I am hitting the bug as explained above. I removed the dial-plan and it
worked. I added the dialplan and changed the extension in CUE to OTHER
(e.164 number) and it worked.
Thanks guys
On Thu, Apr 3, 2008 at 4:27 PM, Erick Bergquist <[EMAIL PROTECTED]> wrote:
> Does the user in CUE have the righ
Correction, 20ms, you are right bad at math
Jonathan
On Thu, Apr 3, 2008 at 12:42 AM, Jonathan Charles <[EMAIL PROTECTED]> wrote:
> Serialization delay is a function of how long it will take to get X
> bits onto the wire at speed Y
>
> And is simply X/Y
>
> So, a 1500-byte ping will take
Serialization delay is a function of how long it will take to get X
bits onto the wire at speed Y
And is simply X/Y
So, a 1500-byte ping will take 214ms to be put on the wire at a line
speed of 56 bps
(1500*8)/56000 = .214
So, you list the speed of the circuit 768000
The fragment size
Does the user in CUE have the right extension configured that the call
is coming from? I've seen that be the issue before and changing it
and/or also setting the e.164 number to where it is coming from gets
it going. You could also briefly get rid of the dialplan pattern
command if it is doing som
http://www.cisco.com/en/US/products/sw/voicesw/ps5520/products_field_notice09186a008023cfe2.shtml
I don't think it applies...
But it might...
Jonathan
On Wed, Apr 2, 2008 at 9:36 PM, Paul and Bobs <[EMAIL PROTECTED]> wrote:
> self enroll
>
> I see from another reply it could be a bug with
Did you have the command 'dcm-manager music-on-hold' ?
You still need that one.
Mark Snow
Sr Technical Instructor
IPexpert, Inc.
Sent from my iPhone
On Apr 2, 2008, at 9:23 PM, "jason sung" <[EMAIL PROTECTED]> wrote:
Not SRST, I meant an actual H323 gateway.
Yes your assumption is correct,
self enroll
I see from another reply it could be a bug with the dial plan
I will look into that
Thanks
On Thu, Apr 3, 2008 at 11:26 AM, Jonathan Charles <[EMAIL PROTECTED]> wrote:
> What do you mean by 'access the mailbox'? Do you mean try to leave a
> message or try to self-enroll?
>
>
>
> Jo
Actually, there are a few 'minor' commands you are not allowed to
change/configure and they are listed on the first page of the exam.
Otherwise, the lab questions will let you know whether or not you can do
certain things.
Read the entire exam carefully!
On Wed, Apr 2, 2008 at 12:25 AM, Jonathan
??? - I don't understand. You can configure LFI either can't you?
What is the difference below?
1)
policy-map shape
class class-default
shape average 729600 7296 0
map-class frame-relay frf12
frame-relay fragment 960
service-policy out shape
2)
map-class frame-relay frf12
frame-relay frag
One reason you might want to use FRF.12 over legacy FRTS is because you can
fragment the packets.
On Wed, Apr 2, 2008 at 8:28 PM, Scott Monasmith <[EMAIL PROTECTED]> wrote:
> Does it matter if you use FRTS or legacy FRTS with FRF.12? Is there really
> any difference? Why would you use one over
Can someone confirm my calculations? I am trying to calculate delay based on
port speed and CIR.
Port speed: 768
CIR: 384
fragment size = 768 *10/8 = 960
delay = fragment size *8/ CIR
= 960*8/384
= 20 ms
20 ms sounds too high though
TIA
Does it matter if you use FRTS or legacy FRTS with FRF.12? Is there really
any difference? Why would you use one over the other?
--
"There are only 10 types of people in the world: Those who understand
binary, and those who don't"
Not SRST, I meant an actual H323 gateway.
Yes your assumption is correct, this is MoH to the PSTN PRI trunk.
Here is what I did.
Since it worked with MGCP, i configured siteB router as MGCP and pulled CCM
trace.
Next I wiped out MGCP and configured as H323 gateway to compare CCM traces,
but now
It wouldn't work - Multicast doesn't allow transcoders.
Are you in SRST fallback when you are stating that you are in H323?
Also I assume this is MoH to the PSTN PRI Trunk - since IP phones
speak SCCP not MGCP or H323 - am I correct on that assumption?
Anyway - next test you would need to do
dialplan pattern bug. have you verified that???
On Wed, Apr 2, 2008 at 7:23 PM, Paul and Bobs <[EMAIL PROTECTED]> wrote:
> Hi
>
> I am having an issue with my CUE. Whenever I setup a new mailbox for a
> user and try to access the mailbox it is saying that I can only access it
> from the primary e
Region is set to 711,
streaming service is set to 711 and 729.
729 works no 711.
If i change the streaming service to 711 only, N0 MOH
On Wed, Apr 2, 2008 at 7:25 PM, Jonathan Charles <[EMAIL PROTECTED]> wrote:
> Well, wait
>
> The region is set to 711-only, the MOH is set to 711-only, the
> v
What do you mean by 'access the mailbox'? Do you mean try to leave a
message or try to self-enroll?
Jonathan
On undefined, Paul and Bobs <[EMAIL PROTECTED]> wrote:
> Hi
>
> I am having an issue with my CUE. Whenever I setup a new mailbox for a user
> and try to access the mailbox it is saying t
Well, wait
The region is set to 711-only, the MOH is set to 711-only, the
voice-class codec is set to 711 as first preference, but 729 works...
What is the region for the base voice stream?
Jonathan
On undefined, jason sung <[EMAIL PROTECTED]> wrote:
> This is weird, I set the voice class to
This is weird, I set the voice class to use 711 as 1st preference and both
codecs work
I set it back to the way I had it g729 1st and 711 stops working.
I thought codec would be negotiated on voice-class codec. no?
I am going to erase and do everything over.
On Wed, Apr 2, 2008 at 7:02 PM, Jonat
Hi
I am having an issue with my CUE. Whenever I setup a new mailbox for a user
and try to access the mailbox it is saying that I can only access it from
the primary extension for security reasons thens hangs up.
This happens on all new mailboxes.
Paul
Nope - doesn't work that way - read the rest of the thread that came
in a few minutes ago :)
--
Mark Snow
CCIE #14073 (Voice, Security)
CCSI #31583
Senior Technical Instructor - IPexpert, Inc.
A Cisco Learning Partner - We Accept Learning Credits!
Telephone: +1.810.326.1444
Fax: +1.309.413.40
Indeed - take a break! it can sometimes be just as important as hard
study.
I knew you knew that one anyway ;-)
--
Mark Snow
CCIE #14073 (Voice, Security)
CCSI #31583
Senior Technical Instructor - IPexpert, Inc.
A Cisco Learning Partner - We Accept Learning Credits!
Telephone: +1.810.326.144
yes preference 1 is g729
On Wed, Apr 2, 2008 at 7:02 PM, Jonathan Charles <[EMAIL PROTECTED]> wrote:
> A voice-class codec on the dial peer would do it, with no G.711 on
> it... What are the codec preferences?
>
> What is preference 1? Is it G.729a?
>
>
>
>
>
> Jonathan
>
> On undefined, jason su
A voice-class codec on the dial peer would do it, with no G.711 on
it... What are the codec preferences?
What is preference 1? Is it G.729a?
Jonathan
On undefined, jason sung <[EMAIL PROTECTED]> wrote:
> Ok, I eliminated the 1st option by only allowing g711. (removed g729 under
> ip streamin
I am confused by this statement:
"I have the
gateways expecting 10 digits on inbound calls and have
external phone number masks assigned to make the phone
numbers 10 digits long. "
If your gateways are sending 10 digits, and you are expecting 10
digits, and your phones have 4 digits, then you eit
Ok, I eliminated the 1st option by only allowing g711. (removed g729 under
ip streaming services)
2nd option you mention. How would that happen?
On Wed, Apr 2, 2008 at 6:57 PM, Jonathan Charles <[EMAIL PROTECTED]> wrote:
> And we know from Mark Snow that the region setting only says 'this is
>
And we know from Mark Snow that the region setting only says 'this is
the MAXIMUM bandwidth' we can use, so if it set to G.711, then G.729
is still available for use.
If you want it to use G.711 for MoH disable G.729 under the IP Voice
Media Streaming App Service Parameter
Or.
Your H.323 destina
Mark,
You are a genius, labbing too often can actually make
you less effective if you don't rest and let it sink
in. Thanks again.
--- Mark Snow <[EMAIL PROTECTED]> wrote:
> External Number Mask is only used on Outgoing calls
> to make the DN
> look like its full DID if the checkbox is checked
External Number Mask is only used on Outgoing calls to make the DN
look like its full DID if the checkbox is checked on the outgoing
Route Pattern (and for AAR purposes) - but it will not get a call
inbound from a PSTN or any other GW or Trunk to route the call to that
DN.
You must either u
All,
I have been getting alot better with voice while using
the proctorlabs setup but tonight is making me
question what I am doing! :) I can't seem to get
calls to work from the PSTN (IP Blue) into BR1 or HQ.
I have the gateways configured correctly and have
inbound to outbound calls working fi
I have a small problem.
My multicast MOH works fine at the Remote site using MGCP, but fails to use
g711 at remote site when I use H323.
CallManager configurations stay the same. When I switch from H323 to MGCP
everything works fine, if I switch back to H323, MOH works only for g729 and
not g711.
Which raises another question.
Since the labs use a web link to UniverCD and UniverCD is a complete
disaster of dead links... are they now migrating the labs to the new
Doc location?
Jonathan
On Wed, Apr 2, 2008 at 12:35 PM, Mark Snow <[EMAIL PROTECTED]> wrote:
> Well - without giving you the
Then I would need to know what the specific Task is asking in order to
answer your question.
Please respond with a Task as it might be given to your - possibly one
from our WB or one you have come up with yourself.
Again - there is no answer without know what the task was to begin
with! :)
Yes. For the LAB preparation.
Gustavo Sánchez
From: Mark Snow [mailto:[EMAIL PROTECTED]
Sent: Miércoles, 02 de Abril de 2008 12:40 p.m.
To: Sanchez Galarza, Gustavo - (Col)
Cc: ccie_voice@onlinestudylist.com; [EMAIL PROTECTED]
Subject: Re: [OSL | CCI
AAR and IPMA are inherently incompatible in CUCM 4.1 - as is AAR with
ANY CTI Route Points.
--
Mark Snow
CCIE #14073 (Voice, Security)
CCSI #31583
Senior Technical Instructor - IPexpert, Inc.
A Cisco Learning Partner - We Accept Learning Credits!
Telephone: +1.810.326.1444
Fax: +1.309.413.409
In real life or for CCIE Lab preparation?
Because the answers to which you are configuring for are VERY
different. :)
For CCIE Lab preparation - you are to ONLY do whatever the lab tells
you to do for a given task - so I have no quick answer to your
question - it would completely depend on t
I am not at all sure what your question is - or really what it relates
to other than something about dialing out an E1 and something about SIP.
Clarify please?
--
Mark Snow
CCIE #14073 (Voice, Security)
CCSI #31583
Senior Technical Instructor - IPexpert, Inc.
A Cisco Learning Partner - We Acce
Well - without giving you the answer (learn to fish method :) - how
could you find out yourself?
Remember being resourceful is being a CCIE.
Hint - Test it yourself - or if you don't have immediate access to a
pod - then http://cisco.com/univercd and lookup the Cmd Reference to
see what tha
Is that a hunt timer or a timeout for the entire hunt?
Jonathan
On Wed, Apr 2, 2008 at 4:48 AM, ccievoice1 <[EMAIL PROTECTED]> wrote:
> I will customized the timeout value, as default is 180s
>
>
> !
> ephone-hunt 1 sequential
> pilot 3111
> list 3100,3101
> final
> timeout 6
> !
>
> HTH
>
I will customized the timeout value, as default is 180s
!
ephone-hunt 1 sequential
pilot 3111
list 3100,3101
final
timeout 6
!
HTH
On Wed, Apr 2, 2008 at 2:03 PM, Paul and Bobs <[EMAIL PROTECTED]> wrote:
> Howdy
> When creating hunt groups and haveing the hunt group hunt to phone 1 on
> s
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