Re: [OSL | CCIE_Voice] CME diap-peer question

[Steve Denney (stdenney)]

Instead of using destination-pattern 9001212T in dial-peers 2 and 3, use
something more specific, such as 9001212...$ instead.

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Randall
Crumm
Sent: Thursday, June 02, 2011 11:59 AM
To: Online Study
Subject: [OSL | CCIE_Voice] CME diap-peer question

 

HI,

I have a question.

Say I have a CME and have a dial-peer for intl calls that goies out
local GW

I also need two dial-peers(1 backupo) to route calls through
Gatekeeper@HQ, 1st choice, or through the branch 1 gw as a backup.

 

I get calls to work through the GK but when it is down my calls go
through dial-peer 1 instead of dial-peer 3

 

My question is how is how do make sure the call goes though dial-peer 3
and not dial-peer 1 

 

 

(leaving some info out)

 

dial-peer voice 1 pots - local gw

dest-pattern 900T

prefix 00

port0/0/0:15

 

dial-peer voice 2 voip

dest-patt

9001212T

session target ras

 

dial-peer voice 3 voip

dest-pattern 9001212T

session target br1

pref 1

 

 

thanks

randall

 

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Re: [OSL | CCIE_Voice] drop packets from 3750 output queue 2

[Steve Denney (stdenney)]

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

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Re: [OSL | CCIE_Voice] ISDN bchan selection

[Steve Denney (stdenney)]

It helps me to visualize a ladder, with the rungs being the channels,
and rung (channel) number 1 being at the top. 

 

Ascending and descending refer to *numerically* ascending or
descending - so ascending numerically (from channel 1 to channel N)
means going top down on the ladder.

Likewise, descending numerically (the default; from channel N down to
channel 1) means going bottom up on the ladder.

 

So Randall, your statement is incorrect. Top down = descending =
channel 1 to 23.

 

Just my little mnemonic. YMMV :)

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cristobal
Priego
Sent: Wednesday, June 01, 2011 1:38 PM
To: Randall Crumm
Cc: Online Study
Subject: Re: [OSL | CCIE_Voice] ISDN bchan selection

 

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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www.PlatinumPlacement.com

 

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Re: [OSL | CCIE_Voice] ISDN bchan selection

[Steve Denney (stdenney)]

Whoops. I was doing fine up until that last cut and paste error. :)

 

Bill's absolutely correct; that last line should have read

Top down = *ascending* = channel 1 to 23.

 

cheers, sd

 

From: Bill Lake [mailto:whl...@gmail.com] 
Sent: Wednesday, June 01, 2011 4:34 PM
To: Steve Denney (stdenney)
Cc: Cristobal Priego; Randall Crumm; Online Study
Subject: Re: [OSL | CCIE_Voice] ISDN bchan selection

 

ascending is 1-23, just did this at a customer so I know that is right
and confirmed with show isdn q931

descending is 23-1 as is the default.

 

On Wed, Jun 1, 2011 at 1:54 PM, Steve Denney (stdenney)
stden...@cisco.com wrote:

It helps me to visualize a ladder, with the rungs being the channels,
and rung (channel) number 1 being at the top. 

 

Ascending and descending refer to *numerically* ascending or
descending - so ascending numerically (from channel 1 to channel N)
means going top down on the ladder.

Likewise, descending numerically (the default; from channel N down to
channel 1) means going bottom up on the ladder.

 

So Randall, your statement is incorrect. Top down = descending =
channel 1 to 23.

 

Just my little mnemonic. YMMV :)

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cristobal
Priego
Sent: Wednesday, June 01, 2011 1:38 PM
To: Randall Crumm
Cc: Online Study
Subject: Re: [OSL | CCIE_Voice] ISDN bchan selection

 

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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please visit www.ipexpert.com http://www.ipexpert.com/ 

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

 


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please visit www.ipexpert.com http://www.ipexpert.com/ 

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

 

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Re: [OSL | CCIE_Voice] Vol2 lab 3 q7.2

[Steve Denney (stdenney)]

Use cisco / cisco (not ipexpert / cisco) for the FTP server username /
password. This is a recurring typo in the courseware.

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ahmed
Ellboudy
Sent: Wednesday, May 18, 2011 5:45 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Vol2 lab 3 q7.2

 

Dears ,

I am working on copying the license for CUE/CCM
(cue-vm-license_50mbx_ccm_7.0.5.pkg)file using freeftpd but I get an
(error access denied 530 ) on the CUE.

Any help for this issue ?

 

Thanks,

 

 

Ahmed Ellboudy | CCNP, CCVP.

 

Networking Team Leader

Raya IT - Professional Networking Services

Mobile: +20100770837

Tel  : +20238276000 Ext. 2338

Fax : +20238372930

Email  : ahmed_ellbo...@rayacorp.com
mailto:nadia_khal...@rayacorp.com 
Address : El Motamayez District - 6th of October

 

 

 

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 this message by a third party or as a
result of any malicious code or virus 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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Re: [OSL | CCIE_Voice] Volume 2 Lab 5 Question 2.2

[Steve Denney (stdenney)]

The VoD indeed takes a different approach, ignoring that particular
requirement. But it's to your benefit if you know how to configure it
both ways. :)

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Chevy
Sent: Wednesday, May 18, 2011 10:42 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Volume 2 Lab 5 Question 2.2

 

I'm looking at the video for Volume 2 Lab 5.  In question 2.2 it says
you are not permitted to create any new Route Patterns to achieve this
task.  However in the video, Vik is creating new route patterns.  Is
this a typo?

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Re: [OSL | CCIE_Voice] UCCX script editor in standalone mode

[Steve Denney (stdenney)]

IIRC, you'll need to use UCCX editor 8.0 or higher for standalone mode
to work properly. I'm using v8.0(1.1) and it works fine. 

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of adam
compton
Sent: Monday, May 09, 2011 8:35 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] UCCX script editor in standalone mode

 

All,

 

I'm having a hell of a time trying to get UCCX editor to work in
standalone mode.  I use proctorlabs for my UCCX studying, because I
don't have a UCCX server.  I am able to connect to the UCCX with the
script editor, but when I'm done with the lab and try the editor at
home, it fails everytime.  It's getting really frustrating because this
is an area I need some time in.  Does anybody have any good ideas on
getting that running locally by itself?

 

Adam Compton

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Re: [OSL | CCIE_Voice] Finally succeeded ..Got CCIE

[Steve Denney (stdenney)]

Congrats Shrini!

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Shrini
Sent: Wednesday, May 04, 2011 4:02 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Finally succeeded ..Got CCIE

 

Hi Experts,

Today I am officially announced as Voice CCIE.

Thanks to one and all for your valuable suggestions and help throughout
this journey.

Special thanks to Vik and IP Expert team for hosting this excellent
mailer list and helping us.

Thanks again
Shrini

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Re: [OSL | CCIE_Voice] Workbook 2 Lab 2 Task 6.3

[Steve Denney (stdenney)]

You know, I think you're right. The 10 50 20 20 would seem to be
correct for *share* not shape (since 50 / (10+50+20+20) = 50/100 = 1/2 =
50%).

 

Anyone else?

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Pablo
Meneses
Sent: Saturday, April 30, 2011 7:46 PM
To: ccie voice
Subject: [OSL | CCIE_Voice] Workbook 2 Lab 2 Task 6.3

 

Hello Experts,

I was wondering if one of you could give me an explanation on this task
since I am a bit lost:

Workbook 2 Lab 2 Task 6.3:

The task says:

Ensure that the SRR scheduler for phones at the HQ switch shapes Q2 to
50% of the interface bandwidth.

On the Solution Guide, it says:

HQ-3750(config)#interface FastEthernet 1/0/2
HQ-3750(config-if)#srr-queue bandwidth shape 10 50 20 20

After I did that I got the following output:

HQ-3750#show mls qos interface fastEthernet 1/0/2 queueing
FastEthernet1/0/2
Egress Priority Queue : enabled
Shaped queue weights (absolute) :  10 50 20 20
Shared queue weights  :  10 10 60 20
The port bandwidth limit : 100  (Operational Bandwidth:100.0)
The port is mapped to qset : 2

However, I then check CCO and found the following:

The bandwidth weight for queue 1 is 1/8, which is 12.5 percent:
Switch(config)# interface gigabitethernet2/0/1
Switch(config-if)# srr-queue bandwidth shape 8 0 0 0

http://www.cisco.com/en/US/docs/switches/lan/catalyst3750/software/relea
se/12.2_50_se/configuration/guide/swqos.html#wp1163879

The question is:

Why is it configured on the solution as 10 50 20 20? Shouldn't it be
configured as 0 2 0 0 since 1/2 equals 0.5?

Looking forward to your response.

-Pablo Meneses.

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Re: [OSL | CCIE_Voice] RSVP based CAC

[Steve Denney (stdenney)]

If you bump up the ip rsvp bandwidth command on the routers to a much higher 
number (say 120), does the call go through?

 

If so, do you have any shared lines configured across sites on the phones? I 
recall a bug whereby this can trigger a “two call worst case bandwidth” 
scenario (e.g., need 2 reservations at 40k instead of one).

 

Also, is there any other CAC method configured besides RSVP?

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vinay Kumar6
Sent: Friday, April 29, 2011 7:10 AM
To: Naoufal Kerboute
Cc: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] RSVP based CAC

 

Yes, It is registered. 

Apr 29 10:47:45.600: RSVP 10.10.11.1_17198-10.10.10.1_17944[0.0.0.0]: RESV: no 
path information for 10.10.10.1 

Where 10.10.10.1 is HQ MTP and 10.10.11.1 is BR MTP IP. 

because of which the RSVP is not getting initiated. 

Not sure how to fix this. 

Warm Regards, 
Vinay Kumar
MTS-Remote Support Centre.
IBM India Private Limited, Subramanya Arcade1, 12, Bannerghatta Main Road, 
Bangalore 560 029 (India) Telephone: Direct +91-80-40683977, Board +91-80-4068 
3000, Extn: 83977, Fax +91-80-26787711  Email : vinayjaisw...@in.ibm.com





From: 

Naoufal Kerboute naou...@mhdinfotech.com 

To: 

Vinay Kumar6/India/IBM@IBMIN 

Cc: 

ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com, 
ccie_voice-boun...@onlinestudylist.com 
ccie_voice-boun...@onlinestudylist.com 

Date: 

04/29/2011 04:21 PM 

Subject: 

RE: [OSL | CCIE_Voice] RSVP based CAC

 






Have you checked the status of your MTP on CUCM (make sure It’s registred) 
Naoufal 
  
  
  
From: Vinay Kumar6 [mailto:vinayjaisw...@in.ibm.com 
mailto:vinayjaisw...@in.ibm.com ] 
Sent: Friday, April 29, 2011 12:33 PM
To: Naoufal Kerboute
Cc: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] RSVP based CAC 
  
Codec is g729 and g722 is disabled in enterprise parameters. 

From the debug on the gateways I found  a message which says RSVP Confirmation 
not required. 

Warm Regards, 
Vinay 



From: 

Naoufal Kerboute naou...@mhdinfotech.com 

To: 

Vinay Kumar6/India/IBM@IBMIN 

Cc: 

ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com, 
ccie_voice-boun...@onlinestudylist.com 
ccie_voice-boun...@onlinestudylist.com 

Date: 

04/29/2011 01:32 PM 

Subject: 

RE: [OSL | CCIE_Voice] RSVP based CAC


  

 







What is the codec between regions? Make sure you’re using g729 and disable the 
g722 advertising and also the iLBC 
 
From: ccie_voice-boun...@onlinestudylist.com 
mailto:ccie_voice-boun...@onlinestudylist.com  
[mailto:ccie_voice-boun...@onlinestudylist.com 
mailto:ccie_voice-boun...@onlinestudylist.com ] On Behalf Of Vinay Kumar6
Sent: Friday, April 29, 2011 11:28 AM
Cc: ccie_voice@onlinestudylist.com mailto:ccie_voice@onlinestudylist.com ; 
ccie_voice-boun...@onlinestudylist.com 
mailto:ccie_voice-boun...@onlinestudylist.com 
Subject: Re: [OSL | CCIE_Voice] RSVP based CAC 
 
Following is the config on the routers. 


interface Serial0/0/0.1 point-to-point 
ip address 10.10.1.1 255.255.255.0 
ip ospf mtu-ignore 
snmp trap link-status 
frame-relay interface-dlci 401 
ip rsvp bandwidth 40 

dspfarm profile 2 mtp 
codec g729r8 
codec pass-through 
rsvp 
maximum sessions software 4 
associate application SCCP 





dspfarm profile 1 mtp 
codec g729r8 
codec pass-through 
rsvp 
maximum sessions software 4 
associate application SCCP 


interface Serial0/2/0.1 point-to-point 
ip address 10.10.1.2 255.255.255.0 
ip ospf mtu-ignore 
snmp trap link-status 
frame-relay interface-dlci 501 
ip rsvp bandwidth 40 



Is there any step by step guide to configure or troubleshoot it? 

Warm Regards, 
Vinay Kumar

From: 

Vinay Kumar6/India/IBM@IBMIN 

To: 

Rogers Ochieng rogersochi...@gmail.com 

Cc: 

ccie_voice@onlinestudylist.com 

Date: 

04/29/2011 11:46 AM 

Subject: 

Re: [OSL | CCIE_Voice] RSVP based CAC


  


  








Yes, it is there. 

Warm Regards, 
Vinay Kumar 

From: 

Rogers Ochieng rogersochi...@gmail.com 

To: 

Vinay Kumar6/India/IBM@IBMIN 

Cc: 

ccie_voice@onlinestudylist.com 

Date: 

04/29/2011 10:25 AM 

Subject: 

Re: [OSL | CCIE_Voice] RSVP based CAC


  


  








do you have this under your IOS dspfarm profile Mtp configuration 

rsvp 

codec pass-through 



On 29 April 2011 07:24, Vinay Kumar6 vinayjaisw...@in.ibm.com wrote: 
Hi, 

Trying to configure Location based CAC using RSVP, have done the configuration 
buut it always says not enough bandwidth even though i have given ample 
bandwidth on the serial interfaces. 

Steps used to configure: 

Configured MTP on the HQ and Branches-Registered to HQ and Branch DP. 
Assigned the HQ and Branch MTPs to respective DPs using MRGL. 
Created Location for HQ and 

Re: [OSL | CCIE_Voice] One Week Lab Experience - April 11 thru 15 - Who will attend?

[Steve Denney (stdenney)]

I'll be there as well (assuming you meant April 11th and not 19th) :)

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roger
Carpio
Sent: Friday, April 01, 2011 7:46 PM
To: Pablo Meneses
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] One Week Lab Experience - April 11 thru
15 - Who will attend?

 

Pablo,

I'll see you there :-)

Regards,
Roger Carpio.

On Fri, Apr 1, 2011 at 1:39 PM, Pablo Meneses pmenese...@gmail.com
wrote:

Hi there,

I am just wondering how will attend to April 19th's One Week Lab
Experience boot camp in San Jose for CCIE Voice?

Regards.

-Pablo Meneses.

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please visit www.ipexpert.com

 

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Re: [OSL | CCIE_Voice] Vol2 Lab 7 Question 3.2 MVA

[Steve Denney (stdenney)]

Hi ShinGei,

 

I see you have already changed the CUCM Service Parameter Inbound
Calling Search Space for Remote Destination from its default to RDP +
Line CSS, which is good.

 

Have you tried restarting the mobility services (and the CUCM service as
a last resort)?

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ShinGei
Yong
Sent: Monday, March 21, 2011 7:34 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Vol2 Lab 7 Question 3.2 MVA

 

Hi,

I've gone thru the entire OSL list regarding the problem as I'm facing,
but unfortunately didn't manage to get the answer.

MVA has configured on BR2 gw, the remote destination manage to call in
to MVA# and authenticated successfully. When press option #1, which make
a call to either internal or external, it just failed.

RDP CSS: css-snr-3002 / pt-snr-3002
Rerouting CSS: css-br2-unrestricted /pt-uk-emer, pt-uk-national,
pt-uk-international

Mobility Service Parameters:
Partial Match 10 Digits, RDP + Line CSS

** As per question required, we need xlation rule to display mobile ANI
instead of internal DN **
Translation Pattern
 / pt-snr-3002
CSS: css-phones / pt-phones
Use Calling Party EPNM: Checked

1002/pt-phones
5002/pt-phones

I can call up the internal extension [15]002, IF I include the pt-phones
into RDP CSS, but that will cause ANI display become internal DN instead
of mobile ANI because of closest match.
Any idea why internal calling doesn't work?



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Re: [OSL | CCIE_Voice] CCIE PASSE

[Steve Denney (stdenney)]

You all do know what a troll is, right? 

http://en.wikipedia.org/wiki/Troll_(Internet)

 

You'll notice that Suzie Suzie claimed to have passed the CCIE, but
never told us what his (her?) number is? Hmm, I wonder why not?

I hate to even chime in and keep this useless, time-wasting thread alive
any longer, but please folks, don't feed the trolls any more. :)

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of PaulG
Sent: Monday, March 21, 2011 7:03 AM
To: voice boy
Cc: OSL Questions
Subject: Re: [OSL | CCIE_Voice] CCIE PASSE

 

 

 

IT'S TIME TO MOVE ON SUZIE

 


THANKS.

2011/3/21 voice boy voice...@hotmail.com

 
Why OSL let this silly talking from that SUZIE
he pass his exam using his bootcamps
 
And after taking his number, come to shout here to make the exam more
hard with a lot of pre-requisites
 
why SUZIE didn't shout before taking your number, or you need only to
stop the growing of this number to imagine your self from the littles
who acheive that
 
Please OSL stop this very bad useless discussion and let us follow to
acheive our goal
 
thanks
 
 From: ccie_voice-requ...@onlinestudylist.com
 Subject: CCIE_Voice Digest, Vol 61, Issue 124 


 To: ccie_voice@onlinestudylist.com

 Date: Mon, 21 Mar 2011 02:55:55 -0400
 
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 Today's Topics:
 
 1. Re: CUE/CME MWI config (Rogers Ochieng)
 2. Re: CUE/CME MWI config (Rogers Ochieng)
 3. Re: CCIE PASSE (Jon 1992)
 
 
 --
 
 Message: 1
 Date: Mon, 21 Mar 2011 09:41:06 +0300
 From: Rogers Ochieng rogersochi...@gmail.com
 To: Michael Luo hout...@gmail.com 


 Cc: ccie_voice@onlinestudylist.com

 Subject: Re: [OSL | CCIE_Voice] CUE/CME MWI config
 Message-ID:
 AANLkTi=ivgphff9ugkdnh1qc-vmbvgyem81fxfuxs...@mail.gmail.com
 Content-Type: text/plain; charset=iso-8859-1
 

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps5520/products_c
onfiguration_example09186a0080289ef0.shtml
 
 On 21 March 2011 07:38, Michael Luo hout...@gmail.com wrote:
 
  Can anyone send me a link to CUE/CME MWI configuration example?
 
  Whenever I left a message, CUE VM pilot will call the phone instead
of
  lighting up the phone.
 
  Attached are the config and debug.
 
  Thanks!
  Michael 


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please
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 Message: 2
 Date: Mon, 21 Mar 2011 09:43:34 +0300
 From: Rogers Ochieng rogersochi...@gmail.com
 To: Michael Luo hout...@gmail.com 


 Cc: ccie_voice@onlinestudylist.com

 Subject: Re: [OSL | CCIE_Voice] CUE/CME MWI config
 Message-ID:
 aanlktimobyeixnexlxaqf6hkh9y+gcsaoczmsx+2s...@mail.gmail.com
mailto:aanlktimobyeixnexlxaqf6hkh9y%2bgcsaoczmsx%2b2s...@mail.gmail.com
 
 Content-Type: text/plain; charset=iso-8859-1
 
 Use this
 

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps5520/products_c
onfiguration_example09186a008037f2a9.shtml
 
 On 21 March 2011 07:38, Michael Luo hout...@gmail.com wrote:
 
  Can anyone send me a link to CUE/CME MWI configuration example?
 
  Whenever I left a message, CUE VM pilot will call the phone instead
of
  lighting up the phone.
 
  Attached are the config and debug.
 
  Thanks!
  Michael 


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

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 Message: 3
 Date: Mon, 21 Mar 2011 15:55:50 +0900
 From: Jon 1992 jon1...@hotmail.com
 To: Antonio Dee antonio_...@hotmail.com, 


 ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] CCIE PASSE

 Message-ID: blu167-ds887e42d90e91561448e76a7...@phx.gbl
 Content-Type: text/plain; charset=iso-8859-1 


 
 Antonio
 
 Well said.
 I personally am putting the CCIE on back burner, work, and family took
priority after my attempt at the beast.
 With over 15 years experience in IT, and many years in voice as well,
the CCIE Voice is a very hard lab test, but it doesnt completely test
your abilities.
 The 

Re: [OSL | CCIE_Voice] Calling and Called Party Number Type

[Steve Denney (stdenney)]

Agreed with your logic, but in your first example (calling HQ phone from
BR1 Phone using local gateway), shouldn't the calling party be
Subscriber (BR1 phone calling out from BR1 GW)?

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Shrini
Sent: Sunday, March 06, 2011 5:28 PM
To: Tamer Ismail
Cc: 'CCIE Study'
Subject: Re: [OSL | CCIE_Voice] Calling and Called Party Number Type

 

Called and Calling party number type will be always referenced with the
GW you are using.

Examples:

If you are calling HQ phone from BR1 Phone using local gateway.

Called party is National (because you are sending 10 digits for national
call)  agreed
calling party is national. ???

If calling BR1 PSTN from BR1 phone using local gateway

Called - Subscriber
Calling - Subscriber

If BR1 phone calling HQ PSTN using HQ GW 

Calling is National
Called is Subscriber

Yes this is confusing, but always use the GW you are using as reference.

On 3/3/2011 3:30 AM, Tamer Ismail wrote: 


Calling Party Number Type to: Subscriber. or National

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Re: [OSL | CCIE_Voice] Calling and Called Party Number Type

[Steve Denney (stdenney)]

I agree with Roger's response 100%, with one small comment: If nothing is 
stated in the lab, then do not waste time configuring any of those settings. 
Extra or best practice configuration, if not required, will not get you any 
points.

 

If it's not in the lab, then it is not a requirement, and hence it does not 
matter what values are set. Just be sure you have read the question 
*completely* and understand exactly what is being asked for. :)

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roger Källberg
Sent: Friday, March 04, 2011 3:54 AM
To: Ccie Voice; CCIE Study
Subject: Re: [OSL | CCIE_Voice] Calling and Called Party Number Type

 

As other have answered, if needed you will be instructed in the lab material 
how these settings should be set. That's about all that anyone can say about 
this, and other stuff covered in the lab, without breaking the NDA.

 

As a side note, Vik Malhi and Amy Ryan recommendation about this on the 
boot-camp were that you always should do whatever the lab material states, 
pretty obvious J, and if nothing is stated you should set it to the proper 
value for that type of call. This applies to both Type of number (TON) and 
Number plan information (NPI). The later is mostly set to ISDN, if nothing else 
is said.

 

Sincerely

 

Roger Källberg
CCIE # 26199 (Voice)
Unified Communication Consultant
Cygate AB

 

From: Ccie Voice [mailto:v.c...@yahoo.com] 
Sent: den 3 mars 2011 19:52
To: CCIE Study
Subject: Re: [OSL | CCIE_Voice] Calling and Called Party Number Type

 

Thank you all for your reply,

I just need to know if the PSTN router in the LAB will accept the call or no if 
it is not set to the proper value.

If the PSTN router will not accept the call then it is OK I can play with these 
values and solve the problem.

But the problem if the PSTN router accepts all calls based on called party 
number and later on the proctor will check if you set the values correctly or 
not.


for me what I understood before is the way that Roger sent. (thank you Roger) 

Regards,



From: Roger Källberg roger.kallb...@cygate.se
To: Ccie Voice v.c...@yahoo.com; CCIE Study ccie_voice@onlinestudylist.com
Sent: Thu, March 3, 2011 6:41:12 PM
Subject: SV: [OSL | CCIE_Voice] Calling and Called Party Number Type



Hi,

You need to look at this from the originating endpoint and the outgoing 
gateway. For a more detailed explanation see my response in line with your 
mail. 

Sincerely

Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB
Eric Perssons väg 21, SE-217 62 MALMÖ

 



Från: Ccie Voice [v.c...@yahoo.com]
Skickat: den 3 mars 2011 02:49
Till: CCIE Study
Ämne: [OSL | CCIE_Voice] Calling and Called Party Number Type

Hi All,

I am a little bit confused about how to set the value for Calling and Called 
Party Number Plan.

let us say HQ Phone 1 Calls local Call in this case I think I have to set:
Calling Party Number Type to: Subscriber.
Called Party Number Type to: Subscriber.

This is correct

What about Long Distance:
Calling Party Number Type to: Subscriber or National

From the perspective of caller and VGW this is a call that came from a local 
site , aka it's subscriber
Called Party Number Type to: National
From the perspective of called and VGW this is a call goes to a remote phone, 
aka it's national


it will be more complicated if we need to use TEHO, So if HQ Phone 1 calls BR1 
Local PSTN number what I should set the values?

Long Distance, using BR1 Router

Calling Party Number Type to: Subscriber or National

From the perspective of caller and VGW this is a call that came from a remote 
site , aka it's national
Called Party Number Type to: National or Subscriber 

From the perspective of called and VGW this is a call goes to a local phone, 
aka it's subscriber

Long Distance, backup for BR1 using HQ Router

Calling Party Number Type to: Subscriber or National

From the perspective of caller and VGW this is a call that came from a local 
site , aka it's subscriber
Called Party Number Type to: National or Subscriber I am using BR1 Router 

From the perspective of called and VGW this is a call goes to a remote phone, 
aka it's national

Regards,

 

 

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Re: [OSL | CCIE_Voice] Simplest way to record a custom prompt for UCCX?

[Steve Denney (stdenney)]

For the benefit of the archives: I was going through some older emails today 
and playing with this method of recording custom UCCX prompts (and a very nice 
method it is, Kobel, thank you!)

 

But I was having issues when trying to set the properties for the Upload Prompt 
step (# 2 below). Instead of bringing up the customizer window, I was getting 
an error message with the words “Customizer Error”, a big red X, and no text. 
Not very helpful. ;)

 

Some creative Googling led me to this: 
https://supportforums.cisco.com/thread/2036653 which resolved the error. 
Essentially, if you’re running the UCCX editor on Windows 7 or Vista, you have 
to set its properties in the compatibility tab to disable visual themes, and 
run as administrator (the link has the details). I’m running UCCX editor 
v8.0(1.1) since I can run it offline on my laptop; don’t know if other versions 
are affected, but wanted to post this fix to hopefully save others some time if 
you hit this same issue.

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Miron Kobelski
Sent: Monday, January 10, 2011 4:53 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Simplest way to record a custom prompt for UCCX?

 

Hello,

I was playing with UCCX today and I think I found a quick way to record a 
custom prompt without using CUE or CUC. 

1) open script editor on UCCX desktop and go to file  new  IVR tab  Spoken 
name upload
2) at the end of the SNU script, before Terminate step, add Upload Prompt 
step:
 * language en_US
 * name: prompt.wav
 * Document: recording (variable used by the script)
 * User: user (variable used by the script)
3) save the script to script repo and create the application (you would need it 
anyway for the task), but temporarily assign it with this modified SNU script
4) use the SNU IVR to record the prompts (remember to rename the file or it 
will get overwritten), which will be put directly to prompt repository. You 
need to authenticate with whatever the UCCX user (e.g. admin or agent) has in 
his Name Dialing field in CUCM's User Configuration page + his PIN.
5) Reuse your application by replacing the SNU script with the real one 
required by your task.

Benefits:
 * no need for CUE/CUC integration
 * no need to enable outgoing call routing for CUC
 * no need to configure Telephony Administration for prompt recording

What's your method of choice and why? ;)




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Re: [OSL | CCIE_Voice] BUG ????? QoS with MLP LFI

[Steve Denney (stdenney)]

I’ve seen this a few times also. In the OWLE class, Vik mentioned that this is 
a known bug that shows up sometimes on their pod HQ routers because they are 
running a slightly older IOS (in order to allow the Gatekeeper feature to run 
without purchasing a feature license). Not sure which specific IOS versions are 
affected. Sometimes you can get the virtual interfaces to come up by bouncing 
the physical serial interfaces (not always).

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Friderich Claude
Sent: Monday, February 21, 2011 10:25 AM
To: ccie_voice-boun...@onlinestudylist.com; ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] BUG ? QoS with MLP LFI

 

Hello Guys,

 

Does anybody  already encountered the problem with Virtual access  ??

 

Let me explain …..

 

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-Access110.10.112.1YES
TFTP up   down

 

BR2

Interface IP-AddressOK? 
  Method   StatusProtocol

Virtual-Access110.10.112.2YES
TFTP up   down

 

It’s not the first time !!! I tried to change the IOS but in vain.  The same 
behavior with IOS 12.4(20)T2,T1  and 12.4(24)T2.

 

If something like this happens during the lab, what can I do   Just ask the 
proctor and tell him that I have an unexpected behavior and that I’m sure about 
the qos config ?

 

I have usually this type of behavior after rebooting my routers.

 

Any comments or solutions would be appreciated.

 

Regards

 

Claude.

 

Claude Friderich

PreSales Support

 

NETCORE PSF S.A.

49 rue du Baerendall

B.P.65 L-8201 Mamer

Téléphone: 31 33 80-407

Fax: 31 33 80 8-407

GSM: 621 303 616

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

 

 
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Re: [OSL | CCIE_Voice] Vol2 Lab3 - Changing CUE License - FTP Issues

[Steve Denney (stdenney)]

Thanks to everyone who responded. I suspect it was a simple
user/password mistake (docs say username ipexpert, but many replies
said username should be cisco). 

Could have sworn that I checked the FTP server and tried that combo, but
it was late and I was tired, so maybe not. :) Thanks again.

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Steve
Denney (stdenney)
Sent: Friday, February 18, 2011 5:20 PM
To: OSL Questions
Subject: [OSL | CCIE_Voice] Vol2 Lab3 - Changing CUE License - FTP
Issues

 

Sanity check...I was doing Vol2 Lab3 yesterday, and hitting an issue
with not being able to FTP the CUE CCM license file over from the UCCX
server.

I had the CUE up and online, and issued this command:

software install clean url
ftp://10.10.210.5/cue-vm-license_12mbx_ccm_7.0.1.pkg user ipexpert
password cisco

 

And got the following result:

WARNING:: This command will install the necessary software to 

WARNING:: complete a clean install.  It is recommended that a backup be
done 

WARNING:: before installing software. 

 

Would you like to continue? [n]y

Downloading ftp cue-vm-license_12mbx_ccm_7.0.1.pkg

 

Error: Download error  

Can not download cue-vm-license_12mbx_ccm_7.0.1.pkg 

error code 0 : error type 'couldn't connect to host'

cue-br2#

 

It certainly looks like an FTP server issue. But Filezilla was up and
running on the UCCX server, the file was there in the FTP directory
(double checked the filename for exact match), and I had network
connectivity (ping OK) between CUE and the UCCX server. I gave up after
a while and moved on, but what glaringly obvious thing could I have
overlooked - any ideas?

 

cheers, sd

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[OSL | CCIE_Voice] Vol2 Lab3 - Changing CUE License - FTP Issues

[Steve Denney (stdenney)]

Sanity check...I was doing Vol2 Lab3 yesterday, and hitting an issue
with not being able to FTP the CUE CCM license file over from the UCCX
server.

I had the CUE up and online, and issued this command:

software install clean url
ftp://10.10.210.5/cue-vm-license_12mbx_ccm_7.0.1.pkg user ipexpert
password cisco

 

And got the following result:

WARNING:: This command will install the necessary software to 

WARNING:: complete a clean install.  It is recommended that a backup be
done 

WARNING:: before installing software. 

 

Would you like to continue? [n]y

Downloading ftp cue-vm-license_12mbx_ccm_7.0.1.pkg

 

Error: Download error  

Can not download cue-vm-license_12mbx_ccm_7.0.1.pkg 

error code 0 : error type 'couldn't connect to host'

cue-br2#

 

It certainly looks like an FTP server issue. But Filezilla was up and
running on the UCCX server, the file was there in the FTP directory
(double checked the filename for exact match), and I had network
connectivity (ping OK) between CUE and the UCCX server. I gave up after
a while and moved on, but what glaringly obvious thing could I have
overlooked - any ideas?

 

cheers, sd

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Re: [OSL | CCIE_Voice] How to troubleshoot CUBE call?

[Steve Denney (stdenney)]

Try adding codec g729r8 to your transcoder profile. (Also remove
g729abr8 - nothing to do with your issue, but vad is bad)

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Michael Luo
Sent: Tuesday, February 01, 2011 7:01 PM
To: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] How to troubleshoot CUBE call?

 

On HQ-RTR, debug voip ccapi inout:

Feb  1 18:53:08.822 EST: //30/00AB25620700/CCAPI/ccCallDisconnect:
   Cause Value=47, Tag=0x0, Call Entry(Previous Disconnect Cause=0,
Disconnect Cause=47)

It looks like a codec issue.  But I have configure transcoder per the
proctor guide:

HQ-RTR#sh sdspfarm units

mtp-1 Device:hq-xcoder TCP socket:[1]  REGISTERED in SCCP ver 17/10
actual_stream:6 max_stream 6 IP:10.10.200.3  35686  MTP Dixieland
keepalive 151  
Supported codec:
 G711Ulaw
 G711Alaw
 G729a
 G729ab

 max-mtps:1, max-streams:6, alloc-streams:6, act-streams:0

HQ-RTR#sh run | b teleph
telephony-service
 sdspfarm units 1
 sdspfarm transcode sessions 3
 sdspfarm tag 1 hq-xcoder
 max-ephones 1
 max-dn 1
 ip source-address 10.10.200.3 port 2000

How does CUBE know where to look for XCoder resource?

Thanks!
Michael





On Tue, Feb 1, 2011 at 5:51 PM, Michael Luo hout...@gmail.com wrote:

I was doing IPExpert vol2 lab 1 task 4.2.

When HQ(5001) calls BR2(3001), phone rings.  But when even if I pressed
answer on BR2 phone, HQ phone kept ringing for a couple seconds, then
disconnected.

I've read some thread like
http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg17555.html
.  But it didn't see to apply to my case.

What's the systematic way to debug this issue?  What debug commands we
could use?

Thank you very much!
Michael

 

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Re: [OSL | CCIE_Voice] 3750 QoS: service-policy + mls qos trustcommands on the same port

[Steve Denney (stdenney)]

To answer your second question - the Enterprise QoS SRND is here:

http://www.cisco.com/en/US/partner/docs/solutions/Enterprise/WAN_and_MAN/QoS_SRND/QoS-SRND-Book.html

 

AFAIK it’s not accessible via the support URL available to you in the lab 
(http://www.cisco.com/cisco/web/psa/default.html) – which is why they give you 
a pdf copy on the candidate desktop.

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Miron Kobelski
Sent: Wednesday, January 26, 2011 1:07 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] 3750 QoS: service-policy + mls qos trustcommands on 
the same port

 

Hello,

I'm working on Vol2 Lab8 QoS section. Task 5.2 requires to conditionally trust 
DSCP markings from the Cisco IP phone, which can be accomplished with:

mls qos trust device cisco-phone
mls qos trust dscp

But 5.3 requires policing and remarking using service-policy for the same 
switch port.
In the Enterprise QoS SRND page 106 we have:

At the time of writing, the Catalyst 2970/3560/3750 does not support a trust 
statement (such as mls qos 
trust device cisco-phone) in conjunction with a service-policy input statement 
applied to given port at 
the same time. While this may be configurable, if the switch is reset, one or 
the other statement may be 
removed when the switch reloads. This limitation is to be addressed; consult 
the latest Catalyst 
2970/3560/3750 QoS documentation for updates on this limitation

PG's solution seems to ignore this fact. What's your opinion on this? I was 
unable to find anything on this in the archive.

BTW, how can I find QoS SRND via cisco.com documentation portal?

regards
kobel



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Re: [OSL | CCIE_Voice] IPExpert's PSTN configuration problem

[Steve Denney (stdenney)]

This thread from the OSL archives explains the PSTN router setup,
including why those commands are there:

 

http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg18568.html

 

cheers, sd

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of George
Goglidze
Sent: Sunday, January 23, 2011 12:43 PM
To: Tam Nhu
Cc: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] IPExpert's PSTN configuration problem

 

Hi all,

 

I'm doing right now an online session of IPExpert, and ran into this
problem.

on PSTN router, on incoming voice-port they have a translation pattern
which translates all numbers to 1234!!! thus call failing.

 

Is it me? Maybe I loaded wrong configuration of PSTN router,  because
it's just not going to work like this.

 

thanks, 

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Re: [OSL | CCIE_Voice] IPPA service url

[Steve Denney (stdenney)]

IPPA (IP Phone Agent) and One Button Login

Cisco.com  UCCX support page  Configuration Examples  One Button
Login

 

http://www.cisco.com/en/US/partner/products/sw/custcosw/ps1846/prod_conf
iguration_examples_list.html

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mritunjay
Kumar
Sent: Thursday, January 20, 2011 9:19 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] IPPA service url

 

Hi All,

Where i can find service url for IPPA. I did not find in cucm  and ccx
help page ?

any doc which is accessible in lab ?

regards,
Mritunjay

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Re: [OSL | CCIE_Voice] T1 Pri Issue

[Steve Denney (stdenney)]

Pretty common issue (as a search of the OSL archives would show). :)

 

Remove, and then replace, the isdn bind-l3 ccm-manager command under the
serial interface. That should bring up the circuit. 

Alternative method is to remove, and then replace, the mgcp bind
control interface command. Either way should work.

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Deepak
sidana
Sent: Thursday, January 06, 2011 4:25 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] T1 Pri Issue

 

Hi All,

 

I am trying to connect the T1 from Br1-RTR to PSTN-WAN. Only when i use
service mgcp, under controller, layer 2 isdn staus as TEI_ASSIGNED. 

 

At PSTN-WAN Router, i am using isdn protocol-emulate network under
s0/0/0:23

 

Branch1 Config:-

 

BR1-RTR#sh isdn sta
Global ISDN Switchtype = primary-net5
ISDN Serial0/0/0:23 interface
 dsl 0, interface ISDN Switchtype = primary-net5
Layer 1 Status:
 ACTIVE
Layer 2 Status:
 TEI = 0, Ces = 1, SAPI = 0, State = TEI_ASSIGNED

 

!

controller T1 0/0/1
 framing esf
 linecode b8zs
 cablelength long 0db

!

interface Serial0/0/0:23
 no ip address
 encapsulation hdlc
 isdn switch-type primary-net5
 isdn bind-l3 ccm-manager
 isdn incoming-voice voice
 no cdp enable

 

Please share you experince, if some one faced the same issue.
 

ThanksRgds
Deepak Sidana.

 

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Re: [OSL | CCIE_Voice] CIPC hangs at Registering when connecting to lab CUCM

[Steve Denney (stdenney)]

...which would be good advice for a hard phone, but this is a CIPC :)

 

Just some random thoughts: You might have a DB replication issue. Check
to make sure your device type is correct on the Pub/Sub (e.g. CIPC). If
so, shut down the CallManager service on the Sub and see if it registers
to the Pub (might need to restart the service on the Pub also). If so,
run a utils dbreplication repair all from the Pub CLI.

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccieid1ot
Sent: Tuesday, November 30, 2010 1:25 PM
To: Ginther, Scott [NTK]
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CIPC hangs at Registering when
connecting to lab CUCM

 

Go into your settings unlock, then Device and erased your CCM stuff.

On Tue, Nov 30, 2010 at 11:58 AM, Ginther, Scott [NTK] 
scott.gint...@sprint.com wrote:

Cristobal,

 

That's not a silly question at all.

 

I have CIPC pointed at the CUCM Pub (10.10.210.10). The same TFTP server
that I have my IP Blue client pointed at. The IP Blue client is
registered and working. The CIPC client just says registering.

 

Both the IP Blue and CIPC phones are configured in CUCM as SCCP with the
correct MAC addresses. Nothing fancy.

 

 

 

From: Cristobal Priego [mailto:cristobalpri...@gmail.com] 
Sent: Tuesday, November 30, 2010 8:54 AM
To: Ginther, Scott [NTK]
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CIPC hangs at Registering when
connecting to lab CUCM

 

A silly question

 

Dd you change the tftp settings of your Ipc to point to the proper
server?

Right click on the ipc, preferences, tftp

Sent from my iPhone


On Nov 30, 2010, at 6:45, Ginther, Scott [NTK] 
scott.gint...@sprint.com wrote:

I'm having trouble connecting CIPC 7.0.5 to the virtual lab
CUCM. CIPC hangs at Registering, and does not give an error.

 

If I disconnect my virtual lab VPN connection and connect to my
corporate network, CIPC will register to our corporate CUCM cluster. Am
I missing something? 

 

I can't register my CIPC phone to the lab CUCM via SCCP or SIP.
I could use a little help getting it configured correctly for the
virtual lab.

 

Thanks,

 

-Scott-

 




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Re: [OSL | CCIE_Voice] Support Needed!

[Steve Denney (stdenney)]

Hi Scott,

 

Try emailing supp...@ipexpert.com

 

And best of luck on your lab!

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Scott
Newberry
Sent: Sunday, November 14, 2010 4:23 PM
To: osl osl
Subject: [OSL | CCIE_Voice] Support Needed!

 

Anybody know how to contact ProctorLabs After-Hours Support if you're
not in a current rack session?  I had sessions scheduled for today that
are no longer scheduled.  And of course, I can't schedule now since it's
past the start time.

My lab exam is tomorrow...  Just wanted to run through a few things.

Scott

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Re: [OSL | CCIE_Voice] Volume 2 Lab 3 Task 2.4, Calling Party Transformation

[Steve Denney (stdenney)]

Hi ShinGei,

 

I suspect you are just hitting a known limitation of the CIPC and IP
Blue clients. 

When you localize using calling party transformations, it will affect
not only the Ringing state (as on a regular hard phone), but also the
call history.

It's nothing you are doing wrong; simply the way these clients function.

 

cheers, sd

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ShinGei
Yong
Sent: Monday, October 18, 2010 10:47 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Volume 2 Lab 3 Task 2.4,Calling Party
Transformation

 

Hi all,

I'm trying to configure the CngPTP to transform (locallize) the calling
party number from E164 to 4 digit as per WB3, task2.4 stated.
I managed to transform the calling display to 4 digit, but some how the
missed and received call also displayed the 4 digit calling number 
instead of E164.

Prior to Localization, the Globalization was work as expected, the
calling  called number displayed in E164 format in either missed or
received.
Once i applied the CngPTP under DP, the calling number localize to 4
digit, but it does localize the missed and received entry, which is
not desired.

I'm testing this question by using 1 CIPC and 1 IPBlue, and calling each
other to verify.

Am i miss out anything?

TIA
Shingei.

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Re: [OSL | CCIE_Voice] Passwords on the Lab

[Steve Denney (stdenney)]

Congratulations on passing your written. 

Even if someone did know the usernames/passwords, to post them would be
in violation of NDA.
Just use something short and simple for your own lab, such as admin |
cisc0123.

cheers, sd



-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Maciej
Karpinski
Sent: Thursday, October 14, 2010 2:31 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Passwords on the Lab

Hi

I passed my CCIE Voice written a couple of weeks ago and will start to 
build a lab. Does any one know usernames/passwords used on the lab?
Would 
want to use the same in my own lab.
/Maciej
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Re: [OSL | CCIE_Voice] Speed for taking the Lab

[Steve Denney (stdenney)]

Also highly recommended: 

Vik's Voice Lab Strategy vLecture, available via IPexpert's Facebook
page:

http://www.facebook.com/pages/IPexpert/24586557119?v=app_7146470109

 

Direct link:

http://ipexpert.acrobat.com/p93148979/

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Daniel
Berlinski
Sent: Thursday, September 30, 2010 5:23 PM
To: Amp
Cc: ccie_voice@onlinestudylist.com; Pithog Oil
Subject: Re: [OSL | CCIE_Voice] Speed for taking the Lab

 

HI Pithog

Here comes my suggestion:
Choose one lab and change the IP addresses in your environment, the
number plan, and the physical position of the phones you work with on
your desk, introduce infrastructure probs, make sure you make it very
hard for your phones to register.

Review the troubleshooting IP Tel book on chapter  3 I think speaks
about phone registration and there is a white paper on cisco.com that
talks about common phone registration probs, make sure you are familiar
with those and practice those scenarios so you can see the symptoms
happening before you in the stress room chamber

On Fri, Oct 1, 2010 at 9:38 AM, Amp amccar...@cciequest.com wrote:

Hey Pithog,
You ask a tough question my friend. I think some of the things that you
need to consider are how well do you know the core technologies and how
fast can you correctly configure them? Based upon the forums and the
practice labs there are going to be some things that you will need to
know how to configure rather swiftly. Will you have CME with SCCP and
SIP phones to configure on your lab? Who knows but it would be a good
idea to know how to configure CME in a matter of minutes. Can you
configure IOS media resources as fast as you can type your name? If not
then ask yourself why not. Start configuring H323, MGCP, and Gatekeepers
in notepad. If you can do it in notepad with little to no screw-ups then
you can do it in the router lightning fast. Are you able to read the
question and not over-complicate what's being asked? How fast can you
configure COR? Furthermore what's your strategy? Do you plan on
configuring once and copying, modifying, and pasting? What do you know
really well and what do you need help in? Spend as much time trying to
master the areas that you are weak in. Also remember, it is very
possible that the IPX labs are more difficult than the actual lab so if
you can't get through the IPX labs in less than 8 hours, can you do the
core of what's being asked in a timely manner? So in my opinion,
configure as much as you can in notepad to ensure you know the
configuration steps inside and out. During your steps write out the
steps to configure what's being asked. Do this without looking it up and
see where you are coming up short. I know I didn't directly answer your
question but I hope that helps.

Amp



Quoting Pithog Oil pithog...@yahoo.com:

Please i will like to know if my speed is okay and good enough for the
exam, 
it takes me 8 hours at the moment to finish the ipexpert, Labs,
suggestions are welcome on how i can shorthen the time to 4 hours, i
really hope its possible, please i need assistance on this.
Ultimately i want to know how to manage my time better.
Thanks







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Re: [OSL | CCIE_Voice] WHERE CAN I FIND THE UCCX LICENCE

[Steve Denney (stdenney)]

This is a great example of how searching the OSL archives first might
save you a lot of time and frustration...

http://www.mail-archive.com/search?l=ccie_vo...@onlinestudylist.comq=uc
cx+license+file

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Pithog Oil
Sent: Friday, September 17, 2010 2:43 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] WHERE CAN I FIND THE UCCX LICENCE

 

Hi Experts

 

Please i need someone to help me locate the UCCX license , i spent a
long time loking for it while working on Volume 1 lab 12 but i did not
find it.

 

Pithog oil

 

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Re: [OSL | CCIE_Voice] Layer 2 Overhead brainstorm

[Steve Denney (stdenney)]

The right value to use is the one that will get you the points in the
lab exam. And as Daniel pointed out yesterday, Ben Ng (the lab author)
has clearly stated that he uses the values in the QoS SRND.

 

You do have a certain amount of leeway in the calculations (some unknown
percentage factor allowed by the graders). Many of the IPexpert
materials use what they consider to be a closer to real-world value.
Will that value be within the leeway, and get you the points? That's up
to you to decide. Personally - I do wish that the materials were more
consistent in the values chosen. 

 

As has been stated countless times in the archives - The lab is not a
test of best practices or real world scenarios. It's a test to see how
well you can interpret and follow the blueprint and the directions that
are given. Draw your own conclusions. :)

 

good luck,

sd

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Tam Nhu
Sent: Wednesday, September 15, 2010 10:31 AM
To: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Layer 2 Overhead brainstorm

 

Still confusing.

We know that QoS SRND ues MLP overhead is 13 bytes, but the IPExpert PG
always uses 9 bytes.  

Also, for FRF.12, QoS SRND uses 4 bytes, but the PG uses 6 or 7 bytes,
change per lab basic.

So what is the right value to use? 

Thanks,
TN.

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Re: [OSL | CCIE_Voice] Voice L_A_B Q_U_E_R_I_E_S

[Steve Denney (stdenney)]

One general comment: I believe that the voice proctors will stop grading
an exam once it is clear that you will not have enough points to pass.
So once you have missed 20 points, any sections following that will
likely be shown with 0 points (meaning that they were not graded). You
may well have gotten the solution correct; your 0 simply means that
that section was not graded.

 

Disclaimer: I don't know this for sure, and have no specific knowledge
of test grading procedures - this is just based on my personal
experience and opinion.

 

Better luck on your next Disdo / Disco attempt :)

 

cheers, sd

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of voicerack
voicerack
Sent: Wednesday, July 21, 2010 11:05 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Voice L_A_B Q_U_E_R_I_E_S

 

Hi,

QUESTION 

1) Background image 

Background image on CME PHONES

Cisco do not want these files to be uploaded from CUCM but via SSH :-
HOW TO ACHIEVE THIS WITHOUT UPLOADING TO CUCM TFTP


MY COMMENTS

Why the hell cisco want us to use 3rd party when we can use CUCM??
Why they have nt mentioned the use of SSH in the blue-pint??

How to achieve this without using cucm TFTP??

 

2) MEET ME

LA-PH1 only can initiate the meet me conference The other users can call
the meet me number
and get connected to the conference. PSTN can also access the conference
bridge/
1234 is the number for the meet me.

Make sure when user join and leave the conference beeps are heard


SOLUTION 1


ephone-dn  7  octo-line
 number 1234 no-reg both
 conference meetme
 no huntstop

ephone-dn  8  octo-line
 number 1234 no-reg both
 conference meetme
 preference 1


voice class custom-cptone leavetone
dualtone conference
frequency 400 800
cadence 400 50 200 50
!
voice class custom-cptone jointone
dualtone conference
frequency 600 900
cadence 300 150 

dspfarm profile 2 conference  
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 codec g729r8
 codec g729br8
 maximum sessions 3
 conference-join custom-cptone jointone
 conference-leave custom-cptone leavetone
 associate application SCCP


SOLUTION 2

ephone-dn  7  octo-line
 number 1234 no-reg both
 conference meetme


voice class custom-cptone leavetone
dualtone conference
frequency 400 800
cadence 400 50 200 50 200 50
!
voice class custom-cptone jointone
dualtone conference
frequency 600 900
cadence 300 150 300 100 300 50

dspfarm profile 2 conference  
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 codec g729r8
 codec g729br8
 maximum sessions 3
 conference-join custom-cptone jointone
 conference-leave custom-cptone leavetone
 associate application SCCP


FOR BOTH SOLUTION DISDO IS GIVING 0  why why why?

 

 

3) Presence

 

a) LA-PH-2 should be able to monitor LA-PH1 line 1 

their should be a 3rd line on LA-PH-2 that monitors this phone.  

When you push this button it should speed dial to 4001.  

when 4001 is on the phone this button should show red


b. When LA-PH-1 line 1(4001) is on the phone you should see the status
of this call in the  
  local directory of phone 1 

SOLUTION 1

Presence

ip http server

sip-ua
 presence enable

presence
 presence call-list

telephony-service
  directory entery 1 4001 name SC Phone 1
  directory entery 2 4002 name SC Phone 2
  url directories http://142.102.66.254/localdirectory

ephone-dn  1 octo-line
  name SC Phone 1
  allow watch

ephone-dn 2
  name SC Phone 2
   allow watch
  
ephone  2
  blf-speed-dial 1 4001 label SpeedDial-4001

.SOLUTION 2

presence
 presence call-list

ephone-dn  1 octo-line
  name LA Phone 1
  allow watch

ephone-dn 2
  name LA Phone 2
   allow watch
  
ephone  2
Presenc call-list
butt 1:2 2:4 3m1


AGAIN THE QUESTION WHY THE HELL DISDO IS GIVING THIS WRONG, AFTER ASKING
TROCTOR he said if it is not define you can use any key word e.g 3w1 or
blf speed dial

Then why the hell we are not getting marks on the same.

 

4) QUESTION 

a) 
Queue 1 5 in the priority queue .
queue 2 4,6,7 
queue 3 2, 3  
queue 4 0 

b. guarantee Queue 1 has the 25% of the bandwidth. the other queues
should share the bandwidth as 30 40 30.

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

 

SOLUTION

mls qos
mls qos map cos-dscp 0 8 16 24 32 46 48 56
mls qos srr-queue output cos-map queue 1 threshold 3 5
mls qos srr-queue output cos-map queue 2 threshold 1 4
mls qos srr-queue output cos-map queue 2 threshold 3 6 7
mls qos srr-queue output cos-map queue 3 threshold 3 2 3
mls qos srr-queue output cos-map queue 4 threshold 3 0 


===

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


int f0/1
 *** connected to LA-RTR ***
 srr-queue bandwidth share 1 30 40 30
 srr-queue bandwidth shape 4 0 0 0
 queue-set 2
 mls qos trust dscp !!!
 

int range f0/13 - 15
 *** connected to IP Phones ***
 mls qos trust device cisco-phone
 mls qos trust cos


int range f0/3-4
 *** 

Re: [OSL | CCIE_Voice] isdn plan

[Steve Denney (stdenney)]

That triggered another thought... :)

What would the appropriate Calling type be, if (for example) you have a
phone, local to the egress gateway, calling an International number? 
My thought is that you would set Calling type=Subscriber (since it's a
local phone), or National (if you're using TEHO and going through
another in-country egress gateway), and Called type=International
(unless otherwise specified, of course). Or would it be more appropriate
to actually set the Calling type to International as well, since the
destination is an international number? That doesn't seem intuitive, but
not sure if I've seen this addressed anywhere.

thx, sd

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Berry,
Matthew J.
Sent: Friday, July 09, 2010 8:47 AM
To: Mark Holloway
Cc: OSL osl
Subject: Re: [OSL | CCIE_Voice] isdn plan

I set it for everything, but that's just me.

Matthew Berry, CCVP, Sr. Unified Communications Engineer
mjbe...@kroll.com

-Original Message-
From: Mark Holloway [mailto:m...@markholloway.com] 
Sent: Thursday, July 08, 2010 11:42 PM
To: Berry, Matthew J.
Cc: OSL osl
Subject: Re: [OSL | CCIE_Voice] isdn plan

Are you setting plan/type for both the called and calling numbers or
just one of them?  For example, if a task says the pstn provider wants
the called party number type set and you set the plan/type for the
called number, are you just leaving the calling portion set to
CallManager or are you setting the plan/type for that as well?


On Jul 7, 2010, at 11:43 AM, Berry, Matthew J. wrote:

 I make a habit of always setting the plan to ISDN.
 
 Matthew Berry, CCVP, Sr. Unified Communications Engineer 
 mjbe...@kroll.com
 
 
 -Original Message-
 From: ccie_voice-boun...@onlinestudylist.com 
 [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark 
 Holloway
 Sent: Wednesday, July 07, 2010 1:40 PM
 To: OSL osl
 Subject: [OSL | CCIE_Voice] isdn plan
 
 When tasked with setting the call type to unknown, subscriber,
national, or international, are you guys also setting the plan to isdn
or are you just specifying the type and leaving the plan as unknown even
though all the pstn access is isdn?
 
 
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Re: [OSL | CCIE_Voice] Taking the written Monday!

[Steve Denney (stdenney)]

Good job! Now the real fun begins :) :)

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Rob
Sent: Thursday, July 01, 2010 12:31 PM
To: OSL osl
Subject: Re: [OSL | CCIE_Voice] Taking the written Monday!

 

CCIE Written passed!!  Score was an 820!  =D

On Fri, Jun 25, 2010 at 10:21 PM, Jon1992 jon1...@hotmail.com wrote:

Rob

 

You can do it, get that # and come back a winner :-

 

Jon

 

From: Rob mailto:rloe...@gmail.com  

Sent: Saturday, June 26, 2010 10:53 AM

To: ccie_voice@onlinestudylist.com 

Subject: [OSL | CCIE_Voice] Taking the written Monday!

 

I am taking the written test on Monday at Cisco Live in Vegas.  I will
post how I did when I get back.

I want to thank you for sharing your knowledge here, it has truly been
helpful to my studies!

Rob



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[OSL | CCIE_Voice] X-Lite SIP Client not registering to BR2 CME

[Steve Denney (stdenney)]

I've noticed some odd behavior the last few weeks with my X-Lite SIP
client (BR2 Ph4, DN 3006; using software VPN to connect to the vRacks).

The X-Lite SIP client doesn't seem to want to register to the CME. It
comes up with the error Registration error 408 - Request Timeout on
the display.

 

The only other time I've hit this symptom was a couple of months ago,
when I and a few other folks hit the IP addressing issue discussed in
this thread:

http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg16149.html

 

Since then I've been careful to run this fix on the HQ and BR2 routers
before each lab, and none of the other symptoms mentioned in that thread
have occurred. 

But the Registration error 408 - Request Timeout has popped up again
for the last few weeks. 

Different pods, too: twice on pod 14 and twice on pod 12. 

 

As per another archive thread, I removed type 7960 from the voice
register pool 2 - didn't help. Have reloaded the router, no create prof
/ create prof (all the usual fixes). 

No issues with the BR2 SCCP (IPblue) softclient. 

Searched the archives and the Web but no further hits. Any ideas?

 

cheers, sd

 

 

Steve Denney, CISSP

Systems Engineer - Technology Solutions Network

Voice and Unified Communications Products

Cisco Systems, Inc.

125 High Street, 21st Floor

Boston, MA  02110

978-936-4048 (Office)

617-872-5031 (Mobile)

stden...@cisco.com mailto:stden...@cisco.com  

 

 

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Re: [OSL | CCIE_Voice] X-Lite SIP Client not registering to BR2 CME

[Steve Denney (stdenney)]

Forgot to attach BR2 config, sorry.

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Steve
Denney (stdenney)
Sent: Friday, June 11, 2010 8:38 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] X-Lite SIP Client not registering to BR2 CME

 

I've noticed some odd behavior the last few weeks with my X-Lite SIP
client (BR2 Ph4, DN 3006; using software VPN to connect to the vRacks).

The X-Lite SIP client doesn't seem to want to register to the CME. It
comes up with the error Registration error 408 - Request Timeout on
the display.

 

The only other time I've hit this symptom was a couple of months ago,
when I and a few other folks hit the IP addressing issue discussed in
this thread:

http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg16149.html

 

Since then I've been careful to run this fix on the HQ and BR2 routers
before each lab, and none of the other symptoms mentioned in that thread
have occurred. 

But the Registration error 408 - Request Timeout has popped up again
for the last few weeks. 

Different pods, too: twice on pod 14 and twice on pod 12. 

 

As per another archive thread, I removed type 7960 from the voice
register pool 2 - didn't help. Have reloaded the router, no create prof
/ create prof (all the usual fixes). 

No issues with the BR2 SCCP (IPblue) softclient. 

Searched the archives and the Web but no further hits. Any ideas?

 

cheers, sd

 

 

Steve Denney, CISSP

Systems Engineer - Technology Solutions Network

Voice and Unified Communications Products

Cisco Systems, Inc.

125 High Street, 21st Floor

Boston, MA  02110

978-936-4048 (Office)

617-872-5031 (Mobile)

stden...@cisco.com mailto:stden...@cisco.com  

 

 

BR2-RTR#
BR2-RTR#sh cdp ne
Capability Codes: R - Router, T - Trans Bridge, B - Source Route Bridge
  S - Switch, H - Host, I - IGMP, r - Repeater

Device IDLocal Intrfce HoldtmeCapability  Platform  Port ID
SEP001193B6EC51  Fas 0/3/0  157   H   IP Phone  Port 1
SEP0002FD3BA793  Fas 0/3/1  161   H   IP Phone  Port 1
HQ-RTR   Ser 0/1/0:0.1  129 R S I 2811  Ser 
0/0/1:0.2


BR2-RTR#sh run
Building configuration...

Current configuration : 6564 bytes
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname BR2-RTR
!
boot-start-marker
warm-reboot
boot-end-marker
!
logging message-counter syslog
logging buffered 4096
!
no aaa new-model
memory-size iomem 20
clock timezone CET 1
clock summer-time CET recurring 1 Sun Apr 1:00 last Sun Oct 1:00
network-clock-participate wic 0 
no network-clock-participate wic 1 
!
dot11 syslog
ip source-route
!
!
ip cef
ip dhcp excluded-address 10.10.202.1 10.10.202.49
ip dhcp excluded-address 10.10.202.70 10.10.202.254
!
ip dhcp pool CME
   network 10.10.202.0 255.255.255.0
   default-router 10.10.202.1 
   option 150 ip 10.10.110.3 
!
!
no ip domain lookup
no ipv6 cef
!
multilink bundle-name authenticated
!
!
isdn switch-type primary-net5
!
!
!
voice service voip 
 sip
  registrar server
!
!
!
!
!
!
!
!
!
!
!
!
!
voice register global
 mode cme
 source-address 10.10.202.1 port 5060
 max-dn 2
 max-pool 2
 load 7960-7940 P0S3-08-9-00
 authenticate register
 timezone 13
 time-format 24
 date-format D/M/Y
 tftp-path flash:
 create profile sync 300632851775
 ntp-server 10.10.100.2 mode unicast
!
voice register dn  1
 number 3005
 name br2 phn2
!
voice register dn  2
 number 3006
 name br2 phn4
!
voice register template  1
 dialplan 1
!
voice register dialplan  1
 type 7940-7960-others
 pattern 1 3...
!
voice register pool  1
 id mac 001B.D4C6.C139
 type 7960
 number 1 dn 1
 template 1
 dtmf-relay rtp-nte
 username 3005 password cisco
 description 32143005
 codec g711ulaw
!
voice register pool  2
 id mac 000D.299D.348C
 number 1 dn 2
 template 1
 dtmf-relay rtp-nte
 username 3006 password cisco
 description 32143006
 codec g711ulaw
!
!
!
voice-card 0
 no dspfarm
 dsp services dspfarm
!
!
!
!
!
archive
 log config
  hidekeys
! 
!
!
!
!
controller E1 0/0/0
 pri-group timeslots 1-3,16
!
controller T1 0/1/0
 framing esf
 linecode b8zs
 channel-group 0 timeslots 1-24
!
!
!
!
!
interface Loopback0
 ip address 10.10.110.3 255.255.255.255
 ip ospf network point-to-point
!
interface FastEthernet0/0
 no ip address
 shutdown
 duplex auto
 speed auto
!
interface Service-Engine0/0
 no ip address
!
interface FastEthernet0/1
 no ip address
 shutdown
 duplex auto
 speed auto
!
interface FastEthernet0/3/0
 switchport trunk native vlan 200
 switchport mode trunk
 switchport voice vlan 400
 switchport priority extend cos 0
 shutdown
!
interface FastEthernet0/3/1
 switchport trunk native vlan 200
 switchport mode trunk
 switchport voice vlan 400
 switchport priority extend cos 0
 shutdown
!
interface FastEthernet0/3/2
 shutdown
!
interface FastEthernet0/3/3
 shutdown
!
interface Serial0/0/0:15
 no ip

[OSL | CCIE_Voice] CUE - Destination unreachable error

[Steve Denney (stdenney)]

Working on Vol 2 Lab 2 Question 8.2.

 

When trying to session into the CUE, I get this error:

BR2-RTR#service-module service-Engine 0/0 sess  

Trying 10.10.202.1, 2194 ... 

% Destination unreachable; gateway or host down

 

Module status looks good:

BR2-RTR#service-module service-Engine 0/0 status

Service Module is Cisco Service-Engine0/0

Service Module supports session via TTY line 194

Service Module is in Steady state

Service Module heartbeat-reset is enabled

Getting status from the Service Module, please wait..

 

Cisco Unity Express 7.0.1

CUE Running on AIM

 

IP route looks good:

BR2-RTR#sh ip route 10.10.202.2

Routing entry for 10.10.202.2/32

  Known via static, distance 1, metric 0 (connected)

  Routing Descriptor Blocks:

  * directly connected, via Service-Engine0/0

  Route metric is 0, traffic share count is 1

 

Config is plain enough:

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

 

Have reloaded the router, and did a ser ser 0/0 reset - still no joy.

What obvious thing am I missing?

 

cheers, sd

 

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Re: [OSL | CCIE_Voice] CUE - Destination unreachable error

[Steve Denney (stdenney)]

Bingo!  I had shut down both of the remote BR2 phones earlier, thus no
active endpoints on Vlan400, and thus...Vlan400 was up | down. Great
catch Amy, thanks!

 

cheers, sd

 

 

From: Amy Ryan [mailto:ar...@ipexpert.com] 
Sent: Friday, June 11, 2010 3:03 PM
To: Steve Denney (stdenney); ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CUE - Destination unreachable error

 

Do a sh ip int bri  is vlan 400 up up?

---
Amy Ryan - CCIE #24677 (Voice)
Technical Instructor - IPexpert, Inc.
Mailto: ar...@ipexpert.com
Telephone: +1.810.326.1444
Live Assistance, Please visit: www.ipexpert.com/chat
http://www.ipexpert.com/chat 
eFax: +1.810.454.0130 

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, Security  Service Provider) certification(s) with
training locations throughout the United States, Europe, South Asia and
Australia. Be sure to visit our online communities at
www.ipexpert.com/communities http://www.ipexpert.com/communities  and
our public website at www.ipexpert.com http://www.ipexpert.com/  






From: Steve Denney (stdenney) stden...@cisco.com
Date: Fri, 11 Jun 2010 13:56:42 -0500
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] CUE - Destination unreachable error

Working on Vol 2 Lab 2 Question 8.2.
 
When trying to session into the CUE, I get this error:
BR2-RTR#service-module service-Engine 0/0 sess  
Trying 10.10.202.1, 2194 ... 
% Destination unreachable; gateway or host down

Module status looks good:
BR2-RTR#service-module service-Engine 0/0 status
Service Module is Cisco Service-Engine0/0
Service Module supports session via TTY line 194
Service Module is in Steady state
Service Module heartbeat-reset is enabled
Getting status from the Service Module, please wait..
 
Cisco Unity Express 7.0.1
CUE Running on AIM
 
IP route looks good:
BR2-RTR#sh ip route 10.10.202.2
Routing entry for 10.10.202.2/32
 Known via static, distance 1, metric 0 (connected)
 Routing Descriptor Blocks:
 * directly connected, via Service-Engine0/0
 Route metric is 0, traffic share count is 1
 
Config is plain enough:
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
 
Have reloaded the router, and did a ser ser 0/0 reset - still no joy.
What obvious thing am I missing?
 
cheers, sd





___
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please visit www.ipexpert.com

___
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Re: [OSL | CCIE_Voice] CUE - Destination unreachable error

[Steve Denney (stdenney)]

By the way - This fix had the added bonus of also fixing my X-Lite SIP
Client not registering to BR2 CME issue from earlier.

 

Amazing how much stuff breaks when the Vlan is down...sheesh :)

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Steve
Denney (stdenney)
Sent: Friday, June 11, 2010 3:09 PM
To: Amy Ryan; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CUE - Destination unreachable error

 

Bingo!  I had shut down both of the remote BR2 phones earlier, thus no
active endpoints on Vlan400, and thus...Vlan400 was up | down. Great
catch Amy, thanks!

 

cheers, sd

 

 

From: Amy Ryan [mailto:ar...@ipexpert.com] 
Sent: Friday, June 11, 2010 3:03 PM
To: Steve Denney (stdenney); ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CUE - Destination unreachable error

 

Do a sh ip int bri  is vlan 400 up up?

---
Amy Ryan - CCIE #24677 (Voice)
Technical Instructor - IPexpert, Inc.
Mailto: ar...@ipexpert.com
Telephone: +1.810.326.1444
Live Assistance, Please visit: www.ipexpert.com/chat
http://www.ipexpert.com/chat 
eFax: +1.810.454.0130 

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, Security  Service Provider) certification(s) with
training locations throughout the United States, Europe, South Asia and
Australia. Be sure to visit our online communities at
www.ipexpert.com/communities http://www.ipexpert.com/communities  and
our public website at www.ipexpert.com http://www.ipexpert.com/  





From: Steve Denney (stdenney) stden...@cisco.com
Date: Fri, 11 Jun 2010 13:56:42 -0500
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] CUE - Destination unreachable error

Working on Vol 2 Lab 2 Question 8.2.
 
When trying to session into the CUE, I get this error:
BR2-RTR#service-module service-Engine 0/0 sess  
Trying 10.10.202.1, 2194 ... 
% Destination unreachable; gateway or host down

Module status looks good:
BR2-RTR#service-module service-Engine 0/0 status
Service Module is Cisco Service-Engine0/0
Service Module supports session via TTY line 194
Service Module is in Steady state
Service Module heartbeat-reset is enabled
Getting status from the Service Module, please wait..
 
Cisco Unity Express 7.0.1
CUE Running on AIM
 
IP route looks good:
BR2-RTR#sh ip route 10.10.202.2
Routing entry for 10.10.202.2/32
 Known via static, distance 1, metric 0 (connected)
 Routing Descriptor Blocks:
 * directly connected, via Service-Engine0/0
 Route metric is 0, traffic share count is 1
 
Config is plain enough:
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
 
Have reloaded the router, and did a ser ser 0/0 reset - still no joy.
What obvious thing am I missing?
 
cheers, sd



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

___
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Re: [OSL | CCIE_Voice] X-Lite SIP Client not registering to BR2 CME [RESOLVED]

[Steve Denney (stdenney)]

Just responding for the archives:

 

Interface Vlan400 was up | down. I had shut down the phones connected to
the BR2 ESW and never brought them back up. This put Vlan400 in up |
down state, preventing my local client from registering.

Doing a no shut on one of the BR2 phones (fa0/3/0 or 0/3/1) brought the
Vlan400 up | up, and the X-Lite registered immediately. :)

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Steve
Denney (stdenney)
Sent: Friday, June 11, 2010 8:38 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] X-Lite SIP Client not registering to BR2 CME

 

I've noticed some odd behavior the last few weeks with my X-Lite SIP
client (BR2 Ph4, DN 3006; using software VPN to connect to the vRacks).

The X-Lite SIP client doesn't seem to want to register to the CME. It
comes up with the error Registration error 408 - Request Timeout on
the display.

 

The only other time I've hit this symptom was a couple of months ago,
when I and a few other folks hit the IP addressing issue discussed in
this thread:

http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg16149.html

 

Since then I've been careful to run this fix on the HQ and BR2 routers
before each lab, and none of the other symptoms mentioned in that thread
have occurred. 

But the Registration error 408 - Request Timeout has popped up again
for the last few weeks. 

Different pods, too: twice on pod 14 and twice on pod 12. 

 

As per another archive thread, I removed type 7960 from the voice
register pool 2 - didn't help. Have reloaded the router, no create prof
/ create prof (all the usual fixes). 

No issues with the BR2 SCCP (IPblue) softclient. 

Searched the archives and the Web but no further hits. Any ideas?

 

cheers, sd

 

 

Steve Denney, CISSP

Systems Engineer - Technology Solutions Network

Voice and Unified Communications Products

Cisco Systems, Inc.

125 High Street, 21st Floor

Boston, MA  02110

978-936-4048 (Office)

617-872-5031 (Mobile)

stden...@cisco.com mailto:stden...@cisco.com  

 

 

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For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] FRTS and MLP over a Serial with Sub-Interfaces

[Steve Denney (stdenney)]

Hi Matthew,

Volume 2 Lab 1 Question 5.3 covers this scenario. If one of the
sub-interfaces uses FRTS, you'll indeed have to use FRTS on both - but
this does not preclude using FRF.12. 

The proctor guide has a good explanation.

cheers, sd

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Matthew
Berry
Sent: Thursday, June 10, 2010 10:36 AM
To: OSL Group
Subject: [OSL | CCIE_Voice] FRTS and MLP over a Serial with
Sub-Interfaces

 

Quick question.

In the lab, if the HQ site is setup with two sub-interfaces that connect
to BR1 and BR2 (i.e. meaning, they're both running off the same
interface), how would you configure MLP for one site and FRF.12 for
another site?

According to my understanding, MLP will require that frame-relay
traffic-shaping is enabled on the serial interface.  However, this
would botch up your FRF.12 configuration on the other sub-interface.

QoS is a weak  area for me so I might be missing something obvious in
this question.  However, it came up so I thought I would ask.

Thanks

-- 



Matthew Berry

A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written

 

Vitals:

GVoice: +1.612.424.5044

Gmail: ciscovoiceg...@gmail.com

Skype: ciscovoiceguru

Twitter: ciscovoiceguru

 

Cert Stats:

Cisco Cert Journey Began: Jan 1, 2009

1st Lab Attempt: Aug 16, 2010

___
For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] Vol1 Lab 4A - X-Lite Issues

[Steve Denney (stdenney)]

Hi Brian,

I had run out of time on that lab, but some replies I got afterwards suggested 
that the most likely culprit was a codec mismatch. Check the codec settings 
between the regions, and the codec that the X-Lite client is set to use, and 
set up a transcoder if need be. 

Also ensure that any SIP dial peers are using a hard-coded codec and not voice 
class codec to set a codec preference - although it looks fine and won't give 
you an error message, the dial peer apparently won't actually use the voice 
class codec settings, so you may have a mismatch that isn't obvious.

cheers, sd


Steve Denney, CISSP
Systems Engineer - Technology Solutions Network
Voice and Unified Communications Products
Cisco Systems, Inc.
125 High Street, 21st Floor
Boston, MA  02110
978-936-4048 (Office)
617-872-5031 (Mobile)
stden...@cisco.com 
 

-Original Message-
From: Brian Valentine [mailto:bkvalent...@gmail.com] 
Sent: Saturday, May 15, 2010 9:40 PM
To: Steve Denney (stdenney)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Vol1 Lab 4A - X-Lite Issues

Steve,

I'm hitting this tonight.  Ever figure it out?

Thanks,

Brian

On Thu, Feb 4, 2010 at 4:24 PM, Steve Denney (stdenney)
stden...@cisco.com wrote:
 Hitting an interesting problem and just wondering if anyone else has seen
 similar symptoms...



 Working on Vol1, Lab 4A, Task 4.5.

 This is the task where you set up a SIP Route Pattern and use SIP URI
 dialing to dial the X-Lite CME SIP Phone (BR2 Ph 4, DN 3006) from the CIPC
 SIP Phone (HQ Ph2, DN 5002).



 When dialing from 5002 to 3006 (using the corporate directory on CIPC, as
 shown in the lab), the X-Lite rings, but hangs up immediately after the call
 is answered.

 The output of debug ccsip mess is attached. Looks like the X-Lite is sending
 a SIP BYE message with the description of Illegal Sdp Negotiation.



 I tried a call in the other direction as well - direct dial from 3006 to
 5002. The CIPC rings, but you cannot actually answer the call.

 The debug in this case shows a 503 Service Unavailable message, and the
 display on the X-Lite says Call failed: Service Unavailable.



 I've double and triple checked all configs (including allow-connections sip
 to sip), reloaded all routers, Googled for similar issues, and am now
 officially stumped. :)

 Debugs attached. Any ideas?



 cheers, steve





 ___
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 visit www.ipexpert.com


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[OSL | CCIE_Voice] Issues with accessing voice racks this morning?

[Steve Denney (stdenney)]

Anyone else having issues with accessing voice racks this morning? I can
ping proctorlabs.com, but can't get the Web site to come up at all...

 

thx, sd

 

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Re: [OSL | CCIE_Voice] Issues with accessing voice racks this morning?

[Steve Denney (stdenney)]

Back up now. Lost 45 mins of lab time though :(

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Steve
Denney (stdenney)
Sent: Friday, May 14, 2010 8:15 AM
To: ccie_voice@onlinestudylist.com
Cc: supp...@ipexpert.com
Subject: [OSL | CCIE_Voice] Issues with accessing voice racks this
morning?

 

Anyone else having issues with accessing voice racks this morning? I can
ping proctorlabs.com, but can't get the Web site to come up at all...

 

thx, sd

 

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[OSL | CCIE_Voice] Vol 2 Lab 1 Question 4.1 - Calls from PSTN to HQ failing

[Steve Denney (stdenney)]

Sigh...another week, another strange behavior from my PSTN phone :)

When placing a call from the PSTN phone (from any line) to any HQ phone
(5001 or 5002), I'm getting the following strange symptoms.

 

I hear ringback on the PSTN phone very briefly, then it shows
Connected but I hear dial tone - as if the PSTN line is simply off
hook.

After about 10 seconds I hear a fast busy. The phone at HQ never rings.

 

Strangely, the debug isdn q931 output from both the PSTN and HQ gateways
looks like a successful call (see attached).

So it seems like the call is hitting the HQ GW, but the HQ phone never
actually rings.

 

On CUCM, the significant digits for inbound calls is set to 4.

 

Calls from PSTN to BR1 phones are working fine.

 

I've reset and restarted everything (all routers, HQ switch, Pub, and
Sub). 

Yes, I double checked the linecode, framing, and switch type settings
after last week's embarrassing incident :)

There are no remote destinations set up that might be causing a conflict
with any of the PSTN numbers.

Show isdn status looks normal from both sides.

 

I've run out of troubleshooting ideas - any suggestions?

 

cheers, sd

 

=
debug isdn q931 from PSTN router:

May 14 21:34:17.806: ISDN Se0/3/0:23 Q931: Applying typeplan for sw-type 0xD is 
0x0 0x0, Calling num 911
May 14 21:34:17.806: ISDN Se0/3/0:23 Q931: Applying typeplan for sw-type 0xD is 
0x2 0x1, Called num 2123945002
May 14 21:34:17.806: ISDN Se0/3/0:23 Q931: TX - SETUP pd = 8  callref = 0x0099 
Bearer Capability i = 0x8090A2 
Standard = CCITT 
Transfer Capability = Speech  
Transfer Mode = Circuit 
Transfer Rate = 64 kbit/s 
Channel ID i = 0xA98381 
Exclusive, Channel 1 
Progress Ind i = 0x8583 - Origination address is non-ISDN  
Display i = 'Emergency Services' 
Calling Party Number i = 0x0080, '911' 
Plan:Unknown, Type:Unknown 
Called Party Number i = 0xA1, '2123945002' 
Plan:ISDN, Type:National
May 14 21:34:17.838: ISDN Se0/3/0:23 Q931: RX - CALL_PROC pd = 8  callref = 
0x8099 
Channel ID i = 0xA98381 
Exclusive, Channel 1
May 14 21:34:17.842: ISDN Se0/3/0:23 Q931: RX - CONNECT pd = 8  callref = 
0x8099
May 14 21:34:17.846: %ISDN-6-CONNECT: Interface Serial0/3/0:0 is now connected 
to 2123945002 N/A

May 14 21:34:17.846: ISDN Se0/3/0:23 Q931: TX - CONNECT_ACK pd = 8  callref = 
0x0099

May 14 21:34:23.846: %ISDN-6-CONNECT: Interface Serial0/3/0:0 is now connected 
to 2123945002 N/A

May 14 21:34:29.594: %ISDN-6-DISCONNECT: Interface Serial0/3/0:0  disconnected 
from 2123945002 , call lasted 11 seconds

May 14 21:34:29.598: ISDN Se0/3/0:23 Q931: TX - DISCONNECT pd = 8  callref = 
0x0099 
Cause i = 0x8290 - Normal call clearing
May 14 21:34:29.610: ISDN Se0/3/0:23 Q931: RX - RELEASE pd = 8  callref = 
0x8099
May 14 21:34:29.614: ISDN Se0/3/0:23 Q931: TX - RELEASE_COMP pd = 8  callref = 
0x0099

=
debug isdn q931 from HQ router:

*May 14 17:33:10.023: ISDN Se0/0/0:23 Q931: RX - SETUP pd = 8  callref = 
0x0099 
Bearer Capability i = 0x8090A2 
Standard = CCITT 
Transfer Capability = Speech  
Transfer Mode = Circuit 
Transfer Rate = 64 kbit/s 
Channel ID i = 0xA98381 
Exclusive, Channel 1 
Progress Ind i = 0x8583 - Origination address is non-ISDN  
Display i = 'Emergency Services' 
Calling Party Number i = 0x0080, '911' 
Plan:Unknown, Type:Unknown 
Called Party Number i = 0xA1, '2123945002' 
Plan:ISDN, Type:National
*May 14 17:33:10.039: ISDN Se0/0/0:23 Q931: TX - CALL_PROC pd = 8  callref = 
0x8099 
Channel ID i = 0xA98381 
Exclusive, Channel 1
*May 14 17:33:10.039: ISDN Se0/0/0:23 Q931: TX - CONNECT pd = 8  callref = 
0x8099
*May 14 17:33:10.055: ISDN Se0/0/0:23 Q931: RX - CONNECT_ACK pd = 8  callref = 
0x0099
HQ-RTR#
*May 14 17:33:10.055: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now connected 
to 911 N/A
HQ-RTR#
*May 14 17:33:16.055: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now connected 
to 911 N/A
HQ-RTR#
*May 14 17:33:21.803: ISDN Se0/0/0:23 Q931: RX - DISCONNECT pd = 8  callref = 
0x0099 
Cause i = 0x8290 - Normal call clearing
*May 14 17:33:21.807: %ISDN-6-DISCONNECT: Interface Serial0/0/0:0  disconnected 
from 911 , call lasted 11 seconds
HQ-RTR#
*May 14 17:33:21.811: ISDN Se0/0/0:23 Q931: TX - RELEASE pd = 8  callref = 
0x8099
*May 14 17:33:21.819: ISDN Se0/0/0:23 Q931: RX - RELEASE_COMP pd = 8  callref 
= 0x0099
HQ-RTR#

=
Config from HQ-RTR:

HQ-RTR#sh run
Building configuration...
Current configuration : 3091 bytes
!
version 12.4
service timestamps debug datetime msec
service 

Re: [OSL | CCIE_Voice] Vol 2 Lab 1 Question 4.1 - Calls from PSTN to HQ failing

[Steve Denney (stdenney)]

D'oh!  

 

I swear I configured the inbound pots dial-peer - I really did. I
suspect someone sneaked in while I was not looking and deleted it. ;)

Thanks for the brain check!

 

cheers, sd

 

From: Devin Chamberlain [mailto:dchamberl...@unislumin.com] 
Sent: Friday, May 14, 2010 1:50 PM
To: Steve Denney (stdenney); ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] Vol 2 Lab 1 Question 4.1 - Calls from
PSTN to HQ failing

 

Is the HQ gateway a h.323 gateway ... if it is it sounds like you do not
have an inbound DID dial-peer.  If you do not have you dial-peer one
will match and that will give you the dial tone.

 

Devin Chamberlain
(403) 560-8599

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Steve
Denney (stdenney)
Sent: Friday, May 14, 2010 11:43 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Vol 2 Lab 1 Question 4.1 - Calls from PSTN
to HQ failing

 

Sigh...another week, another strange behavior from my PSTN phone :)

When placing a call from the PSTN phone (from any line) to any HQ phone
(5001 or 5002), I'm getting the following strange symptoms.

 

I hear ringback on the PSTN phone very briefly, then it shows
Connected but I hear dial tone - as if the PSTN line is simply off
hook.

After about 10 seconds I hear a fast busy. The phone at HQ never rings.

 

Strangely, the debug isdn q931 output from both the PSTN and HQ gateways
looks like a successful call (see attached).

So it seems like the call is hitting the HQ GW, but the HQ phone never
actually rings.

 

On CUCM, the significant digits for inbound calls is set to 4.

 

Calls from PSTN to BR1 phones are working fine.

 

I've reset and restarted everything (all routers, HQ switch, Pub, and
Sub). 

Yes, I double checked the linecode, framing, and switch type settings
after last week's embarrassing incident :)

There are no remote destinations set up that might be causing a conflict
with any of the PSTN numbers.

Show isdn status looks normal from both sides.

 

I've run out of troubleshooting ideas - any suggestions?

 

cheers, sd

 

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Re: [OSL | CCIE_Voice] Problem in LAB 4.3 A vol 1

[Steve Denney (stdenney)]

Do a show isdn status. 

 

If you don't see MULTIPLE_FRAME_ESTABLISHED state under Layer 2 Status,
then do a no isdn bind-l3 ccm-manager command under interface
Serial0/0/0:23, wait a few seconds, then put the bind-l3 command back. 

 

I find this usually works (much faster than reloading the router).

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of amr gaber
Sent: Friday, May 07, 2010 4:42 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Problem in LAB 4.3 A vol 1

 

Dear,
in this lab section we should add BR1-RTR as an MGCP gateway.
I made the configuration AS Proctorlab guide and I got gateway register
(sh ccm-manager)
But when I (debug isdn q931) nothing appeared  (I dialed as match as the
Route Pattern)

I attach the BR1 Configuration
Please advice



Amt THABT
CCNP, CCVP ,CUSS,UCCX,CCNA, MCP

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[OSL | CCIE_Voice] Vol 2 Lab 1 - ISDN Restart Errors

[Steve Denney (stdenney)]

Hi,

 

Seeing some errors today that I haven't encountered before in any other
lab...wh! :)

 

I'm working on Vol 2 Lab 1 Question 4.1, and trying to get calls from
the PSTN working into HQ. 

Pretty straightforward stuff, except the calls never seem to get out of
the PSTN router. 

 

When dialing the HQ phone from the PSTN phone (regardless of line
selected), I get the following debug isdn q931 errors from the PSTN
router:

May  6 23:11:09.243: ISDN Se0/3/0:23 Q931: Applying typeplan for sw-type
0xD is 0x0 0x0, Calling num 911

May  6 23:11:09.243: ISDN Se0/3/0:23 **ERROR**: CCPMSG_OutCall: fails
with cause 0x22

 

And every 30 seconds, I see the same batch of 4 ISDN Restart messages,
like this (also from the PSTN router):

May  6 23:12:04.539: ISDN Se0/3/0:23 Q931: TX - RESTART pd = 8  callref
= 0x 

Restart Indicator i = 0x87

May  6 23:12:05.539: ISDN Se0/3/0:23 Q931: TX - RESTART pd = 8  callref
= 0x 

Restart Indicator i = 0x87

 

May  6 23:12:06.539: ISDN Se0/3/0:23 Q931: TX - RESTART pd = 8  callref
= 0x 

Restart Indicator i = 0x87

May  6 23:12:07.539: ISDN Se0/3/0:23 Q931: TX - RESTART pd = 8  callref
= 0x 

Restart Indicator i = 0x87

 

Show isdn status on the PSTN router looks normal for this interface:

ISDN Serial0/3/0:23 interface

*** Network side configuration *** 

dsl 1, interface ISDN Switchtype = primary-ni

Layer 1 Status:

ACTIVE

Layer 2 Status:

TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED

Layer 3 Status:

0 Active Layer 3 Call(s)

Active dsl 1 CCBs = 0

The Free Channel Mask:  0x8000

Number of L2 Discards = 0, L2 Session ID = 0

 

 

Attaching show run and show isdn status as well for the HQ router (the
other end) just for troubleshooting completeness, but there's no
indication of anything amiss, nor any debug messages at all, on the HQ
router. The call never gets that far.

 

I started this morning on Voice Pod 11 and hit this. Ryan was kind
enough to move me over to Voice Pod 16, but I'm hitting the same issue
here. 

OSL archive and Google search turned up nothing concrete, other than a
general theme of it sounds like your telco / carrier has issues.  :)

 

Any ideas?

 

Cheers and TIA, sd

 

HQ-RTR#sh isdn stat

Global ISDN Switchtype = primary-ni
ISDN Serial0/0/0:23 interface
dsl 0, interface ISDN Switchtype = primary-ni
Layer 1 Status:
ACTIVE
Layer 2 Status:
TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
Layer 3 Status:
0 Active Layer 3 Call(s)
Active dsl 0 CCBs = 0
The Free Channel Mask:  0x8007
Number of L2 Discards = 0, L2 Session ID = 26
Total Allocated ISDN CCBs = 0


HQ-RTR#sh run
Building configuration...

Current configuration : 3412 bytes
!
! Last configuration change at 19:00:05 UTC Thu May 6 2010
! NVRAM config last updated at 19:00:07 UTC Thu May 6 2010
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname HQ-RTR
!
boot-start-marker
warm-reboot
boot-end-marker
!
logging buffered 51200 warnings
!
no aaa new-model
memory-size iomem 20
network-clock-participate wic 0 
network-clock-select 1 T1 0/0/0
dot11 syslog
no ip source-route
!
!
ip cef
!
!
no ip domain lookup
!
multilink bundle-name authenticated
!
isdn switch-type primary-ni
!
voice-card 0
 no dspfarm
 dsp services dspfarm
!
!
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8
!
!
voice class h323 1
 h225 timeout tcp establish 3
!
!
vtp domain home.com
vtp mode transparent
archive
 log config
  hidekeys
! 
!
controller T1 0/0/0
 framing esf
 linecode ami
 pri-group timeslots 1-3,24
!
controller T1 0/0/1
 framing esf
 linecode b8zs
 channel-group 0 timeslots 1-24
!
!
interface Loopback0
 ip address 10.10.110.1 255.255.255.255
!
interface FastEthernet0/0
 no ip address
 duplex full
 speed 100
 ip rsvp bandwidth
!
interface FastEthernet0/0.10
 encapsulation dot1Q 10 native
 ip address 10.10.100.1 255.255.255.0
!
interface FastEthernet0/0.20
 encapsulation dot1Q 20
 ip address 10.10.200.3 255.255.255.0
 ip helper-address 10.10.210.11
!
interface FastEthernet0/0.30
 encapsulation dot1Q 30
 ip address 10.10.210.1 255.255.255.0
!
interface FastEthernet0/1
 no ip address
 shutdown
 duplex auto
 speed auto
!
interface Serial0/0/0:23
 no ip address
 encapsulation hdlc
 isdn switch-type primary-ni
 isdn incoming-voice voice
 isdn outgoing display-ie
 no cdp enable
!
interface Serial0/0/1:0
 no ip address
 encapsulation frame-relay
 fair-queue 64 256 36
 frame-relay lmi-type ansi
 ip rsvp bandwidth
!
interface Serial0/0/1:0.1 point-to-point
 ip address 10.10.111.1 255.255.255.0
 ip ospf mtu-ignore
 snmp trap link-status
 frame-relay interface-dlci 201   
!
interface Serial0/0/1:0.2 point-to-point
 ip address 10.10.112.1 255.255.255.0
 ip ospf mtu-ignore
 snmp trap link-status
 

Re: [OSL | CCIE_Voice] Vol 2 Lab 1 - ISDN Restart Errors

[Steve Denney (stdenney)]

Good points, Jeff and vccie, and thanks. But if this is occurring on the
PSTN-WAN router...well...how could that config possibly be wrong? :)

 

Here's the relevant sections from the PSTN router:

controller T1 0/3/0

 framing esf

 clock source internal

 linecode b8zs

 pri-group timeslots 1-3,23-24

 description ** T1 VOICE CONNECTION TO HQ-RTR **

 

interface Serial0/3/0:23

 description ** T1 PRI D-CHANNEL TO HQ-RTR **

 no ip address

 encapsulation hdlc

 isdn switch-type primary-ni

 isdn protocol-emulate network

 isdn incoming-voice voice

 isdn outgoing display-ie

 no cdp enable

 

And from the HQ-RTR:

 

controller T1 0/0/0

 framing esf

 linecode ami

 pri-group timeslots 1-3,24

 

interface Serial0/0/0:23

 no ip address

 encapsulation hdlc

 isdn switch-type primary-ni

 isdn incoming-voice voice

 no cdp enable

 

The only thing that looks a little odd to me is the pri-group timeslots
1-3,23-24 on the PSTN T1 controller - channel 23 being a backup D
channel or something (not that I could change this even if I wanted to
without write access).

 

And I've also restarted the PSTN router, to no avail. 

 

Any other ideas?

thx, sd

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Price
(jeffpric)
Sent: Thursday, May 06, 2010 3:41 PM
To: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Vol 2 Lab 1 - ISDN Restart Errors

 

This is true.  I forgot about this.  Normally it has to do with
oversubscribing the DSP resources.  Meaning, make sure you aren't using
all of them and the PSTN has enough to use.  Try decreasing the amount
of channels you create under the pri-group timeslots command.  Good
point.

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of vccie2010
Sent: Thursday, May 06, 2010 12:39 PM
To: Steve Denney (stdenney)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Vol 2 Lab 1 - ISDN Restart Errors

 

cause 0x22 - generaly means no channel availableplease checj you
have right number of slots defined in pri-group timeslots statement
and double check your PSTN router for same.

On Thu, May 6, 2010 at 12:26 PM, Steve Denney (stdenney)
stden...@cisco.com wrote:

Hi,

 

Seeing some errors today that I haven't encountered before in any other
lab...wh! :)

 

I'm working on Vol 2 Lab 1 Question 4.1, and trying to get calls from
the PSTN working into HQ. 

Pretty straightforward stuff, except the calls never seem to get out of
the PSTN router. 

 

When dialing the HQ phone from the PSTN phone (regardless of line
selected), I get the following debug isdn q931 errors from the PSTN
router:

May  6 23:11:09.243: ISDN Se0/3/0:23 Q931: Applying typeplan for sw-type
0xD is 0x0 0x0, Calling num 911

May  6 23:11:09.243: ISDN Se0/3/0:23 **ERROR**: CCPMSG_OutCall: fails
with cause 0x22

 

And every 30 seconds, I see the same batch of 4 ISDN Restart messages,
like this (also from the PSTN router):

May  6 23:12:04.539: ISDN Se0/3/0:23 Q931: TX - RESTART pd = 8  callref
= 0x 

Restart Indicator i = 0x87

May  6 23:12:05.539: ISDN Se0/3/0:23 Q931: TX - RESTART pd = 8  callref
= 0x 

Restart Indicator i = 0x87

 

May  6 23:12:06.539: ISDN Se0/3/0:23 Q931: TX - RESTART pd = 8  callref
= 0x 

Restart Indicator i = 0x87

May  6 23:12:07.539: ISDN Se0/3/0:23 Q931: TX - RESTART pd = 8  callref
= 0x 

Restart Indicator i = 0x87

 

Show isdn status on the PSTN router looks normal for this interface:

ISDN Serial0/3/0:23 interface

*** Network side configuration *** 

dsl 1, interface ISDN Switchtype = primary-ni

Layer 1 Status:

ACTIVE

Layer 2 Status:

TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED

Layer 3 Status:

0 Active Layer 3 Call(s)

Active dsl 1 CCBs = 0

The Free Channel Mask:  0x8000

Number of L2 Discards = 0, L2 Session ID = 0

 

 

Attaching show run and show isdn status as well for the HQ router (the
other end) just for troubleshooting completeness, but there's no
indication of anything amiss, nor any debug messages at all, on the HQ
router. The call never gets that far.

 

I started this morning on Voice Pod 11 and hit this. Ryan was kind
enough to move me over to Voice Pod 16, but I'm hitting the same issue
here. 

OSL archive and Google search turned up nothing concrete, other than a
general theme of it sounds like your telco / carrier has issues.  :)

 

Any ideas?

 

Cheers and TIA, sd

 


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please visit www.ipexpert.com http://www.ipexpert.com/ 

 

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Re: [OSL | CCIE_Voice] about IP blue VTGO multilab

[Steve Denney (stdenney)]

Agreed. I tried using VTGO Multilab back in March (version 2.11.1.307,
which is the same version still posted). 

 

I found it to be too unstable to use, and had to revert back to
2.11.1.230 (which turned out to be a complete uninstall / reinstall
process, including some manual registry key deletion - not trivial).

Documentation on the new version is also not yet available, so you're on
your own in terms of figuring out how to save/access the multiple
instances.

 

IPblue makes excellent products, and I'm sure this one will be as well,
but I would recommend sticking with 2.11.1.230 until Multilab is a bit
more stable and documented. 

As always YMMV. :)

 

cheers, sd

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roger
Henderson
Sent: Sunday, May 02, 2010 7:42 PM
To: Ravindra Lakpriya
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] about IP blue VTGO multilab

 

I found the multilab software really buggy (it crashed every time I
registered a phone), so I use the current version of VTGO-Advanced with
the /d switch (for multiple devices) - it works great (although only
simulates 7961's and has some shortcomings).

On Mon, May 3, 2010 at 4:48 AM, Ravindra Lakpriya lakpr...@gmail.com
wrote:

Hi Guys,

I saw in the new multilab soft-phone trial you can see that there are
option to simulate 7965.

Any of you have used that ?

Can we really depend on that with out going for hardware phones

Best Regards
--
Ravindra Lakpriya
+94 773 532 094
___
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please visit www.ipexpert.com

 

___
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Re: [OSL | CCIE_Voice] IPExpert Workbook vol2

[Steve Denney (stdenney)]

Vik, here's my public feedback :)

 

I recently received your newest v3 Voice VoD and Vol 1 video
walkthroughs. They are excellent. You have done a great job in balancing
theory (i.e., learning the concepts) and practical components (i.e.,
specific tips to help pass the exam). The only thing I don't understand
is, when do you sleep. :)

 

Thanks for your tireless efforts to improve the quality of the materials
(not to mention your time spent here on the OSL).

 

Best regards,

 

Steve Denney, CISSP

Systems Engineer - Technology Solutions Network

Voice and Unified Communications Products

Cisco Systems, Inc.

125 High Street, 21st Floor

Boston, MA  02110

978-936-4048 (Office)

617-872-5031 (Mobile)

stden...@cisco.com 

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vik Malhi
Sent: Sunday, May 02, 2010 4:56 PM
To: Wael Agina; Tom
Cc: OSL Group
Subject: Re: [OSL | CCIE_Voice] IPExpert Workbook vol2

 

Thanks Wael. Fingers crossed for your next attempt.

We are finally reaching a point where we have completed all the labs,
VOD and Vol 1 walkthru's etc, etc. Our next task, once the vol 2
walkthru's are complete, will be to revisit all the WB labs/solutions
and correct typos and add more detailed explanations where necessary. It
has been a difficult job to record/write new material and at the same
time support the existing labs- hopefully we can improve the job we are
doing on the product maintenance from now on.

I, as always, would appreciate any feedback (privately if bad, publicly
if good- LOL:-) you have on the VOD, WB and Walkthrus.

-- 
Vik Malhi - CCIE #13890
Instructor - IPexpert, Inc.
Mailto: vma...@ipexpert.com
Telephone: +1.810.326.1444
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, Security  Service Provider) certification(s) with
training locations throughout the United States, Europe, South Asia and
Australia. Be sure to visit our online communities at
www.ipexpert.com/communities http://www.ipexpert.com/communities  and
our public website at www.ipexpert.com http://www.ipexpert.com/  






From: Wael Agina waelag...@gmail.com
Date: Sun, 2 May 2010 19:47:49 +0300
To: Tom tom.c...@gmail.com
Cc: OSL Group ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] IPExpert Workbook vol2

Dear Tom,

  Actually I bought both INE and IPE.
I want to say that IPE matriel is much better with more and more indepth
tasks and solution and more explanations.
The point that the proctor guide in some labs - especially volume II -
not 100 % accurate , so you will have to try a little .

Also I recommend you IPE Walkthrough videos, which is a must [ I heared
some of volume one videos and it is really excellent].

For INE the best is thier VoD by Josh Fink which is really very good.

In brief, The IPE Workbooks 1,2 are the best along with walkthrough
video's.

Note: I already tried lab last month with no luck and found IPE material
is very good along with the Outstanding bootcamp with Vik.



Thanks and Best Regards,
Wael Agina



___
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please visit www.ipexpert.com

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

Re: [OSL | CCIE_Voice] IPExpert Workbook vol2

[Steve Denney (stdenney)]

Hi Wayne,

 

As I just posted in response on the OSL, Vik’s new materials are outstanding. 
He has done an excellent job in balancing theory and practical topics and is to 
be commended for this excellent work.

 

I will say that I’m a little surprised by the harsh comments regarding Mark’s 
work. While I agree that the previous Voice v3 VoD was not up to the standards 
that Vik reaches in the newer version, “terrible quality” might be overstating 
it a bit. I learned a lot from his version, as well as his previous v2 version. 
It seems a bit odd and unnecessary to call out a former employee in a public 
forum in this manner. 

 

Just my opinion.

 

Regards, Steve

 

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Wayne Lawson
Sent: Sunday, May 02, 2010 3:31 PM
To: A A
Cc: ccie_voice@onlinestudylist.com; tom.c...@gmail.com
Subject: Re: [OSL | CCIE_Voice] IPExpert Workbook vol2

 

AA  Gang -

 

 Thanks for the positive feedback. I'm interested in hearing more about 
what people think about our new Voice VOD course done by Vik. The old one 
(terrible quality) that was done by Mark Snow (before he was replaced) has been 
replaced by one redone by Vik (began shipping about a month ago). (People can 
get their update - if they haven't already received it - by emailing 
sa...@ipexpert.com.) 

Regards,

 

Wayne A. Lawson II - CCIE #5244 (RS)

Founder, President  CEO - IPexpert, Inc., Proctor Labs, Inc.  Platinum 
Solutions Group, LLC. 

Mailto: wlaw...@ipexpert.com

Telephone: +1.810.334.1564

eFax: +1.810.454.0130

 

::Message sent from iPhone::

 

IPexpert  Proctor Labs are premier providers of Self-Study Workbooks, Video on 
Demand, Audio Tools, Online Hardware Rental and Classroom Training for the 
Cisco CCIE (RS, Voice, Security  Service Provider) certification(s) with 
training locations throughout the United States, Europe, South Asia and 
Australia. Be sure to visit our online communities at 
www.ipexpert.com/communities and our public website at www.ipexpert.com or 
www.proctorlabs.com. 

 

Platinum Solutions Group (PSG) provides high-end consulting services with a 
primary emphasis on Cisco's Data Center Solutions, Service Provider Solutions, 
Unified Communications and Security-enabled infrastructures. Be sure to visit 
www.platinumsolutionsgroup.com. 


On May 2, 2010, at 1:27 PM, A A f...@hotmail.com wrote:

Totally agreed with Wael here, Let's not be mean  INE products are 
good to learn the technologies, whereas IPX products are good for both learning 
the technologies and tackle the exam:) ... . Josh Finke and Vik videos are 
awesome. I have both and I'm satisfied.. planning to take the exam end of 2010. 
 
Best regards,
Ahmad Azeem

 



Date: Sun, 2 May 2010 19:47:49 +0300
From: waelag...@gmail.com
To: tom.c...@gmail.com
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] IPExpert Workbook vol2

Dear Tom,

  Actually I bought both INE and IPE.
I want to say that IPE matriel is much better with more and more 
indepth tasks and solution and more explanations.
The point that the proctor guide in some labs - especially volume II - 
not 100 % accurate , so you will have to try a little .

Also I recommend you IPE Walkthrough videos, which is a must [ I heared 
some of volume one videos and it is really excellent].

For INE the best is thier VoD by Josh Fink which is really very good.

In brief, The IPE Workbooks 1,2 are the best along with walkthrough 
video's.

Note: I already tried lab last month with no luck and found IPE 
material is very good along with the Outstanding bootcamp with Vik.



Thanks and Best Regards,
Wael Agina

 



Get a free e-mail account with Hotmail. Sign-up now. 

___
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please visit www.ipexpert.com

___
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www.ipexpert.com

Re: [OSL | CCIE_Voice] 7965 Phone tftp-server statements

[Steve Denney (stdenney)]

FYI, here's a very helpful link to find exactly which load files are
included on all CME versions:

 

Cisco Unified CME 7.0(1) Supported Firmware, Platforms, Memory, and
Voice Products

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/requirements/guide/c
me701spc.htm

(this is for CME v7.0(1) - just navigate up one level to the
Compatibility Information link to find any other version)

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of vccie2010
Sent: Thursday, April 29, 2010 3:11 AM
To: Roger Henderson
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] 7965 Phone tftp-server statements

 

That helps a lot Roger. Appreciate it !!!

On Thu, Apr 29, 2010 at 12:05 AM, Roger Henderson rhender...@gmail.com
wrote:

These are the files on my Proctorlabs router:

 

103 2496963 Oct 19 2007 07:11:16 PHONE/7945-7965/apps45.8-3-2-27.sbn
104 585536 Oct 19 2007 07:11:14 PHONE/7945-7965/cnu45.8-3-2-27.sbn
105 2453202 Dec 18 2008 12:46:18 PHONE/7945-7965/cvm45sccp.8-3-2-27.sbn
106 326315 Oct 19 2007 07:11:18 PHONE/7945-7965/dsp45.8-3-2-27.sbn
107 555406 Dec 18 2008 12:46:30 PHONE/7945-7965/jar45sccp.8-3-2-27.sbn
108 638 Dec 18 2008 12:46:30 PHONE/7945-7965/SCCP45.8-3-3S.loads
109 642 Oct 19 2007 07:11:10 PHONE/7945-7965/term45.default.loads
110 642 Oct 19 2007 07:11:12 PHONE/7945-7965/term65.default.loads
111 650 Sep 18 2009 08:20:06 PHONE/7945-7965/SIP45.8-5-3S.loads
112 2817568 Oct 19 2007 07:11:20 PHONE/7945-7965/cvm45sip.8-3-2-27.sbn
113 557452 Oct 19 2007 07:11:18 PHONE/7945-7965/jar45sip.8-3-2-27.sbn
114 638 Oct 19 2007 07:11:08 PHONE/7945-7965/SIP45.8-3-3S.loads

 

I hope that helps.

 

Roger

 

On Thu, Apr 29, 2010 at 4:59 PM, vccie2010 vccie2...@gmail.com wrote:

Unfortunately I have 7962 phones in my lab and will appreciate
if someone can help me with with all tftp-server statements required on
CME for a 7965 SIP and 7965 SCCP phones...

 

Appreciate your help on this...

 

thx / V

 

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training, please visit www.ipexpert.com http://www.ipexpert.com/ 

 

 

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[OSL | CCIE_Voice] BR2 Registration Issues

[Steve Denney (stdenney)]

Been racking my brains and burning up lots of rack time (and support's
time as well) over this one. Would greatly appreciate any insights.

 

Suddenly, I am having major issues getting my softphones registered to
the BR2 CME router. This started last week on Voice Pod 14. Since it was
the first time I'd used that particular pod, I chalked it up to a
possible flaky pod. But this week, the same thing is happening on Pod 13
- which I've used before several times. It's as if something has
changed, but I have no idea what. I've triple checked all of the soft
client settings and MAC addresses and router configs. Like I said, this
has always worked in the past - it's not as if I have changed any of my
lab procedures which have worked up to now. I am using Software VPN.
Everything about HQ and BR1 seems fine - the only issues are with BR2.

 

Specific symptoms are:

My IPblue client, which should register as ephone 2, is not registering
to BR2, but to SRST somewhere (!!). The client is pointing to
10.10.110.3, as it should, with the SRST box unchecked - no changes.

 

My X-Lite client, which should register as voice register pool 2, is
getting Registration error 408 - Request Timeout.

 

Finally, a third issue, not related to softphones: the BR2 Ph2 (7960 at
Proctor Labs) will not convert from SCCP to SIP. I've hit this many
times in the past and was always able to use the time-tested procedure
of pointing the DHCP option 150 to the CUCM Sub, and letting it get its
SIP files from there. But since last week, that procedure no longer
works either.

 

I have purposely left out a lot of the troubleshooting details because I
did not want to overburden the list :)  Has anyone else experienced
anything like this recently?

 

Any tips appreciated...essentially dead in the water in regards to BR2 /
CME / CUE until I can get this figured out. *sigh*

 

cheers, sd

 

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Re: [OSL | CCIE_Voice] BR2 Registration Issues - SOLVED

[Steve Denney (stdenney)]

Lo and behold, I just hit the same issue, and got the same fix, courtesy
of Ryan :)

Note that changes need to be made on both HQ and BR2 routers, as
follows:

 

HQ-RTR(config)#interface Serial0/0/1:0 

HQ-RTR(config-if)#no  frame-relay traffic-shaping 

HQ-RTR(config)#interface Serial0/0/1:0.2 point-to-point 

HQ-RTR(config-subif)#no  frame-relay interface-dlci 202  

HQ-RTR(config-subif)#frame-relay interface-dlci 202 

HQ-RTR(config-fr-dlci)# ip address 10.10.112.1 255.255.255.0 

 

BR2-RTR(config)#interface Serial0/1/0:0 

BR2-RTR(config-if)#no frame-relay traffic 

BR2-RTR(config)#interface Serial0/1/0:0.1 point-to-point 

BR2-RTR(config-subif)#no frame-relay interface-dlci 102 

BR2-RTR(config-subif)# frame-relay interface-dlci 102   

BR2-RTR(config-fr-dlci)# ip address 10.10.112.2 255.255.255.0

 

In troubleshooting, I was reverting the BR2 router back to earlier labs,
suspecting a problem with its 11A config - but that was not helping, as
the HQ router was going untouched.

Working like a champ now. :)

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Matthew
Berry
Sent: Friday, April 23, 2010 12:39 PM
To: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] BR2 Registration Issues - SOLVED

 

Steve -

I just recently had the same issue.  It started popping up out of
nowhere.  I opened a support requested with Proctor Labs and got it
figure out.  This worked like a charm for me.  The following is a quote
from support:

Matthew,

I had some time to do some testing on Volume 1 Lab 11a and I found that
the initial and final config did not configure an IP address between
HQ-RTR and BR2-RTR. Drew and myself worked for a couple days on the
issue and could not resolve it, banging our heads against the wall, as
imagine you were too, 3 hours I think you said. In the end the issue was
above our pay grade :-) We had to pull Amy and Vik into the mix to help
us out. I apologize that I did not know this Sunday night. 

The issue is a known IOS bug. Once the configs load and if you run into
the issue again here are the configs to resolve the issue on those two
devices:

HQ-RTR(config)#interface Serial0/0/1:0
HQ-RTR(config-if)#no  frame-relay traffic-shaping
HQ-RTR(config)#interface Serial0/0/1:0.2 point-to-point
HQ-RTR(config-subif)#no  frame-relay interface-dlci 202 
HQ-RTR(config-subif)#frame-relay interface-dlci 202
HQ-RTR(config-fr-dlci)# ip address 10.10.112.1 255.255.255.0

Matthew,

I had some time to do some testing on Volume 1 Lab 11a and I found that
the initial and final config did not configure an IP address between
HQ-RTR and BR2-RTR. Drew and myself worked for a couple days on the
issue and could not resolve it, banging our heads against the wall, as
imagine you were too, 3 hours I think you said. In the end the issue was
above our pay grade :-) We had to pull Amy and Vik into the mix to help
us out. I apologize that I did not know this Sunday night. 

The issue is a known IOS bug. Once the configs load and if you run into
the issue again here are the configs to resolve the issue on those two
devices:

HQ-RTR(config)#interface Serial0/0/1:0
HQ-RTR(config-if)#no  frame-relay traffic-shaping
HQ-RTR(config)#interface Serial0/0/1:0.2 point-to-point
HQ-RTR(config-subif)#no  frame-relay interface-dlci 202 
HQ-RTR(config-subif)#frame-relay interface-dlci 202
HQ-RTR(config-fr-dlci)# ip address 10.10.112.1 255.255.255.0

 

Matthew Berry

A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written

 

Vitals:

GVoice: +1.612.424.5044

Gmail: ciscovoiceg...@gmail.com

Skype: ciscovoiceguru

Twitter: ciscovoiceguru

 

Cert Stats:

Cisco Cert Journey Began: Jan 1, 2009

1st Lab Attempt: Aug 16, 2010


On 4/23/2010 11:06 AM, Steve Denney (stdenney) wrote: 

Been racking my brains and burning up lots of rack time (and support's
time as well) over this one. Would greatly appreciate any insights.

 

Suddenly, I am having major issues getting my softphones registered to
the BR2 CME router. This started last week on Voice Pod 14. Since it was
the first time I'd used that particular pod, I chalked it up to a
possible flaky pod. But this week, the same thing is happening on Pod 13
- which I've used before several times. It's as if something has
changed, but I have no idea what. I've triple checked all of the soft
client settings and MAC addresses and router configs. Like I said, this
has always worked in the past - it's not as if I have changed any of my
lab procedures which have worked up to now. I am using Software VPN.
Everything about HQ and BR1 seems fine - the only issues are with BR2.

 

Specific symptoms are:

My IPblue client, which should register as ephone 2, is not registering
to BR2, but to SRST somewhere (!!). The client is pointing to
10.10.110.3, as it should, with the SRST box unchecked - no changes.

 

My X-Lite client, which should register as voice register pool 2, is
getting Registration error 408 - Request Timeout

[OSL | CCIE_Voice] Vol 1 Lab 12 Task 1 - IPPA Agent phone goes Not Ready and won't accept the incoming call

[Steve Denney (stdenney)]

Working on UCCX (Vol 1 Lab 12), and wondering if anyone's seen this
behavior before...Having trouble getting the IPPA agent to pick up an
incoming ICD call.

 

I'm dialing the ICD CTI route point (5710) from the PSTN phone. UCCX
answers, and the queue prompt is played. 

The agent phone (HQ Phn2) is running the IPPA service and in Ready
state. But the incoming call never gets picked up, and the IPPA agent
gets automatically put into Not Ready state. 

 

Meanwhile, the line on the HQ Phn2 (5002) shows a status of Remote In
Use (I can see the call coming in from the CTI port, 5701). 

But I can't even pick up this call manually (the only softkeys available
are Barge and New Call).

And if I try to put the IPPA service back in the Ready state, I get this
error: You cannot change to the requested state from your current
state.

 

When the caller finally gets tired and hangs up, I can actually *then*
pick up the call from the CTI port on the agent phone (the call shows as
on hold, and the Resume softkey becomes available) - and I hear the
queue prompt from the CTI port.

 

Very odd. Any ideas on what/how to troubleshoot? 

My most obvious thought was that the CTI ports did not have a CSS that
could see the agent phones - but they do, and yes they are registered to
CUCM. 

I also reset all the CTI ports, the CTI route point, the phones, and the
UCCX engine. Wheee.. :)

 

cheers, sd

 

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Re: [OSL | CCIE_Voice] Vol 1 Lab 12 Task 1 - IPPA Agent phone goes NotReady and won't accept the incoming call - SOLVED

[Steve Denney (stdenney)]

Dug a little deeper - Interestingly, the behavior I described below
occurred ONLY if the incoming call was from the 2123942123 line on the
PSTN phone - the IPPA agent worked as expected if the call is coming
from the 911 or 6178632683 lines.

 

So then I exited the IPPA service completely, and just dialed 2123945002
directly from the 2123942123 line of PSTN phone - which triggered *two*
incoming calls on 5002 - at which point I remembered somewhat sheepishly
that 2123942123 was still set up as a remote destination for 5002.

 

Moral: Mobility can mess with you when you least expect it. :)

 

Have a good weekend all...

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Steve
Denney (stdenney)
Sent: Friday, April 23, 2010 5:28 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Vol 1 Lab 12 Task 1 - IPPA Agent phone goes
NotReady and won't accept the incoming call

 

Working on UCCX (Vol 1 Lab 12), and wondering if anyone's seen this
behavior before...Having trouble getting the IPPA agent to pick up an
incoming ICD call.

 

I'm dialing the ICD CTI route point (5710) from the PSTN phone. UCCX
answers, and the queue prompt is played. 

The agent phone (HQ Phn2) is running the IPPA service and in Ready
state. But the incoming call never gets picked up, and the IPPA agent
gets automatically put into Not Ready state. 

 

Meanwhile, the line on the HQ Phn2 (5002) shows a status of Remote In
Use (I can see the call coming in from the CTI port, 5701). 

But I can't even pick up this call manually (the only softkeys available
are Barge and New Call).

And if I try to put the IPPA service back in the Ready state, I get this
error: You cannot change to the requested state from your current
state.

 

When the caller finally gets tired and hangs up, I can actually *then*
pick up the call from the CTI port on the agent phone (the call shows as
on hold, and the Resume softkey becomes available) - and I hear the
queue prompt from the CTI port.

 

Very odd. Any ideas on what/how to troubleshoot? 

My most obvious thought was that the CTI ports did not have a CSS that
could see the agent phones - but they do, and yes they are registered to
CUCM. 

I also reset all the CTI ports, the CTI route point, the phones, and the
UCCX engine. Wheee.. :)

 

cheers, sd

 

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Re: [OSL | CCIE_Voice] Translation rule?

[Steve Denney (stdenney)]

That is a good trick. :) 

 

Couldn't you also use a backslash in front of the ? to escape the
wildcard? Not in front of a console right now so can't verify...

 

cheers, sd 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ashar
Siddiqui
Sent: Saturday, April 03, 2010 10:52 PM
To: Hough, Earl
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Translation rule?

 

Cheers mate! it worked.


On 04/04/2010 03:47, Hough, Earl wrote: 

It's an old RS lab trick.  

 

Hold down Ctrl+v first, then release and type the question mark.  It
should come in as a literal character and not a navigation key of the
CLI. 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ashar
Siddiqui
Sent: Saturday, April 03, 2010 10:45 PM
To: CCIE Voice OSL (ccie_voice@onlinestudylist.com)
Subject: [OSL | CCIE_Voice] Translation rule?

 

Hi,

How can you enter the following translation rule on a router?

voice translation-rule 100
rule 1 /5+44?\(598\)$/ /693\1/

I cannot add this command on my routeras soon as I enter ? , the
router thinks that I am asking about some command!

-- 

 
ProctorLaben
ProctorLab#conf t
Enter configuration commands, one per line.  End with CNTL/Z.
ProctorLab(config)#voice trans
ProctorLab(config)#voice translation-rul
ProctorLab(config)#voice translation-rule 100
ProctorLab(cfg-translation-rule)#rule 1 /5+44?
WORD/  
 
ProctorLab(cfg-translation-rule)#rule 1 /5+44 ?
% Unrecognized command
ProctorLab(cfg-translation-rule)#rule 1 /5+44?
WORD/  
 
ProctorLab(cfg-translation-rule)#rule 1 /5+44\?% trailing \
% Unrecognized command
ProctorLab(cfg-translation-rule)#rule 1 /5+44/?
/  
 
ProctorLab(cfg-translation-rule)#rule 1 /5+44


Any clue?



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Re: [OSL | CCIE_Voice] SIP SRST - What application to use?

[Steve Denney (stdenney)]

According to the SIP SRST Admin Guide

(http://www.cisco.com/en/US/docs/voice_ip_comm/cusrst/admin/sipsrst/conf
iguration/guide/spsrst2.html):

 

application application-name 

Selects the session-level application on the VoIP dial peer. Use the
application-name argument to define a specific interactive voice
response (IVR) application.

Example: Router(config-register-pool)# application SIP.App

 

I haven't played with this very much, so real-world anecdotes are
welcomed. :)

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Berry,
Matthew J.
Sent: Thursday, March 25, 2010 8:39 AM
To: Angel Perez; osl osl
Subject: Re: [OSL | CCIE_Voice] SIP SRST - What application to use?

 

So what you're saying is that SIP SRST seems to work properly even
without the sip.app application specified?

 

I haven't been able to tell a different without the application, which
is what raised the question about its function.

 

M

 

From: Angel Perez [mailto:gorr...@hotmail.com] 
Sent: Thursday, March 25, 2010 6:16 AM
To: Berry, Matthew J.; osl osl
Subject: RE: [OSL | CCIE_Voice] SIP SRST - What application to use?

 

Hello:
 
The second example is not shown...
 
My experience tell me that if you use application sip.app the gw won't
find the app, then you will need application global service alternate
Default (similar to mgcp srst) this way  the gw will use h323 and call
will work. A better aproach that worked for me is just delete this
command application sip.app
 
I know that this doesn't answer your question but could help
 
Regards

 



From: mjbe...@krollontrack.com
To: ccie_voice@onlinestudylist.com
Date: Thu, 25 Mar 2010 06:00:35 -0500
Subject: [OSL | CCIE_Voice] SIP SRST - What application to use?

All -

 

Here's a sample section from a SIP SRST setup from the SIP SRND Admin
Guide:

 

voice register pool 1

  id network 10.10.201.0 mask 255.255.255.0

  application sip.app

  preference 2

  incoming called-number

  cor incoming css-internal default

  codec g711ulaw

 

What the heck is this application command used for?  Later on, I came
across this config example:

 

 

 

 

 



Hotmail: Trusted email with Microsoft's powerful SPAM protection. Sign
up now. https://signup.live.com/signup.aspx?id=60969 

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Re: [OSL | CCIE_Voice] Type 7962 IP Blue softphone

[Steve Denney (stdenney)]

Do be careful with this version :)  

 

It is a new Multiab version, which in addition to adding 7962 support,
is supposed to make it easier to manage multiple instances (up to 5).

 

However, I found it to be unstable and had to revert back to 2.11.1.230
(which turned out to be a complete uninstall / manual registry key
deletion / reinstall process).

Documentation on the new version is also not yet available, so you're on
your own in terms of figuring out how to save/access the multiple
instances.

 

YMMV, but I would recommend sticking with 2.11.1.230 until this version
is a bit more stable and documented.

 

cheers, sd

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Wael Agina
Sent: Wednesday, March 03, 2010 10:25 AM
To: OSL Group
Subject: Re: [OSL | CCIE_Voice] Type 7962 IP Blue softphone

 

Dear's,

 

  IP Blue released new version supports 7962 and other newer.

Version: 2.11.1.307 - Date: February 23, 2010



-- 

Thanks and Best Regards,
Wael Agina

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Re: [OSL | CCIE_Voice] Type 7962 IP Blue softphone

[Steve Denney (stdenney)]

Typo; that's supposed to be Multilab :)

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Steve
Denney (stdenney)
Sent: Wednesday, March 03, 2010 10:37 AM
To: Wael Agina; OSL Group
Subject: Re: [OSL | CCIE_Voice] Type 7962 IP Blue softphone

 

Do be careful with this version :)  

 

It is a new Multiab version, which in addition to adding 7962 support,
is supposed to make it easier to manage multiple instances (up to 5).

 

However, I found it to be unstable and had to revert back to 2.11.1.230
(which turned out to be a complete uninstall / manual registry key
deletion / reinstall process).

Documentation on the new version is also not yet available, so you're on
your own in terms of figuring out how to save/access the multiple
instances.

 

YMMV, but I would recommend sticking with 2.11.1.230 until this version
is a bit more stable and documented.

 

cheers, sd

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Wael Agina
Sent: Wednesday, March 03, 2010 10:25 AM
To: OSL Group
Subject: Re: [OSL | CCIE_Voice] Type 7962 IP Blue softphone

 

Dear's,

 

  IP Blue released new version supports 7962 and other newer.

Version: 2.11.1.307 - Date: February 23, 2010



-- 

Thanks and Best Regards,
Wael Agina

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[OSL | CCIE_Voice] Vol 1 Lab 5C - CME SIP Phone not registering

[Steve Denney (stdenney)]

Hi Gang,

 

Just need a sanity check here. :) As usual I'm having trouble getting my
remote CME SIP Phone (BR2 Ph 2, DN 3005) registered to CME. 

 

After many previous efforts at getting the right SIP FW loads via the
TFTP server on the BR2 router, this time I took advice ;) and pointed
the DHCP option 150 to the CUCM Sub. That seemed to work as far as
getting the proper SIP load onto the 7960, since sh cdp ne now shows a
SIP (not SEP, as it was when I first booted)
device name. So I think that part is OK. The phone is still not
registering to CME though. (My remote X-Lite SIP client is registered
fine.)

 

I've done another shut on the interface, did a no crea prof / crea prof,
still no joy. I am quite sure I'm missing an obvious step. :) 

Any suggestions for troubleshooting? 

 

cheers, sd

 

___
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Re: [OSL | CCIE_Voice] Vol 1 Lab 5C - CME SIP Phone not registering - SOLVED

[Steve Denney (stdenney)]

Brain lock resolved. All phones registered and happy  :)

 

Steps taken (all from the BR2 CME router):

 

1) Change DHCP option 150 to the CUCM Sub

2) shut / no shut on fa0/3/1 (allow BR2 Ph2 to get its SIP load from
Sub)

3) Wait patiently until show cdp ne confirms SIP (not
SEP) on fa0/3/1

4) Edit tftp-server flash: commands to REMOVE pointers to any CME SIP
loads (leave only pointers to the SCCP loads) - notepad cut  paste is
your friend here

5) In voice register global, REMOVE the old SIP load statement (example
in my case: no load 7960-7940 P0S3-08-9-00)

6) In voice register global, no crea prof / crea prof

7) Write mem, reload router, wait for all phones to register happily :)

 

Thanks all for the tips, and hopefully this is helpful for others.

 

cheers, sd

 

From: CCIETalk.com [mailto:cciet...@gmail.com] 
Sent: Tuesday, March 02, 2010 9:49 AM
To: Steve Denney (stdenney)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 5C - CME SIP Phone not
registering

 

Can you post your configs along with show cdp neigh detail for that
port? Sounds something basic for sure. I ran into this the other day but
resolved it after getting my load from the SUB.

On Tue, Mar 2, 2010 at 9:17 AM, Steve Denney (stdenney)
stden...@cisco.com wrote:

Hi Gang,

 

Just need a sanity check here. :) As usual I'm having trouble getting my
remote CME SIP Phone (BR2 Ph 2, DN 3005) registered to CME. 

 

After many previous efforts at getting the right SIP FW loads via the
TFTP server on the BR2 router, this time I took advice ;) and pointed
the DHCP option 150 to the CUCM Sub. That seemed to work as far as
getting the proper SIP load onto the 7960, since sh cdp ne now shows a
SIP (not SEP, as it was when I first booted)
device name. So I think that part is OK. The phone is still not
registering to CME though. (My remote X-Lite SIP client is registered
fine.)

 

I've done another shut on the interface, did a no crea prof / crea prof,
still no joy. I am quite sure I'm missing an obvious step. :) 

Any suggestions for troubleshooting? 

 

cheers, sd

 


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please visit www.ipexpert.com




-- 
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Re: [OSL | CCIE_Voice] Vol 1 Lab 5C - CME SIP Phone not registering- SOLVED

[Steve Denney (stdenney)]

Sorry...brain lock apparently continues in regards to
documentation...Left out a very important step. ;)

 

1) Change DHCP option 150 to the CUCM Sub

2) shut / no shut on fa0/3/1 (allow BR2 Ph2 to get its SIP load from
Sub)

3) Wait patiently until show cdp ne confirms SIP (not
SEP) on fa0/3/1

4) Edit tftp-server flash: commands to REMOVE pointers to any CME SIP
loads (leave only pointers to the SCCP loads) - notepad cut  paste is
your friend here

5) In voice register global, REMOVE the old SIP load statement (example
in my case: no load 7960-7940 P0S3-08-9-00)

6) In voice register global, no crea prof / crea prof

7) Change DHCP option 150 back to the CME router

8) Write mem, reload router, wait for all phones to register happily :)

 

cheers, sd 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Steve
Denney (stdenney)
Sent: Tuesday, March 02, 2010 10:05 AM
To: CCIETalk.com
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 5C - CME SIP Phone not
registering- SOLVED

 

Brain lock resolved. All phones registered and happy  :)

 

Steps taken (all from the BR2 CME router):

 

1) Change DHCP option 150 to the CUCM Sub

2) shut / no shut on fa0/3/1 (allow BR2 Ph2 to get its SIP load from
Sub)

3) Wait patiently until show cdp ne confirms SIP (not
SEP) on fa0/3/1

4) Edit tftp-server flash: commands to REMOVE pointers to any CME SIP
loads (leave only pointers to the SCCP loads) - notepad cut  paste is
your friend here

5) In voice register global, REMOVE the old SIP load statement (example
in my case: no load 7960-7940 P0S3-08-9-00)

6) In voice register global, no crea prof / crea prof

7) Write mem, reload router, wait for all phones to register happily :)

 

Thanks all for the tips, and hopefully this is helpful for others.

 

cheers, sd

 

From: CCIETalk.com [mailto:cciet...@gmail.com] 
Sent: Tuesday, March 02, 2010 9:49 AM
To: Steve Denney (stdenney)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 5C - CME SIP Phone not
registering

 

Can you post your configs along with show cdp neigh detail for that
port? Sounds something basic for sure. I ran into this the other day but
resolved it after getting my load from the SUB.

On Tue, Mar 2, 2010 at 9:17 AM, Steve Denney (stdenney)
stden...@cisco.com wrote:

Hi Gang,

 

Just need a sanity check here. :) As usual I'm having trouble getting my
remote CME SIP Phone (BR2 Ph 2, DN 3005) registered to CME. 

 

After many previous efforts at getting the right SIP FW loads via the
TFTP server on the BR2 router, this time I took advice ;) and pointed
the DHCP option 150 to the CUCM Sub. That seemed to work as far as
getting the proper SIP load onto the 7960, since sh cdp ne now shows a
SIP (not SEP, as it was when I first booted)
device name. So I think that part is OK. The phone is still not
registering to CME though. (My remote X-Lite SIP client is registered
fine.)

 

I've done another shut on the interface, did a no crea prof / crea prof,
still no joy. I am quite sure I'm missing an obvious step. :) 

Any suggestions for troubleshooting? 

 

cheers, sd

 


___
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please visit www.ipexpert.com




-- 
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[OSL | CCIE_Voice] Lab 5c Task 5.3 - 911 calls from HQ disconnecting

[Steve Denney (stdenney)]

Having trouble with calls from HQ Ph2 (SIP CIPC DN 5002) to 911 (PSTN
phone).

This is similar to what happened last time I did this lab (on a
different pod). 

 

The call hits the GW fine, but I get a reorder tone on the CIPC. 

The call actually gets as far as the PSTN router, then disconnects as
per the debug ISDN q931 output (below). 

The PSTN phone alerts as if it's trying to accept the call, but the call
can't be picked up.

 

Calls from BR1 to 911 work fine. 

Debug shows Dial peer matching on HQ router is correct. 

Have restarted *everything* (all routers including PSTN, phones,
pub/sub, HQ 3750). 

 

HQ router config attached.

 

Any ideas? About to chalk it up to a bug and move on...

 

cheers, sd

 

 

HQ-RTR Debug:

 

Mar  2 17:41:20.667: ISDN Se0/0/0:23 Q931: pak_private_number: Invalid
type/plan 0x0 0x0 may be overriden; sw-type 13

Mar  2 17:41:20.667: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type
0xD is 0x2 0x1, Calling num 2123945002

Mar  2 17:41:20.667: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type
0xD is 0x0 0x0, Called num 911

Mar  2 17:41:20.671: ISDN Se0/0/0:23 Q931: TX - SETUP pd = 8  callref =
0x0087 

Bearer Capability i = 0x8090A2 

Standard = CCITT 

Transfer Capability = Speech  

Transfer Mode = Circuit 

Transfer Rate = 64 kbit/s 

Channel ID i = 0xA98383 

Exclusive, Channel 3 

Calling Party Number i = 0x2181, '2123945002' 

Plan:ISDN, Type:National 

Called Party Number i = 0x80, '911' 

Plan:Unknown, Type:Unknown

Mar  2 17:41:20.703: ISDN Se0/0/0:23 Q931: RX - CALL_PROC pd = 8
callref = 0x8087 

Channel ID i = 0xA98383 

Exclusive, Channel 3

Mar  2 17:41:20.715: ISDN Se0/0/0:23 Q931: RX - ALERTING pd = 8
callref = 0x8087 

Progress Ind i = 0x8188 - In-band info or appropriate now
available 

Mar  2 17:41:20.755: ISDN Se0/0/0:23 Q931: TX - DISCONNECT pd = 8
callref = 0x0087 

Cause i = 0x80AC - Requested circuit/channel not available

Mar  2 17:41:20.767: ISDN Se0/0/0:23 Q931: RX - RELEASE pd = 8  callref
= 0x8087

Mar  2 17:41:20.771: ISDN Se0/0/0:23 Q931: TX - RELEASE_COMP pd = 8
callref = 0x0087

 

 

PSTN-WAN Router Debug:

 

Mar  2 17:41:20.675: ISDN Se0/3/0:23 Q931: RX - SETUP pd = 8  callref =
0x0087 

Bearer Capability i = 0x8090A2 

Standard = CCITT 

Transfer Capability = Speech  

Transfer Mode = Circuit 

Transfer Rate = 64 kbit/s 

Channel ID i = 0xA98383 

Exclusive, Channel 3 

Calling Party Number i = 0x2181, '2123945002' 

Plan:ISDN, Type:National 

Called Party Number i = 0x80, '911' 

Plan:Unknown, Type:Unknown

Mar  2 17:41:20.699: ISDN Se0/3/0:23 Q931: TX - CALL_PROC pd = 8
callref = 0x8087 

Channel ID i = 0xA98383 

Exclusive, Channel 3

 

Mar  2 17:41:20.711: ISDN Se0/3/0:23 Q931: TX - ALERTING pd = 8
callref = 0x8087 

Progress Ind i = 0x8188 - In-band info or appropriate now
available 

Mar  2 17:41:20.759: ISDN Se0/3/0:23 Q931: RX - DISCONNECT pd = 8
callref = 0x0087 

Cause i = 0x80AC - Requested circuit/channel not available

Mar  2 17:41:20.763: ISDN Se0/3/0:23 Q931: TX - RELEASE pd = 8  callref
= 0x8087

Mar  2 17:41:20.775: ISDN Se0/3/0:23 Q931: RX - RELEASE_COMP pd = 8
callref = 0x0087

 

 

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Re: [OSL | CCIE_Voice] Lab 5c Task 5.3 - 911 calls from HQ disconnecting

[Steve Denney (stdenney)]

With attachment this time :)

 

 

From: Steve Denney (stdenney) 
Sent: Tuesday, March 02, 2010 12:52 PM
To: ccie_voice@onlinestudylist.com
Subject: Lab 5c Task 5.3 - 911 calls from HQ disconnecting

 

Having trouble with calls from HQ Ph2 (SIP CIPC DN 5002) to 911 (PSTN
phone).

This is similar to what happened last time I did this lab (on a
different pod). 

 

The call hits the GW fine, but I get a reorder tone on the CIPC. 

The call actually gets as far as the PSTN router, then disconnects as
per the debug ISDN q931 output (below). 

The PSTN phone alerts as if it's trying to accept the call, but the call
can't be picked up.

 

Calls from BR1 to 911 work fine. 

Debug shows Dial peer matching on HQ router is correct. 

Have restarted *everything* (all routers including PSTN, phones,
pub/sub, HQ 3750). 

 

HQ router config attached.

 

Any ideas? About to chalk it up to a bug and move on...

 

cheers, sd

 

 

HQ-RTR Debug:

 

Mar  2 17:41:20.667: ISDN Se0/0/0:23 Q931: pak_private_number: Invalid
type/plan 0x0 0x0 may be overriden; sw-type 13

Mar  2 17:41:20.667: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type
0xD is 0x2 0x1, Calling num 2123945002

Mar  2 17:41:20.667: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type
0xD is 0x0 0x0, Called num 911

Mar  2 17:41:20.671: ISDN Se0/0/0:23 Q931: TX - SETUP pd = 8  callref =
0x0087 

Bearer Capability i = 0x8090A2 

Standard = CCITT 

Transfer Capability = Speech  

Transfer Mode = Circuit 

Transfer Rate = 64 kbit/s 

Channel ID i = 0xA98383 

Exclusive, Channel 3 

Calling Party Number i = 0x2181, '2123945002' 

Plan:ISDN, Type:National 

Called Party Number i = 0x80, '911' 

Plan:Unknown, Type:Unknown

Mar  2 17:41:20.703: ISDN Se0/0/0:23 Q931: RX - CALL_PROC pd = 8
callref = 0x8087 

Channel ID i = 0xA98383 

Exclusive, Channel 3

Mar  2 17:41:20.715: ISDN Se0/0/0:23 Q931: RX - ALERTING pd = 8
callref = 0x8087 

Progress Ind i = 0x8188 - In-band info or appropriate now
available 

Mar  2 17:41:20.755: ISDN Se0/0/0:23 Q931: TX - DISCONNECT pd = 8
callref = 0x0087 

Cause i = 0x80AC - Requested circuit/channel not available

Mar  2 17:41:20.767: ISDN Se0/0/0:23 Q931: RX - RELEASE pd = 8  callref
= 0x8087

Mar  2 17:41:20.771: ISDN Se0/0/0:23 Q931: TX - RELEASE_COMP pd = 8
callref = 0x0087

 

 

PSTN-WAN Router Debug:

 

Mar  2 17:41:20.675: ISDN Se0/3/0:23 Q931: RX - SETUP pd = 8  callref =
0x0087 

Bearer Capability i = 0x8090A2 

Standard = CCITT 

Transfer Capability = Speech  

Transfer Mode = Circuit 

Transfer Rate = 64 kbit/s 

Channel ID i = 0xA98383 

Exclusive, Channel 3 

Calling Party Number i = 0x2181, '2123945002' 

Plan:ISDN, Type:National 

Called Party Number i = 0x80, '911' 

Plan:Unknown, Type:Unknown

Mar  2 17:41:20.699: ISDN Se0/3/0:23 Q931: TX - CALL_PROC pd = 8
callref = 0x8087 

Channel ID i = 0xA98383 

Exclusive, Channel 3

 

Mar  2 17:41:20.711: ISDN Se0/3/0:23 Q931: TX - ALERTING pd = 8
callref = 0x8087 

Progress Ind i = 0x8188 - In-band info or appropriate now
available 

Mar  2 17:41:20.759: ISDN Se0/3/0:23 Q931: RX - DISCONNECT pd = 8
callref = 0x0087 

Cause i = 0x80AC - Requested circuit/channel not available

Mar  2 17:41:20.763: ISDN Se0/3/0:23 Q931: TX - RELEASE pd = 8  callref
= 0x8087

Mar  2 17:41:20.775: ISDN Se0/3/0:23 Q931: RX - RELEASE_COMP pd = 8
callref = 0x0087

 

 

HQ-RTR#sh run
Building configuration...


Current configuration : 3620 bytes
!
! No configuration change since last restart
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname HQ-RTR
!
boot-start-marker
warm-reboot
boot-end-marker
!
logging buffered 51200 warnings
!
no aaa new-model
memory-size iomem 20
network-clock-participate wic 0 
network-clock-select 1 T1 0/0/0
dot11 syslog
no ip source-route
!
!
ip cef
!
!
no ip domain lookup
ip multicast-routing 
!
multilink bundle-name authenticated
!
isdn switch-type primary-ni
!
voice-card 0
 no dspfarm
 dsp services dspfarm
!
!
!
voice service voip 
 allow-connections h323 to sip
!
!
! 
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8
!
!
!
voice class h323 1
 h225 timeout tcp establish 3
!
!
!
!
!
!
!
!
!
!
!
!
!
!
! 
archive
 log config
  hidekeys
! 
!
!
!
controller T1 0/0/0
 framing esf
 linecode b8zs
 pri-group timeslots 1-3,24
!
controller T1 0/0/1
 framing esf
 linecode b8zs
 channel-group 0 timeslots 1-24
!
!
!
!
!
interface Loopback0
 ip address 10.10.110.1 255.255.255.255
!
interface FastEthernet0/0
 no ip address
 duplex full
 speed 100
!
interface FastEthernet0/0.10
 encapsulation

Re: [OSL | CCIE_Voice] Lab 5c Task 5.3 - 911 calls from HQ disconnecting - SOLVED

[Steve Denney (stdenney)]

As usual Otto, you have discovered the blatantly obvious item I
overlooked. :)

 

The problem was no incoming dial-peer in HQ-RTR to receive H.323 calls
from CUCM:

dial-peer voice 5000 voip

incoming called-number .

 

I revisited this task after completing the following task (5.4, in which
the incoming dial-peer was added) - and 5.3 now works.

Lesson learned for today: Requested circuit/channel not available
sometimes actually means Check your dial-peers again, stupid :)

 

Thanks Otto!

 

cheers, sd

 

 

From: Otto Sanchez [mailto:o...@ipexpert.com] 
Sent: Tuesday, March 02, 2010 1:43 PM
To: Steve Denney (stdenney)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Lab 5c Task 5.3 - 911 calls from HQ
disconnecting

 

Hi Steve,

What's the output for the following commands:

-. sh controllers t1
-. sh isdn status
-. sh voice port summary

Also, please configure an incoming dial-peer in hq-rtr to receive h.323
calls from ucm, 
dial-peer voice x voip
incoming called-number .

Thanks,



On Tue, Mar 2, 2010 at 1:22 PM, Steve Denney (stdenney)
stden...@cisco.com wrote:

With attachment this time :)

 

 

From: Steve Denney (stdenney) 
Sent: Tuesday, March 02, 2010 12:52 PM
To: ccie_voice@onlinestudylist.com
Subject: Lab 5c Task 5.3 - 911 calls from HQ disconnecting

 

Having trouble with calls from HQ Ph2 (SIP CIPC DN 5002) to 911 (PSTN
phone).

This is similar to what happened last time I did this lab (on a
different pod). 

 

The call hits the GW fine, but I get a reorder tone on the CIPC. 

The call actually gets as far as the PSTN router, then disconnects as
per the debug ISDN q931 output (below). 

The PSTN phone alerts as if it's trying to accept the call, but the call
can't be picked up.

 

Calls from BR1 to 911 work fine. 

Debug shows Dial peer matching on HQ router is correct. 

Have restarted *everything* (all routers including PSTN, phones,
pub/sub, HQ 3750). 

 

HQ router config attached.

 

Any ideas? About to chalk it up to a bug and move on...

 

cheers, sd

 

 

HQ-RTR Debug:

 

Mar  2 17:41:20.667: ISDN Se0/0/0:23 Q931: pak_private_number: Invalid
type/plan 0x0 0x0 may be overriden; sw-type 13

Mar  2 17:41:20.667: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type
0xD is 0x2 0x1, Calling num 2123945002

Mar  2 17:41:20.667: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type
0xD is 0x0 0x0, Called num 911

Mar  2 17:41:20.671: ISDN Se0/0/0:23 Q931: TX - SETUP pd = 8  callref =
0x0087 

Bearer Capability i = 0x8090A2 

Standard = CCITT 

Transfer Capability = Speech  

Transfer Mode = Circuit 

Transfer Rate = 64 kbit/s 

Channel ID i = 0xA98383 

Exclusive, Channel 3 

Calling Party Number i = 0x2181, '2123945002' 

Plan:ISDN, Type:National 

Called Party Number i = 0x80, '911' 

Plan:Unknown, Type:Unknown

Mar  2 17:41:20.703: ISDN Se0/0/0:23 Q931: RX - CALL_PROC pd = 8
callref = 0x8087 

Channel ID i = 0xA98383 

Exclusive, Channel 3

Mar  2 17:41:20.715: ISDN Se0/0/0:23 Q931: RX - ALERTING pd = 8
callref = 0x8087 

Progress Ind i = 0x8188 - In-band info or appropriate now
available 

Mar  2 17:41:20.755: ISDN Se0/0/0:23 Q931: TX - DISCONNECT pd = 8
callref = 0x0087 

Cause i = 0x80AC - Requested circuit/channel not available

Mar  2 17:41:20.767: ISDN Se0/0/0:23 Q931: RX - RELEASE pd = 8  callref
= 0x8087

Mar  2 17:41:20.771: ISDN Se0/0/0:23 Q931: TX - RELEASE_COMP pd = 8
callref = 0x0087

 

 

PSTN-WAN Router Debug:

 

Mar  2 17:41:20.675: ISDN Se0/3/0:23 Q931: RX - SETUP pd = 8  callref =
0x0087 

Bearer Capability i = 0x8090A2 

Standard = CCITT 

Transfer Capability = Speech  

Transfer Mode = Circuit 

Transfer Rate = 64 kbit/s 

Channel ID i = 0xA98383 

Exclusive, Channel 3 

Calling Party Number i = 0x2181, '2123945002' 

Plan:ISDN, Type:National 

Called Party Number i = 0x80, '911' 

Plan:Unknown, Type:Unknown

Mar  2 17:41:20.699: ISDN Se0/3/0:23 Q931: TX - CALL_PROC pd = 8
callref = 0x8087 

Channel ID i = 0xA98383 

Exclusive, Channel 3

 

Mar  2 17:41:20.711: ISDN Se0/3/0:23 Q931: TX - ALERTING pd = 8
callref = 0x8087 

Progress Ind i = 0x8188 - In-band info or appropriate now
available 

Mar  2 17:41:20.759: ISDN Se0/3/0:23 Q931: RX - DISCONNECT pd = 8
callref = 0x0087 

Cause i = 0x80AC - Requested circuit/channel not available

Mar  2 17:41:20.763: ISDN Se0/3/0:23 Q931: TX - RELEASE pd = 8  callref
= 0x8087

Mar  2 17:41:20.775: ISDN Se0/3/0:23 Q931: RX - RELEASE_COMP pd = 8
callref = 0x0087

 

 


___
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please visit www.ipexpert.com




-- 
Regards,

Otto

Re: [OSL | CCIE_Voice] Blocking a router pattern and PrecedenceMessage in Vol 1 Lab 5c Task 5.6

[Steve Denney (stdenney)]

Thanks Otto, I noticed that behavior as well.

 

Hijacking the thread - :) I also noticed something else interesting
while working on this same task - I was able to bypass the 91900 block
if I dialed from the call history directories. This was true on both
CICP (SIP) and IP Blue (SCCP) clients. 

Tried the 91900 RP both with and without Urgent Priority checked; no
change. 

Maybe I needed to add the ? at the end of the pattern to match the
whole string? Ran out of lab time before trying that...

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Otto
Sanchez
Sent: Tuesday, February 23, 2010 10:20 AM
To: CCIETalk.com
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Blocking a router pattern and
PrecedenceMessage in Vol 1 Lab 5c Task 5.6

 

Hi,

That's the expected behavior since the annunciator does not support sip
phones, 




On Tue, Feb 23, 2010 at 10:44 AM, CCIETalk.com cciet...@gmail.com
wrote:

So i was working Vol 1 Lab 5c task 5.6 where it asks to block 9-1-900
calls. Appeared straight forward and I was able to get this done.
However my SIP phones dont play this message and just disconnect the
call after I dial 9-1-900 while my SCCP phones play the message. Any
reason?

-- 
www.ccietalk.com

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




-- 
Regards,

Otto Sanchez 
CCIE #25592 (Voice) 
Support Engineer - IPexpert, Inc.
URL: http://www.IPexpert.com

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

Re: [OSL | CCIE_Voice] Blocking a router pattern and PrecedenceMessage in Vol 1 Lab 5c Task 5.6

[Steve Denney (stdenney)]

Thanks Otto! That's what I suspected. Will try it next lab.

 

cheers, sd

 

 

From: Otto Sanchez [mailto:o...@ipexpert.com] 
Sent: Tuesday, February 23, 2010 2:04 PM
To: CCIETalk.com
Cc: Steve Denney (stdenney); ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Blocking a router pattern and
PrecedenceMessage in Vol 1 Lab 5c Task 5.6

 

Hello,

Since sccp phones will send digits in one go (en block dialing) when
calling from directories, the pattern that you might have been created
will not be matched (I assume 91900 with urgent priority) under that
circumstances, so in that case, please create another 91900! pattern
with similar characteristics to match calls made from the directories
and play the same precedence level exceeded message,

HTH, 

On Tue, Feb 23, 2010 at 11:05 AM, CCIETalk.com cciet...@gmail.com
wrote:

I Am thinking that since dialing from call history isn't really a digit
press may be that's why? Let Otto shed some light :D

 

Steve - it appears we are on the same workbook lab... When are you
attempting the lab?

 

On Tue, Feb 23, 2010 at 10:30 AM, Steve Denney (stdenney)
stden...@cisco.com wrote:

Thanks Otto, I noticed that behavior as well.

 

Hijacking the thread - :) I also noticed something else interesting
while working on this same task - I was able to bypass the 91900 block
if I dialed from the call history directories. This was true on both
CICP (SIP) and IP Blue (SCCP) clients. 

Tried the 91900 RP both with and without Urgent Priority checked; no
change. 

Maybe I needed to add the ? at the end of the pattern to match the
whole string? Ran out of lab time before trying that...

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Otto
Sanchez
Sent: Tuesday, February 23, 2010 10:20 AM
To: CCIETalk.com
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Blocking a router pattern and
PrecedenceMessage in Vol 1 Lab 5c Task 5.6

 

Hi,

That's the expected behavior since the annunciator does not support sip
phones, 



On Tue, Feb 23, 2010 at 10:44 AM, CCIETalk.com cciet...@gmail.com
wrote:

So i was working Vol 1 Lab 5c task 5.6 where it asks to block 9-1-900
calls. Appeared straight forward and I was able to get this done.
However my SIP phones dont play this message and just disconnect the
call after I dial 9-1-900 while my SCCP phones play the message. Any
reason?

-- 
www.ccietalk.com

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




-- 
Regards,

Otto Sanchez 
CCIE #25592 (Voice) 
Support Engineer - IPexpert, Inc.
URL: http://www.IPexpert.com





-- 
www.ccietalk.com




-- 
Regards,

Otto Sanchez 
CCIE #25592 (Voice) 
Support Engineer - IPexpert, Inc.
URL: http://www.IPexpert.com

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

Re: [OSL | CCIE_Voice] Volume 1 Lab 5 Proctor Guide needs to bereworked

[Steve Denney (stdenney)]

Wayne, I am pleading with you now: 

Can we *please* get some kind of version control in the members download areas, 
so we know exactly which files are being posted?

 

The old site at least had “last updated” tags on the uploaded files, which made 
life much easier.

 

Thanks.

 

 

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Wayne Lawson
Sent: Monday, February 22, 2010 10:19 AM
To: CCIETalk.com
Cc: ccie_voice@onlinestudylist.com; Vik Malhi; Ryan Barnum
Subject: Re: [OSL | CCIE_Voice] Volume 1 Lab 5 Proctor Guide needs to bereworked

 

We'll make sure the latest files are uploaded. All of this was updated and 
fixed. 

 

Regards,

 

Wayne A. Lawson II - CCIE #5244

Founder  President - IPexpert

Mailto: wlaw...@ipexpert.com

Telephone: +1.810.326.1444, ext. 101

Live Assistance, Please visit: www.ipexpert.com/chat

eFax: +1.810.454.0130

 

::Message sent from iPhone::

 

IPexpert is a premier provider of Classroom and Self-Study Cisco CCNA (RS, 
Voice  Security), CCNP, CCVP, CCSP and CCIE (RS, Voice, Security  Service 
Provider) Certification Training with locations throughout the United States, 
Europe and Australia. Be sure to check out our online communities at 
www.ipexpert.com/communities and our public website at www.ipexpert.com. 


On Feb 22, 2010, at 10:10 AM, CCIETalk.com cciet...@gmail.com wrote:

Yeah it has TONS of errors and not for the faint hearted :D

On Mon, Feb 22, 2010 at 8:50 AM, t n tnn...@gmail.com wrote:

Hello,

I am currently on lab 5. From the beginning, it's already missing a few 
crucial items. For example, there is no discussion of the partitions/CSS for 
the phones. Also, the previous lab had the gateways used in the RPs. It's also 
missing the verifications and explanations seen in the previous 4 labs. 

It seems like the person who wrote the PG had a lot of energy for the 
first 4 labs. Then in lab 5 he/she got tired or bored and  decided it was 
alright to skip many steps.

Can someone at Ipexpert please look at the the PG for this lab? The 
quality is inconsistent with what I have seen so far. 
-- 
Thanks.

tnn314.wordpress.com

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




-- 
www.ccietalk.com

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

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

Re: [OSL | CCIE_Voice] Volume 1 Lab 5 Proctor Guide needs to bereworked

[Steve Denney (stdenney)]

Thanks Greg, but that is not what I'm referring to.

I am asking for the files that are posted to be tagged *on the Web site* so we 
know what version they are *before* downloading them.

A secondary issue here is that the v985 footer you refer to has been 
unchanged through the last several revisions of PGs I have seen - so, if that 
is intended as version control, it’s not reliable.

Thanks, Steve


-Original Message-
From: Pulos, Greg [mailto:gpu...@doc.gov] 
Sent: Monday, February 22, 2010 10:52 AM
To: Steve Denney (stdenney); Wayne Lawson; CCIETalk.com
Cc: ccie_voice@onlinestudylist.com; Vik Malhi; Ryan Barnum
Subject: RE: [OSL | CCIE_Voice] Volume 1 Lab 5 Proctor Guide needs to bereworked

It seems there is version control.

The version is identified on the bottom of every page of every WB and PG; in 
the footer.

The current version is: v985

Thank you.

greg


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Steve Denney 
(stdenney)
Sent: Monday, February 22, 2010 10:44 AM
To: Wayne Lawson; CCIETalk.com
Cc: ccie_voice@onlinestudylist.com; Vik Malhi; Ryan Barnum
Subject: Re: [OSL | CCIE_Voice] Volume 1 Lab 5 Proctor Guide needs to bereworked

Wayne, I am pleading with you now: 

Can we *please* get some kind of version control in the members download areas, 
so we know exactly which files are being posted?

 

The old site at least had “last updated” tags on the uploaded files, which made 
life much easier.

 

Thanks.

 

 

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Wayne Lawson
Sent: Monday, February 22, 2010 10:19 AM
To: CCIETalk.com
Cc: ccie_voice@onlinestudylist.com; Vik Malhi; Ryan Barnum
Subject: Re: [OSL | CCIE_Voice] Volume 1 Lab 5 Proctor Guide needs to bereworked

 

We'll make sure the latest files are uploaded. All of this was updated and 
fixed. 

 

Regards,

 

Wayne A. Lawson II - CCIE #5244

Founder  President - IPexpert

Mailto: wlaw...@ipexpert.com

Telephone: +1.810.326.1444, ext. 101

Live Assistance, Please visit: www.ipexpert.com/chat

eFax: +1.810.454.0130

 

::Message sent from iPhone::

 

IPexpert is a premier provider of Classroom and Self-Study Cisco CCNA (RS, 
Voice  Security), CCNP, CCVP, CCSP and CCIE (RS, Voice, Security  Service 
Provider) Certification Training with locations throughout the United States, 
Europe and Australia. Be sure to check out our online communities at 
www.ipexpert.com/communities and our public website at www.ipexpert.com. 


On Feb 22, 2010, at 10:10 AM, CCIETalk.com cciet...@gmail.com wrote:

Yeah it has TONS of errors and not for the faint hearted :D

On Mon, Feb 22, 2010 at 8:50 AM, t n tnn...@gmail.com wrote:

Hello,

I am currently on lab 5. From the beginning, it's already missing a few 
crucial items. For example, there is no discussion of the partitions/CSS for 
the phones. Also, the previous lab had the gateways used in the RPs. It's also 
missing the verifications and explanations seen in the previous 4 labs. 

It seems like the person who wrote the PG had a lot of energy for the 
first 4 labs. Then in lab 5 he/she got tired or bored and  decided it was 
alright to skip many steps.

Can someone at Ipexpert please look at the the PG for this lab? The 
quality is inconsistent with what I have seen so far. 
-- 
Thanks.

tnn314.wordpress.com

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please visit www.ipexpert.com




-- 
www.ccietalk.com

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Re: [OSL | CCIE_Voice] Voice Vol 1 Lab 5c task 5.3

[Steve Denney (stdenney)]

Did you include clid strip name in your outgoing 911 pots dial peer?

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of
CCIETalk.com
Sent: Monday, February 22, 2010 2:03 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Voice Vol 1 Lab 5c task 5.3

 

So I got this working without a big deal but for some reason my caller
name is NOT getting blocked from HQ Phones to 911. I looked up the
solution guide and they did a voice dial peer on HQ GW. I tried that
route but still no work. When I do debug isdn q931 on HQ GW I see both
calling name and number going to PSTN.

 



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Re: [OSL | CCIE_Voice] Vol 1 Lab 5A Task 5.2 - \Cannot reach thenumber\ display

[Steve Denney (stdenney)]

I assume you mean BR1 Ph2. 
I encountered this same problem again on another pod. Restarting the CUCMs 
finally fixed it.




From: Wael Agina wag...@thrupoint.net 
To: Steve Denney (stdenney) 
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com; 
waelag...@gmail.com waelag...@gmail.com 
Sent: Sun Feb 21 05:29:10 2010
Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 5A Task 5.2 - \Cannot reach 
thenumber\ display 



Dear Team,

 

  I am facing exactly the same issue here.

Br2 ph 2 cant place any call to 911.

It is exactly same situation – just one RP 911 point to RL_LOCAL and hence the 
DP RG points to the Br1 GW.

The GW seems not to get any call at all.

 

Any more suggestions ?

 

Note: I am working on Pod16.

 

Regards,

Wael Agina



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immediately by replying to the message and deleting it from your computer. 
Thankyou. ThruPoint Ltd. 

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[OSL | CCIE_Voice] PG Vol 1 Lab 5A Task 5.9 - no pt-internal partition

[Steve Denney (stdenney)]

Happy Sunday, fellow lab rats  :)

 

Noticed something odd with PG Vol 1 Lab 5A Task 5.9 (setting up a hunt
group). 

 

The solution calls for the hunt pilot to be placed in the pt-internal
partition, as well as for the line group members to be in that same
partition. But this partition does not exist. 

It appears that this is a holdover from the earlier version of Lab 5A
(in which a pt-internal partition and a css-internal CSS were created in
earlier tasks, which no longer appear in the revised version of Lab 5A).

 

This isn't really an issue (just using the None partition works fine)
until you get to the next task, 5.10. At that point, partition order
(due to time schedules) becomes important, and the pilot will need to be
assigned to some partition. I ran out of lab time before getting much
further, but here's the question: What steps need to be added prior to
task 5.9 to get the partitions / CSS's in order? (I can of course cobble
together my own solution, but I don't know what ripple effect it might
have on the rest of the lab tasks.)

 

Secondary question: Is this (no pt-internal or css-internal) also an
issue for Lab 5C?

 

Final note: There is still a mismatch between the WB and PG numbering
and content, starting at WB / PG 5.7. 

It would really be helpful if the docs posted on the Web site were
tagged with last revised dates (like they used to be on the old site),
so we would know at a glance when content has changed.

 

thanx, sd

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Re: [OSL | CCIE_Voice] Vol 1 Lab 5A Task 5.2 - Cannot reach the number display

[Steve Denney (stdenney)]

Thanks for the tips - I did reset everything in sight :) 

 

Just to close the loop, I strongly suspect some kind of issue with the
connection between the BR1 and the PSTN-WAN router - even in subsequent
tasks, no calls at all were ever able to route out the BR1 PSTN, even
though the L2 connectivity looked OK from both sides. I had major
problems with the initial load of the PSTN-WAN router during this lab,
and needed tech support to manually go in and load the initial configs -
since I never found a smoking gun, I'm blaming my subsequent issues on
gremlins related to that. ;)

 

cheers, sd

 

From: vccie2010 [mailto:vccie2...@gmail.com] 
Sent: Sunday, February 14, 2010 9:41 PM
To: Steve Denney (stdenney)
Cc: OSL Group
Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 5A Task 5.2 - Cannot reach
the number display

 

I hope you tried Reset DP and last resort CCM service too ?

On Fri, Feb 12, 2010 at 8:50 AM, Steve Denney (stdenney)
stden...@cisco.com wrote:

Racking my brains a bit over this one...

 

Trying to place a call from BR1 Ph2 (SCCP IP Blue) to 911 (PSTN phone). 

Getting a fast busy and this display on the phone: Cannot reach the
number.

 

Calls from HQ Ph2 (SIP CICP) to 911 work fine - so I know the CUCM route
pattern for 911 is working.

Also, getting the secondary dial tone when the 9 is dialed from BR1 Ph2
- so the phone is definitely hitting the route pattern.

The call just never seems to reach the BR1 MGCP gateway. 

 

911 Route Pattern (None partition) points to rl-local-gw, which points
to Standard Local Route Group. 

Device Pool BR1 is configured to use Local Route Group of rg-br1.

Route group rg-br1 has the proper GW selected
(S0/SU0/ds...@br-rtr.proctorlabs.com).

BR1 GW is cleanly registered to CUCM, with multiple_frame_established.

Have bounced no mgcp / mgcp on BR1 GW, and have reset the phone and
route list, multiple times.

There are no other route patterns configured which could be conflicting.

 

Any thoughts?

 

thx, sd

 


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[OSL | CCIE_Voice] Vol 1 Lab 5A Task 5.2 - Cannot reach the number display

[Steve Denney (stdenney)]

Racking my brains a bit over this one...

 

Trying to place a call from BR1 Ph2 (SCCP IP Blue) to 911 (PSTN phone). 

Getting a fast busy and this display on the phone: Cannot reach the
number.

 

Calls from HQ Ph2 (SIP CICP) to 911 work fine - so I know the CUCM route
pattern for 911 is working.

Also, getting the secondary dial tone when the 9 is dialed from BR1 Ph2
- so the phone is definitely hitting the route pattern.

The call just never seems to reach the BR1 MGCP gateway. 

 

911 Route Pattern (None partition) points to rl-local-gw, which points
to Standard Local Route Group. 

Device Pool BR1 is configured to use Local Route Group of rg-br1.

Route group rg-br1 has the proper GW selected
(S0/SU0/ds...@br-rtr.proctorlabs.com).

BR1 GW is cleanly registered to CUCM, with multiple_frame_established.

Have bounced no mgcp / mgcp on BR1 GW, and have reset the phone and
route list, multiple times.

There are no other route patterns configured which could be conflicting.

 

Any thoughts?

 

thx, sd

 

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Re: [OSL | CCIE_Voice] Vol 1 Lab 5A Task 5.2 - Cannot reach the number display

[Steve Denney (stdenney)]

That's part of the problem. :) The call never reaches the BR1 GW, so
there is no debug isdn q931 output at all.

 

 

From: vccie2010 [mailto:vccie2...@gmail.com] 
Sent: Friday, February 12, 2010 2:03 PM
To: Steve Denney (stdenney)
Cc: OSL Group
Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 5A Task 5.2 - Cannot reach
the number display

 

Can you pls post the debug isdn q931 output here to further
troubleshoot it.

On Fri, Feb 12, 2010 at 8:50 AM, Steve Denney (stdenney)
stden...@cisco.com wrote:

Racking my brains a bit over this one...

 

Trying to place a call from BR1 Ph2 (SCCP IP Blue) to 911 (PSTN phone). 

Getting a fast busy and this display on the phone: Cannot reach the
number.

 

Calls from HQ Ph2 (SIP CICP) to 911 work fine - so I know the CUCM route
pattern for 911 is working.

Also, getting the secondary dial tone when the 9 is dialed from BR1 Ph2
- so the phone is definitely hitting the route pattern.

The call just never seems to reach the BR1 MGCP gateway. 

 

911 Route Pattern (None partition) points to rl-local-gw, which points
to Standard Local Route Group. 

Device Pool BR1 is configured to use Local Route Group of rg-br1.

Route group rg-br1 has the proper GW selected
(S0/SU0/ds...@br-rtr.proctorlabs.com).

BR1 GW is cleanly registered to CUCM, with multiple_frame_established.

Have bounced no mgcp / mgcp on BR1 GW, and have reset the phone and
route list, multiple times.

There are no other route patterns configured which could be conflicting.

 

Any thoughts?

 

thx, sd

 


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please visit www.ipexpert.com http://www.ipexpert.com/ 

 

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Re: [OSL | CCIE_Voice] Vol 1 Lab 5A Task 5.2 - Cannot reach thenumber display

[Steve Denney (stdenney)]

DNA output shows that the call is making it to the Route List
rl-local-gw:

*   Results Summary 

*   Calling Party Information 

*   Calling Party = 1002 
*   Partition = 
*   Device CSS = 
*   Line CSS = 
*   AAR Group Name = 
*   AAR CSS = 

*   Dialed Digits = 911 
*   Match Result = RouteThisPattern 
*   Matched Pattern Information 

*   Pattern = 911 
*   Partition = 
*   Time Schedule = 

*   Called Party Number = 911 
*   Time Zone = 
*   End Device = rl-local-gw 
*   Call Classification = OffNet 
*   InterDigit Timeout = NO 
*   Device Override = Disabled 
*   Outside Dial Tone = NO 

*   Call Flow 

*   TranslationPattern :Pattern= 

*   Positional Match List = 911 
*   Calling Party Number = 1002 
*   PreTransform Calling Party Number = 
*   PreTransform Called Party Number = 
*   Calling Party Transformations 

*   External Phone Number Mask = NO 
*   Calling Party Mask = 
*   Prefix = 
*   CallingLineId Presentation = 
*   CallingName Presentation = 
*   Calling Party Number = 1002 

*   ConnectedParty Transformations 

*   ConnectedLineId Presentation = 
*   ConnectedName Presentation = 

*   Called Party Transformations 

*   Called Party Mask = 
*   Discard Digits Instruction = 
*   Prefix = 
*   Called Number = 

*   Route Pattern :Pattern= 911 

*   Positional Match List = 911 
*   DialPlan = 
*   Route Filter 

*   Filter Name = 
*   Filter Clause = 

*   Require Forced Authorization Code = No 
*   Authorization Level = 0 
*   Require Client Matter Code = No 
*   Call Classification = 
*   PreTransform Calling Party Number = 1002 
*   PreTransform Called Party Number = 911 
*   Calling Party Transformations 

*   External Phone Number Mask = NO 
*   Calling Party Mask = 
*   Prefix = 
*   CallingLineId Presentation = Default 
*   CallingName Presentation = Default 
*   Calling Party Number = 1002 

*   ConnectedParty Transformations 

*   ConnectedLineId Presentation = Default 
*   ConnectedName Presentation = Default 

*   Called Party Transformations 

*   Called Party Mask = 
*   Discard Digits Instruction = None 
*   Prefix = 
*   Called Number = 911 

*   Route List :Route List Name= rl-local-gw 

*   RouteGroup :RouteGroup Name= Standard Local
Route Group 

*   PreTransform Calling Party Number = 1002

*   PreTransform Called Party Number = 911 
*   Calling Party Transformations 

*   External Phone Number Mask =
Default 
*   Calling Party Mask = 
*   Prefix = 
*   Calling Party Number = 1002 

*   Called Party Transformations 

*   Called Party Mask = 
*   Discard Digits Instructions = 
*   Prefix = 
*   Called Number = 911 

*   Alternate Matches 

*   Note: Information Not Available 

Seems like it's not getting past the rl-local-gw. 

But rl-local-gw *is* working when the call comes from a phone in the HQ
DP. 

It's only with the BR1 phone that there is an issue.

 

Sigh...thanx for the tips so far...

 

 

 

-Original Message-
From: Kevin Damisch [mailto:kevin.dami...@vitalsite.com] 
Sent: Friday, February 12, 2010 2:39 PM
To: Steve Denney (stdenney); vccie2010
Cc: OSL Group
Subject: RE: [OSL

Re: [OSL | CCIE_Voice] Vol1 Lab 5 - Mismatch on WB and PG?

[Steve Denney (stdenney)]

A few follow-up questions... :)

1) Which is the correct question for 5.7 (the one in the WB about CME,
or the one in the PG about 900 #s)?
2) Has the doc (whichever one was incorrect) been updated yet?
3) (now we get to my real question)...Would it be possible to include
Last Revision Date info alongside the files in the downloads area,
similar to how they used to be listed on the old Web site? Would
probably save us all a lot of time and trouble if we could see at a
glance when a doc has been updated.

thanx, sd


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vik Malhi
Sent: Monday, February 01, 2010 12:28 AM
To: Stephen Greszczyszyn; OSL Group
Subject: Re: [OSL | CCIE_Voice] Vol1 Lab 5 - Mismatch on WB and PG?

Yep. You got me. Missed Q5.7 in lab 5A. Should be resolved. Thanks.

-- 
Vik Malhi - CCIE #13890
Instructor - IPexpert, Inc.
Mailto: vma...@ipexpert.com
Telephone: +1.810.326.1444
Fax: +1.810.454.0130
Live Assistance, Please visit: www.ipexpert.com/chat
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IPexpert is a premier provider of Classroom and Self-Study Cisco CCNA
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communities at
www.ipexpert.com/communities http://www.ipexpert.com/communities  and
our
public website at www.ipexpert.com http://www.ipexpert.com .


 From: Stephen Greszczyszyn sgres...@gmail.com
 Date: Sun, 31 Jan 2010 19:10:10 +
 To: OSL Group ccie_voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice] Vol1 Lab 5 - Mismatch on WB and PG?
 
 I'm working through the new Vol1 Lab 5, and there seems to be a
 mismatch in the questions asked and the proctor guide solutions
 starting at section 5.7 in Lab A.
 
 In the Work Book, it talks about UCME routing.  In the Proctor Guide
 it describes adding 900-number restrictions (I think from the old
 version of Vol1 Lab 5).
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 visit www.ipexpert.com


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[OSL | CCIE_Voice] Vol1 Lab 4A - X-Lite Issues

[Steve Denney (stdenney)]

Hitting an interesting problem and just wondering if anyone else has
seen similar symptoms...

 

Working on Vol1, Lab 4A, Task 4.5. 

This is the task where you set up a SIP Route Pattern and use SIP URI
dialing to dial the X-Lite CME SIP Phone (BR2 Ph 4, DN 3006) from the
CIPC SIP Phone (HQ Ph2, DN 5002).

 

When dialing from 5002 to 3006 (using the corporate directory on CIPC,
as shown in the lab), the X-Lite rings, but hangs up immediately after
the call is answered.

The output of debug ccsip mess is attached. Looks like the X-Lite is
sending a SIP BYE message with the description of Illegal Sdp
Negotiation.

 

I tried a call in the other direction as well - direct dial from 3006 to
5002. The CIPC rings, but you cannot actually answer the call. 

The debug in this case shows a 503 Service Unavailable message, and
the display on the X-Lite says Call failed: Service Unavailable.

 

I've double and triple checked all configs (including allow-connections
sip to sip), reloaded all routers, Googled for similar issues, and am
now officially stumped. :)

Debugs attached. Any ideas?

 

cheers, steve

 

 

!
! Call flow: HQ Ph2 (CIPC SIP - DN 5002) calls BR2 Ph4 (X-Lite SIP - DN 3006)
!

BR2-RTR#
Feb  4 19:57:49.795: //-1//SIP/Msg/ccsipDisplayMsg:
Received: 
INVITE sip:3...@ipxcme.com:5060 SIP/2.0
Date: Thu, 04 Feb 2010 22:09:06 GMT
Call-Info: 
sip:10.10.210.11:5060;method=NOTIFY;Event=telephone-event;Duration=500
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, 
SUBSCRIBE, NOTIFY
From: HQ Ph2 
sip:5...@10.10.210.11;tag=ad370c0e-63ed-49fa-adb9-1bd65232089c-44886080
Allow-Events: presence, kpml
P-Asserted-Identity: HQ Ph2 sip:5...@10.10.210.11
Supported: 100rel,timer,resource-priority,replaces
Min-SE:  1800
Remote-Party-ID: HQ Ph2 
sip:5...@10.10.210.11;party=calling;screen=yes;privacy=off
Content-Length: 0
User-Agent: Cisco-CUCM7.0
To: sip:3...@10.10.202.1
Contact: sip:5...@10.10.210.11:5060;transport=tcp
Expires: 180
Call-ID: e7d63280-b6b14582-4d-bd20...@10.10.210.11
Via: SIP/2.0/TCP 10.10.210.11:5060;branch=z9hG4bK3054718c7
CSeq: 101 INVITE
Session-Expires:  1800
Max-Forwards: 69


BR2-RTR#
Feb  4 19:57:49.815: //-1//SIP/Msg/ccsipDisplayMsg:
Sent: 
INVITE sip:3...@10.10.0.98:43748 SIP/2.0
Via: SIP/2.0/UDP 10.10.202.1:5060;branch=z9hG4bK91678
Remote-Party-ID: HQ Ph2 
sip:5...@10.10.202.1;party=calling;screen=yes;privacy=off
From: HQ Ph2 sip:5...@10.10.202.1;tag=3B5C6C-863
To: sip:3...@10.10.0.98
Date: Thu, 04 Feb 2010 19:57:49 GMT
Call-ID: 6811fa76-10fe11df-8049d3ba-227bb...@10.10.202.1
Supported: 100rel,timer,resource-priority,replaces
Min-SE:  1800
Cisco-Guid: 1745808662-285086175-2151928762-578533463
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, 
NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1265313469
Contact: sip:5...@10.10.202.1:5060
Call-Info: 
sip:10.10.202.1:5060;method=NOTIFY;Event=telephone-event;Duration=2000
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 68
Session-Expires:  1800
Content-Length: 0


Feb  4 19:57:49.819: //-1//SIP/Msg/ccsipDisplayMsg:
Sent: 
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.10.210.11:5060;branch=z9hG4bK3054718c7
From: HQ Ph2 
sip:5...@10.10.210.11;tag=ad370c0e-63ed-49fa-adb9-1bd65232089c-44886080
To: sip:3...@10.10.202.1
Date: Thu, 04 Feb 2010 19:57:49 GMT
Call-ID: e7d63280-b6b14582-4d-bd20...@10.10.210.11
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 INVITE
Allow-Events: telephone-event
Content-Length: 0


Feb  4 19:57:50.047: //-1//SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.10.202.1:5060;branch=z9hG4bK91678
Contact: sip:3...@10.10.0.98:43748
To: sip:3...@10.10.0.98;tag=c97b3344
From: HQ Ph2sip:5...@10.10.202.1;tag=3B5C6C-863
Call-ID: 6811fa76-10fe11df-8049d3ba-227bb...@10.10.202.1
CSeq: 101 INVITE
User-Agent: X-Lite release 1103k stamp 53621
Content-Length: 0


Feb  4 19:57:50.051: //-1//SIP/Msg/ccsipDisplayMsg:
Sent: 
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 10.10.210.11:5060;branch=z9hG4bK3054718c7
From: HQ Ph2 
sip:5...@10.10.210.11;tag=ad370c0e-63ed-49fa-adb9-1bd65232089c-44886080
To: sip:3...@10.10.202.1;tag=3B5D58-D78
Date: Thu, 04 Feb 2010 19:57:49 GMT
Call-ID: e7d63280-b6b14582-4d-bd20...@10.10.210.11
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, 
NOTIFY, INFO, REGISTER
Require: 100rel
RSeq: 510
Allow-Events: telephone-event
Remote-Party-ID: sip:3...@10.10.202.1;party=called;screen=no;privacy=off
Contact: sip:3...@10.10.202.1:5060;transport=tcp
Content-Length: 0


Feb  4 19:57:50.071: //-1//SIP/Msg/ccsipDisplayMsg:
Received: 
PRACK sip:3...@10.10.202.1:5060;transport=tcp SIP/2.0
Date: Thu, 04 Feb 2010 22:09:06 GMT
From: HQ Ph2 
sip:5...@10.10.210.11;tag=ad370c0e-63ed-49fa-adb9-1bd65232089c-44886080
RAck: 510 101 INVITE
Allow-Events: presence, kpml
Content-Length: 0
To: 

Re: [OSL | CCIE_Voice] CME files for phones

[Steve Denney (stdenney)]

Very good info Matthew. 
FYI, the actual load files for 7945/65s with CME 7.0(1) (i.e., 12.4.22(T); 
phone firmware version 8.3.3S) are as follows:
(SCCP Files):
SCCP45.8-3-3S.loads
term45.default.loads 
term65.default.loads 
apps45.8-3-2-27.sbn 
cnu45.8-3-2-27.sbn 
cvm45sccp.8-3-2-27.sbn 
dsp45.8-3-2-27.sbn 
jar45sccp.8-3-2-27.sbn 

(SIP Files - same as SCCP files, except for these 3 differences):
SIP45.8-3-3S.loads
cvm45sip.8-3-2-27.sbn 
jar45sip.8-3-2-27.sbn

I highly suggest taking a look at this page before sitting for the actual lab:
Cisco Unified CME 7.0(1) Supported Firmware, Platforms, Memory, and Voice 
Products
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/requirements/guide/cme701spc.htm

cheers, sd


Steve Denney, CISSP
Systems Engineer - Technology Solutions Network
Voice and Unified Communications Products
Cisco Systems, Inc.
125 High Street, 21st Floor
Boston, MA  02110
978-936-4048 (Office)
617-872-5031 (Mobile)
stden...@cisco.com



-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Berry, Matthew J.
Sent: Wednesday, January 27, 2010 8:25 AM
To: Randall Crumm; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CME files for phones

Randall -

Those four file types make up the firmware that the Cisco IP phone uses.  The 
7940/7960 phones use the .sb2 and .bin commands.  Proctor Labs (ie. IP Expert) 
uses 7960s in their racks.  However, the actual lab is going to use 7965s.  If 
you look at the newer phone models, such as the 7965, they don't use the same 
images and file types as previous versions.

Take a look at the CUCME SRND:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeinstl.html#wp1070512

The following example shows a list of phone firmware files that are installed 
in flash memory for the Cisco Unified IP Phone 7911:

tftp-server flash:SCCP11.7-2-1-0S.loads
tftp-server flash:term06.default.loads
tftp-server flash:term11.default.loads
tftp-server flash:cvm11.7-2-0-66.sbn
tftp-server flash:jar11.7-2-0-66.sbn
tftp-server flash:dsp11.1-0-0-73.sbn
tftp-server flash:apps11.1-0-0-72.sbn
tftp-server flash:cnu11.3-0-0-81.sbn

Here's an example of the firmware files used for the 7911 IP Phone.  The format 
will be the same for the 7941, 7961, 7965, etc.  Become familiar with this new 
file format because that's what you'll see on the lab.

Hope this helps!

Thanks,

Matthew Berry
Office 952 516 3748  |  Mobile 952 221 2814|  mjbe...@krollontrack.com

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Randall Crumm
Sent: Tuesday, January 26, 2010 11:48 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] CME files for phones

Hi,
I want to know what are the four files in CME? I don't work with CME and I am 
looking at lab 3A
.bin
.loads
.sb2
.sbn

Thanks,
Randall

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Re: [OSL | CCIE_Voice] SIP load for 7941/61 and CME

[Steve Denney (stdenney)]

Hi Bill,

 

This site might be helpful for you:

 

Cisco Unified CME 7.0(1) Supported Firmware, Platforms, Memory, and
Voice Products

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/requirements/guide/c
me701spc.htm

 

cheers, sd

 

Steve Denney, CISSP

Systems Engineer - Technology Solutions Network

Voice and Unified Communications Products

Cisco Systems, Inc.

125 High Street, 21st Floor

Boston, MA  02110

978-936-4048 (Office)

617-872-5031 (Mobile)

stden...@cisco.com mailto:stden...@cisco.com  

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Bill
Hatcher
Sent: Monday, January 25, 2010 6:58 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] SIP load for 7941/61 and CME

 

Can someone please tell me what load to use for SIP on the CME for the
Cisco 7941/61 phone model?  I see 3 types of files to download, the .zip
the .cop and the .cop.sgn file on Cisco's site, but they all referance
CallManager only.

 

Bill

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[OSL | CCIE_Voice] Question on Terminal Emulators

[Steve Denney (stdenney)]

General question for the list - What terminal emulator / SSH client do
you like to use for labs? 

I know SecureCRT is recommended...are there reasonably good freeware
alternatives that people like?

 

Please don't suggest Windows HyperTerm :)

 

cheers, sd

 

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[OSL | CCIE_Voice] Continuing Web Site Issues

[Steve Denney (stdenney)]

I really hate to ask this in the public forum, but since support is no
longer responding to my emails, I guess I have no other choice...

 

I've been hearing hang on for a few more hours for three days now. 

Is there any more updated information about when paying customers will
be able to log in to the Web site, and open their secure Adobe docs?

 

Between this little kerfuffle, the numerous typos in the voice
workbooks, the delays in production of the balance of the long-promised
v3 material, and the nickel-and-diming attitude towards the product
updates that *do* come out, my patience is pretty much at an end, as is
my recommendation of IPexpert as a vendor. I can't tell you how
disappointed I am.

 

Regards, Steve

 

Steve Denney, CISSP

Systems Engineer - Technology Solutions Network

Voice and Unified Communications Products

Cisco Systems, Inc.

125 High Street, 21st Floor

Boston, MA  02110

stden...@cisco.com mailto:stden...@cisco.com  

 

 

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Re: [OSL | CCIE_Voice] Attention IPexpert Members

[Steve Denney (stdenney)]

No Brian, you're not alone. 

 

I was also informed yesterday that I'd have to re-download all of my
ebooks (once I'm able to log in to the new site, whenever that may be)
in order for the Adobe secure authentication to work. (As if I had
nothing better to do in my copious spare time...) 

 

This is beyond frustrating. I'm trying to give IPexpert the benefit of
the doubt here, but I'm about ready to cut my losses and look for
another training vendor.

 

 

Steve Denney, CISSP

Systems Engineer - Technology Solutions Network

Voice and Unified Communications Products

Cisco Systems, Inc.

125 High Street, 21st Floor

Boston, MA  02110

stden...@cisco.com 

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Brian
Valentine
Sent: Wednesday, December 09, 2009 7:26 AM
To: 'Drew LePla'; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Attention IPexpert Members

 

Since this deadline has passed, should I assume that my account is the
only one that still doesn't work?

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Drew LePla
Sent: Monday, December 07, 2009 2:54 PM
To: ccie...@onlinestudylist.com; ccie_voice@onlinestudylist.com;
ccie_secur...@onlinestudylist.com; ccie...@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Attention IPexpert Members

 

Attention IPexpert Members,

 

 As most of you are aware, we launched a new website on Friday
(December 4th). The new website is tightly integrated with our
Salesforce account structure, therefore in order to better support our
client base, we've changed the login method that needs to be followed
when accessing your IPexpert Members Account.

 

 We are currently in the process of cutting over all accounts.  This
new solution will be effective and in place no later than Tuesday,
December 8th, at noon EST.  The process to login will be as follows:

 

1.   Visit the www.ipexpert.com website, click on Client Login

2.   On the left side of the page, you will find a Current
Customers area with Email / Username and Password, enter your
CURRENT username and password.

3.   You will then be walked through an Account Migration process.
Your FileOpen login and Members Login will be converted to your email
address and your password of choice upon confirming your email address
on file.

 

 If you have any issues or problems, please contact
supp...@ipexpert.com or call at +1.810.326.1444.

 

 

Regards,

 

Drew LePla - COMP TIA A+, CCNA - IPexpert

Lead Technical Support Engineer

Mailto: dle...@ipexpert.com

Telephone: +1.810.326.1444, ext. 204

Live Assistance, Please visit: www.ipexpert.com/chat

eFax: +1.810.454.0130

 

IPexpert is a premier provider of Classroom and Self-Study Cisco CCNA
(RS, Voice  Security), CCNP, CCVP, CCSP and CCIE (RS, Voice, Security
 Service Provider) Certification Training with locations throughout the
United States, Europe and Australia. Be sure to check out our online
communities at www.ipexpert.com/communities and our public website at
www.ipexpert.com

 

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[OSL | CCIE_Voice] IPexpert Web site - still issues?

[Steve Denney (stdenney)]

Sorry to clog up the study list, but like (apparently) many others
today, I'm having ongoing issues with the new IPexpert Web site.

 

Response / refresh time is horrible, with pages refreshing unacceptably
slow, or not correctly at all (missing images, etc.).

Also, have not been able to log in to the Members area, using either my
old user ID or my email address.

 

Thinking that you might want to consider a backout / contingency plan at
this point, folks...until the new site has been properly tested...

 

Regards,

 

Steve Denney, CISSP

Systems Engineer - Technology Solutions Network

Voice and Unified Communications Products

Cisco Systems, Inc.

125 High Street, 21st Floor

Boston, MA  02110

stden...@cisco.com mailto:stden...@cisco.com  

 

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Re: [OSL | CCIE_Voice] IPExpert - Chock up Another Winner!

[Steve Denney (stdenney)]

Way to go Kevin! Congrats.

 

 

Steve Denney, CISSP

Systems Engineer - Technology Solutions Network

Voice and Unified Communications Products

Cisco Systems, Inc.

125 High Street, 21st Floor

Boston, MA  02110

978-936-4048 (Office)

617-872-5031 (Mobile)

stden...@cisco.com 

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kevin
Porter
Sent: Friday, November 20, 2009 5:21 PM
To: 'ccie_voice@onlinestudylist.com'
Subject: [OSL | CCIE_Voice] IPExpert - Chock up Another Winner!

 

Thanks IPExpert,

I just got the news that I PASSED the Lab yesterday.  Pulled number
25730!!!

 

Everyone out there that is studying IPExpert Materials, keep on
truckin', you'll get your day.  

I know everyone wants to know about preparation, well when you get to
the point of completing the Volume 2 labs in about 7 hours, you should
be in pretty good shape.  Also, all I can say is, over 10,000 Cisco IP
Phones deployed, 26 CUCM Clusters deployed (version 3.3 through 7.1.3a),
with Unity (version 4.0 through 7.0), Unity Connections (1.2 through
7.1.3a), CUE (all versions), IPCC/UCCX (version 4.0 through 7.0,
Presence versions 7.0 and 7.1.3a), Emergency Responder (version 1.3
through 7.0), etc...Basically, I'm saying you cannot beat real-world
experience along with the excellent practice labs from IPExpert.  People
have been saying for years that the Lab was also so far out in the not
Real-world land that it is unfair, well, I can truthfully say that the
Lab is not that far away from real-world now and I personally think the
two will completely intersect in the very near future, especially with
Fully compliant E.164 providers now on the scene.  For instance, I just
turned up a SIP trunk with Level 3 communications that sends all calling
and called party numbers in + format and demands that outbound calls do
the same

Anyway, Thanks again IPExpert for the awesome training materials and
everyone, keep pounding away and you'll get a number too...

Thanks,

Kevin Porter

 

Kevin S. Porter
Systems Engineer

kpor...@netelligent.com mailto:kpor...@netelligent.com 

(p) 314.392.6921
(f) 314.392.5421

  http://www.netelligent.com/ 

 

 

 



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