I agree to Thomas.It would be nice to know that IPexpert does care for
existing customers by giving some incentive on new products.
On Wed, Dec 30, 2009 at 10:01 PM, Thomas Bader wrote:
> Wayne,
>
> although this is a sales question I ask in the OSL as the question seems to
> be common:
>
> can
Hi everyone,
Sorry for off-topic question but just wondering to know if anyone knows
about whether New Voice Labs opened in Beijing,Hong Kong,and Bangalore are
local or remote ? I have heard remote labs have sluggish performance so,just
want to confirm if anyone has been there lately.
Thanks.
__
I can't seem to access IPexpert.com . Is anyone else having same issue ?
On Fri, Dec 11, 2009 at 1:38 PM, Graham Hopkins wrote:
> May also be worth pointing out that your old PDFs will now longer open with
> the new username/password and that you will need to re-download them - at
> least I did.
Congrats James !
On Thu, Dec 10, 2009 at 10:35 PM, Amp wrote:
> I have to say it again, congrats bro. I haven't met anyone as humble
> as you during this entire process and if anyone deserves the digits
> it's you!!!
> Quoting James Key :
>
> > Received the news yesterday that I passed my lab in
ity &
>> Service Provider) Certification Training with locations throughout the
>> United States, Europe and Australia. Be sure to check out our online
>> communities at www.ipexpert.com/communities and our public website at
>> www.ipexpert.com.
>>
>> On Oct 2
I think Vik Malhi is a great great resource.His labs/even vlectures are
always orientated towards practical demonstration while most others just use
slides.I wish Vik had produced Voice VoD too.I have great faith in him and
Wayne,and i am positive , IPexpert will keep dominating voice track.
On Tu
Congrats! ..Please share your strategy and any tips you have for rest of us.
On Sun, Oct 25, 2009 at 6:02 AM, Wilson Bolanos wrote:
> Congratulations! So what resources did you use? IPExperts Mock
> labs? Please let us know.
>
> -Original Message-
> From: ccie_voice-boun..
Hello Team,
I heard sometime back that ProctorLabs is planning down the road to have 4
hr rack sessions in addition to current 8 hrs sessions to facilitate people
like me. I have/had lots of problem w.r.t electricity outage/internet
outage. Having 4 hr sessions will also help us if we need to foc
That seems right.Module 13 as mentioned in Table of Contents for UCCX is not
part of package.I hope it will be available as a download sometime sooner.
Thnx
On Sat, Sep 19, 2009 at 8:12 AM, Brian Valentine wrote:
> Looking through the new VOD table of contents, I don’t see a UCCX
> module. Cur
Philip,
You would have to drop an email to supp...@ipexpert.com to add other files
in your account.
On Mon, Jun 29, 2009 at 6:05 PM, GRAFL Philipp
wrote:
> Has anyone got more yet than the 5A_5B_5C Labs in his ebook section?
>
>
>
> Philipp
>
>
>
> *Von:* ccie_voice-boun...@onlinestudylist.com
enable
offline
restore factory default
On Sat, May 9, 2009 at 1:51 PM, Clawson, Von wrote:
> I was working on a virtual rack session today and tried to setup the
> CUE. However, when I sessioned into the CUE the configuration was not
> clean. I tried wr er and a reset, as well as erase startup
Hi,
Can anyone tell me how long it takes for CCIE voice lab result to come ? I
know expected time is b/w 24-48 hours.But since 34+ hours have already
passed since my lab, i am getting nerveless to know what could be causing
delay ?
Thanks
Yeah..I did that but putting DN didn't work.You would need Secondary AC
pilot #. Besides, I prefer to use Unity rather than going through this
method.At least for lab, it won't be advisable unless strictly asked to do
so.
On Tue, Feb 3, 2009 at 5:02 PM, Kapil Atrish wrote:
> Cool...I did not che
Hi,
When dialing from BR1 into Unity , calls and digit input work .During
AAR/SRST , callers dialing via PSTN are able to leave voice-mail
messages.However, when subscriber dials directly into unity during AAR/SRST,
unity plays subscriber greeting but it doesn't recognize any input dtmf
digit inp
Kapil,
If you dial your first AC pilot # , you should hear greeting .If you dial
second(dummy) AC pilot # , you should hear user busy.Now, When you link 2nd
to first,i.e add 2nd AC pilot # as Always Route Member , after hold time
expires, call will be routed to dummy pilot point and you will get
Its very easy. Go here:
http://corner-il.blogspot.com/2007/07/using-multiple-ipblue-phones-on-one-pc.html
Cheerz
On Tue, Feb 3, 2009 at 9:58 AM, omar itani wrote:
> hi guys ,can any 1 tell me how to use 2 instances of vtgo lite in the same
> time ,what is the procedure need before run on the
;
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *kamal yousaf
> *Sent:* Friday, 30 January 2009 12:35 PM
> *To:* ccie_voice@onlinestudylist.com
> *Subject:* [OSL | CCIE_Voice] CCM abnormal behaviour
>
>
>
> Can someone sp
Can someone spot what could be issue when CCM server starts showing all
gateways/ip phones/ route lists/route groups/every MRG/MRGL as Status"NOT
FOUND" Or "NOT REGISTRED" . I have restarted CCM services 2 times. Then,
rebooted box twice thinking problem will vanish but it didn't. SQL
replication
the hunt
>
>
>
> - Original Message -
> *From:* kamal yousaf
> *To:* ccie_voice@onlinestudylist.com
> *Sent:* Thursday, January 29, 2009 1:07 PM
> *Subject:* [OSL | CCIE_Voice] B-ACD VoiceMail
>
> If B-ACD script causes call to be sent to VoiceMail number defined using
>
If B-ACD script causes call to be sent to VoiceMail number defined using
'param voicemail 5000' , which mailbox is the call routed to ? Is it Pilot
Point OR HuntGroup Number ?
I tested it and it works great.Thanks Anthony for kind help.
On Thu, Jan 29, 2009 at 3:52 PM, Anthony Yeung wrote:
> What you can try is assign a dummy AC pilot point to the original AC
> Pilot Point as the 'Always Route Member' creating a linked hunt group.
> Then for this second dummy AC pilot
Is it possible to change cal ltransfer number of Call Handler that is being
managed using Greetings Administrator.Sure,you can modify greeting but can
you modify forwarding number too ?
I tried same way.It plays greeting only once.I also changed service
parameter for Cisco TCD "Allow Routing with Unknown Line State" to True ,and
retried.Call still doesn't
end.
Kapil,
how did you add TP as member in HuntGroup.In my case, it gives error saying
that member should be a valid DN on s
No need to memorize anything.If readme file is available,then its
good.Otherwise,you DocCD and copy-paste script for your use.
Here is the navigation Structure:
DocCD:
*Voice and Unified Communications --> IP Telephony Call Control -->CUCME -->
Configuration Guidelines --> Cisco Unified CME B-ACD
(do not clear calls when the
datalink goes down).
!
On Wed, Jan 21, 2009 at 2:14 PM, kamal yousaf wrote:
> James/Ryan,
>
> What if you change CCM Service Parameter *Clear Calls Flag When Datalink
> Is Down = False* ? Have you tested and seen if MGCP fallback call gets
> preser
>> *From:* ccie_voice-boun...@onlinestudylist.com [
>> ccie_voice-boun...@onlinestudylist.com] On Behalf Of kamal yousaf [
>> lovingprin...@gmail.com]
>> *Sent:* Tuesday, January 20, 2009 5:32 PM
>> *To:* Ryan Trauernicht
>> *Cc:* ccie_voice@onlinestudylis
, Ryan Trauernicht wrote:
> You can not preserve calls from MGCP to SRST.
> In the IOS of the code on the lab I dont think you can preserve the calls
> from H323 to SRST (even though they are 1 in the same)
>
> Thanks,
> Ryan Trauernicht
>
> On Tue, Jan 20, 2009 at 1:52 PM,
You would use said translation-rule (10) if you don't want to set ISDN
numbering plan or numbering type on outgoing calls.This might be a
requirement by PSTN provider.However, this will not strip ANI.
To strip ANI , use clid restrict under your outgoing dial-peer.so,in your
case,configuration wil
Is it possible to preserve calls from Br1 MGCP gateway when it falls back
into SRST mode ? I know calls can be preserved while using H323 gateway but
not using MGCP gateway since L3 binding is terminated from gateway to CCM
when falling back to SRST mode and hence calls cannot be preserved ?
Any t
Wondering to know if someone can give me a clue ?
Also, If assumingly, one of CCM server dies, will these services still work
? Does CCM provide redundancy in b/w pub and sub for EM/IPMA etc.
On Sat, Jan 17, 2009 at 10:37 PM, kamal yousaf wrote:
> Hi,
>
> Assuming *10.1.10.5* =
'call-forward max-length 4'
You can restrict either under individual ephone-dn Or system-wide.
On Mon, Jan 19, 2009 at 3:49 PM, karuna durai wrote:
>
>
> Hi,
>
> You can do using alias command as below...
>
>
> alias 1 1001 to 1001 preference 1 cfw 2001 timeout 20
>
>
> karuna
>
>
> On Mon, Jan
Hi,
Assuming *10.1.10.5* = Publisher (Backup Call Processing Engine) /IPCC
*10.1.10.6* = Subscriber (Primary Call Processing Engine)
We have a number of parameters in UCM where we assign ip addresses for
services to operate properly, I believe we assign Services Like EM , IPMA
(
Ph1 and Ph 2 should be members of GDM and Ph1 should be owner as
well.Remember,you would need to add a button(line) using GDM Extension on
Ph1 and Ph2 for lighting up MWI there in order to check messages.
On Thu, Jan 15, 2009 at 10:30 PM, Kumar, Narinder <
narinder.ku...@uxcg.com.au> wrote:
> Al
equires
> placing MOH servers in G.711-only region/DP as per
> http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configuration_example09186a00803f8950.shtml
> which
> I haven't mentioned in my previous email.
> Rgds
> Alex
>
> - Original Message
Self-Study, Classroom-Based,
> Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE
> R&S Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and
> CCIE Storage Lab Certifications.
>
>
>
>
>
>
>
> -
Alex,
Regarding your comment, If MOH server uses 239.1.1.3 to stream G729 to
remote phones, shouldn't we enable G.729 for MOH ? Only exception is using
transcoder but since it can't be used for multicast,don't we have to enable
G.729 ?
>
> Alex schrieb:
>
>> Your mcast group IP@ in below debug
Hi ,
I have MOH Sub and Pub configured to support Multicast for Br1 phones and
Unicast for HQ phones (using MRGL).I placed MOH sub in G711 only device pool
so that it communicates using G711 only. Now, since BR1phones/BR1 MGCP gw
are in a device pool which communicates G729 to other device pools,
Hi All,
In workbooks, integration b/w *unity* and *cue* is accomplished by creating
a *New Lookup Zone* in addition to A and MX records.However,this is a
redundant step and can be avoided. Now, if integrating between Unity to
Unity,does this step become mandatory ? I assume "YES". Please comment
Hi,
Can anyone guide me about which steps can be done to troubleshoot VPIM
integration from Unity to other Servers(Unity,CUE). I have unity-to-unity
integration done exactly as per documenation.All components(voice
connected,smtp,ad schema) are installed and OK.VPIM messages are sent from
each Un
Hi,
When i use IPCC Agent login using IP Phone Services, it logs me in
successfully and Screen shows "Not Ready". When i hit "update" to enter
into "Ready state",it throws error saying "host not found".
I have double-checked all service parameters in CCM and they are pointing
to right IP addr
Hi ,
If we are required to configure CME such that all outbound calls through
PSTN use same E.164 number i.e 9197684000, this can be manipulated using
Voice Translation Rules. However, if its not allowed to use Voice
translation rule, how will this be achieved? Can we use station-id number
91976
Thanks Alex for invaluable input !
On Tue, Dec 30, 2008 at 6:44 AM, anil batra wrote:
> UR DA MAN Alex. HEre is your CCIE number XX007 :)
>
> --- On *Tue, 12/30/08, Sergio Polizer * wrote:
>
> From: Sergio Polizer
> Subject: Re: [OSL | CCIE_Voice] Forwarding a PSTN incoing call to
> extension's
ny other possible combinations. This is one of the possible
> resolution.
>
>
>
> On Mon, Dec 29, 2008 at 10:16 AM, kamal yousaf
> wrote:
> > Hi,
> >
> > If we have 2 dial-peers,
> >
> > One with:
> >
> > dial-peer voice 1 voip
&
Normally, all required files B-ACD scripts and audio files will be available
on flash.However, if you need to copy them over , simply use archive
command:
archive tar /xtract tftp://x.x.x.x/cme-bacd.2.1.0.0.tar flash:
This will extract all files to your flash.
On Mon, Dec 29, 2008 at 5:54 PM, a
Hi,
If we have 2 dial-peers,
One with:
dial-peer voice 1 voip
destination-pattern 24004001
session target ipv4:10.10.10.10
incoming called-number 24004001
codec g711ulaw
no vad
dtmf-relay h245-alpha
2nd:
dial-peer voice 2 voip
destination-pattern 2...
session target ipv4:10.10.10.10
Hi,
While integrating Unity with CCM cluster, there has to be a primary CCM
server and secondary CCM server. Will Publisher fit into primary OR will the
primary processing node (Subscriber) fit into primary settings of Unity ?My
thought is it has to be publisher and not the primary call processin
If you record message in unity ,copy over to TFTP directory in CCM, then
move it to flash on CME router, you would have to reload audio prompt.
audio-prompt load flash:en_bacd_welcome.au
//If an audio prompt file has been changed, reload it.
Rgds
On Fri, Dec 26, 2008 at 6:00 PM, anil batra wrot
Hi,
I have setup B-ACD script on my SiteC router.When i call from SiteC phones
or test using 'csim start 4000', it works great. From SiteC phones, i can
hear sound and options too.
When i call script from HQ site i.e phone registered to HQ site , call
works and script answers. Yet, no sound is
Hi,
I am here referring to GK Vlecture. Now, though Remote PSTN GK is out of
scope in CCIE lab i.e we don't require to configure it but for my home lab,
i need to set this up.
Scenario was like this :
When remote Zone was configured , Call was routed in this fashion:
CCM >GK Local(CCM)--
Hi,
Can someone proivde me with CCM/IPCC clean install script . I have plenty
of configurations and every time when i need to start over, i have to revert
back to older snaphost. Please provide me with .bat script file.
Thanks.
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