Re: [OSL | CCIE_Voice] BLF Speed Dials

2012-01-17 Thread datucha123 datucha123
Thank you George

On Mon, Jan 16, 2012 at 8:09 PM, George Goglidze gogli...@gmail.com wrote:

 Hi Datucha,

 Actually for CUPS you need only line association. you do not need to
 specify owner user id.

 Misread the question a little bit initially.

 the owner user id on the phone is for SNR.

 Cheers,

 On Mon, Jan 16, 2012 at 4:01 PM, George Goglidze gogli...@gmail.comwrote:

 correctisimo  :-)



  On Mon, Jan 16, 2012 at 3:25 PM, datucha123 datucha123 
 datucha...@gmail.com wrote:

  When configuring the simple BLF Speed Dial, we need to configure the
 Subscribe CSS for watching Device. So that it could the the Watched DN.

 But, the Owner User ID,  Line Association with End User and other kinds
 of associations are not required for this BLF, right?

 Even for Call List Presence.



 The Owner User ID and Line Association with End User, along with License
 Capability Assignements are required only for CUPS Presence method.

 Is it correct?

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Re: [OSL | CCIE_Voice] Cant connect to Sub

2012-01-17 Thread datucha123 datucha123
Also check the DB Replication Status

On Tue, Jan 17, 2012 at 6:17 AM, Ashraf Ayyash ash.ayy...@gmail.com wrote:

 Hello Errol ,

 This issue usually mean you have incorrect Host/ processNode files in
 your PUB or the Sub and so the communication is broken between the PUB
 and the SUB and you have to find what table we have corrupted and fix
 it , or you have a big timing difference reported by your NTP ,

 can you check what service you have running on both pub and sub
 specially the A Cisco DB and let me know what you will get .

 Ash

 On Mon, Jan 16, 2012 at 5:15 PM, Errol Abrahams eabraham2...@gmail.com
 wrote:
 
 
  Hi All,
 
  I had a problem with my VMWARE Server and I had to rebuilt the system
 from
  scratch. I have reloaded PUB,SUB,CUPS,CUC and CUCCX and all virtual
  addresses are pingable. When I activate the services for the PUB from the
  Cisco Unified Serviceability screen, it worked. But, when I try to access
  the SUB from same screen then it displays'Connection to the Server
 cannot be
  established(unable to access Remote Node).
 
  Has anybody had a problem like this and how can I fix this problem. Your
  help is appreciated..thnx.
 
  Chhers
 
  EA
 
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[OSL | CCIE_Voice] Voicemail access during AAR

2012-01-17 Thread Vega Wong
Hi experts

I am working on Vol 2. Lab 7 and try to fully understand the topic of voice 
mail access during AAR, but I am struggling to have a clear picture. Here is 
the setup

HQ GW - MGCP
BR GW - H323
HQ Phones, and BR Phones are both SCCP. The line on both phone been assigned to 
the same AAR group. Only HQ phone assigned with AAR group and CSS AAR on the 
device level. 
voicemail for BR is on CUE which is integrated with CUCM using CTI route point.
All integration is configured and tested ok

AAR between the phones are working, the HQ Phone will reroute out to PSTN to 
reach BR Phones when there is not enough bandwidth. 
Also, HQ Phone can directly dial in to the voicemail pilot (reroute out to 
PSTN) and reach the CUE log in. 
BR phone can press the message button and reach the sign-in prompt for CUE 
(only asking for PIN)

However, when HQ phone calls BR phone and BR Phone doesnt pick up, just as it 
should transfer to voicemail, I get fast busy tone on HQ Phone. 

I am trying to understand, at this instance, I am still using the AAR CSS on HQ 
Phone to reach the voice mail pilot right? I imagine if HQ Phone can 
successfully call to voicemail directly during AAR, it shouldnt be different 
when it is transferred by BR Phone?

Also, I am trying to understand when do we need to assign AAR group and AAR CSS 
to the gateway? and why?

Please help

Thanks in advance

Vega
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[OSL | CCIE_Voice] SRST problem: Phones in srst mode have a different configuration than they were registered to CUCM

2012-01-17 Thread The Masterplan
Hello,

I have a problem regarding srst. The 2811 router than now is a srst was a
acting as a cme in past in a demo lab and the 7960 phone was registered to
it with extension . Now, the 7960 phone is registered in UCM with
extension 5001 and the 2811 router is configured in telephony service srst
mode. The problem is that although the old configuration of the router was
erased, when it goes in srst fallback mode, the 7960 gets extension 
instead of 5001 and the command show telephony-service ephone shows the
specified phone with message:This is an srst fallback phone.

Thank you for your answer
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[OSL | CCIE_Voice] Presence on CUPC

2012-01-17 Thread datucha123 datucha123
Hello,

Based on my testings, CUPC support Presence Status change in its Contacts
only when they are imported from AD.
Manually created contacts does not support Presence Status Change.

Is it correct?
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Re: [OSL | CCIE_Voice] Voicemail access during AAR

2012-01-17 Thread Farkas Péter
- Can AAR CSS assigned to HQ DN/Device reach Voicemail Pilot over PSTN?

GW AAR CSS is used when there is an incoming call from PSTN to BR IP phone and 
the phone has call forwarding set to voicemail.  Also VM is in other CAC 
location that has insufficient bandwidth which triggers AAR function. Call is 
terminated on gw and will do forwarding instead of IP Phone at the branch. Here 
gw's AAR CSS will route the call toward the Voicemail's pilot.

Peter

- Original Message -
From: Vega Wong vega2...@yahoo.com.au
Date: Tuesday, January 17, 2012 2:01 pm
Subject: [OSL | CCIE_Voice] Voicemail access during AAR
To: ccie_voice@onlinestudylist.com


 Hi experts
  
  I am working on Vol 2. Lab 7 and try to fully understand the topic of voice 
 mail access during 
 AAR, but I am struggling to have a clear picture. Here is the setup
  
  HQ GW - MGCP
  BR GW - H323
  HQ Phones, and BR Phones are both SCCP. The line on both phone been assigned 
 to the same AAR 
 group. Only HQ phone assigned with AAR group and CSS AAR on the device level. 
  voicemail for BR is on CUE which is integrated with CUCM using CTI route 
 point.
  All integration is configured and tested ok
  
  AAR between the phones are working, the HQ Phone will reroute out to PSTN to 
 reach BR Phones 
 when there is not enough bandwidth. 
  Also, HQ Phone can directly dial in to the voicemail pilot (reroute out to 
 PSTN) and reach the 
 CUE log in. 
  BR phone can press the message button and reach the sign-in prompt for CUE 
 (only asking for PIN)
  
  However, when HQ phone calls BR phone and BR Phone doesnt pick up, just as 
 it should transfer 
 to voicemail, I get fast busy tone on HQ Phone. 
  
  I am trying to understand, at this instance, I am still using the AAR CSS on 
 HQ Phone to reach 
 the voice mail pilot right? I imagine if HQ Phone can successfully call to 
 voicemail directly 
 during AAR, it shouldnt be different when it is transferred by BR Phone?
  
  Also, I am trying to understand when do we need to assign AAR group and AAR 
 CSS to the 
 gateway? and why?
  
  Please help
  
  Thanks in advance
  
  Vega 
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 visit www.ipexpert.com
  
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[OSL | CCIE_Voice] CUE error

2012-01-17 Thread study buddy
Hi

While accessing my voicemail via VoiceView, I get the following error when
I click on listen

Unknown error code {0}. Report this error to your system administrator

Has anyone run into this error before? I tried several things but I cant
get over this error

TR
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Re: [OSL | CCIE_Voice] Presence on CUPC

2012-01-17 Thread George Goglidze
yes it's possible, as long as it's correctly done:

http://www.cisco.com/en/US/products/ps6837/products_tech_note09186a00808a2b0d.shtml

Unable to View Directory Information Problem

With CUPC 8.x, you are unable to view directory information, you are unable
to see the status/Presence information for any of the added users, search
does not show any results, and you are unable to initiate a chat session
with another user.
Solution

This issue might occur due to an LDAP integration issue. If LDAP is not
configured, you cannot search any contact.

The reason for the user not able to chat with a manually added user is that
when manually adding the user if the proper domain name is not provided,
then CUPS will not be able to validate that user with CallManager. So, if
you are not using LDAP Integration, then it is suggested to add the user
through the CUPS User Option Page to resolve the issue. Perform these steps:

   1.

   Delete the user from CUPC.
   2.

   Add the user from the User Option Page:
   1.

  Enter the CUPS IP address, and click *User Option*.
  2.

  Enter the user ID and credentials of the end user.
   3.

   Now, click on *Settings  Contacts  Add New* and then type the user ID
   of the user you want to add.

Once these steps are complete, you should be able to see the user ID
populated in the CUPC, and you should be able to chat with these users.


On Tue, Jan 17, 2012 at 2:52 PM, datucha123 datucha123 datucha...@gmail.com
 wrote:

 Hello,

 Based on my testings, CUPC support Presence Status change in its Contacts
 only when they are imported from AD.
 Manually created contacts does not support Presence Status Change.

 Is it correct?

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[OSL | CCIE_Voice] BAT File on Excel 2011 for Mac

2012-01-17 Thread Emanuel Damasceno
Hello Experts,

Does anybody ever made the BAT file work just fine on Excel 2011? I am
asking this because it is a pain to open my VMWare Fusion, load Windows
with Excel 2007 every time I need to do something with the BAT file.

Any ideas?
*Emanuel Damasceno*
CCNP Voice
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[OSL | CCIE_Voice] Unity Connection Integration

2012-01-17 Thread Ken Wyan
Hi,

When adding a new port group to unity connection , it prompts to enter
only one CUCM ip address. Should we essentially provide CUCM Publisher here?

My Integration is CUCM Sub First  CUCM Pub Second as we expect to do in
CCIE Exam.

When adding a phone system we can provide Subscriber (first)  Publisher
(second) in order (AXL Server)

When adding a new portgroup , it asks for only one ip address. The ip
address I provide is displayed under Port Group Configuration (CUCM  TFTP)
 we can later add another CUCM address also.

Any guideline?
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Re: [OSL | CCIE_Voice] SRST problem: Phones in srst mode have a different configuration than they were registered to CUCM

2012-01-17 Thread Ken Wyan
Did you enter no create cnf-files  create cnf-files on CME Router ?

Which srst mode are you running? srst mode auto provision all | dn | none ?
Better to post full telephony-service configuration here.

(In SRST phone may be downloading previous xml configuration file from the
router. Delete it from flash if so)



On Tue, Jan 17, 2012 at 6:53 PM, The Masterplan winmasterp...@gmail.comwrote:

 Hello,

 I have a problem regarding srst. The 2811 router than now is a srst was a
 acting as a cme in past in a demo lab and the 7960 phone was registered to
 it with extension . Now, the 7960 phone is registered in UCM with
 extension 5001 and the 2811 router is configured in telephony service srst
 mode. The problem is that although the old configuration of the router was
 erased, when it goes in srst fallback mode, the 7960 gets extension 
 instead of 5001 and the command show telephony-service ephone shows the
 specified phone with message:This is an srst fallback phone.

 Thank you for your answer

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Re: [OSL | CCIE_Voice] Presence on CUPC

2012-01-17 Thread Mohd Baqari
No its supporting any kind of users (manual or AD imported). Also, it can be 
another CUPC or IPPM.

Regards,
Mohammed Al Baqari

Sent from my iPhone

On Jan 17, 2012, at 5:52 PM, datucha123 datucha123 datucha...@gmail.com wrote:

 Hello,
  
 Based on my testings, CUPC support Presence Status change in its Contacts 
 only when they are imported from AD.
 Manually created contacts does not support Presence Status Change.
  
 Is it correct?
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Re: [OSL | CCIE_Voice] SRST problem: Phones in srst mode have a different configuration than they were registered to CUCM

2012-01-17 Thread Mohd Baqari
Try to do factory reset for the phone and then test again.

Regards,
Mohammed Al Baqari

Sent from my iPhone

On Jan 17, 2012, at 5:23 PM, The Masterplan winmasterp...@gmail.com wrote:

 Hello,
 
 I have a problem regarding srst. The 2811 router than now is a srst was a 
 acting as a cme in past in a demo lab and the 7960 phone was registered to it 
 with extension . Now, the 7960 phone is registered in UCM with extension 
 5001 and the 2811 router is configured in telephony service srst mode. The 
 problem is that although the old configuration of the router was erased, when 
 it goes in srst fallback mode, the 7960 gets extension  instead of 5001 
 and the command show telephony-service ephone shows the specified phone with 
 message:This is an srst fallback phone.
 
 Thank you for your answer 
 ___
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Re: [OSL | CCIE_Voice] Unity Connection Integration

2012-01-17 Thread Hough, Earl
I guess it would determine what your requirements are.  If, for example, the 
global requirements were that your subscriber were to be the primary server for 
all call processing, you might want to take that into account when choosing 
whether to only use one server or two and which should be the primary and 
secondary.

Earl Hough
CCIE #16508 (RS/Security/Voice)

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ken Wyan
Sent: Tuesday, January 17, 2012 11:42 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Unity Connection Integration

Hi,

When adding a new port group to unity connection , it prompts to enter  only 
one CUCM ip address. Should we essentially provide CUCM Publisher here?

My Integration is CUCM Sub First  CUCM Pub Second as we expect to do in CCIE 
Exam.

When adding a phone system we can provide Subscriber (first)  Publisher 
(second) in order (AXL Server)

When adding a new portgroup , it asks for only one ip address. The ip address I 
provide is displayed under Port Group Configuration (CUCM  TFTP)  we can 
later add another CUCM address also.

Any guideline?

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

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] Who is passing and how?

2012-01-17 Thread cciev wannabe
I do not want to mis-inform at anybody. Since I have been through the pain
and sufferings I want other to have happiness.

Let me get to the point, the LAB is very passable if you can configure with
very high speed and know land mines.
Again, I could be wrong but this is how I felt. Doing fast outweight
knowing more. Knowing all technologies in and out is good but you can still
fail if you don't have speed.
So, please practice your speed. It is very very important.

About land mines, I am not sure how to prepare. There are some who posted
already on this and other forums on trick questions.
I agreed that most of the questions are very reasonable but there are some
which isn't best practice nor a real world problems.
So sorry that I can't give you advice on how to prepare but the good thing
is there are only a couple crazy questions par LAB.

Good luck guys!

On Sat, Jan 14, 2012 at 10:31 PM, Edgar Feliz ejzi...@gmail.com wrote:

 Well I know the proctor I had at RTP would not pass it, I had a issue with
 something and he had no clue, proctor's use script they are like the
 telemarketers that call you to bother you.

 My thought on your main question is this and I do not know how many times
 you have taken the lab, I have taken it once, be prepared. I know looking
 back I was not prepared the first time not like I thought I was, I will be
 going again soon and I feel better prepared for this attempt will I pass
 hard to say but I will try, there seems to be a lot of variables that no
 one can explain. The grading of the questions I have heard different things
 from people, proctors, and instructor saying things are like this but they
 are saying different things, do you get partial credit or not

 Someone I know at Cisco that has taken the lab 3 times, last time finished
 in 6 hours double checked everything said everything was 100% verified and
 failed worse then her first attempthow can that be?

 Concentrate on what is giving you trouble why work on the stuff you know
 as much as the stuff you don't, work harder on what you don't have a good
 grasp on. I have a lot of friends that have taken the lab 3, 4 and more
 times and some have stopped because it's expensive and time consuming. I
 always tell people the reward for passing is a lifetime of work.

 E

 On Fri, Jan 13, 2012 at 9:44 PM, cciev wannabe cciev.wann...@gmail.comwrote:

 Hi All,

 My questions to you especially to those who has taken the lab: how to
 pass the lab?
 I took a few time already and failed each time. It looks very passable
 after reading all the questions.
 But for me, time isn't enough to troubleshoot and do the configurations.

 Not to insult anybody but I sincerely wonder can even proctor pass these
 lab if they haven't seen it?

 Any thoughts?

 Thanks,

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Re: [OSL | CCIE_Voice] Unity Connection Integration

2012-01-17 Thread datucha123 datucha123
That parameter actually does not effect anything.

On Tue, Jan 17, 2012 at 9:22 PM, Hough, Earl
earl.ho...@pcmallservices.comwrote:

  I guess it would determine what your requirements are.  If, for example,
 the global requirements were that your subscriber were to be the primary
 server for all call processing, you might want to take that into account
 when choosing whether to only use one server or two and which should be the
 primary and secondary.

 ** **

 Earl Hough

 CCIE #16508 (RS/Security/Voice)

 ** **

 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Ken Wyan
 *Sent:* Tuesday, January 17, 2012 11:42 AM
 *To:* ccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] Unity Connection Integration

 ** **

 Hi,

  

 When adding a new port group to unity connection , it prompts to enter
 only one CUCM ip address. Should we essentially provide CUCM Publisher here?
 

  

 My Integration is CUCM Sub First  CUCM Pub Second as we expect to do in
 CCIE Exam.

  

 When adding a phone system we can provide Subscriber (first)  Publisher
 (second) in order (AXL Server)

  

 When adding a new portgroup , it asks for only one ip address. The ip
 address I provide is displayed under Port Group Configuration (CUCM  TFTP)
  we can later add another CUCM address also.

  

 Any guideline?

  

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

 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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[OSL | CCIE_Voice] Gatekeeper VIAZONE - Could not find an IPIPGW

2012-01-17 Thread Steven

Hi there,
i got some problems with my viazone (CUBE) at HQ-RTR.
I already checked the Tech prefix match and it seems to succeed.
But i'm clueless how to debug/resolve the Could not find an 
IPIPGW-problem.

I also checked the dial-peers on HQ-RTR.

Any help appreciated.

Regards,
Steven


! *** Begin tech details:

HQ-RTR#debug gatekeeper main 10! tried to call from HQ (5002) to 
BR2 (3006)


Jan 17 19:56:37.786: ////GK/gk_process: 
QUEUE_EVENT (minor 0) wakeup
Jan 17 19:56:37.786: ////GK/gk_rassrv_arq: 
arqp=0x48F0C08C,crv=0x7, answerCall=0
Jan 17 19:56:37.786: ////GK/gk_rassrv_sep_arq: 
ARQ Didn't use GK_AAA_PROC
Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/gk_dns_query: No 
Name servers
Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_get_addrinfo: 
(3006) Tech-prefix match failed.
Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_get_addrinfo: 
(3006) Matched zone prefix 3 and remainder 006
Jan 17 19:56:37.786: 
////GK/gk_rassrv_get_ingress_network: returning 
default ingress network = 1
Jan 17 19:56:37.786: 
//80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: about to check 
the source side, src_zonep=0x4793079C
Jan 17 19:56:37.786: 
//80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: matched zone 
is UCM, and z_invianamelen=0
Jan 17 19:56:37.786: 
//80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: about to check 
the destination side, dst_zonep=0x47930A08
Jan 17 19:56:37.786: 
//80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: matched zone 
is UCME, and z_outvianamelen=4
Jan 17 19:56:37.786: 
//80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone  and 
z_outvianamep=CUBE
Jan 17 19:56:37.786: 
//80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: Received ARQ 
for a zone (UCME) that has an outviazone (CUBE) specified.  Pick an 
IP-IP gateway in that viazone.
Jan 17 19:56:37.786: 
////GK/gk_gw_select_ipipgw_random: zonep: 
0x47930C74, tpp: 0x0, current_endpt: 0
Jan 17 19:56:37.786: 
////GK/gk_gw_select_ipipgw_random: Selecting any 
IPIPGW. qelemp.head=0x46F0FE88, use_count=1, current_endpt=0
Jan 17 19:56:37.786: 
////GK/gk_gw_select_ipipgw_random: 
qelemp=0x46F0FE88, loop_count=0
Jan 17 19:56:37.786: 
////GK/gk_gw_select_ipipgw_random: Examining 
tgwp 0x46F253E0, g_supp_prots: 0x50 qelemp: 0x46F0FE88, loop_count: 1
Jan 17 19:56:37.786: 
////GK/gk_gw_select_ipipgw_random: Searched 
through the entire gateway list. loop_count=0
Jan 17 19:56:37.786: 
////GK/gk_gw_select_ipipgw_random: Could not 
find an IPIPGW.
Jan 17 19:56:37.786: 
//80AC69450700/80AC69450700/GK/rassrv_get_addrinfo(3006): Viazone 
gateway selection failed for zone CUBE
Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/gk_rassrv_sep_arq: 
rassrv_get_addrinfo() failed (return code = 0x805)



HQ-RTR#show gatekeeper gw-type-prefix
GATEWAY TYPE PREFIX TABLE
=
Prefix: 1#*(Default gateway-technology)
  Zone CUBE master gateway list:
10.10.110.1:1720 HQ-RTR
  Zone UCM master gateway list:
10.10.210.10:44248 gk-trunk_1
10.10.210.11:36641 gk-trunk_2

Prefix: 3#*
  Zone UCME master gateway list:
10.10.110.3:1720 BR2-RTR


HQ-RTR#show gatekeeper zone prefix
  ZONE PREFIX TABLE
  =
GK-NAME   E164-PREFIX
---   ---
UCME  3...
UCM   5...


HQ-RTR#show running-config interface loopback 0
interface Loopback0
 ip address 10.10.110.1 255.255.255.255
 h323-gateway voip interface
 h323-gateway voip id CUBE ipaddr 10.10.110.1 1719
 h323-gateway voip h323-id HQ-RTR
 h323-gateway voip tech-prefix 1#


 HQ-RTR#show running-config | section gatekeeper
gatekeeper
 zone local UCM ipexpert.com
 zone local UCME ipexpert.com outvia CUBE
 zone local CUBE ipexpert.com
 zone prefix UCME 3...
 zone prefix UCM 5...
 gw-type-prefix 1#* default-technology
 no shutdown
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Re: [OSL | CCIE_Voice] Unity Connection Integration

2012-01-17 Thread datucha123 datucha123
You can set it to any server you like initially, but when you go to Server
configuration in Port Group page, there you have to set them as necessary
and required.

On Wed, Jan 18, 2012 at 12:18 AM, datucha123 datucha123 
datucha...@gmail.com wrote:

 That parameter actually does not effect anything.

   On Tue, Jan 17, 2012 at 9:22 PM, Hough, Earl 
 earl.ho...@pcmallservices.com wrote:

I guess it would determine what your requirements are.  If, for
 example, the global requirements were that your subscriber were to be the
 primary server for all call processing, you might want to take that into
 account when choosing whether to only use one server or two and which
 should be the primary and secondary.

 ** **

 Earl Hough

 CCIE #16508 (RS/Security/Voice)

 ** **

 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Ken Wyan
 *Sent:* Tuesday, January 17, 2012 11:42 AM
 *To:* ccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] Unity Connection Integration

 ** **

 Hi,

  

 When adding a new port group to unity connection , it prompts to enter
 only one CUCM ip address. Should we essentially provide CUCM Publisher here?
 

  

 My Integration is CUCM Sub First  CUCM Pub Second as we expect to do in
 CCIE Exam.

  

 When adding a phone system we can provide Subscriber (first)  Publisher
 (second) in order (AXL Server)

  

 When adding a new portgroup , it asks for only one ip address. The ip
 address I provide is displayed under Port Group Configuration (CUCM  TFTP)
  we can later add another CUCM address also.

  

 Any guideline?

  

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

 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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 visit www.ipexpert.com

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 www.PlatinumPlacement.com http://www.platinumplacement.com/



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Re: [OSL | CCIE_Voice] CUE error

2012-01-17 Thread datucha123 datucha123
First of all ensure that the Authentication URL is set correctly.

Also you can try to reset the CUE module - just in case.

On Tue, Jan 17, 2012 at 6:30 PM, study buddy studybudd...@gmail.com wrote:

 Hi

 While accessing my voicemail via VoiceView, I get the following error when
 I click on listen

 Unknown error code {0}. Report this error to your system administrator

 Has anyone run into this error before? I tried several things but I cant
 get over this error

 TR

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[OSL | CCIE_Voice] Study Partner wanted

2012-01-17 Thread Duncan Hamilton-Walker
Hi,

 

Looking for a study partner... am working towards taking my first attempt of
the Lab. 

Am in the Geneva/French alps area 

 

Thanks

D

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[OSL | CCIE_Voice] redundancy for SIP dialpeers

2012-01-17 Thread Juan Lopez
When configuring 2 SIP dialpeers for redundancy, together with:
sip-ua
retry invite 2

This should generate in total 3 INVITES sent to the primary UCM via the
first dialpeer, before going over to the second sip dialpeer, right?
Doing debug ccsip messages only shows 1 invite sent to the primary, and
then 1 invite to the secondary.
am I missing something?
thanks, Juan
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[OSL | CCIE_Voice] GK routed calls into UCM

2012-01-17 Thread Juan Lopez
When 2 sites BR1 and BR2 call into the UCM phones via gatekeeper, these 2
sites send H225 call setup to the UCM.
In this case there is no way to choose a codec based on calling (= GK trunk
on the UCM) and called endpoint for *both situations*?

example:
GK-trunk in DP BR1calls from BR1 GW will have g711 for calls to BR1
phones. However, calls to BR2 phones that enter the BR2 gw will then use
the g729 codec, instead of the intra-site g711 codec
Setting the GK trunk in a device pool that speaks g711 with all is not good
either, as this would mean a xfer from BR1 to BR2 would create a g711 call
between BR1 GW and BR2 phone...
Am I missing a valid solution ?

Another question:
For GK call routing, it is not necessary for the UCM to know the h323
source address of both remote branches, only the gatekeeper needs to be
defined. How does UCM know it can accept H225 messages sourced by both
gateways - as they are not defined? Is there a 'GK' flag set in the h225
setup that triggers the UCM to consult the GKand therefore accept the call
from the undefined h323 gateways?
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Re: [OSL | CCIE_Voice] Gatekeeper VIAZONE - Could not find an IPIPGW

2012-01-17 Thread Boris
Hi Steve,

Do you have This in your config? 

Voice service voip
 Allow h323 to h323

If not, add it and do no gateway/gateway

Your cube should appear as H323 type in show gatek end.

Sent from my mobile device, sorry for typos.
---
Regards
Boris

On 18/01/2012, at 7:22, Steven forum.ccie.onlinestudyl...@nocer.net wrote:

 Hi there,
 i got some problems with my viazone (CUBE) at HQ-RTR.
 I already checked the Tech prefix match and it seems to succeed.
 But i'm clueless how to debug/resolve the Could not find an IPIPGW-problem.
 I also checked the dial-peers on HQ-RTR.
 
 Any help appreciated.
 
 Regards,
 Steven
 
 
 ! *** Begin tech details:
 
 HQ-RTR#debug gatekeeper main 10! tried to call from HQ (5002) to BR2 
 (3006)
 
 Jan 17 19:56:37.786: ////GK/gk_process: QUEUE_EVENT 
 (minor 0) wakeup
 Jan 17 19:56:37.786: ////GK/gk_rassrv_arq: 
 arqp=0x48F0C08C,crv=0x7, answerCall=0
 Jan 17 19:56:37.786: ////GK/gk_rassrv_sep_arq: ARQ 
 Didn't use GK_AAA_PROC
 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/gk_dns_query: No Name 
 servers
 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_get_addrinfo: 
 (3006) Tech-prefix match failed.
 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_get_addrinfo: 
 (3006) Matched zone prefix 3 and remainder 006
 Jan 17 19:56:37.786: 
 ////GK/gk_rassrv_get_ingress_network: returning 
 default ingress network = 1
 Jan 17 19:56:37.786: 
 //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: about to check the 
 source side, src_zonep=0x4793079C
 Jan 17 19:56:37.786: 
 //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: matched zone is 
 UCM, and z_invianamelen=0
 Jan 17 19:56:37.786: 
 //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: about to check the 
 destination side, dst_zonep=0x47930A08
 Jan 17 19:56:37.786: 
 //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: matched zone is 
 UCME, and z_outvianamelen=4
 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone 
  and z_outvianamep=CUBE
 Jan 17 19:56:37.786: 
 //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: Received ARQ for a 
 zone (UCME) that has an outviazone (CUBE) specified.  Pick an IP-IP gateway 
 in that viazone.
 Jan 17 19:56:37.786: 
 ////GK/gk_gw_select_ipipgw_random: zonep: 0x47930C74, 
 tpp: 0x0, current_endpt: 0
 Jan 17 19:56:37.786: 
 ////GK/gk_gw_select_ipipgw_random: Selecting any 
 IPIPGW. qelemp.head=0x46F0FE88, use_count=1, current_endpt=0
 Jan 17 19:56:37.786: 
 ////GK/gk_gw_select_ipipgw_random: qelemp=0x46F0FE88, 
 loop_count=0
 Jan 17 19:56:37.786: 
 ////GK/gk_gw_select_ipipgw_random: Examining tgwp 
 0x46F253E0, g_supp_prots: 0x50 qelemp: 0x46F0FE88, loop_count: 1
 Jan 17 19:56:37.786: 
 ////GK/gk_gw_select_ipipgw_random: Searched through 
 the entire gateway list. loop_count=0
 Jan 17 19:56:37.786: 
 ////GK/gk_gw_select_ipipgw_random: Could not find an 
 IPIPGW.
 Jan 17 19:56:37.786: 
 //80AC69450700/80AC69450700/GK/rassrv_get_addrinfo(3006): Viazone gateway 
 selection failed for zone CUBE
 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/gk_rassrv_sep_arq: 
 rassrv_get_addrinfo() failed (return code = 0x805)
 
 
 HQ-RTR#show gatekeeper gw-type-prefix
 GATEWAY TYPE PREFIX TABLE
 =
 Prefix: 1#*(Default gateway-technology)
  Zone CUBE master gateway list:
10.10.110.1:1720 HQ-RTR
  Zone UCM master gateway list:
10.10.210.10:44248 gk-trunk_1
10.10.210.11:36641 gk-trunk_2
 
 Prefix: 3#*
  Zone UCME master gateway list:
10.10.110.3:1720 BR2-RTR
 
 
 HQ-RTR#show gatekeeper zone prefix
  ZONE PREFIX TABLE
  =
 GK-NAME   E164-PREFIX
 ---   ---
 UCME  3...
 UCM   5...
 
 
 HQ-RTR#show running-config interface loopback 0
 interface Loopback0
 ip address 10.10.110.1 255.255.255.255
 h323-gateway voip interface
 h323-gateway voip id CUBE ipaddr 10.10.110.1 1719
 h323-gateway voip h323-id HQ-RTR
 h323-gateway voip tech-prefix 1#
 
 
 HQ-RTR#show running-config | section gatekeeper
 gatekeeper
 zone local UCM ipexpert.com
 zone local UCME ipexpert.com outvia CUBE
 zone local CUBE ipexpert.com
 zone prefix UCME 3...
 zone prefix UCM 5...
 gw-type-prefix 1#* default-technology
 no shutdown
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Re: [OSL | CCIE_Voice] Gatekeeper VIAZONE - Could not find an IPIPGW

2012-01-17 Thread Leslie Meade
and do you have at least one h323 dial-peer on the HQ router



Leslie Meade 


- Original Message -
From: Boris boris.k...@gmail.com
To: Steven forum.ccie.onlinestudyl...@nocer.net
Cc: ccie voice ccie_voice@onlinestudylist.com
Sent: Tuesday, January 17, 2012 1:19:21 PM
Subject: Re: [OSL | CCIE_Voice] Gatekeeper VIAZONE - Could not find an IPIPGW

Hi Steve,

Do you have This in your config? 

Voice service voip
 Allow h323 to h323

If not, add it and do no gateway/gateway

Your cube should appear as H323 type in show gatek end.

Sent from my mobile device, sorry for typos.
---
Regards
Boris

On 18/01/2012, at 7:22, Steven forum.ccie.onlinestudyl...@nocer.net wrote:

 Hi there,
 i got some problems with my viazone (CUBE) at HQ-RTR.
 I already checked the Tech prefix match and it seems to succeed.
 But i'm clueless how to debug/resolve the Could not find an IPIPGW-problem.
 I also checked the dial-peers on HQ-RTR.
 
 Any help appreciated.
 
 Regards,
 Steven
 
 
 ! *** Begin tech details:
 
 HQ-RTR#debug gatekeeper main 10! tried to call from HQ (5002) to BR2 
 (3006)
 
 Jan 17 19:56:37.786: ////GK/gk_process: QUEUE_EVENT 
 (minor 0) wakeup
 Jan 17 19:56:37.786: ////GK/gk_rassrv_arq: 
 arqp=0x48F0C08C,crv=0x7, answerCall=0
 Jan 17 19:56:37.786: ////GK/gk_rassrv_sep_arq: ARQ 
 Didn't use GK_AAA_PROC
 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/gk_dns_query: No Name 
 servers
 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_get_addrinfo: 
 (3006) Tech-prefix match failed.
 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_get_addrinfo: 
 (3006) Matched zone prefix 3 and remainder 006
 Jan 17 19:56:37.786: 
 ////GK/gk_rassrv_get_ingress_network: returning 
 default ingress network = 1
 Jan 17 19:56:37.786: 
 //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: about to check the 
 source side, src_zonep=0x4793079C
 Jan 17 19:56:37.786: 
 //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: matched zone is 
 UCM, and z_invianamelen=0
 Jan 17 19:56:37.786: 
 //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: about to check the 
 destination side, dst_zonep=0x47930A08
 Jan 17 19:56:37.786: 
 //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: matched zone is 
 UCME, and z_outvianamelen=4
 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone 
  and z_outvianamep=CUBE
 Jan 17 19:56:37.786: 
 //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: Received ARQ for a 
 zone (UCME) that has an outviazone (CUBE) specified.  Pick an IP-IP gateway 
 in that viazone.
 Jan 17 19:56:37.786: 
 ////GK/gk_gw_select_ipipgw_random: zonep: 0x47930C74, 
 tpp: 0x0, current_endpt: 0
 Jan 17 19:56:37.786: 
 ////GK/gk_gw_select_ipipgw_random: Selecting any 
 IPIPGW. qelemp.head=0x46F0FE88, use_count=1, current_endpt=0
 Jan 17 19:56:37.786: 
 ////GK/gk_gw_select_ipipgw_random: qelemp=0x46F0FE88, 
 loop_count=0
 Jan 17 19:56:37.786: 
 ////GK/gk_gw_select_ipipgw_random: Examining tgwp 
 0x46F253E0, g_supp_prots: 0x50 qelemp: 0x46F0FE88, loop_count: 1
 Jan 17 19:56:37.786: 
 ////GK/gk_gw_select_ipipgw_random: Searched through 
 the entire gateway list. loop_count=0
 Jan 17 19:56:37.786: 
 ////GK/gk_gw_select_ipipgw_random: Could not find an 
 IPIPGW.
 Jan 17 19:56:37.786: 
 //80AC69450700/80AC69450700/GK/rassrv_get_addrinfo(3006): Viazone gateway 
 selection failed for zone CUBE
 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/gk_rassrv_sep_arq: 
 rassrv_get_addrinfo() failed (return code = 0x805)
 
 
 HQ-RTR#show gatekeeper gw-type-prefix
 GATEWAY TYPE PREFIX TABLE
 =
 Prefix: 1#*(Default gateway-technology)
  Zone CUBE master gateway list:
10.10.110.1:1720 HQ-RTR
  Zone UCM master gateway list:
10.10.210.10:44248 gk-trunk_1
10.10.210.11:36641 gk-trunk_2
 
 Prefix: 3#*
  Zone UCME master gateway list:
10.10.110.3:1720 BR2-RTR
 
 
 HQ-RTR#show gatekeeper zone prefix
  ZONE PREFIX TABLE
  =
 GK-NAME   E164-PREFIX
 ---   ---
 UCME  3...
 UCM   5...
 
 
 HQ-RTR#show running-config interface loopback 0
 interface Loopback0
 ip address 10.10.110.1 255.255.255.255
 h323-gateway voip interface
 h323-gateway voip id CUBE ipaddr 10.10.110.1 1719
 h323-gateway voip h323-id HQ-RTR
 h323-gateway voip tech-prefix 1#
 
 
 HQ-RTR#show running-config | section gatekeeper
 gatekeeper
 zone local UCM ipexpert.com
 zone local UCME ipexpert.com outvia CUBE
 zone local CUBE ipexpert.com
 zone prefix UCME 3...
 zone prefix UCM 5...
 gw-type-prefix 1#* default-technology
 no shutdown
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Re: [OSL | CCIE_Voice] Gatekeeper VIAZONE - Could not find an IPIPGW

2012-01-17 Thread amit batra
If the Voice service Voip commands are all configured...A restart has always 
helped me ...





 From: Boris boris.k...@gmail.com
To: Steven forum.ccie.onlinestudyl...@nocer.net 
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com 
Sent: Wednesday, January 18, 2012 10:19 AM
Subject: Re: [OSL | CCIE_Voice] Gatekeeper VIAZONE - Could not find an IPIPGW
 
Hi Steve,

Do you have This in your config? 

Voice service voip
Allow h323 to h323

If not, add it and do no gateway/gateway

Your cube should appear as H323 type in show gatek end.

Sent from my mobile device, sorry for typos.
---
Regards
Boris

On 18/01/2012, at 7:22, Steven forum.ccie.onlinestudyl...@nocer.net wrote:

 Hi there,
 i got some problems with my viazone (CUBE) at HQ-RTR.
 I already checked the Tech prefix match and it seems to succeed.
 But i'm clueless how to debug/resolve the Could not find an IPIPGW-problem.
 I also checked the dial-peers on HQ-RTR.
 
 Any help appreciated.
 
 Regards,
 Steven
 
 
 ! *** Begin tech details:
 
 HQ-RTR#debug gatekeeper main 10        ! tried to call from HQ (5002) to BR2 
 (3006)
 
 Jan 17 19:56:37.786: ////GK/gk_process: QUEUE_EVENT 
 (minor 0) wakeup
 Jan 17 19:56:37.786: ////GK/gk_rassrv_arq: 
 arqp=0x48F0C08C,crv=0x7, answerCall=0
 Jan 17 19:56:37.786: ////GK/gk_rassrv_sep_arq: ARQ 
 Didn't use GK_AAA_PROC
 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/gk_dns_query: No Name 
 servers
 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_get_addrinfo: 
 (3006) Tech-prefix match failed.
 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_get_addrinfo: 
 (3006) Matched zone prefix 3 and remainder 006
 Jan 17 19:56:37.786: 
 ////GK/gk_rassrv_get_ingress_network: returning 
 default ingress network = 1
 Jan 17 19:56:37.786: 
 //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: about to check the 
 source side, src_zonep=0x4793079C
 Jan 17 19:56:37.786: 
 //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: matched zone is 
 UCM, and z_invianamelen=0
 Jan 17 19:56:37.786: 
 //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: about to check the 
 destination side, dst_zonep=0x47930A08
 Jan 17 19:56:37.786: 
 //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: matched zone is 
 UCME, and z_outvianamelen=4
 Jan 17 19:56:37.786: 
 //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone      and 
 z_outvianamep=CUBE
 Jan 17 19:56:37.786: 
 //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: Received ARQ for a 
 zone (UCME) that has an outviazone (CUBE) specified.  Pick an IP-IP gateway 
 in that viazone.
 Jan 17 19:56:37.786: 
 ////GK/gk_gw_select_ipipgw_random: zonep: 0x47930C74, 
 tpp: 0x0, current_endpt: 0
 Jan 17 19:56:37.786: 
 ////GK/gk_gw_select_ipipgw_random: Selecting any 
 IPIPGW. qelemp.head=0x46F0FE88, use_count=1, current_endpt=0
 Jan 17 19:56:37.786: 
 ////GK/gk_gw_select_ipipgw_random: qelemp=0x46F0FE88, 
 loop_count=0
 Jan 17 19:56:37.786: 
 ////GK/gk_gw_select_ipipgw_random: Examining tgwp 
 0x46F253E0, g_supp_prots: 0x50 qelemp: 0x46F0FE88, loop_count: 1
 Jan 17 19:56:37.786: 
 ////GK/gk_gw_select_ipipgw_random: Searched through 
 the entire gateway list. loop_count=0
 Jan 17 19:56:37.786: 
 ////GK/gk_gw_select_ipipgw_random: Could not find an 
 IPIPGW.
 Jan 17 19:56:37.786: 
 //80AC69450700/80AC69450700/GK/rassrv_get_addrinfo(3006): Viazone gateway 
 selection failed for zone CUBE
 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/gk_rassrv_sep_arq: 
 rassrv_get_addrinfo() failed (return code = 0x805)
 
 
 HQ-RTR#show gatekeeper gw-type-prefix
 GATEWAY TYPE PREFIX TABLE
 =
 Prefix: 1#*    (Default gateway-technology)
  Zone CUBE master gateway list:
    10.10.110.1:1720 HQ-RTR
  Zone UCM master gateway list:
    10.10.210.10:44248 gk-trunk_1
    10.10.210.11:36641 gk-trunk_2
 
 Prefix: 3#*
  Zone UCME master gateway list:
    10.10.110.3:1720 BR2-RTR
 
 
 HQ-RTR#show gatekeeper zone prefix
      ZONE PREFIX TABLE
      =
 GK-NAME               E164-PREFIX
 ---               ---
 UCME                  3...
 UCM                   5...
 
 
 HQ-RTR#show running-config interface loopback 0
 interface Loopback0
 ip address 10.10.110.1 255.255.255.255
 h323-gateway voip interface
 h323-gateway voip id CUBE ipaddr 10.10.110.1 1719
 h323-gateway voip h323-id HQ-RTR
 h323-gateway voip tech-prefix 1#
 
 
 HQ-RTR#show running-config | section gatekeeper
 gatekeeper
 zone local UCM ipexpert.com
 zone local UCME ipexpert.com outvia CUBE
 zone local CUBE ipexpert.com
 zone prefix UCME 3...
 zone prefix UCM 5...
 gw-type-prefix 1#* default-technology
 no shutdown
 ___
 For 

Re: [OSL | CCIE_Voice] QoS question on Workbook2 Lab 10

2012-01-17 Thread Anthony Alba
...I just did a check: in Workbook 2 Lab 6, Tasks 7.1, 7.2 we are trusting
the phones+HWIC-4ESW on both BR1  BR2 , the class-map used is

class-map match-all wan-rtp
 match dscp ef
etc. etc

...so as I thought, the DSG is not consistent here...

On Wed, Jan 18, 2012 at 12:52 PM, Anthony Alba ascanio.al...@gmail.comwrote:

  Hello,

 This is what I thought the DSG was pointing too:

 the HWIC-4ESW is a cheapo low-end device and we're not sure what it does
 with the  markings from the phone so let's re-classify and re-mark at BR1's
 WAN egress interface to be safe (i.e., don't depend on what phone +
 HWIC-4ESW passes to us)

 BTW, I have no knowledge that the HWIC-4ESW spoils markings so this is
 more a case of being paranoid.

 Now if you had your phones attached via another 3750 to BR1 then by all
 means use trust.

 (I'm not sure the DSG is entirely consistent about this: I'm sure there
 are other solutions where the phone+HWIC-4ESW is trusted.)






 On Tue, Jan 17, 2012 at 10:18 AM, John McGaughey (jomcgaug) 
 jomcg...@cisco.com wrote:

 Hello,

 ** **

 In Workbook 2, Lab 10, question 5.2  it asks you to setup MLP LFI between
 HQ and BR1.  In the solution guide it has you use auto qos trust on the HQ
 side but does not use trust on the BR1 side.  The DSG guide says the reason
 for not using the trust key word is because of the following:

 ** **

 *Note that we have not done any prior QOS classification/marking on the
 ESW module therefore we will use class-based marking (no use of the trust
 keyword when running auto qos).*

 * *

 But the phones use the following markings by default.

 ** **

 signaling (SCCP or SIP) - CoS 3 / cs3

 media (RTP) - CoS 5 / DSCP 46 (EF)

 ** **

 Why couldn’t we just use the trust keyword on BR1 as well since the phone
 is already marking the packets correctly?

 ** **

 John

 ___
 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] QoS question on Workbook2 Lab 10

2012-01-17 Thread Rrcrumm
If the question says to trust packets, should you use the trust keyword or is 
that there to throw you off.

Sent from my iPhone

On Jan 16, 2012, at 6:54 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote:

 Hello John ,
 
 You thinking is correct however as  the Packet will traverse over the
 wan , it will be always subject to get modified by a lot of SW to
 change specially the DSCP Value and so the QOS SRND recommend to Not
 trust the packets coming from the wan and so we dont trust in the
 Branches routers .
 
 The right answer for any QOS Question is what is in the SRND and i
 believe thats why you will find it in the real lap in the desktop
 
 Ash
 
 On Mon, Jan 16, 2012 at 8:18 PM, John McGaughey (jomcgaug)
 jomcg...@cisco.com wrote:
 Hello,
 
 
 
 In Workbook 2, Lab 10, question 5.2  it asks you to setup MLP LFI between HQ
 and BR1.  In the solution guide it has you use auto qos trust on the HQ side
 but does not use trust on the BR1 side.  The DSG guide says the reason for
 not using the trust key word is because of the following:
 
 
 
 Note that we have not done any prior QOS classification/marking on the ESW
 module therefore we will use class-based marking (no use of the trust
 keyword when running auto qos).
 
 
 
 But the phones use the following markings by default.
 
 
 
 signaling (SCCP or SIP) - CoS 3 / cs3
 
 media (RTP) - CoS 5 / DSCP 46 (EF)
 
 
 
 Why couldn’t we just use the trust keyword on BR1 as well since the phone is
 already marking the packets correctly?
 
 
 
 John
 
 
 ___
 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