Hi all,
I have made two vRack sessions so far based on Remote Phones and
Softphones. For my next session I want to use the hardware vpn session.
I have a 2811 with 4-Port Hwic and 3 7962 phones, but as written on
proctorlabs it seems that this is not enough (requirements are 5 phones
and a
Hi all,
this question is not related to the CCIE labs, but hopefully you can
help me to find an answer.
When a caller reach a user voicemail and choose the caller input option
to be forwarded to an alternate number (in this case the mobile number)
I can see in the isdn debug that extension
I found it,
CUCM Service Parameter:
Display Original Calling Number on Transfer from Cisco Unity – Value needs to
be set to true.
Von: Cristobal Priego [mailto:cristobalpri...@gmail.com]
Gesendet: Donnerstag, 15. September 2011 03:55
An: Steffen Bruening
Cc: ccie_voice
Hi all,
I have the BLS from IPexpert and I feel very comfortable with the 10 labs
of it. Now I got a marketing email for 5 extra labs made by Ipexpert to
learn for the real lab.
www.ipexpert.com/cisco/ccie/voice/handbook
Does somebody of you buyed this product? Is it it worth? What are the
Hi all,
there is a picture missing on page 197 in the solution guide, which should
show how to setup the trace settings. Can somebody explain me how I should
setup the trace?
Regards
Steffen
___
For more information regarding industry leading CCIE Lab
Hi all,
during all my last 4 proctorlab sessions I missed phones connected to the
switch/switch-modules. Often only one phone per site is available,
sometimes a complete site has none. It is really really annoying to find a
solution about how to bring enough phones (own physical devices or soft
-More
Phone: +1.810.326.1444 ext 206
Email: jd...@ipexpert.com
Twitter: www.Twitter.com/IPexpert
Check out OUR CATALOG: http://www.ipexpert.com/catalog
On Jul 13, 2012, at 1:47 AM, Steffen Bruening stbruen...@gmail.com
wrote:
Hi all,
during all my last 4 proctorlab sessions I missed phones
Yes, the first three commands I put in on every device are, cdp run, cdp ad
and cdp t 5.
2012/7/16 Ronmac ron...@solcon.nl
Hi,
Maybe stupid remark but,
Is cdp enabled. AdverV2!
V1 os not ok
Regards ron
Send from my mobile
Op 16 jul. 2012 om 12:41 heeft Steffen Bruening stbruen
Hi all,
in section 5.1 is written Configure a transcoder on Site A gateway...Site
A devices *must have the privileges* to use the transcoder when necessary.
Maybe somebody can help me to get a better understanding on this. Does this
mean the Site A is the only site which should allow to use it,
Hi Ramy,
does the question say that you should preserve this? There is no iDivert
softey in CME/SRST 7.X. iDivert is supported for SIP Phones in CME with 8.5
or so but not for phones falling back to SRST. You can only preserve the
feature with workarounds like this:
Transfer a call directly
Hi all,
when ever I go through the call routing section I find several ways to
answer the question, some straight, some inconvient. But I don't know for
what I can get the points. Does the way matter to reach the goal/answer the
question (as long as it not vialote the question itself)? Or is
Hi,
I don't know whether you are using your own equipment or remote
proctorlabs. You should check your show ip route output. Also you can try
to restart the dhcp monitor service or to disable the cisco security agent
with utils csa disable (server reboot required).
2012/8/29 Krishna
Hi all,
this question is not ccie lab related. I have 2911 integrated via sip to
cucm 8.6. When a call terminates because of user busy/unallocated number or
what ever, the cisco phone still rings for 30 seconds because the sip
disconnect message was send 30 seconds after pstn disconnect. How
Krishna.
--
*From:* Steffen Bruening stbruen...@gmail.com
*To:* ccie_voice ccie_voice@onlinestudylist.com
*Sent:* Thursday, August 30, 2012 4:56 AM
*Subject:* [OSL | CCIE_Voice] sip gateway PSTN disconnect cause delayed
send to cisco phone
Hi all
Hi all,
I know that usually we would configure multicast moh for CME in this way
multicast' moh 239.1.1.1 port 16384 route . I read through the CME
admin guide and found the following sentence:
*port port-number—Media port for multicast. Range is 2000 to 65535. We
recommend port 2000 because
Hi all,
I have a fractional PRI with MGCP (8 channels). When I look int sh voice
dsp group all I could see that 8 channels of dsp 5 are used. Maybe someone
can explain how the dsp allocation algorithm is working. I would like to
know why not dsp 1 will be used. Is there maybe an option that I
media port numbers using voice
moh-group number command. Document Author seems referring to the
live-feed with media port 2000 inside telephony-service configuration.
*From:* Jason Aarons (AM) jason.aar...@dimensiondata.com
*To:* Steffen Bruening stbruen...@gmail.com; ccie_voice
ccie_voice
Hi Krishna,
you will not see Cos 5 in Threshold 1 because this is the default Threshold
for Cos 5. The running-config presents always only the non-default values,
e.g if you put an prefererence command on your dial-peers you will this
only as long as it is not the default value 0.
Regards
On Oct 20, 2012, at 4:32 PM, Steffen Bruening stbruen...@gmail.com
wrote:
Hi,
I have 3 Sites, all of them configured on the CUCM, Site C has Voicemail
with local CUE. When I am dialing from Site B to C codec g279 will be used
and I can reach the voicemail, so I know When I am dialing from
I have this seen this also, to be honest I think it shouldn't matter
whether it is in threshold 1 or 3 as long as no other COS is in same
Threshold of queue 1 of queset 2. When you leave in in threshold 3 I think
you should be fine with:
mls qos queue-set output *2* threshold 1 100* *100* 75
where it
will take/borrow the memory value from reserved threshold when desired.
long story short... right way of doing it either assign it to t2 or t1 and
assign threshold value of 75% for correct approach...
thank you
krishna.
--
*From:* Steffen Bruening
Hi,
you don't need such a bunch of dial-peers for CUE
You are using MWI outcalling, therefore you only need the both DN's also
you can send Voicemal and AVT on the same Dial-Peer to the CUE, use 400[45]
as destination-pattern.
If outcalling is not request in the question I would prefer
Q931 doesnt lie. Something is missing in your config, maybe on the cucm,
maybe on the gateway (for h323). It depends on where your stripping your
DIDs to the internal number format.
2012/10/30 otunola Akerele otunola.aker...@gmail.com
hi all, am new to the forum, ples am having some issues am
Recommed by whom? Ipexpert, I know, but the problem is that the
Proctors/Scripts which marks your lab are not from IPExpert they are from
Cisco. Therefore it should be better to follow the cisco guidelines:
You should configure voice VLAN on switch access ports; voice VLAN is not
supported on
Here you can find some scripting guides:
http://www.cisco.com/en/US/products/sw/custcosw/ps1846/products_programming_reference_guides_list.html
2012/11/1 sanity insanity networksanitytoinsan...@gmail.com
hi Guys,
I really need your help to understand UCCX scripts ...How they are made?
and
Oh okay. I apologize for the confusion I made.
2012/11/1 Cory Gray corygray22...@hotmail.com
IPexpert and myself have run into problems doing it the traditional way.
But for the authoritative source you should be looking at the LAN Switching
Guide for IOS 12.4T which confirms that
Steffen
2012/11/1 Steffen Bruening stbruen...@gmail.com
Oh okay. I apologize for the confusion I made.
2012/11/1 Cory Gray corygray22...@hotmail.com
IPexpert and myself have run into problems doing it the traditional
way. But for the authoritative source you should be looking at the LAN
Hi all,
I was in Bruessels yesterday and I failed. I was suprised about my score
report and about the amount of section were lost points. I only know one
definitly mistake, about the other things I could only suppose.
It seems that I am just to stupid to read the questions. I was finished
with
Hi Bill,
When you ask for callers ahead of you, you should go with
ContactsWaiting+decrement step, because position in Queue is your position
and when nobody else is in the queue you will hear a - 1 after the
decrement, don't know whether there is also an increment step.
At the end it is your
Hi Bill,
I had my first attempt in November, but failed because I don't read
some question corectly
in every detail.
I had this problem with -1 in some of my lab sessions (not proctor
labs) and it worked with contacts waiting for me. I tested it with 3
call-in users and all get an indivudually
Generally it doesn't matter to which you bind your media resources. But
maybe the question ask you for a specific binding.
The only thing you never should do is to bind Cube and Gatekeeper to the
same Interface.
Regards
Steffen
Am Samstag, 1. Dezember 2012 schrieb virajith :
Hi All,
I am
the the routing as well as traffic from loopback or
virtual interfaces needs to reach the callmanger and other devices also if
the routing needs to setup then this is going to eat into the allocated
exam time?
-Vir
From: Steffen Bruening stbruen...@gmail.com javascript:_e({}, 'cvml',
'stbruen
Thats not allowed via CLI, so no syntax available.
Am Montag, 3. Dezember 2012 schrieb Ramcharan Arya :
Hello,
Can you someone please tell me command syntax how to restart Cisco Tftp
service from CUCM CLI.?
Thanks Regards,
Ramcharan Arya
CCIE # 28926 (RS)
Hi,
I would always go with a pots dial-peer.
Your external BACD dial in number is truncated to +343500 and therefore
it could not find a matching dial-peer, check your voice translation.
Am Montag, 17. Dezember 2012 schrieb Ramcharan Arya :
Hello,
I have configured BACD on Brach2 site
Hi,
you need one identy document, passport for example. Lunch is included in
the exam fee, you will get a coupon for one meal, salat, drink and
dessert. In Brussels waa also a machine near the exam room were you get
water and tea for free during the exam.
Regards
Steffen
Am Donnerstag, 20.
7962s are not guranteed. HQ SW is connected with one 7960 and one 7962,
site B also. Only site C has two 7962s at most of the PODs. The problem is
that for SiteB and C the phones not always connected directly, sometime
they are remotly connect via vpn, therefore the port is up, but cdp shows
Why is predot trailing needed for mgcp? I use always only predot and it
works perfect.
2012/12/22 Chrysostomos Christofi ch.christ...@logicom.net
Ramcharan , hi
** **
Both route patterns for the below are correct
It depends what the question ask , or what is your
What so you want to see? You are calling from a PSTN number into the XML
Script. When you call a internal number through the mva, you will the
number am when the destination hooks up and the call is connected you will
also see the name.
Am Sonntag, 23. Dezember 2012 schrieb Rrcrumm :
Hello
I
of hqph1 I just see the extension 3001 is
calling, no name.
Should I see a name and number?
Thanks
Randall
On Dec 22, 2012, at 10:36 PM, Steffen Bruening
stbruen...@gmail.comjavascript:_e({}, 'cvml', 'stbruen...@gmail.com');
wrote:
What so you want to see? You are calling from a PSTN number
Hi singh,
connet to the cue, check/install license then put the module offline (conf
t, offline) when it is in offline state run restore factory default and
wait till it ask to press any key for reload. After the reload it will go
through the intial configuration wizard, when you completed that
When they ask for 10 digits you have to send only the last ten digits. your
RP looks good, on your dial-peer you don't need the forward digits command,
because 9 and 1 are the only digits which are clearly seen they will be
stripped automatically
Am Samstag, 29. Dezember 2012 schrieb sanity
I can confirm it is SecureCRT, but an old version without tabs. I training
with desktop shortcuts to the device connections which opens in seperate
windows.
2012/12/29 Marko Milivojevic mar...@ipexpert.com
I *think* Voice lab is still using SecureCRT. Does it matter whether
it's SSH or
Maybe it is a good idea to use just the default profile for all the phones
which have CUCN mailbox and only to add a new profile to the users who get
a CUE mailbox.
2012/12/29 CCIEing aboaz...@gmail.com
Hi virajith,
You may update all your phone settings using bulk edit.
On Sat, Dec 29,
There is a trial version for 30 days.
Am Sonntag, 30. Dezember 2012 schrieb singh :
hi Guys,
Thanks for your inputs.
I generally use command prompt and putty to access my routers and servers
hence I am not sure how different is secure crt from these terminals.
Besides this I see it is
Did you see the call incoming on the cme? debug voip dial-peer?
Am Freitag, 4. Januar 2013 schrieb Cory Gray :
All,
** **
I noticed during the following setup that I cannot call DIRECTLY into CUE
but if SiteC-Phone1 is busy or does not answer, forwarding the voicemail
works. I am not
The problem with your example is that the frame-relay ip rtp
header-compression is out-dated and the header compression should be in the
policy-map in the rtp class.
Also I don't see any less flexibilty in using auto qos, you can adjust all
the values as with the manual way.
Regards
Steffen
Am
No grading is not affected, because it is allowed to use manual qos config
and so you can use whatever you want for the names.
Am Samstag, 5. Januar 2013 schrieb Pixar Perfect :
Ignore the RTP header compressions part. My question is not technical but
from strategy point. Do you think the
I went on Thursday to Brussels for my 2nd attempt and I got my results
suprisingly today. I passed. CCIE 37891.
I like to say thank you to all on you on the list which shares your
questions and answers about the blueprint task and ipexpert labs.
I also wanna say thank you to IPexpert for their
Its not allowed to bring any electronical device into the exam room.
Am Samstag, 5. Januar 2013 schrieb Marko Milivojevic :
I would guess not. If you're concerned about it, you could open a case
with the certifications support.
--
Marko Milivojevic - CCIE #18427 (SP RS)
Senior CCIE
The dbreplication issues which you have at ipexpert/proctorlabs session you
can fix with an easy utils dbreplication repair all, nothing more.
Am Samstag, 5. Januar 2013 schrieb Chrysostomos Christofi :
Hi Ram
** **
** **
First check in PUB
** **
utils network connectivity
You don't need to write a complete cue config, most is done with factory
defaults, only focus on integration, so ccn subsystem jtapi ( for cucm) ccn
subsystem sip (for cme and srst) ccn trigger for sip and jtapi and how to
create users and voicemail boxes.
Am Samstag, 5. Januar 2013 schrieb
No don't do that, also don't add or remove domain names.
Am Sonntag, 6. Januar 2013 schrieb sanity insanity :
hi Guys,
I am just wondering if we are allowed to change host names on the routers
. For example if the changing the router name on R1 to HQ and so too for
the branches.
-MJ
Hi Nicolas,
I had this problem too on some pods, where only one of the HQ phones want
to register, same for site B, dhcp addressing is working but no
registration occurs. Because I had no physical access on the phones to
check what the display is showing ( I guess something like registration
Both is available on the desktop (in Brussels)
Am Mittwoch, 9. Januar 2013 schrieb Abdullin Kamil :
Hi men,
On lab what program is used for remote access on UCCX?
windows remote desktop, VNC or any other?
Thanks!
___
For more information
You can define a gw-priority for the trunks to each endpoint.
Am Mittwoch, 9. Januar 2013 schrieb Ramcharan Arya :
Hello,
I have a small query about Gatekeeper in single zone configuration it is
prefer PUB as call processing ( as per ARQ message).
if lab requirement says make subscriber as
I always put it down to 10s, not less because sometimes the wan connection
is flapping/very low and so the phones flapping between cucm and srst.
Am Mittwoch, 9. Januar 2013 schrieb Pixar Perfect :
Hello,
is it advisable to bring down the Connection Monitor Duration under Device
Pool to a
Maybe you configured more channels for the controller as available on the
PSTN Router Inbound calls are taking the first channel, outgoing the last
channel.
As far as I remember the HQ router on proctorlabs has only 6 channels, site
B and C have 4 channels.
Am Donnerstag, 10. Januar 2013 schrieb
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