[OSL | CCIE_Voice] Single Number Reach

2010-10-01 Thread Mark Holloway
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

Re: [OSL | CCIE_Voice] UCX preventing from answering calls that it originated

2010-10-01 Thread Stutz, Bernhard
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

Re: [OSL | CCIE_Voice] Single Number Reach

2010-10-01 Thread Tam Nhu
Hi Mark, The EPNM does not work at RP for SNR. Have you try to set EPNM at RL level? TN. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com

[OSL | CCIE_Voice] How to install CUCM

2010-10-01 Thread Pithog Oil
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

Re: [OSL | CCIE_Voice] Single Number Reach

2010-10-01 Thread groganhockey
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

Re: [OSL | CCIE_Voice] Single Number Reach

2010-10-01 Thread Graham Hopkins
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.

Re: [OSL | CCIE_Voice] Single Number Reach

2010-10-01 Thread Mark Holloway
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

Re: [OSL | CCIE_Voice] Single Number Reach

2010-10-01 Thread Mark Holloway
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,

[OSL | CCIE_Voice] CUPC client

2010-10-01 Thread Erwan Erwan
hi all,   is anyone know proctor rack has CUPC client installed in the PC ?   tks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com

Re: [OSL | CCIE_Voice] Single Number Reach

2010-10-01 Thread Graham Hopkins
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

Re: [OSL | CCIE_Voice] Single Number Reach

2010-10-01 Thread Mark Holloway
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

Re: [OSL | CCIE_Voice] QOS solution required.

2010-10-01 Thread sisiaji
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

Re: [OSL | CCIE_Voice] Single Number Reach

2010-10-01 Thread sisiaji
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

Re: [OSL | CCIE_Voice] Single Number Reach

2010-10-01 Thread Mark Holloway
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

Re: [OSL | CCIE_Voice] QOS solution required.

2010-10-01 Thread Ashar Siddiqui
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

Re: [OSL | CCIE_Voice] QOS solution required.

2010-10-01 Thread CCIE Voice GMAIL
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:

Re: [OSL | CCIE_Voice] Single Number Reach

2010-10-01 Thread sisiaji
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

Re: [OSL | CCIE_Voice] Single Number Reach

2010-10-01 Thread Mark Holloway
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

Re: [OSL | CCIE_Voice] QOS solution required.

2010-10-01 Thread sisiaji
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

Re: [OSL | CCIE_Voice] Single Number Reach

2010-10-01 Thread Graham Hopkins
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

Re: [OSL | CCIE_Voice] QOS solution required.

2010-10-01 Thread CCIE Voice GMAIL
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

Re: [OSL | CCIE_Voice] Speed for taking the Lab

2010-10-01 Thread Steve Denney (stdenney)
Also highly recommended: Vik's Voice Lab Strategy vLecture, available via IPexpert's Facebook page: http://www.facebook.com/pages/IPexpert/24586557119?v=app_7146470109 Direct link: http://ipexpert.acrobat.com/p93148979/ cheers, sd From: ccie_voice-boun...@onlinestudylist.com

Re: [OSL | CCIE_Voice] Single Number Reach

2010-10-01 Thread Jason Aarons (US)
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]

[OSL | CCIE_Voice] +Dialing to Site C without Gatekeeper

2010-10-01 Thread ayman labib
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.

Re: [OSL | CCIE_Voice] +Dialing to Site C without Gatekeeper

2010-10-01 Thread CCIE Voice GMAIL
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]

[OSL | CCIE_Voice] SIP Phones in CME

2010-10-01 Thread CCIE Voice GMAIL
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

Re: [OSL | CCIE_Voice] SIP Phones in CME

2010-10-01 Thread Daniel Berlinski
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

Re: [OSL | CCIE_Voice] SIP Phones in CME

2010-10-01 Thread CCIE Voice GMAIL
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

Re: [OSL | CCIE_Voice] SIP Phones in CME

2010-10-01 Thread Daniel Berlinski
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

[OSL | CCIE_Voice] RES: +Dialing to Site C without Gatekeeper

2010-10-01 Thread Marcelo Alexandria
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

Re: [OSL | CCIE_Voice] SIP Phones in CME

2010-10-01 Thread CCIE Voice GMAIL
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 don’t know why it would be though, as

Re: [OSL | CCIE_Voice] SIP Phones in CME

2010-10-01 Thread Daniel Berlinski
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

[OSL | CCIE_Voice] Voice Lab Equipment on Sale

2010-10-01 Thread Duke
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

Re: [OSL | CCIE_Voice] Single Number Reach

2010-10-01 Thread Mark Holloway
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 -