First, thanks everyone for all of your responses. I tried everything that
you suggested. Matthew's suggestion of adding the voip tech-prefix 1#
actually gets the call to route, but it breaks the requirements of the
question. I'm only allowed to have the CUCM and CUCME under the 1# default
techno
Hi CCIE Voice (hopefully, I wish you to start using ur number soon ;-) ),
Well, I may suggest something different, I have two suggestions:
1st, assign an ip address for your gatekeeper in first zone local command
2nd, use different interface for your VIA zone and your gatekeeper
Run a "debug voip dialpeer " on the CUBE. I had the EXACT same issue as you
and, for me, it turned out to be an issue with my dial-peers not matching
correctly.
You need to add "h323-gateway voip tech-prefix 1#" under your loopback
interface.
On Sun, Jun 20, 2010 at 6:21 PM, CCIE VOICE wrote:
I have BR1 phones configured with a shared line. During normal operation
when phone 1 calls a number (ie 911), phone 2 can see the number of the
caller who is connected to phone 1. So privacy is off. This was done by
changing the service parameter in callmanager and leaving the phones to
default
Just One more thing. I think for such requirement, we should not enable the
priority-queue under the interface. It will overwrite the shape/share which
is not mentioned in the question. Please correct me, if I am wrong. Thanks.
Cheers,
Vincent
2010/6/20 Matthew Hall <1.matt.h...@gmail.com>
> F
I believe you need a zone prefix for zone VIA. Have you tried to put that
on?
On Mon, Jun 21, 2010 at 12:21 PM, CCIE VOICE wrote:
> Hey everyone...I have NO IDEA what is causing my issue and I was hoping for
> your assistance. I am currently working on Volume 2, Lab 1, Task 4.2 with
> no succes
I have felt your pain ...try this...
!
gatekeeper
shut
no shut
!
Hopefully it works.
Regards,
Mike
On Sun, Jun 20, 2010 at 8:21 PM, CCIE VOICE wrote:
> Hey everyone...I have NO IDEA what is causing my issue and I was hoping for
> your assistance. I am currently working on Volume 2, Lab 1,
Hey everyone...I have NO IDEA what is causing my issue and I was hoping for
your assistance. I am currently working on Volume 2, Lab 1, Task 4.2 with
no success. The goal is to dial 3XXX from HQ or BR1 and route the call from
CUCM-->GK-->CUBE-->BR2-RTR. I am getting the *Viazone gateway selectio
Anyone in SJC this week for the OWLE? Shoot me an email if you want to grab a
drink tonight.
- Sent from my Blackberry
___
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Thank You Nasser & Ashar for your expert answers
It worked
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Yes it does show 5002 is calling.
From: bkvalent...@gmail.com [mailto:bkvalent...@gmail.com]
Sent: 18 June 2010 22:24
To: Jones, Brett; 'bkvalent...@gmail.com'; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 5C Q12 Mobile Voice Access
From your remote destination, when y
Thanks,
but I forgot to mention - all the gateways are MGCP. I need to solve this on
CUCM.
On Sun, Jun 20, 2010 at 9:47 PM, naoufal.kerboute
wrote:
> Try to do apply a translation-rule on the dial peer routing call to UC
> using ur internal extension.
>
>
___
Try to do apply a translation-rule on the dial peer routing call to UC using ur
internal extension.
Message d'origine
De: ccie_voice-boun...@onlinestudylist.com de la part de kobel
Date: dim. 6/20/2010 3:01
À: ccie_voice@onlinestudylist.com
Objet : [OSL | CCIE_Voice] AAR and AN
Hello,
I'm playing with AAR and VM. When out of bandwidth condition occurs, AAR
correctly kicks in for different types of calls (between HQ and BR1, direct
calls to VM from BR1, for incoming PSTN calls to BR1 forwarded to voicemail
in HQ). It seems that the configuration is ok.
But I've an issu
Your Config looks fine to me... Maybe they found an issue in their FR
switch?
Great troubleshooting notes posted in this thread, though. Would put you on
very good footing to take the issue to the proctor in the actual lab.
On Jun 20, 2010 9:54 AM, "naoufal.kerboute" wrote:
This is on PL racks
The PL engineer told me that The cable was severed
so a waste of time
Thank you guys
Message d'origine
De: Graham Hopkins [mailto:ghopk...@wolf-rock.co.uk]
Date: dim. 6/20/2010 1:50
À: naoufal.kerboute
Cc: ccie_voice@onlinestudylist.com
Objet : Re: [OSL | CCIE_Voice] Problem Con
This is on PL racks, Now it's resolved by proctor, but I want to know whats is
the problem.
Proctor engineer told me to post the issue in the rack then they told me it's
resolved, but what is the problem because the configuration seems OK.
I'm comfused :s
Message d'origine
De:
Is this on your on kit or one of the PL racks ?
What is the status of the frame relay PVCs ?
sh frame-relay pvc
sh frame-relay pvc 101 etc
and the lmi to the frame switch
sh frame-relay lmi
check that the status messages are being sent and received thus
LMI Statistics for interface Serial1/0
Hi,
I've a connection issue between HQ and BR1, I can't bring the interface dlci
201 up. below my configuration:
BR1:
interface Serial0/0/1:0
no ip address
encapsulation frame-relay IETF
no fair-queue
frame-relay lmi-type ansi
!
interface Serial0/0/1:0.1 point-to-point
ip address 10.10.111.
i tested bot the RP first.. then i did a no mgcp command on GW1
On Sun, Jun 20, 2010 at 4:52 PM, Angel Perez wrote:
> Hi:
>
> Did you test both rp alone first to make sure it working correctly?
>
> Did you shutdown controller at br1 before testing backup path?
>
> thx
>
> -
ok sorry for that i didn't get the question i thought he as problem with ANI
i'll try to figure it out from previous mails thx Ash
On Sun, Jun 20, 2010 at 7:22 AM, Angel Perez wrote:
> Hi:
>
> Did you test both rp alone first to make sure it working correctly?
>
> Did you shutdown co
Hi:
Did you test both rp alone first to make sure it working correctly?
Did you shutdown controller at br1 before testing backup path?
thx
Date: Sun, 20 Jun 2010 11:49:27 +0100
From: siddas...@gmail.com
To: voip.ccieci...@gmail.com
CC: gorr...@hotmail.com; ccie_voice@onlinestudyli
@Shadow
He is not talking about changing ANI.
Ash>
Shadow of Voice wrote:
ok try this way and i am
assuming u r calling from Br1 first call should go thru BR1-GW if fails
then go to BR2-GW and both are MGCP also assuming number your are
dialling 972 xxx and make sure under servi
ok try this way and i am assuming u r calling from Br1 first call
should go thru BR1-GW if fails then go to BR2-GW and both are MGCP also
assuming number your are dialling 972 xxx and make sure under service
parameters these parameter should be false
Stop Routing on Out of Bandwidth Flag
Angel,
Yes you are correct to play with the display we set calling party or
called party transformation. The problem in this scenario is that you
have to change the number depending on the gateway it is egressing ...
the one I mentioned may do the trick (not sure)I will have to test
this o
Did you also try what I suggested? masking Called party at RL detail
level!
cisco voip wrote:
I tried this just now. and it is not working,
So what i was thinking is correct, it can match only one route pattern
and call cannot come back.
Is there any other way anyone would think of?
I missed one question of yours - CUCM PUBLISH trunk is also configured in
CUPS (presence -> settings). There you can check the checkbox to enable
PUBLISH method and select on of the SIP trunk on CUCM which should be used
for this purpose. This also changes the CCM service paramter for CCM via
AXL.
Hi,
I'm not aware of any document describing this explicitly. This is the only
document I know:
http://docwiki.cisco.com/wiki/Cisco_Unified_Presence%2C_Release_7.x_--_How_to_Configure_the_SIP_Trunk_on_Cisco_Unified_Communications_Manager#How_to_Configure_the_SIP_Trunk_on_Cisco_Unified_Communicatio
Hi Kobel,
Is there any document for how to configure the CUP Publish trunk method?
I could understand from the posts that we still need to create a SIP trunk,
configure it in the CUP Publish service parameter field, and assign each user
to a line appearence
Anyway, if the CUCM gatewa
FYI, I think this is too complex to answer simply:
mls qos queue-set output 1 threshold 2 40 60 100 200
This line gives threshold 1 40% of the buffer space and threshold 2 60% of the
buffer space, total buffer space reserved is 100% and max (threshold 3) is
200%.
Reserved at 100% tells the
I tried this just now. and it is not working,
So what i was thinking is correct, it can match only one route pattern and
call cannot come back.
Is there any other way anyone would think of??
On Sun, Jun 20, 2010 at 12:00 AM, Angel Perez wrote:
> Hi Ash, I think that to change calling n
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