Calling Party manipulations performed on the RP or RL/RG details will
NOT override your Calling Party XForms. Transformations will override
any manipulations in RP/RL-Level.
In your case:
Your Calling Party XForm Pattern 212394500X will never be matched,
because your calling party is 5001.
There are 2 steps missing:
Under Application/Deskphone Control/User Assingments give the user the
permission to use Deskphone Control.
Under Application/CUPC/CTI Profile assing user to appropriate profile.
Nara Shikamaru wrote:
In working though this section and doublechecking my work in
Thanks, Phil. Much appreciated.
On Mon, Oct 12, 2009 at 1:03 AM, Phil G pgciscov...@gmx.net wrote:
There are 2 steps missing:
Under Application/Deskphone Control/User Assingments give the user the
permission to use Deskphone Control.
Under Application/CUPC/CTI Profile assing user to
What does the voice class codec 1 look like?
And try replacing the voice class codec with just codec g711u. Just to see
what happens.
From: Girard, Jeffrey COL MIL USA [mailto:jeffrey.gir...@us.army.mil]
Sent: Sunday, October 11, 2009 11:35 PM
To: Michael Ciarfello;
This is a housekeeping, organization suggestion for IP Expert.
There are so many email threads that go back and forth about specific labs
in the guide. Have you ever thought about creating a forum, with a
different folder for each lab? That way, things would be better organized
for future
Voice class codec 1
Codec preference 1 g711ulaw
Codec preference 2 g729r8
The SIP phones are already at g711 from the previos lab
If I remove the voice-class codec 1 command from the voip DP and replace it
with codec g711 - there are no changes. SIP phone cannot call through GK
Vik/Mark - any
Can you confirm the GK is not failing the call due to CAC (if you are using
bandwidth cac for example). Debug ras.
Also ensure you have allow-connections sip to h323 within voice service
voip.
Voice-class codec is configurable but not supported in voice register pool-
therefore don¹t use it-
Hmm.. Assuming they mean transfer to a live operator/receptionist, the operator
input wouldn't necessarily require its own CH for the transfer.
It's possible that caller input 0 is configured for Attempt Transfer to an
existing CH within the 7 mentioned below, where as the other caller input
Matthew,
You raise a good point- we used to have exactly what you describe
(CertificationTalk). There were a few problems associated with this but I¹ll
raise the question.
Vik
--
Vik Malhi CCIE #13890
Senior Technical Instructor - IPexpert, Inc.
Telephone: +1.810.326.1444
Fax:
Something that has some feedback such as resolved, etc. Netpro has a good
concept on that. That way we could hopefully sort on resolved vs unresolved.
Perform stronger searches than what's offered on OSL archives, etc.
CertificationTalk was kind of too stratified. Also need the ability to
Along the same lines, have you (IP Expert) considered correcting the errors
and clarifications that are uncovered on this forum to an online (and
constantly updated) errata?
It would make it much easier for the people using the study guides if they
could go to one place to see this information,
WHAT?! YOU CAN DO THAT?!?!?!?!?!
My brain hurts.
On Mon, Oct 12, 2009 at 3:43 PM, Vik Malhi vma...@ipexpert.com wrote:
I think by far and away the quickest and best way is to *not *do firmware
uploads on CME- do it on UCM. That means you set your TFTP to be the PUB and
add the device in
Yes we can do that UNLESS question specifically says to perform firmware
conversion locally on CME
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Nara
Shikamaru
Sent: Tuesday, 13 October 2009 9:58 AM
To: Vik Malhi
Cc: OSL Group
I'm going through them so I'm thinking out loud- pardon the suggestion
without having read through all your debugs.
But...the first thing I want you to try is to modify the dtmf type in the
voice register pools to be rtp-nte. Cre prof/reset and try the call again.
Failure in DTMF negotiation
How are they going to know that?
And the CCM version will probably be different than the CCME version so you
will be upgrading again.
We know all the phones are 7965's. Is there the upgrade issues that were
described before? The 65's are new enough that they should be able to have a
decent
Hello,
i have added the transcoder to the ports and route point and its not working
Also, the CTI Port are not registering to the calmanager
What can i do
thanks
On Sun, Oct 11, 2009 at 7:02 AM, Aamir Panjwani
aamir.panjw...@ivision.com.au wrote:
Make sure voicemail port, route point has
Make sure your cti ports and route point is associated with cue-jtapi
app user and this user should have standard CTI enabled rights
From: Omotayo [mailto:adefilabi...@gmail.com]
Sent: Tuesday, 13 October 2009 12:32 PM
To: Aamir Panjwani
Cc: OSL Group
Subject: Re: [OSL | CCIE_Voice]
I tried it like Vik mentioned, of course it works - but, yes, I'm concerned
that they may specify a specific type of firmware download (CME, not CUCM).
A two-step would be a pretty rough thing to have to deal with in this
context. Maybe I should just work on sorting out the tftp-server syntax in
I think you will be fine to do it on UCM in the lab. They won't
specify that you have to do it on CME. It was raised in the techtorial
at networkers.
Vik Malhi - CCIE#13890
Senior Technical Instructor - IPexpert Inc
Telephone: +1.810.326.1444
Fax: +1.810.454.0130
Mailto: vma...@ipexpert.com
Vik,
Are there any recordings of this techtorial? I have a CiscoLive
virtual account. Is it somewhere there in one of the sessions you can
listen to online?
Chris
On Oct 12, 2009, at 10:02 PM, Vik Malhi wrote:
I think you will be fine to do it on UCM in the lab. They won't
specify that
Don't quote me on this- but i think you only get this if you attended.
Vik Malhi - CCIE#13890
Senior Technical Instructor - IPexpert Inc
Telephone: +1.810.326.1444
Fax: +1.810.454.0130
Mailto: vma...@ipexpert.com
Join IPexpert's Free CCIE Peer Groups Study Communities at
Vik,
I think your'e right. I have gone through the CLV site top to bottom
and didn't see anything in there. But it's a heap of content (much of
it very good btw well worth it). You happen to recall what the
techtorial was called?
Chris
On Oct 12, 2009, at 10:12 PM, Vik Malhi wrote:
Also on a slight tangent- I noticed if you gracefully put cue in srst
mode by stopping the ccm service, the cue never comes out of srst
(after starting the ccm service) until you reboot it. So I think your
next step after checking the config is to reload cue.
If you shutdown the wan
Ccie voice technical seminar (I think). TECCCIE-3002
Vik Malhi - CCIE#13890
Senior Technical Instructor - IPexpert Inc
Telephone: +1.810.326.1444
Fax: +1.810.454.0130
Mailto: vma...@ipexpert.com
Join IPexpert's Free CCIE Peer Groups Study Communities at
www.IPexpert.com/communities
On Oct
I checked both and the Subscribe CSS is the same for all phones and the BLF
for call lists is enabled
On Mon, Oct 12, 2009 at 1:56 PM, Jonathan Charles jonv...@gmail.com wrote:
Make sure BLF for call lists is enabled in Enterprise Parameters...
also make sure they have a subscribe css...
On
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