I'm having a hard time when an internal extension calls another internal
extension that uses SNR, the From phone number on the PSTN phone is 4 digits
instead of 7. For example, extension 2001 calls 2003, and 2003 simultaneously
rings a PSTN phone number. The display on the PSTN phone says
Hi Miron,
Thanks for this.
There is even more:
Enable for Supervised Transfers - Check this check box so that Cisco Unity
Connection uses DTMF to detect and reject calls that have been transferred to
another extension (by using supervised transfer) and that have been transferred
Hi Mark,
The EPNM does not work at RP for SNR. Have you try to set EPNM at RL level?
TN.
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For more information regarding industry leading CCIE Lab training, please visit
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http://chikkis.blogspot.com/2009/10/installing-cucm-7-pub-and-sub-on-vmware.html
This link will help you get thru.
you can also go to you tube there are lots of videos on you tube, that will
help you complete your installation successfully.
Pithog oil
If I'm following your example correctly, Mark, then you aren't hitting on
the translation pattern.
The SNR call is matching the \+1408.6347694 RP, to go out, why would it be
hitting the translation pattern? Perhaps you meant to configure this as a
Calling Party Transformation?
mike
On
Just hit the same problem in Vol2 Lab4 and I can confirm that this doesn't work
at the RP level but does work at the RL level. Is this a known bug ?
Graham
On 1 Oct 2010, at 13:35, Tam Nhu wrote:
Hi Mark,
The EPNM does not work at RP for SNR. Have you try to set EPNM at RL level?
TN.
Graham, same thing here.
This is a summary of what I've done to get it working correctly. I eliminated
using Translation Profiles as I didn't find them necessary for this.
Create PT_SNR which is assigned to CSS_SNR
Create a Remote Destination Profile and assign CSS_SNR to both Calling Search
Sorry, I meant Translation Patterns, not Profiles. Still working on the From
number presentation. I'm assuming that if HQ1 calls HQ3 the PSTN phone should
show a 7 digit From number, but if BR1 calls 2003 the PSTN should show a 10
digit From number. Would you guys agree?
On Oct 1, 2010,
hi all,
is anyone know proctor rack has CUPC client installed in the PC ?
tks
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Well I'm just showing the full E.164 as that's what the lab I'm looking at
looks for. However I guess you could strip the HQ area code at the gateway with
the calling party transformation.
In the real world (plan to visit that soon) then the remote destination is
likely to be a mobile phone
The crazy thing is I tried this but I couldn't get it to work.
PT_HQ_CALLING_OUT assigned to CSS_HQ_CALLING_OUT assigned to CALLING Number
Transform on the Outbound portion of the HQ gateway.
Calling Party Transformation: PT_HQ_CALLING_OUT pattern is \+1480.!(replace
480 with what your
well, that is why we have this group for - to help each other and test our
config and knowledge before the real test.
so gang,
for WAN, when you need MLP, just go to subinterface, enter bandwidth and go
to DLCI and type:
auto qos voip [trust] fr-atm
(trust is optional, if nothing else is
calling party transformation is done without prefix \
On Fri, Oct 1, 2010 at 7:00 PM, Mark Holloway m...@markholloway.com wrote:
The crazy thing is I tried this but I couldn't get it to work.
PT_HQ_CALLING_OUT assigned to CSS_HQ_CALLING_OUT assigned to CALLING Number
Transform on the
When doing it under Call Routing Transformation Pattern Calling Party
Transformation you have to use \+
When doing it on the Calling Party transform mask on a Route Pattern or Route
List you don't use \
On Oct 1, 2010, at 10:44 AM, sisiaji wrote:
calling party transformation is done
Yeah it is but if you have tested some solution yourself already and
need some guidance as to where you are making mistake.
This list, I presume, is not for those who post questions with even exam
points mentioned having no clue what they are asking about and then
expecting a reply! Sorry I
I feel that there has been too many people blatantly breaking their NDAs on
this list lately...please ask questions related to your studies, not exact
exam questions !
Then I know I will be glad to help and I'm sure there are more on the list
who will too...
-Original Message-
From:
yeah, you are right, I was referring to RP/RL transformations...
i tested it and i got the same in my lab
so i guess, as you already mentioned before, the way to do it is to actually
put Calling Party Transform Mask to be XXX on the RL (for RG member).
On Fri, Oct 1, 2010 at 7:52 PM, Mark
The only issue with this is you don't know if the calling party is Subscriber,
National, or International, so you can't use XXX because if BR2 or BR1
calls HQ3 the From number would only show the first 7 digits.
On Oct 1, 2010, at 11:21 AM, sisiaji wrote:
yeah, you are right, I was
are you sure those are real questions?
i thought they are pasting some tasks from ipexpert or other's mock labs or
so...as i myself don't have those mock labs...
how can i distinguish between those?
On Fri, Oct 1, 2010 at 8:15 PM, CCIE Voice GMAIL
givemeccievoice2...@gmail.com wrote:
I feel
Same here , I was beginning to think that no patterns are matched in calling
number transformations - but I tested with a pattern of ! and a mask of 12345
and that works.
So it would appear that there is a mismatch between \+1480.! and the calling
number, which does seem odd as if you leave
I may be wrong in this particular instance, but if you've been paying
attention to the list for the past 3 months or so you've seen it happen more
than a few times.
I apologize if this isn't the case here.
However, I agree with Ash, this isn't a list for you to just get answers,
this
Also highly recommended:
Vik's Voice Lab Strategy vLecture, available via IPexpert's Facebook
page:
http://www.facebook.com/pages/IPexpert/24586557119?v=app_7146470109
Direct link:
http://ipexpert.acrobat.com/p93148979/
cheers, sd
From: ccie_voice-boun...@onlinestudylist.com
Remote Destination Profile has a Rerouting Calling Search Space.
-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Holloway
Sent: Friday, October 01, 2010 3:39 AM
To: osl osl
Subject: [OSL | CCIE_Voice]
Hello Experts,
I'm working on TEHO to Site C with Gatekeeper. Everything works except for the
+ sign. On the debug ISDN Q931 on both sites, I see the + being sent and
received, but on the phone it only shows my 11 digits (12123945003), but on the
bottom of the phone it shows my plus sign.
That's a bug in CME I believe. That is the correct behavior.
Jeff
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ayman labib
Sent: Friday, October 01, 2010 2:37 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice]
Hi everyone,
I am having a hard time remembering what command will affect the number
displayed in the upper-right of the phones for CME.
With SCCP, I know the description command will effect that number. How do
you change this value for SIP phones registered to CME?
Thanks for the
description in voice register pool config mode.
On Sat, Oct 2, 2010 at 10:51 AM, CCIE Voice GMAIL
givemeccievoice2...@gmail.com wrote:
Hi everyone,
I am having a hard time remembering what command will affect the number
displayed in the upper-right of the phones for CME.
With SCCP, I
Hi Dan,
Thanks for the response.
Before I could test what you had told me, I must have screwed something up.
I keep getting a message on both of my CME phones saying Unprovisioned. I
have reloaded my router and re-configured everything again but I am still
getting that message.
Has
Make sure after each change you commit to your SIP phones configuration
files you do a create profile under voice reg global. Verify you have your
config file ready for download by issueing show voice register tftp and
check the mac address of your SIP phone is in the list.
A couple of other
Its normal Behavior, no worries.
De: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] Em nome de ayman labib
Enviada em: sexta-feira, 1 de outubro de 2010 18:37
Para: ccie_voice@onlinestudylist.com
Assunto: [OSL | CCIE_Voice] +Dialing to Site C without
Hi again,
I actually reloaded my router with clean configuration and then
re-configured CME, however I am still seeing the same problem. I erased the
configurations on the phones before this all happened, so I assume this is
maybe part of the problem. I dont know why it would be though, as
I beleive you are missing tftp path flash: under voice register global.
Can you try, create profile and let us know?
On Sat, Oct 2, 2010 at 1:11 PM, CCIE Voice GMAIL
givemeccievoice2...@gmail.com wrote:
Hi again,
I actually reloaded my router with clean configuration and then
Hi guys,
Now that I'm done with my lab, I have the following voice lab equipment on
sale. Please let me know if you are interested. Thanks.
1 x 3640 router
2 x 2811 router
3550 24 port POE switch
2509 router (Access Server)
2522 router (FR Switch)
AIM-CUE
HWIC-4ESW
PVDM2-48 x 1
PVDM2-16 x 1
6 x
Is there a specific setting to force the ip phone to display an in use
message in the event the pstn phone answers the incoming call?
On Oct 1, 2010, at 11:42 AM, Graham Hopkins wrote:
Same here , I was beginning to think that no patterns are matched in calling
number transformations -
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