to do is to create a speed dial to vm

and set service parameter transfer on-hook to true

as transfer softkey is already there in connected state.

 

regards, 

 

From: ikizoo hello [mailto:ikiz...@hotmail.com] 
Sent: Friday, May 24, 2013 12:00 PM
To: ali raza; ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] xfer to VM in 2key stroke

 

in the question, shows the phone screen clearly that line button and
xsfer-VM[button]: speed dial only. no Transfer button.

thanks

-ikizoo

  _  

From: ccie2...@gmail.com
To: ccie_voice@onlinestudylist.com
Date: Fri, 24 May 2013 11:26:54 +0500
Subject: Re: [OSL | CCIE_Voice] xfer to VM in 2key stroke

I believe there are two ways of doing this

 

 

First way (must be the correct way)

 

Assign "transfer" key to a phone button

Assign speed dial to VM-Box  on another button

Set transfer on-hook service parameter to true.

 

call comes in pickup the call

hit transfer

hit VM speed dial

hang-up

 

Second Way (I would call it stupid way)

 

create a soft-key template for "Ring In" state

add iDivert soft-key 

add two or three (or as many are needed) "Undefined" between iDivert key and
other keys( i.e. Answer, DND keys),

so that the iDivert key would come after hitting "more" key.

 

now when the call rings

hit "more" (key stroke 1)

hit "iDivert" ( key stroke 2)

stupid I believe but its two key strokes, I call it stupid cuz actually that
was one key solution, which we made two key solution.

 

Regards,

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Avrille Alvaris
Sent: Friday, May 24, 2013 10:44 AM
To: ikizoo hello; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] xfer to VM in 2key stroke

 

No you are wrong!

 

Yes transfer on hook is right

 

thanks

 

  _  

From: ikizoo hello 
To: "ccie_voice@onlinestudylist.com"  
Sent: Friday, May 24, 2013 8:01 AM
Subject: [OSL | CCIE_Voice] xfer to VM in 2key stroke

 

question says 2 key stroke but the solution using vm pilot mask need 3 key
stroke to complete transfer caller to VM

1. transfer[soft key]

2. transfer-VM[button]: speed dial

3. transfer[soft key]

 

transfer on-hook is only solution?

 

- ikizoo


___
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