Re: [OSL | CCIE_Voice] OT : Missing normal softkeys from 9971.

2011-03-17 Thread Pulos, Greg
WOW!!!  Somebody feels entitled.

Quit with the complaining about the help/response you get from these very hard 
working, hard studying people and perhaps be more focused on the problem you 
need answered; and marking an OT in the subject line..?!?!?!?!

How in the world is someone supposed to know you mean 'Off Topic'???I read 
it as 'Out of Touch'.

As you've noticed, not many others have commented, likely due to the severe 
tongue lashing you give when you don't get the answer you expect.

I for one am with the first response.CISCO.COM has all your answers and if 
you expect to be any good in the networking field with cisco technologies, you 
better KNOW how to find the answers yourself.

greg

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Rogers Ochieng
Sent: Thursday, March 17, 2011 2:02 AM
To: ccielabrat
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] OT : Missing normal softkeys from 9971.


Lets be polite guys! People ask OT questions all the time, if i have a quick 
answer i'll give it or politely mention a useful link or forum. I saw someone 
throwing a tirade at Ash sometime back for OT, i think he didn't know that the 
guy is a CCIE and has a great blog,the kind of contribution the guy has been 
making. The point is that the very person you are being rude to can be the one 
with a goof insight on your future challenges unless you already got your number
On Mar 17, 2011 7:36 AM, "ccielabrat" 
mailto:ccielab...@gmail.com>> wrote:
Randall & Roger,

Are you kidding me?

I admit it was a dumb question and I tried to be polite by marking it OT: (Off 
Topic)
I was stuck in a lab with no direct internet access or a manual for a phone I 
never used before.
I leveraged the mailer group on my phone looking for a hand.
A group, which for the most part, is helpful whether or not it has anything 
specifically to do with the voice lab.

I got several quick replies with useful information.
The thread could have died 5 minutes after I posted it.
In the time it took to reply, both of you could have deleted it and moved on 
with your day.
I don't see either of you replying back to morons who blatantly ask for NDA 
information.

Your arrogance is laughable, to think either of you have a place to suggest 
what does or doesn't belong on this mailing list.

If you have a problem with it , take it up with the mailer admin.
Or email me a picture of your mailer police badge , and then I'll consider your 
email.

I honestly hope if you are ever in the same position, you both get a helpful 
reply instead of the nonsense you both provided.




On Wed, Mar 16, 2011 at 9:21 PM, Randall Saborío Cubero 
mailto:ill2...@gmail.com>> wrote:
>
> It's diff...

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Re: [OSL | CCIE_Voice] why is vad bad?

2011-02-09 Thread Pulos, Greg
Per Cisco, when VAD is enabled, it contributes to a slightly degraded sound 
quality.

Also, VAD is to be disabled on both sides of a FAX connection as it can 
interfere with the successful reception of the fax traffic.

Per experience, unless VAD actually alleviates bandwidth problem, it is likely 
best to keep it disabled in most applications.

Please see the following link for more information on VAD:

http://www.cisco.com/en/US/docs/ios/voice/command/reference/vr_v1.html#wp1103127

Thank you.

Greg

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Stutz, Bernhard
Sent: Wednesday, February 09, 2011 10:43 AM
To: matt...@ciscovoiceguru.com
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] why is vad bad?

Hm this depends on your available ISDN Channels.
You may have 1000 phones but you can only have for example max 30 or lets say 
60 channels available if you have a standard PRI to PSTN.
You may run into this issue if you got a SIP Trunk to a SIP Provider with 
multiple channels but 60 voip legs shouldn't be a problem with a 2921 I hope ;-)
Or is a SIP Trunk now Standard at US?

Cheers,
Bernhard


Von: matt...@ciscovoiceguru.com [mailto:matt...@ciscovoiceguru.com]
Gesendet: Mittwoch, 9. Februar 2011 16:25
An: Stutz, Bernhard
Cc: ccie_voice@onlinestudylist.com
Betreff: Re: [OSL | CCIE_Voice] why is vad bad?

I've always understood that VAD results in a higher CPU utilization.  For a 
site of 10 phones running a 2921 it wouldn't be an issue.  However, if you're 
running several hundred (or thousand) users running off the same pool of 
devices then you'd run into a significant impact on CPU performance.

Matthew Berry, CCIE #26721

Email: matt...@ciscovoiceguru.com
Twitter: http://twitter.com/CiscoVoiceGuru
Blog: http://ciscovoiceguru.com

On Feb 9, 2011, at 9:38 AM, Stutz, Bernhard wrote:

Hi,

i am just wondering why vad is bad and we all learn as a rule of thumb to 
disable vad on all voip dial peers?

When you have a look for what vad has been designed for it looks to me as a 
valuable algorithm (http://en.wikipedia.org/wiki/Voice_activity_detection)

Whats the reason we disable it all the time?
Is Cisco not able to support vad correctly or is it user experience that they 
want to hear a noise otherwise they think of that the connection has been lost? 
But therefore you have comfort-noise isn't it?

Kindly regards,
Bernhard

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Re: [OSL | CCIE_Voice] IP dhcp pool - back to basics

2011-01-26 Thread Pulos, Greg

Your example seems to make no sense since you talk about an IP range of one 
subnet, but then discuss excluding addresses from a completely different subnet.

Bottom line.if your IOS dhcp is handing out actual broadcast IDs as IP 
addys, then you have an IOS bug.

I would doubt this is the case though as I've never seen this in working with 
IOS for 20+ years.

If .255 was being handed out, then you are likely not using a 24 bit mask on 
your dhcp pool.

Also, I see no where in any cisco doc that states a properly configured 24 bit 
subnet in dhcp may "possibly hand out a broadcast address". it just can't 
happen if IOS is not buggy and the configuration of dhcp is correct.

greg


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Shrini
Sent: Wednesday, January 26, 2011 12:54 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] IP dhcp pool - back to basics

Greetings:

If I need phones to be assigned only specific range of IPs , say from 
10.10.202.100 to 10.10.202.150 in a /24 network

ip dhcp excluded-address 10.21.20.1 10.21.20.99
ip dhcp excluded-address 10.21.20.151 10.21.20.255

http://www.cisco.com/en/US/docs/ios/12_3/ipaddr/command/reference/ip1_i1g.html#wp1080002

I have seen that some use the end address as 10.10.202.254 because 10.21.20.255 
is broadcast address.

But according to above document if we do not specify there is a possibility 
that 10.21.20.255 will be assigned, and happened with me couple of times.

Experts opinion requested.

Regards
Shrini
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Re: [OSL | CCIE_Voice] Deleting and Creating VM Box on Unity 5.0

2010-12-10 Thread Pulos, Greg
You CAN delete a voicemail box and KEEP the greetings and voicemails.


1)  Just pull the voicemails off via outlook express and save them.

2)  Then pull the greetings off via Unity Administration.

3)  Delete the mailbox and recreate.

4)  Then you can add the voicemails back into the new mailbox and copy the 
greetings back to their respective places via Unity Administration.

greg


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Padilla
Sent: Friday, December 10, 2010 1:39 PM
To: tanner.ez...@gmail.com
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Deleting and Creating VM Box on Unity 5.0

Will Unity allow you to re add the mailbox, even if the account already resides 
on exchange? I thought you would get an error.

> Date: Fri, 10 Dec 2010 10:35:33 -0800
> Subject: Re: [OSL | CCIE_Voice] Deleting and Creating VM Box on Unity 5.0
> From: tanner.ez...@gmail.com
> To: clane...@hotmail.com
> CC: ccie_voice@onlinestudylist.com
>
> You cannot delete the voice mail box and save the greeting. Messages
> should be fine so long as the actual mailbox isn't deleted.
>
> On Fri, Dec 10, 2010 at 10:20 AM, Mark Padilla  wrote:
> > I was wondering if anyone knew of a way to delete and create and or move a
> > users VM Box with saving the users VM and Greeting. Please let me know if
> > anyone has a better way of moving this information other than deleting and
> > having the user recreate evertyhing.
> >
> >> From: ccie_voice-requ...@onlinestudylist.com
> >> Subject: CCIE_Voice Digest, Vol 58, Issue 32
> >> To: ccie_voice@onlinestudylist.com
> >> Date: Fri, 10 Dec 2010 12:00:03 -0500
> >>
> >> Send CCIE_Voice mailing list submissions to
> >> ccie_voice@onlinestudylist.com
> >>
> >> To subscribe or unsubscribe via the World Wide Web, visit
> >> http://onlinestudylist.com/mailman/listinfo/ccie_voice
> >> or, via email, send a message with subject or body 'help' to
> >> ccie_voice-requ...@onlinestudylist.com
> >>
> >> You can reach the person managing the list at
> >> ccie_voice-ow...@onlinestudylist.com
> >>
> >> When replying, please edit your Subject line so it is more specific
> >> than "Re: Contents of CCIE_Voice digest..."
> >>
> >>
> >> Today's Topics:
> >>
> >> 1. CFUR and SRST - is there a bug? (Tam Nhu)
> >> 2. Re: CFUR display For and By (Tam Nhu)
> >> 3. UCCX and CCM Integration Issue (shafqut hamid)
> >> 4. Re: UCCX and CCM Integration Issue (Prashant Patel)
> >>
> >>
> >> --
> >>
> >> Message: 1
> >> Date: Fri, 10 Dec 2010 09:11:47 -0600
> >> From: Tam Nhu 
> >> To: ccie_voice@onlinestudylist.com
> >> Subject: [OSL | CCIE_Voice] CFUR and SRST - is there a bug?
> >> Message-ID:
> >> 
> >> Content-Type: text/plain; charset="iso-8859-1"
> >>
> >> Hi Experts,
> >>
> >> Not sure if this subject has been posted before, but I could not find it
> >> in
> >> this forum. Here is the problem I just discovered a couple days back and
> >> can be duplicated in many labs.
> >>
> >> I have configured CFUR to display the correct full E164 digits in FOR and
> >> BY
> >> field; everything configured on CUCM, not the router. But when I finished
> >> my testing in SRST, and things are back to normal CUCM mode; I tested it
> >> again by shutdown the serial interface again to force it go back to SRST
> >> mode. Now, the FOR and BY field only display 4-digit, not full E164 as on
> >> the first test.
> >>
> >> I rebooted the router and went to SRST mode again, then everything is
> >> working with FOR and BY as expected. I then performed the same step above,
> >> no shut to back to CUCM mode, then shut to go in SRST mode, now I
> >> duplicated
> >> the same issue with FOR and BY display only 4-digit.
> >>
> >> I duplicated this many times on my own lab and IPX vRack lab as well. It
> >> does not matter the remote gateway is MGCP or H323 (I did test both with
> >> the
> >> same result). So looks like it is a bug to me and required to reboot the
> >> router every time after it went to SRST. I didn't know about this issue
> >> since we always have a fresh vRack session, and I only tested SRST once
> >> time, but not 2nd time on the same vrack session.
> >>
> >> Has anyone run into this before, and what is the bug ID if you have it? Is
> >> there another work-round to fix this?
> >>
> >> Thanks,
> >> Tam.
> >> -- next part --
> >> An HTML attachment was scrubbed...
> >> URL:
> >> 
> >>
> >> --
> >>
> >> Message: 2
> >> Date: Fri, 10 Dec 2010 09:17:44 -0600
> >> From: Tam Nhu 
> >> To: ccie_voice@onlinestudylist.com
> >> Subject: Re: [OSL | CCIE_Voice] CFUR display For and By
> >> Message-ID:
> >> 
> >> Content-Type: text/plain; charset="iso-8859-1"
> >>
> >> Hi study2b,
> >>
> >> You can look at this thread and make it works.
> >>
> >> http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg19418.html
> >>
> >>
>

Re: [OSL | CCIE_Voice] QoS Issue Lab 4 & 5 Vol 2

2010-10-20 Thread Pulos, Greg
Just to be clear

Your question was

"So you mean i can specify any number other than 4 but keep sure that both of 
them configured with the same vlaue, is that correct."

The responder stated "Yep :)"

To which, I corrected"No, both values do not have to be the same." And 
if you change them to be the same, say from 4 + 4, to 5 + 5, your bandwidth 
will not  be 50%, therefore the answer to "So you mean i can specify any number 
other than 4 but keep sure that both of them configured with the same vlaue, is 
that correct.", is no, not necessarily; it depends on the amount of bandwidth 
you want serviced.

Please see the following link for more info on the 'mls qos srr-queue input 
bandwidth' command.

http://www.cisco.com/en/US/docs/switches/lan/catalyst3750/software/release/12.2_18_se/command/reference/cli1.html#wp2128913


greg


From: Amr Sherif [mailto:miroale...@hotmail.com]
Sent: Wednesday, October 20, 2010 12:14 PM
To: Pulos, Greg; findko...@gmail.com
Cc: ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] QoS Issue Lab 4 & 5 Vol 2


But the question was specifying that both the two queues has to be equally 
shared, right .So that's why in this situation it's used for both values

Best regards,

Amr Sherif
Senior Network Voice Engineer
CCNA,CCNP,CCVP and CCIE Voice Written (Certified)
CCIE Voice Lab (In Progress)



From: gpu...@doc.gov
To: findko...@gmail.com; miroale...@hotmail.com
CC: ccie_voice@onlinestudylist.com
Date: Wed, 20 Oct 2010 11:55:16 -0400
Subject: RE: [OSL | CCIE_Voice] QoS Issue Lab 4 & 5 Vol 2
I disagree...

These 2 values DO NOT have to be the same.

Thank you.

greg


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Miron Kobelski
Sent: Wednesday, October 20, 2010 11:51 AM
To: Amr Sherif
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] QoS Issue Lab 4 & 5 Vol 2

yep :)
--
Sent from my mobile device.
On 20 Oct 2010 17:48, "Amr Sherif" 
mailto:miroale...@hotmail.com>> wrote:

So you mean i can specify any number other than 4 but keep sure that both of 
them configured with the same vlaue,is that correct.



Best regards,


Amr Sherif
Senior Network Voice Engineer
CCNA,CCNP,CCVP and CCIE Voice Written (C...
Date: Wed, 20 Oct 2010 17:27:20 +0200
Subject: Re: [OSL | CCIE_Voice] QoS Issue Lab 4 & 5 Vol 2
From: findko...@gmail.com<mailto:findko...@gmail.com>
To: miroale...@hotmail.com<mailto:miroale...@hotmail.com>
CC: ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com>


those two 4s are relative to each other. so each of 2 queues will get 4/(4+4) 
50% after servicing ...
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Re: [OSL | CCIE_Voice] QoS Issue Lab 4 & 5 Vol 2

2010-10-20 Thread Pulos, Greg
I disagree…

These 2 values DO NOT have to be the same.

Thank you.

greg


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Miron Kobelski
Sent: Wednesday, October 20, 2010 11:51 AM
To: Amr Sherif
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] QoS Issue Lab 4 & 5 Vol 2


yep :)

--
Sent from my mobile device.
On 20 Oct 2010 17:48, "Amr Sherif" 
mailto:miroale...@hotmail.com>> wrote:

So you mean i can specify any number other than 4 but keep sure that both of 
them configured with the same vlaue,is that correct.



Best regards,


Amr Sherif
Senior Network Voice Engineer
CCNA,CCNP,CCVP and CCIE Voice Written (C...
Date: Wed, 20 Oct 2010 17:27:20 +0200
Subject: Re: [OSL | CCIE_Voice] QoS Issue Lab 4 & 5 Vol 2
From: findko...@gmail.com
To: miroale...@hotmail.com
CC: ccie_voice@onlinestudylist.com


those two 4s are relative to each other. so each of 2 queues will get 4/(4+4) 
50% after servicing ...
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Re: [OSL | CCIE_Voice] QoS Issue Lab 4 & 5 Vol 2

2010-10-20 Thread Pulos, Greg
Please see the following link for the info you need on the 'mls qos srr-queue 
input bandwidth' command.

http://www.cisco.com/en/US/docs/switches/lan/catalyst3750/software/release/12.2_18_se/command/reference/cli1.html#wp2128913

greg


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Amr Sherif
Sent: Wednesday, October 20, 2010 11:47 AM
To: findko...@gmail.com
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] QoS Issue Lab 4 & 5 Vol 2


So you mean i can specify any number other than 4 but keep sure that both of 
them configured with the same vlaue,is that correct.


Best regards,

Amr Sherif
Senior Network Voice Engineer
CCNA,CCNP,CCVP and CCIE Voice Written (Certified)
CCIE Voice Lab (In Progress)



Date: Wed, 20 Oct 2010 17:27:20 +0200
Subject: Re: [OSL | CCIE_Voice] QoS Issue Lab 4 & 5 Vol 2
From: findko...@gmail.com
To: miroale...@hotmail.com
CC: ccie_voice@onlinestudylist.com

those two 4s are relative to each other. so each of 2 queues will get 4/(4+4) 
50% after servicing the pq
regards
--
Sent from my mobile device.
On 20 Oct 2010 17:18, "Amr Sherif" 
mailto:miroale...@hotmail.com>> wrote:

Hello Everyone,

Well in Vol 2 lab 4 & 5 for qos section PG always use "mls qos srr-queue input 
bandwidth 4 4" ,why it's specifying 4 ? .The question was " Ensure that input 
queue #1 is configured as the priority queue with 20% and requesting that the 
remaining bandwidth should be serviced by SRR equally".

So any ideas why he specify 4 in this answer?



Best regards,


Amr Sherif
Senior Network Voice Engineer
CCNA,CCNP,CCVP and CCIE Voice Written (Certified)
CCIE Voice Lab (In Progress)



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Re: [OSL | CCIE_Voice] IOS Image

2010-09-23 Thread Pulos, Greg
You can use the cisco Feature Navigator link below to determine what features 
your IOS supports:

http://tools.cisco.com/ITDIT/CFN/jsp/index.jsp

Thank you.

greg


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Santi Cuni
Sent: Thursday, September 23, 2010 9:10 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] IOS Image


Hello Everyone

Could you tell me if these two IOS version has the Gatekeeper capabilities?
c2800nm-adventerprisek9_ivs_li-mz.124-15.T8.bin
c2800nm-adventerprisek9_ivs-mz.124-20.T1.bin

Thanks Regards

Santi Cuni




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Re: [OSL | CCIE_Voice] Unity Question

2010-07-28 Thread Pulos, Greg
I think you answered your own question in a way.

You say you want to setup a 'test' environment, and tie it to your 'production' 
system. That alone is a bad practice.

Don't learn the hard way how bad it can get when you integrate 'test' systems 
into your production environment.

Just my 2 cents!

greg


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mav
Sent: Tuesday, July 27, 2010 8:49 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Unity Question

Hello,

I'm building a testing environment that will run in parallel to an existing 
cluster that has CUCM 6.x and Unified Unity 4. The test environment is 
configured with different IP addresses then the one in production.  My concern 
is with Unity , extending the schema in Active Directory, running the 
permission wizard and integrating the new unity with the exchange in use by the 
old Unity.  Is it going to effect the Unity in production?

Thanks in advance
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Re: [OSL | CCIE_Voice] NTP

2010-07-09 Thread Pulos, Greg
To identify the most accurate stratum, think of the stratum as the number of 
'hops' from the master time source.

ie: your publisher server is sync'd to your internal router; your internal 
router is sync'd to your Internet router; your Internet router is sync'd to a 
master time source on the net; that's three stratum 'hops' right there.

A lowest 'root' master time source would use stratum 0.

ie: Master = stratum 0 >>> internet router = stratum 1 >>> internal router = 
stratum 2 >>> publisher = stratum 3

At this point, any device syncing to the publisher would use stratum 4. (or 
higher)
Any device syncing to the internal router would use stratum 3 (or 
higher)etc.

(of course, your internet router may or may not sync to a stratum 0 master; it 
could be a higher stratum which in that case you'd increment your stratum 
numbers down the line appropriately)

greg


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of 
bkvalent...@gmail.com
Sent: Friday, July 09, 2010 7:07 AM
To: cisco voip; Mark Holloway
Cc: OSL osl
Subject: Re: [OSL | CCIE_Voice] NTP

A router that has "ntp server" configured also acts as a server.  You don't 
need ntp master to act as a server.

Brian

- Reply message -
From: "cisco voip" 
Date: Thu, Jul 8, 2010 10:42 pm
Subject: [OSL | CCIE_Voice] NTP
To: "Mark Holloway" 
Cc: "OSL osl" 


But "ntp master" command is necessary for the router to be able to be ntp 
server for anyone else. i agree about stratum chain, but without ntp master 
command publisher should not be able to get its clock.

you can check show ntp associations on the router, after ntp master command 
only it will show you the local interface.

now if you want to use the stratum to be 1 more from whatever the stratum is on 
actual source, you do not need stratum command, for anything else you need that 
commnd

On Fri, Jul 9, 2010 at 4:42 AM, Mark Holloway  wrote:

> Cool, thanks Graham and Randall.
>
> On Jul 8, 2010, at 4:11 PM, Graham Hopkins wrote:
>
> > Default stratum is 8 so a simple ntp master will work
> >
> > Graham
> >
> > On 8 Jul 2010, at 23:59, Mark Holloway  wrote:
> >
> >> Yikes, I meant "ntp master stratum X" not "ntp server stratum X"
> >>
> >> On Jul 8, 2010, at 3:57 PM, Mark Holloway wrote:
> >>
> >>> If a router (for example, HQ) is configured with the "ntp server
> x.x.x.x" command to sync time from another source, but I want another 
> device (such as PUB) to get its time from the HQ router, do I also 
> need to configure the HQ router with "ntp server stratum X" or can UCM 
> simply get the time sync from HQ without the stratum command?
> >>> ___
> >>> 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
>
> ___
> For more information regarding industry leading CCIE Lab training, 
> please visit www.ipexpert.com
>



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Re: [OSL | CCIE_Voice] External Phone # mask in Unity

2010-05-27 Thread Pulos, Greg
I use external voicemail masks on all 72 voicemail ports we have. Works fine.

greg


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of cisco voip
Sent: Wednesday, May 26, 2010 11:36 PM
To: Salman Shaikh
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] External Phone # mask in Unity

You can use CTI Route point


On Thu, May 27, 2010 at 8:22 AM, Salman Shaikh  
wrote:


Is there any way to assign " external phone number mask" to voicemail
pilot number (in CUE and UC both)? on display when you call from PSTN 
VM Pilot number 
I appreciate your advise.
 
Thanks
sal


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Re: [OSL | CCIE_Voice] Basic VMWare Server Lab Question

2010-05-20 Thread Pulos, Greg
To get the software, contact your Cisco Rep and request.

There used to be a site which is the Cisco Market Place to order your NFR (not 
for resale) Kit of the entire UC 7.01 platform; but this seems to show it is 
not available currently. (may require cco login)
https://cisco.mediuscorp.com/market/networkers/listSubCat.se.work?TRGT=10&/nxt/rcrs/=1180

Thank you.

greg

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of 
amir.safa...@memorialhealthsystem.com
Sent: Wednesday, May 19, 2010 4:12 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Basic VMWare Server Lab Question


I am looking for the best approach to building a VMWare server for running
CUCM, UCCX and UC.  I have an IBM 3650 M2 with lots of CPU, Memory and
drive space.  \

What is the best OS and VMWare application to use as my foundation?  Once
that is built, what is the best way to get the installation media for 7.x
versions of CUCM, UCCX and UC testing in our lab?  We are a large
enterprise customer and we have software subscriptions from Cisco.  Often
those subscriptions only provide an "upgrade" path and not the original
installation media.

We're running 4.x  in production on all the above mentioned platforms and
certainly don't want to perform original installations of 4.x and then use
our software subscription to upgrade to 7.x

Amir

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Re: [OSL | CCIE_Voice] Quick question regarding dial peer zero

2010-05-02 Thread Pulos, Greg
nope.

greg


From: ccie_voice-boun...@onlinestudylist.com 
[ccie_voice-boun...@onlinestudylist.com] On Behalf Of Matthew Berry 
[ciscovoiceg...@gmail.com]
Sent: Sunday, May 02, 2010 7:25 AM
To: CCIE Voice OSL
Subject: [OSL | CCIE_Voice] Quick question regarding dial peer zero

Can you completely shutdown dial peer zero just like you can shutdown
manually created voice dial peers? Such a command might be helpful
during a lab to ensure you don't overlook that pesky feature.

Matthew Berry

**Sent from my iPhone**
Skype/Twitter: ciscovoiceguru
Google Voice: +1 612 424 5044
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Re: [OSL | CCIE_Voice] UCCX Scripting

2010-04-22 Thread Pulos, Greg
It's possiblebut you'll have to change the way in which you provide the 1 
for customer support and the option to dial-by-extension since your extensions 
begin with 1 as well. Just a bit like building an algorithm.think it out 
clearly first. 

Please see the following link for the CCX scripting information you'll need.

http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_7_0/user/guide/uccx70edgs.pdf

Thank you.

greg


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cristobal Priego
Sent: Wednesday, April 21, 2010 7:45 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] UCCX Scripting

Hello All, 

I have 2 questions for you

I need to create a script for an AA, the main prompt will allow the user to 
dial the extension if the know it or wait to listen to the menu options

for the menu I need to use a menu step no big deal, and i'm thinking to create 
a variable that will store the digits the customer pressed and then with a call 
transfer try to reach that extension. however my problem or question comes how 
do i merge this 2 options together, allow the caller to dial the extension 
while at the same time know when he presses 1 for customer service and is not 
trying to reach ext 1002 ?


also is there a way to prompt the option to the caller if he is in the queue to 
leave a vm, however instead of leaving a vm record the prompt, put it on an 
e-mail and then sent it to a queue and then have the e-mail to be delivered to 
the next available agent ?
I don't think it's possible but the sales engineer sold this option and i have 
to implement it and i just don't think is possible

maybe you've done it before or something similar

thanks

Cristobal

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Re: [OSL | CCIE_Voice] CIPC strange behavior !

2010-03-15 Thread Pulos, Greg
The benefit of SRNDs.
1) you WONT pass the CCIE without knowing the information contained within them
2) you WONT know Cisco best practices and recommendations of cisco technology 
design
3) you WONT know A LOT of design information that is very valuable in the real 
world troubleshooting and design

The SRNDs are the 'solution reference network design' guides for all things 
Cisco Design.

Please see the following link for more info on CUCM SRND 7.x.

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/uc7_0.html

See the following link for more Design Zone and SRND info and links:

http://www.cisco.com/en/US/netsol/ns818/networking_solutions_program_home.html


Thank you.

greg

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of jonn cozak
Sent: Monday, March 15, 2010 12:54 AM
To: Brian Mulgrew; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CIPC strange behavior !

Dear Sir, 

I have found it !! 

Now if possible can you kindly also let me know what is SRND all about and what 
benefit i may get from these docs ?

The question is for you and for anyone who wishes to answer.

Thanks again for the support 





From: Brian Mulgrew 
To: cisco.jon...@yahoo.com; ccie_voice@onlinestudylist.com
Sent: Sun, March 14, 2010 1:12:53 AM
Subject: RE: [OSL | CCIE_Voice] CIPC strange behavior !


Hi - think it is touched on briefly on the SRND under Network Infrastructure / 
Network Svcs / TFTP section:

When a device requests a configuration file from the TFTP server, the TFTP 
server searches for the configuration file in its internal caches, the disk, 
and then alternate Cisco file servers (if specified). If the TFTP server finds 
the configuration file, it sends it to the device. If the configuration file 
provides Unified CM names, the device resolves the name by using DNS and opens 
a connection to the Unified CM. If the device does not receive an IP address or 
name, it uses the TFTP server name or IP address to attempt a registration 
connection. If the TFTP server cannot find the configuration file, it sends a 
"file not found" message to the device.


cheers
b


Date: Sat, 13 Mar 2010 08:06:47 -0800
From: cisco.jon...@yahoo.com
Subject: Re: [OSL | CCIE_Voice] CIPC strange behavior !
To: btmulg...@hotmail.com; ccie_voice@onlinestudylist.com


Dear Brian, thats exactly what is confusing me. Why this happens ? i mean i 
havent read this on any cisco technology doc no matter i much i search. Why 
this behaviour is not documented ? is it a random behavior ? 

i tried the same thing with IP Blue with 7960 selected but it didnt register 
!!, it wasnt able to resolve the hostname and hence registeration wasnt 
successful. Perhaps this is the behaviour that is defined in various study 
guides, is it a special case with CIPC ?





From: Brian Mulgrew 
To: cisco.jon...@yahoo.com; ccie_voice@onlinestudylist.com
Sent: Sat, March 13, 2010 5:46:31 PM
Subject: RE: [OSL | CCIE_Voice] CIPC strange behavior !

Hi - If I remember correctly if the IP Phone / CIPC cannot find the hostname of 
the CUCM in the cnf.xml it will register with  ccm service running on the tftp 
server - so if your tftp is set as IP this could be the reason why it still 
registers.

thks
b




Date: Fri, 12 Mar 2010 23:45:10 -0800
From: cisco.jon...@yahoo.com
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] CIPC strange behavior !


Hi all. I am using CUCM 6. CIPC is installed on my pc. I am using single 
callmanager server that is also my tftp server for endpoints. Now while doing 
my studies i wanted to check how DNS might cause issues. In System->Server, i 
am using hostname instead of IP. Now what happens is that, CIPC after getting 
the .cnf.xml file, registers with tftp server successfully (which is also my 
callmanager server). Now what i read in student guide was, ip phone should not 
be able to register if its not able to resolve the hostname through DNS. (i am 
not using any DNS server and nor the entry for the hostname is present in my pc 
host file).

Can someone tell me why is this the case ? i have searched alot but i couldnt 
find any thing stating that ip phone will fall back to tftp server in case 
primary callmanager fails !!

Any input on this pls ?





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Re: [OSL | CCIE_Voice] Barge

2010-03-03 Thread Pulos, Greg
The Cisco SRNDs for unified communications will tell you everything you need to 
know about how, where, why conferencing/transcoding/mtp resources are required.

Please see the following link for more info on SRNDs for unified communications.

http://www.cisco.com/iam/unified/ipt1/Using_SRND_Documents.htm

greg


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Omotayo
Sent: Wednesday, March 03, 2010 7:19 AM
To: Otto Sanchez
Cc: OSL Group
Subject: Re: [OSL | CCIE_Voice] Barge

Hello Otto,
 
I want to make few clarification.
 
If i have a centralized call processing system 
 
Hq phones need to barge into br1 phone. where do i need to configure the 
conferencing resouces ( hq router or br1 router) and which conferencing 
resources is used
 
 
Also for transcoder, if br1 phones needs to access a server resources that is 
using g711 and br1 phones is coming with g729 cuz its going across the wan. 
where do i configure the transcoding resources- the hq router or br1 router
 
thanks


On Sun, Feb 21, 2010 at 8:10 PM, Otto Sanchez  wrote:


If you are barging br1phn2 the br1 conference resources are used, not 
the hq ones. Also make sure that br1phn2 has privacy off by the use of the 
privacy button or has privacy off in the device configuration,

Finally, make sure that when cbarging from the shared line button, your 
hq phone config has the cbarge setting configured,


On Sat, Feb 20, 2010 at 4:24 PM, Omotayo  wrote:


Hello,
working on Volume 1 lab 8
IOS conference has been cnfigured on hq router
when i tried with question 8.1 i used hq phones as bri phone of 
the question and vice versa
on pressing the button on br1 phone 2 when in In Use Remote, i 
still see the barge softkey and it gives the message No Conference Bridge
CBarge was enabled on the service parameter
 
Below is a proof the conference bridge on the hq router is 
working
 
HQ-RTR#sh sccp connections 
sess_idconn_idstype mode codec   ripaddr 
rport sport
33557433   33554446   conf  sendrecv g729b   192.168.3.16
16386 16524
33557433   3355   conf  sendrecv g729b   192.168.3.12
18674 17838
33557433   33554442   conf  sendrecv g711u   192.168.3.18
28554 18950

Any ideas on what the issue is

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-- 
Regards,

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



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Re: [OSL | CCIE_Voice] Volume 1 Lab 5 Proctor Guide needs to bereworked

2010-02-22 Thread Pulos, Greg
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 (R&S, 
Voice & Security), CCNP, CCVP, CCSP and CCIE (R&S, 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"  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  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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Re: [OSL | CCIE_Voice] CCIE Voice Vol 1 Lab 5c - Transcoder

2010-02-22 Thread Pulos, Greg
Please see the following link for NM-HDV Transcoder configuration information 
and examples:

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configuration_example09186a0080334294.shtml

Thank you.

greg

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Otto Sanchez
Sent: Monday, February 22, 2010 7:49 AM
To: CCIETalk.com
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE Voice Vol 1 Lab 5c - Transcoder

Hi,

Would you please send us your 3725 configuration?,

Thanks,


On Sun, Feb 21, 2010 at 10:27 PM, CCIETalk.com  wrote:


Well it appears that NM-HDV doesn't support transcoding either :\


On Sun, Feb 21, 2010 at 9:55 PM, CCIETalk.com  
wrote:


I added NM-HDV with pvdm-12 but I still see the same issue.


On Sun, Feb 14, 2010 at 3:51 PM, Otto Sanchez 
 wrote:


Hi, 

I don't think the AIM-VOICE-30 supports transcoding or 
conferencing but voice termination services only, so in this case you may need 
to install a NM in your 3725 to move on,

Thanks,


On Sun, Feb 14, 2010 at 12:07 PM, CCIETalk.com 
 wrote:


I was working through lab 5c and came across 
the task where I had to configure a transcoder. I am using a 3725 with AIM-30

- one voice pri with 3 channels
- one data T1

I try to create the dspfarm profile and get 
this erro

HQ-RTR(config-dspfarm-profile)#codec ?
% Unrecognized command

Any idea?

-- 
www.ccietalk.com


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-- 
Regards,

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





-- 
www.ccietalk.com





-- 
www.ccietalk.com





-- 
Regards,

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

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Re: [OSL | CCIE_Voice] UCCX Scripting

2010-02-18 Thread Pulos, Greg
Knowing CCX scripting is vital if you acutally want to be able to provide 
solutions to customers and to troubleshoot CRS application problems.

Please see the link below for more information on CCX 7.01 scripting.

http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_7_0/user/guide/uccx70edgs.pdf

Thank you.

greg


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roger Henderson
Sent: Thursday, February 18, 2010 6:53 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] UCCX Scripting

Hey Everyone,

What is the best resource(s) to learn the various UCCX scripting methods needed 
for the lab? Does anyone have any good resources online? How complicated is it 
likely to get for the lab and how much time should we dedicate to this?


Thanks,

Roger

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Re: [OSL | CCIE_Voice] Unity Connection with Exchange using IMAP

2010-02-12 Thread Pulos, Greg
You can find the Unity Connection integration and IMAP setup in the following 
Cisco document.

'System Administration Guide for Cisco Unity Connection 7.x'

http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/administration/guide/7xcucsag175.html

greg


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Sivakumar 
Mahalingam
Sent: Friday, February 12, 2010 1:36 PM
To: OSL Group
Subject: [OSL | CCIE_Voice] Unity Connection with Exchange using IMAP

Hi all,
 
Can anyone help me in briefing me the step by step process to integrate unity 
connection with exchange using IMAP. And configure Outlook client with IMAP 
protocol.
 
I need this for a production environment and any help is highly appericiated.
 
Thanks in advance.
 
Simah.
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Re: [OSL | CCIE_Voice] Disconnect BR1-PSTN: 0x80A9 - Temporary failure

2010-02-03 Thread Pulos, Greg
Ooops...i see you have some q921 debug running already

greg


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Stephen 
Greszczyszyn
Sent: Wednesday, February 03, 2010 8:16 AM
To: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Disconnect BR1-PSTN: 0x80A9 - Temporary failure

I also get some weird stuff around the time of call failure when I
enable MGCP debugging:

debug mgcp errors
debug mgcp packet

It seems as though my br1 MGCP gateway tells CUCM to drop the call and
also sends the disconnect to the PSTN:

BR1:
Feb  3 13:09:00.557: ISDN Se0/1/0:15 Q921: User TX -> INFO sapi=0
tei=0, ns=14 nr=15
Feb  3 13:09:00.557: ISDN Se0/1/0:15 Q931: DISCONNECT pd = 8  callref = 0x0003
Cause i = 0x80A9 - Temporary failure

PSTN:
*Feb  3 13:15:23.583: ISDN Se0/3/0:15 Q921: Net RX <- INFO sapi=0
tei=0, ns=14 nr=15
*Feb  3 13:15:23.583: ISDN Se0/3/0:15 Q931: DISCONNECT pd = 8  callref = 0x0003
Cause i = 0x80A9 - Temporary failure

(Previous call traces on BR1 - MGCP debug)

Feb  3 12:52:45.713: Following traceback is for INFO ONLY. -Traceback=
0x40F3126Cz 0x40F5BA6Cz 0x40F4296Cz 0x40F3ABC4z 0x40F333D4z
0x40EEED44z 0x40EF43F0z 0x435D69D8z 0x435D69BCz
Feb  3 12:52:45.717: MGCP Packet sent to 10.10.210.11:2427--->
DLCX 205358175 S0/SU1/DS1-0/1...@br1.proctorlabs.com MGCP 0.1
C: D2a512c900F50002
I: 3
P: PS=5000, OS=80, PR=4978, OR=796480, PL=22, JI=7, LA=0
E: 502
<---

Feb  3 12:52:45.725: MGCP Packet received from 10.10.210.11:2427--->
200 205358175
<---

Feb  3 12:52:45.725:
//-1//MGCP/mgcp_mp_get_not_entity(830):[lvl=2]Invalid
parameter (pkt 0x4A428FE0 pkt->mgcp_parm_lines 0x)
Feb  3 12:52:45.733: ISDN Se0/1/0:15 Q931: TX -> DISCONNECT pd = 8
callref = 0x0002
Cause i = 0x80A9 - Temporary failure
Feb  3 12:52:45.745: ISDN Se0/1/0:15 Q931: RX <- RELEASE pd = 8
callref = 0x8002
Feb  3 12:52:45.757: ISDN Se0/1/0:15 Q931: TX -> RELEASE_COMP pd = 8
callref = 0x0002
Feb  3 12:53:04.977: MGCP Packet sent to 10.10.210.11:2427--->
NTFY 205358176 *...@br1.proctorlabs.com MGCP 0.1
X: 0
O:
<---

In the CUCM traces, it isn't clear to me why the call is being dumped:

02/03/2010 12:52:45.690 CCM|MGCPHandler received msg from: 10.10.110.2
DLCX 205358175 S0/SU1/DS1-0/1...@br1.proctorlabs.com MGCP 0.1
C: D2a512c900F50002
I: 3
P: PS=5000, OS=80, PR=4978, OR=796480, PL=22, JI=7, LA=0
E: 502
|
02/03/2010 12:52:45.690
CCM|
02/03/2010 12:52:45.691 CCM|MGCPHandler send msg SUCCESSFULLY to: 10.10.110.2
200 205358175
|
02/03/2010 12:52:45.692 CCM|ConnectionManager -
wait_AuDisconnectRequest(44372680,44372681),disconnectType(1),
IFHandling(0,0)|
02/03/2010 12:52:45.692 CCM|ConnectionManager -
storeMediaInfo(44372680): EXISTING ENTRY DISCOVERED,
size=2|
02/03/2010 12:52:45.692 CCM|ConnectionManager -
storeMediaInfo(44372681): EXISTING ENTRY DISCOVERED,
size=2|
02/03/2010 12:52:45.692 CCM|MediaCoordinator -
wait_AuDisconnectRequest,CI(44372680,44372681),IFCreated(1,1)|
02/03/2010 12:52:45.692 CCM|MediaCoordinator -
wait_AuDisconnectRequest - sending disconnect to
MediaManager(48)|

...

02/03/2010 12:52:45.694 CCM|MGCPpn9d - Dump portInfo table:
 portInfo[00] endpoint=S0/SU1/DS1-0/1...@br1.proctorlabs.com, ci=44372681
 portInfo[01] endpoint=S0/SU1/DS1-0/2...@br1.proctorlabs.com, ci=0
 portInfo[02] endpoint=S0/SU1/DS1-0/3...@br1.proctorlabs.com, ci=0

...

02/03/2010 12:52:45.698 CCM|Out Message -- PriDisconnectMsg --
Protocol= 
PriEuroProtocol|
02/03/2010 12:52:45.698 CCM|Ie - Q931CauseIe IEData= 08 02 80 A9
|


On Wed, Feb 3, 2010 at 12:31 PM, Stephen Greszczyszyn
 wrote:
> I have been testing some calls as part of Vol1 Lab 5, and for some
> reason my calls from BR1 (MGCP) to PSTN drop suddenly after about 1:30
> to 2 minutes.  I don't get this when I call from HQ (H.323) to PSTN,
> so this would rule out the PSTN router?  I also used the B-Channel
> Maintenance Status Service Parameters to busy out all but the 4 E1
> channels that I am using.
>
> Could this be a timer issue with the clocking between PSTN router and
> MGCP gateway?
>
> I get this in the ISDN traces:
>
> BR1:
> Feb  3 11:53:33.851: ISDN Se0/1/0:15 Q931: TX -> DISCONNECT pd = 8
> callref = 0x0007
>        Cause i = 0x80A9 - Temporary failure
>
> PSTN:
> *Feb  3 11:59:57.011: ISDN Se0/3/0:15 Q931: RX <- DISCONNECT pd = 8
> callref = 0x0007
>        Cause i = 0x80A9 - Temporary failure
> *Feb  3 11:59:57.015: %ISDN-6-DISCONNECT: Interface Serial0/3/0:0
> disconnected from +16178631002 , call lasted 72 seconds
>
> I can't really see why this is happening?  I did some searching on
> Google, where people suggested to test the DSPs.  The DSPs looked OK
> during the call:
>
> BR1:
> br1#sh voice dsp group all
> DSP groups on slot 0:
> dsp 1:
>  State: UP, firmware: 24.3.2
>  Max signal/voice channel: 16/16
>  Max credits: 240
>  num_of_sig_chnls_allocated: 8
>  Transcoding channels allocated: 0
>  Group:

Re: [OSL | CCIE_Voice] Disconnect BR1-PSTN: 0x80A9 - Temporary failure

2010-02-03 Thread Pulos, Greg
Cause code 0x80A9 identifies

1) 80 = router
*** identifies the point where the 'disconnect' cause code appears (vs from the 
telco switch, etc)

2) A9 = temporary failure
*** identifies the problem occurring due to a network failure/malfunction

You may want to debug q921 as well to verify layer 2 events for the isdn.
You may want to verify your mgcp config and/or see if mgcp is 'bouncing' to the 
ccm which would cause the mgcp DLCX (delete connection) to occur.
(by bouncing I mean mgcp gateways continually registering/reregistering; 
usually due to a ccm/mgcp config problem)

Thank you.

greg


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Stephen 
Greszczyszyn
Sent: Wednesday, February 03, 2010 8:16 AM
To: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Disconnect BR1-PSTN: 0x80A9 - Temporary failure

I also get some weird stuff around the time of call failure when I
enable MGCP debugging:

debug mgcp errors
debug mgcp packet

It seems as though my br1 MGCP gateway tells CUCM to drop the call and
also sends the disconnect to the PSTN:

BR1:
Feb  3 13:09:00.557: ISDN Se0/1/0:15 Q921: User TX -> INFO sapi=0
tei=0, ns=14 nr=15
Feb  3 13:09:00.557: ISDN Se0/1/0:15 Q931: DISCONNECT pd = 8  callref = 0x0003
Cause i = 0x80A9 - Temporary failure

PSTN:
*Feb  3 13:15:23.583: ISDN Se0/3/0:15 Q921: Net RX <- INFO sapi=0
tei=0, ns=14 nr=15
*Feb  3 13:15:23.583: ISDN Se0/3/0:15 Q931: DISCONNECT pd = 8  callref = 0x0003
Cause i = 0x80A9 - Temporary failure

(Previous call traces on BR1 - MGCP debug)

Feb  3 12:52:45.713: Following traceback is for INFO ONLY. -Traceback=
0x40F3126Cz 0x40F5BA6Cz 0x40F4296Cz 0x40F3ABC4z 0x40F333D4z
0x40EEED44z 0x40EF43F0z 0x435D69D8z 0x435D69BCz
Feb  3 12:52:45.717: MGCP Packet sent to 10.10.210.11:2427--->
DLCX 205358175 S0/SU1/DS1-0/1...@br1.proctorlabs.com MGCP 0.1
C: D2a512c900F50002
I: 3
P: PS=5000, OS=80, PR=4978, OR=796480, PL=22, JI=7, LA=0
E: 502
<---

Feb  3 12:52:45.725: MGCP Packet received from 10.10.210.11:2427--->
200 205358175
<---

Feb  3 12:52:45.725:
//-1//MGCP/mgcp_mp_get_not_entity(830):[lvl=2]Invalid
parameter (pkt 0x4A428FE0 pkt->mgcp_parm_lines 0x)
Feb  3 12:52:45.733: ISDN Se0/1/0:15 Q931: TX -> DISCONNECT pd = 8
callref = 0x0002
Cause i = 0x80A9 - Temporary failure
Feb  3 12:52:45.745: ISDN Se0/1/0:15 Q931: RX <- RELEASE pd = 8
callref = 0x8002
Feb  3 12:52:45.757: ISDN Se0/1/0:15 Q931: TX -> RELEASE_COMP pd = 8
callref = 0x0002
Feb  3 12:53:04.977: MGCP Packet sent to 10.10.210.11:2427--->
NTFY 205358176 *...@br1.proctorlabs.com MGCP 0.1
X: 0
O:
<---

In the CUCM traces, it isn't clear to me why the call is being dumped:

02/03/2010 12:52:45.690 CCM|MGCPHandler received msg from: 10.10.110.2
DLCX 205358175 S0/SU1/DS1-0/1...@br1.proctorlabs.com MGCP 0.1
C: D2a512c900F50002
I: 3
P: PS=5000, OS=80, PR=4978, OR=796480, PL=22, JI=7, LA=0
E: 502
|
02/03/2010 12:52:45.690
CCM|
02/03/2010 12:52:45.691 CCM|MGCPHandler send msg SUCCESSFULLY to: 10.10.110.2
200 205358175
|
02/03/2010 12:52:45.692 CCM|ConnectionManager -
wait_AuDisconnectRequest(44372680,44372681),disconnectType(1),
IFHandling(0,0)|
02/03/2010 12:52:45.692 CCM|ConnectionManager -
storeMediaInfo(44372680): EXISTING ENTRY DISCOVERED,
size=2|
02/03/2010 12:52:45.692 CCM|ConnectionManager -
storeMediaInfo(44372681): EXISTING ENTRY DISCOVERED,
size=2|
02/03/2010 12:52:45.692 CCM|MediaCoordinator -
wait_AuDisconnectRequest,CI(44372680,44372681),IFCreated(1,1)|
02/03/2010 12:52:45.692 CCM|MediaCoordinator -
wait_AuDisconnectRequest - sending disconnect to
MediaManager(48)|

...

02/03/2010 12:52:45.694 CCM|MGCPpn9d - Dump portInfo table:
 portInfo[00] endpoint=S0/SU1/DS1-0/1...@br1.proctorlabs.com, ci=44372681
 portInfo[01] endpoint=S0/SU1/DS1-0/2...@br1.proctorlabs.com, ci=0
 portInfo[02] endpoint=S0/SU1/DS1-0/3...@br1.proctorlabs.com, ci=0

...

02/03/2010 12:52:45.698 CCM|Out Message -- PriDisconnectMsg --
Protocol= 
PriEuroProtocol|
02/03/2010 12:52:45.698 CCM|Ie - Q931CauseIe IEData= 08 02 80 A9
|


On Wed, Feb 3, 2010 at 12:31 PM, Stephen Greszczyszyn
 wrote:
> I have been testing some calls as part of Vol1 Lab 5, and for some
> reason my calls from BR1 (MGCP) to PSTN drop suddenly after about 1:30
> to 2 minutes.  I don't get this when I call from HQ (H.323) to PSTN,
> so this would rule out the PSTN router?  I also used the B-Channel
> Maintenance Status Service Parameters to busy out all but the 4 E1
> channels that I am using.
>
> Could this be a timer issue with the clocking between PSTN router and
> MGCP gateway?
>
> I get this in the ISDN traces:
>
> BR1:
> Feb  3 11:53:33.851: ISDN Se0/1/0:15 Q931: TX -> DISCONNECT pd = 8
> callref = 0x0007
>        Cause i = 0x80A9 - Temporary failure
>
> PSTN:
> *Feb  3 11:59:57.011: ISDN Se0/3/0:15 Q931: RX <- DISCONNECT pd = 8
> callref = 0x0007
>        Cause i = 0x80A9 - Temp

Re: [OSL | CCIE_Voice] Recovery of CallManager subscriber 4.2

2010-01-19 Thread Pulos, Greg
Please see the following link for installing/replacing a 4.2 CCM Subscriber.

NOTE: select the 'Server Replacement' option when prompted during the install 
procedure.

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/install/4_2/cm421ins.html

Basically you rebuild the server from scratch; Hardware Detect/OS, CCM, etc. 
then it gets the DB from the pub.

Thank you.

greg

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of asif raza
Sent: Tuesday, January 19, 2010 3:07 AM
To: CCIE-Voice
Subject: [OSL | CCIE_Voice] Recovery of CallManager subscriber 4.2

Hi Friends
 
CallManager Subscriber 4.2 has been crashed, due to some hardware failure etc, 
now it is giving me a blue screen (dumb error). Primary server is working fine. 
Any idea how to recover it. 
because cisco Disaster recovery utitlity can only be utilized from version 5. 
of Call manager.

Thanks and Kind Regards 

 
Asif Raza
 





Hotmail: Trusted email with powerful SPAM protection. Sign up now. 
 
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Re: [OSL | CCIE_Voice] UCCX clock set

2010-01-14 Thread Pulos, Greg
Use windows time or xntp provided by Cisco.

Please see the link below for more info on time servers for UCXX.

http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_5_0/maintenance/admin/crs501ag.pdf

Page 4-12; Modifying NTP Configuration

Thank you.

greg


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Omotayo
Sent: Thursday, January 14, 2010 10:03 AM
To: OSL Group
Subject: [OSL | CCIE_Voice] UCCX clock set

Hello,
 
Am working on Lab 2
 
how do i synchronize the uccx time to that of the callmanager or the 
ntp(10.10.100.2)
 
thanks for the anticipated help
 
rgd
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Re: [OSL | CCIE_Voice] Vol 2 Lab3 UC 7.1

2010-01-14 Thread Pulos, Greg
This is stating that the UC should not be able to 'loop' calls to itself.

In other words, a call to the Unity system does not get routed by unity such 
that unity has to receive/answer the call again.

ie: a unity port being allowed to answer a call from another unity port.

Please see the following link for more info on preventing Unity call routing 
loops.

http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/administration/guide/7xcucsag240.html#wp1049790

Thank you.

greg

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of kavi ten
Sent: Wednesday, January 13, 2010 10:03 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Vol 2 Lab3 UC 7.1

All,

What does this Question mean  :The UC Server should be prevented from answering 
calls that its originated"

Thanks, 

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Re: [OSL | CCIE_Voice] Activating Services from the CLI

2010-01-08 Thread Pulos, Greg
Please see the following link for info on changing the CUCM 7x server name.

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/install/7_0_1/ipchange/ipchg701.html

Thank you.

greg

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jefferson Wilson
Sent: Friday, January 08, 2010 12:07 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Activating Services from the CLI

Good Afternoon.   (EST)

I thought someone asked this question but  I can't find the responses in
my stored e-mail.

>From the CLI on UCM you can start and stop services.  I don't believe
you can activate services though correct?

Mine are not starting on my VM home lab and I can't web to the service
since I rebuilt it and changed the name of the server to an IP address.

Next time I will activate the services then change the name and see it
that fixes it.

Either way my UCM can't be browsed.  I will rebuild if needed.  

Jefferson

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of
ccie_voice-requ...@onlinestudylist.com
Sent: Friday, January 08, 2010 12:00 PM
To: ccie_voice@onlinestudylist.com
Subject: CCIE_Voice Digest, Vol 47, Issue 30

Send CCIE_Voice mailing list submissions to
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Today's Topics:

   1. Re: Upgrading SCCP Phones to SIP - IPExpert v. Cisco
  Discrepancy? (Roger K?llberg)


--

Message: 1
Date: Fri, 8 Jan 2010 10:38:34 +0100
From: Roger K?llberg 
Subject: Re: [OSL | CCIE_Voice] Upgrading SCCP Phones to SIP -
IPExpert v. Cisco   Discrepancy?
To: "Berry, Matthew J." , "CCIE Voice OSL
(ccie_voice@onlinestudylist.com)"

Message-ID:
<79fa99add19eda4c9880d26d736e50ef2cf7e6e...@ex2-sth.domain.root>
Content-Type: text/plain; charset="windows-1252"

I have run through the SIP CME upgrade sometimes now and I newer needed
to issue the upgrade command under voice register global.

Besides from that I believe that the example you found on CCO isn't
complete, since you at least need to have the max-dn, max-pool,
source-address and create profile configured to get CME to run at all.

Roger K?llberg
Unified Communication Consultant
Cygate AB


From: Berry, Matthew J. [mailto:mjbe...@krollontrack.com]
Sent: den 8 januari 2010 04:28
To: CCIE Voice OSL (ccie_voice@onlinestudylist.com)
Cc: Berry, Matthew J.
Subject: [OSL | CCIE_Voice] Upgrading SCCP Phones to SIP - IPExpert v.
Cisco Discrepancy?

All -

When comparing the Volume 1 PG Appendix A-B v6 511 1 PDF file with
Cisco's documentation, there seems to be a discrepancy.  On page "165"
of the IP Expert document, it shows the following commands:


voice register global

  mode cme

  source-address 10.10.202.1 port 5060

  max-dn 2

  pax-pool 2

  load 7940-7960 P0S3-08-6-00 [SIP firmware upgrading to]

  tftp-path flash:

  create profile

However, on the CUCME System Admin Guide, these commands are displayed:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/
guide/cmeinstl.html#wp1070512


Router(config)# telephony-service

Router(config-telephony)# no create cnf

CNF files deleted

Router(config-telephony)# voice register global

Router(config-register-global)# mode cme

Router(config-register-global)# load 7960 P0S3-07-4-00

Router(config-register-global)# upgrade

Router(config-register-global)# create profile


Difference:
Cisco displays the UPGRADE command under VOICE REGISTER GLOBAL.

Is that a typo?  Is that really required to do the upgrade?
I would run it through my lab, but I don't have access to it right now.
I'm interested in anyone's input.

Thanks!


Digital Footprint:
Skype: ciscovoiceguru
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For more information re

Re: [OSL | CCIE_Voice] Volume 1 (CCIE Voice 3.0) Walk-Through Video Tutorials

2009-12-30 Thread Pulos, Greg
I will rescind my comments that seem to suggest I am looking for a product I've 
never paid forperhaps I misunderstood the 'walk-through' vs. the 'VOD' 
product were different; my mistake.

Thank you for the clarification.

greg


-Original Message-
From: Wayne Lawson [mailto:groupst...@ipexpert.com] 
Sent: Wednesday, December 30, 2009 9:41 AM
To: Dew Swen
Cc: Pulos, Greg; ; Amy Ryan; Vik Malhi
Subject: Re: [OSL | CCIE_Voice] Volume 1 (CCIE Voice 3.0) Walk-Through Video 
Tutorials

Greg (and Dew) - A totally new VOD product is going to ship at no charge (you 
can check our website for the ETA). This newly overhauled VOD is much, much 
better and was redone by Vik Malhi. We totally agree with your comments on the 
initial one that was done by a former employee.  

Regarding the Volume 1 Video Walk Throughd - these have never been included in 
the BLS. These are a totally new product, thus there's no "free update".  

Also - regarding your comment about you assuming that our products should be 
updated (without a cost) due to mandated test changes - we're doing that...are 
you implying that we're not?  


Regards,

Wayne A. Lawson II - CCIE #5244
Founder & President - IPexpert
Mailto: <mailto:wlaw...@ipexpert.com> wlaw...@ipexpert.com
Telephone: +1.810.326.1444, ext. 101
Live Assistance, Please visit: <http://www.ipexpert.com/chat> 
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 (R&S, 
Voice & Security), CCNP, CCVP, CCSP and CCIE (R&S, 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 Dec 30, 2009, at 9:23 AM, Dew Swen  wrote:



Totally agree Greg.


    Dew Swen



On Wed, Dec 30, 2009 at 3:24 PM, Pulos, Greg < <mailto:gpu...@doc.gov> 
gpu...@doc.gov> wrote:


WOW...and I thought the $3500+ I've already spent on IP Expert 
self based learning products would contain updates that are mandated by test 
changes.

And to think I have to pay more for the new Video series when 
all we had was that 'mark SNORE' crap that was only good as a sleeping aid!!!

Well, needless to say, kind of disappointing to know the video 
material for the new lab version is not included, even though we've paid for 
the old video crap that I wouldn't use as toilet paper

ARG

I think I'll be saving my money this time as I doubt anything 
new can be so much better that it is worth paying for a second time.

greg



-Original Message-
From: <mailto:ccie_voice-boun...@onlinestudylist.com> 
ccie_voice-boun...@onlinestudylist.com [mailto: 
<mailto:ccie_voice-boun...@onlinestudylist.com> 
ccie_voice-boun...@onlinestudylist.com] On Behalf Of Wayne Lawson
Sent: Friday, December 25, 2009 7:56 PM
To: A A
Cc: < <mailto:ccie_voice@onlinestudylist.com> 
ccie_voice@onlinestudylist.com>; < <mailto:vma...@ipexpert.com> 
vma...@ipexpert.com>
Subject: Re: [OSL | CCIE_Voice] Volume 1 (CCIE Voice 3.0) 
Walk-Through Video Tutorials


This new prodict was not part of the BLS until a week or so ago 
- so if you purchased a BLS PRIOR to our new website going live (which we have 
all of this information on record) - it's not included (and not free) and there 
will be a charge - no exceptions. Again - these "sales" types of questions 
should be sent to yor Training Advisor.


Regards,

Wayne A. Lawson II - CCIE #5244
Founder & President - IPexpert

Mailto: mailto:wlaw...@ipexpert.com> 
wlaw...@ipexpert.com> <mailto:wlaw...@ipexpert.com> wlaw...@ipexpert.com

Telephone: +1.810.326.1444, ext. 101

Live Assistance, Please visit: < <http://www.ipexpert.com/chat> 
http://www.ipexpert.com/chat> <http://www.ipexpert.com/chat> 
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 (R&S, Voice & Security), CCNP, CCVP, CCSP and CCIE (R&S, Voice, 
Se

Re: [OSL | CCIE_Voice] Volume 1 (CCIE Voice 3.0) Walk-Through Video Tutorials

2009-12-30 Thread Pulos, Greg
WOW...and I thought the $3500+ I've already spent on IP Expert self based 
learning products would contain updates that are mandated by test changes.

And to think I have to pay more for the new Video series when all we had was 
that 'mark SNORE' crap that was only good as a sleeping aid!!!

Well, needless to say, kind of disappointing to know the video material for the 
new lab version is not included, even though we've paid for the old video crap 
that I wouldn't use as toilet paper

ARG

I think I'll be saving my money this time as I doubt anything new can be so 
much better that it is worth paying for a second time.

greg


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Wayne Lawson
Sent: Friday, December 25, 2009 7:56 PM
To: A A
Cc: ; 
Subject: Re: [OSL | CCIE_Voice] Volume 1 (CCIE Voice 3.0) Walk-Through Video 
Tutorials

This new prodict was not part of the BLS until a week or so ago - so if you 
purchased a BLS PRIOR to our new website going live (which we have all of this 
information on record) - it's not included (and not free) and there will be a 
charge - no exceptions. Again - these "sales" types of questions should be sent 
to yor Training Advisor. 


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 (R&S, 
Voice & Security), CCNP, CCVP, CCSP and CCIE (R&S, 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 Dec 25, 2009, at 7:27 PM, A A  wrote:



Can vik or wayne confirm this please ?
 



Date: Fri, 25 Dec 2009 19:25:01 -0500
Subject: Re: [OSL | CCIE_Voice] Volume 1 (CCIE Voice 3.0) Walk-Through 
Video Tutorials
From:  cciet...@gmail.com
To:  f...@hotmail.com
CC: groupst...@ipexpert.com; ssh...@gmail.com; 
ccie_voice@onlinestudylist.com;  vma...@ipexpert.com

I think the new IPX policy is that if the version changes then that's a 
new product. Atleast thats how I understood.


On Fri, Dec 25, 2009 at 7:23 PM, A A <  
f...@hotmail.com> wrote:


Entirely new?
 
I thought this is part of the BLS product. does it mean 
I'll need to pay extra to get them ?

 
> From:  groupst...@ipexpert.com
> To:  ssh...@gmail.com
> Date: Fri, 25 Dec 2009 19:15:29 -0500
> CC:  
ccie_voice@onlinestudylist.com;  vma...@ipexpert.com
> Subject: Re: [OSL | CCIE_Voice] Volume 1 (CCIE Voice 3.0) 
Walk-Through Video Tutorials 

> 
> The Vol 1 Video Solutions are an entirely new product - 
pricing 
> details are on our website.
> 
> 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 
> (R&S, Voice & Security), CCNP, CCVP, CCSP and CCIE (R&S, 
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 Dec 25, 2009, at 6:53 PM, Lala Lander < 
 ssh...@gmail.com> wrote:
> 
> > So how existing custo era cab get this product
> >
> > On Friday, December 25, 2009, CCIETalk.com < 
 cciet...@gmail.com> 

Re: [OSL | CCIE_Voice] IPexpert Web site - still issues?

2009-12-07 Thread Pulos, Greg
Ditto

Unable to login with any username, new or old.

gpulos


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Steve Denney 
(stdenney)
Sent: Monday, December 07, 2009 2:04 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] IPexpert Web site - still issues?

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   

 

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Re: [OSL | CCIE_Voice] Unified Contact Center Express for CCIE Voice v 3

2009-12-04 Thread Pulos, Greg
An excellent source is the ccx getting started with scripts document.

http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_7_0/user/guide/uccx70edgs.pdf

thank you.

gpulos

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Talmadge Almand
Sent: Thursday, December 03, 2009 4:15 PM
To: akash patel
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Unified Contact Center Express for CCIE Voice v 
3

Yes to the troubleshooting being fairgame, and you can also expect any 
variation for scripting, I would recommend the advanced scripting training, or 
practice some custom scenarios, the ones that IPExpert details are a good 
measure of what to expect.


On Thu, Dec 3, 2009 at 3:30 PM, akash patel  wrote:


I understood from previous blog/post that UCCX will be pre integrated 
with UCM in the lab, however we will be expected to troubleshoot the 
integration.  I can take help from SRND, 
 
however as far 2nd piece of UCCX which is scripting, what material do 
you recommend to study? 
 
 Is there any document outlining what version of UCXX is in test 
(premier, enhance or standard) and /or what topics will be tesable (can imagine 
standard AA, ACD), but weather email, DB dig out, web application, java applet 
creation could be testable as well?  
 
Thank you

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-- 
Regards,

Talmadge Almand
CCIE #20901 (Voice)
Sr, Support Engineer - IPexpert, Inc.
URL:http://www.IPexpert.com

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Re: [OSL | CCIE_Voice] CME QUESTION

2009-11-25 Thread Pulos, Greg
I think a more accurate answer to the questions are as follows:

1) NOSRST does not come as a licensed IOS feature by default. You will have 
to obtain the SRST license to be legal.

2) YES...if you do extract a CME tar file to the flash on your router, you 
should be able to see the files with 'dir flash:'.


Thank you.

greg



-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Wilson Bolanos
Sent: Wednesday, November 25, 2009 11:02 AM
To: Wilson Bolanos; iptuser55; ccie voice
Subject: Re: [OSL | CCIE_Voice] CME QUESTION

Also, I have attached a file that will probably answer your SRST question.  
Below is a sh ver of my 2821 router that will show what resources I have on it:

 

 

BR2#conf t

Enter configuration commands, one per line.  End with CNTL/Z.

BR2(config)#telephony-service

BR2(config-telephony)#^Z

BR2#

*Nov 25 15:54:30.847: %SYS-5-CONFIG_I: Configured from console by console

BR2#sh ver

Cisco IOS Software, 2800 Software (C2800NM-ADVENTERPRISEK9_IVS_LI-M), Version 
12  
   .4(22)T, RELEASE SOFTWARE (fc1)

Technical Support: http://www.cisco.com/techsupport

Copyright (c) 1986-2008 by Cisco Systems, Inc.

Compiled Fri 10-Oct-08 00:05 by prod_rel_team

 

ROM: System Bootstrap, Version 12.4(13r)T11, RELEASE SOFTWARE (fc1)

 

BR2 uptime is 2 minutes

System returned to ROM by power-on

System image file is "flash:c2800nm-adventerprisek9_ivs_li-mz.124-22.T.bin"

 

 

This product contains cryptographic features and is subject to United

States and local country laws governing import, export, transfer and

use. Delivery of Cisco cryptographic products does not imply

third-party authority to import, export, distribute or use encryption.

Importers, exporters, distributors and users are responsible for

compliance with U.S. and local country laws. By using this product you

agree to comply with applicable laws and regulations. If you are unable

to comply with U.S. and local laws, return this product immediately.

 

A summary of U.S. laws governing Cisco cryptographic products may be found at:

http://www.cisco.com/wwl/export/crypto/tool/stqrg.html

 

If you require further assistance please contact us by sending email to

exp...@cisco.com.

 

Cisco 2821 (revision 53.50) with 243712K/18432K bytes of memory.

Processor board ID FTX1342AJGB

2 Gigabit Ethernet interfaces

1 Serial interface

1 Channelized/Clear T1/PRI port

1 Virtual Private Network (VPN) Module

DRAM configuration is 64 bits wide with parity enabled.

239K bytes of non-volatile configuration memory.

509544K bytes of ATA CompactFlash (Read/Write)

 

Configuration register is 0x2102

 

BR2#dir flash:

Directory of flash:/

 

1  -rw-58816728  
c2800nm-adventerprisek9_ivs_li-mz.124-22.T.bin

2  -rw-   18836  
bacdprompts/app-b-acd-2.1.2.2-ReadMe.txt

3  -rw-   24985  
bacdprompts/app-b-acd-2.1.2.2.tcl

4  -rw-   35485  
bacdprompts/app-b-acd-aa-2.1.2.2.tcl

5  -rw-   75650  
bacdprompts/en_bacd_allagentsbusy.au

6  -rw-   83291  
bacdprompts/en_bacd_disconnect.au

7  -rw-   63055  
bacdprompts/en_bacd_enter_dest.au

8  -rw-   37952  
bacdprompts/en_bacd_invalidoption.au

9  -rw-  496521  
bacdprompts/en_bacd_music_on_hold.au

   10  -rw-  123446  
bacdprompts/en_bacd_options_menu.au

   11  -rw-   42978  
bacdprompts/en_bacd_welcome.au

 

An important file that will come up when you do a dir flash: is 
CME-7-0-full-readme-v.1.0.txt.  Do a command: more 
CME-7-0-full-readme-v.1.0.txt to read it.  It will help.  Below is a small 
preview of that file:

 

CAVEAT: Only CME systems with a minimum of 128MB of flash should use the 
cme-full-7-0-0-0.tar file;

with smaller flash sizes, customers should use the cme-basic-7-0-0-0.tar file 
and then individually add phone

firmwares, ringtones, backgrounds, and b-acd prompts as needed.

 

Please follow the procedure below to upgrade to CME 7.0 from any earlier 
version of CME.  This is to be done

after upgrading the IOS to support CME 7.0.  For a list of IOS/CME 
compatibility matrix, please refer to

the following link.

 

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/requirements/guide/33matrix.htm

 

Recommended TFTP server: Unix TFTP or JouninTFTP server for windows.  When 
downloading large

files via TFTP, some free tftp servers can't handle files larger than 32mb.  
Older non-32 bit tftp-servers

can't handle large files and thus tftp transfers may fail.

 

NOTE:

(1) These new tar packages can only be used with routers that support unix file 
structure such

as 37xx, 28xx, 3

Re: [OSL | CCIE_Voice] Cisco Telephony OS / VMWare ESXi

2009-10-30 Thread Pulos, Greg
No special reg patch needed but you do need some reg entries which are as 
follows:

KEY - Hkey_local_machine/software/Cisco Systems/Model
String Value - Hardware 7835H04
String Value - Memory 2048
String Value - Speed 3400

1) install windows 2003 server (any ver, not MCS disk)
2) put above entries into registry after server install
3) install MCS winOS disk or download install from cco
4) install CRS

Note: you need physical 2048mb memory but not 7835 or 3400 speed.

greg


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Iwan
Sent: Friday, October 30, 2009 6:59 AM
To: 'Jeff Garvas'; OSL Group
Subject: Re: [OSL | CCIE_Voice] Cisco Telephony OS / VMWare ESXi

Jeff, 

 

What are you trying to install exactly?

I used The Cisco OS (cisco iptelephony server os2003.1.4) which I used to 
install UCCX.

I know that all the versions below 1.4 are not Vmware aware and you need a 
special .reg patch for this.

 

This can be found here:

http://tannerezell.com/cisco/?p=164  

 

 

 ing. Iwan Hoogendoorn

CCIE3#13084

 

Van: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] Namens Jeff Garvas
Verzonden: vrijdag 30 oktober 2009 7:05
Aan: OSL Group
Onderwerp: [OSL | CCIE_Voice] Cisco Telephony OS / VMWare ESXi

 


I think I'm missing something obvious but I can't find anything with google 
that gives this one away. Has anyone successfully loaded Cisco Telephony OS for 
uccx in vmware ESXi without installing it on an actual MCS server first and 
creating a image?   I'm using the HP 2003.1.1 disk, but I'm installing on a 
simple Intel CPU box.

I built a 80gb partition, 3.7gb of memory, and I keep getting this error early 
in the install:   "The installation program has determined that this server is 
not supported by this media"

What is the best way to install the OS on a generic ESXi machine?

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Re: [OSL | CCIE_Voice] AIM-VOICE-30 for PSTN simulator

2009-10-26 Thread Pulos, Greg
AIM-VOICE-30 DSPs can only be used for TDM termination on T1/E1.

DSP sharing is not available for this module.

Please see the following link for more info:

http://www.cisco.com/en/US/docs/ios/12_2t/12_2t8/feature/guide/ft_04gin.html#wp1053375

(notice the dsp allocation can be shared when using a T1/E1 packet voice 
network module; not the AIM dsps)

Thank you.

Greg

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Allan Ren
Sent: Sunday, October 25, 2009 6:37 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] AIM-VOICE-30 for PSTN simulator

Hi,
 
Im trying to build up my home lab using a 2621XM router as PSTN simulator. An 
AIM-VOICE-30 was supposed to provide DSPs to vwic-2mft-e1 and vwic-2mft-t1. The 
problem is I cant create the second pri-group. The first pri-group was created 
successfully. When I create the second pri-group, it failed with following 
error prompt:
 
% Not enough DSP resources available to configure pri-group on controller T1 
0/2 % The remaining dsp resources are enough for 28 time slots.
% For current codec complexity, 0 extra dsp(s) are required to create this 
voice port.
 
I guess AIM-VOICE-30 don't support DSPs sharing. Once it was assigned to e1 or 
t1, the remaining DSPs cant be assigned to another t1 or e1. But previous email 
in this list, someone posted lab equipment list that AIM-VOICE-30 is the DSP 
resources for both E1 and T1. Did I get wrong understanding? Any response would 
be much appreciated :)
 
Cheers,
Allan
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Re: [OSL | CCIE_Voice] b-channel order in H323

2009-10-16 Thread Pulos, Greg
It looks as though you've selected 1-12 AND 16 as your active PRI timeslots.

As they are numbered from 0 to 23, 16 would actually equal S0/0/0:15. (as shown 
in your output)

You stated you were only trying to turn on 12 channels but you've turned on 13. 
Is the 13th channel, aka S0/0/0:15 available?

In this case it is not. Remove the ,16 from your pri-group command and it 
should work both ways.

greg


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of P N
Sent: Friday, October 16, 2009 12:52 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] b-channel order in H323

Hi All,
 
I've a problem on b-channel usage order on a fractional H323, if I only turned 
on 12 b channels, and use it as default - descending, I will hit fast busy and 
get "Cause i = 0x82AC18 - Requested circuit/channel not available". But if I 
use it as ascending, then call can go thru. Shouldn't be H323 takes care of 
fractional PRI in pri-group already?
 
Config: 
 
controller E1 0/0/0
 pri-group timeslots 1-12,16
!
interface Serial0/0/0:15
 no ip address
 encapsulation hdlc
 isdn switch-type primary-net5
 isdn incoming-voice voice
 isdn outgoing display-ie
 no cdp enable
!
interface GigabitEthernet0/0.103
 encapsulation dot1Q 103
 ip address 142.103.66.254 255.255.255.0  h323-gateway voip interface  
h323-gateway voip bind srcaddr 142.103.66.254 !
 
Debug:
 
Oct 16 04:49:01.973: ISDN Se0/0/0:15 Q931: TX -> SETUP pd = 8  callref = 0x008C 
Bearer Capability i = 0x8090A3 
Standard = CCITT 
Transfer Capability = Speech  
Transfer Mode = Circuit 
Transfer Rate = 64 kbit/s 
Channel ID i = 0xA9838C 
Exclusive, Channel 12 
Progress Ind i = 0x8183 - Origination address is non-ISDN  
Calling Party Number i = 0x0180, '32143001' 
Plan:ISDN, Type:Unknown 
Called Party Number i = 0x81, '999' 
Plan:ISDN, Type:Unknown
Oct 16 04:49:01.985: ISDN Se0/0/0:15 Q931: RX <- RELEASE_COMP pd = 8  callref = 
0x808C 
Cause i = 0x82AC18 - Requested circuit/channel not available
 
Thanks
Patrick Ng

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Re: [OSL | CCIE_Voice] T1 CAS

2009-09-28 Thread Pulos, Greg
A specific isdn switch type depends on your telco. You must verify with them 
what type of switch is terminating the PRI.

If this is the IP Expert labs, then the switch type should be identified in 
your Lab Info sheets which show IP addresses, circuit info, etc.

You'll also need to setup the Serial port created by the PRI Group. Your serial 
interface will be S0/3/0:23. Don't forget binding the ISDN L3 to ccm on that 
interface. (this is how the gateway backhauls the PRI signaling to the ccm)

Greg.




-Original Message-
From: J Hogan [mailto:j.jho...@gmail.com] 
Sent: Monday, September 28, 2009 12:05 PM
To: Pulos, Greg
Cc: James Key; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] T1 CAS

I did not know that it needs to be pri- group. Also is there a  
specific isdn switch type?

Thanks

Sent from my iPhone

On Sep 28, 2009, at 6:24 AM, "Pulos, Greg"  wrote:

> How is it you're attempting to configure CAS channels to talk using  
> CCS signaling? (aka ISDN)
>
> Change your DS0 group command to PRI group.
>
> greg
>
>
> -Original Message-
> From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice- 
> boun...@onlinestudylist.com] On Behalf Of J Hogan
> Sent: Sunday, September 27, 2009 5:42 PM
> To: James Key
> Cc: ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] T1 CAS
>
> I am using the ccm-manager bind command and the interface is up/up  
> as this is also the interface I telnet into.
>
>
> On Sun, Sep 27, 2009 at 8:56 AM, James Key  wrote:
>
>
>Remove both MGCP bind commands, see if it comes up, then add back  
> in.
> 
>
>From: ccie_voice-boun...@onlinestudylist.com [ccie_voice- 
> boun...@onlinestudylist.com] On Behalf Of J Hogan [j.jho...@gmail.com]
>Sent: Sunday, September 27, 2009 8:41 AM
>To: Nara Shikamaru
>Cc: ccie_voice@onlinestudylist.com
>Subject: Re: [OSL | CCIE_Voice] T1 CAS
>
>
>the domanin name i removed. and yes the router host name matches  
> too?
>
>
>On Sat, Sep 26, 2009 at 8:32 PM, Nara Shikamaru  > wrote:
>
>
>Verify that the gateway's hostname.domain-name matches the  
> Domain Name field on the router.
>
>
>On Sat, Sep 26, 2009 at 1:11 PM, J Hogan   
> wrote:
>
>
>Hello All
>
>   I have a T1 configured and I keep getting registration  
> rejected. I beleive I am doing everything right
>1.) I restarted MGCPno mgcp, mgcp
>2.) I only have 16 DSPs
>2.) here is what i did   thughts
>
>voice class codec 1
> codec preference 1 g729r8
> codec preference 2 g711ulaw
>
>card type t1 0 3
>
>controller T1 0/3/0
>
>controller T1 0/3/0
> framing esf
> linecode b8zs
> cablelength short 110ft
> ds0-group 1 timeslots 1-10 type e&m-wink-start
>
>ccm-manager mgcp
>no ccm-manager fax protocol cisco
>ccm-manager music-on-hold
>ccm-manager config server 192.168.60.6
>ccm-manager config
>!
>mgcp
>mgcp call-agent 192.168.60.6 service-type mgcp version 0.1
>mgcp rtp unreachable timeout 1000 action notify
>mgcp modem passthrough voip mode nse
>mgcp package-capability rtp-package
>mgcp package-capability sst-package
>mgcp package-capability pre-package
>no mgcp package-capability res-package
>no mgcp timer receive-rtcp
>mgcp sdp simple
>mgcp fax t38 inhibit
>mgcp rtp payload-type g726r16 static
>mgcp bind control source-interface FastEthernet0/0
>mgcp bind media source-interface FastEthernet0/0
>
>
>
>
>--
>J. Hogan MCP,CCDA,CCDP, CCNA, CCNP, CCSP, CCAI
>Yahoo ID: jhogan552000
>AIM ID: jhogan55
>MSN ID: jhogan55
>ICQ ID: 257599283
>
>Live Life And Do Not Kill Time.
>
>
>___
>For more information regarding industry leading CCIE Lab  
> training, please visit www.ipexpert.com <http://www.ipexpert.com/>
>
>
>
>
>
>
>--
>-Shikamaru
>
>
>
>
>
>--
>J. Hogan MCP,CCDA,CCDP, CCNA, CCNP, CCSP, CCAI
>Yahoo ID: jhogan552000
>AIM ID: jhogan55
>MSN ID: jhogan55
>ICQ ID: 257599283
>
>Live Life And Do Not Kill Time.
>
>NOTICE: This electronic ma

Re: [OSL | CCIE_Voice] T1 CAS

2009-09-28 Thread Pulos, Greg
How is it you're attempting to configure CAS channels to talk using CCS 
signaling? (aka ISDN)

Change your DS0 group command to PRI group.

greg


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of J Hogan
Sent: Sunday, September 27, 2009 5:42 PM
To: James Key
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] T1 CAS

I am using the ccm-manager bind command and the interface is up/up as this is 
also the interface I telnet into.


On Sun, Sep 27, 2009 at 8:56 AM, James Key  wrote:


Remove both MGCP bind commands, see if it comes up, then add back in.


From: ccie_voice-boun...@onlinestudylist.com 
[ccie_voice-boun...@onlinestudylist.com] On Behalf Of J Hogan 
[j.jho...@gmail.com]
Sent: Sunday, September 27, 2009 8:41 AM
To: Nara Shikamaru
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] T1 CAS


the domanin name i removed. and yes the router host name matches too?


On Sat, Sep 26, 2009 at 8:32 PM, Nara Shikamaru  
wrote:


Verify that the gateway's hostname.domain-name matches the 
Domain Name field on the router.


On Sat, Sep 26, 2009 at 1:11 PM, J Hogan  
wrote:


Hello All

   I have a T1 configured and I keep getting 
registration rejected. I beleive I am doing everything right
1.) I restarted MGCPno mgcp, mgcp
2.) I only have 16 DSPs 
2.) here is what i did   thughts

voice class codec 1
 codec preference 1 g729r8
 codec preference 2 g711ulaw

card type t1 0 3

controller T1 0/3/0

controller T1 0/3/0
 framing esf
 linecode b8zs
 cablelength short 110ft
 ds0-group 1 timeslots 1-10 type e&m-wink-start

ccm-manager mgcp
no ccm-manager fax protocol cisco
ccm-manager music-on-hold
ccm-manager config server 192.168.60.6
ccm-manager config
!
mgcp
mgcp call-agent 192.168.60.6 service-type mgcp version 
0.1
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp package-capability sst-package
mgcp package-capability pre-package
no mgcp package-capability res-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp fax t38 inhibit
mgcp rtp payload-type g726r16 static
mgcp bind control source-interface FastEthernet0/0
mgcp bind media source-interface FastEthernet0/0




-- 
J. Hogan MCP,CCDA,CCDP, CCNA, CCNP, CCSP, CCAI
Yahoo ID: jhogan552000
AIM ID: jhogan55
MSN ID: jhogan55
ICQ ID: 257599283

Live Life And Do Not Kill Time.


___
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Lab training, please visit www.ipexpert.com  






-- 
-Shikamaru





-- 
J. Hogan MCP,CCDA,CCDP, CCNA, CCNP, CCSP, CCAI
Yahoo ID: jhogan552000
AIM ID: jhogan55
MSN ID: jhogan55
ICQ ID: 257599283

Live Life And Do Not Kill Time.

NOTICE: This electronic mail message and any files transmitted with it 
are intended
exclusively for the individual or entity to which it is addressed. The 
message, 
together with any attachment, may contain confidential and/or 
privileged information.
Any unauthorized review, use, printing, saving, copying, disclosure or 
distribution 
is strictly prohibited. If you have received this 

Re: [OSL | CCIE_Voice] Problem in CDR reporting

2009-03-13 Thread Pulos, Greg
This shows the date(s) you are attempting to report on does not exist.

Either you don't have CDR collection running properly or it wasn't during the 
date(s) you are selecting.

1) Make sure your CDR Insert is running on the pub. (doesn't run on subs)
2) Make sure your CDR parameters properly enabled and are acceptable for the 
amount of CDR data you wish to collect, etc.

Please see the following link for more info on CDR query failure:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a0080576153.shtml

greg


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of asif raza
Sent: Friday, March 13, 2009 1:06 AM
To: CCIE-Voice
Subject: [OSL | CCIE_Voice] Problem in CDR reporting

Hi Friends,

I am using CCM 4.1, and receiving problem with CDR reporting, ScreenShot is 
attached.


Thanks and Best Regards

Asif Raza





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Re: [OSL | CCIE_Voice] Not able to start the synchronization between Pub - Sub

2009-03-12 Thread Pulos, Greg
You're publisher doesn't have the Distributor either configured correctly 
and/or running.

You're DB is there so all you have to do is setup the Distributor to push it to 
the sub.

If you're not familiar with SQL and getting the DBs synchronizing, use 
DBHelper, which comes with CCM to see if it can get the Distribution going.

Or..

Try to goto the Distributor Agent on the pub and start it. Once started, you 
may need to start the Snapshot Agent to snap the DB and push it to the sub.

Please see the following link for more info on reestablishing a broken SQL 
subscription for CCM:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a00801d11a6.shtml

greg

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of kapil atrish
Sent: Thursday, March 12, 2009 12:31 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Not able to start the synchronization between Pub - 
Sub

  Hi 
List,

See the attached screen-shots from Publisher Enterprise Manager. I've 
re-registered both servers, started Agent but still Subscriber is not able to 
pull the database successfully. I've Hosts/Lmhosts file in place and name 
resolution is working fine. Reboot didn't help. I didn't do any thing at 
Subscriber>>enterprise manager except updating the hosts/lmhosts file.

Now, none of my config is getting synchronized with the sub. Any inputs on this 
is highly appreciated.






Re: [OSL | CCIE_Voice] Unity>>UTIM>>Message Notification VS Dialout MWI

2009-02-18 Thread Pulos, Greg
You are correct that MWI has no related to Message Notification.

To help prevent Unity from looping through its ports you can goto UTIM and on 
the ports page, put the DN for the specific port(s) as they are configured in 
CCM.

This will allow Unity to know the call came from one of its ports and not to 
loop back to it.

greg

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kapil Atrish
Sent: Wednesday, February 18, 2009 1:16 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Unity>>UTIM>>Message Notification VS Dialout MWI

Hi List,

I'v a question about Message notification, I read about it, pl let me know if 
my understanding is correct.

I understand Message Notification ports are used when susbcriber has 
notifications enabled on additional devices under Subscriber>>Message 
notification settings, like Home Phone/Pager etc.

So if I don't have notification enabled on additional devices do I still need 
to uncheck "Message Notification" on Ports to avoid Unity looping or is there 
something I am missing?

MWI has nothing to do with "Message Notification ports" but only with "Dialout 
MWI"ports, correct?

Thanks in advance...






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acquisitions and deals on BPO Watch. Try it now! 



Re: [OSL | CCIE_Voice] NTP

2009-01-31 Thread Pulos, Greg
Ntp Fudge is a 'keyword utility for ntp which can be used to configure 
reference clocks in specific ways. It can be used to define calibration 
constants to force a time offset to a particular time standard.'
(ie: to set or 'fudge' a timesource and stratum)

Example:

Server 23.230.1.10
Fudge 127.127.1.0 statum 14

The above example creates two ntp time sources. The first source is a valid ntp 
server on the network. The second entry is a 'backup' which points to the 
systems own local clock. Since the local clock time is not very accurate, you 
should 'fudge' it to a low stratum.

The stratum can almost be considered as 'ntp hops'; how far from the radio 
clock is the ntp server. It defines the 'ntp hierarchical structure'. Stratum 1 
would be the highest priority in the ntp hierarchical structure of a device 
directly synched to a radio clock source, which would be stratum 0.

We would likely use stratum 3 or higher on most simple ntp setups.
Statum 0 would be the radio clock source; by default.
Stratum 1 would be your router that has a direct synch with that radio clock 
source
Stratum 2 would be perhaps an internal ntp server that syncs to the router.
Stratum 3 would be used for any devices that synch to your internal ntp server.

Greg


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ryan Trauernicht
Sent: Friday, January 30, 2009 2:19 PM
To: Mark Snow
Cc: OSL Group
Subject: Re: [OSL | CCIE_Voice] NTP

Thank you for the reply Mark.  What exactly is the Fudge line for

My drift file also only has a number in it (23.121).  I have never really 
understood what the number is for.

My ntp.config file looks just like below:

server 192.168.187.11 # Set Local Clock to Authoritive Time Source  <-- NTP 
source IP
driftfile C:\WINNT\system32\drivers\etc\ntp.drift # path for drift file

I took out the fudge command and I cant remember what was in it untill i 
rebuild my VM box again tonight, but what should the fudge command be?

thanks,
Ryan Trauernicht



I have set my BR1 to my HQ router (which is the master) and it sync's just fine.


On Fri, Jan 30, 2009 at 7:39 AM, Mark Snow  wrote:


Ryan,

Try everything again except in your step 5, don't delete the "fudge" 
line of code. Then restart the the NTP service however without running the 
NTPdate.exe.

ntpdate.exe and the ntp.conf/ntp service are mutually exclusive form 
one another.


Also, instead of setting your system clock to within 10 mins of the 
correct time on your ntp master router- set it to an entirely different hour 
and maybe even year.
Every time you stop and start the ntp service, ntp will attempt to 
update.

BTW, let's say your NTP Master router is the HQ router, did you ever 
try setting up say a BR1 router to be a NTP client to see if it syncs properly 
with the master first - to make sure the problem doesn't lie with the master 
instead of the UCM server?
If not - be sure to try that first.

Cheers,

Mark Snow
Sr Technical Instructor
IPexpert, Inc.

Sent from my iPhone

On Jan 30, 2009, at 8:22 AM, Chris Parker  wrote:



Did you set your driftfile in ntp.conf?




From: Ryan Trauernicht 
Sent: Thursday, January 29, 2009 11:39 PM
To: OSL Group 
Subject: [OSL | CCIE_Voice] NTP

I know this topic has been covered as well... so sorry again 
for the spam.

I have never gotten the NTP on CM to work properly.  The 
process I have always followed is below:


1. Start>Run>Services.msc
2. Verify the Windows Network Time is disabled
3. Stop Network Time Protocol (leave as automatic)
4. Set local CM time to close to real time (within 10 mins)
5. Update ntp.config (c:\winnt\system32\drivers\etc) with the 
"server x.x.x.x" of the NTP server and delete the fudge item in the text file)(
5. open up command prompt and navigate to c:\program 
files\cisco\xntp
6. run "ntpdate.exe x.x.x.x"
7. Start Network Time Protocol


Once this process is done, CM updates just fine untill it is 
rebooted.  Once it is rebooted my CM goes back to a time I don't know where it 
is pulling it from.  It is a dedicated box to CM so not a VMware.  I have lost 
points on this and I don't want to lose them again any ideas what is 
missing from this process?

thanks!
Ryan





Re: [OSL | CCIE_Voice] CCIE lab DOC-CD (univercd replacement)

2009-01-07 Thread Pulos, Greg
I'm going to disagree with this one.

I know you CAN go into the ccie lab and remember everything required. It's the 
difference between really knowing/memorizing and kinda knowing/memorizing the 
required material. (a lot of people may not do it, but it is very possible)

If one really wanted to, I can't see how a person cannot remember everything; 
this isn't rocket science. All the information is well documented in a number 
of ways which makes it very possible to learn/know it all before a lab is even 
attempted.

I've been networking cisco for over 20 years and would never attempt a ccie lab 
if I have any hint that there is information on the lab test that requires me 
looking it up. The ccie blueprints are very detailed on what exactly can be 
covered, so not knowing something by heart before going into the lab just 
doesn't make much sense.

As for the speed issue, I've always found it much quicker and easier to just 
enter the commands quick from memory. The way IOS uses ambiguity in its CLI 
only makes it that much faster and easier.   (that's just me and my experience 
and I know people are not the same)

So what is the major difference between kinda knowing and really knowing? It 
almost always comes down to experience. The more you have with the technology, 
the more ingrained it is in our entire being.

greg

From: Jacob Owen [mailto:certmas...@gmail.com]
Sent: Tuesday, January 06, 2009 3:04 PM
To: Pulos, Greg
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE lab DOC-CD (univercd replacement)

Greg,
Having taken 10 CCIE labs in my day I can tell you that you CAN NOT go in 
remembering everything that you may see when it comes to a CCIE Lab.  Many 
times I would have to trade off between remembering only 1 of 2 things and just 
remembering where to find the item I chose not to remember was in the 
documentation.  It's also a matter of speed, if you can copy and paste QoS 
configs from the SRND to a notepad, make your changes and then copy and paste 
into your device that can save you far more time than trying to remember the 
syntax of every command.  I don't say go into the Lab expecting to use the 
documentation for the majority of it, just get a good sense of where the 
commonly hit items are in case you need to look something up.
On Tue, Jan 6, 2009 at 1:49 PM, Pulos, Greg 
mailto:gpu...@doc.gov>> wrote:
I've seen this question about what documents are available in the lab so much 
that I feel I have to comment.

How can someone expect to go into the lab thinking they may need to use 
documentation to find answers?

I believe if a person feels they may need the documentation available in the 
lab, then they're not ready for the lab.

greg

-Original Message-
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 Chris Parker
Sent: Tuesday, January 06, 2009 12:01 PM
To: Wieland, John
Cc: ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com>
Subject: Re: [OSL | CCIE_Voice] CCIE lab DOC-CD (univercd replacement)

Keeping with the NDA ...

Based on my experience, I think its very likely that section will not be
available.


Wieland, John wrote:
> Team,
>
> There was some question as to if the new Cisco Connection Online (CCO - 
> replacement for univercd)  was going to 'strip' out the "configuration 
> examples and technotes" links when made available during the CCIE Lab.
> Does anyone know if this is true, or can we study with the understanding that 
> these examples and technotes will be accessible to us during the lab?
>
> Thanks in advance,
>
> john
>
>
>




--
Jacob Owen
CCIE #14063 (R&S, Voice, Service Provider), CCDP


Re: [OSL | CCIE_Voice] CCIE lab DOC-CD (univercd replacement)

2009-01-06 Thread Pulos, Greg
I've seen this question about what documents are available in the lab so much 
that I feel I have to comment.

How can someone expect to go into the lab thinking they may need to use 
documentation to find answers?

I believe if a person feels they may need the documentation available in the 
lab, then they're not ready for the lab.

greg

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Chris Parker
Sent: Tuesday, January 06, 2009 12:01 PM
To: Wieland, John
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE lab DOC-CD (univercd replacement)

Keeping with the NDA ...

Based on my experience, I think its very likely that section will not be
available.


Wieland, John wrote:
> Team,
>
> There was some question as to if the new Cisco Connection Online (CCO - 
> replacement for univercd)  was going to 'strip' out the "configuration 
> examples and technotes" links when made available during the CCIE Lab.
> Does anyone know if this is true, or can we study with the understanding that 
> these examples and technotes will be accessible to us during the lab?
>
> Thanks in advance,
>
> john
>
>
>




Re: [OSL | CCIE_Voice] Please Kindly advise about service activation on pub and sub in lab exam

2009-01-05 Thread Pulos, Greg
Needed services are dependent upon features required and what server they 
should run on.

For example, why enable 'CDR Insert' on a subscriber?? It can never be used on 
a Sub.

Use the help in CCM to learn about the services, what they do and what servers 
they should be enabled on.

Not all services are enabled on all servers.

greg

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of jeremy co
Sent: Sunday, January 04, 2009 2:10 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Please Kindly advise about service activation on 
pub and sub in lab exam

Hi folks,

Can anyone give me some tips on activating services on lab exam?


which services should be enabled on pup?
which services should be enabled on sub?
Is there any negative point to enable all of services on pub and sub?

My practice is
Pub : all services except ipmanager, extension mobility and webdialer
Sub : all services except TFTP, ipmanager, extension mobility and webdialer


ipmanager and extention mobility and webdialer are obvious since they should be 
enabled upon a request. ( But if they asked one of these features, should it be 
enabled on sub too?


Cheers

Jeremy



Re: [OSL | CCIE_Voice] using Change B-Channel Maintenance for IOS T1 or not?

2008-12-24 Thread Pulos, Greg
The b channel maintenance should be used to allow you to change the state of 
the channels in real time while the circuit is up. Commonly used for 
troubleshooting.

CallMan can change the state of the channels in real time to either:

0 - in service
1 - graceful out of service
2 - forceful out of service

greg

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of 
saralilin2...@yahoo.co.jp
Sent: Wednesday, December 24, 2008 1:36 AM
To: co jeremy; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] using Change B-Channel Maintenance for IOS T1 
or not?


we only need to do b-channel maintenance if question ask for bottom up right? 
if we choose top down this is not needed, am i right?

Sara

jeremy co  wrote:

Hi,

I've seen some workbooks use Change B-Channel Maintenance option to busyout 
unused channels on T1 of IOS GWs as well as 6500 T1 while some of them only use 
this option on 6500 T1.

In cisco Docs, I can it specified to use thi option for  "MGCP gateways"


So which method should be used?

btw, I use both and both works for IOS .


Jeremy




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Re: [OSL | CCIE_Voice] crs editor ?

2008-12-12 Thread Pulos, Greg
This is telling you the 'trigger', likely in this case the 'ContactTrigger' 
component, is set wrong.
(it says it's a RemoteApplicationTrigger)

The debug should show you where the error was generated so you can confirm the 
component.

Did you edit the script? Is there a typo or mistake if you did?

Post the script and I can review for you if you like.

greg
-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Erwan Erwan
Sent: Thursday, December 11, 2008 6:15 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] crs editor ?

hi,

anybody experience this ?  I tried to debug  CRS script using CRS Editor and it 
show error msg, like this

Wrong trigger type: com.cisco.app.remote.RemoteApplicationTrigger

thks






Re: [OSL | CCIE_Voice] IPCC with CCM integration problem , trace : ROUTE_FAILED

2008-11-18 Thread Pulos, Greg
It looks as though your CTI Port that is selected to be used to redirect/route 
the call is in use. (per the trace)

I see you have 1801 and 1802 cti ports. The trigger is attempting to use 1801 
and finding it unavailable after it is selected. Are the CTI Ports registered? 
If not, verify they are associated with the proper jtapi user you created in 
the GD.

Also verify the trigger max connections settings and the mediaGroup it is 
supposed to use. These are setup/found in the CRS configuration of the jtapi 
trigger and mediaGroup configuration.

You can view the operation logs of all the 'behind-the-scenes' components of 
ipccx by using the 'dumplog' command on the separate CRS components which can 
give you a broader view of the CRS engine and how its components are operating.

Please see the following link for more info on IPCC(x) troubleshooting:
http://www.cisco.com/en/US/products/sw/custcosw/ps1846/prod_troubleshooting_guides_list.html

This link shows recommended trace levels for troubleshooting ipcc:
http://www.cisco.com/en/US/products/sw/custcosw/ps1844/products_tech_note09186a0080094b22.shtml

greg

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cyrus
Sent: Tuesday, November 18, 2008 6:04 AM
To: Adrian Clinton - Watkins
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] IPCC with CCM integration problem , trace : 
ROUTE_FAILED

Hi Adrian,


I have no PT or CSS in CCM, Every thing is in none partition.

 just to try to walk be4 running.


Cheers


On Tue, Nov 18, 2008 at 9:25 PM, Adrian Clinton - Watkins <[EMAIL PROTECTED]> 
wrote:


Check partitions of CTI ports and RPs. I think you will find your CTI 
ports are not in a part that cannot be dialed from the source CSS.

Adrian Clinton - Watkins
CCIE #21806, CCDP, CCNP. CCVP, MCSE
GGR Communications Limited
DDI Phone: +44 (0)1905 825923
Office: +44 (0)1905 825900
Fax: +44 (0)1905 825901
Email: [EMAIL PROTECTED]

http://www.ggr.net 

For technical support please email [EMAIL PROTECTED] or call +44 
(0)1905 825999.





From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cyrus
Sent: 18 November 2008 10:01 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] IPCC with CCM integration problem ,trace : 
ROUTE_FAILED


Hi  fellas,

I got a problem in integrating IPCC and CCM , here is the trace file.

if anyone knew what "ROUTE_FAILED" meaning in this trace file, let me 
know.
I guess it cannot connect to CTI ports of ccm.

ICD RT point :1800
CTI ports :1801 ,1802

1764: Nov 17 01:14:16.796 EST %MIVR-SS_TEL-7-UNK:Route 
Connection=[1800/(P1-jtapi_1_1) GCID=(1,15)->ACTIVE]->OFFERED, reason=1, Event= 
CallCtlConnOfferedEv, cause=100, metacode=128, isMaster=true
1765: Nov 17 01:14:16.875 EST %MIVR-SS_CM-7-UNK:RmCm contact -1[null] 
.setAppContact(0) from null
1766: Nov 17 01:14:16.875 EST %MIVR-SS_TEL-7-UNK:Call.received() 
JTAPICallContact[id=0,implId=15/1,inbound=true,App 
name=ICD,task=null,session=null,seq 
num=-1,cn=1800,dn=1800,cgn=3002,ani=null,dnis=null,clid=null,atype=DIRECT,lrd=null,ocn=1800,route=TR[num=1800],TP=null
1767: Nov 17 01:14:16.890 EST %MIVR-SS_CM-7-UNK:ICDContactAdapter 0 : 
ContactSessionChanged received for App FW contact 0, iefSourceContact is 
16777231 [15/1] (0)
1768: Nov 17 01:14:16.890 EST %MIVR-SS_CM-7-UNK:Ignoring 
contactSessionChanged because either the new or old session is null
1769: Nov 17 01:14:16.937 EST %MIVR-SS_TEL-7-UNK:Route Connection: 
[1800/(P1-jtapi_1_1) GCID=(1,15)->ACTIVE]->OFFERED, CTI Port selected: 
TP[id=1,implId=1801,state=IN_USE]
1770: Nov 17 01:14:16.968 EST %MIVR-SS_TEL-3-ROUTE_FAILED:Route failed: 
All Call ids=JTAPICallContact[id=0,implId=15/1,inbound=true,App 
name=ICD,task=null,session=501,seq 
num=0,cn=1800,dn=1800,cgn=3002,ani=null,dnis=null,clid=null,atype=DIRECT,lrd=null,ocn=1800,route=TR[num=1800],TP=null,List
 of Active Connections=[1800/(P1-jtapi_1_1) 
GCID=(1,15)->ACTIVE]->OFFERED,Extension=1801,Exception=com.cisco.jtapi.PlatformExceptionImpl:
 CCNException not handled: com.cisco.cti.client.CCNException: redirect 
failure,Failure reason= call will be rejected, 
CTIERR_REDIRECT_CALL_MEDIA_CONNECTION_FAILED=0x8ccc0036,Contact.Reject.reason=TRIGGER_FAIL,(SelectRouteTime,ObtainingIdleChannelTime,RedirectTime=125,47,31)
1771: Nov 17 01:14:16.968 EST 
%MIVR-SS_TEL-3-EXCEPTION:com.cisco.jtapi.PlatformExceptionImpl: CCNException 
not handled: com.cisco.cti.client.CCNException: redirect failure
1772: Nov 17 01:14:16.968 EST %MIVR-SS_TEL-3-EXCEPTION:at 
com.cisco.jtapi.ConnectionImpl.redirect(com/cisco/jtapi/ConnectionImpl)
1773: Nov 17 01:14:16.968 EST %MIVR-SS_TEL-3-EXCEPTION:at 
com.cisco.jtapi.ConnectionImpl.

Re: [OSL | CCIE_Voice] 6608 failed call

2008-11-18 Thread Pulos, Greg

Hi Erwan,

This dick tracy seems to show your T203 timer is reaching it's 'maximum time in 
seconds allowed without frames being exchanged between the endpoints of the 
link'. This is causing MGCP to delete the connection it had just created 10 
seconds before when attempting to setup the call.

By default, the T203 timer is 10 seconds.

A couple of things to note:

1) you may have more help determining the problem if by collecting CCM traces 
on the call.

2) you may want to verify your config of the PRI. Ie: framing, line encoding, 
speed of channels, channels to be used, top-down or bottom-up channel 
selection, etc.

3) if all config is 100% correct, perhaps there is some sort of delay on the 
D-channel causing the exchange of frames to take longer than 10 seconds. (the 
T203 timer value that is set) perhaps increasing this just a few seconds will 
allow more time to get the initial frames exchanged for the connection to 
succeed.
(a physical problem will likely not allow any increase in T203 to help)

To increase the T203 timer, use the following command on your D-channel 
interface of the PRI.

Ie:
interface S1/0:23
  isdn T203 15000 (or other msec value greater than the default 1)

greg

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erwan Erwan
Sent: Monday, November 17, 2008 4:03 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] 6608 failed call

hi,

I config the 6608 PRI , and it registered in CCM , however I can't make 
outgoing call. 7 digit and 10 digit

I capture debug from Tracy Tool.
Can anybody help me understand what is wrong from my debug ?



Thks



Re: [OSL | CCIE_Voice] Configuring a frame relay switch for lab

2008-11-10 Thread Pulos, Greg
You seem to have forgotten your frame-relay interface type which would be DCE 
on the FR switch side. (frame-relay intf-type dce)

Please see the following link for more info on setting up FR switch and setting 
DCE via the interface type:
http://www.cisco.com/en/US/docs/ios/12_2/wan/configuration/guide/wcffrely_ps1835_TSD_Products_Configuration_Guide_Chapter.html#wp1065161

Thank you.

greg
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Kagadis 
(kagadis.com)
Sent: Saturday, November 08, 2008 2:41 PM
To: OSL CCIE Voice Lab Exam
Subject: [OSL | CCIE_Voice] Configuring a frame relay switch for lab

I've been trying to set up a frame switch for my home lab and would like to 
know if anyone may be able to help.  None of my frame-relay PVCs are active and 
I'm wondering what I am doing wrong.

1) For the frame switch I am using a 2610 with c2600-ik9o3s3-mz.123-21.bin and 
three WIC-1DSU-T1 cards (one for each branch)
2) The branch routers are 2811 with WIC-1DSU-T1-V2
3) The connections are all crossover T1 and layer 1 is working


I can't seem to find any good documentation on setting this up properly, any 
wisdom is very much appreciated.

On Frame Relay switch

interface Serial0/0 (connected to HQ router)
 description HQ
 no ip address
 encapsulation frame-relay IETF
 no fair-queue
 service-module t1 clock source internal
 service-module t1 timeslots 1-24
 frame-relay lmi-type ansi
 frame-relay route 101 interface Serial1/0 103
 frame-relay route 102 interface Serial1/1 104
!
interface Serial1/0 (connected to BR1 router)
 description BR1
 no ip address
 encapsulation frame-relay IETF
 no fair-queue
 service-module t1 clock source internal
 service-module t1 timeslots 1-24
 frame-relay lmi-type ansi
 frame-relay route 103 interface Serial0/0 101
!
interface Serial1/1 (connected to BR2 router)
 description BR2
 no ip address
 encapsulation frame-relay IETF
 no fair-queue
 service-module t1 clock source internal
 service-module t1 timeslots 1-24
 frame-relay lmi-type ansi
 frame-relay route 104 interface Serial0/0 102




On HQ router

interface Serial0/1/0
 no ip address
 encapsulation frame-relay IETF
 no keepalive
 service-module t1 clock source internal
!
interface Serial0/1/0.1 point-to-point
 ip address 172.16.100.1 255.255.255.252
 snmp trap link-status
 frame-relay interface-dlci 101
!
interface Serial0/1/0.2 point-to-point
 ip address 172.16.100.5 255.255.255.252
 snmp trap link-status
 frame-relay interface-dlci 102



On BR1 router

interface Serial0/1/0
 no ip address
 encapsulation frame-relay IETF
 no keepalive
!
interface Serial0/1/0.1 point-to-point
 ip address 172.16.100.2 255.255.255.252
 snmp trap link-status
 frame-relay interface-dlci 103


On BR2 router

interface Serial0/0/0
 no ip address
 encapsulation frame-relay IETF
 no keepalive
!
interface Serial0/0/0.1 point-to-point
 ip address 172.16.100.6 255.255.255.252
 frame-relay interface-dlci 104








PVC Statistics for interface Serial0/0 (Frame Relay DTE)

  Active Inactive  Deleted   Static
  Local  0000
  Switched   0200
  Unused 0000

DLCI = 101, DLCI USAGE = SWITCHED, PVC STATUS = INACTIVE, INTERFACE = Serial0/0

  input pkts 564   output pkts 0in bytes 65930
  out bytes 0  dropped pkts 6   in pkts dropped 6
  out pkts dropped 0out bytes dropped 0
  in FECN pkts 0   in BECN pkts 0   out FECN pkts 0
  out BECN pkts 0  in DE pkts 0 out DE pkts 0
  out bcast pkts 0 out bcast bytes 0
  30 second input rate 0 bits/sec, 0 packets/sec
  30 second output rate 0 bits/sec, 0 packets/sec
  switched pkts 1
  Detailed packet drop counters:
  no out intf 0out intf down 0  no out PVC 0
  in PVC down 0out PVC down 6   pkt too big 0
  shaping Q full 0 pkt above DE 0   policing drop 0
  pvc create time 12:18:46, last time pvc status changed 00:30:22

DLCI = 102, DLCI USAGE = SWITCHED, PVC STATUS = INACTIVE, INTERFACE = Serial0/0

  input pkts 533   output pkts 0in bytes 62730
  out bytes 0  dropped pkts 10  in pkts dropped 10
  out pkts dropped 0out bytes dropped 0
  in FECN pkts 0   in BECN pkts 0   out FECN pkts 0
  out BECN pkts 0  in DE pkts 0 out DE pkts 0
  out bcast pkts 0 out bcast bytes 0
  30 second input rate 0 bits/sec, 0 packets/sec
  30 second output rate 0 bits/sec, 0 packets/sec
  switched pkts 0
  Detailed packet drop counters:
  no out intf 0out intf down 0  no out PVC 0
  in PVC down 0out PVC down 10  pkt too big 0
  shaping Q full 0 pkt above DE 0   policing drop 0
  pvc create time 12:18:31, last time pvc status changed 00:

Re: [OSL | CCIE_Voice] 6500 port issue

2008-10-07 Thread Pulos, Greg
Ipexpert should have this RMA'd and replaced with TAC.

This is indicative of a blade going bad if it is not dead for good already.

Sometimes a reset, reseat or complete reconfig of the blade will fix 
temporarily but I've seen this too many timesthat 6608 needs to be replaced 
asap!

greg

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Edi Hamlet
Sent: Monday, October 06, 2008 5:12 PM
To: James Key; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] 6500 port issue

this port is failure. I had same problem with you. I reported this to after 
hour support & they move me to new POD which i usually get, POD 21 which is the 
perfect one.

yesterday i got POD 23, all T1 ports for PSTN, Conf & Transcoder had problem 
and got same error message "Port host processor not online". I reported this 
but no one answer me. So that I couldn't play with my PSTN, CONF & XCODER at HQ 
site during all the time on may lab session.


- Original Message 
From: James Key <[EMAIL PROTECTED]>
To: ccie_voice@onlinestudylist.com
Sent: Tuesday, October 7, 2008 3:13:03 AM
Subject: [OSL | CCIE_Voice] 6500 port issue



I was on Pod 23 this weekend and ran into an issue with my conference bridge 
port on the 6500.  When I tried to assign it to the proper vlan, I would get 
this error message "Port host processor not online".  I tried disable and 
enable put it didn't resolve.  My transcoder and pstn ports were fine.  Has 
anyone else run into this and what is the best way to troubleshoot/resolve?



James Key





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Re: [OSL | CCIE_Voice] NM-HDV2 with PVDM2??

2008-09-22 Thread Pulos, Greg
The NM-HDV2 does not come with any PVDM2s on it.

It does have 4 slots for up to 4 PVDM2s to be added to it though.
(can have up to 4 PVDM2 (8-64) modules on the HDV2 card; or a mix of different 
PVDM2 modules)

Thank you.

greg

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WorkerBee
Sent: Sunday, September 21, 2008 11:23 PM
To: CCIE Voice Maillist
Subject: [OSL | CCIE_Voice] NM-HDV2 with PVDM2??

Does it comes with PVDM2 when I purchase this module?

http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a008039c333.shtml#table2

Thanks.



Re: [OSL | CCIE_Voice] CTI Super provider, when do we use it?

2008-07-16 Thread Pulos, Greg
Well, it is a valid option to use, in the scenarios as I described earlier so 
perhaps, as I've seen a few times, that specific TAC engineer was not so 
familiar with the option.
(actually, you'll rarely, if ever, see it checked in production as usually not 
all CTI devices need a single CTI application to exercise control over all of 
them)

Please see the following link for more info on 'CTI Super' provider:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/4_1_3/ccmsys/a08cti.html#wp1033158

Thank you.

greg
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Charles
Sent: Wednesday, July 16, 2008 2:10 PM
To: Pulos, Greg
Cc: OSL CCIE Voice Lab Exam
Subject: Re: [OSL | CCIE_Voice] CTI Super provider, when do we use it?

Cool, thanks...

So, why are we (via TAC) instructed not to use it?



Jonathan

On Wed, Jul 16, 2008 at 11:53 AM, Pulos, Greg <[EMAIL PROTECTED]> wrote:
> You use a CTI SUPER provider when you want that 'user' account to be able to 
> control ALL CTI devices. (not just the devices you assign under device 
> association)
>
> For example, PGUSER for IPCC Enterprise could be a super cti user if you only 
> had IPCC CTI devices defined. (as I do on my ipcc only ccm cluster)
>
> greg
>
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan 
> Charles
> Sent: Wednesday, July 16, 2008 12:35 PM
> To: OSL CCIE Voice Lab Exam
> Subject: [OSL | CCIE_Voice] CTI Super provider, when do we use it?
>
> In CCM 5/6 we have CTI control of all devices, and we assign users to
> that... I had TAC once say to never set anyone to CTI Super
> Provider...
>
> So, do we ever use it, and if so, when?
>
>
>
>
> Jonathan
>
>



Re: [OSL | CCIE_Voice] CTI Super provider, when do we use it?

2008-07-16 Thread Pulos, Greg
You use a CTI SUPER provider when you want that 'user' account to be able to 
control ALL CTI devices. (not just the devices you assign under device 
association)

For example, PGUSER for IPCC Enterprise could be a super cti user if you only 
had IPCC CTI devices defined. (as I do on my ipcc only ccm cluster)

greg

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Charles
Sent: Wednesday, July 16, 2008 12:35 PM
To: OSL CCIE Voice Lab Exam
Subject: [OSL | CCIE_Voice] CTI Super provider, when do we use it?

In CCM 5/6 we have CTI control of all devices, and we assign users to
that... I had TAC once say to never set anyone to CTI Super
Provider...

So, do we ever use it, and if so, when?




Jonathan



Re: [OSL | CCIE_Voice] CTI Super provider, when do we use it?

2008-07-16 Thread Pulos, Greg
You use a CTI SUPER provider when you want that 'user' account to be able to 
control ALL CTI devices. (not just the devices you assign under device 
association)

For example, PGUSER for IPCC Enterprise could be a super cti user if you only 
had IPCC CTI devices defined. (as I do on my ipcc only ccm cluster)

greg

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Charles
Sent: Wednesday, July 16, 2008 12:35 PM
To: OSL CCIE Voice Lab Exam
Subject: [OSL | CCIE_Voice] CTI Super provider, when do we use it?

In CCM 5/6 we have CTI control of all devices, and we assign users to
that... I had TAC once say to never set anyone to CTI Super
Provider...

So, do we ever use it, and if so, when?




Jonathan



Re: [OSL | CCIE_Voice] DSP problems on BR2

2008-05-21 Thread Pulos, Greg
In the config below, you have no sccp ccm group configured.

Also, you can use 'debug sccp' and 'debug dsp-resources flex all' to see what 
your resources are trying to do and whey their failing.

greg

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul and Bobs
Sent: Tuesday, May 20, 2008 5:49 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] DSP problems on BR2

My config is pasted below. Fro some reson when I enter dspfarm this command
disappears. and when i enter dspfarm transcode maximum session ? i get
<0-0>.

Once I get these configured how can i check to see that the dsp are
correctly configured and is there a command to see how many dsp are left nad
how many are confgiure for different services.

controller E1 0/0/0
 pri-group timeslots 1-3,16

voice-card 0
 dspfarm
 dsp services dspfarm

sccp local FastEthernet0/0.240
sccp ccm 10.4.202.1   priority 1
sccp

telephony-service
 max-ephones 30
 max-dn 30
 ip source-address 10.4.202.1   port 2000
 system message Your current options
 sdspfarm units 2
 sdspfarm transcode sessions 1
 sdspfarm tag 1 mtp00128031cca8
 time-zone 42
 time-format 24
 voicemail 4111
 max-conferences 8 gain -6
 call-forward pattern .T
 dn-webedit
 time-webedit
 secondary-dialtone 9



Re: [OSL | CCIE_Voice] Contact Center Express 5.02 VMWare

2008-04-23 Thread Pulos, Greg
I've just completed two call managers and unity on two new vmWare systems I 
built this week.

I'm going to put the IPCCX on there today and will provide you the results and 
steps when completed.

greg

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian
Sent: Tuesday, April 22, 2008 2:54 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Contact Center Express 5.02 VMWare

Any advice/directions on creating a Contact Center Express 5.0(2) VMWare box?

I likely won't be taking the CCIE Voice for 1 to 2 years so the lab I am 
building VMWare at work is UCM 6.x etc.

Thank you.

Brian Schear, CCVP
Unified Communications Engineer






Re: [OSL | CCIE_Voice] 6608 Incoming call troubleshooting

2008-04-11 Thread Pulos, Greg
You have to use the Dick Tracy tool.

Please see the following link for more information:

http://www.ciscopress.com/bookstore/product.asp?isbn=1587050757#info5

On this page, click the 'downloads' link under the More Inforation section.

There at the bottom of the page will be the link to the Dick Tracy tool.

Understand that this is officially not supported by cisco but it is a very 
useful tool once you understand how to work it and read the results.

greg

P.Si highly recommend the book Troubleshooting IP Telephony from 
ciscopress. Also I recommend the 'q.931 translator' tool.



-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Onur Tufekci
Sent: Friday, April 11, 2008 9:07 AM
To: CCIE Maillist
Subject: [OSL | CCIE_Voice] 6608 Incoming call troubleshooting

Hello everyone,

Is there a way to debug the incoming call on 6608 T1 module?

Thank you,

Onur.



Re: [OSL | CCIE_Voice] QoS on CM port

2008-04-08 Thread Pulos, Greg
The SRND states that allowing a TRUST from a server opens up the possibilities 
that a compromised server now has priority access, among other issues, as taken 
from the SRND below:

Servers-Certain servers, within the data center or otherwise, might be capable 
of correctly marking their traffic on their NICs. In such cases, the network 
administrator can choose to trust such markings. However, enforcing such a 
trust boundary requires cooperation between network administrators and system 
or server administrators, an alliance that is often fragile, at best, and 
usually involves considerable finger pointing. Additionally, network 
administrators should bear in mind that the majority of DoS/ worm attacks 
target servers. Infected servers not only might spew profuse amounts of traffic 
onto the network, but, in such cases, they might do so with trusted markings. 
There's no hard-and-fast rule that will apply to every situation. Some 
administrators prefer to trust certain servers, like Cisco CallManagers, due to 
the large number of ports that may be in use to provide services rather than 
administer complex access lists.

I'd rather do the work ourselves as engineers to do our best to alleviate the 
inherent problems that can (and surely will) arise with trusting a server.

Of course as it states, some admins would just as soon trust the server as it 
is much easier for them than doing the work up front themselves. I don't go for 
easyI go for secure and assured and error on the side of caution/safety.
(of course, if the lab tells you to trust the server, then you trust the 
server...that's the only time I will)

greg

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Charles
Sent: Monday, April 07, 2008 4:34 PM
To: Pulos, Greg
Cc: ccie_voice@onlinestudylist.com; Sven Hansen (svhansen)
Subject: Re: [OSL | CCIE_Voice] QoS on CM port

I disagree... the QoS SRND said to trust the CCM and Unity for DSCP...



Jonathan

On Mon, Apr 7, 2008 at 6:52 AM, Pulos, Greg <[EMAIL PROTECTED]> wrote:
>
>
>
>
> A general rule of thumb is to never trust a server, as a server can be
> compromised by a virus, worm or other malware thereby allowing an infected
> server to generate massive amounts of traffic that would always get priority
> and possibly causing everything from heavy uplink congestion to outright
> DoS.
>
>
>
> You should mark traffic from a server port respective to the type of
> traffic, ingress, that the server puts onto the wire; then police that
> traffic to DSCP 8 (scavenger), if not outright drop, when exceed-burst
> thresholds are exceeded.
>
>
>
>
>
> greg
>
>  
>
>
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Sven Hansen
> (svhansen)
>  Sent: Monday, April 07, 2008 7:38 AM
>  To: ccie_voice@onlinestudylist.com
>  Subject: [OSL | CCIE_Voice] QoS on CM port
>
>
>
>
>
>
> Hi all
>
>
> What sort of QoS is recommended for the CallManager port? Trusted server as
> per the SRND?
>
>
> Thanks
>
>
> Sven
>
>
>
>
>
>
>
>
>
> Sven Hansen
>  Systems Engineer
>  Sales / Channels
>
>  [EMAIL PROTECTED]
>  Phone :+27 (0) 11 267 1039
>  Mobile :+27 (0) 83 616 6216
>  Fax :+27 (0) 11 267 1100
>
>
>
>  Sub-Saharan Africa
>  www.cisco.com/global/ZA/
>
>
>
>
>
>
>
>
>
>
>
> This e-mail may contain confidential and privileged material for the sole
> use of the intended recipient. Any review, use, distribution or disclosure
> by others is strictly prohibited. If you are not the intended recipient (or
> authorized to receive for the recipient), please contact the sender by reply
> e-mail and delete all copies of this message.
>
>
>
>
>
>
>
>
>
>
>
>
>



Re: [OSL | CCIE_Voice] QoS on CM port

2008-04-07 Thread Pulos, Greg
A general rule of thumb is to never trust a server, as a server can be 
compromised by a virus, worm or other malware thereby allowing an infected 
server to generate massive amounts of traffic that would always get priority 
and possibly causing everything from heavy uplink congestion to outright DoS.

You should mark traffic from a server port respective to the type of traffic, 
ingress, that the server puts onto the wire; then police that traffic to DSCP 8 
(scavenger), if not outright drop, when exceed-burst thresholds are exceeded.

greg

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sven Hansen 
(svhansen)
Sent: Monday, April 07, 2008 7:38 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] QoS on CM port

Hi all
What sort of QoS is recommended for the CallManager port? Trusted server as per 
the SRND?
Thanks
Sven

[cid:image001.gif@01C89884.62F8B2B0]


Sven Hansen
Systems Engineer
Sales / Channels

[EMAIL PROTECTED]
Phone :+27 (0) 11 267 1039
Mobile :+27 (0) 83 616 6216
Fax :+27 (0) 11 267 1100



Sub-Saharan Africa
www.cisco.com/global/ZA/







This e-mail may contain confidential and privileged material for the sole use 
of the intended recipient. Any review, use, distribution or disclosure by 
others is strictly prohibited. If you are not the intended recipient (or 
authorized to receive for the recipient), please contact the sender by reply 
e-mail and delete all copies of this message.



[cid:image003.gif@01C89884.62F8B2B0]




<><><>

Re: [OSL | CCIE_Voice] [cisco-voip] SIlly QoS Question

2008-04-07 Thread Pulos, Greg
Just to clarify:

AFx3 has a higher drop preference than AFx1, not the other way around.

eg: AFx1 has a lower drop preference than AFx3 as defined in rfc2597.
(the drop preference of assured forwarding is 1-3; 1 being lowest, 3 being 
highest)

We can manipulate the 'randomness' of WRED by assigning higher drop preferences 
to the packets such as in the case of using policing to remark an AF21 packet 
to AF23 if we would like it to be dropped more aggresively in preference of 
other data types such as AF31 for example.

CS3 does not have a higher drop probability in the cases where you remark AF31 
to AF32/AF33.

greg

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Charles
Sent: Sunday, April 06, 2008 12:10 PM
To: Peter Smale
Cc: CCIE Maillist; [EMAIL PROTECTED]; Pulos, Greg
Subject: Re: [OSL | CCIE_Voice] [cisco-voip] SIlly QoS Question

My working assumption is that any lost packets are bad... and only
should be experienced with wireless...

My question was more to my own understanding of CS3 vs. AD31, and why
Cisco would decide to change it to something that appears to have a
higher drop probability...


Jonathan

On Sun, Apr 6, 2008 at 11:04 AM, Peter Smale <[EMAIL PROTECTED]> wrote:
> I think there are only 3 drop thresholds on 3550 very primitive RED
>  C1 is better than CS2 which is better than CS3 etc.   But within a class as
>  for as RED or WRED the opposite is true.
>  AD31 is dropped before AF32  etc.   Traffic in CS3 should not be eligible
>  for drop at all.
>  Been a long time since I have taught QoS but that's how I remember it.
>
>  I made the argument sometime ago that control traffic SKINNY RS323 SIP etc.
>  is so negligible that it could just be classified with the RTP stream itself
>  CS2.
>
>  But I guess that would make things too easy.
>
>  By the way if you are dropping packets on a 100 or even 10 meg link in a
>  short Range campus i.e. 10 millisecond range something has gone very wrong.
>  Broadcast storm, virus etc.  in any event you've got a problem to fix.
>  Peter.  :-)
>
>
>
>  -Original Message-
>  From: [EMAIL PROTECTED]
>  [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Charles
>  Sent: Saturday, April 05, 2008 5:46 PM
>  To: Pulos, Greg
>  Cc: CCIE Maillist; [EMAIL PROTECTED]
>  Subject: Re: [cisco-voip] [OSL | CCIE_Voice] SIlly QoS Question
>
>  So, if CS3 has a higher drop probability than AF31, why change the CS3?
>
>  It seems to me that it would be good to get dial-tone when I pick up
>  the receiver, even if there is congestion.
>
>  This seems like a bad idea...
>
>
>
>  Jonathan
>
>  On Sat, Apr 5, 2008 at 6:47 PM, Pulos, Greg <[EMAIL PROTECTED]> wrote:
>  > I think it sounds a bit confusing too but the concept of rfc2474 defining
>  the CSx as 'not subject to dropping' is correct. (although in fact the
>  taildrops can/will drop cs3 if absolutely required by the qos design and
>  traffic requirements.)
>  >
>  >  I think the bigger point that is being made by the migration is to allow
>  for MissionCriticalData to be assigned af31period. (therefore allowing
>  greater drop preference scalability, ie: AFx(1-3), depending on the qos
>  design and exact traffic type)
>  >
>  > greg
>  >
>  >
>  >
>  >  -Original Message-
>  >  From: [EMAIL PROTECTED]
>  [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
>  Charles
>  >  Sent: Saturday, April 05, 2008 6:11 PM
>  >  To: CCIE Maillist; [EMAIL PROTECTED]
>  >  Subject: [OSL | CCIE_Voice] SIlly QoS Question
>  >
>  >
>  >
>  > [cross-posting to Puck and the CCIE Online Study List...if you guys
>  >  aren't familiar with each other, you should be...]
>  >
>  >
>  >  Ok, so Cisco is migrating control traffic from AF31 to CS3...
>  >
>  >  I am curious as to why... the QoS SRND says they are moving to CS3
>  >  "because Class Selector code points, as defined in RFC 2474, were not
>  >  subject to markdown/aggressive dropping"
>  >
>  >  OK, I guess...
>  >
>  >  But.
>  >
>  >  WRED is enabled on each non-priority queue on the Cisco 3550 (queues
>  >  1,2 and 3)...
>  >
>  >  Doesn't the lack of any drop thresholds (low, medium and high) in the
>  >  CSX PHB levels, cause this traffic to be potentially dumped in the
>  >  case of congestion?
>  >
>  >  In other words, since AF31 has a binary value of 011010 and CS3 has a
>  >  binary value of 011000, that means that CS3 has nothing set for Delay,
>  >  Throughput or Reliability; whereas AF31 had high throughput set.
>  >
>  >  How is CS3 better?
>  >
>  >
>  >
>  >  Jonathan
>  >
>  >
>  ___
>  cisco-voip mailing list
>
> [EMAIL PROTECTED]
>  https://puck.nether.net/mailman/listinfo/cisco-voip
>
>



Re: [OSL | CCIE_Voice] SIlly QoS Question

2008-04-05 Thread Pulos, Greg
I think it sounds a bit confusing too but the concept of rfc2474 defining the 
CSx as 'not subject to dropping' is correct. (although in fact the taildrops 
can/will drop cs3 if absolutely required by the qos design and traffic 
requirements.)

I think the bigger point that is being made by the migration is to allow for 
MissionCriticalData to be assigned af31period. (therefore allowing greater 
drop preference scalability, ie: AFx(1-3), depending on the qos design and 
exact traffic type)

Gregory T. Pulos II
Sr. VoIP Engineer
U.S. Department of Commerce
Office of the Secretary
OCIO/ONTO/NOC
202.482.5010
[EMAIL PROTECTED]

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Charles
Sent: Saturday, April 05, 2008 6:11 PM
To: CCIE Maillist; [EMAIL PROTECTED]
Subject: [OSL | CCIE_Voice] SIlly QoS Question

[cross-posting to Puck and the CCIE Online Study List...if you guys
aren't familiar with each other, you should be...]


Ok, so Cisco is migrating control traffic from AF31 to CS3...

I am curious as to why... the QoS SRND says they are moving to CS3
"because Class Selector code points, as defined in RFC 2474, were not
subject to markdown/aggressive dropping"

OK, I guess...

But.

WRED is enabled on each non-priority queue on the Cisco 3550 (queues
1,2 and 3)...

Doesn't the lack of any drop thresholds (low, medium and high) in the
CSX PHB levels, cause this traffic to be potentially dumped in the
case of congestion?

In other words, since AF31 has a binary value of 011010 and CS3 has a
binary value of 011000, that means that CS3 has nothing set for Delay,
Throughput or Reliability; whereas AF31 had high throughput set.

How is CS3 better?



Jonathan



Re: [OSL | CCIE_Voice] CS3 or AF31

2008-03-31 Thread Pulos, Greg
I believe cisco has made up their minds on this matter.

They are migrating to CS3 for call signaling.

Currently a lot of devices mark AF31 but all new devices are to be marking CS3. 
(I'm guessing the 79x2s mark this way, for example, although I've not gotten my 
hands on any to verify)

I have not found a defined date that states when migration will be completed so 
just follow the rules they provide...CS3/AF31 should be marked CS3 and any 
mission-critical-data should be marked DSCP 25 until otherwise informed.
(in the cases of advanced model policing in which some FE ports can only have 8 
policers max such as with the 3550, then you would combine the softphone 
signaling with the mission-critical-data access-list to allow for a single 
policer for both) as defined in the qos srnd.

greg

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Charles
Sent: Sunday, March 30, 2008 11:51 AM
To: CCIE Maillist
Subject: [OSL | CCIE_Voice] CS3 or AF31

Which one?

I only ask, because I am seeing a lot of conflicting documentation on
Cisco's website.

One says that all IP Phones mark as AF31, but CCM, etc will mark as CS3...

However, I saw a note in the QoS SRND that all markings will migrate
to CS3 (when?)

It probably doesn't much matter as I am doing a match-any on my
class-maps, but it seems that Cisco hasn't quite made up their minds
yet...



Jonathan



Re: [OSL | CCIE_Voice] WAN QOS question

2008-03-26 Thread Pulos, Greg
You should use the voice bandwidth formula as stated in the QoS SRND as follows:

Voice bandwidth = ((voice payload) + (40byte ip/udp/rtp header)) * 8 * (PPS) + 
((L2 header) * 8 * (PPS))

L2 Headers = dot1q - 32 bytes; ppp - 12 bytes; mlppp - 13 bytes; frameRelay - 4 
bytes; FRF.12 - 8 bytes
PPP = 50pps for 20ms sampling; 33pps for 30 ms sampling

Makes your table look like the following: (assuming 20ms sample)(round bw up to 
nearest kbps)



g711

g729

PPP

84800bps

28800bps

MLP

85200bps

29200bps

Frame w/FRF.12

83200bps

27200bps




greg

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jason sung
Sent: Tuesday, March 25, 2008 8:32 PM
To: CCIE Maillist
Subject: [OSL | CCIE_Voice] WAN QOS question

I am trying to understand QOS over WAN and want to make sure my math is correct 
as well.

I am trying to calculate 4 g711/g729 calls over a WAN link with layer2 overhead.



g711

g729

PPP

84x4=336

37x4=148

MLP

86x4=344

30x4=120

Frame w/FRF.12

84x4=336

28x4=112


Can anyone comment on this?

Thanks in advance.


[OSL | CCIE_Voice] CCM Registraion

2008-03-17 Thread Pulos, Greg
what server should be the registration server? sub only right??

the workbook states no pub exists (although a pub does exist...ie: 
10.x.200.21), so i assume the sub should be the registraton server.

no phones register although many 'transient connection' error msgs in the event 
viewer...reason code 6 (which is a 'call manager reset')

i can get over 4000 phones on 80+ vlans in 65+ closets to register to multiple 
clusters in the office but i can't get a single phone on the same vlan as the 
ccm to register on ipexpert?!?!

both call managers can ping all phones at BR1, BR2 and HQ

ccmsub control center services:
Cisco CallManager   Activated
Cisco Tftp  Activated

br1-rtr config:
interface FastEthernet0/0
 no ip address
 shutdown
 duplex auto
 speed auto
!
interface FastEthernet0/1
 no ip address
 shutdown
 duplex auto
 speed auto
!
interface Serial0/1/0
 no ip address
 encapsulation frame-relay IETF
 no fair-queue
 frame-relay lmi-type ansi
!
interface Serial0/1/0.1 point-to-point
 ip address 162.6.101.2 255.255.255.0
 ip ospf mtu-ignore
 frame-relay interface-dlci 101
!
interface FastEthernet1/0
 description PHONE
 switchport access vlan 160
 switchport trunk native vlan 160
 switchport mode trunk
 switchport voice vlan 260
!
interface FastEthernet1/1
!
interface FastEthernet1/2
!
interface FastEthernet1/3
!
interface FastEthernet1/4
!
interface FastEthernet1/5
!
interface FastEthernet1/6
!
interface FastEthernet1/7
!
interface FastEthernet1/8
 description PHONE
 switchport access vlan 160
 switchport trunk native vlan 160
 switchport mode trunk
 switchport voice vlan 260
!
interface FastEthernet1/9
!
interface FastEthernet1/10
!
interface FastEthernet1/11
!
interface FastEthernet1/12
!
interface FastEthernet1/13
!
interface FastEthernet1/14
!
interface FastEthernet1/15
!
interface Vlan1
 no ip address
 shutdown
!
interface Vlan160
 description DATA VLAN
 no ip address
!
interface Vlan260
 description VOICE VLAN
 ip address 10.6.201.1 255.255.255.0
 ip helper-address 10.6.200.21


hq-rtr config:
interface FastEthernet0/0
 no ip address
 duplex auto
 speed auto
!
interface FastEthernet0/0.160
 description DATA VLAN
 encapsulation dot1Q 160 native
 no snmp trap link-status
!
interface FastEthernet0/0.260
 description VOICE VLAN
 encapsulation dot1Q 260
 ip address 10.6.200.3 255.255.255.0
 no snmp trap link-status
!
interface FastEthernet0/1
 no ip address
 shutdown
 duplex auto
 speed auto
!
interface Serial0/1/0:0
 no ip address
 encapsulation frame-relay IETF
 frame-relay lmi-type ansi
!
interface Serial0/1/0:0.1 point-to-point
 ip address 162.6.101.1 255.255.255.0
 ip ospf mtu-ignore
 frame-relay interface-dlci 201
!
interface Serial0/1/0:0.2 point-to-point
 ip address 162.6.102.1 255.255.255.0
 ip ospf mtu-ignore
 frame-relay interface-dlci 202


6500 config:
 2/34 on   dot1q  trunking  160

2/31 POD16-HQ-PHONE1  connected  160a-full   a-100 10/100BaseTX
 2/32 POD16-HQ-PHONE2  connected  160a-full   a-100 10/100BaseTX
 2/33 POD16-ATA-186connected  260a-halfa-10 10/100BaseTX
 2/34 POD16-HQ-RTR connected  trunk  a-full   a-100 10/100BaseTX
 2/35 POD16-CCM-UNITY  connected  260  full 100 10/100BaseTX

Port  Name Status Vlan   Duplex Speed   Type
-  -- -- -- --- 
 2/32 POD16-HQ-PHONE2  connected  160a-full   a-100 10/100BaseTX
Port  AuxiliaryVlan AuxVlan-Status
- - --
 2/32 260   active
Port  InlinePowered  PowerAllocated   Device IEEE class
From PS   To PD
  Admin  Oper   mWattsmWatts
- -- -- ---   --- -- --
 2/32 auto   on 6300  6300cisco  none
Port  MaximumPower  ActualConsumption
  mWattsmWatts
-   -
 2/32 7000  6300

(other phone the same)

any advice??? thank you.


From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of senthil natarajan [EMAIL 
PROTECTED]
Sent: Sunday, March 16, 2008 8:18 AM
To: CCIE Maillist
Subject: [OSL | CCIE_Voice] Proctorlabs.com not available

Can anyone access proctorlabs.com now? I have been 
trying for the past 17 min and its still down?

-Senthil


Re: [OSL | CCIE_Voice] 3550 QoS : Policing and Marking

2008-03-16 Thread Pulos, Greg
Is proctorlabs.com currently down?? all other internet sites are available but 
proctorlabs keeps getting cannot display webbapge? If so, do we get our session 
back if this site is unavailable when we attempt to access it?

Also, it seems IP Blue does not work with Vista, is this correct or is there 
another compatible IP Blue client we can use?

greg


Re: [OSL | CCIE_Voice] gatekeep registration

2008-03-14 Thread Pulos, Greg
You can use the 'debug h225 asn1' and 'debug ras' to determine the processes 
and failures of your GK registration. (maybe one at a time since the h225 debug 
provides a lot of output alone)

This will help point you to the cause of the problem.

Thank you.

Gregory T. Pulos II
Sr. VoIP Engineer
U.S. Department of Commerce
Office of the Secretary
OCIO/ONTO/NOC
[EMAIL PROTECTED]

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Friday, March 14, 2008 7:34 AM
To: Jonathan Charles; ccievoice1
Cc: ccie_voice@onlinestudylist.com; [EMAIL PROTECTED]
Subject: Re: [OSL | CCIE_Voice] gatekeep registration

ya i am sure, could it be something wrong with the vmware? i was using pod31. 
follow exactly the proctor guide


Jonathan Charles <[EMAIL PROTECTED]> wrote:
As soon as you create a GK controlled trunk, the trunk should register
to the GK, you sure you have your zone and type prefix correct?



Jonathan

On Thu, Mar 13, 2008 at 7:15 PM, ccievoice1 wrote:
> Well, CallManager will not register the endpoints to Gatekeeper.
>
> However you can use alias static to manually add-in the CallManager's
> endpoints to Gatekeeper. While, a H323 gateway is able auto registered its
> endpoints to gatekeeper.
>
> On Fri, Mar 14, 2008 at 8:05 AM, wrote:
> > Hi all,
> >
> > I am doing the lab task 5.1, followed the proctor guide, configured the
> following at the hq-rtr
> > gatekeeper
> > zone local HQ-RTR ipexpert.com 172.31.100.1
> > zone remote PSTN-WAN ipexpert.com 10.31.200.2 1719
> > gw-type-prefix 1#* default-technology
> > bandwidth remote 144
> > no shutdown
> > and added a gatekeeper and h225 gatekeeper controlled trunk.
> > however when i do show gatekeeper endpoint
> > there are no endpoint registered at gatekeeper.
> > am i missing anything?
> >
> > Sara
> >
> >
> >
> >
> >
> > 
> Easy + Joy + Powerful = Yahoo! Bookmarks x Toolbar
> >
>
>







Easy + Joy + Powerful = Yahoo! Bookmarks x 
Toolbar


Re: [OSL | CCIE_Voice] ipcc scripting

2007-11-28 Thread Pulos, Greg
Oops..

Forgot the link I mentioned in the last email:

http://cisco.com/application/pdf/en/us/guest/products/ps7255/c1667/ccmigration_09186a00807fc027.pdf

Gregory T. Pulos II
Sr. VoIP Engineer
[EMAIL PROTECTED]

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of sean hurricane
Sent: Sunday, November 25, 2007 5:51 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] ipcc scripting

i get the following error message when i debug the attached script.
The error message is

Accept Step: Trigger is not a ContactApplication Trigger


hence a hear a system error message when i call into my RP.


any ideas?


Re: [OSL | CCIE_Voice] ipcc scripting

2007-11-28 Thread Pulos, Greg
The error "Accept Step: Trigger is not a ContactApplicationTrigger" is OK.

This is only a message stating the first trigger is not a CAT, which is normal 
for most IVR scripts. You likely have no contacts defined other than the system 
default, which is normal in most CRA installs.
(please see the following link for more info:

The problem of hearing a system error message when calling your RP could be a 
few things. Most likely it is the IVR script either named incorrectly in the 
IVR vs how it's named in ICM or the script has an error or it needs to be 
reloaded to the CRA engine via the script manager. If the only error you get 
when debugging your IVR script is the Trigger error, then the script is most 
likely OK.

I'm attempting to review the .aef although my CRA editor will not open it. I'll 
see if it is a file issue or a java issue and provide you more status on the 
file if I get any.

Also, sometimes when the IVR provides the system error message, it is due to 
the CRA engine and/or the scripts.

Do a refresh of the ICD.aef script and/or just delete/re-add to the script 
manager. Also, do not forget to refresh the applications the scripts are tied 
too.

Last but not least, verify your ICM script that calls the IVR functions. Check 
for .aef name in the RunExternalScript node. Reload the script in ICM and 
verify any connection problems to the script and do a ICM script check and also 
verify the active script.

Thank you.

gtp
Sr. VoIP Engineer

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of sean hurricane
Sent: Sunday, November 25, 2007 5:51 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] ipcc scripting

i get the following error message when i debug the attached script.
The error message is

Accept Step: Trigger is not a ContactApplication Trigger


hence a hear a system error message when i call into my RP.


any ideas?