Stephen,
If you want to see best practices for the global dialplan, I would
pull up the CUCM SRND. Under the Call Routing section for the 7.x
and 7.1 srnd there is a new features section where they have a graph
of the global dialplan where it breaks down partitions, Css, Xlations,
and
will put out a more detailed debrief/how to study later but I am still
in shock and shaking :)
Alex Hannah
CCIE Voice 25853
___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
Jeff,
What's even better Check out their extremely professional
website. There is no phone number, just a gmail account and you
have to pay for their product using paypal, western union, or get this
a wire transfer!!!
Now I did a whois and the site is hosted/registered in brussels,
Guys,
I have a very stupid question regarding CUCM 7.1(X) and the new Apply
Config button. In regards to phones... if you configure the device then
the line on a phone, what the heck is the difference in resetting vs
apply config? It looks to me after reading the description of the apply
config
Guys,
I have a CUCM 7.x cluster ( 7.1(2) ) with a Pub and a Sub with the Sub being
the primary server for phones to register to via the CM Group, and auto
registration is turned on for the Sub. DHCP is running on the Pub, the
phones get an IP Address, and download the firmware fine, when they
Hello everyone,
If you wanted to limit the number of conferences in SRST mode to 2, AND
limit those conferences to a maximum of 3 participants, would the following
be the proper approach?
!
call-manager-fallback
max-conference 2 ( where X would represent the total number of
conferences
Hello everyone.
I am attempting to create the following QoS policy on a 3750 port with an
IP Phone plugged in behind it.
The policy will police signalling ( SCCP ) 32k down to 8k and remark to DSCP
8. I have read through most of the SRND guide for the 3750, the model I am
following is the:
-
From: Michael Ciarfello mciarfe...@iplogic.com
Date: Wednesday, November 11, 2009 8:56 pm
Subject: Re: [OSL | CCIE_Voice] 3750 QoS Question
To: Alex Hannah alex.han...@gmail.com, ccie_voice@onlinestudylist.com
ccie_voice@onlinestudylist.com
Here are some hints for you to research:
I
Hey Guys,
I have AAR configured on my CUCM Server here in my lab. Below is a brief
rundown of the config.
1. I have enabled the AAR Service Parameter
2. Restarted the Call Manager Service and have rebooted ( just to
be dilligent )
3. Created 2 AAR Groups, 1 for each site
4. Created AAR PT
Of *Alex Hannah
*Sent:* Saturday, 3 October 2009 2:55 PM
*To:* ccie_voice@onlinestudylist.com
*Subject:* [OSL | CCIE_Voice] AAR Won't kick in on CUCM 7
Hey Guys,
I have AAR configured on my CUCM Server here in my lab. Below is a brief
rundown of the config.
1. I have enabled the AAR
Guys,
I have CME 7x loaded on a 2811 here in my lab. I am attaching the config to
this email. I have placed the SIP firmware for a 7960 on the flash, added
the necessary commands into the TFTP-Server command and I have followed Lab
3A fairly closely. I am still recieving Registration Rejected
Guys,
I have configured Device Mobility and Local Route Lists and have them both
working very nicely with one another. My question is pretty simply but I
cannot figure out where to do it. Is there a place in CUCM ( probably in
Real Time Monitoring Tool ) where you can see what phones are
Guys,
I am trying to wrap my head around the local route list concept. I know it
in effect decouples the route list from the route group. I have
successfully configured it in my lab here at home with two sites ( simulated
NYC and San Jose )... And I can run debug isdn q931 and DNA and see the
, Sep 21, 2009 at 8:00 PM, Alex Hannah alex.han...@gmail.com wrote:
Jonathan,
Thanks for your answer... I had one follow up question.
With regards to Extension Mobility - If a user in San Jose logs into an IP
Phone in NYC, then Device Mobility kicks in and as you stated... sees that
the phone
For the UCCX node manager terminated unexpectedly folks... Find the
MIVR logs at c:\program files\wfavvid\logs\mivr and search those it
may help you. From past experiences the patches often mess with the
JRE and that often causes the issie
Alex
Sent from my iPhone
On Aug 10, 2009, at
Vik or Mark,
Just out of curiousity when will volume 2 be out? If we purchased
version 1 workbook and proctor guide are we entitled to volume 2 as a
download? Also are the audio and videos being updated as well? Any
eta on those?
Thanks a bunch...
Alex
Sent from my iPhone
Hello,
I am having a nerve racking problem with my IP to IP Gateway in my home lab.
I can dial from the Call Manager cluster (Extensions 3-4XXX) to the CME
phone via the IP2IP Gateway. The CME registered phone ( extension 4xxx )
will ring and I can answer the phone call. I receive no audio
Experts,
In the IP Expert workbook (old version ) on Section 4: Gateways task 4.7,
the workbook has you register BR2's ephone-dn's and FXO ports to a SIP
Registrar. In my lab environment at home I would like to be able to
simulate this on the network side ( in the cloud/PSTN ). What software
Correct me if I'm wrong but the CME version on Blueprint v2 is 3.3 correct?
Thanks,
Alex
DSCP46EF,
I just took the lab in RTP last Thursday. You DO have access to
www.cisco.com/univercd via internet explorer. You also have the QoS SRND,
CM 4.1 SRND, and CME SRND on the desktop. I spent some time after I
realized I wasn't going to pass :) studying what I had access to via
UniverCD.
I am attaching the Sh Run and Debugs from my CME router that I have
attempted to configure Transcoding resources on. I am following page 356 of
the CME Administration Guide, however when I do a sh sdsp units command I
don't see any registration of the Transcoder. I have already done a no sccp
I am taking my first attempt at the IE Lab in RTP on Feb 7th. I was
wondering if anyone had a rough idea ( or list ) of the known documentation
that they give you on the Documentation CD? More specifically I am curious
about the following:
1. Do they provide any Voice IOS Command
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