...@gmail.com]
Sent: Wednesday, July 21, 2010 8:28 AM
To: Berry, Matthew J.
Cc: Mark Holloway; osl osl
Subject: Re: [OSL | CCIE_Voice] CBarge in SRST ~(Again)
Sorry to jump in on the topic. Matt, just curious were you successful with
this configuration? It does not work for me with auto-provision none
Is someone going to ban this guys' email for sharing NDA material on this list?
I don't care what people choose to share offline, that's their business. But
don't blast this stuff out for the whole world to see.
Thanks for posting it (sarcasm).
I bet these questions will be removed from the
Scott -
You need to change the pri-group timeslots 1-3,23-24 to pri-group timeslots
1-3 (H.323 gateway) or pri-group timeslots 1-3 service mgcp (MGCP gateway).
If you look at your running config, IOS will add the ,24 for the D channel of
the circuit. However, if you try to copy your config
I set it for everything, but that's just me.
Matthew Berry, CCVP, Sr. Unified Communications Engineer
mjbe...@kroll.com
-Original Message-
From: Mark Holloway [mailto:m...@markholloway.com]
Sent: Thursday, July 08, 2010 11:42 PM
To: Berry, Matthew J.
Cc: OSL osl
Subject: Re: [OSL
I make a habit of always setting the plan to ISDN.
Matthew Berry, CCVP, Sr. Unified Communications Engineer
mjbe...@kroll.com
-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Holloway
Sent: Wednesday, July
Mark -
Make sure that g729r8 is added under the dspfarm profile. Also, make sure you
CUE dial-peer is hardcoded to be G711ulaw. Otherwise, it will try to use the
default which is g729.
Matthew Berry, CCVP, Sr. Unified Communications Engineer
mjbe...@kroll.com
-Original Message-
Bo,
When you change the frame-relay CIR settings, your fragment size will also
change if you want to stay in sync with the 95% PVC best practice.
Page 3-27 of the QoS SRND:
Fragment Size in Bytes = (PVC Speed in kbps * Maximum Allowed Jitter in ms) / 8
If your PVC is 95% then the calculation
Graham –
But if you modify the CIR, it would seem to affect the calculation used for the
fragment size?
Fragment Size in Bytes = (PVC Speed in kbps * Maximum Allowed Jitter in ms) / 8
Matthew Berry, CCVP, Sr. Unified Communications Engineer
mjbe...@kroll.commailto:david.ra...@kroll.com
I guess that makes sense. You're not actually making the link slower, so the
fragment size wouldn't change.
We'd only need to change the minCIR, CIR, and bc?
Matthew Berry, CCVP, Sr. Unified Communications Engineer
mjbe...@kroll.commailto:david.ra...@kroll.com
From:
CUCME does not support intercluster multicast MOH.
Matthew Berry, CCVP, Sr. Unified Communications Engineer
mjbe...@kroll.commailto:david.ra...@kroll.com
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Afzal Bhutta
Sent: Monday, June 28,
) [mailto:jason.aar...@us.didata.com]
Sent: Monday, June 28, 2010 1:08 PM
To: Berry, Matthew J.; Afzal Bhutta; ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] Music on Hold
How did you infer he's using multicast and/or CME? I must be missing something.
Check IP Voice Media Streaming App
Make you you have bound the SIP media and control to an interface under voice
service voip / sip
- Sent from my Blackberry
From: ccie_voice-boun...@onlinestudylist.com
ccie_voice-boun...@onlinestudylist.com
To: naoufal.kerbo...@cbi.ma naoufal.kerbo...@cbi.ma;
Daniel,
You best bet would be to do the manipulation at the route list level for such a
request.
- Sent from my Blackberry
From: ccie_voice-boun...@onlinestudylist.com
ccie_voice-boun...@onlinestudylist.com
To: Angel Perez gorr...@hotmail.com
Cc: osl osl
Anyone in SJC this week for the OWLE? Shoot me an email if you want to grab a
drink tonight.
- Sent from my Blackberry
___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
Congratulations, Ash!
Matthew Berry, CCVP, Sr. Unified Communications Engineer
mjbe...@kroll.commailto:david.ra...@kroll.com
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ashar Siddiqui
Sent: Friday, June 18, 2010 1:46 PM
To:
That worked! Thanks!
Matthew Berry, CCVP, Sr. Unified Communications Engineer
mjbe...@kroll.commailto:david.ra...@kroll.com
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Peter Farkas
Sent: Wednesday, May 26, 2010 1:49 AM
To: Matthew
All -
I had an issue last night on Vol 2 Lab 2. I am sending calls from HQ (Region =
HQ) to BR2 over my H.225 trunk (Region = GK). Region setting between HQ and GK
specifies G.729. I have a transcoder registered on the BR2 router.
When I call across the gatekeeper, my endpoints show G.729,
Matthew Berry, CCVP, Sr. Unified Communications Engineer
mjbe...@kroll.com
-Original Message-
From: ccieid1ot [mailto:ccieid...@gmail.com]
Sent: Friday, May 21, 2010 10:23 AM
To: Berry, Matthew J.
Cc: CCIE Voice OSL (ccie_voice@onlinestudylist.com)
Subject: Re: [OSL | CCIE_Voice] Show
the show gatekeeper calls output listed
in the question.
Matthew Berry, CCVP, Sr. Unified Communications Engineer
mjbe...@kroll.commailto:david.ra...@kroll.com
From: Roger Källberg [mailto:roger.kallb...@cygate.se]
Sent: Friday, May 21, 2010 10:44 AM
To: Berry, Matthew J.; CCIE Voice OSL (ccie_voice
service parameters to g729, or increase zone bandwidth setting on the
gatekeeper
Matthew Berry, CCVP, Sr. Unified Communications Engineer
mjbe...@kroll.commailto:david.ra...@kroll.com
From: Roger Källberg [mailto:roger.kallb...@cygate.se]
Sent: Friday, May 21, 2010 10:44 AM
To: Berry, Matthew J
Without LDAP integration, you will not be able to do directory search. To
enable chat with LDAP, you must first send a message from the IP phone presence
client to the CUPC. You will then be able to add the contact and chat.
Matthew Berry, CCVP, Sr. Unified Communications Engineer
Do you have the appropriate called party transformation CSS configured on your
gateway? Make sure you have unchecked the option to use the device pool CSS
for transformations. It sounds like your transformation is not being invoked
on the gateway.
Matthew Berry, CCVP, Sr. Unified
Bo -
You're right. I changed the max sessions value from 45 to 2. This is the
new output from my show sdspfarm units:
mtp-1 Device:RTR-XCODE TCP socket:[1] REGISTERED in SCCP ver 17/10
actual_stream:4 max_stream 4 IP:192.168.99.1 57105 MTP Dixieland keepalive 0
Supported codec:
Vik is absolutely right.
In Chapter 9 of the CUCM SRND (9-12):
Location Hub_None is a special location that is configured by default with
unlimited audio and video bandwidth, and location Hub_None cannot be deleted.
If the devices at a branch location are configured in the Hub_None location,
I encountered this linecode issue last week. Changing it to B8ZS fixed my
problem.
Matthew Berry, CCVP, Sr. Unified Communications Engineer
mjbe...@kroll.commailto:david.ra...@kroll.com
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of
Here's my question:
What kind of real-world (or lab) scenario would require disabling SIP
supplementary services on an IOS gateway?
Quote from CUCM SRND:
SIP Refer or SIP 302 Moved Temporarily messages can be used for supplementary
services such as call
transfer or call forward on Unified CME
CUPC STATUS MENU IS GRAYED-OUT
This usually means that CUPC failed to connect to the Presence Engine. This
could be caused by:
* Digest credential or Incoming ACL was not configured
* Proxy domain was not configured properly
* Network issue
Matthew Berry,
Jeff -
Use the username and password that is used to login to AppAdmin. Pay
attention! The login is case-sensitive. If AppAdam/CUCM sees your login ID as
JCotter, you better enter it exactly (jcotter will not work).
There should be a button on the login screen to login anonymously. Really
If using PL, did you try admin and c1sc0123?
From: vccie2010 [mailto:vccie2...@gmail.com]
Sent: Thursday, April 29, 2010 1:20 PM
To: Berry, Matthew J.
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Unable to open .aef file on CRS editor
no luck tired with AppAdmin
Amy -
So you recommend that we rely purely on CUCM to do the firmware conversion for
the CUCME phones? I know there are varying opinions about whether this is a
good option or not. Most of it, in my opinion, stems from the (legitimate?)
fear that the lab will specifically ask us to change
Great post, Otto. I miss chatting with you! I'll need to keep my eyes open
for you on Skype.
Matthew Berry, CCVP, Sr. Unified Communications Engineer
mjbe...@kroll.commailto:david.ra...@kroll.com
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On
Jeff -
Did you ever find out the answer to your question? I'm curious.
Matthew Berry, CCVP, Sr. Unified Communications Engineer
mjbe...@kroll.commailto:david.ra...@kroll.com
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Cotter
Jeremy -
It's important to remember that translation pattern modify the number whereas
route patterns and route lists do not. Once the modification takes place, as a
call passes from an internal IP phone toward the gateway, the transformations
at each point can be overridden. The
Thanks, Amy. Good tips!
Matthew Berry, CCVP, Sr. Unified Communications Engineer
mjbe...@kroll.commailto:david.ra...@kroll.com
From: Amy Ryan [mailto:ar...@ipexpert.com]
Sent: Wednesday, April 21, 2010 11:19 AM
To: Berry, Matthew J.; amr gaber; ccie_voice@onlinestudylist.com
Subject: Re: [OSL
I read the same thing the other day. You can actually configure multiple CUCME
sites with a single CUE. You will need to setup the CUCME hosting the CUE as
an MWI relay server.
To Configure the SIP MWI Server (Multiple CUCME Routers)
o Go into the SIP user-agent config and
Sergio -
Did you ever figure this out?
Matthew Berry, CCVP, Sr. Unified Communications Engineer
mjbe...@kroll.commailto:david.ra...@kroll.com
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Sergio Polizer
Sent: Wednesday, April 14, 2010
Does anyone know if there is a way to define two skill groups for an agent, but
give a particular skill group a priority over the other one? Meaning, if two
calls are in queue, each for different skill groups, I can adjust which call
will get priority over the other based on the type of skill?
. Unified Communications Engineer
mjbe...@kroll.com
-Original Message-
From: bkvalent...@gmail.com [mailto:bkvalent...@gmail.com]
Sent: Monday, April 19, 2010 9:21 AM
To: Berry, Matthew J.; ccie_voice-boun...@onlinestudylist.com; OSL
Subject: Re: [OSL | CCIE_Voice] UCCX Skill Groups
Yes
Wilson,
You Meet-Me numbers will need to belong to a partition that the inbound gateway
CSS can see. Make sure that the significant digits set on the gateway will
deliver the correct number of digits to CUCM to match up with the Meet-Me
ranges.
To my knowledge, there is no way to set a PIN
From the CUE Design Guide:
Call-Agent Time-of-Day Routing: Time-of-Day (ToD) routing of calls to a
receptionist (in contrast to the AA) requires a ToD routing feature on your
call agent. With Cisco CME, this can be done by using a Tool Command Language
(TCL) 2.0 script named Time of Day Routing
You will need to add a voice translation pattern to append the + to all
incoming/outgoing calls. Then, apply the rule to the voice-port for your T1/E1.
Matthew Berry, CCVP, Sr. Unified Communications Engineer
mjbe...@kroll.commailto:david.ra...@kroll.com
From:
It took me two days to realize that EPNM = External Phone Number Mask
#fail
:)
Matthew Berry, CCVP, Sr. Unified Communications Engineer
mjbe...@kroll.commailto:david.ra...@kroll.com
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roger
I don't know much about QSIG, so pardon the simple minded question: Could QSIG
be on the lab and, if so, in what ways could we see it manifest?
Thanks!
- Sent from my Blackberry
___
For more information regarding industry leading CCIE Lab training,
I actually just emailed Ben. We'll see.
- Sent from my Blackberry
From: ccie_voice-boun...@onlinestudylist.com
ccie_voice-boun...@onlinestudylist.com
To: roger.kallb...@cygate.se roger.kallb...@cygate.se;
ciscovoiceg...@gmail.com ciscovoiceg...@gmail.com; CCIE
All -
Here's a sample section from a SIP SRST setup from the SIP SRND Admin Guide:
voice register pool 1
id network 10.10.201.0 mask 255.255.255.0
application sip.app
preference 2
incoming called-number
cor incoming css-internal default
codec g711ulaw
What the heck is this
, March 25, 2010 6:16 AM
To: Berry, Matthew J.; osl osl
Subject: RE: [OSL | CCIE_Voice] SIP SRST - What application to use?
Hello:
The second example is not shown...
My experience tell me that if you use application sip.app the gw won't find the
app, then you will need application global service
Ashar -
You need to change the display name on your ephone. The +617 Is in the
place where you'd normally put the caller ID.
Your are displaying the digits correctly. You just need to change the CID.
From: ccie_voice-boun...@onlinestudylist.com
Do you have the authenticate register command entered?
- Sent from my Blackberry
From: ccie_voice-boun...@onlinestudylist.com
ccie_voice-boun...@onlinestudylist.com
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Sat Mar 20 09:30:11 2010
Whoops. Should have read Otto's email. Sorry for the duplicate.
- Sent from my Blackberry
From: ccie_voice-boun...@onlinestudylist.com
ccie_voice-boun...@onlinestudylist.com
To: Kalyan iyer kparam2...@gmail.com
Cc: ccie_voice@onlinestudylist.com
Angel,
For starters, the only scenario where you're need to worry about a plus being
sent to an IOS gateway from CUCM would be an MGCP/SIP gateway. H.323 cannot
receive a plus from CUCM; it will simply strip it off before it hits any
translation rules.
That said, say you are asked to strip
The QoS SRND states that the auto qos voip command adds the following config
to the router:
C2970(config)# mls qos
C2970(config)# mls qos map cos-dscp 0 8 16 26 32 46 48 56
Earlier in the SRND, around page 40, it says that the old marking for audio
signaling was AF31 (26). That is the same
Pulling from the QoS SRND, the following configuration is only supposed to
allow the bandwidth for one voice call per switchport VLAN. Obviously, based
on the 128k, we're focused on G.711 calls (so my next question will not apply
to G.729).
I want to know if the following command would
Anupam,
Can you please provide more detail? Take opportunities like this to thoroughly
explain what you do know, including config examples and outputs from show
statements. Your issues could be caused by any number of nuances.
Approach these scenarios as if you were already a CCIE. Provide
Jean,
Are you using Proctor Labs or your own lab? The PSTN router should take care of
the calling number type.
You should also make sure you don't have any translation patterns on the BR2
gateway that would modify the type. Also check your H323 gateway to ensure the
same thing.
- Sent from my
a national type.
- Sent from my Blackberry
From: Jean M. Thewissen m...@mnet.com.mx
To: Berry, Matthew J.; 'ccie_voice@onlinestudylist.com'
ccie_voice@onlinestudylist.com
Sent: Wed Mar 17 20:55:13 2010
Subject: RE: [OSL | CCIE_Voice] about globalization
How many seconds go by? Could you be hitting the T38 timeout value? Perhaps
something with call capabilities are not being setup?
- Sent from my Blackberry
- Original Message -
From: ccie_voice-boun...@onlinestudylist.com
ccie_voice-boun...@onlinestudylist.com
To: Jason Granat
Sorry. Thinking of wrong term.
Sounds like capabilities are not being established. Are you using h323 fast
start?
- Sent from my Blackberry
- Original Message -
From: Jason Granat j...@slash128.com
To: Berry, Matthew J.
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Is anyone aware of good design documentation (akin to SRNDs) for Unity
Connection?
All I've been able to find is the following:
* CUC Admin
* CUC CUCM SCCP Integration Guide
* CUC CUCME SCCP Integration Guide
* CUC Design Guide (very limited, not very
Working on Vol 1 Lab 10A, Question 10.4
The Proctor Guide calculates L2 MLPoFR as 9 bytes per packet. However, the QoS
SRND defines the following on page 1-15:
- PPP = 12 bytes
- MLP = 13 bytes
- FR = 4 bytes
- FR with FRF.12 = 8 bytes
None of those match up. Why did IPexpert chose 9 bytes
Might be simple, but do you have your H323 dial peers setup correctly?
- Sent from my Blackberry
From: ccie_voice-boun...@onlinestudylist.com
ccie_voice-boun...@onlinestudylist.com
To: Mike Todd michaelt...@gmail.com
Cc: ccie_voice ccie_voice@onlinestudylist.com
All -
I am setting up + dialing on a self-made lab. A question has come up as to
where the Called Party Number Type should be set. For this exercise, I want to
find the best way to route calls through a system, utilizing alternate paths
for failover scenarios. Those this does not take TEHO
Check the CSS on the remote destination profile you're calling from.
If you do a debug isdn q931 on the PSTN gateway, do you see the call hit the
gateway?
Your rerouting CSS on the RDP is used for calls out to your RD.
Your CSS on the RDP is used for calls through MVA that are routed out through
Does anyone know of where Cisco's UCCX/IVR sample script repository is? I
can't find it.
___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
Omotayo,
How do you have your regions setup in CUCM? The CUCME trunk through the HQ
gateway should be placed in the HQ region.
Can you also send me the HQ config as an attached file. Make sure your dspfarm
has a 'no shutdown issued. Also, make sure your transcoder is registered to
CUCM
Have you issued a no shut' on dspfarm profile 1 transcode?
From: Omotayo [mailto:adefilabi...@gmail.com]
Sent: Wednesday, March 10, 2010 5:38 PM
To: Berry, Matthew J.
Cc: OSL Group
Subject: Re: [OSL | CCIE_Voice] calls from hq to and from cme sip phones
The trunk DP has a region that speaks
If all else fails, save your configs and reboot BR2 and HQ routers. And test
again.
From: Omotayo [mailto:adefilabi...@gmail.com]
Sent: Wednesday, March 10, 2010 5:38 PM
To: Berry, Matthew J.
Cc: OSL Group
Subject: Re: [OSL | CCIE_Voice] calls from hq to and from cme sip phones
The trunk DP
I am going to attend the IPexpert five day bootcamp in San Jose this coming
May. Would anyone be interested in splitting a hotel room to save some money?
Please let me know soon as I will probably start looking for deals this week.
I will also do the mock labs in June. If you'll be at that
All -
I have several theory questions that came up during my time spent in labs 5a,
5b, and 5c. I am hoping that some super smart CCIE candidates may have found
the answer to these questions.
Roger and Ossamah, I'm looking to you guys to represent. :)
1. How do you source RAS messages
Would anyone be willing to bring a microphone and record Ben's 8 hour
techtorial?
- Sent from my Blackberry
From: ccie_voice-boun...@onlinestudylist.com
ccie_voice-boun...@onlinestudylist.com
To: OSL Group ccie_voice@onlinestudylist.com
Sent: Tue Mar 02
Good point, Roger. CUCM Pub needs something to validate against.
- Sent from my Blackberry
From: ccie_voice-boun...@onlinestudylist.com
ccie_voice-boun...@onlinestudylist.com
To: ShinGei Yong shingei.y...@gmail.com; ccie_voice@onlinestudylist.com
Easy workaround is to use a different interface or, even better, setup another
loopback.
- Sent from my Blackberry
From: ccie_voice-boun...@onlinestudylist.com
ccie_voice-boun...@onlinestudylist.com
To: Wael Agina waelag...@gmail.com; OSL Group
I had a similar issue yesterdat, Randall. When I tried to revert, it wouldn't
go back to the original version. Although, I don't fully understand how PLs
revert command works.
- Sent from my Blackberry
- Original Message -
From: ccie_voice-boun...@onlinestudylist.com
I was going though Mark Snow's VoD for v3. In the call routing video, Mark
touches on IP-to-IP gateway functionality, but I felt there was quite a bit
left out. It didn't seem to complete to me. One of the questions that came
out of watching that video is what is the big difference in
That's good to know.
Pursuing a CCIE is almost as much a crash-course in the disjointed way that
Cisco utilizes licensing and hides documents on their website. :)
From: Roger Källberg [mailto:roger.kallb...@cygate.se]
Sent: Tuesday, February 23, 2010 7:27 AM
To: Berry, Matthew J.; ccie_voice
Is there a plan to add that? Ben said it could be a testable topic.
- Sent from my Blackberry
From: ccie_voice-boun...@onlinestudylist.com
ccie_voice-boun...@onlinestudylist.com
To: scott carruthers scarruthe...@hotmail.com
Cc: ccie_voice@onlinestudylist.com
All -
I did Vol 1 Lab 5a yesterday and ran into a question about Calling Party
Numbering Plan. I have pasted Cisco's explanation below, but I'm looking for
some insight as to why we would ever use this setting and how I would know if
the lab was trying to validate my understanding of this
...@krollontrack.commailto:agutz...@krollontrack.com
From: Jeff Knuckle [mailto:jknuc...@nationwidelab.com]
Sent: Monday, February 15, 2010 4:06 PM
To: Berry, Matthew J.; ccie_voice@onlinestudylist.com
Subject: RE: Calling Party Numbering Plan
To answer the first part of your question, Calling party
Vol 1 - Lab 5 - Task 5.5 / Setting up TEHO for 212 calls from BR1 through HQ
H.323 GW
I can get TEHO to work when dialing a 617 area code number from HQ Phone 2,
routing the call over the WAN, out the BR1 MGCP gateway. It works like a
charm. It appends the + which seems to come from the
All -
This is somewhat off-topic for the CCIE Voice lab... Well, it is actually
completely off-topic.
Is anyone here using a SNMP-based tool to monitor in real-time the DS0 usage of
your PRIs? We have been using PRTG enterprise for our data traffic. However,
I'm looking for something to
Steve,
What codec work is going on between the sites? If you are calling from HQ to
BR2, your region will be using G.729r8. However, you BR2 SIP phone is likely
setup for G.711ulaw. That could be why when you answer the call, it drops.
Codec negotiation could fail.
Remember, SIP dial
Guys -
From what Ben Ng told me, OSPF would be setup and not a part of
troubleshooting. I think it's safe to say we can focus on what the blueprint
says and the blueprint alone. If it doesn't say OSPF, then we can ignore OSPF.
Thanks,
Matthew Berry
Office 952 516 3748 | Mobile 952 221
Wilson,
There is a known bug in CUCM 7.0 (not sure if it's in 7.1) where CSA will
disallow DHCP requests if you initially installed the CUCM software and did not
configure DHCP during the install process.
1.If CSA is enabled, CUCM-facilitated DHCP may fail. You may need to
disable
According to Vik's vol 1 walkthrough, that willk not work. CUCME SIP phones
support one codec only.
- Sent from my Blackberry
From: ccie_voice-boun...@onlinestudylist.com
ccie_voice-boun...@onlinestudylist.com
To: Mike Brooks 2xcci...@gmail.com
Cc: OSL Group
...@krollontrack.com
From: Wayne Lawson [mailto:groupst...@ipexpert.com]
Sent: Tuesday, January 26, 2010 9:53 PM
To: Berry, Matthew J.
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CUCME SIP Issues
Matthew - How are the Vol 1 Video Solutions working out? Keep in touch!
Regards
Randall -
Those four file types make up the firmware that the Cisco IP phone uses. The
7940/7960 phones use the .sb2 and .bin commands. Proctor Labs (ie. IP Expert)
uses 7960s in their racks. However, the actual lab is going to use 7965s. If
you look at the newer phone models, such as the
Scenario:
I have an X-Lite softphone setup with a dn of 20004. I also setup another dn
of 20005 to call forward all to 20004. The dn of 20005 is not assigned to
another phone. In this scenario, there is only one phone registered to the
CUCME SIP instance.
Problem:
I go off-hook, dial
All -
Last week, I received the new IP Expert Volume 1 video walkthroughs that were
recorded by Vik. Today is my labbing day so I decided to pop in the DVD and
listen to it after each lab was finished.
WOW! That's what I have to say.
Vik brought up a ton of stuff that I otherwise would not
Cristobal mentioned that he is using CME 7.1.1.0 in his lab.
According to the lab blueprint, it seems that the version that will be on the
lab is CME 7.0.
Is this correct? Are minor releases supported in the labs? For example, could
we be tested on CUCM 7.1(2) and the new features that are
Will CME 7.1 be in the lab? I thought 7.0 was on the blueprint.
- Sent from my Blackberry
From: ccie_voice-boun...@onlinestudylist.com
ccie_voice-boun...@onlinestudylist.com
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Fri Jan 22
, 2010 12:55 PM
To: Berry, Matthew J.
Cc: vccie2010; OSL Group; Vik Malhi
Subject: Re: [OSL | CCIE_Voice] Issues Configuring SCCP UCME 7.0
Matthew,
Did you solve this issue?
Is the tftp file being generated? do a sh telephony-service tftp and send us
the results, would you also please send us a sh
All -
It'd be great if someone could comment on this issue I'm encountering. This is
the first time I've setup UCME 7.0 (as compared to CME 3.x, 4.x).
So I transferred the SCCP files for a 7961 phone to the root of flash. This is
different than the IP Expert Appendix A documentation and Mark
SCCP phones should register without an ephone-dn. I know from experience last
week that SIP phones will not register without a DN.
I am still trying to work through this issue. I've noticed that when I run
debug tftp packets I get the following output about a file not being found:
Jan 21
Are you changing the script name under the application in UCCX?
Have you added the script under the RM subsystem? I think that's where it is
although I may be thinking of IP-IVR.
- Sent from my Blackberry
From: ccie_voice-boun...@onlinestudylist.com
AM
To: Berry, Matthew J.; OSL Group
Subject: Re: [OSL | CCIE_Voice] SCCP to SIP Firmware Upgrade
I would use UCM to do the firmware upgrade. Follow these steps:
(1) Add the CME phone into the UCM database (add a new phone).
(2) Assign a number to the phone- dummy number will do, just necessary
After listening to Mark Snow's comments on SCCP to SIP firmware upgrades in
CUCME, I'm not surprised that I has issues tonight. I entered all the
necessary commands (I think, check me on this) to allow for registration. TFTP
aliases are in the config. The files are also there, preloaded for
Kavi,
Directed call park allows you to park a call at a specific directed call park
number, instead of randomly being assigned a call park number.
I recreated the scenario at work, but I could not get call retrieval to fail
with a status of blocked. As long as the directed call park number is
]
Sent: Sunday, January 17, 2010 9:22 PM
To: Arun Kumar
Cc: Berry, Matthew J.; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Trunk Port Configuration
Hello,
If you define a specific speed/duplex setting on the port, you should do the
same on the phone, you don't want to have one
I am having an issue with phones that are connected to the Proctor Labs rack
over EZVPN. If I go off-hook by lifting up the handset or pressing the
speakerphone button, I will not get a dial tone. The phone is registered to
CUCM. After this occurs, I will see the phone re-register. It is
In workbook 1, lab 1, we are told to configure a trunk port this way:
Standard Catalyst 3750 Configuration for Trunk Port
Vlan 10
Name DATA
State active
Interface FastEthernet 1/0/2
Switchport trunk encapsulation dot1q
Switchport mode trunk
Switchport trunk native vlan 10
Speed 100
Can someone explain these two concepts to me? I'm not sure I understand the
practical application of these options.
Rerouting Calling Search Space
From the drop-down list box, choose a calling search space to use for
rerouting.
The rerouting calling search space of the referrer gets used
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