http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configuration_example09186a008076f8d2.shtml
Old versions, but has everything you need.
regards
Peter
On 4/28/2013 8:00 AM, CISCO CCIE VOICE wrote:
HI Thanks,
but if suppose i want to use 6 channel from T1 PRI on HQ Router and
hello peeps
please who can be kind enough to share cvl labv3 guide with me please.
its is urgently needed.
thanks soo much
pete
___
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Are you a CCNP or CC
I'd like to see what difference it would make to my times by trying this
(radically) different approach!
regards
Peter
Peter Simmons
On 4/3/2013 5:59 PM, William Bell wrote:
I am not familiar with Marko's approach for "on-screen" window placement.
I actually don't have a s
Hi Friends,
There is now new lab in HK my friend got it on 27th March be careful
andifanyone want to be study partner please email me back so that we can
talk more.
we have got few questions now.
thanks
___
For more information regarding industry leadi
hi guys,
call to users vm works just fine but when transfering to a vm extension it
drops. pls what could be the problem??
___
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www.ipexpert.com
Are you a CCNP or CCIE and l
Got lab 7 in SJC very big lab i completed in 6 hrs time was having full
hopes will clear the test
Lost in few sections like VG i got only 53% and HA less marks and did not
got marks in LAN qos.
I did exactly the way cisco expected this is really bugging me about VG
section as it was total 17 mark
so that we can talk more rest all
sections i got 100% so if needed i can share the solutions.
On Sat, Mar 23, 2013 at 3:57 AM, Josh Petro wrote:
> Good luck and God Bless!
> On Mar 22, 2013 10:39 AM, "Peter Rody" wrote:
>
>> Hello friends,
>>
>> Ready for
Hello friends,
Ready for the attempt just starting from hotel now its long practice please
pray for my attempt.
Everyone comes out say i pass its feel so good but i think i might be first
guy need your prayers starting from my hotel now. Not able to sleep whole
night properly.
Feel like on a top
Sounds like you didn't set the MVA details under Media Resources.
I usually get these symptoms if I forget to add this.
CUCM Administration/Media Resource/Mobile Voice Access - set the MVA
number and locale.
Let us know how you get on?
regards
Peter
On 3/17/2013 12:53 PM, sanity ins
the ANI in SRST/CME.
You call as before, press "transfer" and then the speed-dial - it then
forwards "blindly" (!).
Let us know how you get on.
regards
Peter
On 3/15/2013 3:11 PM, Suresh Bhandari wrote:
I had the voicemail callerid configured in cueeven then there is
hi guys
i have tried the dbreplication force and dbrep repair and and my
subscriber is stil not replicating, please i have been on this issue for a
while now and the issue is not resolved yet.please what else can i do??
peter
___
For more
hi guys
am using proctorlabs and ipcc extension is missin on the end user page i
tried a crs "FRESH_INSTALL" and i cants stil stil not there. please what
should i do???
thanks
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u have multiple DSPs installed across several PVDMs, then you might
try :
voice-card 0
dspfarm
dsp services dspfarm
if they're all on the same module.
First thing for us to see would be output from"show inventory" I reckon
we can look a bit more closely at why you're out o
Sorry, in my haste I forgot to say thankyou to everyone who offered
suggestions, either via the list or privately - all appreciated!
regards
Peter
On 10/25/2012 3:06 PM, Peter Simmons wrote:
Dear all,
I have resolution of sorts for the VVX issue I was seeing.
I have this working now - with
I shut down the sub, and
when that was instantly resolved I tried the VoiceView Express message
playback again and it worked fine.
Oh well, if it doesn't kill you.
I hope this helps someone avoid a few hours of pointless effort :-)
regards
Peter
On 10/20/2012 9:45 AM, Peter Sim
uter: 2811 IOS 12.4(24)T4
I'll do some traces on CUE today and see what that shows up, if
anything, then next up I plan to see if it works using CME integration,
that'll be for next week though :-)
Many thanks for coming back to me on this, your input is appreciated.
regards
Peter
O
CUE module for a while and see if this
draws anything else out I haven't seen already.
Again, thankyou for your feedback - I appreciate it.
regards
Peter
On 10/19/2012 11:36 PM, Dan Quinlan (daquinla) wrote:
Try adding "standard CTI enabled" to the jtapi user. I've seen issues w
RTP User ID Phone MAC Address
4001No SCPhone100C8.D0B1.C0E9
1 session(s)
0 active RTP stream(s)
CUE#
I'm pretty much out of ideas, so happy to hear suggestions from the list
members on this if you have them.
--
regards
Peter
a pity the " features blocked Confrn" option in the ephone
template doesn't do anything for Meetme conferences, it would seem to be
the thing to use otherwise.
regards
Peter
On 10/17/2012 9:32 PM, Kevin Spicer wrote:
Another slightly different idea...
Instead of a dummy epho
set differently from you, but
these are the settings I alter to make this appear or not appear in my Lab.
What version of CUCM are you running? I am slighlty "up level" from
7.0.1, so it may work in my lab but not in _the_ Lab, if you follow me :-)
YMMV, but hope this helps.
regards
Try to add:
!
voice service voip
sip
midcall-signaling passthru
!
Peter
- Original Message -
From: "Mohd Baqari"
To: "san r"
Cc: ; "The Masterplan"
Sent: Wednesday, May 23, 2012 9:15 PM
Subject: Re: [OSL | CCIE_Voice] Gatekeeper trunk
Hi,
I
+ unity with *
IPCC was a new script transfer to operator by mistake :)
So all is set done i am very happy
Guys can email me if anyone needed anything i will surely help :)
Regards
Peter
___
For more information regarding industry leading CCIE Lab
Hi,
I am looking for the partner for purchase
If anyone interested please email me the same.
I just found the link if anyone is interested we can share the cost.
http://www. c e r t k n o w l e d g e
.com/forum/index.php?/topic/24-gb-ccievoicelabscom-real-labs/
Thanks
On Sat, Feb 4, 2012 at 8
Hi,
I am looking for the partner for purchase
If anyone interested please email me the same.
I just found the link if anyone is interested we can share the cost.
http://www. c e r t k n o w l e d g e .com
/forum/index.php?/topic/24-gb-ccievoicelabscom-real-labs/
Thanks
__
Hi,
I am looking for the partner for purchase
If anyone interested please email me the same.
I just found the link if anyone is interested we can share the cost.
http://www. c e r t k n o w l e d g e
.com/forum/index.php?/topic/24-gb-ccievoicelabscom-real-labs/
Thanks
Peter
Chris,
Looks to me like you need the "switchport voice detect" command here.
Regards
Peter
On 09/12/2011 07:57, datucha123 datucha123 wrote:
Can you please update the solution to this question, when you will get
it to work
On Thu, Dec 8, 2011 at 7:50 PM, Chris Martin ma
Ashraf i think you are from rea l lab company as we are sharing it for free is
paining you ha ha i really dont want yr return comments bec all is useless
until u are nt a ccie dont say anything heheheh
yes there is lab 6 out now and we have to work on it to clear the lab there is
srst cor also
Here is something new which i was stucked with
2.2 IP Phone customization (Part I) Directory Services
SB Phone 1 user is alleging unauthorized access of his Corporate directory
services from his phone and has asked to disable access to his IP Phone’s
Corporate Directory. You have management a
From: Julien Krieger
To: Randall Crumm
Cc: Peter Jeff ; Mark Reed ;
"ccie_voice@onlinestudylist.com" ; Ken Wyan
Sent: Tuesday, November 22, 2011 2:09 PM
Subject: Re: [OSL | CCIE_Voice] lab - 6
Hi Randall,
I agree with you don't want to go this road. Certainly Voice Lab
Ha Ha,
Always r eal lab question is different that is why it is r eal lab question :)
Ha ha
I have full lab now :)
Thanks
From: Randall Crumm
To: Peter Jeff ; Mark Reed
Cc: "ccie_voice@onlinestudylist.com" ; Ken Wyan
Sent: Tuesday, N
thanks marks easy one
From: Mark Reed
To: Peter Jeff
Cc: Ken Wyan ; "ccie_voice@onlinestudylist.com"
Sent: Tuesday, November 22, 2011 3:22 AM
Subject: Re: [OSL | CCIE_Voice] lab - 6
The answer is in this document.
http://www.cisco.com/en/U
MGCP TS lab (6) what we can do for one way audio explanation on debugs
thanks
___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
Are you a CCNP or CCIE and looking for a job? Check out
www.PlatinumPlac
>
>
>
>On Sun, Nov 20, 2011 at 2:22 PM, datucha123 datucha123
>wrote:
>
>Do you mean under the DHCP configuration, that such question has been on the
>exam?
>>
>>What do you mean under the SIP early offer? or MGCP TS?
>>
>>
>>
&
you mean under the SIP early offer? or MGCP TS?
>>
>>
>>
>>
>>On Sun, Nov 20, 2011 at 8:24 PM, Peter Jeff wrote:
>>
>>Hi Guys,
>>>
>>>
>>>It was my 5th attempt and i got lab 6 frustration is on peak everytime i
>>>went for the lab i get
Hi Guys,
It was my 5th attempt and i got lab 6 frustration is on peak everytime i went
for the lab i get new lab
From last 2 months all my JR guys passed in dubai bec they got lab 3 and lab 4
in Dubai since last 2 months i saw lot of guys cleared from dubai
Now lab 6 i am so so so frustrated
Hi Guys,
It was my 5th attempt and i got lab 6 frustration is on peak everytime i went
for the lab i get new lab i dont know why
From last 2 months all my JR guys passed in dubai bec they got lab 3 and lab 4
in Dubai since last 2 months i saw lot of guys cleared from dubai
Now lab 6 i am so
Hi,
Is it best practice to configure "ccm-manager switchback immediate" command
while configuring to change the default behaviuor of "Graceful" method?
In the LAB, do we lose marks if I configure this, even if this is not
explicitly asked?
Peter___
Hi,
You mean the Display Name for DN associated the RDP, right.
I have tried that already, no luck?
Peter
On Mon, 18 Jul 2011 00:47:50 +0530 wrote
>Add display name to rdp and let me know if it helps
Sent from my iPad
On Jul 17, 2011, at 1:14 PM, "Peter" wrote:
Hi,
Hi Ashraf,
Thanks for the response.
My question is, if there is no specific requirement in the Gatekeeper section
of the lab to have priority for Subscriber, are you supposed to do this
configuration using gw-priority command?
By doing, Are you going to lose any marks?
Peter
On Sun, 17 Jul
Hi,
MVA has been configured for BR1 which is a H.323 Gateway. Users are getting
authenticated and are able to make calls. But for the outgoing calls, calling
name is not getting displayed. Calling Party ID is going as per the
configuration. Only the calling name is not going to the PSTN Network
Hi,
In the lab, suppose there is a general requirement like Subscriber should be
the primary call processing with Publisher as backup.
When you have a setup like HQ as CUCM and BR2 with CUCME, and Gatekeeper
between them. There is nothing specific mentioned in the Gatekeeper section,
related
Hi Shabeeb,
Thanks for the help.
Can you please let me know why Point no. 1 is invalid? Does it violate any of
the requirements?
Peter
On Sun, 17 Jul 2011 10:06:36 +0530 wrote
>point number 2 should be enough..
On Sun, Jul 17, 2011 at 7:18 AM, Peter wrote:
Hi,
Suppose there is s
ed extensions?
2) By removing the Park or Meetme softkey from other phones
3) Both of the above.
Peter ___
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www.ipexpert.com
Are you a CCNP or CCIE and looking
Hi,
Is there any way by which I can generate a RTMT Email alert only once.
The solution in which the alert frequency being kept as 1 for 999 (maximum)
minutes doesn't seem to be convincing, because it will generate only 1 alert
for 999 minutes regardless of the number of occurrence of the event
Corporate Directory
Missed Calls
Personal Directory
Placed Calls
Received Calls.
Since the order is different, will it result in losing marks?
Or anybody has better solution ?
Peter ___
For more information regarding industry leading CCIE Lab training, pl
George is asking the right questions - the gateway is most likely not
wanting to do the appropriate variant of g.729. Let's see that gateway
configuration =) The way he's saying to fix it is most likely correct.
-Pete
On Fri, Jun 3, 2011 at 10:24 AM, Brian Mulgrew wrote:
> cuc sccp vmail port
This thread shows blatant disregard for the CCIE NDA. You really
should not be posting your lab questions to anyone, let alone an
entire user group.
Please be more careful,
Peter
On Thu, May 19, 2011 at 12:54 PM, faustin ntchana wrote:
> Hi ,
>
> If I can help please make sure t
In CME busy-trigger is the same common value for the shared line. Let it be
only one then if either instance serve a call it does not hunt for an idle.
This differs in UCM where all line is independent in case of busy-trigger.
Peter
- Original Message -
From: Roig Borrell, Francesc
Hi all,
I have some q about auto-registering in CUCM:
- If default auto-registering protocol is changed than the whole cluster is
needed to be restarted or only the node on which this function is enabled?
Ex.sub obly.
- After migrating a phone from SCCP to SIP via BAT enabling auto-register SI
also when i said "international escape" i was thinking "office code."
Oops. Hey, it's late here.
-Pete
On Sat, Mar 19, 2011 at 4:32 AM, Peter Slow wrote:
> I think the most concise way of describing its purpose is to say its
> "job" is to provide a c
I think the most concise way of describing its purpose is to say its
"job" is to provide a context, if you will, for the phone number.
Since there's variation in numbering formats as they're programmed
into various systems, switch B needs switch A to tell it if the first
couple numbers are a countr
t 7.1(5) or the like would run. You generally dont need to know
these exact numbers. the CUCM version you configure for the ccm
command in IOS determines what features your media device assumes CUCM
is able to support.
-Peter
On Tue, Oct 12, 2010 at 1:32 PM, Vik Malhi wrote:
> HQ-RTR#sh scc
the MAC address only needs to be used for the old style transcoders
that used the original PVDMs. on these types of devices, there was no
sccp group /" associate profile 1 register" command TO configure the
name it'd register as.
the people that did all the documentation kept that practice alive
u
region and the called device's dp's region
...do all devices support the required codec?
-Peter
On Wed, Sep 8, 2010 at 5:18 AM, Ki Wi wrote:
> I'm currently doing V1 Lab 5C where Q5.2 requires a transcoder.
> If i placed the transcoder in DP_HQ while i purposely put the SIP
please show us "sh ephone sum" and a show run. both from teh branch
router, and tell us what interface the phone is on.
-Peter
On Fri, Jul 2, 2010 at 7:52 AM, Afzal Bhutta wrote:
> Sorry Folks not providing details in first attempt.
>
> Thanks for all and special thanks t
also set 'transfer-pattern .T' under telephony-service to allow transfer to VM
- Original Message -
From: Phillip Day
To: ccie_voice@onlinestudylist.com
Sent: Wednesday, June 23, 2010 12:13 PM
Subject: [OSL | CCIE_Voice] Simple transfer to VM issue
Has anyone got a sensible
You can record a spoken name via AvT menu. It works for me.
- Original Message -
From: Angel Perez
To: osl osl
Sent: Tuesday, June 15, 2010 2:23 PM
Subject: [OSL | CCIE_Voice] CUE caller id
Hi:
I was trying to setup CUE to say voicemail user name instead of phone num
Any consideration why the recommended method is using a CTI port instead of a
hunt pilot?
If a hunt pilot is configured for AA then a user sees the call as a directed
rather than forwarded which happens in case of CTI.
- Original Message -
From: Peter Farkas
To: ccie list
Sent
I get the sccp service down on HQ which is an RSVP agent. After this when the
CUC would reach (by MWI or transfer) a phone at BR1 AAR is activated. In my
opinion AAR CSS of a VM port has to take effect in this case.
However this CSS is here but MWI still works. In addition a call can be
transf
Supposed method to call in a CUC from UCM is via a CTI RP which is forwarded to
the Voicemail.
However my solution was to set up a new hunt pilot that points to the same hunt
list that was created for VM. I've also created a new route to the AA's Call
handler in CUC Direct Call Routing Rules t
In case of ESW the suggested method is trunk mode.
http://www.cisco.com/en/US/docs/ios/lanswitch/configuration/guide/lsw_hwic_ethsw_ic_ps6441_TSD_Products_Configuration_Guide_Chapter.html#wp1051730
- Original Message -
From: Peter Farkas
To: Angel Perez
Cc: osl osl
Sent
I also cheked mine. Further interesting these different output of the same
config.
BR2-RTR#sh vlan-switch
VLAN Name StatusPorts
- ---
1default activeFa0/1/2
ccess international but not 911. :(
the right syntax under call-manager-fallback:
cor incoming default
I hope it helps.
- Original Message -
From: Ashar Siddiqui
To: Peter Farkas
Cc: ccie_voice@onlinestudylist.com
Sent: Monday, May 31, 2010 8:19 PM
Subject: Re: [OSL | CCIE_Voice
If u do not attach any incoming COR to the SB2, as can be seen in the config,
then it can access all dial-peers. For this reason it is a good practice to
config a default incoming cor under call-manager-fallback to restrict.
SB1 has correct configuration, it also can access dial-peer 9011.
Conference is supported at HQ site but CIPC(SIP) IP phone cannot use
Select/Join softkey to bulid up a conference. It failes with Unavailable
Feature message on the display. However Confrn softkey works as expected.
CIPC or SIP supports buliding a conference by Select/Join method, at all?___
Display ID of RDP's DN is missing. When shared line is created then only the
Alerting Name is copied from the line. Go to the DN Configuration of 5002 and
select the RDP from Associated Devices list and use Edit Line Appaearance
button to modify.
- Original Message -
From: Matthew B
>From Ben at ask the expert forum:
"Please see answers inline:
1-The version of Unified applications (Communication Manager, Presence,
Unity
Connection, IPCCX) is 7.0 or it is the latest service release.
: 7.0(1)
4-CUPC and CIPC Versions.
: 7.0(1)"
CUCME 7.0
- Original Message -
Fr
http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/command/reference/CUECmdRef.html
http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/administrator/AA_and_VM/guide/vmadmin_book.html
Moreover GUI covers around half of the options.
- Original Message -
From: Matthew Berry
Thank you for the link however my case is different a bit.
Finally I could step over by looking sdi traces in depth: H.323 gw put '+1' at
the begining of the ANI to be in E.164 format but the RD was defined without
that. The behaviour was strange to me since Mobile Connect works as expected so
Hi Segio
Yes it is confusing isn't it. The columns represent the number of packets per
dscp value. As an example there are 133518 packets in the best effort class
(dscp 0) and 34243 packets marked with dscp 24. There are also 40177 voice
packets, dscp value 46.
Regards
Peter
From: S
platform
port-asic stats drop show the queues with thresholds or the dropped packet per
threshold.
I hope that helps.
Regards
Peter
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mad Kiwi
Sent: 6. mai 2010 23:49
To: Sergio Polizer
Cc
I was missed that there is no need to apply the 'session protocol cisco'
command at all since it is the default. Can be checked by show dial-peer.
- Original Message -
From: Paul Kruger
To: Peter Farkas
Cc: ccie list
Sent: Monday, May 03, 2010 2:05 PM
Subject
Hi,
I'm a bit confused why the session protocol command is not necessary for an
incoming VoIP dial-peer.
Even if it is H.323 dial-peer (session protocol cisco) a SIP call can match.
Any ANI/DNIS matching rule is enough.
Sb can makes me clear?
thx
P__
do you mean device pool or MRGL? No CFB means CUCM couldnt select a
CFB from the MRGL it was using for selection.
-Peter
On Wed, Mar 17, 2010 at 10:46 AM, Omotayo wrote:
> Hello,
> i meant to say i put all the software conference brige in a device pool that
> is not assigned to
so can we see the dial-peers without any sort of CUE configuration?
-Peter
On Mon, Feb 15, 2010 at 2:38 PM, sean hurricane wrote:
> this is not CUE, this is for UC.
>
> On Mon, Feb 15, 2010 at 12:49 PM, Peter Slow wrote:
>>
>> where is your CUE's interface config
command the amount of voice traffic
dropped to 25% of the bandwidth. I hope this helps.
Regards
Peter
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of scott carruthers
Sent: 16. februar 2010 04:31
To: ccie_voice@onlinestudylist.com
*busy signal*
Haha, busty signal. Oops.
-Peter
On Mon, Feb 15, 2010 at 1:50 PM, Peter Slow wrote:
> they would usually carry an in-band alerting tone coming from the
> "telco." sometimes ringing tone can't e signaled with q.931 or other
> protocols and so voice cut t
messages comgin from the far end such as a busty signal,
tri-tone or various "we're sorry, your call could not be..." etc. It's
a common occurrence.
-Peter
On Sun, Feb 14, 2010 at 11:32 PM, Allen Su wrote:
> Hi folks,
>
>
>
> When testing Vol. 1 Lab 10 question 10
yes.
On Mon, Feb 15, 2010 at 10:55 AM, A. Venkatesan wrote:
> hi all,
>
> Can any one tell me, is it necessary to know the complete CCX script
> programming for CCIE vocie lab exam.
>
> Thank you.
>
> Regards,
>
> Venkat A
>
>
> ___
> For more informati
where is your CUE's interface configuration amd the associated dialpeers?
On Mon, Feb 15, 2010 at 12:26 PM, sean hurricane wrote:
> sip mwi does not work
>
> see config below:
>
>
> BR1#sh run | s voice register
> voice register global
> mode cme
> source-address 10.10.201.1 port 5060
> max-dn
lable in the kindle store, but i already
OWN all the books i want. I would LOVE to be able to take my library
on site with me!
-Peter
On Thu, Jan 28, 2010 at 11:00 PM, Tanner Ezell wrote:
> There's always the iPad... :)
>
> On Thu, Jan 28, 2010 at 10:27 PM, Jason Granat wrote:
&g
; On Thu, Jan 14, 2010 at 12:16 PM, Peter Slow wrote:
>>
>> honestly, its going to vary depending on your telco.
>>
>> i know you were asking abotu caller ID from the PSTN, but this sorta
>> goes both ways. Generally speaking, sending 011 AND setting your
>
honestly, its going to vary depending on your telco.
i know you were asking abotu caller ID from the PSTN, but this sorta
goes both ways. Generally speaking, sending 011 AND setting your
called party type to international doesn't make sense. 011 is the
international escape for USA - it tells the
an 20 will be terminated on
two different switches with either GLBP og HSRP between them to give you
default gateway redundancy.
Regards
Peter
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Cotter
Sent: 9. januar 2010 19:
nitor
is not working.
Unity Connection is on a VM workstation but I have had this working before with
a slightly different setup.
Regards
Peter Weeks
Senior System Engineer
CCIE #14880 SP
Grenseveien 95
Pb. 6528 Etterstad
0606 Oslo
peter.we...@datametr
All of your dialpeers use the same codec. you have to hard set the
incoming and outgoing dial-peers, to use g729 on one side, and 711 on
the other side. this is what triggers transcoder invocation in CUBE.
-Pete
On Sun, Jan 3, 2010 at 10:39 PM, Paul Dardinski wrote:
> All,
>
>
>
> Maybe I have b
no... the IPAgentInitial.jsp page will prompt you for that stuff even
if you send the ID, Pwd and Ext parameters.
On Sun, Jan 3, 2010 at 7:04 PM, Anil Batra wrote:
>
> I m not using "single button login"
> so do I still need to config ext and pass to
> Make it work.
>
> Sent from my iPhone!
> On
You dont see it because your call isnt redirected =) when you hit the
messages button, you're making a direct call. in all likelihood, your
call is getting to the VM pilot with its full caller ID. unity is
going to need to see the call as comign from the approriate 4-digit
extension for it to recog
Hi
Debug tftp events shows the correct files being downloaded infact that was the
debug included with my mail. The SIP upgrade has worked fine unfortunately the
phones won't register with the cme.
Peter
-Original Message-
From: bkvalent...@gmail.com [mailto:bkvalent...@gmai
as a check. I have found
that the auto config CUCM should be the same as the option 150.
I hope this helps
Peter
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Alex Hannah
Sent: 3. januar 2010 10:30
To: ccie_voice@onlinestudylist.com
Su
size 1237
for process 231
Jan 3 09:27:34.232: TFTP: Finished flash:/SIP001121115FAF.cnf, time 00:00:00
for process 231
____
Peter Weeks
Senior System Engineer
CCIE #14880 SP
Grenseveien 95
Pb. 6528 Etterstad
0606 Oslo
peter.we...@datametrix.no<mailto:stef...@
suggestions?
Phone 1002 also resets every now and again for no reason that I can see.
Peter Weeks
Senior System Engineer
CCIE #14880 SP
Grenseveien 95
Pb. 6528 Etterstad
0606 Oslo
peter.we...@datametrix.no<mailto:stef...@datametrix.no>
www.datametrix.n
o get moh file:
file get activelog mohprep/SampleAudioSource.alaw.wav
-Peter
On Sun, Dec 27, 2009 at 11:02 PM, Vccie Vccie wrote:
>
> From th 7.0 CUCM Administrators guide.
>
> You can manage the audio files that the Music On Hold feature uses as audio
> sources. The Media Re
On Wed, Oct 28, 2009 at 12:47 AM, Michael Ciarfello
wrote:
> If you are waiting for more labs to come out, I twisted lab5 around to add
Do lab 3 but only use a total of 3 route patterns to do it.
you can Ignore post dial delay requirements.
all your transformations and callback numbers still
no... you need to use DRS to be able to back stuff up and import it back
into CUCM.
If you jsut want to print out your dialplan, you can use the SQL scrit that
i wrote. It shoudl be in the archives if you search for it.
-Peter
On Sat, Oct 17, 2009 at 8:17 PM, Erwan Erwan wrote:
> hi
i think, if i understand what hes askign correctly, you guys might
want to read about
Hunt Forward Settings > Use Personal Preferences checkbox
-Peter
On Sat, Oct 17, 2009 at 12:21 PM, Daniel Rodriguez
wrote:
> No problem. Also, if you weren’t using Unity or Unity Connection and addin
the actual things that make CME work are in IOS.
if you want the webpages / GUI to work, you need to copy the
associated files there. the CUE files are on flash on the CUE, which
is a separate module with it's own filesystem.
Some of the things used for CME that are created during runtime go
into
, and 7975
On Sat, Oct 17, 2009 at 4:58 PM, Peter Slow wrote:
> 7960s/40s dont support it. You are correct.
>
> On Sat, Oct 17, 2009 at 11:12 AM, Dave Wong wrote:
>> Hi
>> I've configured vol 2 lab 3 globalization and localization according to the
>> proctor gu
7960s/40s dont support it. You are correct.
On Sat, Oct 17, 2009 at 11:12 AM, Dave Wong wrote:
> Hi
> I've configured vol 2 lab 3 globalization and localization according to the
> proctor guide, ie, globalization using the translation pattern and
> localization using calling party transformation
er.
Let us know how this goes.
-Peter
On Sat, Oct 17, 2009 at 1:52 AM, Mike Thompson wrote:
> Actually that looks like before that point. Make sure that you have dsp
> services dspfarm configured under the voice card.
> Sent from my phone, apologies for any typos.
> On Oct 16, 2009, at
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