There is a CCM Service parameter to check calling no for MVA. You may set it to
partial match to avoid issues with not having exact digit by digit match. See
if it helps.
--- On Thu, 6/24/10, naoufal.kerboute naoufal.kerbo...@cbi.ma wrote:
From: naoufal.kerboute naoufal.kerbo...@cbi.ma
Hi List,
I've discussed this with PL team and taken their permission before posting this.
I've around 20 odd vouchers available at minimal price. Those are left
overs after passing my lab. If anyone interested pl PM me. All vouchers are
valid for V3.
Thanks
I cracked mine on 3rd go in V2. Be consistent...good luck
--- On Wed, 9/2/09, Ravindra Lakpriya lakpr...@gmail.com wrote:
From: Ravindra Lakpriya lakpr...@gmail.com
Subject: Re: [OSL | CCIE_Voice] V3 Attempt Two
To: Tanner Ezell tanner.ez...@gmail.com
Cc: ccie_voice@onlinestudylist.com
Date:
available at
http://www.ipexpert.com/index.cfm/a/p/vlectures. There is lot of relevant
information available in those vlectures.
Wish good luck to all,
Thanks,
Kapil Atrish
--- On Mon, 6/29/09, Cristobal Priego cristobalpri...@gmail.com wrote:
From: Cristobal Priego cristobalpri...@gmail.com
Hi list,
I've passed the lab exam and would like to thanks each one of you for your
contributions to this list.
I would also like to extend my thanks to PL After-hour support team for
providing the instant support on various issues I faced during practice
sessions.
Thanks,
Kapil Atrish
To: kapil atrish nice_cha...@yahoo.com, Cristi Radescu
cristian.rade...@crescendo.ro
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Date: Tuesday, June 23, 2009, 7:50 AM
What about 3101 requirement?
From: kapil atrish
[mailto:nice_cha...@yahoo.com]
Sent
That's a common issue. Create one DP on CME as following:
dial-peer voice xxx pots
incoming called-nu .
direct-inward-dial
!
--- On Thu, 5/28/09, ccieid1ot ccieid...@gmail.com wrote:
From: ccieid1ot ccieid...@gmail.com
Subject: Re: [OSL | CCIE_Voice] Problem with outgoing calls from Branch
image...:(
I did shake it, opened the box and tried pressing the little button but no
good.:(
--- On Sun, 5/3/09, Michael Ciarfello mciarfe...@iplogic.com wrote:
From: Michael Ciarfello mciarfe...@iplogic.com
Subject: RE: [OSL | CCIE_Voice] ATA IVR not responding
To: kapil atrish nice_cha
| CCIE_Voice] ATA IVR not responding
To: kapil atrish nice_cha...@yahoo.com, ccie_voice@onlinestudylist.com
Date: Saturday, May 2, 2009, 11:30 PM
Did you push the button on top of the ATA after
picking up the phone on port 1? That's how you activate the IVR
menu.
What exactly have you done
Hi list,
I want to know if CCM 7 is supported on VMWare workstation 5.0? The hardware
I've is AMD quad-core, 4gig ram. Will that work, if someone who has tested it
can comment please?
Thanks..
I've tested in on CCM and CME but over the PSTN.
On CCM side:
Create a Voice-Mail profile to_vm, select the Voice-mail pilot and put the
external phone number mask as .
Create a Route Point with DN 22xxx, do call-forward all to VM, select the VM
profile to_vm.
To make it work over the
:
zone-prefix GK 1* gw-priority 10 Sub_Trunk_1
zone-prefix GK 1* gw-priority 9 Pub_Trunk_2
--- On Sun, 3/29/09, CCIE OSL ccie...@gmail.com wrote:
From: CCIE OSL ccie...@gmail.com
Subject: Re: [OSL | CCIE_Voice] gatekeeper question
To: kapil atrish nice_cha...@yahoo.com
Cc: ccie Me ccievoic
You can try this:
Create two trunks between CCM and GK having only one CCM in each trunk i.e one
with Pub and another one with Sub. Create two set of regions (codec G729), two
locations (24kbps to allow only single call over the trunk), and two DPs. Bind
all this with respective trunks.
On
You are absolutely righttwo ccm groups required having one ccm in each.
--- On Sun, 3/29/09, anil batra anil...@yahoo.com wrote:
From: anil batra anil...@yahoo.com
Subject: Re: [OSL | CCIE_Voice] gatekeeper question
To: ccie Me ccievoic...@yahoo.com, CCIE OSL ccie...@gmail.com, kapil
atrish
Hi List,
Inside Unity Call Handler or SubscriberCall Transfer options you have the
Checkbox to enable/disable the prompt Wait while I transfer your call. I
noted in a unity system this option is not there at all.
Has anybody seen this? Does anyone know if there is a way in Unity to get the
Priego cristobalpri...@gmail.com wrote:
From: Cristobal Priego cristobalpri...@gmail.com
Subject: Re: [OSL | CCIE_Voice] Unity - Wait while I transfer option not
available
To: kapil atrish nice_cha...@yahoo.com
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Date: Tuesday, March
following statement should also be added for H245 traffic:
set qos acl ip POD15_SERVER dscp 24 tcp any any ran 11000 65535
--- On Sun, 3/22/09, Christian Hennrich christian.hennr...@intact-is.com
wrote:
From: Christian Hennrich christian.hennr...@intact-is.com
Subject: Re: [OSL | CCIE_Voice]
Hi list,
Has anyone any thoughts on this?
--- On Fri, 3/13/09, kapil atrish nice_cha...@yahoo.com wrote:
From: kapil atrish nice_cha...@yahoo.com
Subject: Policer on Cat6k - Aggregate or Microflow
To: ccie_voice@onlinestudylist.com
Date: Friday, March 13, 2009, 1:46 AM
HI,
I am looking
You are not ref to the label, correct?
I know of following two ways:
1) You can simply apply the same ephone-dn to button 1 and 2 on same phone.
2) You can create multiple DNs with same number and apply to two different
buttons of same ephone.
--- On Fri, 3/13/09, CCIE OSL ccie...@gmail.com
Put max-conn under dial-peer.
--- On Fri, 3/13/09, CCIE OSL ccie...@gmail.com wrote:
From: CCIE OSL ccie...@gmail.com
Subject: [OSL | CCIE_Voice] How to only allow one international or LD call?
To: OSL Group ccie_voice@onlinestudylist.com
Date: Friday, March 13, 2009, 12:56 AM
Question
How do
HI,
I am looking for a clarification on the policer to be used on Cat 6k.
Question says limit sccp traffic from phones to 32k, I've see few posts where
an aggregate policer has been configured and implemented on voice vlan.
Shouldn't there be a microflow policer since I want to limit sccp
Hi,
I've a query regarding CNAME display when calling PSTN.
1) CCM Phone calls PSTN Phone
Pstn phone can see the CNAME of CCM Phone but CCM Phone can't see the name of
called party.
2) PSTN Phone calls CCM Phone
CCM phone can see the CNAME of PSTN phone but PSTN Phone can't see the name of
Training Tools for the Cisco CCIE RS Lab, CCIE
Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab
Certifications.
From: Kapil Atrish kapilatr...@hotmail.com
Date: Tue, 10 Mar 2009 02:23:28 +0530
To: OSL Group ccie_voice@onlinestudylist.com
Subject: [OSL
] On Behalf Of Kapil Atrish
Sent: Saturday, March 07, 2009
2:38 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] ATA186
- not able to change anything
Hi list,
I've an ATA on POD11 which has port 1 enabled by default. I want to enable Port
2 as well. Whenever I click on SCCP
You need Xcoder. BACD supports only G711 and since you are doing G729, you'll
face this problem. To verify, make end to end G711 and you should be able to
reach BACD successfully.
--- On Wed, 3/11/09, Jiahong - tobeccie Fang mo...@hotmail.com wrote:
From: Jiahong - tobeccie Fang
Hi List,
I tried to mark SCCP/RTP traffic on ESW module, but the policy got rejected
with a message the it doesn't support range keyword in access-list.
Any other workaround to mark all SCCP and RTP packets to respective DSCP values
on this module? I could successfully do it on Cat 3550 with
Hi,
I observed both ports on ATA register with dual-line even though max-dn
dual-line is not configured. Is this expected behavior?
I couldn't find much information on cisco reg this. See following capture
interface FastEthernet0/0
/call-mana
filtering...
call-manager-fallback
After switching over to max-dn dual-line I realised ATA actually registers with
two channels now. Earlier it showed two buttons both associated to Channel 1.
If someone can comment on the usage of two buttons with single channe or if
there is something wrongl, that'll be great.
See below
In the route pattern or route-group level, don't use Use external phone number
mask and put area-code (3 digits) as Prefix to calling number.
--- On Mon, 3/2/09, hasan khalife hasan_khal...@hotmail.com wrote:
From: hasan khalife hasan_khal...@hotmail.com
Subject: [OSL | CCIE_Voice] 6608 gw
To:
Instead of rolling it over to HQ Unrestricted, I created a CSS having access to
only 1x which internally points to HQ GW. The TP can only be accessed
from GK and am doing PreDot 1# at TP level. It always works for me.
Another TP which looks for 1#.[1-2]xxx. I do a PreDot and roll it
Hi list,
Actually, it all started when I tried to send a G.729 stream to CME with G.711
at CCM side. I initially had Xcoder on CME but never got invoked. I tried IPIP
GW but same result. Below is my scneario and results of the testing so far:
I've a trunk from CCM to GK with codec G.711 set.
Put following on BR1 to trigger SRST:
ip route ccm pub ip /mask null 0
ip route ccm sub ip/mask null 0
--- On Sun, 3/1/09, Mike Brooks 2xcci...@gmail.com wrote:
From: Mike Brooks 2xcci...@gmail.com
Subject: Re: [OSL | CCIE_Voice] BR1 - SRST???
To: Cliff McGlamry cl...@mcglamry.net
Cc:
-To: anil...@yahoo.com
Date: Fri, 20 Feb 2009 10:08:12 -0800 (PST)
To: OSL Group ccie_voice@onlinestudylist.com, Kapil Atrish
kapilatr...@hotmail.com
Subject: Re: [OSL | CCIE_Voice] CCM to GK trunkcan load balancing be achieved
Not sure if it possible, In CCM when we create RG-GK we can't define
Hi List,
I want to confirm if there is a way to play MOH from BR1 flash in case Primary
MOH server is down.
thanks,
_
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Regards
- Basant
On Wed, Feb 25, 2009 at 12:44 PM, Kapil Atrish kapilatr...@hotmail.com wrote:
Hi List,
I want to confirm if there is a way to play MOH from BR1 flash in case Primary
MOH server is down.
thanks
February 2009 12:04 AM
To: Kapil Atrish; basant.ya...@gmail.com
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] MOH from BR1 Flash in case Primary CCM down
Kapil,
I haven’t tested this but .
If Sub unicast to BR1 than you can easily achieve this by spoofing the PUB
CCIE RS Lab, CCIE
Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab
Certifications.
From: Kapil Atrish kapilatr...@hotmail.com
Date: Thu, 19 Feb 2009 09:59:28 +0530
To: vma...@ipexpert.com, ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice
Hi List,
Quick question on LLQ with FRTS through MQC. I am not able to recall but I read
some where the class-default is not required when FRTS is implemented using MQC
alongwith LLQ.
Below is the config shap-shot:
policy-map voice
class EF
priority percent 50
class AF
bandwidth percent 5
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and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE
Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab
Certifications.
From: Kapil Atrish kapilatr...@hotmail.com
Date: Fri, 20 Feb 2009 17:08:26 +0530
Hi,
CCM cluster (pub, sub) is registered to a GK. GK correctly shows two trunks
with _1 and _2.
Requirement is to enable load-balancing of outgoing calls from CCM to GK over
those two trunks , how can it be achieved? I can achieve GK to CCM
load-balancing via gw-priority, but requirement is
Hi List,
My cue module is in rebooting state. Reboot of Router didn't help. It always
comes to this stage and halt:
System Now Booting ...[BOOT-ASM]
7
Please enter '***' to change boot configuration:
__
I've observed following during
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 SubscriberMessage
notification settings, like Home
...@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] UnityUTIMMessage Notification VS Dialout MWI
Hi List,
I'v a question about Message
.
On Tue, Feb 3, 2009 at 5:02 PM, Kapil Atrish kapilatr...@hotmail.com wrote:
Cool...I did not check for the TCD Service Parameter. I think if I set this
parameter the second AC would not be required. I may simply put a DN as Always
route member to extend fast busy to caller after initial
of 'userbusy'.
On Wed, Jan 28, 2009 at 11:15 AM, Kapil Atrish kapilatr...@hotmail.com
wrote: I did not put the TP directly inside the Hunt-Group. I put a CTIRP as
Always Route Member and on CTIRP I did a forward all to the TP. I am yet to
try the solution given by Christian. I'll put the call
Pilot Point. After the hold time expires for the first AC pilot,
the call will be forwarded to the second AC pilot. Since queuing is
disabled, the call should drop BUT w/ a disconnect cause of 'user
busy'.
On Wed, Jan 28, 2009 at 11:15 AM, Kapil Atrish kapilatr...@hotmail.com wrote:
I did
was able to add phone/CTIRP DNs
though.
On Tue, Jan 27, 2009 at 8:48 PM, Kapil Atrish kapilatr...@hotmail.com wrote:
I tried with RP/TP Block this pattern and in that case call
stays in queue. AC takes the call out of the queue only when it is
routed to a registered end-point that's what
expires for the first AC pilot, the call will be forwarded to the second AC
pilot. Since queuing is disabled, the call should drop BUT w/ a disconnect
cause of 'user busy'. On Wed, Jan 28, 2009 at 11:15 AM, Kapil Atrish
kapilatr...@hotmail.com wrote: I did not put the TP directly inside
Hi,
Do you've dspfarm and dsp services dspfarm under voice-cards??
Daniel Sobrinho dani...@hotmail.com wrote:
Hello,
Could please help me with a doubt? I've been made an upgrade of DSPs in my
router 2851 to increase the capacity for transcoder and conference bridge.
disonncted, it seems
the queue is holdin git for forever. Anyone here has tested this and have some
workaround please.
--- On Tue, 1/27/09, Kapil Atrish kapilatr...@hotmail.com wrote:
From: Kapil Atrish kapilatr...@hotmail.com
Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN
?
what about routing to a number CUCM, which does not exist, or even to a
PSTN number, which is unallocated?
Christian
Kapil Atrish schrieb:
The requirement is to drop the call within CCM itself. I don't want to
use Unity/IPCCX/TCL for this purpose
, I am not able to disconnect the call.
The message keeps on playing until caller drops the call.
thanks,
Kapil Atrish
Date: Mon, 26 Jan 2009 18:57:28 +0100
From: christian.hennr...@intact-is.com
To: cpar...@cparker.us
CC: ryanstudyvo...@gmail.com; kapilatr...@hotmail.com;
ccie_voice
, I am not able to disconnect the call.
The message keeps on playing until caller drops the call.
thanks,
Kapil Atrish
Date: Mon, 26 Jan 2009 18:57:28 +0100
From: christian.hennr...@intact-is.com
To: cpar...@cparker.us
CC: ryanstudyvo...@gmail.com; kapilatr...@hotmail.com;
ccie_voice
?
Thanks,
Kapil Atrish
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?
Thanks,
Kapil Atrish
_
Plug in to the MSN Tech channel for a full update on the latest gizmos that
made an impact.
http://computing.in.msn.com/
the help you provided.
thanks,
Kapil Atrish
From: narinder.ku...@uxcg.com.auto: kapilatr...@hotmail.com;
ccie_vo...@onlinestudylist.comdate: Mon, 19 Jan 2009 23:39:09 +1100Subject: RE:
[OSL | CCIE_Voice] CME BACD - drop-through not working
Kapil,
This is the configuration which is working
Yes ReadMe is available in router flash. I've used it. But be aware it doesn't
cover each and every parameter (for ex. drop-through).
DocCD is no more provided in the Lab. It has been replaced with following URL:
http://www.cisco.com/web/psa/products/index.html
kamal
Nope. I've done it manier times, you don't need to restart anything.
Mike O mik...@msn.com wrote: When you change to a different script in
IPCC do you need to restart any services?
Thanks,
Mike
Hi,
Can someone pl suggest how to debug this issue? I am still not able to make it
work?
Thanks...
From: kapilatr...@hotmail.comto: ccie_vo...@onlinestudylist.comsubject:
AC-Broadcast hunting not workingDate: Mon, 22 Dec 2008 22:43:43 +0530
Hi list,I've configured Attendant Console and
It won't be a normal call even if annunciator answers the PSTN call (firstly it
doesn't) because difference in Call Clearing Cause Code issued by GW/CCM for
unallocated number, number busy or Call cleared normally etc..
Chris Parker [EMAIL PROTECTED] wrote:
Sounds correct to me.
I
Put Site A phones in AAR Group say SiteA, location SiteA. Put Site B phones in
AAR Group say SiteB, location SiteB. Set AAR prefix, AAR CSS and route-patterns.
Lower down the b/w and AAR triggers.
jeremy co [EMAIL PROTECTED] wrote: Hi,
assuming following scenario:
Is your question about overhead calculation for voice calls or something else?
My explanation for calculating overhead for voice calls.
Example:
CIR 512,
Allow priority b/w for 5 g.729 calls with FRF.12 (can be MLPPP, FR:
Allow 10% overhead.
I would calculate 5x 27.2kbps = 136kbps
Add 10%
Possible your transcoder is not getting invoked.
You've HQ region set to use G.711 within itself and G.729 with others.
I believe CTI RP and ports would be in HQ region and IPCC Express is configured
for G.711 codec. So you get fast-busy. When you change the region settings to
use G.711 you
On route-pattern you may set CLID Name/Number to restricted/allowed.
James Key [EMAIL PROTECTED] wrote: Block calling nameWhat is the
best way to block calling name on certain route patterns, while still allowing
it on others? Example: hq local send calling name + number, hq
translation-profile incoming on ephone-dn and translate the voicemail number to
CUE or Unity Pilot. Leave other phone without translation.
Balamurugan Singaram [EMAIL PROTECTED] wrote: Hi,
We have two cme phones in BR2 two different unity systems:
1st phone press messages button and go
I ran into this problem number of times. I initially put number without
no-reg option under ephone-dn and when integrating it with GK later I simply
put the command number no-reg: and I faced this issue.
I need to do no number and number no-reg to resolve this. Same way for
You may need to upload the welcome-prompt under respective language (en_US,
en, Default).
Did you validate the script before uploading under CRS Script editor?
You may've run the script in reactive mode and verify whether the call was
hitting the script or not and which stage it
100% correct. I had the same issue and I had to remove the dialplan pattern and
use TP under voice-port to meet the requirement of 4 digit CLID to HQ 10
digit DID to PSTN . I've made it a practice not to use dialplan pattern.
However one small confusion when having BACD. For ex Question says:
HI,
Not sure if it has been asked and answered earlier.
Requirement, play CCM annunciator on incoming call from PSTN to any unassigned
DID.
For ex DID range 200-300. Ext 250 to 300 are not assigned to any device. If
PSTN calls any of these DIDs, the caller gets fast-busy. Instead of
When adding CAS circuit in CCM (MGCP) it gives option to enable channels for
Outbound/Inbound/Bothways. I never tried but won't that work for us.
Secondly, if CAS circuit in H.323 mode you may create multiple DS0 and point
DPs to respective voice-ports for outbound calls leaving others for
Isn't the call processing depends upon which CCM (Sub/Pub) the IP Phone/IPMA
CTI Port is registering to? If Phone/CTI RP are registered to Sub, all calls
will be processed by Sub even though IPMA points to Pub.
Correct me if I am missing something..
Yung Hung [EMAIL PROTECTED] wrote:
Does the codec sampling rate need to match at CCM and H.323 GW or whatever
configured at CCM H.323 GW auto-negotiates?
What if different codec sampling rate at CME, does CCM also need to have the
same sampling rate?
Appreciate any comments on this
Hi,
I've configured CUE/CME and I need few phones to test. My softphones fail to
register with CUE. I've tried wit IP Blue and IPC which keeps on
registering/unregistering. I've verified loads are present in flash and CME is
configured as TFTP for all the load files. I am doing manual
One quick question:
Although I've it configured on sub-if and virtual-template, when doing MLPP
fragmentation, I need to put ip pim-dense mode only on int virtual-template and
not on the sub-if?
From: [EMAIL PROTECTED]
To: ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice]
Hi,
Can pl let me know any debug commands to confirm multicast MOH reaching BR1
Router (or Phones if possible). I checked through perfomance monitor on CCM
and could see one MOH Multicast resource active when call was put on hold but
there was no MOH on phone.
I checked show ip pim
Hi,
Following scenario:
IP Phone ---CME---sip trunk---CCM--IP Phone
Using g729 call fails and works fine on 711. MTP is selected on trunk, infact
I've created an IOS enchanced software MTP on a router and given it to SIP
trunk. Bt that's software only and I understand it would support only
From: Kapil Atrish [EMAIL PROTECTED]
To: ccie_voice@onlinestudylist.com
Sent: Monday, October 6, 2008 6:47:23 PM
Subject: [OSL | CCIE_Voice] SIP call fails on G729
Hi,
Following scenario:
IP Phone ---CME---sip trunk---CCM--IP Phone
Using g729 call fails and works fine on 711. MTP
Lab and CCIE Storage Lab Certifications.
On Oct 4, 2008, at 11:43 AM, Kapil Atrish wrote:Its local call from CME phone
to bacd. No gatekeeper in between.
PH1---CME with AA/ACD---ephone=hunt
CC: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com
From: [EMAIL PROTECTED]
To: [EMAIL
ShavrovSent: Monday, October 06, 2008 8:54 AMTo: Edi Hamlet; Kapil Atrish;
[EMAIL PROTECTED]: Re: [OSL | CCIE_Voice] SIP call fails on G729
So. may be we should exclude the software MTP from the MRGL, and keep the only
hardware MTP/XCoder?
- Original Message -
From: Edi Hamlet
HI,
I am getting following when trying to register Cat6K port to CCM on Pod 20.
I've tried enabling/disabling the ports but all three ports (T1, Xcode and CFB)
are in same state. Reset the DHCP service, other devices are able to take IP
Address from DHCP and enough IPs are available in the
debug voice ccapi inout - using two dial-peers
debug voice ccapi inout - using single dial-peer.
Result is same, Unknown number, fast-busy tone.
From: [EMAIL PROTECTED]
To: ccie_voice@onlinestudylist.com
Subject: BACD issue - No welcome prompt
Date: Sat, 4 Oct 2008 17:27:31 +0530
HI,
HI,
Attached is my config. I get fast busy tone and Unknown number on display when
I dial the pilot number from any CME phone. I can dial hunt-pilot directly and
call get routed correctly or give the aa-pilot to hunt-pilot and ring the
phones fine. Call in between phones are setup using
? Scott Hardesty | Cisco
Engineer | MidAtlantic | Presidio Networked Solutions7601 Ora Glen Drive, Suite
100, Greenbelt, MD 20770 | [EMAIL PROTECTED]: 301.313.2041 | C: 443.789.1219 |
www.presidio.com From: [EMAIL PROTECTED]:[EMAIL PROTECTED] On Behalf Of Kapil
Atrish
Sent: Saturday, October 04
Storage Lab
Certifications.
On Oct 4, 2008, at 10:00 AM, Kapil Atrish wrote:
The attached file has full config and debug output if you wish to see.
!
dial-peer voice 15 voip
destination-pattern 3700
session target ipv4:172.22.102.1
dtmf-relay h245-alphanumeric
codec g711ulaw
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Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice
Lab and CCIE Storage Lab Certifications.
On Oct 4, 2008, at 10:00 AM, Kapil Atrish wrote:The attached file has full
config and debug output if you wish
i came to know that candidates are being interviewed by the proctor before
sitting for the lab exam. Has anyone come across this and can share a bit about
the i/v?
_
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any harm but not sure how will that
be graded.
If anybody has followed this approach and any idea how it'll be graded?
Thanks,
Kapil Atrish
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Hi,
Today I was configuring NTP on CCM. I didn't find NTP service available in
services console. I tried to install but install.bat was not present inside
XNTP folder. I checked another CCM installation and the file was not there as
well.
Any thoughts how can i restore install.bat if
Hi,
Following is the URL available in Lab exam now:
http://www.cisco.com/web/psa/products/index.html
Go to Voice Unified communications Cisco Unified CCM Maintain and
Operate Guides there you'll find features and services guide...
Jacob Owen [EMAIL PROTECTED] wrote:
For IPPA URL:
PRODUCT SUPPORT
VOICE AND UNIFIED COMMUNICATIONS
CISCO AGENT DESKTOP
INSTALL AND UPGRADE
Install and Upgrade Guides
See CAD installation guide.
kapil atrish [EMAIL PROTECTED] wrote:
Hi
Hi,
This link doesn't even give access to SRND. Not sure, let's wait for someone
who attends the lab to confirm.
I got that URL info from cisco.com/go/certsupport. I raised a case
specifially to know the new URL and they gave me this one.
Thanks,
Kapil Atrish
Robert Schuknecht
I was working on eBook Volume 1 lab 4. Q 31.
What I understood from there is that When doing PPP multilink fragmentation
alogwith LLQ and frame-relay adaptive traffic-shaping, there is no class map to
be associatecd to DLCI?
Sample configuration:
policy-map LLQ
!
Policy-map GTS
shape
Hi,
1) Is there any way in Unity to check the detailed call treatment other then
looking at the Port Status Monitor and Call Viewer? I want to check which Route
Handler (Direct Call or Forwarded Call) the call hit initially and step by step
flow from there onwards, if any option
Hi,
When Fax-Relay is enabled on 6608 module, what should be the Fax speed i.e.
7200bps or 14400 bps? Should it be configured 7200bps if far end is a Router
and 1400bps if far end is a Cat 6608 module? What should it be set to if far
end is an ATA/VG248?
When
Hi,
I've a subscriber (2001) configured in Unity. I want to route the call as
following:
If someone tries to reach 2001 and the call goes unanswered, it should ring at
2002 and if that extention also doesn't answer the call, the caller should get
routed to Voice Mailbox of 2001.
I
.
Kapil Atrish
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