Re: [OSL | CCIE_Voice] Calling Number during Unity Connection Call Transfer to external number

2011-09-14 Thread Cristobal Priego
I there is a service parameter that you have to chante to preserve the  
caller id. I don't remember the name of the option though

Hth

Enviado desde mi iPhone

El Sep 14, 2011, a las 14:21, Steffen Bruening steffen.bruen...@intact-is.com 
 escribió:



Hi all,



this question is not related to the CCIE labs, but hopefully you can  
help me to find an answer.




When a caller reach a user voicemail and choose the caller input  
option to be forwarded to an alternate number (in this case the  
mobile number) I can see in the isdn debug that extension of the  
used voicemail-port for this call transfer is used as calling number  
instead of the originating number. Any ideas how I could achieve  
that the originating phone number will be presented? The german  
telco do not check or restrict the calling number presentation (CLIP  
no screening).




Regards

Steffen




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Re: [OSL | CCIE_Voice] 4 g729 calls

2011-09-10 Thread Cristobal Priego

Is it compressed audio maybe?

Enviado desde mi iPhone

El Sep 10, 2011, a las 4:49, Ray jonha...@yahoo.com escribió:


in lab5
 task 7.2... it says 16k for singnaling and 4 g729 calls. the  
question is why is 47  used  for bandwidth instead of 96 (24x4=96).

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Re: [OSL | CCIE_Voice] CUPS integration and CUPC time consuming

2011-07-24 Thread Cristobal Priego
i spend 8 hours. 1 proctor-lab session just configuring/troubleshooting CUPS
in my lab session before i got all the glitches and solutions and found a
way to configure it real fast
once i had it configured i'd revert the pod and start again, and again until
you're up to speed and get your cupc clients and ippm working all the time,
the very first time

2011/7/24 Michael Luo hout...@gmail.com

 You should be able to do it within 10 to 15 minutes.

 Michael

 On Sat, Jul 23, 2011 at 9:01 PM, Adil Shaikh adil.sha...@gmail.comwrote:

 hi guys,

 i am finding it takes me 25 to 30 minutes to configure CUCM for CUPS
 (including 2 x CUPC and 1 x IPPM) and CUPS.

 just want to know, is it the time frame you guys are taking?

 thanks
 -adil

 --
   .. . .
 _7___|___|_|_|adil.sha...@gmail.com

 . .



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Re: [OSL | CCIE_Voice] Passed

2011-07-22 Thread Cristobal Priego
congratulations man !!

2011/7/22 Hough, Earl earl.ho...@pcmallservices.com

 Congratulations, and enjoy a well-deserved break!

 ** **

 Earl Hough

 CCIE #16508 (RS/Security/Voice)

 ** **

 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *adam compton
 *Sent:* Friday, July 22, 2011 9:36 AM
 *To:* ccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] Passed

 ** **

 Just found out the good news.  I passed  CCIE Voice #29657.  This must
 be a good week to take the lab.  Congrats to the others who passed this week
 as well

  

 Adam Compton

 _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _

 The information contained in this transmission is confidential. It is
 intended solely for the use of the individual(s) or organization(s) to
 whom it is addressed. Any disclosure, copying or further distribution is
 not permitted unless such privilege is explicitly granted in writing by
 PC Mall, Inc. Furthermore, PC Mall, Inc. is not responsible for
 the proper and complete transmission of the substance of this
 communication, nor for any delay in its receipt.



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Re: [OSL | CCIE_Voice] QoS bandwidth calculations

2011-07-21 Thread Cristobal Priego
i have always used the values from the QoS SRND and i haven't had any
problems


take a look at the following analysis

http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg06644.html


2011/7/21 CCIEVoiceKP ccievoic...@gmail.com

 I'm good with how to calculate BW from the QoS SRND, however it seems to
 get a little squishy when it comes to values for cRTP and rounding the
 output.  For example, say I have a requirement to allow 5 g729 calls and I'm
 using MLP with header compression.  We know that by default CRTp should be 2
 without checksum and 4 with checksum but in going through the Vol1 and Vol 2
 labs and VoDs I've seen both values used with no metion of whether or not
 checksum was a requirement.  So if we calcualte first with cRTP of 2 bytes
 we get 13 + 2 + 20 = 35 * 8 * 50 = 14000 = *14kbps*   with cRTP of 4
 we get 13 + 4 + 20 = 14800 = *14.8kbps.*  Now if we multiply this out by
 the number of calls we get 14kbps * 5 = *70kbps* and 14.8 * 5 = *74kbps 
 *respectively
  or do we?  Because I've seen the rounding take place earlier on so we
 would in essence have 15kbps * 5 = *75kbps.*

 Keep in mind this is if your following the SRND.  Vik and Amy recommend
 using 9bytes as the L2 value for MLP as opposed to 13bytes... I realize this
 is a more accurate number but do we have any official guidance as to how the
 proctors are grading this?  Are they using values strictly from the SRND?
 How big is the window or margin of error for these calculations?

 I know this sounds like splitting hairs but I've taken the lab before and
 walked away with 0 points in QoS not knowing why.  Thanks in advance for any
 insight?

 KP

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Re: [OSL | CCIE_Voice] CCIE #29619

2011-07-20 Thread Cristobal Priego
congratulations Emin
good job man !!

2011/7/20 Ashraf Ayyash ash.ayy...@gmail.com

 Congratulatuion Emin , all the best ,

 Ash

 On Wed, Jul 20, 2011 at 6:22 PM, Emin Guliyev eguli...@fidelus.com
 wrote:
  Guys,
 
 
 
  I passed the lab yesterday on second attempt. I
 appreciate
  all the help I have gotten from OSL.
 
 
 
  Regards,
 
 
 
  Emin
 
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[OSL | CCIE_Voice] passed CCIE 29606

2011-07-19 Thread Cristobal Priego
hello guys,

My 2 year journey just came to an end. I took my 3rd attempt yesterday in
San Jose and i just got my score 5 mins ago with good news
I'm CCIE # 29606
It was long hours of study time and lab time
I couldn't afford a home lab so all i did is rent equipment from Proctorlabs
i was able to get physical phones to test. which was pretty good

my strategy was a bit different from what people usually recommend

I started with Infrastructure of course
switch config all that needed to be done in the switch including QoS, so i
could forget about that device
then i moved to HQ, SiteB and SiteC NTP, vlans, and then QoS right away
after that i started working on CUE (because i know it takes long time)
then i moved to PRI configs
then i started working on configuring my phones, basic config
then i moved over to VM, then UCCX, then all the features requested
and at the end i did call routing

something that actually helped me out a lot to improve my speed was
i created my configs in notepad and i called those my templates, so during
the week i didn't have access to any rack rental sessions so i was
practicing in notepad

i did the workbooks 1 and 2 only once then i was using the rack rental
sessions to practice pretty heavily on a tech basis everthing from inside
out and every once in a while start adding more technologies

that was my experience i didn't know how to approach this beast at the
beginning I was reading all the documentation all the pdf's and books and i
was getting bored
so i dediced to read all the documentation, pdf's and books on a technology
basis i printed a calendar and with the blue print on the other hand i
marked which days i was going to spend studying X item on the blue print so
by the time it was the weekend i could practice that in my lab sessions
during the weekend and get as much as i could from the rack rental sessions

thank you very much for all your support and also for my study partners from
all over the world
I'll try to help you to get your number as well, you guys helped me, i'm
going to try to do the same

have a great day !
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[OSL | CCIE_Voice] Proctors in San Jose

2011-07-19 Thread Cristobal Priego
i forgot to mention

Tom the proctor in San Jose is retired, he took early retirement
so there are only 2 proctors in San Jose now, Tong and Ben

this week the proctor in San Jose will be Scott, a guy from Australia very
friendly, very nice guy
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Re: [OSL | CCIE_Voice] passed CCIE 29606

2011-07-19 Thread Cristobal Priego

Media Resource Management, Cisco Unified Communications Manager
System Guide, Release 7.0(1)

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmsys/a05media.html

Annunciator,

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmsys/a05ann.html

Conference Bridge,

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmsys/a05confb.html

Transcoders,

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmsys/a05trans.html

MTP,

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmsys/a05mtp.html

DSP resources for Transcoding, Conferencig, MTP

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmsys/a05dsp.html

DSP, MTP types, configuration of HW Conference and Transcoder resources,
CVOICE
Chapter 2, pages 91-120

DSP, MTP types, configuration of HW Conference and Transcoder resources,
Cisco Voice Gateways and Gatekeepers
Chapter 14, pages 411-435

CIPT1, MEDIA RESOURCES
Chapter 15, pages 367 - 405

 8. CM Features: Extension Mobility, IPMA, Attendant Console, Call
Park, Pickup

EXTENSION MOBILITY, Cisco Unified Communications Manager Features and
Services Guide

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmfeat/fsem.pdf

ATTENDANT CONSOLE, Cisco Unified Communications Manager Features and
Services Guide

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmfeat/fsccmac.pdf

CALL PARK, Cisco Unified Communications Manager Features and Services
Guide

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmfeat/fscallpk.pdf

PICKUP, Cisco Unified Communications Manager Features and Services Guide

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmfeat/fscpickg.pdf

CIPT 1, USER FEATURES
CHAPTER 16, pages407 - 414

CIPT 2, IMPLEMENTING EXTENSION MOBILITY
CHAPTER 12, pages 301 - 320

IPMA, Proxy Line Support,  Cisco Unified Communications Manager Features
and Services Guide

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmfeat/fsipma.pdf

IPMA, Shared Line Support,  Cisco Unified Communications Manager
Features and Services Guide

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmfeat/fsipmasl.pdf

 9. Phone Settings

10. CTI, TAPI and JTAPI Interface

CTI, SRND
  CTI  8-13, 8-18, 13-19, 16-2, 16-44
CTI Manager 8-4, 8-13

CTI, Cisco Unified Communications Manager System Guide, Release 7.0(1)

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmsys/a08cti.html#wpmkr1026233

JTAPI, SRND
JTAPI  8-13, 16-2


2011/7/19 Hough, Earl earl.ho...@pcmallservices.com

 Congratulations!  That’s a great feeling of accomplishment, I know!

 ** **

 Earl Hough

 CCIE #16508 (RS/Security/Voice)

 ** **

 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Cristobal Priego
 *Sent:* Tuesday, July 19, 2011 11:49 AM

 *To:* ccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] passed CCIE 29606

 ** **

 hello guys,


 My 2 year journey just came to an end. I took my 3rd attempt yesterday in
 San Jose and i just got my score 5 mins ago with good news
 I'm CCIE # 29606
 It was long hours of study time and lab time
 I couldn't afford a home lab so all i did is rent equipment from
 Proctorlabs i was able to get physical phones to test. which was pretty good

 my strategy was a bit different from what people usually recommend

 I started with Infrastructure of course
 switch config all that needed to be done in the switch including QoS, so i
 could forget about that device
 then i moved to HQ, SiteB and SiteC NTP, vlans, and then QoS right away
 after that i started working on CUE (because i know it takes long time)
 then i moved to PRI configs
 then i started working on configuring my phones, basic config
 then i moved over to VM, then UCCX, then all the features requested
 and at the end i did call routing

 something that actually helped me out a lot to improve my speed was
 i created my configs in notepad and i called those my templates, so during
 the week i didn't have access to any rack rental sessions so i was
 practicing in notepad

 i did the workbooks 1 and 2 only once then i was using the rack rental
 sessions to practice pretty heavily on a tech basis everthing from inside
 out and every once in a while start adding more technologies

 that was my experience i didn't know how to approach this beast at the
 beginning I was reading all the documentation all the pdf's and books and i
 was getting bored
 so i dediced to read all the documentation, pdf's and books on a technology
 basis i printed a calendar and with the blue print on the other hand i
 marked which days i was going to spend studying X item on the blue print so
 by the time it was the weekend i could practice that in my lab

Re: [OSL | CCIE_Voice] CUCM username password

2011-07-16 Thread Cristobal Priego
admin
c1sc0123

2011/7/15 narayan sarma narayan.na...@hotmail.com

  Hi,

 Can anyone tell the username  password of the CUCM in ipexpert. I m having
 a seesion now..but i m stuck outside of the CUCM because of not having the
 username  pass..





 Thanks  Regards
 Narayan Dev Sarma

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Re: [OSL | CCIE_Voice] Logistic info for the lab

2011-07-13 Thread Cristobal Priego
to answer your questions

you can bring food, water, pens to the lab, however at your desk you can
only have the water and pens, you can't have the food
the lunch break depends on the place where you take the lab in San Jose is
between 30-40 mins break
you don't have to do any patching at all
the only paper provided is the lab questions, in a binder
you get 2 sheets of paper on your desk to make your notes, and you can ask
the proctor for more paper if needed

2011/7/13 Vega Wong vega2...@yahoo.com.au

 Hi experts

 Many experts had advised that time management is a very important ( if not
 the most important) aspect during the lab. This leads me to wonder some of
 the smaller things during the lab:
 1. What can we bring in to the lab? Water bottle?
 2. How long is the lunch break? Is it allocated at a specific time?
 3. Do we have to do the patching of cables?
 4. What do they provide apart from the Cisco PDF? Any writing papers?

 Obviously I havent had my first attempt yet, but I will have mine soon.
 Just thought the more I know the better,

 Hope those who had the experience can share

 Cheers

 Vega

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Re: [OSL | CCIE_Voice] Logistic info for the lab

2011-07-13 Thread Cristobal Priego
in San Jose
I asked the question about the food to Tom
and he told me that you can bring food and water, only water is allowed on
your desk if you want to eat something you have to go to the table that is
close to the door to eat it. you're allowed to bring your own food, in fact
i did it for my 2nd attempt in  San Jose, i had apples, oranges, fruitbars
and water, no problems

2011/7/13 cciev wannabe cciev.wann...@gmail.com

 Answers:
 1. Nothing. They provide you food and water/soda. I have seen one guy who
 brought his own lunch because of religious practice and they allow it
 2. I have been to RTP only where 30 minutes lunch in a small conf room
 attached to the lab
 3. No you don't have to do any hardware stuffs
 4. two A4 size papers and bunch of colored pencils and cisco support page
 is accessible (don't type url in, just hit home button)

 On Wed, Jul 13, 2011 at 10:36 AM, Vega Wong vega2...@yahoo.com.au wrote:

 Hi experts

 Many experts had advised that time management is a very important ( if not
 the most important) aspect during the lab. This leads me to wonder some of
 the smaller things during the lab:
 1. What can we bring in to the lab? Water bottle?
 2. How long is the lunch break? Is it allocated at a specific time?
 3. Do we have to do the patching of cables?
 4. What do they provide apart from the Cisco PDF? Any writing papers?

 Obviously I havent had my first attempt yet, but I will have mine soon.
 Just thought the more I know the better,

 Hope those who had the experience can share

 Cheers

 Vega

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Re: [OSL | CCIE_Voice] Second Attempt

2011-07-13 Thread Cristobal Priego
sometimes it's hard to keep the motivation after you've failed the test and
you see your score.  try to find a study partner that will make you commit
to get the cert, get updates frequently and stuff. i have taken the lab
twice in the last 2 months and i'm going for my third attempt next week.
it's great to have a motivation like a raise, i don't know if i'm going to
get a raise if i get my cert, but i know i'm going to gain credibility and
trust, and perhaps a job offer.  I live and work in northern California and
i wish there was a great market for ccie's or maybe there is but i haven't
been looked at those opportunities yet. there are great people knowledgeable
willing to pass this test, try to find them and keep it up
i've been studying for a bit over 1.5 years now for this test and i'd love
to pass it sooner than later.

2011/7/13 adam compton com...@gmail.com

 Believe me, I was pissed after my second attempt and failed as well.  I'm
 taking 3rd attempt next thursday.  Some things changed from when I took it
 in March and in June. so the best advice I could give is to try it over and
 over as quick as possible until you get it.  I've got a big raise on the
 line, so the incentive to pass is big.  I just started a new job that was a
 huge raise from my last job, and that was because I was studying CCIE
 Voice.  I probably wouldn't have gotten the interview without it.  I cannot
 speak for all countries and areas, but in the Southeast US,  this
 certification is HIGHLY in demand.  More in demand than CCIE RS by a long
 shot.

 I agree it seems like cisco is hell-bent on failing everyone with silly
 requirements, and being shady about telling you what to study, but the fact
 of the matter is, if you want the recognition that comes with CCIE Voice,
 you have to grin, bear it, and take it like a woman ;)

 Adam Compton

 On Wed, Jul 13, 2011 at 11:00 AM, Ccie Voice v.c...@yahoo.com wrote:

   Hi All,


 I did my first attempt, I did not pass but the report was very good and
 most of my marks 100%. The report encourages me to do my second attempt. I
 prepared for the second attempt and I did my second attempt.



 Second attempt was much better than the first one. I studied very well and
 tried to avoid my mistakes in my first attempt. Unfortunately, the report
 was very bad. Most of my marks 40%, 20%, and some 0%.



 But why???

 Some questions are exactly the same with first attempt? Why I am getting
 100 in my first attempt and 0 in my second attempt.



 Some of the requirements I was not able to solve it and thought I will get
 0 because I did not solve it but in the report 100%. Is it a magic?



 I am now very pessimistic I am thinking to leave my CCIE and I am
 sometimes I thinking to leave Cisco and work with other vendors.



 I am very sorry for this bad e-mail. Specially, for who is preparing right
 now. But I am really very pessimistic from my second attempt and I need some
 advises from your side


 Regards,



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Re: [OSL | CCIE_Voice] WAN QoS MLP LFI

2011-07-11 Thread Cristobal Priego
sure no problem

i configured MLP LFI manually and tested connectivity from BR2 and BR1 to
HQ, BR2 no problems, because there's no virtual-template
however from BR2 to HQ the connectivity was fine unless you reload BR2
router, once the router came back up, I lost connectivity to HQ and of
course all the servers
after that I tried reloading the HQ router, no difference same issue

so i applied the following workaround

I went to the sub-interface on BR2 router
removed the frame-relay interface-dlci 201 ppp virtual-template 1   command
re-added the command
re-added the class-map to the fram-relay interface
connectivity came back up

however if you reload the BR2 router again, i was having the same problem

so at the end what it did was
fix the connectivity problem, once I had connectivity back, what i did is
Reload HQ router

once HQ came back up, I could reload Br2 router multiple times and the
connectivity issue was resloved

hope this helps

2011/7/11 Ronmac ron...@solcon.nl

 Hi,

 Maybe a hint on what was wrong?

 Share the knowledge.


 Send from my iPad2

 Op 11 jul. 2011 om 00:32 heeft Cristobal Priego cristobalpri...@gmail.com
 het volgende geschreven:

 nevermind
 i figured it out

 thanks

 2011/7/10 Cristobal Priego  cristobalpri...@gmail.com
 cristobalpri...@gmail.com

 hello all

 I know we have address this issue several times

 i'm configuring MLP manually and this is how i have HQ configured

 int serial 0/0/1:0.1
 bandwidth 384

 int serial 0/0/1:0
 frame-relay traffic-shapping



 class-map VOICE
 match ip dscp ef
 class-map match-any SIG
 match ip dscp cs3
 match ip dscp af31


 policy-map VOIP
 class VOICE
 priority percent 33
 class SIG
 bandwidth percent 5
 class class-default
 fair-queue


 map-class frame-relay myconfig
 frame-relay cir 384000
 frame-relay mincir 384000
 frame-relay be 0
 frame-relay bc 3840


 int virtual-template 1
 ip address 10.10.111.1 255.255.255.0
 bandwidth 384
 ppp multilink
 ppp multilint interleave
 ppp multilink fragment delay 10
 service-policy out VOIP


 int serial 0/0/1:0.1

 no ip add
 no frame-relay interface-dlci
 frame-relay interface-dlci 201 ppp virtual-template 1
 no sh
 class myconfig


 Branch1 is pretty much the same

 when i reload Branch1 I lose connectivity to HQ

 so i go to HQ and go to the sub interface
 and remove
  frame-relay interface-dlci 201 ppp virtual-template1

 add it back in
 do the same on branch1

 and i have connectivity back

 however as soon as i reload either the branc1 or hq router
 I lose connectivity again and i have to issue the same workaround to get
 it to work again

 what do you guys recommend on this issue ?


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[OSL | CCIE_Voice] QRT logs

2011-07-11 Thread Cristobal Priego
hello all

cold you please let me know if the jitter shown on the QRT reports is one
way or round trip ?

thank you
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Re: [OSL | CCIE_Voice] CME H323 signal and media packet source

2011-07-11 Thread Cristobal Priego
on h323 that command h323-gateway v b s voice vlan ip addr binds the media
and the control signaling to that ip address

2011/7/11 Adil Shaikh adil.sha...@gmail.com

 hi expert,

 if a customer ask you to source H323 signal and media from voice vlan
 interface on CME router then what commands be issued?


 i would do following:
 1) Telephony-service
  ip source address voice vlan interface ip address
 2) int vlan voice vlan id
  h323 v b s voice vlan ip add


 Is this what you would do or something else?


 thanks
 -adil



 --
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 _7___|___|_|_|adil.sha...@gmail.com

 . .



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[OSL | CCIE_Voice] WAN QoS MLP LFI

2011-07-10 Thread Cristobal Priego
hello all

I know we have address this issue several times

i'm configuring MLP manually and this is how i have HQ configured

int serial 0/0/1:0.1
bandwidth 384

int serial 0/0/1:0
frame-relay traffic-shapping



class-map VOICE
match ip dscp ef
class-map match-any SIG
match ip dscp cs3
match ip dscp af31


policy-map VOIP
class VOICE
priority percent 33
class SIG
bandwidth percent 5
class class-default
fair-queue


map-class frame-relay myconfig
frame-relay cir 384000
frame-relay mincir 384000
frame-relay be 0
frame-relay bc 3840


int virtual-template 1
ip address 10.10.111.1 255.255.255.0
bandwidth 384
ppp multilink
ppp multilint interleave
ppp multilink fragment delay 10
service-policy out VOIP


int serial 0/0/1:0.1

no ip add
no frame-relay interface-dlci
frame-relay interface-dlci 201 ppp virtual-template 1
no sh
class myconfig


Branch1 is pretty much the same

when i reload Branch1 I lose connectivity to HQ

so i go to HQ and go to the sub interface
and remove
 frame-relay interface-dlci 201 ppp virtual-template1

add it back in
do the same on branch1

and i have connectivity back

however as soon as i reload either the branc1 or hq router
I lose connectivity again and i have to issue the same workaround to get it
to work again

what do you guys recommend on this issue ?
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Re: [OSL | CCIE_Voice] WAN QoS MLP LFI

2011-07-10 Thread Cristobal Priego
nevermind
i figured it out

thanks

2011/7/10 Cristobal Priego cristobalpri...@gmail.com

 hello all

 I know we have address this issue several times

 i'm configuring MLP manually and this is how i have HQ configured

 int serial 0/0/1:0.1
 bandwidth 384

 int serial 0/0/1:0
 frame-relay traffic-shapping



 class-map VOICE
 match ip dscp ef
 class-map match-any SIG
 match ip dscp cs3
 match ip dscp af31


 policy-map VOIP
 class VOICE
 priority percent 33
 class SIG
 bandwidth percent 5
 class class-default
 fair-queue


 map-class frame-relay myconfig
 frame-relay cir 384000
 frame-relay mincir 384000
 frame-relay be 0
 frame-relay bc 3840


 int virtual-template 1
 ip address 10.10.111.1 255.255.255.0
 bandwidth 384
 ppp multilink
 ppp multilint interleave
 ppp multilink fragment delay 10
 service-policy out VOIP


 int serial 0/0/1:0.1

 no ip add
 no frame-relay interface-dlci
 frame-relay interface-dlci 201 ppp virtual-template 1
 no sh
 class myconfig


 Branch1 is pretty much the same

 when i reload Branch1 I lose connectivity to HQ

 so i go to HQ and go to the sub interface
 and remove
  frame-relay interface-dlci 201 ppp virtual-template1

 add it back in
 do the same on branch1

 and i have connectivity back

 however as soon as i reload either the branc1 or hq router
 I lose connectivity again and i have to issue the same workaround to get it
 to work again

 what do you guys recommend on this issue ?


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Re: [OSL | CCIE_Voice] PSTN-WAN call drop scenario

2011-07-09 Thread Cristobal Priego
hello Adil,

I just had the same problem with proctorlabs.

i check the pstn-wan router and the codec gas set to g711

all my regions, my dialpeers for my cube was set to g729

and when i was calling from hq or br1 i was able to call the pstn phone no
problems

i was able to resolve it by going to the h225 trunk and uncheck the checkbox
for Wait for far end h245 terminal capability set

after that i reset the trunk

i called again and as soon as i answer the pstn line

my call dropped

hth

2011/7/7 George Goglidze gogli...@gmail.com

 Hi Adil,

 Without debug h245 asn1 it's difficult to be sure, but I bet on this other
 rack rental they do support g729 codec on incoming dial-peer on PSTN-WAN
 router and thats why the call succeeds.

 Sent from my iPad

 On 7 Jul 2011, at 02:08, Adil Shaikh adil.sha...@gmail.com wrote:

 hi experts,


 in ipexpert lab,  i make call from HQ phone to PSTN-WAN via CUBE, when the
 call is answered by PSTN-WAN phone, call fails. This is due to codec
 mismatch and i can get 'TSC rejected' message in 'debug h245 asn1'. I have
 been succeful in simulating this on everytime in last 5 attempts on ipexpert
 lab.

 few days back, i used other providers lab for which i paid lot of money. i
 did same configuration as i do on ipexpert lab (that is create g729 region
 which talks to HQ and BR1 region as g729. Assign g729 region to GK device
 pool. Assign this device pool to GK trunk).
 (note: I use this trunk to make a 4 digit call to BR2 phone and when i
 do 'show gatekeeper call', i see 16k used. So, i know that GK trunk is
 definately using g729 codec.)
 for PSTN-WAN, i configured CUBE as well. When i make a call to
 PSTN-WAN when the call is answered call does not drop. I was expecting it to
 drop due to codec mismatch.


 Here is my configuration on HQ router:

 voice service voip
 all h t h

 gatkeeper
 zone local VIA http://ipexpert.comipexpert.com
 zone local GK http://ipexpert.comipexpert.com 10.10.110.1
 using local loopback
 zone remote PSTN-WAN http://joblog.comjoblog.com outvia VIA
 zone prefix PSTN-WAN 01191*
 zone prefix GK 212* gw-priority 10 gk-trunk_2
 zone prefix GK 212* gw-priority 9 gk-trunk_1
 zone prefix GK 617* gw-priority 10 gk-trunk_2
 zone prefix GK 617* gw-priority 9 gk-trunk_1
 zone prefix GK 212* gw-priority 0 CME
 zone prefix GK 617* gw-priority 0 CME
 no shut

 int lo 0
 ip add 10.10.110.1 255.255.255.0
 h323-g v b s 10.10.110.1
 h323-g v inter
 h323-g v id VIA ipadd 10.10.110.1
 h323-g v h323-id HQ-RTR

 gateway

 dial-p voice 91 voip
 destination-p 01191T
 incoming called 01191
 session target ras
 dtmf-r h245-al
 no vad


 On CUCM:
 a route pattern 9.011916737485173
 partition: pt-hq-CallToIndia
 discard digit predot

 routelist: rl-gk-CallToIndia
 contains gk trunk
 calling party External number mask: On


 Do you guys see any problem in above config. Have you come across similar
 situation in non-IPexpert lab for which you have paid lot of money?
 If so, how did you resolve it.


 Thanks
 -adil

 --
   .. . .
 _7___|___|_|_| adil.sha...@gmail.comadil.sha...@gmail.com

 . .


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Re: [OSL | CCIE_Voice] CME EM Host not found

2011-07-08 Thread Cristobal Priego
what i'd personally do is

go to telephony service

telephony-service
  no authentication credential secretname psswrd
  no url services http://10.10.202.1/CMEserverForPhone/serviceurl
  no url authentication
http://10.10.202.1/CCMCIP/authenticate.aspsecretname psswrd

and then go to global config mode

and issue

ip http server

that will configure the default services that you need, at least that's what
i personally do instead of creating the services manually
!

2011/7/8 Farkas Péter wormh...@sch.bme.hu

 Gents,

 I am facing an EM issue on CME cannot step over.

 EM is configured like:

 voice logout-profile 1
  number 3002 type normal
 !
 voice user-profile 1
  max-idle-time 10
  user br2phn3 password adgjm
  number 3005 type normal
  speed-dial 1 3006
 !
 ephone  2
  privacy off
  device-security-mode none
  mac-address 0005.9A3C.7800
  ephone-template 2
  type CIPC
  logout-profile 1
 !
 telephony-service
  authentication credential secretname psswrd
  url services http://10.10.202.1/CMEserverForPhone/serviceurl
  url authentication http://10.10.202.1/CCMCIP/authenticate.asp secretname
 psswrd
 !

 When I press the Services button on CIPC only Host not found error
 message apperars on the phone. So I cannot reach EM menu at all.

 I can see phone tries to reach CME:

 Jul  8 14:07:07.436: Fri, 08 Jul 2011 14:07:07 GMT 192.168.208.1
 /CMEserverForPhone/serviceurl ok
Protocol = HTTP/1.1 Method = GET Query =
 locale=English_United_Statesname=SEP00059A3C7800

 Any thought to resolve?

 Peter
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Re: [OSL | CCIE_Voice] CUE - Voice Mail Access from SIP Phone

2011-07-07 Thread Cristobal Priego
By default, SIP uses in-band signaling, sending the DTMF information in the
voice stream. However, you can configure it to use RTP-NTE, SIP INFO
messages, SIP NOTIFY messages, or KPML for transmitting DTMF tone
information.

RTP-NTE is an in-band DTMF relay method, which uses RTP Named Telephony
Event (NTE) packets to carry DTMF information instead of voice.
If RTP-NTE is configured, SDP is used to negotiate the payload type value
for NTE packets and the events that will be sent using NTE.

Cisco has two out-of-band procedures for DTMF relay. One uses SIP INFO
methods, and the other uses SIP NOTIFY methods.
The SIP INFO method sends DTMF digits in INFO messages. It is always
enabled. When a gateway receives an INFO message containing DTMF relay
information, it sends the corresponding tone.

NOTIFY-based out-of-band DTMF relay is negotiated by including a Call-Info
field in the SIP INVITE and response messages. This field indicates an
ability to use NOTIFY for DTMF tones and the duration of each tone in
milliseconds. Using this method can help SIP gateways interoperate with
Skinny phones.


2011/7/7 adam compton com...@gmail.com

 Yes, this is the behavior I have seen in the lab.  Because there are 2 call
 legs, CME handles the different dtmf types between the phone and CUE.  I
 could be wrong, but I think 7960s and maybe other phones as well do not
 support sip-notify for dtmf-relay.

 Regards,
 Adam Compton

 On Thu, Jul 7, 2011 at 11:06 AM, mgscip gpsvoiceexpe...@yahoo.com wrote:

  Hi Experts.

 Please clarify on this !

   *From:* mgscip gpsvoiceexpe...@yahoo.com
 *To:* ccie ccie_voice@onlinestudylist.com
 *Sent:* Wednesday, July 6, 2011 10:21 AM
 *Subject:* [OSL | CCIE_Voice] CUE - Voice Mail Access from SIP Phone

  Hi All,

 When i dial the Voicemail pilot number from SIP Phone dtmf is not
 recognized by the CUE .

 once enabled the dtmf-relay rtp-nte in the voice register pool it's
 start working.

 is the way it's working ?

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Re: [OSL | CCIE_Voice] CME to CUCM Via CUBE configuration assistant

2011-07-01 Thread Cristobal Priego
if you remove the call start interwork and on your ucm h225 trunk un check
wait for far end TCS set it should work as well

2011/7/1 Art Joe Babakhani Gharibian ciscoie2...@gmail.com

 One more thing. I couldn't answer the call without call start interwork
 so I added the following class to both incoming and outgoing dial-peers.

 voice class h323 5
 h225 timeout tcp establish 5
 h225 display-ie ccm-compatible
 call start interwork

 Thanks,
 Joe
 On Fri, Jul 1, 2011 at 11:34 AM, Art Joe Babakhani Gharibian 
 ciscoie2...@gmail.com wrote:

 thanks, it was the codec I changed it and now is working.


 On Fri, Jul 1, 2011 at 7:57 AM, manishankar pandey 
 manishankar...@yahoo.com wrote:

 Yes this has to do with Codec mismatch. Or the Interpretation of Fast
 start and Slow start.

 Try using the below option

 1) Create Voice class and then bind it to dial-peer
 Example :

 voice class h323 1
  h225 timeout tcp establish 5
  h225 display-ie ccm-compatible
   call start fast
 !
 voice class h323 5
  h225 timeout tcp establish 5
  h225 display-ie ccm-compatible
   call start interwork
 !
 voice class h323 6
  h225 timeout tcp establish 5
  h225 display-ie ccm-compatible
   call start slow

 And under Dial-peer VoIP, try testing with each Voice class h323 command

  voice-class h323 1/5/6
 Thanks
 --- On *Fri, 7/1/11, Emin Guliyev eguli...@fidelus.com* wrote:


 From: Emin Guliyev eguli...@fidelus.com
 Subject: Re: [OSL | CCIE_Voice] CME to CUCM Via CUBE configuration
 assistant
 To: Art Joe Babakhani Gharibian ciscoie2...@gmail.com, 
 ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 Date: Friday, July 1, 2011, 6:01 PM


  Looks like it is failing due to codec negotiation.



 Media negotiation failure

 Typical scenarios include:

 •[image: Description: http://www.cisco.com/en/US/i/templates/blank.gif]No
 codec match occurred.

 •[image: Description: http://www.cisco.com/en/US/i/templates/blank.gif]H.323
 or H.245 problem leading to failure in media negotiation

  65

 CC_CAUSE_BEARER_CAPABILITY_
 NOT_IMPLEMENTED

 Indicates that the equipment sending this cause does not support the
 bearer capability requested.



 Jun 29 21:49:46.402: //33/8A1D25B080B9/H323/cch323_h225_send_release: Cause
 = 65; Location = 0





 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Art Joe Babakhani
 Gharibian
 *Sent:* Friday, July 01, 2011 12:47 AM
 *To:* ccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] CME to CUCM Via CUBE configuration
 assistant



 Has anyone tested a call from CME  to   CUCM via Gatekeeper and CUBE?

 Could you please provide a working configuration sample?

 I have configured the gatekeeper to send the calls from CME to CUCM via
 CUBE. CME calls drop after matching an incoming voip dialpeer in the CUBE.
 I can’t figure out why my calls get disconnected once they match an incoming
 dial-peer. It is worth mentioning that I use that exact same dial-peers to
 send calls from CUCM to CME and they work just fine.

 I am pasting some show and debug outputs.

 Thanks,

 Joe

 *ON the CUBE  show run | s dial-peer  *

 dial-peer voice 20 voip

 incoming called-number .

 dtmf-relay h245-alphanumeric

 no vad

 dial-peer voice 21 voip

 destination-pattern [1,3,5]...

 voice-class codec 1

 session target ras

 dtmf-relay h245-alphanumeric

 no vad





 *show run | s gatek *

 gatekeeper

 zone local Spain ipexpert.com outvia VGK

 zone local US ipexpert.com outvia VGK

 zone local VGK ipexpert.com

 zone remote PSTN-WAN ipexpert.com 10.10.100.2 1719

 zone prefix US 1... gw-priority 10 gk-trunk_1

 zone prefix US 1... gw-priority 9 gk-trunk_2

 zone prefix Spain 3...

 zone prefix US 5... gw-priority 10 gk-trunk_1

 zone prefix US 5... gw-priority 9 gk-trunk_2

 gw-type-prefix 1#* default-technology

 no shutdown



 *on the CUBE debug voice dialpeer *



 Jun 29 21:49:12.911: //-1/762755B380B3/DPM/dpAssociateIncomingPeerCore:

Calling Number=3006, Called Number=5002, Voice-Interface=0x0,

Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
 Type=PEER_TYPE_VOICE,

Peer Info Type=DIALPEER_INFO_SPEECH

 Jun 29 21:49:12.911: //-1/762755B380B3/DPM/dpAssociateIncomingPeerCore:

Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=20

 Jun 29 21:49:12.911: //-1/762755B380B3/DPM/dpAssociateIncomingPeerCore:

Calling Number=3006, Called Number=5002, Voice-Interface=0x0,

Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
 Type=PEER_TYPE_VOICE,

Peer Info Type=DIALPEER_INFO_SPEECH



 Jun 29 21:49:12.911: //-1/762755B380B3/DPM/dpAssociateIncomingPeerCore:

Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=20





 *On the CUBE  debug cch323 h225 *



 Jun 29 21:49:46.398: //-1//H323/cch323_h225_receiver:
 Received msg of type SETUPIND_CHOSEN

 Jun 29 21:49:46.398: //-1//H323/setup_ind: Entry

 Jun 29 21:49:46.398: 

Re: [OSL | CCIE_Voice] B-ACD Drop Through

2011-07-01 Thread Cristobal Priego
i don't think that;s configurable

2011/6/30 Pablo Meneses pmenese...@gmail.com

 Hi experts,

 Is there a way to display 1 call on phone when using B-ACD Drop Through?

 I think that it will display 1 call on queue and that there is no way for
 it to display 1 call on phone, but, I better ask.

 Regards,

 -Pablo Meneses.

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Re: [OSL | CCIE_Voice] CUE in SRST.. Failover and Fall back

2011-06-30 Thread Cristobal Priego

If you test failover by stopping the ccm  service, it is expected
If you shut down the wan ports the ports should register back
Enviado desde mi iPhone

El Jun 29, 2011, a las 21:06, sandeep.ra...@wipro.com escribió:


Hi All,

I have CUE in SRST mode and it works fine, but when i bring back  
CUCM i dont see the CTI Route point and CTI ports getting registered  
at all unless i reboot the CUE module.


Is this the way it normally behaves or am i unaware of some  
configuration to fix this behaviour.


Also can you please jot down few points on how should i configure  
CUE in SRST mode.


Thanks
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[OSL | CCIE_Voice] ESW configuration

2011-06-30 Thread Cristobal Priego
Hello all

for the lab on the ESW modules on SiteB and SiteC which one is the best
approach

switchport trunk encap dotq1
switchport mode trunk
switchport trunk native vlan XX
switchport voice vlan XX
spanning-tree portfast trunk
mls qos trust cos

or

switchport mode acess
switchport access vlan XX
switchport voice vlan XX
spanning-tree portfast ?
mls qos trust cos


This links shows how to configure ESW on ISR 2800 ios 12.4(T)
http://www.cisco.com/en/US/products/hw/modules/ps2797/products_configuration_example09186a00808066b8.shtml#step4


any comments ?
I'd really appreciate those

thanks
wheneve
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Re: [OSL | CCIE_Voice] ESW configuration

2011-06-30 Thread Cristobal Priego
thank you guys


2011/6/30 Peterson, Ryan ryan.peter...@pcmall.com

 I would personally keep it as simple as possible unless the book tells you
 to add something else…

 ** **

 switchport mode trunk
 switchport trunk native vlan XX   Add only if it tells you to have a PC
 behind the phone.
 switchport voice vlan XX

 ** **

 I would not go with the access method unless spanning-tree portfast is a
 requirement as access on ESW is not stable.

 ** **

 Good luck!

 ** **

 Ryan

 ** **

 ** **

 ** **

 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Cristobal Priego
 *Sent:* Thursday, June 30, 2011 7:03 PM
 *To:* ccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] ESW configuration

 ** **

 Hello all

 for the lab on the ESW modules on SiteB and SiteC which one is the best
 approach

 switchport trunk encap dotq1
 switchport mode trunk
 switchport trunk native vlan XX
 switchport voice vlan XX
 spanning-tree portfast trunk
 mls qos trust cos

 or

 switchport mode acess
 switchport access vlan XX
 switchport voice vlan XX
 spanning-tree portfast ?
 mls qos trust cos


 This links shows how to configure ESW on ISR 2800 ios 12.4(T)

 http://www.cisco.com/en/US/products/hw/modules/ps2797/products_configuration_example09186a00808066b8.shtml#step4


 any comments ?
 I'd really appreciate those

 thanks
 wheneve

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Re: [OSL | CCIE_Voice] Passed

2011-06-29 Thread Cristobal Priego
congratulatins
and thank you for the posts and explanations that you gave to the voice
community

2011/6/29 George Goglidze gogli...@gmail.com

 Hi all,

 As the subject suggests, it's official, I'm dual CCIE #19926 RS and Voice
 starting yesterday.

 Finally, it's a relief, no more studying late, spending weekends on a
 computer, making calls, my neighbours think I escaped a psychiatric, after
 hearing voices at night TEST VOICEMAIL FOR HQ PHONE 1 and my personal
 favourite YOUR POSITION IN QUEUE IS :)

 I would like to transmit a very special thank you to IPExpert, and
 especially Vik Malhi. He's definitely made difference for me as a trainer.
 Just when you think you know it all about something, he would come up with
 something to prove me wrong, to show me the gaps in my knowledge I didn't
 know existed.

 Thank you Vik!

 Thanks to everyone on this forum too, there are many good people on the
 forum, with big knowledge, and more importantly willing to share it. It was
 big fun, I enjoyed the process a lot.

 By the way, I made it technically on my first attempt. Well, I payed once
 only, although I went to exam 3 times.
 1st time, technical problems, Cisco gave me free retake voucher, 2nd time,
 again technical problems, again free voucher.
 To be honest after so many free retakes, I'm not even sure if I really
 passed, or I was costing them too much so they decided to give it to me :-),
 good samaritans these cisco guys.

 Wish you all good luck, and don't get frustrated, and most importantly have
 fun.

 Regards,



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[OSL | CCIE_Voice] CME octo line functionality

2011-06-28 Thread Cristobal Priego
Hello all,

I've been reading the blog about octo lines on cme
however i still doesn't clearly understand how the max-calls-per-button and
busy-trigger-per-button work and it's correlation to shared lines
could you point me to a document or if you could, please explain that to me?
I'd really appreciated it
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Re: [OSL | CCIE_Voice] CUE NTP synchronisation with HQ Router Loopback 0

2011-06-28 Thread Cristobal Priego
yes both only if that router is an NTP master, otherwise CUE will ignore it

i've tested this scneario multiple times and connectivity is not an issue

your router could receive time form the pstn:  ntp server 10.10.100.2
and if you want it to provide clocking to your netowrk devices you must make
that router an ntp master, issue the show ntp associations command to check
the stratum of the pstn clock and make the router an ntp master 7 or
whatever

otherwise CUE will ignore an NTP source that is not the service-engine ip
address

you



2011/6/27 Bill Lake whl...@gmail.com

 I have tested this with routers but not with the CUE module.

 Routers will definitely pull their time from a router synced with
 another NTP source.

 On Mon, Jun 27, 2011 at 4:55 PM, Alan Gardner agard...@ctclc.com wrote:
  According to Vik you do not need the NTP Master.
 
 
 
  Try doing a no shut in the Service module interface!!!
 
 
 
  Best Regards,
 
 
 
  Alan Gardner
 
  Chief Executive Officer
 
  Current Technologies Computer Learning Center
 
  PH. 219.764.3800
 
  agard...@ctclc.com
 
  MCT, MCSE (Messaging  Security), MCSD,
 
  MSF Trainer, MCNI, MCNE, MCSA, CCNA, , ITIL
 
  CCNP, CCVP, CCIP, CCDP, CCIE # 22758, CCSI
 
  A+, MCDST, Network+, Server+, IT Project+,
 
  EBIZ, CTT+, MCDBA CCNT, CISSP, CWNA, CTP, CIW
 
  Microsoft Server 2008 Enterprise Administrator
 
 
 
  From: ccie_voice-boun...@onlinestudylist.com
  [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Chris
 Martin
  Sent: Monday, June 27, 2011 2:27 PM
  To: sandeep.ra...@wipro.com
  Cc: ccie_voice@onlinestudylist.com
  Subject: Re: [OSL | CCIE_Voice] CUE NTP synchronisation with HQ Router
  Loopback 0
 
 
 
  Believe you also need ntp master command.
 
  Chris
 
  On Mon, Jun 27, 2011 at 11:56 AM, sandeep.ra...@wipro.com wrote:
 
  Hello All,
 
 
 
 
 
  When asked to configure NTP server for CUE as the HQ Router's loopback IP
 or
  HQ routers Voice Vlan IP,
  should i configure NTP master command on the HQ router besides below
  configuration?
 
 
 
 
 
  HQ Router:
 
  ntp server 10.10.100.2external NTP
  ntp source loopback 0
  clock timezone PST -8
  clock summer-time PDT recurring
 
 
 
 
 
  I tried configuring the CUE module with HQ routers loopback-IP and Voice
  VLAN IP.
  But faced errors saying NTP server is not accessible. Can any one please
  tell me what extra should i configure on HQ router or CUE router in order
 to
  get the NTP synchronization on the CUE successful.
 
 
 
 
 
  Thanks
 
  Please do not print this email unless it is absolutely necessary.
 
  The information contained in this electronic message and any attachments
 to
  this message are intended for the exclusive use of the addressee(s) and
 may
  contain proprietary, confidential or privileged information. If you are
 not
  the intended recipient, you should not disseminate, distribute or copy
 this
  e-mail. Please notify the sender immediately and destroy all copies of
 this
  message and any attachments.
 
  WARNING: Computer viruses can be transmitted via email. The recipient
 should
  check this email and any attachments for the presence of viruses. The
  company accepts no liability for any damage caused by any virus
 transmitted
  by this email.
 
  www.wipro.com
 
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  visit www.ipexpert.com
 
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  www.PlatinumPlacement.com
 
 
 
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Re: [OSL | CCIE_Voice] MLS QoS issue

2011-06-27 Thread Cristobal Priego
the output doesn't show anything like that

you only only posted mls qos map cos-dscp 0 8 16 24 32 46 48 56
which means
CS 0 will be mapped to dscp 0
CS 1 to dscp 8
 CS 2 to dscp 16
 CS 3 to dscp 24
 CS 4 to dscp 32
 CS 5 to dscp 46
CS 6 to dscp 48
CS 7 to dscp 56
2011/6/27 Mike mik...@msn.com

 Hi all I’m trying to lab up a QOS run through and for some reason the
 switch is resetting everything to 0

 ** **

 Switch config:

 ** **

 mls qos map cos-dscp 0 8 16 24 32 46 48 56

 mls qos

 ** **

 interface FastEthernet0/3

 switchport mode access

 switchport voice vlan 10

 mls qos trust dscp

 spanning-tree portfast

 ** **

 monitor session 1 source interface Fa0/3 tx

 monitor session 1 destination interface Fa0/6

 ** **

 Wireshark reports DSCP field 0x00.

 ** **

 ** **

 Any ideas?

 ** **

 I tried mls qos trust cos and same thing.

 ** **

 The phone plugged into the switchport is a 7941

 ** **

 ** **

 Thanks,

 Mike

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Re: [OSL | CCIE_Voice] CUE NTP synchronisation with HQ Router Loopback 0

2011-06-27 Thread Cristobal Priego
you need the ntp master command on the router with the proper stratum set
otherwise when you go through the cue initialization and you enter the
router's loopback it;s going to say not reachable and is going to ask for a
different ntp server

2011/6/27 Alan Gardner agard...@ctclc.com

 According to Vik you do not need the NTP Master.

 ** **

 Try doing a no shut in the Service module interface!!!

 ** **

 Best Regards,

 ** **

 Alan Gardner

 Chief Executive Officer

 Current Technologies Computer Learning Center

 PH. 219.764.3800

 agard...@ctclc.com

 MCT, MCSE (Messaging  Security), MCSD, 

 MSF Trainer, MCNI, MCNE, MCSA, CCNA, , ITIL

 CCNP, CCVP, CCIP, CCDP, CCIE # 22758, CCSI

 A+, MCDST, Network+, Server+, IT Project+,

 EBIZ, CTT+, MCDBA CCNT, CISSP, CWNA, CTP, CIW

 Microsoft Server 2008 Enterprise Administrator

 ** **

 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Chris Martin
 *Sent:* Monday, June 27, 2011 2:27 PM
 *To:* sandeep.ra...@wipro.com
 *Cc:* ccie_voice@onlinestudylist.com

 *Subject:* Re: [OSL | CCIE_Voice] CUE NTP synchronisation with HQ Router
 Loopback 0

 ** **

 Believe you also need ntp master command.

 Chris

 On Mon, Jun 27, 2011 at 11:56 AM, sandeep.ra...@wipro.com wrote:

 Hello All,

  

  

 When asked to configure NTP server for CUE as the HQ Router's loopback IP
 or HQ routers Voice Vlan IP,
 should i configure NTP master command on the HQ router besides below
 configuration?

  

  

 HQ Router:

 ntp server 10.10.100.2external NTP
 ntp source loopback 0
 clock timezone PST -8
 clock summer-time PDT recurring

  

  

 I tried configuring the CUE module with HQ routers loopback-IP and Voice
 VLAN IP.
 But faced errors saying NTP server is not accessible. Can any one please
 tell me what extra should i configure on HQ router or CUE router in order to
 get the NTP synchronization on the CUE successful.

  

  

 Thanks

 *Please do not print this email unless it is absolutely necessary. *

 The information contained in this electronic message and any attachments to
 this message are intended for the exclusive use of the addressee(s) and may
 contain proprietary, confidential or privileged information. If you are not
 the intended recipient, you should not disseminate, distribute or copy this
 e-mail. Please notify the sender immediately and destroy all copies of this
 message and any attachments. 

 WARNING: Computer viruses can be transmitted via email. The recipient
 should check this email and any attachments for the presence of viruses. The
 company accepts no liability for any damage caused by any virus transmitted
 by this email. 

 www.wipro.com 


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Re: [OSL | CCIE_Voice] IPPM getting invalid device

2011-06-26 Thread Cristobal Priego
does yourapp  user matches the user in CUPS ? is ip phone messenger on  in
CUPS?
do you have the CTI enabled and the CTI controll all devices group
associated to your app user ?

2011/6/26 Alan Gardner agard...@ctclc.com

 Invalid Device

 You were trying to access IP Phone Messenger service from a device not
 provisioned on Cisco CallManager server. Please work with your system
 administrator to get this device configured.

  

 I am running CUCM 7 and CUPS 7 and I have completed the following steps:**
 **

  

 1. Configured IP PhoneMSG service with CUPS in URL

 2. Created PhoneMessenger application user and added IPC and 7965 phones
 in Controlled Devices

 3. Associated end users with primary DNs on IPC and 7965 and configured end
 users with Standard CTI Enabled user group permissions.

 4. Subscribed both phones with IP PhoneMSG service

 ** **

 Any ideas

 ** **

 ** **

 Best Regards,

 ** **

 Alan Gardner

 ** **

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[OSL | CCIE_Voice] FRF.12 question

2011-06-26 Thread Cristobal Priego
hello all

when you configure FRF.12 manually on your seial interfaces on HQ and BR2
on the HQ router where the same physical interface is used to connect BR1
and BR2,  BR1 link isn't affected at all because traffic shaping isn't
enabled on the physical interface, correct ?
so i can pretty much ignore that link in regards to a basic QoS config if
not needed

thanks
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Re: [OSL | CCIE_Voice] priority-queue out

2011-06-25 Thread Cristobal Priego
I have the same understanding the priority queue when enabled will be
serviced until the last frame has been transmitted and the sape/share will
be ignored

2011/6/24 Paul Dardinski pa...@marshallcomm.com

 My understanding is the same as yours. Include priority queue and at that
 point presumably all ranged (assigned) traffic will go out as priority on
 that queue regardless. Assuming this traffic was to exceed allocations on
 other queues, they are indeed crowded out, making share/shape irrelevant…..
 

 ** **

 Paul (#16842 RS/Sec)

 ** **

 ** **

 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Dial Peer
 *Sent:* Friday, June 24, 2011 3:02 PM

 *To:* ccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] priority-queue out

 ** **

 I'm confused by all the discussions about priority-queue out command.***
 *

  

 From my understanding, this command eanble strict priority, which means
 none of the other queues can transmit until the priority queue was emptied.
 

  

 If this command was enabled, both the srr-queue bandwidth share and
 srr-queue bandwidth shape will be pointless.  Because the switch will
 neither share nor shape until the priority queue was emptied.

  

 Did I understand this wrong?

  

 Thanks!

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Re: [OSL | CCIE_Voice] MVA calling name

2011-06-25 Thread Cristobal Priego
make sure that on your RDP you have the Display field set, it will not
inherit the name from the line you're associating it with

2011/6/24 Victor Malyuga victor_maly...@yahoo.com

 If I make a call from the remote destination number via MVA the calling
 name is not displayed on the called phone while the phone is ringing. Only
 the extension associated with the RDP is displayed. Once the call is
 connected the name and extension is displayed on the called phone.

 Did someone come across that?

 Line in the RDP profile has display name configured. RDP number matches
 when I call the mva number because I  am not prompted to enter RDP number.

 The same problem is described in this post

 https://supportforums.cisco.com/message/3284389

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Re: [OSL | CCIE_Voice] Lab result status

2011-06-23 Thread Cristobal Priego
yes if you go to the ccie policies uder FAQs it actually says is usually 48
hours

2011/6/23 Adil Shaikh adil.sha...@gmail.com

 One week no result. This is too long.

 Has anyone read any fine print which specifies within what timeframe they
 will be providing result?

 -adil

 On Fri, Jun 24, 2011 at 3:50 AM, Pablo Meneses pmenese...@gmail.comwrote:

 I just sent her an email. Let's see who it goes.

 -Pablo Meneses.

 On Thu, Jun 23, 2011 at 10:38 AM, William Affeldt 
 william.affe...@yahoo.com wrote:

 Her name is Kathe Saccenti. I would just call Support and ask to talk to
 her. She is in charge of the proctors and the testing environment.

 --- On *Thu, 6/23/11, Pablo Meneses pmenese...@gmail.com* wrote:


 From: Pablo Meneses pmenese...@gmail.com
 Subject: Re: [OSL | CCIE_Voice] Lab result status
 To: William Affeldt william.affe...@yahoo.com
 Cc: George Goglidze gogli...@gmail.com, ccie voice 
 ccie_voice@onlinestudylist.com
 Date: Thursday, June 23, 2011, 4:23 PM


 Hello Bill,

 Thanks for that information. Do you know Kathe's last name?

 Regards.

 -Pablo Meneses.

 On Wed, Jun 22, 2011 at 9:09 PM, William Affeldt 
 william.affe...@yahoo.comhttp://us.mc1146.mail.yahoo.com/mc/compose?to=william.affe...@yahoo.com
  wrote:

  You should look at this as a positive. If they deleted your test they
 may just give you a passing grade. You will never know though. The person
 you should contact is the lab ops manager. Her name is Kathe. She is a nice
 person and will take your call at any time.

 Sent from my iPhone

 On Jun 22, 2011, at 4:56 PM, George Goglidze 
 gogli...@gmail.comhttp://us.mc1146.mail.yahoo.com/mc/compose?to=gogli...@gmail.com
 wrote:

  WTF man, I can't believe this is happening to you.
 I wouldn't be able to sleep until I got the results.


 On Wed, Jun 22, 2011 at 10:46 PM, Pablo Meneses 
 http://us.mc1146.mail.yahoo.com/mc/compose?to=pmenese...@gmail.com
 pmenese...@gmail.comhttp://us.mc1146.mail.yahoo.com/mc/compose?to=pmenese...@gmail.com
  wrote:

 Can you believe that I haven't received the answer yet!
 This has been the longest wait ever.
 I even opened a ticket with Cert Support and so far I have just received
 templates.
 Does someone else have gone through this?
 I'm starting to think that they messed up on my exam.

 -Pablo Meneses.


 On Wed, Jun 22, 2011 at 2:47 AM, Julien Krieger 
 http://us.mc1146.mail.yahoo.com/mc/compose?to=krieger.jul...@gmail.com
 krieger.jul...@gmail.comhttp://us.mc1146.mail.yahoo.com/mc/compose?to=krieger.jul...@gmail.com
  wrote:

 Hi Pablo,
 How did it go? Have you passed?


 2011/6/21 Pablo Meneses 
 http://us.mc1146.mail.yahoo.com/mc/compose?to=pmenese...@gmail.com
 pmenese...@gmail.comhttp://us.mc1146.mail.yahoo.com/mc/compose?to=pmenese...@gmail.com
 

 Thanks Cris.

 Looking forward to getting my result.


 On Tue, Jun 21, 2011 at 1:23 PM, Cristobal Priego 
 http://us.mc1146.mail.yahoo.com/mc/compose?to=cristobalpri...@gmail.com
 cristobalpri...@gmail.comhttp://us.mc1146.mail.yahoo.com/mc/compose?to=cristobalpri...@gmail.com
  wrote:

 i know man
 i took my last attempt on a Friday as well In San Jose

 i didn't get my score until next Monday at 14:45 PST


 2011/6/21 Pablo Meneses 
 http://us.mc1146.mail.yahoo.com/mc/compose?to=pmenese...@gmail.com
 pmenese...@gmail.comhttp://us.mc1146.mail.yahoo.com/mc/compose?to=pmenese...@gmail.com
 

 Hopefully that's true. Longest wait EVER!!!

 -Pablo Meneses.


 On Tue, Jun 21, 2011 at 1:19 PM, Cristobal Priego 
 http://us.mc1146.mail.yahoo.com/mc/compose?to=cristobalpri...@gmail.com
 cristobalpri...@gmail.comhttp://us.mc1146.mail.yahoo.com/mc/compose?to=cristobalpri...@gmail.com
  wrote:

 you will get today, wait 2.5 more hours
 that's what happens when you take it on Friday

   2011/6/21 Pablo Meneses 
 http://us.mc1146.mail.yahoo.com/mc/compose?to=pmenese...@gmail.com
 pmenese...@gmail.comhttp://us.mc1146.mail.yahoo.com/mc/compose?to=pmenese...@gmail.com
 

  Hi experts,

 I am wondering how long it takes for you to received your score report
 this days. I've taken my lab last Friday and I have NOT received the answer
 yet. Yesterday I logged to the CCIE Online Tool and it said HOLD on the
 PASS/FAIL field, does someone know what that means? Has someone else gone
 through this?

 I will look forward to your reply as I figure most of you understand the
 feeling of not having the answer.

 Regards.

 -Pablo Meneses.

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Re: [OSL | CCIE_Voice] route group distribution algorithm

2011-06-22 Thread Cristobal Priego
the reason why i am asking is because in the lab they're not that specific

2011/6/22 CCIE STUDENT cciefo...@hotmail.com

 You rarely even do it in the real world
 -Original Message-
 From: Randall Saborio ill2...@gmail.com
 Date: Wed, 22 Jun 2011 13:26:24
 To: cristobalpri...@gmail.com
 Cc: ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] route group distribution algorithm

 Can you name when on the lab would you ever configure two devices on same
 route group?

 I don't think this is ever done on the lab. Even if it did, its not a
 matter of preference, but a matter of matching the task requirements.


 On Tue, Jun 21, 2011 at 7:41 PM, Cristobal Priego 
 cristobalpri...@gmail.com mailto:cristobalpri...@gmail.com  wrote:
  guys,

 for the lab, whenever you configure your route groups

 which distribution algorithm is better

 circular or  top down

 i use top down all the time, but i'd like to know your opinion on this

 thanks

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 --
 Randall da ill Saborio
  CCIE Voice Wannabe #10054675811

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Re: [OSL | CCIE_Voice] UCM LOCATION QUESTION

2011-06-22 Thread Cristobal Priego
Stephen,

the call will fail
you're going to get a message on the phone saying not enough bandwidth and
the call will drop

2011/6/22 Stephen Manuel srman...@bellsouth.net

 Experts, 

 ** **

 I have a somewhat basic question just clear up something that is lingering
 in my mind. 

 ** **

 The example is this:

 ** **

 In a multiple remote location site company, with a Centralized UCM
 architecture, each site has it’s own device pool, region, and location
 settings. 

 ** **

 All the DN’s on the phones for all sites are in the same PT, meaning calls
 between sites are on net via their existing WAN infrastructure and not the
 SIP Trunk/PSTN. 

 ** **

 Remote site A is rather large 50-60 phones.

 ** **

 Remote site B is rather small 10-15 phones. 

 ** **

 Calls between regions use g729

 Calls to the PSTN using a SIP Trunk use g711

 ** **

 The location setting for site A is unlimited.

 The location setting for site B is 256

 ** **

 What I am wondering is this, Site B has 3 calls to the PSTN using the SIP
 Trunk using g711 @ 80k per call which equals 240k. The SIP trunk is also in
 it’s own Device Pool, region, etc…

 ** **

 Now someone from Site A calls site B, the bandwidth at Site A is unlimited
 so no issue there. 

 According to the Cisco Research I’ve done, a call requires 2 streams. 

 Since Site B is using 240 of 256k of bandwidth on the 3 PSTN/SIP Trunk
 calls, when a g729 calls comes inbound  which requires 24k  which causes
 Site B location setting to be exceeded, will the call fail or succeed ??**
 **

 ** **

 IMO, the call will fail.

 ** **

 Thanks, 

 Stephen Manuel

 ** **

 ** **

 ** **

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Re: [OSL | CCIE_Voice] UCM LOCATION QUESTION

2011-06-22 Thread Cristobal Priego
my thoughts are if you want to use QoS you actually can't do a lot ob CAC
with QoS
you will degrade the quality of the call, but i don't think you can fully
implement CAC with QoS. this is just my thought though.
i've used only 2 types of CAC mechanisim. locations CAC or GK CAC
my 2 cents
hth

2011/6/22 Stephen Manuel srman...@bellsouth.net

 That’s what I thought, but have not tested it to know for sure. 

 ** **

 As a follow up, as a general design what would be better. 

 ** **

 Using the Location setting as a form of Call-Admission-Control, or use
 various QOS and IOS mechanisms to implement CAC on the Gateway Routers. **
 **

 ** **

 IMO, I would rather have CAC implemented on the Routers, leaving UCM to
 process calls, which is why I have usually set the Audio Bandwidth setting
 under Locations to Unlimited. 

 ** **

 Thanks, 

 ** **

 Stephen Manuel

 ** **

 *From:* Cristobal Priego [mailto:cristobalpri...@gmail.com]
 *Sent:* Wednesday, June 22, 2011 2:37 PM
 *To:* Stephen Manuel
 *Cc:* ccie_voice@onlinestudylist.com
 *Subject:* Re: [OSL | CCIE_Voice] UCM LOCATION QUESTION

 ** **

 Stephen,

 the call will fail
 you're going to get a message on the phone saying not enough bandwidth
 and the call will drop

 2011/6/22 Stephen Manuel srman...@bellsouth.net

 Experts, 

  

 I have a somewhat basic question just clear up something that is lingering
 in my mind. 

  

 The example is this:

  

 In a multiple remote location site company, with a Centralized UCM
 architecture, each site has it’s own device pool, region, and location
 settings. 

  

 All the DN’s on the phones for all sites are in the same PT, meaning calls
 between sites are on net via their existing WAN infrastructure and not the
 SIP Trunk/PSTN. 

  

 Remote site A is rather large 50-60 phones.

  

 Remote site B is rather small 10-15 phones. 

  

 Calls between regions use g729

 Calls to the PSTN using a SIP Trunk use g711

  

 The location setting for site A is unlimited.

 The location setting for site B is 256

  

 What I am wondering is this, Site B has 3 calls to the PSTN using the SIP
 Trunk using g711 @ 80k per call which equals 240k. The SIP trunk is also in
 it’s own Device Pool, region, etc…

  

 Now someone from Site A calls site B, the bandwidth at Site A is unlimited
 so no issue there. 

 According to the Cisco Research I’ve done, a call requires 2 streams. 

 Since Site B is using 240 of 256k of bandwidth on the 3 PSTN/SIP Trunk
 calls, when a g729 calls comes inbound  which requires 24k  which causes
 Site B location setting to be exceeded, will the call fail or succeed ??**
 **

  

 IMO, the call will fail.

  

 Thanks, 

 Stephen Manuel

  

  

  


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 ** **

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Re: [OSL | CCIE_Voice] how to find b-acd on cisco documentation

2011-06-22 Thread Cristobal Priego
you will be able to open pdf files

2011/6/22 Chris Green voice5...@yahoo.com

 Actually, this is ended with PDF file, can we open PDF file during exam
 time.

 --
 *From:* ccie_voice-requ...@onlinestudylist.com 
 ccie_voice-requ...@onlinestudylist.com

 *To:* ccie_voice@onlinestudylist.com
 *Sent:* Wed, 22 June, 2011 11:37:28
 *Subject:* CCIE_Voice Digest, Vol 64, Issue 177

 Send CCIE_Voice mailing list submissions to
 ccie_voice@onlinestudylist.com

 To subscribe or unsubscribe via the World Wide Web, visit
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 than Re: Contents of CCIE_Voice digest...


 Today's Topics:

   1. Re: route group distribution algorithm (Randall Saborio)
   2. UCM LOCATION QUESTION (Stephen Manuel)
   3. Re: route group distribution algorithm
   (givemeccievoice2...@gmail.com)
   4. Re: UCM LOCATION QUESTION (Cristobal Priego)


 --

 Message: 1
 Date: Wed, 22 Jun 2011 10:06:48 -0600
 From: Randall Saborio ill2...@gmail.com
 To: Cristobal Priego cristobalpri...@gmail.com
 Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] route group distribution algorithm
 Message-ID: BANLkTin5zSzWXV08XPM=+x5je2tspgf...@mail.gmail.com
 Content-Type: text/plain; charset=iso-8859-1

 Or maybe because it is never used or it never matters. If it would matter,
 then they would be specific about it.

 On Wed, Jun 22, 2011 at 9:59 AM, Cristobal Priego 
 cristobalpri...@gmail.com
  wrote:

  the reason why i am asking is because in the lab they're not that
 specific
 
 
  2011/6/22 CCIE STUDENT cciefo...@hotmail.com
 
  You rarely even do it in the real world
  -Original Message-
  From: Randall Saborio ill2...@gmail.com
  Date: Wed, 22 Jun 2011 13:26:24
  To: cristobalpri...@gmail.com
  Cc: ccie_voice@onlinestudylist.com
  Subject: Re: [OSL | CCIE_Voice] route group distribution algorithm
 
  Can you name when on the lab would you ever configure two devices on
 same
  route group?
 
  I don't think this is ever done on the lab. Even if it did, its not a
  matter of preference, but a matter of matching the task requirements.
 
 
  On Tue, Jun 21, 2011 at 7:41 PM, Cristobal Priego 
  cristobalpri...@gmail.com mailto:cristobalpri...@gmail.com  wrote:
   guys,
 
  for the lab, whenever you configure your route groups
 
  which distribution algorithm is better
 
  circular or  top down
 
  i use top down all the time, but i'd like to know your opinion on this

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

 
   Are you a CCNP or CCIE and looking for a job? Check out
  www.PlatinumPlacement.com http://www.PlatinumPlacement.com

 
 
 
  --
  Randall da ill Saborio
   CCIE Voice Wannabe #10054675811
 
 
 


 --
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 CCIE Voice Wannabe #10054675811
 -- next part --
 An HTML attachment was scrubbed...
 URL:
 /archives/ccie_voice/attachments/20110622/650f2f6f/attachment-0001.html

 --

 Message: 2
 Date: Wed, 22 Jun 2011 13:18:18 -0400
 From: Stephen Manuel srman...@bellsouth.net

 To: ccie_voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice] UCM LOCATION QUESTION
 Message-ID: 007201cc3100$615f7a70$241e6f50$@bellsouth.net
 Content-Type: text/plain; charset=us-ascii

 Experts,



 I have a somewhat basic question just clear up something that is lingering
 in my mind.



 The example is this:



 In a multiple remote location site company, with a Centralized UCM
 architecture, each site has it's own device pool, region, and location
 settings.



 All the DN's on the phones for all sites are in the same PT, meaning calls
 between sites are on net via their existing WAN infrastructure and not the
 SIP Trunk/PSTN.



 Remote site A is rather large 50-60 phones.



 Remote site B is rather small 10-15 phones.



 Calls between regions use g729

 Calls to the PSTN using a SIP Trunk use g711



 The location setting for site A is unlimited.

 The location setting for site B is 256



 What I am wondering is this, Site B has 3 calls to the PSTN using the SIP
 Trunk using g711 @ 80k per call which equals 240k. The SIP trunk is also in
 it's own Device Pool, region, etc.



 Now someone from Site A calls site B, the bandwidth at Site A is unlimited
 so no issue there.

 According to the Cisco Research I've done, a call requires 2 streams.

 Since Site B is using 240 of 256k of bandwidth on the 3 PSTN/SIP Trunk
 calls, when a g729 calls comes inbound  which requires 24k  which

Re: [OSL | CCIE_Voice] Lab result status

2011-06-21 Thread Cristobal Priego
you will get today, wait 2.5 more hours
that's what happens when you take it on Friday

2011/6/21 Pablo Meneses pmenese...@gmail.com

 Hi experts,

 I am wondering how long it takes for you to received your score report this
 days. I've taken my lab last Friday and I have NOT received the answer yet.
 Yesterday I logged to the CCIE Online Tool and it said HOLD on the PASS/FAIL
 field, does someone know what that means? Has someone else gone through
 this?

 I will look forward to your reply as I figure most of you understand the
 feeling of not having the answer.

 Regards.

 -Pablo Meneses.

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Re: [OSL | CCIE_Voice] Lab result status

2011-06-21 Thread Cristobal Priego
i know man
i took my last attempt on a Friday as well In San Jose

i didn't get my score until next Monday at 14:45 PST

2011/6/21 Pablo Meneses pmenese...@gmail.com

 Hopefully that's true. Longest wait EVER!!!

 -Pablo Meneses.


 On Tue, Jun 21, 2011 at 1:19 PM, Cristobal Priego 
 cristobalpri...@gmail.com wrote:

 you will get today, wait 2.5 more hours
 that's what happens when you take it on Friday

 2011/6/21 Pablo Meneses pmenese...@gmail.com

 Hi experts,

 I am wondering how long it takes for you to received your score report
 this days. I've taken my lab last Friday and I have NOT received the answer
 yet. Yesterday I logged to the CCIE Online Tool and it said HOLD on the
 PASS/FAIL field, does someone know what that means? Has someone else gone
 through this?

 I will look forward to your reply as I figure most of you understand the
 feeling of not having the answer.

 Regards.

 -Pablo Meneses.

 ___
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 visit www.ipexpert.com

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 www.PlatinumPlacement.com




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Re: [OSL | CCIE_Voice] How to avoid phone firmware upgrade in the lab

2011-06-21 Thread Cristobal Priego
shut down the ports on your switch or/and ESW



2011/6/21 Roger Carpio roger.car...@gmail.com

 If the previous candidate using your lab spot changed the protocol as
 requested, let's say his lab had SIP and now yours is SCCP; there is nothing
 you can do to avoid this. It is not necessarily a waste of time; unless you
 keep staring at the phones LOL

 Regards,
 Roger Carpio.

 On Tue, Jun 21, 2011 at 2:16 PM, Pithog Oil pithog...@yahoo.com wrote:

 Hi Experts,

 what should i do so as to avoid phone firmware upgrade at the lab, i have
 discovered that some times i unknowingly have a phone firmware upgrade from
 one version to another.

 Kind assist it is usually a waste of time.

 Pithog Oil

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[OSL | CCIE_Voice] route group distribution algorithm

2011-06-21 Thread Cristobal Priego
guys,

for the lab, whenever you configure your route groups

which distribution algorithm is better

circular or  top down

i use top down all the time, but i'd like to know your opinion on this

thanks
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Re: [OSL | CCIE_Voice] (no subject)

2011-06-19 Thread Cristobal Priego
it could be a replication problem
if it says reject reset the phone quite a few times and it will register
eventually

2011/6/19 Santiago Figueroa sfigue...@mnet.com.mx

  Hello I have a doubt what is reason the ip phone 7941 do not registered
 to cucm the status reject, I am sure the MAC and phone type in the cucm is
 add correctly.

 ** **

 Thanks.

 ** **

 ** **

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Re: [OSL | CCIE_Voice] CUC issue

2011-06-19 Thread Cristobal Priego
do you have  a sip trunk integration ?
if you open up RTMT to check the vm ports
what's the forwarding extension ?

2011/6/19 John Anzar janzar1...@live.com

  Hi Guys,

 i'm doing some testing on the unity connection. i have created the users at
 the CUCM and i have intergrate CUC with the CUCM. ports registered and
 everything is fine. But i have an strange issue. From user-a when i dial
 voicemail it will ask for the pin. But for the user-b its
 giving standard message

 *Hello*: Cisco *Unity Connection Messaging System*. From *a touchtone
 phone you may dial* an extension at *any* time.

 But i have check the configuration of the both users. its 100% identical. I
 wonder what is this. Any idea ?

 Best Regards
 John

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Re: [OSL | CCIE_Voice] interesting post

2011-06-19 Thread Cristobal Priego
that's why i said it's interesting, i didn't mean anything else


2011/6/19 George Goglidze gogli...@gmail.com

 ohhh, don't get me started on this one mate...  I could say much... too
 much.

 On Sun, Jun 19, 2011 at 7:22 PM, Cristobal Priego 
 cristobalpri...@gmail.com wrote:


 https://supportforums.cisco.com/message/3380695#3380695

 what do you think ?

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 visit www.ipexpert.com

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 www.PlatinumPlacement.com



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Re: [OSL | CCIE_Voice] CUC issue

2011-06-19 Thread Cristobal Priego
do you have both users configured under the same phone system ?

2011/6/19 John Anzar janzar1...@live.com

  Hi

 yes i had a sip trunk to CUC from my previous exercise. But i have remove
 that. Under the users of the CUC i can see the extension configurations
 fine.  I'm forwarding it to the vm pilot.

 Thanks

 --
 Date: Sun, 19 Jun 2011 14:00:36 -0700
 Subject: Re: [OSL | CCIE_Voice] CUC issue
 From: cristobalpri...@gmail.com
 To: janzar1...@live.com
 CC: ccie_voice@onlinestudylist.com


 do you have  a sip trunk integration ?
 if you open up RTMT to check the vm ports
 what's the forwarding extension ?

 2011/6/19 John Anzar janzar1...@live.com

  Hi Guys,

 i'm doing some testing on the unity connection. i have created the users at
 the CUCM and i have intergrate CUC with the CUCM. ports registered and
 everything is fine. But i have an strange issue. From user-a when i dial
 voicemail it will ask for the pin. But for the user-b its
 giving standard message

 *Hello*: Cisco *Unity Connection Messaging System*. From *a touchtone
 phone you may dial* an extension at *any* time.

 But i have check the configuration of the both users. its 100% identical. I
 wonder what is this. Any idea ?

 Best Regards
 John

 ___
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 visit www.ipexpert.com

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 www.PlatinumPlacement.com



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Re: [OSL | CCIE_Voice] CUC issue

2011-06-19 Thread Cristobal Priego
yes i have the following link

http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/troubleshooting/guide/7xcuctsg010.html#wp1064561

i think that the microtraces will provide the info on your issue

2011/6/19 John Anzar janzar1...@live.com

  yeah under the same phone system. Chris this does not make any sense.

 Do u have a link for the CUC trace information. i want to check what CUC
 receive from the CUCM end.

 --
 Date: Sun, 19 Jun 2011 14:07:57 -0700

 Subject: Re: [OSL | CCIE_Voice] CUC issue
 From: cristobalpri...@gmail.com
 To: janzar1...@live.com
 CC: ccie_voice@onlinestudylist.com

 do you have both users configured under the same phone system ?

 2011/6/19 John Anzar janzar1...@live.com

  Hi

 yes i had a sip trunk to CUC from my previous exercise. But i have remove
 that. Under the users of the CUC i can see the extension configurations
 fine.  I'm forwarding it to the vm pilot.

 Thanks

 --
 Date: Sun, 19 Jun 2011 14:00:36 -0700
 Subject: Re: [OSL | CCIE_Voice] CUC issue
 From: cristobalpri...@gmail.com
 To: janzar1...@live.com
 CC: ccie_voice@onlinestudylist.com


 do you have  a sip trunk integration ?
 if you open up RTMT to check the vm ports
 what's the forwarding extension ?

 2011/6/19 John Anzar janzar1...@live.com

  Hi Guys,

 i'm doing some testing on the unity connection. i have created the users at
 the CUCM and i have intergrate CUC with the CUCM. ports registered and
 everything is fine. But i have an strange issue. From user-a when i dial
 voicemail it will ask for the pin. But for the user-b its
 giving standard message

 *Hello*: Cisco *Unity Connection Messaging System*. From *a touchtone
 phone you may dial* an extension at *any* time.

 But i have check the configuration of the both users. its 100% identical. I
 wonder what is this. Any idea ?

 Best Regards
 John

 ___
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 visit www.ipexpert.com

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 www.PlatinumPlacement.com




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Re: [OSL | CCIE_Voice] interesting post

2011-06-19 Thread Cristobal Priego
twice in San Jose
going for my third in San Jose

2011/6/19 Vega Wong vega2...@yahoo.com.au

 hi guys

 Where did you attempt your lab exam?

 Cheers

 --- On *Mon, 20/6/11, Cristobal Priego cristobalpri...@gmail.com* wrote:


 From: Cristobal Priego cristobalpri...@gmail.com
 Subject: Re: [OSL | CCIE_Voice] interesting post
 To: George Goglidze gogli...@gmail.com
 Cc: ccie_voice@onlinestudylist.com
 Received: Monday, 20 June, 2011, 7:01 AM


 that's why i said it's interesting, i didn't mean anything else


 2011/6/19 George Goglidze 
 gogli...@gmail.comhttp://mc/compose?to=gogli...@gmail.com
 

 ohhh, don't get me started on this one mate...  I could say much... too
 much.

 On Sun, Jun 19, 2011 at 7:22 PM, Cristobal Priego 
 cristobalpri...@gmail.com http://mc/compose?to=cristobalpri...@gmail.com
  wrote:


 https://supportforums.cisco.com/message/3380695#3380695

 what do you think ?

 ___
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 visit www.ipexpert.com

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 www.PlatinumPlacement.com




 -Inline Attachment Follows-


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Re: [OSL | CCIE_Voice] CUPC - hit and miss

2011-06-18 Thread Cristobal Priego
i had run into similar problems
and what i had to do to resolve the problem is
reset the sip trunk in CUCM



2011/6/15 Adil Shaikh adil.sha...@gmail.com

 did i write hit and miss? sorry, it shoudl be read as hit or miss.


 On Thu, Jun 16, 2011 at 2:11 PM, Adil Shaikh adil.sha...@gmail.comwrote:

 Hi All,

 Is there any gottcha for CUPC to see the presence of another phone when
 it goes off hook?

 I have done CUPS integration almost 6-7 times. Sometime the presence of
 another phone going off hook is displayed on CUPC and sometime not.
 If the CUPS integration is done today, then presence of contact in
 CUPC will be working perfectly fine for this integration.
 if tomorrow, i follow the same procedure for new integration for new lab,
 it will not work whether i reboot CUPS or not.

 The contact in CUPC is added from User Web Interface on CUPS.

 Has anyone come across similar situation?

 Another issue with CUPC is voicemail. Sometime to refresh the
 indication of arrival of new voicemail on CUPC, i need to log out of CUPC
 and log back in. Sometime the referesh of new voicemail will happen without
 logout/login to CUPC. Voicemail integration seems to be working for every
 CUPC integration I have done. There is no hit and miss like contact presence
 raised above.


 Thanks
 -adil

 --
   .. . .
 _7___|___|_|_|adil.sha...@gmail.com

 . .





 --
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 _7___|___|_|_|adil.sha...@gmail.com

 . .



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Re: [OSL | CCIE_Voice] SRST - Phones should work same as it isregistered with CUCM

2011-06-18 Thread Cristobal Priego
for calling name you will need to use
cme in srst mode
and use either auto provision all or dn
and then force your failover so the router will create the ephones and
ephone-dn configs
and then you will have to edit the ephone-dn to get the calling name



2011/6/15 Randall Crumm randall.cr...@flextronics.com

 ** **

 I would add softkeys as well..

 ** **

 Randall

 ** **

 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Greg
 *Sent:* Wednesday, June 15, 2011 8:53 AM
 *To:* ccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] SRST - Phones should work same as it
 isregistered with CUCM

 ** **

 Can anyone give suggestion on how to meet the requirement of Phones should
 work same as it is registered with CUCM?

  

 It's a very broad topic since there are so many features on CUCM.

  

 Here's what I can think of:

  

 1) Basic call routing (phone to phone, intra and inter sites, PSTN calls)
 both outgoing and incoming.

 2) Secondary dial tone

 3) Voicemail (voicemail button, voicemail pilot number, CFwd Busy, CFwd No
 answer)

 4) multi-channel support, hold, conference.

 5) Do we need to care about corporate directory?  How about calling names
 in SRST mode?

  

 Thanks!
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Re: [OSL | CCIE_Voice] NTP summer time

2011-06-18 Thread Cristobal Priego
personally
i do
EST, EDT
or
PST, PDT



2011/6/18 Cristobal Priego cristobalpri...@gmail.com

 i think is just a label,

 EST, EDT or it could be whatever you want

 2011/6/18 Adil Shaikh adil.sha...@gmail.com

 Hi All,

 Question regarding NTP summer time.

 I configure summertime as per IPexpert material:

 clock timezone EST -8
 clock summer EST recurr

 However, many places i have seen this to be configured as:

 clock timezone EST -8
 clock summer EDT recurr


 Which one shoud be done in exam? Any clue?

 thanks
 -adil

 --
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 _7___|___|_|_|adil.sha...@gmail.com

 . .



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Re: [OSL | CCIE_Voice] NTP Concept

2011-06-17 Thread Cristobal Priego
if they don't tell you to configure NTP on Branch 1 is up to you
if they're saying to source it from the loopback then issue the command
ntp source loopback0 if they don't tell you anything is up to you

you can use the CUE as an NTP, indeed is the default
you can also use the HQ-RTR as an NTP source, in that case the ntp master
command will be needed

2011/6/17 sandeep.ra...@wipro.com

  Hi ,

 If  the question says HQ-RTR should be synchronised with external NTP
 server.

 CUCM should be synchronised with HQ-RTR's Loopback0 or VLAN interface.

 My config on HQ RTR looks like :



 ntp server x.x.x.x external ntp server

 ntp source loopback0

 clock timezone 

 clock summertime  recurring



 For BR1 RTR there is no mention in the question about wherei should
 synchronise the NTP, should i synchronise it with external NTP?



 Lets assume for BR1 i configure NTP as HQ RTR's Loopback 0 interface.Should
 i configure ntp source loopback 0 ?



 For BR 2 RTR there is no mention in the question about where i should
 synchronise the NTP, should i synchronise it with external NTP or can i
 synchronise it with the HQ RTR loopback IP?



 Need For NTP master on HQ-RTR is as below:



 If question says i need to configure NTP as HQ RTR loopback for CUE



 When i configure HQ-RTR (loopback 0) as the NTP server for CUE module, i
 would have to configure HQ RTR as an NTP master right?



 Can i configure CUE NTP as CME IP address, if yes in what instances can i
 configure it?





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Re: [OSL | CCIE_Voice] CCS

2011-06-17 Thread Cristobal Priego
i don't think that a command on the CLI will give you that info
you can get that info from the GUI interface

2011/6/17 Santiago Figueroa sfigue...@mnet.com.mx

  Hello all

 ** **

 Canyone tell me if exist any command from CLI in CUCM to shown me that
 include within a Calling Search space? What Partitions are in that CCS?***
 *

 ** **

 ** **

 ** **

 ** **

 Thanks.

 ** **

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Re: [OSL | CCIE_Voice] UCX

2011-06-17 Thread Cristobal Priego
i don't think you should have a problem but you can confirm with the
compatibility matrix

2011/6/17 Santiago Figueroa sfigue...@mnet.com.mx

  Hello I have doubt CUCM 7.1.3 can work  Unity Connection 7.1.5 ?are
 compatible

 ** **

 ** **

 Thanks.

 ** **

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Re: [OSL | CCIE_Voice] ccme

2011-06-16 Thread Cristobal Priego

Spa phones aré only supported on uc500

Enviado desde mi iPhone

El Jun 16, 2011, a las 13:18, Santiago Figueroa  
sfigue...@mnet.com.mx escribió:


Hello anyone know if SPA502G work CCME? can anyone send  
configuration example for that?






Thanks


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CONFIDENTIAL and only for the EXCLUSIVE USE of the receivers.  
Circulation and/or copy without permission is not allowed. If you  
have received this e-mail and you are not the intended recipient,  
please let us know and erase the message and attached files. Thank  
you.

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Re: [OSL | CCIE_Voice] CUPS Presence only showing Devices from BR1 and not HQ / Disconnecting (pending retry) under Server health

2011-06-14 Thread Cristobal Priego
do you have your access list set on Presence or your digest credentials on
that particular user ?

2011/6/14 Randall Crumm rrcr...@yahoo.com

 Have you checked the owner of the phone?
 rc

 --
 *From:* Alan Gardner agard...@ctclc.com
 *To:* ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 *Sent:* Tuesday, June 14, 2011 8:12 AM
 *Subject:* [OSL | CCIE_Voice] CUPS Presence only showing Devices from BR1
 and not HQ / Disconnecting (pending retry) under Server health

 Done the integration and if I go to User management and end users all of my
 BR1 show devices, but all my HQ phone do not show devices.

 I have checked my line to user association all good.

 I have checked my End user in cucm- all good. I have compared all of them
 and cannot figure out what could be different.

 Any ideas.


 [image: Description: cid:image001.png@01CC2A79.DC0E35F0]

 Also getting  Disconnecting (pending retry) under Server health


 Best Regards,

 Alan Gardner
 Chief Executive Officer
 Current Technologies Computer Learning Center
 PH. 219.764.3800
 agard...@ctclc.com
 MCT, MCSE (Messaging  Security), MCSD,
 MSF Trainer, MCNI, MCNE, MCSA, CCNA, , ITIL
 CCNP, CCVP, CCIP, CCDP, CCIE # 22758, CCSI
 A+, MCDST, Network+, Server+, IT Project+,
 EBIZ, CTT+, MCDBA CCNT, CISSP, CWNA, CTP, CIW
 Microsoft Server 2008 Enterprise Administrator



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Re: [OSL | CCIE_Voice] I just failed the LAB 5

2011-06-08 Thread Cristobal Priego
right on !
good luck !

2011/6/8 adam compton com...@gmail.com

 I guess I'm a glutton for punishment.  Scheduled for July 21st in RTP


 On Tue, Jun 7, 2011 at 10:50 AM, adam compton com...@gmail.com wrote:


 Just failed the LAB 5 everybody has been talking about.  I felt very
 prepared, but not for that test.  Alot of things that IPExpert has never
 touched.  That sucks cuz I was trying to use them solely.  I feel like I'm
 going to have to throw down more money for a different study guide...

 Anywho,  I was wondering what the chances are to get the same lab twice.
 If you take a lab, is it pretty much disqualified from you repeating it?

 Adam Compton



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Re: [OSL | CCIE_Voice] Ringlist.xml and List.xml

2011-06-08 Thread Cristobal Priego
what if you have the subscriber working as a TFTP as well ?

2011/6/8 Randall Saborio ill2...@gmail.com

 I still don't follow you.

 If your DHCP has Publisher as TFTP, then why would you change the RingList
 on some other tftp server instead of on Publisher.


 On Wed, Jun 8, 2011 at 3:18 PM, Adil Shaikh adil.sha...@gmail.com wrote:

 Hi Randall,

 IF we do not do step 5 then the phone in question will continue to point
 to old TFTP server which does not have a modified Ringlist.xml file.
 If the question is to delete a particular ringtone from Phone7 at Site Z.
 Then you need to direct Phone7 at Site Z to a TFTP server which has the
 modified Ringlist.xml file.

 Instead of manually changing TFTP address on phone, you can assign
 different TFTP using DHCP server for a particualr MAC address.
 -adil
 On Thu, Jun 9, 2011 at 12:09 AM, Randall Saborio ill2...@gmail.comwrote:

 I don't understand why step 5. Dont you have the phones pointing to that
 server already?

 If you talk about CME phones, then the ringlist must be uploaded to CME,
 not CUCM.

 On Wed, Jun 8, 2011 at 2:16 AM, Adil Shaikh adil.sha...@gmail.comwrote:

 Hi All,

 There are two simple questions (easy score points) raised in some ccie
 voice forums. They are add Background Image (create List.xml) on a phone 
 and
 Removal a particular Ringtone (edit Ringlist.xml) from a phone.

 For BGimage 7965 admin guide shows the clear example of List.xml file.

 However, for the removal of particular ringtone from Ringlist.xml, i am
 practicing following method. If anyone know another method then could you
 please let me know.

 1) ssh to PUB and issue command file view tftp Ringlist.xml
 2) Copy 110 lines to notepad and remove appropriate tag lines and save
 it on desktop as Ringlist.xml
 3) Go to CM OS admin page and upload Ringlist.xml in / directory of
 non-tftp running  CM.
 4) Start  TFTP service on that CM
 5) Change TFTP address on the phone to point to a above CM.  This
 is manually changing TFTP address which was originally assigned from DHCP
 server.


 Thanks
 -adil


 --
   .. . .
 _7___|___|_|_|adil.sha...@gmail.com

 . .



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 please visit www.ipexpert.com

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 --
 Randall da ill Saborio
 CCIE Voice Wannabe #10054675811




 --
   .. . .
 _7___|___|_|_|adil.sha...@gmail.com

 . .





 --
 Randall da ill Saborio
 CCIE Voice Wannabe #10054675811


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Re: [OSL | CCIE_Voice] Unity connection : USER ID should not be asked

2011-06-07 Thread Cristobal Priego
go to advanced, conversation
unckeck the option Request Entry of User ID after Failed Password Entry from
Known Extension

2011/6/7 Julien Krieger krieger.jul...@gmail.com

 Hi Rahul,

 There is an option to uncheck in advanced config menu. Recently, someone,
 posted on thé same subject...

 Julien

 Le 7 juin 2011 à 20:05, Rahul Kapor rahul.kapo...@gmail.com a écrit :

  Hi All,

 trying to achieve following things

 when user is hitting voice mail and entering wrong password , user is
 prompt to enter  USER ID but user  should be prompted for PIN again not for
 USER ID !

 how to achieve  this .

 thx,
 Rahul
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Re: [OSL | CCIE_Voice] Port number document location

2011-06-06 Thread Cristobal Priego
very cool
thank you George

2011/6/6 Rrcrumm rrcr...@yahoo.com

 Awesome George
 Thx
 Sent from my iPhone

 On Jun 6, 2011, at 2:11 AM, George Goglidze gogli...@gmail.com wrote:

 Hi Adam,

 No need to remember all ports by memory at all...

 You have all tools available in IOS to lookup ports:
 h323 signalling:
 VOICEGW#show ip port-map h323
 Default mapping:  h323 tcp port 1720
 system defined

 h323 RAS:
 VOICEGW#show ip port-map h225ras
 Default mapping:  h225ras  udp port 1719
 system defined

 h323 GK mcast discovery:
 VOICEGW#show ip port-map h225rasmcast
 Default mapping:  h225rasMcast udp port 1718
 system defined

 SCCP:
 VOICEGW#show ip port-map skinny
 Default mapping:  skinny   tcp port 2000
 system defined

 MGCP:
 VOICEGW#show ip port-map mgcp
 Default mapping:  mgcp udp port 2427
 system defined

 MGCP backhaul:
 VOICEGW#show ccm-manager backhaul
 Backhaul Link info:
 Link Protocol:  TCP
 Remote Port Number: 2428

 SIP:
 VOICEGW#show ip port-map sip
 Default mapping:  sip  udp port 5060
 system defined
 Default mapping:  sip  tcp port 5060
 system defined


 This is quicker than going into documentation to find ports...

 Hope this helps,


 On Mon, Jun 6, 2011 at 12:21 AM, adam compton  com...@gmail.com
 com...@gmail.com wrote:

 There is a guide on cisco's website that lists all the ports you will need
 to know for the test.  RTP, MGCP, CTI, etc.  Here is the link


 http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/port/7_0/CCM_7.0PortList.pdf
 http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/port/7_0/CCM_7.0PortList.pdf

 If you have to navigate to it (wink wink poke poke) follow this path:

 Voice and Unified Communications  IP Telephony  Call Control  Cisco
 Unified Communications Manager (CallManager)  Maintain and Operate 
 Maintain and Operate Guides  Cisco Unified Communications Manager 7.0 TCP
 and UDP Port Usage

 Regards,
 Adam Compton

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Re: [OSL | CCIE_Voice] DHCP Debug in CUCM

2011-06-06 Thread Cristobal Priego
thanks guys

2011/6/3 Miron Kobelski findko...@gmail.com

 It's feasible as you don't need to gather the trace file - you get the same
 output in CLI as with debug udp on the router (packets received and sent on
 BOOTP/BOOTC udp ports), but my method is indeed more time consuming.

 I need to check the other method with syslog messages. Thanks.


 On Fri, Jun 3, 2011 at 19:01, Adam Frankel (afrankel) 
 afran...@cisco.comwrote:

  Packet capture on CUCM CLI:
 https://supportforums.cisco.com/docs/DOC-11599

 Not really feasible for lab purposes IMO.

 You can also see the DHCP activity in the UCM syslog:

  file tail activelog syslog/messages

 HTH,

 Adam Frankel
 Customer Support Engineer
 Unified Communications Infrastructure
 +1,919-392-7936 9am-5pm EDT (GMT -4)
 CCVP, RHCE, CUP Specialist
 Onshift Weekend Office Hours: 8am-1:30pm EDT



 Original Message--
 From: Miron Kobelski findko...@gmail.com findko...@gmail.com
 Sent: Fri, Jun 03, 2011 11:51:03 Am
 To: Friderich Claude cfrider...@netcore.lu cfrider...@netcore.lu
 CC: CCIE VOICE 
 ccie_voice@onlinestudylist.comccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] DHCP Debug in CUCM

 Much better solution, thanks! :)

 On Fri, Jun 3, 2011 at 11:52, Friderich Claude cfrider...@netcore.luwrote:

  Something very fast and easy ;-)

 Debug ip udp on the hq router or br1 router depending for which site
 dhcp pub is asking for …..







 [image: netcore_signature_4.jpg]





 *Claude FRIDERICH *

 PreSales Voice over IP

 * ***

 *[image: CCIEVoice_UseLogo]*

  CCIE Voice #29035

 *O*ffice**

 +352 31 33 80 – 407**

 *M*obile **

 +352 621 303 616**

 [image: cid:image003.jpg@01CC1F87.400817E0]

 *F*ax  **

 +352 31 33 80 412 %2B352%2031%2033%2080%20412

 *E*mail**

 *cfrider...@netcore.lu x...@netcore.lu***





 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Miron Kobelski
 *Sent:* vendredi 3 juin 2011 08:03
 *To:* Cristobal Priego
 *Cc:* CCIE VOICE
 *Subject:* Re: [OSL | CCIE_Voice] DHCP Debug in CUCM



 Utils network capture

 On Jun 3, 2011 1:55 AM, Cristobal Priego cristobalpri...@gmail.com
 wrote:
  how do you do the packet capture from the cucm cli ?
 
  2011/6/2 Miron Kobelski findko...@gmail.com
 
  Hi,
 
  I was unable to find anything better then packet capture on CUCM CLI -
 it
  allows you to see if the DHCP messages hit the CUCM and are sent back,
 but
  not much more ;)
 
  best regards
 
  On Thu, Jun 2, 2011 at 16:45, John Anzar janzar1...@live.com wrote:
 
 
  Hi Guys
 
  I have a very basic question about DHCP. most of the time this works
 with
  out any issue.
 
  Is there anyway that we can DHCP debug in CUCM ?? Can we get the mac
 biind
  with the IP form CUCM ?
 
  Best Regards
  J anzar
 
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 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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 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] Meet-me

2011-06-06 Thread Cristobal Priego
no it cant.
it has to be initiated from a phone

ip session (ipcelerate) can do the oposite. they call it reverse meet me


2011/6/6 Santiago Figueroa sfigue...@mnet.com.mx

  Hello I need to know if the feature Meet-me can initiate for first call
 from PSTN Gateway instead of cisco ip phone, when Gw have a DID to number
 meet-me.









 Thanks.







 --
 La información incluida en este mensaje y sus anexos es CONFIDENCIAL y para
 USO EXCLUSIVO de sus destinatarios. No está permitida su divulgación y/o
 reproducción sin autorización. Si ha recibido este mensaje y no le incumbe,
 le rogamos nos los comunique y proceda a su borrado. Gracias.

 Information included in this e-mail and attached files is CONFIDENTIAL and
 only for the EXCLUSIVE USE of the receivers. Circulation and/or copy without
 permission is not allowed. If you have received this e-mail and you are not
 the intended recipient, please let us know and erase the message and
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Re: [OSL | CCIE_Voice] Extension Mobility and Local Route Group

2011-06-06 Thread Cristobal Priego
if you have local route groups
when the user logs in to a phone using EM it will use the LRG of the DP that
yo associated to the device. because on the device profile you can't assign
a device pool.

you can indeed use the line device approach

where you allow on the device calling search space and block on the line
and if you share the same dialplan between sites you shouldn't have any
problems

i have used EM and LRG on real life for a few customers and it works pretty
well

2011/6/6 Qq Ccie cci...@yahoo.com

 I though the CSS on device pool mean for Device Mobility only, not for
 Extension Mobility?


 --
 *From:* Brian btmulg...@gmail.com

 *To:* Qq Ccie 
 ccieqqhttps://mail.google.com/mail/?hl=enshva=1#inbox/13060d89e4fd6644if
 y...@yahoo.com cci...@yahoo.com
 *Cc:* ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 *Sent:* Mon, June 6, 2011 1:06:09 PM

 *Subject:* Re: [OSL | CCIE_Voice] Extension Mobility and Local Route Group

 hi - re EM call routing path: we can use device mobility to flip to the
 roaming device pool to support this.

 hth.
 Brian


 Sent from my iPad

 On 6 Jun 2011, at 15:05, Qq Ccie cci...@yahoo.com wrote:


 Thanks Kiwi.  On a side note, calling number transformation has to be
 handled at RL level in order to different calling number requirements
 (local, ld and intl)?

 Also, two RPs (\+1919.XXX placed in two different partitions) have to
 be configured  if Site B has a requirement of TEHO for 919 area code?


 *From:* Ki Wi kiwi.vo...@gmail.com
 *To:* Qq Ccie cci...@yahoo.com
 *Cc:* ccie_voice@onlinestudylist.comccie_voice@onlinestudylist.com
 *Sent:* Sun, June 5, 2011 11:39:36 PM
 *Subject:* Re: [OSL | CCIE_Voice] Extension Mobility and Local Route Group

 In that case, you have a great time of converting everything to e164. =P

 Use called party transformation to strip them into the correct digits at
 the gateway level.

 Example : Site A ( local area code 919) --- calls to local number is
 8123123

 Converted the dialed number to +19198123123

 Use a route pattern +.! and send to local route group


 At each gateway, use called party transformation.
 Match  +1.919! , called transformation mask XXX for Site A gateway

 At Site B, you will strip predot for the other called party transformation.




 On Mon, Jun 6, 2011 at 9:54 AM, Qq Ccie  cci...@yahoo.com
 cci...@yahoo.com wrote:


 Thanks Ahmed for your help.  Yes, I agree that CSS has to be at the line
 level.

 However, the assumption is that HQ and BR1 have configured call routing
 via Standard Local Route Group.  When logged in to EM service, the user
 would not be able to use his original local gateway because the current
 local gateway is based on Device Pool settings.  As a result, his local/long
 distance call might not be correct.  Any suggestion?

 Thanks,

 --
 *From:* Ahmed Ellboudy  ahmed_ellbo...@rayacorp.com
 ahmed_ellbo...@rayacorp.com
 *To:* Qq Ccie  cci...@yahoo.comcci...@yahoo.com;
 ccie_voice@onlinestudylist.comccie_voice@onlinestudylist.com
 *Sent:* Sun, June 5, 2011 1:53:46 PM
 *Subject:* RE: [OSL | CCIE_Voice] Extension Mobility and Local Route
 Group

  Hi,
 I think you need to use Line device approach rule which will solve it by
 open every calls types local/national and international and makes the EM to
 limit your access,refer to the SRND in this point


  Thanks  Regards;
  Ahmed Mahmoud Ellboudy
 --
 *From:* ccie_voice-boun...@onlinestudylist.com
 ccie_voice-boun...@onlinestudylist.com on behalf of Qq Ccie
 *Sent:* Sun 6/5/2011 6:31 PM
 *To:* ccie_voice@onlinestudylist.comccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] Extension Mobility and Local Route Group


  Anyone has any thoughts of the interaction between Extension Mobility
 and Local Route Group?  Assuming HQ and BR1 have configured call routing via
 Local Route Group.  Any suggestions?

 Disclaimer: NOTICE The information contained in this message is
 confidential and is intended for the addressee(s) only. If you have received
 this message in error or there are any problems please notify the originator
 immediately. The unauthorized use, disclosure, copying or alteration of this
 message is strictly forbidden. Raya will not be liable for direct, special,
 indirect or consequential damages arising from alteration of the contents of
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 being passed on. Views expressed in this communication are not necessarily
 those of Raya.If you have received this message in error, please notify the
 sender immediately by email, facsimile or telephone and return and/or
 destroy the original message.

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 For more information regarding industry leading CCIE Lab training, please
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 Are you a CCNP or CCIE and looking for a job? Check out
 

Re: [OSL | CCIE_Voice] Extension Mobility and Local Route Group

2011-06-06 Thread Cristobal Priego
if you have your calling/called party transformation you can also use
globalized dialing

when you you log in with EM the only change on the phone is going to be the
settings assigned to the line, those settings will be overwritten by the
device profile.


2011/6/6 Qq Ccie cci...@yahoo.com

 Thanks Cristobal.  So Globalized dialing is a prerequisite as well?

 --
 *From:* Cristobal Priego cristobalpri...@gmail.com

 *To:* Qq Ccie cci...@yahoo.com
 *Cc:* Brian btmulg...@gmail.com; ccie_voice@onlinestudylist.com 
 ccie_voice@onlinestudylist.com
 *Sent:* Mon, June 6, 2011 3:00:42 PM

 *Subject:* Re: [OSL | CCIE_Voice] Extension Mobility and Local Route Group

 if you have local route groups
 when the user logs in to a phone using EM it will use the LRG of the DP
 that yo associated to the device. because on the device profile you can't
 assign a device pool.

 you can indeed use the line device approach

 where you allow on the device calling search space and block on the line
 and if you share the same dialplan between sites you shouldn't have any
 problems

 i have used EM and LRG on real life for a few customers and it works pretty
 well

 2011/6/6 Qq Ccie cci...@yahoo.com

 I though the CSS on device pool mean for Device Mobility only, not for
 Extension Mobility?


 --
 *From:* Brian btmulg...@gmail.com

 *To:* Qq Ccie 
 ccieqqhttps://mail.google.com/mail/?hl=enshva=1#inbox/13060d89e4fd6644if
 y...@yahoo.com cci...@yahoo.com
 *Cc:* ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 *Sent:* Mon, June 6, 2011 1:06:09 PM

 *Subject:* Re: [OSL | CCIE_Voice] Extension Mobility and Local Route
 Group

 hi - re EM call routing path: we can use device mobility to flip to the
 roaming device pool to support this.

 hth.
 Brian


 Sent from my iPad

 On 6 Jun 2011, at 15:05, Qq Ccie cci...@yahoo.com wrote:


 Thanks Kiwi.  On a side note, calling number transformation has to be
 handled at RL level in order to different calling number requirements
 (local, ld and intl)?

 Also, two RPs (\+1919.XXX placed in two different partitions) have to
 be configured  if Site B has a requirement of TEHO for 919 area code?


 *From:* Ki Wi kiwi.vo...@gmail.com
 *To:* Qq Ccie cci...@yahoo.com
 *Cc:* ccie_voice@onlinestudylist.comccie_voice@onlinestudylist.com
 *Sent:* Sun, June 5, 2011 11:39:36 PM
 *Subject:* Re: [OSL | CCIE_Voice] Extension Mobility and Local Route
 Group

 In that case, you have a great time of converting everything to e164. =P

 Use called party transformation to strip them into the correct digits at
 the gateway level.

 Example : Site A ( local area code 919) --- calls to local number is
 8123123

 Converted the dialed number to +19198123123

 Use a route pattern +.! and send to local route group


 At each gateway, use called party transformation.
 Match  +1.919! , called transformation mask XXX for Site A gateway

 At Site B, you will strip predot for the other called party
 transformation.



 On Mon, Jun 6, 2011 at 9:54 AM, Qq Ccie  cci...@yahoo.com
 cci...@yahoo.com wrote:


 Thanks Ahmed for your help.  Yes, I agree that CSS has to be at the line
 level.

 However, the assumption is that HQ and BR1 have configured call routing
 via Standard Local Route Group.  When logged in to EM service, the user
 would not be able to use his original local gateway because the current
 local gateway is based on Device Pool settings.  As a result, his local/long
 distance call might not be correct.  Any suggestion?

 Thanks,

 --
 *From:* Ahmed Ellboudy  ahmed_ellbo...@rayacorp.com
 ahmed_ellbo...@rayacorp.com
 *To:* Qq Ccie  cci...@yahoo.comcci...@yahoo.com;
 ccie_voice@onlinestudylist.comccie_voice@onlinestudylist.com
 *Sent:* Sun, June 5, 2011 1:53:46 PM
 *Subject:* RE: [OSL | CCIE_Voice] Extension Mobility and Local Route
 Group

  Hi,
 I think you need to use Line device approach rule which will solve it by
 open every calls types local/national and international and makes the EM to
 limit your access,refer to the SRND in this point


  Thanks  Regards;
  Ahmed Mahmoud Ellboudy
 --
 *From:* ccie_voice-boun...@onlinestudylist.com
 ccie_voice-boun...@onlinestudylist.com on behalf of Qq Ccie
 *Sent:* Sun 6/5/2011 6:31 PM
 *To:* ccie_voice@onlinestudylist.comccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] Extension Mobility and Local Route Group


  Anyone has any thoughts of the interaction between Extension Mobility
 and Local Route Group?  Assuming HQ and BR1 have configured call routing via
 Local Route Group.  Any suggestions?

 Disclaimer: NOTICE The information contained in this message is
 confidential and is intended for the addressee(s) only. If you have received
 this message in error or there are any problems please notify the originator
 immediately. The unauthorized use, disclosure, copying or alteration of this
 message is strictly forbidden. Raya

Re: [OSL | CCIE_Voice] Extension Mobility and Local Route Group

2011-06-06 Thread Cristobal Priego
you're not roaming when you use extension mobility because the DP never
changes

if we need to localize the dialing for the user, then i will use the  CSS of
the device profile.




2011/6/6 Brian Mulgrew btmulg...@gmail.com

 Guys - just on this topic; if we had a question which required extension
 mobility be configured on an HQ phone (with no mention of roaming between
 sites) would we still be expected to update the dial plan to support HQ
 localised dialling from a BR1 phone?

 any thoughts appreciated.

 thks
 On Mon, Jun 6, 2011 at 8:30 PM, Cristobal Priego 
 cristobalpri...@gmail.com wrote:

 if you have your calling/called party transformation you can also use
 globalized dialing

 when you you log in with EM the only change on the phone is going to be
 the settings assigned to the line, those settings will be overwritten by the
 device profile.



 2011/6/6 Qq Ccie cci...@yahoo.com

 Thanks Cristobal.  So Globalized dialing is a prerequisite as well?

 --
 *From:* Cristobal Priego cristobalpri...@gmail.com

 *To:* Qq Ccie cci...@yahoo.com
 *Cc:* Brian btmulg...@gmail.com; ccie_voice@onlinestudylist.com 
 ccie_voice@onlinestudylist.com
 *Sent:* Mon, June 6, 2011 3:00:42 PM

 *Subject:* Re: [OSL | CCIE_Voice] Extension Mobility and Local Route
 Group

 if you have local route groups
 when the user logs in to a phone using EM it will use the LRG of the DP
 that yo associated to the device. because on the device profile you can't
 assign a device pool.

 you can indeed use the line device approach

 where you allow on the device calling search space and block on the line
 and if you share the same dialplan between sites you shouldn't have any
 problems

 i have used EM and LRG on real life for a few customers and it works
 pretty well

 2011/6/6 Qq Ccie cci...@yahoo.com

 I though the CSS on device pool mean for Device Mobility only, not for
 Extension Mobility?


 --
 *From:* Brian btmulg...@gmail.com

 *To:* Qq Ccie 
 ccieqqhttps://mail.google.com/mail/?hl=enshva=1#inbox/13060d89e4fd6644if
 y...@yahoo.com cci...@yahoo.com
  *Cc:* ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 
 *Sent:* Mon, June 6, 2011 1:06:09 PM

 *Subject:* Re: [OSL | CCIE_Voice] Extension Mobility and Local Route
 Group

 hi - re EM call routing path: we can use device mobility to flip to the
 roaming device pool to support this.

 hth.
 Brian


 Sent from my iPad

 On 6 Jun 2011, at 15:05, Qq Ccie cci...@yahoo.com wrote:


 Thanks Kiwi.  On a side note, calling number transformation has to be
 handled at RL level in order to different calling number requirements
 (local, ld and intl)?

 Also, two RPs (\+1919.XXX placed in two different partitions) have
 to be configured  if Site B has a requirement of TEHO for 919 area code?


 *From:* Ki Wi kiwi.vo...@gmail.com
 *To:* Qq Ccie cci...@yahoo.com
 *Cc:* ccie_voice@onlinestudylist.comccie_voice@onlinestudylist.com
 *Sent:* Sun, June 5, 2011 11:39:36 PM
 *Subject:* Re: [OSL | CCIE_Voice] Extension Mobility and Local Route
 Group

 In that case, you have a great time of converting everything to e164. =P

 Use called party transformation to strip them into the correct digits at
 the gateway level.

 Example : Site A ( local area code 919) --- calls to local number is
 8123123

 Converted the dialed number to +19198123123

 Use a route pattern +.! and send to local route group


 At each gateway, use called party transformation.
 Match  +1.919! , called transformation mask XXX for Site A gateway

 At Site B, you will strip predot for the other called party
 transformation.



 On Mon, Jun 6, 2011 at 9:54 AM, Qq Ccie  cci...@yahoo.com
 cci...@yahoo.com wrote:


 Thanks Ahmed for your help.  Yes, I agree that CSS has to be at the
 line level.

 However, the assumption is that HQ and BR1 have configured call routing
 via Standard Local Route Group.  When logged in to EM service, the user
 would not be able to use his original local gateway because the current
 local gateway is based on Device Pool settings.  As a result, his 
 local/long
 distance call might not be correct.  Any suggestion?

 Thanks,

 --
 *From:* Ahmed Ellboudy  ahmed_ellbo...@rayacorp.com
 ahmed_ellbo...@rayacorp.com
 *To:* Qq Ccie  cci...@yahoo.comcci...@yahoo.com;
 ccie_voice@onlinestudylist.comccie_voice@onlinestudylist.com
 *Sent:* Sun, June 5, 2011 1:53:46 PM
 *Subject:* RE: [OSL | CCIE_Voice] Extension Mobility and Local Route
 Group

  Hi,
 I think you need to use Line device approach rule which will solve it
 by open every calls types local/national and international and makes the 
 EM
 to limit your access,refer to the SRND in this point


  Thanks  Regards;
  Ahmed Mahmoud Ellboudy
 --
 *From:* ccie_voice-boun...@onlinestudylist.com
 ccie_voice-boun...@onlinestudylist.com on behalf of Qq Ccie
 *Sent:* Sun 6/5/2011 6:31 PM
 *To:* ccie_voice@onlinestudylist.comccie_voice

Re: [OSL | CCIE_Voice] TCL IVR

2011-06-06 Thread Cristobal Priego
you can use B-ACD

2011/6/6 Santiago Figueroa sfigue...@mnet.com.mx

  Hello I have doubt is possible to have a TCL IVR work in Gateway but if
 the flash it fail then to send calls to IVR in Unity in automatic?



 Thnaks

 --
 La información incluida en este mensaje y sus anexos es CONFIDENCIAL y para
 USO EXCLUSIVO de sus destinatarios. No está permitida su divulgación y/o
 reproducción sin autorización. Si ha recibido este mensaje y no le incumbe,
 le rogamos nos los comunique y proceda a su borrado. Gracias.

 Information included in this e-mail and attached files is CONFIDENTIAL and
 only for the EXCLUSIVE USE of the receivers. Circulation and/or copy without
 permission is not allowed. If you have received this e-mail and you are not
 the intended recipient, please let us know and erase the message and
 attached files. Thank you.

 ___
 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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www.ipexpert.com

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

[OSL | CCIE_Voice] study partners

2011-06-06 Thread Cristobal Priego
hello guys

I took my 2nd attempt lab attempt on Friday and i just got my score
i failed with 77%
i'm a bit disappointed but willing to give it a try 1 more time
i'm located in california
i'm willing to give it a try in july the week of the 18th i need to check
the dates i think that's the closest one

please let me know if you're interested
___
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www.ipexpert.com

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Re: [OSL | CCIE_Voice] DHCP Debug in CUCM

2011-06-02 Thread Cristobal Priego
how do you do the packet capture from the cucm cli ?

2011/6/2 Miron Kobelski findko...@gmail.com

 Hi,

 I was unable to find anything better then packet capture on CUCM CLI - it
 allows you to see if the DHCP messages hit the CUCM and are sent back, but
 not much more  ;)

 best regards

 On Thu, Jun 2, 2011 at 16:45, John Anzar janzar1...@live.com wrote:


 Hi Guys

 I have a very basic question about DHCP. most of the time this works with
 out any issue.

 Is there anyway that we can DHCP debug in CUCM ?? Can we get the mac biind
 with the IP form CUCM ?

 Best Regards
 J anzar

 ___
 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] drop packets from 3750 output queue 2

2011-06-02 Thread Cristobal Priego
I agree with Bartosz and Steve's answers
i'd check the threshold and associate COS 4 to Queue 2 on a either threshold
1 or 2 and then modify the threshold to be 60% or whatever they're asking

2011/6/2 Steve Denney (stdenney) stden...@cisco.com

 CoS 4,6,7 are probably mapped to queue 2 threshold 2. Just move CoS 4 to
 queue 2 threshold **1**, then make sure the threshold settings for queue 2
 are 60 100 100 100 (the last 3 values aren’t critical, you just need the 60
 so that CoS 4 packets are dropped starting at threshold 1 – 60%).



 cheers, sd



 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *adam compton
 *Sent:* Thursday, June 02, 2011 1:10 PM
 *To:* ccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] drop packets from 3750 output queue 2



 I have a switch qos item I am not sure how to address:

 When queue 2 reaches 60% capacity COS 4 packets should be dropped.

 COS 4,6,7 is mapped to queue 2.  any ideas?

 Adam Compton

 ___
 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] NTP on the lab

2011-06-02 Thread Cristobal Priego
will it be also valid to use the command

ntp source lo0 for example ?
or if you use the command above you risk the points ?

2011/6/2 Friderich Claude cfrider...@netcore.lu

  Check the time-zone they ask for and if this time-zone contains a
 daylight saving time (compare to the geographical sites), you have to put
 the clock summertime command as well.

 If this site doesn’t contain a Daylight Saving Time, not necessary to put
 this command ….. Normally you gonna find out on which site you have to ….



 Regards,



 [image: netcore_signature_4.jpg]





 *Claude FRIDERICH *

 PreSales Voice over IP

 * ***

 *[image: CCIEVoice_UseLogo]*

  CCIE Voice #29035

 *O*ffice**

 +352 31 33 80 – 407**

 *M*obile **

 +352 621 303 616**

 [image: cid:image003.jpg@01CC1F87.400817E0]

 *F*ax  **

 +352 31 33 80 412

 *E*mail**

 *cfrider...@netcore.lu x...@netcore.lu***





 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *adam compton
 *Sent:* jeudi 2 juin 2011 02:32

 *To:* ccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] NTP on the lab



 When I took the test last time I used the clock timezone command and the
 ntp server command to setup ntp.  I did not get points for this question.

 If the question on the Lab doesn't say anything about clock summertime,
 should we throw the command in anyway on the routers?  I know clock
 summertime without any options uses US daylight savings time dates, but not
 sure if those would apply to other timezones as well.  As best I can tell,
 I've heard of Hong Kong and UK and US as sites.

 Regards,
 Adam Compton



 --

 This email was Anti Virus checked.


 ___
 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

image003.jpgimage004.jpgimage001.jpg___
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] drop packets from 3750 output queue 2

2011-06-02 Thread Cristobal Priego
mls qos queue-set output 2 threshold 2 60 100 100 400

mls qos que-set ouput queue set #   threshold queue-id drop-threshold1
drop-threshold2 reserved-threshold maximum-threshold



2011/6/2 adam compton com...@gmail.com

 So I did a little bit of reading.  Auto QOS was mapping COS values to the
 queues at threshold 3.  I mapped COS 4 to queue 2 threshold 2:

 mls qos srr-queue output cos-map queue 2 threshold 2 4

 So I know that the command is something like someone listed:


 mls qos queue-set output 2 threshold 2 60 100 100 400

 What do each of the numbers mean?  From what I've seen, I don't think the
 last two would apply at all, but it seems like the first two are a min-max
 type setting.  I almost understand it, but just don't get what these values
 are.

 Thanks for all the help

 Adam


 On Thu, Jun 2, 2011 at 5:03 PM, Steve Denney (stdenney) 
 stden...@cisco.com wrote:

 CoS 4,6,7 are probably mapped to queue 2 threshold 2. Just move CoS 4 to
 queue 2 threshold **1**, then make sure the threshold settings for queue
 2 are 60 100 100 100 (the last 3 values aren’t critical, you just need the
 60 so that CoS 4 packets are dropped starting at threshold 1 – 60%).



 cheers, sd



 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *adam compton
 *Sent:* Thursday, June 02, 2011 1:10 PM
 *To:* ccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] drop packets from 3750 output queue 2



 I have a switch qos item I am not sure how to address:

 When queue 2 reaches 60% capacity COS 4 packets should be dropped.

 COS 4,6,7 is mapped to queue 2.  any ideas?

 Adam Compton



 ___
 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] Mobility softkey to send call to cell phone

2011-06-01 Thread Cristobal Priego

It will use the Rdo. Css to be moré specific
It will use the re-routing css

Enviado desde mi iPhone

El Jun 1, 2011, a las 6:18, adam compton com...@gmail.com escribió:

So I have mobility configured, SNR is working fine and dandy.  If I  
make a call and press mobility softkey to send the call to my cell  
phone,  What CSS does it use?  Does is follow phone css or remote  
dest profile CSS?


If it follows the device CSS, I could make sure that the SNR number  
is in a dialable format from the phone.  If it uses the RDP CSS,  
then I could potentially have to create another route pattern.


Thoughts?

Adam Compton
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please visit www.ipexpert.com


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Re: [OSL | CCIE_Voice] Mobility softkey to send call to cell phone

2011-06-01 Thread Cristobal Priego

I usually create a route pttrn for snr

Enviado desde mi iPhone

El Jun 1, 2011, a las 6:18, adam compton com...@gmail.com escribió:

So I have mobility configured, SNR is working fine and dandy.  If I  
make a call and press mobility softkey to send the call to my cell  
phone,  What CSS does it use?  Does is follow phone css or remote  
dest profile CSS?


If it follows the device CSS, I could make sure that the SNR number  
is in a dialable format from the phone.  If it uses the RDP CSS,  
then I could potentially have to create another route pattern.


Thoughts?

Adam Compton
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please visit www.ipexpert.com


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Re: [OSL | CCIE_Voice] 3750 or 3560

2011-06-01 Thread Cristobal Priego
i have a 3560 and i notice a difference yesterday on the QoS

if you run auto QoS on the 3560 it will not configure the service policy on
the port and it will not configure the priority-queue out by default

this is the auto qos config output from a 3750 the highlighted items will
not be configured on the 3560 when you run LAN QoS

Hth


interface FastEthernet0/2
 switchport access vlan 12
 switchport mode access
 switchport voice vlan 11
 srr-queue bandwidth share 10 10 60 20
 priority-queue out
 mls qos trust device cisco-phone
 mls qos trust cos
 auto qos voip cisco-phone
 spanning-tree portfast
 service-policy input AutoQoS-Police-CiscoPhone

2011/6/1 josh west west32...@gmail.com

 I need a switch for my lab and ideally would like to have a 3750 but from
 what I am reading the 3560 is very similar for what is needed on the voice
 lab. Can anyone confirm this or tell me what a 3750 has that the 3560 does
 not.

 Thanks in advanced.



 ___
 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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Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] 3750 or 3560

2011-06-01 Thread Cristobal Priego
yes, i did an ios upgrade before i started my journey and i was playing with
qos and was looking for the policy map and i couldn't find it
i need to check the ios, i'm not home now

2011/6/1 adam compton com...@gmail.com

 I get those commands on my 3560.  are you running advanced ip services on
 the 3560?


 On Wed, Jun 1, 2011 at 12:15 PM, Cristobal Priego 
 cristobalpri...@gmail.com wrote:

 i have a 3560 and i notice a difference yesterday on the QoS

 if you run auto QoS on the 3560 it will not configure the service policy
 on the port and it will not configure the priority-queue out by default

 this is the auto qos config output from a 3750 the highlighted items will
 not be configured on the 3560 when you run LAN QoS

 Hth


 interface FastEthernet0/2
  switchport access vlan 12
  switchport mode access
  switchport voice vlan 11
  srr-queue bandwidth share 10 10 60 20
  priority-queue out
  mls qos trust device cisco-phone
  mls qos trust cos
  auto qos voip cisco-phone
  spanning-tree portfast
  service-policy input AutoQoS-Police-CiscoPhone

 2011/6/1 josh west west32...@gmail.com

 I need a switch for my lab and ideally would like to have a 3750 but from
 what I am reading the 3560 is very similar for what is needed on the voice
 lab. Can anyone confirm this or tell me what a 3750 has that the 3560 does
 not.

 Thanks in advanced.



 ___
 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 http://www.platinumplacement.com/



 ___
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 visit www.ipexpert.com

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com http://www.platinumplacement.com/



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Re: [OSL | CCIE_Voice] ISDN bchan selection

2011-06-01 Thread Cristobal Priego
You can configure the router to select the first available B channel in
ascending order (channel B1) or descending order (channel B23 ).  By
default, the router will select outgoing calls in descending order.

2011/6/1 Randall Crumm rrcr...@yahoo.com

 Just to make sure top down mean channel 23 to 1
 do I would configure isdn bchan descending

 Thanks,
 randall



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Re: [OSL | CCIE_Voice] SW VPN Vs HW VPN during the Proctor Lab Session

2011-05-31 Thread Cristobal Priego
is not a must
with soft clients sometimes you assume your config is working, so you never
troubleshoot because you just can
with hardware phones you can test most off the features if not all of them

2011/5/31 Ilham Jabir ciscotodi...@gmail.com

 Hi,

 Currently i am using software vpn client to connect to my rack session.
 Using IP Blue clients, IP Communicator etc.. to practice the work books. How
 ever i am confused whether i may need hardware phones to test out the
 specific features like + dialing, etc..moving forward.

 Is it a must that we will require hardware phones during the proctor lab
 session to test all the features? Or only the soft phones will help us till
 we succeed the lab?

 Please advice.

 Ilham
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[OSL | CCIE_Voice] 3750 QoS

2011-05-31 Thread Cristobal Priego
guys i have a question for your

my questions is in regards to the shared bandwidth i know it's measured as a
weight and whenever you have shape you ignore the value so
the q2 in this case the weight is indeed 10 / (10+60+20) = so Q2 has 1/9 of
the shared bandwidth right ?

and this is how the config looks like

interface FastEthernet0/1
 switchport access vlan 6
 switchport voice vlan 8
 srr-queue bandwidth share 10 10 60 20
 srr-queue bandwidth shape  10  0  0  0
 mls qos trust device cisco-phone
 mls qos trust cos
 auto qos voip cisco-phone
 spanning-tree portfast

so if the bandwith needs to be shared on q2 40 %, q3 20%, q4 40%

how could i calculate or know what do i need to put on  srr-queue bandwidth
share command



will this be an accurate config  srr-queue bandwidth share 0 40 20 40 ??

thanks
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Re: [OSL | CCIE_Voice] 3750 QoS

2011-05-31 Thread Cristobal Priego
thank you

2011/5/31 ccieid1ot ccieid...@gmail.com

 Answer your 1st question, its 1/10th.

 2nd question is correct.

 duy
 ccie #27737 voice

 tmobile g2
 On May 31, 2011 4:19 PM, Cristobal Priego cristobalpri...@gmail.com
 wrote:
  guys i have a question for your
 
  my questions is in regards to the shared bandwidth i know it's measured
 as a
  weight and whenever you have shape you ignore the value so
  the q2 in this case the weight is indeed 10 / (10+60+20) = so Q2 has 1/9
 of
  the shared bandwidth right ?
 
  and this is how the config looks like
 
  interface FastEthernet0/1
  switchport access vlan 6
  switchport voice vlan 8
  srr-queue bandwidth share 10 10 60 20
  srr-queue bandwidth shape 10 0 0 0
  mls qos trust device cisco-phone
  mls qos trust cos
  auto qos voip cisco-phone
  spanning-tree portfast
 
  so if the bandwith needs to be shared on q2 40 %, q3 20%, q4 40%
 
  how could i calculate or know what do i need to put on srr-queue
 bandwidth
  share command
 
 
 
  will this be an accurate config srr-queue bandwidth share 0 40 20 40 ??
 
  thanks

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Re: [OSL | CCIE_Voice] SW VPN Vs HW VPN during the Proctor Lab Session

2011-05-31 Thread Cristobal Priego
yes no problem
i know 2 guys that got their ccie voice and they were only using soft phones

2011/5/31 Ilham Jabir ciscotodi...@gmail.com

 So i can re use the 2$$$ to purchase few 100 hrs of proctor lab session
 instead of buying the partial hardware which i will use during the rack
 session.

 Bottom line is that we can survive with only with soft phones  proctor
 labs for practicing the work books  prepare for the exam.

 Thanks

 Ilham
 On 01-Jun-2011, at 1:03 AM, Cristobal Priego wrote:

 is not a must
 with soft clients sometimes you assume your config is working, so you never
 troubleshoot because you just can
 with hardware phones you can test most off the features if not all of them

 2011/5/31 Ilham Jabir ciscotodi...@gmail.com

 Hi,

 Currently i am using software vpn client to connect to my rack session.
 Using IP Blue clients, IP Communicator etc.. to practice the work books. How
 ever i am confused whether i may need hardware phones to test out the
 specific features like + dialing, etc..moving forward.

 Is it a must that we will require hardware phones during the proctor lab
 session to test all the features? Or only the soft phones will help us till
 we succeed the lab?

 Please advice.

 Ilham
 ___
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 visit www.ipexpert.com

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 www.PlatinumPlacement.com




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[OSL | CCIE_Voice] LAN QoS

2011-05-31 Thread Cristobal Priego
hello all,

I have a question for you guys

hopefully you can help me understand, this is way too confusing for me

when i issue auto qos on a 3750 switch this is how the config will look like




mls qos srr-queue output cos-map queue 1 threshold 3  5
mls qos srr-queue output cos-map queue 2 threshold 3  3 6 7
mls qos srr-queue output cos-map queue 3 threshold 3  2 4
mls qos srr-queue output cos-map queue 4 threshold 2  1
mls qos srr-queue output cos-map queue 4 threshold 3  0
mls qos srr-queue output dscp-map queue 1 threshold 3  40 41 42 43 44 45 46
47
mls qos srr-queue output dscp-map queue 2 threshold 3  24 25 26 27 28 29 30
31
mls qos srr-queue output dscp-map queue 2 threshold 3  48 49 50 51 52 53 54
55
mls qos srr-queue output dscp-map queue 2 threshold 3  56 57 58 59 60 61 62
63
mls qos srr-queue output dscp-map queue 3 threshold 3  16 17 18 19 20 21 22
23
mls qos srr-queue output dscp-map queue 3 threshold 3  32 33 34 35 36 37 38
39
mls qos srr-queue output dscp-map queue 4 threshold 1  8
mls qos srr-queue output dscp-map queue 4 threshold 2  9 10 11 12 13 14 15
mls qos srr-queue output dscp-map queue 4 threshold 3  0 1 2 3 4 5 6 7
mls qos queue-set output 1 threshold 1 138 138 92 138
mls qos queue-set output 1 threshold 2 138 138 92 400
mls qos queue-set output 1 threshold 3 36 77 100 318
mls qos queue-set output 1 threshold 4 20 50 67 400
mls qos queue-set output 2 threshold 1 149 149 100 149
mls qos queue-set output 2 threshold 2 118 118 100 235
mls qos queue-set output 2 threshold 3 41 68 100 272
mls qos queue-set output 2 threshold 4 42 72 100 242
mls qos queue-set output 1 buffers 10 10 26 54
mls qos queue-set output 2 buffers 16 6 17 61
mls qos


when i'm asked to change the COS or DSCP values from Q to Q

how do I know which threshold to use ?

for exambple

if i'm asked to configure the COS valus 2,3 to Queue 4

how do i know on which threshold withing Q4 ?



does my question make sense ?  this part of the threshold is very confusing
for me

hopefully you can help me out


thanks
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Re: [OSL | CCIE_Voice] auto qos voip trust fr-atm

2011-05-30 Thread Cristobal Priego
do you recall the subject ?
I can't find it
i found some qos discussions, however it was related to LAN qos

thank you

2011/5/30 Bill Lake whl...@gmail.com

 I believe it was about the first week of May ;)

 On Mon, May 30, 2011 at 11:52 AM, Friderich Claude 
 cfrider...@netcore.luwrote:

  Hi,



 Look at the forum history, I spoke about this issue and how to manage and
 resolve this issue a couple of week ago …. ;-)



 *Claude Friderich*

 *PreSales Support*

 *CCIE Voice #29035*

 *[image: CCIEVoice_UseLogo]***

 *NETCORE PSF S.A.***

 49 rue du Baerendall

 B.P.65 L-8201 Mamer

 Téléphone: +352 31 33 80 407

 Fax: +352 31 33 80 412

 GSM: +352 621 303 616

 E-mail: cfrider...@netcore.lu



 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *adam compton
 *Sent:* lundi 30 mai 2011 15:45
 *To:* Cristobal Priego
 *Cc:* ccie_voice@onlinestudylist.com
 *Subject:* Re: [OSL | CCIE_Voice] auto qos voip trust fr-atm



 I've noticed that if you reload BR1 router, it takes a long time for OSPF
 to populate the routes again.  I've had to wait like 5-10 minutes before for
 the routes to get back on the router.  That could possibly be what you're
 seeing.  If that is the case, you should be able to ping the serial
 interfaces between HQ and BR1, but nothing else on HQ or BR2.



 Adam Compton

 On Sun, May 29, 2011 at 11:44 PM, Cristobal Priego 
 cristobalpri...@gmail.com wrote:

 hello all

 while doing a mock lab
  i needed to configure LLQ, MLP LFI cRTP between BR1 and HQ, the bw was
 384

 when i issued auto qos voip trust fr-atm on both interfaces, after setting
 the BW on the inferfaces
 everything was great, no problem

 however once i reloaded the BR1 router
 i lost connectivity to the HQ site

 i reloaded HQ router and still no connectivity


 so i reverted my pod

 and resstarted again only with QOS

 i set the BW
 and ran the command again
 without any customization

 and the same issue happened, where i lost connectivity from BR1 to HQ

 do you guys know how to fix this ?

 thank you

 ___
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 visit www.ipexpert.com

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 www.PlatinumPlacement.com http://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



image001.jpg___
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Re: [OSL | CCIE_Voice] auto qos voip trust fr-atm

2011-05-30 Thread Cristobal Priego
I found it

just in case somebody else is interested this is exactly what happened to me
and indeed Friderich described quite well
i'm going to copy paste the issue and the solution that he found

As I configure qos with MLP LFI between hq and br2 everything was working
fine !!!



After rebooting, no serial access anymore between hq and br2…..



With  *show ip interface brief* I have the following protocol state :



*HQ*

Interface IP-AddressOK?
Method   StatusProtocol

Virtual-Access1 *10.10.112.1*YES
TFTP up   down



*BR2*

Interface IP-AddressOK?
Method   StatusProtocol
Virtual-Access1* 10.10.112.2*YES
TFTP up   down



onclusion:

On HQ, Just have to put the command line no frame-relay interface-dlci 202
ppp Virtual-Template100

and shut the Serial interface  0/1/0 and put the command  frame-relay
interface-dlci 202 ppp Virtual-Template100 and no shut  to make it work
again





Hope this can help you during the real lab …  Think it’s the easiest
solution to make it work …. Have to keep this in mind J

And again, thanks to all for your remarks and suggestions



i'm going to give this a try in 15 mins i just need to have all the
infrastructure stuff fixed




2011/5/30 Cristobal Priego cristobalpri...@gmail.com

 do you recall the subject ?
 I can't find it
 i found some qos discussions, however it was related to LAN qos

 thank you


 2011/5/30 Bill Lake whl...@gmail.com

 I believe it was about the first week of May ;)

 On Mon, May 30, 2011 at 11:52 AM, Friderich Claude cfrider...@netcore.lu
  wrote:

  Hi,



 Look at the forum history, I spoke about this issue and how to manage and
 resolve this issue a couple of week ago …. ;-)



 *Claude Friderich*

 *PreSales Support*

 *CCIE Voice #29035*

 *[image: CCIEVoice_UseLogo]***

 *NETCORE PSF S.A.***

 49 rue du Baerendall

 B.P.65 L-8201 Mamer

 Téléphone: +352 31 33 80 407

 Fax: +352 31 33 80 412

 GSM: +352 621 303 616

 E-mail: cfrider...@netcore.lu



 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *adam compton
 *Sent:* lundi 30 mai 2011 15:45
 *To:* Cristobal Priego
 *Cc:* ccie_voice@onlinestudylist.com
 *Subject:* Re: [OSL | CCIE_Voice] auto qos voip trust fr-atm



 I've noticed that if you reload BR1 router, it takes a long time for OSPF
 to populate the routes again.  I've had to wait like 5-10 minutes before for
 the routes to get back on the router.  That could possibly be what you're
 seeing.  If that is the case, you should be able to ping the serial
 interfaces between HQ and BR1, but nothing else on HQ or BR2.



 Adam Compton

 On Sun, May 29, 2011 at 11:44 PM, Cristobal Priego 
 cristobalpri...@gmail.com wrote:

 hello all

 while doing a mock lab
  i needed to configure LLQ, MLP LFI cRTP between BR1 and HQ, the bw was
 384

 when i issued auto qos voip trust fr-atm on both interfaces, after
 setting the BW on the inferfaces
 everything was great, no problem

 however once i reloaded the BR1 router
 i lost connectivity to the HQ site

 i reloaded HQ router and still no connectivity


 so i reverted my pod

 and resstarted again only with QOS

 i set the BW
 and ran the command again
 without any customization

 and the same issue happened, where i lost connectivity from BR1 to HQ

 do you guys know how to fix this ?

 thank you

 ___
 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 http://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




image001.jpg___
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Re: [OSL | CCIE_Voice] auto qos voip trust fr-atm

2011-05-30 Thread Cristobal Priego
I just tried it
and this is what happens

I reloaded the router and i don't have connectivity to BR1 now, so i started
on the HQ router
this is how my interface looks before i make the change


interface Serial0/0/1:0.1 point-to-point
 bandwidth 384
 ip ospf mtu-ignore
 snmp trap link-status
 frame-relay interface-dlci 201 ppp Virtual-Template200
  class AutoQoS-FR-Se0/0/1:0-201
  auto qos voip trust fr-atm



R1#conf t
Enter configuration commands, one per line.  End with CNTL/Z.
R1(config)#interface Serial0/0/1:0.1 point-to-point
R1(config-subif)#sh
R1(config-subif)#no frame-relay interface-dlci 201 ppp Virtual-Template200

as soon as I enter that command
I get the Errors below:

no frame-relay interface-dlci 201 ppp Virtual-Template200
  ^
% Invalid input detected at '^' marker.

AutoQoS Error: the following command could not be parsed:
no frame-relay interface-dlci 201 ppp Virtual-Template200

R1(config-subif)#
*May 31 01:51:19.535: %LINK-5-CHANGED: Interface Virtual-Template200,
changed state to administratively down
R1(config-subif)#


When I try to re-enter the Command on the sub interface

this is what i get:


R1(config-subif)#frame-relay interface-dlci 201 ppp Virtual-Template200
%vtemplate error; failed to set up PPP-FR circuit
R1(config-subif)#



and my interface looks like this :

!
interface Serial0/0/1:0.1 point-to-point
 bandwidth 384
 ip address 10.10.111.1 255.255.255.0
 ip ospf mtu-ignore
 shutdown
 snmp trap link-status
end




2011/5/30 Friderich Claude cfrider...@netcore.lu

  For MLP LFI, if you create the virtual-template manually (no auto-qos),
 it’s going to work.

 If not, just got to “frame-relay interface-dlci 102 ppp
 Virtual-Template200” and put no frame-relay interface-dlci 102 ppp
 Virtual-Template200

 Shut the serial interface.

 Put frame-relay interface-dlci 102 ppp Virtual-Template200 again and no
 shut the Serial Interface.

 Do this manipulation on the 2 sites configured with MLP.



 But for the Lab, normally you just have to reload the routers to make it
 work.

 Check with show policy-map interface to make sure that the policy-map is
 good configured and that you have traffic for sig and RTP



 Regards,



 *Claude Friderich*

 *PreSales Support*

 *CCIE Voice #29035*

 *[image: CCIEVoice_UseLogo]***

 *NETCORE PSF S.A.***

 49 rue du Baerendall

 B.P.65 L-8201 Mamer

 Téléphone: +352 31 33 80 407

 Fax: +352 31 33 80 412

 GSM: +352 621 303 616

 E-mail: cfrider...@netcore.lu



 *From:* Cristobal Priego [mailto:cristobalpri...@gmail.com]
 *Sent:* lundi 30 mai 2011 22:13
 *To:* Bill Lake
 *Cc:* Friderich Claude; adam compton; ccie_voice@onlinestudylist.com

 *Subject:* Re: [OSL | CCIE_Voice] auto qos voip trust fr-atm



 do you recall the subject ?
 I can't find it
 i found some qos discussions, however it was related to LAN qos

 thank you

 2011/5/30 Bill Lake whl...@gmail.com

 I believe it was about the first week of May ;)



 On Mon, May 30, 2011 at 11:52 AM, Friderich Claude cfrider...@netcore.lu
 wrote:

 Hi,



 Look at the forum history, I spoke about this issue and how to manage and
 resolve this issue a couple of week ago …. ;-)



 *Claude Friderich*

 *PreSales Support*

 *CCIE Voice #29035*

 *[image: CCIEVoice_UseLogo]*

 *NETCORE PSF S.A.*

 49 rue du Baerendall

 B.P.65 L-8201 Mamer

 Téléphone: +352 31 33 80 407

 Fax: +352 31 33 80 412

 GSM: +352 621 303 616

 E-mail: cfrider...@netcore.lu



 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *adam compton
 *Sent:* lundi 30 mai 2011 15:45
 *To:* Cristobal Priego
 *Cc:* ccie_voice@onlinestudylist.com
 *Subject:* Re: [OSL | CCIE_Voice] auto qos voip trust fr-atm



 I've noticed that if you reload BR1 router, it takes a long time for OSPF
 to populate the routes again.  I've had to wait like 5-10 minutes before for
 the routes to get back on the router.  That could possibly be what you're
 seeing.  If that is the case, you should be able to ping the serial
 interfaces between HQ and BR1, but nothing else on HQ or BR2.



 Adam Compton

 On Sun, May 29, 2011 at 11:44 PM, Cristobal Priego 
 cristobalpri...@gmail.com wrote:

 hello all

 while doing a mock lab
  i needed to configure LLQ, MLP LFI cRTP between BR1 and HQ, the bw was 384

 when i issued auto qos voip trust fr-atm on both interfaces, after setting
 the BW on the inferfaces
 everything was great, no problem

 however once i reloaded the BR1 router
 i lost connectivity to the HQ site

 i reloaded HQ router and still no connectivity


 so i reverted my pod

 and resstarted again only with QOS

 i set the BW
 and ran the command again
 without any customization

 and the same issue happened, where i lost connectivity from BR1 to HQ

 do you guys know how to fix this ?

 thank you

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

 Are you

Re: [OSL | CCIE_Voice] Gatekeeper question--RC

2011-05-30 Thread Cristobal Priego
did you reset the Trunk after you made the change on Service Parameters ?

If you have a GK on the lab and if it's telling you to use 1720 and your
output doesn't show port 1720



2011/5/30 Randall Crumm rrcr...@yahoo.com


 Thanks for all the answers.

 I did search the services for 1720 and entered gk-trunk.

 I'm working on WB2 lab 2 and the output from the PG shows port 1720 and my
 output shows 40081 and 33277 respectively below.

 The questions in the WB has output and says the port should show 1720
 no 40081 and 33277. This is what I am trying to do. The PG did not specify
 anything specific as well.

 Should I not worry about this?

 Randall




 --
 *From:* Roger Carpio roger.car...@gmail.com
 *To:* Randall Crumm rrcr...@yahoo.com
 *Cc:* Online Study ccie_voice@onlinestudylist.com
 *Sent:* Monday, May 30, 2011 12:07 PM
 *Subject:* Re: [OSL | CCIE_Voice] Gatekeeper question

 CCM service Device Name of GK-controlled Trunk That Will Use Port 1720;
 just type the trunk name.

 Rgs,
 Roger Carpio.

 On Mon, May 30, 2011 at 11:39 AM, Randall Crumm rrcr...@yahoo.com wrote:

 Hi,
 How do I get gk-trunk_1 and 2  to repflect port 1720?

 Thanks,
 Randall

 HQ-RTR(config)#do show gatekeeper gw
 GATEWAY TYPE PREFIX TABLE
 =
 Prefix: 3*
   Zone HQ master gateway list:
 10.10.110.3:1720 BR2-RTR

 Prefix: 1#*
   Zone HQ master gateway list:
 10.10.210.10:40081 GK-TRUNK_1
 10.10.210.11:33277 GK-TRUNK_2
   Zone HQ prefix 1... priority gateway list(s):
Priority 5:
 10.10.210.10:40081 GK-TRUNK_1
 10.10.210.11:33277 GK-TRUNK_2
   Zone HQ prefix 5... priority gateway list(s):
Priority 5:
 10.10.210.10:40081 GK-TRUNK_1
 10.10.210.11:33277 GK-TRUNK_2



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 visit www.ipexpert.com

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 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] CUE/CME

2011-05-30 Thread Cristobal Priego
no
when you do the initial setup for CUE
it indeed request a user id/password, the last step on the setup

those are the credentials that you should use

the user/name password set under telephony-service
that's indeed user for the integration


2011/5/30 Randall Crumm rrcr...@yahoo.com

 HI,
 Here is a dumb question, but causing me issues.

 My cue is stable and I can log in to the web page with the user from the
 cue cli.

 Now when I start the initialization wizard I keep getting login to CME as
 administrator failed. check CME config.

 Am I suppose to use the user under tele-service? In my case admin/c1sc0123?

 snip-its

 ip http server
 ip http authentication local
 no ip http secure-server
 ip http path flash:/GUI


 telephony-service
  max-ephones 7
  max-dn 7
  ip source-address 10.10.110.3 port 2000
  voicemail 3600
  max-conferences 8 gain -6
  call-forward pattern .T
  web admin system name admin password c1sc0123
  dn-webedit
  transfer-system full-consult
  transfer-pattern .
  transfer-pattern .T


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Re: [OSL | CCIE_Voice] auto qos voip trust fr-atm

2011-05-30 Thread Cristobal Priego
thanks
i indeed found the Thread

i finally got it working
what's happening in my case is

when i go to the sub interface and remove the line
frame-relay interface-dlci 201 ppp Virtual-Template200


the virtual-tempate is gone
so i need to recreate the template first


interface Virtual-Template200
 bandwidth 384
 ip address 10.10.111.1 255.255.255.0
 ppp multilink
 ppp multilink interleave
 ppp multilink fragment delay 10
 service-policy output AutoQoS-Policy-Trust
end


then go back to the sub interface re-issue the command

frame-relay interface-dlci 201 ppp Virtual-Template200

and i get the connectivity back  : )

I feel much better now.



2011/5/30 Miron Kobelski findko...@gmail.com

 http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg21425.html



 On Mon, May 30, 2011 at 22:13, Cristobal Priego cristobalpri...@gmail.com
  wrote:

 do you recall the subject ?
 I can't find it
 i found some qos discussions, however it was related to LAN qos

 thank you


 2011/5/30 Bill Lake whl...@gmail.com

 I believe it was about the first week of May ;)

 On Mon, May 30, 2011 at 11:52 AM, Friderich Claude 
 cfrider...@netcore.lu wrote:

  Hi,



 Look at the forum history, I spoke about this issue and how to manage
 and resolve this issue a couple of week ago …. ;-)



 *Claude Friderich*

 *PreSales Support*

 *CCIE Voice #29035*

 *[image: CCIEVoice_UseLogo]***

 *NETCORE PSF S.A.***

 49 rue du Baerendall

 B.P.65 L-8201 Mamer

 Téléphone: +352 31 33 80 407

 Fax: +352 31 33 80 412

 GSM: +352 621 303 616

 E-mail: cfrider...@netcore.lu



 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *adam compton
 *Sent:* lundi 30 mai 2011 15:45
 *To:* Cristobal Priego
 *Cc:* ccie_voice@onlinestudylist.com
 *Subject:* Re: [OSL | CCIE_Voice] auto qos voip trust fr-atm



 I've noticed that if you reload BR1 router, it takes a long time for
 OSPF to populate the routes again.  I've had to wait like 5-10 minutes
 before for the routes to get back on the router.  That could possibly be
 what you're seeing.  If that is the case, you should be able to ping the
 serial interfaces between HQ and BR1, but nothing else on HQ or BR2.



 Adam Compton

 On Sun, May 29, 2011 at 11:44 PM, Cristobal Priego 
 cristobalpri...@gmail.com wrote:

 hello all

 while doing a mock lab
  i needed to configure LLQ, MLP LFI cRTP between BR1 and HQ, the bw was
 384

 when i issued auto qos voip trust fr-atm on both interfaces, after
 setting the BW on the inferfaces
 everything was great, no problem

 however once i reloaded the BR1 router
 i lost connectivity to the HQ site

 i reloaded HQ router and still no connectivity


 so i reverted my pod

 and resstarted again only with QOS

 i set the BW
 and ran the command again
 without any customization

 and the same issue happened, where i lost connectivity from BR1 to HQ

 do you guys know how to fix this ?

 thank you

 ___
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 For more information regarding industry leading CCIE Lab training,
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 www.PlatinumPlacement.com




 ___
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 www.PlatinumPlacement.com



image001.jpg___
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Re: [OSL | CCIE_Voice] auto qos voip trust fr-atm

2011-05-30 Thread Cristobal Priego
yes you're right

that's what i'm going to do
i have never configure it manually so it's a good practice

will the qos srnd will tell me how to configure it ?

I usually use auto qos, because i'm afraid i'm going to forget something

but its something that i'd definitely want to do configure it manually

if you know any doc that will let me know how to configure it please let me
know
i'd really appreciated it

thanks

2011/5/30 Friderich Claude cfrider...@netcore.lu

  This is because you configured with auto-qos.

 You have to create it manually and it is good for your training and
 understanding ;-)



 *Claude Friderich*

 *PreSales Support*

 *CCIE Voice #29035*

 *[image: CCIEVoice_UseLogo]***

 *NETCORE PSF S.A.***

 49 rue du Baerendall

 B.P.65 L-8201 Mamer

 Téléphone: +352 31 33 80 407

 Fax: +352 31 33 80 412

 GSM: +352 621 303 616

 E-mail: cfrider...@netcore.lu



 *From:* Cristobal Priego [mailto:cristobalpri...@gmail.com]
 *Sent:* lundi 30 mai 2011 22:55
 *To:* Friderich Claude
 *Cc:* Bill Lake; adam compton; ccie_voice@onlinestudylist.com

 *Subject:* Re: [OSL | CCIE_Voice] auto qos voip trust fr-atm



 I just tried it
 and this is what happens

 I reloaded the router and i don't have connectivity to BR1 now, so i
 started on the HQ router
 this is how my interface looks before i make the change


 interface Serial0/0/1:0.1 point-to-point
  bandwidth 384
  ip ospf mtu-ignore
  snmp trap link-status
  frame-relay interface-dlci 201 ppp Virtual-Template200
   class AutoQoS-FR-Se0/0/1:0-201
   auto qos voip trust fr-atm



 R1#conf t
 Enter configuration commands, one per line.  End with CNTL/Z.
 R1(config)#interface Serial0/0/1:0.1 point-to-point
 R1(config-subif)#sh
 R1(config-subif)#no frame-relay interface-dlci 201 ppp Virtual-Template200

 as soon as I enter that command
 I get the Errors below:

 no frame-relay interface-dlci 201 ppp Virtual-Template200
   ^
 % Invalid input detected at '^' marker.

 AutoQoS Error: the following command could not be parsed:
 no frame-relay interface-dlci 201 ppp Virtual-Template200

 R1(config-subif)#
 *May 31 01:51:19.535: %LINK-5-CHANGED: Interface Virtual-Template200,
 changed state to administratively down
 R1(config-subif)#


 When I try to re-enter the Command on the sub interface

 this is what i get:


 R1(config-subif)#frame-relay interface-dlci 201 ppp Virtual-Template200
 %vtemplate error; failed to set up PPP-FR circuit
 R1(config-subif)#



 and my interface looks like this :

 !
 interface Serial0/0/1:0.1 point-to-point
  bandwidth 384
  ip address 10.10.111.1 255.255.255.0
  ip ospf mtu-ignore
  shutdown
  snmp trap link-status
 end



  2011/5/30 Friderich Claude cfrider...@netcore.lu

 For MLP LFI, if you create the virtual-template manually (no auto-qos),
 it’s going to work.

 If not, just got to “frame-relay interface-dlci 102 ppp
 Virtual-Template200” and put no frame-relay interface-dlci 102 ppp
 Virtual-Template200

 Shut the serial interface.

 Put frame-relay interface-dlci 102 ppp Virtual-Template200 again and no
 shut the Serial Interface.

 Do this manipulation on the 2 sites configured with MLP.



 But for the Lab, normally you just have to reload the routers to make it
 work.

 Check with show policy-map interface to make sure that the policy-map is
 good configured and that you have traffic for sig and RTP



 Regards,



 *Claude Friderich*

 *PreSales Support*

 *CCIE Voice #29035*

 *[image: CCIEVoice_UseLogo]*

 *NETCORE PSF S.A.*

 49 rue du Baerendall

 B.P.65 L-8201 Mamer

 Téléphone: +352 31 33 80 407

 Fax: +352 31 33 80 412

 GSM: +352 621 303 616

 E-mail: cfrider...@netcore.lu



 *From:* Cristobal Priego [mailto:cristobalpri...@gmail.com]
 *Sent:* lundi 30 mai 2011 22:13
 *To:* Bill Lake
 *Cc:* Friderich Claude; adam compton; ccie_voice@onlinestudylist.com


 *Subject:* Re: [OSL | CCIE_Voice] auto qos voip trust fr-atm



 do you recall the subject ?
 I can't find it
 i found some qos discussions, however it was related to LAN qos

 thank you

 2011/5/30 Bill Lake whl...@gmail.com

 I believe it was about the first week of May ;)



 On Mon, May 30, 2011 at 11:52 AM, Friderich Claude cfrider...@netcore.lu
 wrote:

 Hi,



 Look at the forum history, I spoke about this issue and how to manage and
 resolve this issue a couple of week ago …. ;-)



 *Claude Friderich*

 *PreSales Support*

 *CCIE Voice #29035*

 *[image: CCIEVoice_UseLogo]*

 *NETCORE PSF S.A.*

 49 rue du Baerendall

 B.P.65 L-8201 Mamer

 Téléphone: +352 31 33 80 407

 Fax: +352 31 33 80 412

 GSM: +352 621 303 616

 E-mail: cfrider...@netcore.lu



 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *adam compton
 *Sent:* lundi 30 mai 2011 15:45
 *To:* Cristobal Priego
 *Cc:* ccie_voice@onlinestudylist.com
 *Subject:* Re: [OSL | CCIE_Voice] auto qos voip trust fr-atm



 I've noticed that if you reload BR1 router, it takes a long time for OSPF

Re: [OSL | CCIE_Voice] auto qos voip trust fr-atm

2011-05-30 Thread Cristobal Priego
thank you very much

2011/5/30 Friderich Claude cfrider...@netcore.lu

  Not so difficult …

 Check the config with auto-qos  and copy paste on notepad

 Reload the router without saving your config



 Put your BW in the sub Serial interface

 Put frame-relay traffic-shaping in your Serial Interface



 Create your map-class (BR2 for example)

 Create your own virtual-template (virtual-template 1 for example)

 Put no ip address and no frame-relay interface-dlci on the sub interface

 Replace with “frame-relay interface-dlci 202 ppp Virtual-Template1” (for
 example)

 Put the new map-class with the name you configured above

 Don’t forget to apply your policy on the virtual-template



 Should work ….



 *Claude Friderich*

 *PreSales Support*

 *CCIE Voice #29035*

 *[image: CCIEVoice_UseLogo]***

 *NETCORE PSF S.A.***

 49 rue du Baerendall

 B.P.65 L-8201 Mamer

 Téléphone: +352 31 33 80 407

 Fax: +352 31 33 80 412

 GSM: +352 621 303 616

 E-mail: cfrider...@netcore.lu



 *From:* Cristobal Priego [mailto:cristobalpri...@gmail.com]
 *Sent:* lundi 30 mai 2011 23:23

 *To:* Friderich Claude
 *Cc:* Bill Lake; adam compton; ccie_voice@onlinestudylist.com
 *Subject:* Re: [OSL | CCIE_Voice] auto qos voip trust fr-atm



 yes you're right

 that's what i'm going to do
 i have never configure it manually so it's a good practice

 will the qos srnd will tell me how to configure it ?

 I usually use auto qos, because i'm afraid i'm going to forget something

 but its something that i'd definitely want to do configure it manually

 if you know any doc that will let me know how to configure it please let me
 know
 i'd really appreciated it

 thanks

 2011/5/30 Friderich Claude cfrider...@netcore.lu

 This is because you configured with auto-qos.

 You have to create it manually and it is good for your training and
 understanding ;-)



 *Claude Friderich*

 *PreSales Support*

 *CCIE Voice #29035*

 *[image: CCIEVoice_UseLogo]*

 *NETCORE PSF S.A.*

 49 rue du Baerendall

 B.P.65 L-8201 Mamer

 Téléphone: +352 31 33 80 407

 Fax: +352 31 33 80 412

 GSM: +352 621 303 616

 E-mail: cfrider...@netcore.lu



 *From:* Cristobal Priego [mailto:cristobalpri...@gmail.com]
 *Sent:* lundi 30 mai 2011 22:55
 *To:* Friderich Claude
 *Cc:* Bill Lake; adam compton; ccie_voice@onlinestudylist.com


 *Subject:* Re: [OSL | CCIE_Voice] auto qos voip trust fr-atm



 I just tried it
 and this is what happens

 I reloaded the router and i don't have connectivity to BR1 now, so i
 started on the HQ router
 this is how my interface looks before i make the change


 interface Serial0/0/1:0.1 point-to-point
  bandwidth 384
  ip ospf mtu-ignore
  snmp trap link-status
  frame-relay interface-dlci 201 ppp Virtual-Template200
   class AutoQoS-FR-Se0/0/1:0-201
   auto qos voip trust fr-atm



 R1#conf t
 Enter configuration commands, one per line.  End with CNTL/Z.
 R1(config)#interface Serial0/0/1:0.1 point-to-point
 R1(config-subif)#sh
 R1(config-subif)#no frame-relay interface-dlci 201 ppp Virtual-Template200

 as soon as I enter that command
 I get the Errors below:

 no frame-relay interface-dlci 201 ppp Virtual-Template200
   ^
 % Invalid input detected at '^' marker.

 AutoQoS Error: the following command could not be parsed:
 no frame-relay interface-dlci 201 ppp Virtual-Template200

 R1(config-subif)#
 *May 31 01:51:19.535: %LINK-5-CHANGED: Interface Virtual-Template200,
 changed state to administratively down
 R1(config-subif)#


 When I try to re-enter the Command on the sub interface

 this is what i get:


 R1(config-subif)#frame-relay interface-dlci 201 ppp Virtual-Template200
 %vtemplate error; failed to set up PPP-FR circuit
 R1(config-subif)#



 and my interface looks like this :

 !
 interface Serial0/0/1:0.1 point-to-point
  bandwidth 384
  ip address 10.10.111.1 255.255.255.0
  ip ospf mtu-ignore
  shutdown
  snmp trap link-status
 end


  2011/5/30 Friderich Claude cfrider...@netcore.lu

 For MLP LFI, if you create the virtual-template manually (no auto-qos),
 it’s going to work.

 If not, just got to “frame-relay interface-dlci 102 ppp
 Virtual-Template200” and put no frame-relay interface-dlci 102 ppp
 Virtual-Template200

 Shut the serial interface.

 Put frame-relay interface-dlci 102 ppp Virtual-Template200 again and no
 shut the Serial Interface.

 Do this manipulation on the 2 sites configured with MLP.



 But for the Lab, normally you just have to reload the routers to make it
 work.

 Check with show policy-map interface to make sure that the policy-map is
 good configured and that you have traffic for sig and RTP



 Regards,



 *Claude Friderich*

 *PreSales Support*

 *CCIE Voice #29035*

 *[image: CCIEVoice_UseLogo]*

 *NETCORE PSF S.A.*

 49 rue du Baerendall

 B.P.65 L-8201 Mamer

 Téléphone: +352 31 33 80 407

 Fax: +352 31 33 80 412

 GSM: +352 621 303 616

 E-mail: cfrider...@netcore.lu



 *From:* Cristobal Priego [mailto:cristobalpri...@gmail.com]
 *Sent

Re: [OSL | CCIE_Voice] RSVP with CUE

2011-05-30 Thread Cristobal Priego
I just tested it too and it works with the transcoder as Randall said

2011/5/30 Randall Saborio ill2...@gmail.com

 Hey guys, just to bring up an old discussion.

 Had to prove to myself if transcoder would work as RSVP and it works just
 fine for me.

 Not sure what you may be missing, but I got it to work the following way:
 BR2:

 dspfarm profile 1 transcode
  codec g711ulaw
  codec g711alaw
  codec g729ar8
  codec g729abr8
  codec g729r8
  codec g729br8
  rsvp
  maximum sessions 4
  associate application SCCP


 BR2-RTR#show dspfarm dsp active
 SLOT DSP VERSION  STATUS CHNL USE   TYPERSC_ID BRIDGE_ID PKTS_TXED
 PKTS_RXED

 02   23.6.1   UP 1USED  xcode   1  0x38130
 364
 02   23.6.1   UP 1USED  xcode   1  0x39431
 299

 Total number of DSPFARM DSP channel(s) 1


 BR2-RTR#show ip rsvp reservation
 ToFrom  Pro DPort Sport Next Hop  I/F  Fi Serv
 BPS
 10.20.110.1   10.20.110.3   UDP 18280 19100 10.20.112.1   Se0/1/0. FF LOAD
 24K
 10.20.110.3   10.20.110.1   UDP 19100 18280 none  none FF LOAD
 24K

 HQ will be exactly the same. What you must take care of when using
 transcoder and RSVP on same profile, is that on Region configuration it is
 important that the transcoders are forced g729 with each other, and then
 g711 towards the local device.

 IOS transcoder will always break if tries to talk g729 to one device and
 g729 towards the other.

 On my tests, you can notice it reserved g729 and on the phones if I hit the
 stats, would bring g711ulaw as that's what is talking towards the
 transcoder.

 On HQ site, it was valid to replace the transcoder with an mtp too, but BR2
 I left it as transcoder and still worked.


 Cheers!
 Randall S.



 2011/5/19 voice boy voice...@hotmail.com

  Again you do it
 Solved :) , you deserve a lot

 --
 CC: ill2...@gmail.com; ccie_voice@onlinestudylist.com
 From: gogli...@gmail.com

 Subject: Re: [OSL | CCIE_Voice] RSVP with CUE
 Date: Thu, 19 May 2011 21:20:12 +0100
 To: voice...@hotmail.com


 Hi voice boy,

 Do not use xcoder for RSVP. I do not think you would be able to do RSVP
 and transcoding at the same time with the same dspfarm profile.

 Do the following:

 Configure MTP for RSVP again in the following manner:
 dspfarm profile 1 MTP
  no codec g711ulaw
  codec g729r
  codec passthrough
   rsvp
  maximum session software 10
  no shutdown

 And on the CUE side configure xcoder resource with g711 and g729r codec
 too.

 It should work now,

 The problem with RSVP MTP agent is that between sites it normally uses
 passthrough codec which means endpoints do the codec negotiation. But when
 the codec negotiation fails between endpoints, like in case of call to CUE,
 the the RSVP MTP agent needs to negotiate the codec itself, and it tries to
 negotiate g711 codec because by default that is what it has configured, but
 it fails because regions have g729 codec limitation.
 So in MTP RSVP agent after passthrough you must have the codec that you
 use between sites.
 In your case g729, but it could be iLBC maybe or anything else.

 Regards,

 Sent from my iPad

 On 19 May 2011, at 19:18, voice boy voice...@hotmail.com wrote:

  Hi Randl,

 I use this configuration on both sides HQ and BR router that contain the
 CUE:

 dspfarm profile 1 transcode
  codec g711ulaw
  codec g711alaw
  codec g729ar8
  codec g729abr8
  codec g729r8
  codec passthrough
  rsvp
  maximum sessions 3
  associate application SCCP



 hq-rsvp in is HQ-DP [hq-region]
 br-rsvp is in BR-DP [br-region]

 both RSVP agents are in different regions, so they talking G729

 So I don't know why CUE don't take benefit of the transcoder when it need
 or RSVP can't invode transcoder !!!



 --
 Date: Thu, 19 May 2011 10:52:46 -0700
 Subject: Re: [OSL | CCIE_Voice] RSVP with CUE
 From: ill2...@gmail.comill2...@gmail.com
 To: voice...@hotmail.comvoice...@hotmail.com
 CC: ccie_voice@onlinestudylist.comccie_voice@onlinestudylist.com

 Take into consideration that the RSVP provisioning is only happening on
 the route path between the RSVP agents. The CUE itself is outside of the
 RSVP path.

 Perhaps what could matter is the codec chosen between the RSVP agents.
 Whats the region configured between the RSVP agents? Make sure it is G729.

 What are you using for RSVP agent on HQ? mtp or transcoder? share the
 config to check.

 2011/5/19 voice boy  voice...@hotmail.comvoice...@hotmail.com

  Hi,

 I'm testing RSVP between 2-ites using RSVP locations
 It work good and AAR route the call as expected
 Also TEHO calls consume from the RSVP bw

 Only the problem is with the CUE at remote site that i think cause it use
 G711u

 I was using MTP for RSVP , then when the call forwarded to CUE,, it give
 not enough bw , call rerouting ,, although there still bw in rsvp
 I decide to use transcoder as rsvp profile to solve CUE codec issue, But
 when it is forwarded to CUE, nothing occue 

Re: [OSL | CCIE_Voice] UCME/CUE

2011-05-29 Thread Cristobal Priego
clear the line
do a show users
and the connection to cue might be in use

once you know which line you're using
do a

clear line X

and then try to get to cue again

2011/5/29 Randall Crumm rrcr...@yahoo.com

 HI,
 I'm having a weird issue. (see snipits below)

 I configured mt CME for CUE(not telephony-service yet) and I cannot ser ser
 0/0 ses to it
 I can ping 10.10.202.2 from hq rtr
 The cli just has a blank like

 Anyone have an ideas or run into this?

 Thanks,
 Randall


 interface Service-Engine0/0
  ip unnumbered Vlan400
  service-module ip address 10.10.202.2 255.255.255.0
  service-module ip default-gateway 10.10.202.1

 ip route 10.10.202.2 255.255.255.255 Service-Engine0/0


 interface Vlan400
  ip address 10.10.202.1 255.255.255.0



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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

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For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] uccx prompts

2011-05-29 Thread Cristobal Priego
i re-configured my script using the default icd.aef
and eveyrhing worked great

thanks

2011/5/29 Miron Kobelski findko...@gmail.com

 I'd check ip routing. rtp from uccx may be routed to the gw, but not to the
 ip phones?
  On May 29, 2011 3:50 AM, Cristobal Priego cristobalpri...@gmail.com
 wrote:

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Re: [OSL | CCIE_Voice] CUE integrated with UCM

2011-05-29 Thread Cristobal Priego
thank you

2011/5/29 Vik Malhi vma...@ipexpert.com

 If you stop the UCM service this is the behavior. If you shut down the WAN
 interface then the CTI RP/P will re-register.

 --
 Vik Malhi – CCIE #13890
 Managing Partner / Instructor - IPexpert, Inc.
 Mailto: *vma...@ipexpert.com
 *Telephone: +1.810.326.1444 ext 420
 Fax: +1.810.454.0130
 Live Assistance, Please visit: www.ipexpert.com/chat *
 http://www.ipexpert.com/chat*

 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand,
 Audio Tools, Online Hardware Rental and Classroom Training for the Cisco
 CCIE (RS, Voice, Wireless, Security  Service Provider) certification(s)
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 www.ipexpert.com/communities *http://www.ipexpert.com/communities*  and
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 From: Cristobal Priego cristobalpri...@gmail.com
 Date: Sat, 28 May 2011 19:06:45 -0700
 To: OSL Group ccie_voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice] CUE integrated with UCM

 hello i'm working on workbook 2 lab 10

 i have the cue configured, registered with ucm, and everything works great
 when i go into srst mode, everything works great
 however when i failback to UCM  my cti ports, cti route point doesn't
 register automatically, unless i reload CUE
 is this expected ?
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Re: [OSL | CCIE_Voice] Proctorlabs SRST ACLs

2011-05-29 Thread Cristobal Priego
cool !!
thank you
i'll try them today

2011/5/28 adam compton com...@gmail.com

 For any body who needs them, here are some acls I use to test full SRST
 functionality

 BR1
 access-list 150 deny ip 10.10.201.0 0.0.0.255 any
 access-list 150 deny ip host 10.10.110.2 any
 access-list 150 deny ip host 192.168.10.18 any(local phone ip address)
 access-list 150 permit ip any any

 HQ:
 access-list 150 deny ip 10.10.200.0 0.0.0.255 any
 access-list 150 deny ip host 10.10.110.1 any
 access-list 150 deny ip host 192.168.10.18 any(local phone ip address)
 access-list 150 permit ip any any

 BR2:
 access-list 150 deny ip 10.10.202.0 0.0.0.255 any
 access-list 150 deny ip host 10.10.110.3 any
 access-list 150 deny ip host 192.168.10.18 any(local phone ip address)
 access-list 150 permit ip any any

 apply as follows:
 int fa0/0.30
  ip access-group 150 out


 You will need to bind MGCP to the loopback interface, but that's required
 half the time anyway.

 Adam Compton

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Re: [OSL | CCIE_Voice] SRST mode auto-provision

2011-05-29 Thread Cristobal Priego
you will have to implemente access list for SRST to work properly
or create a new ccm-group with the pub server only
assign the ccm-group to the proper device pool
and when you're ready to test srst
just stop the ccm service on the pub

2011/5/29 Randall Crumm rrcr...@yahoo.com

 HI,
 Just wondering if it should make a difference if I am running a HW VPN
 because my phones will not register in SRST mode.
 I tried several ways to put br1 rtr in srst no luck.

 I've even manually configured the rtr with ephones and ephone-dn's

 Everything looks configured correctly on CUCM too.



 snipit..

 telephony-service
  srst mode auto-provision all
  max-ephones 3
  max-dn 3
  ip source-address 10.10.110.2 port 2000

 Thanks,
 Randall




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

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Re: [OSL | CCIE_Voice] SRST mode auto-provision

2011-05-29 Thread Cristobal Priego
on your phone check the network settings do they have an SRST reference ?

2011/5/29 Rrcrumm rrcr...@yahoo.com

 I did the latter but with the sub and it did not work

 Randall

 Sent from my iPhone

 On May 29, 2011, at 6:16 PM, Cristobal Priego cristobalpri...@gmail.com
 wrote:

 you will have to implemente access list for SRST to work properly
 or create a new ccm-group with the pub server only
 assign the ccm-group to the proper device pool
 and when you're ready to test srst
 just stop the ccm service on the pub

 2011/5/29 Randall Crumm  rrcr...@yahoo.comrrcr...@yahoo.com

 HI,
 Just wondering if it should make a difference if I am running a HW VPN
 because my phones will not register in SRST mode.
 I tried several ways to put br1 rtr in srst no luck.

 I've even manually configured the rtr with ephones and ephone-dn's

 Everything looks configured correctly on CUCM too.



 snipit..

 telephony-service
  srst mode auto-provision all
  max-ephones 3
  max-dn 3
  ip source-address 10.10.110.2 port 2000

 Thanks,
 Randall




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Re: [OSL | CCIE_Voice] SRST mode auto-provision

2011-05-29 Thread Cristobal Priego
in the lab you can shut down the WAN interface


you can't do that with proctorlabs

2011/5/29 Randall Crumm rrcr...@yahoo.com

 yes.
 I'm going to try tomorrow

 I just need to know the bestway to do this in the lab.

 Randall




 --
 *From:* Cristobal Priego cristobalpri...@gmail.com
 *To:* Rrcrumm rrcr...@yahoo.com
 *Cc:* Online Study ccie_voice@onlinestudylist.com
 *Sent:* Sunday, May 29, 2011 6:23 PM
 *Subject:* Re: [OSL | CCIE_Voice] SRST mode auto-provision

 on your phone check the network settings do they have an SRST reference ?

 2011/5/29 Rrcrumm rrcr...@yahoo.com

 I did the latter but with the sub and it did not work

 Randall

 Sent from my iPhone

 On May 29, 2011, at 6:16 PM, Cristobal Priego cristobalpri...@gmail.com
 wrote:

 you will have to implemente access list for SRST to work properly
 or create a new ccm-group with the pub server only
 assign the ccm-group to the proper device pool
 and when you're ready to test srst
 just stop the ccm service on the pub

 2011/5/29 Randall Crumm  rrcr...@yahoo.comrrcr...@yahoo.com

 HI,
 Just wondering if it should make a difference if I am running a HW VPN
 because my phones will not register in SRST mode.
 I tried several ways to put br1 rtr in srst no luck.

 I've even manually configured the rtr with ephones and ephone-dn's

 Everything looks configured correctly on CUCM too.



 snipit..

 telephony-service
  srst mode auto-provision all
  max-ephones 3
  max-dn 3
  ip source-address 10.10.110.2 port 2000

 Thanks,
 Randall




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

 Are you a CCNP or CCIE and looking for a job? Check out
 http://www.PlatinumPlacement.comwww.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] auto qos voip trust fr-atm

2011-05-29 Thread Cristobal Priego
hello all

while doing a mock lab
 i needed to configure LLQ, MLP LFI cRTP between BR1 and HQ, the bw was 384

when i issued auto qos voip trust fr-atm on both interfaces, after setting
the BW on the inferfaces
everything was great, no problem

however once i reloaded the BR1 router
i lost connectivity to the HQ site

i reloaded HQ router and still no connectivity


so i reverted my pod

and resstarted again only with QOS

i set the BW
and ran the command again
without any customization

and the same issue happened, where i lost connectivity from BR1 to HQ

do you guys know how to fix this ?

thank you
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