I tried both internal and external numbers. Neither work. I ran a
debug ccsip messages and never see CUE try to transfer the call.
John
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Baktha
Muralidharan
Sent: Saturday, February
(jomcgaug)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CFUR does not work
No. it is supported. The destination Phone will just ring a bit later
through PSTN.
On Sun, Feb 5, 2012 at 3:16 AM, John McGaughey (jomcgaug)
jomcg...@cisco.com wrote:
Hi Vik/All
I'm working
Thanks. That works. DSG does say 4-4. I overlooked it. L
From: datucha123 datucha123 [mailto:datucha...@gmail.com]
Sent: Sunday, February 05, 2012 8:43 AM
To: John McGaughey (jomcgaug)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CUE Live Reply
You have press 44
Hello,
I'm in lab 4 of the new 5 labs. Question 6.4. It's asking to configure
Live Reply. The DSG says to just click the check box Enable Live
Reply. I've done so but when I press option 4 it says this message
cannot get a reply. I'm calling from the SiteC phone 1.
I added a user and
Hi Vik/All
I'm working on Lab #4 of the new 5 labs. Quesiton 9.2. They are asking
you to configure CFUR on SiteB phone 2. However this will not work
because of the RDP assigned to the phone.
RDP and CFUR and not supported together. See CSCtg43998.
John
Hello,
From the new 5 labs, Lab 2 question 10.1 it asks the following.
For traffic being sent to the Site A gateway ensure that traffic marked
with COS 5 is dropped if queue 1 is 75% full.
The solution guide says to add queue-set 2 to the fastethernet port and
change the following 2
3rd threshold is 100% implicit. So that wouldn't work since it will drop when
100% full.
-Original Message-
From: Farkas Péter [mailto:wormh...@sch.bme.hu]
Sent: Friday, January 27, 2012 8:04 AM
To: John McGaughey (jomcgaug)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL
In lab 1 of the new 5 labs, question 7.3 it asks that the agent be able
to see the ani and number of calls in queue. The DSG says to add a
salesinq field to the default layout. The problem is that there is
nothing telling IPPA what to populate this field with.
So in my lab I see the
The question doesn’t say anything about trusting or not trusting, which to me
means we can use either.
-Original Message-
From: Rrcrumm [mailto:rrcr...@yahoo.com]
Sent: Wednesday, January 18, 2012 1:14 AM
To: Ashraf Ayyash
Cc: John McGaughey (jomcgaug); ccie_voice@onlinestudylist.com
Hello,
In Workbook 2, Lab 10, question 5.2 it asks you to setup MLP LFI
between HQ and BR1. In the solution guide it has you use auto qos trust
on the HQ side but does not use trust on the BR1 side. The DSG guide
says the reason for not using the trust key word is because of the
following:
Has anyone purchased the five new labs from IPExpert? If so, are they
worth money? What's different about them versus the 10 lab workbook?
Anything new or just more of the same?
John
___
For more information regarding industry leading CCIE Lab
from CUCM (or any
other sources) to use the G729, so that xcoder will get invoked.
On Wed, Dec 28, 2011 at 4:44 AM, John McGaughey (jomcgaug)
jomcg...@cisco.com wrote:
Nevermind. I figured it out. I created a separate inbound dial-peer with
g729 and that caused the xcoder to get invoked
Hello,
I'm in Workbook 2, Lab 8, question 4.5. I have CUE setup properly and
calls between CME phones roll successfully to voicemail. However, they
fail to roll to voice mail when an HQ phone calls a BR2 phone.
The answer guide says to configure an xcoder on the BR2 router. Here's
my
...@gmail.com [mailto:bkvalent...@gmail.com]
Sent: Tuesday, December 27, 2011 6:42 PM
To: John McGaughey (jomcgaug); ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Calls to CUE fail over WAN
Did you enable cube?
Sent from my Verizon Wireless Phone
- Reply message -
From
I think they split it up between each link. There's 2 between HQ and
BR1. 48K on each link.
Doesn't look like a typo to me.
John
-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Rrcrumm
Sent: Wednesday,
I tried adding it to HQ and same issue. Also, the DSG shows that you
should add it to BR2's MRGL not HQ's.
John
From: datucha123 datucha123 [mailto:datucha...@gmail.com]
Sent: Monday, December 12, 2011 4:54 AM
To: John McGaughey (jomcgaug)
Cc: ccie_voice@onlinestudylist.com
Subject: Re
...@onlinestudylist.com
When replying, please edit your Subject line so it is more specific
than Re: Contents of CCIE_Voice digest...
Today's Topics:
1. CBarge with SNR (John McGaughey (jomcgaug))
2. Re: lab/6 MGCP TS got the lab in Dubai really hard luck (Ray)
3. Re: srst behaviour under telephony
Hi Stuart,
I ran into the exact same problem. The answer in the DSG is incorrect.
I got it to work by doing the following.
I created the following dial-peer on BR1.
dial-peer voice 415 pots
destination-pattern 415...
port 1/0/0:23
forward-digits 7
In the teho route list I
OK now it's working for me using this config. It wasn't earlier. Not
sure what I did wrong. Thanks for the clarification.
John
From: Vik Malhi [mailto:vma...@ipexpert.com]
Sent: Saturday, December 03, 2011 3:35 PM
To: John McGaughey (jomcgaug)
Cc: Geoghegan, Stuart; ccie_voice
]
Sent: Thursday, November 10, 2011 10:56 PM
To: John McGaughey (jomcgaug); ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Workbook 2 typos
Did you configure the calling party transform pattern? \.+44.! predot/prefix
0? At least in other labs with similar question this was part
.
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of John McGaughey
(jomcgaug)
Sent: Friday, November 11, 2011 6:04 AM
To: edgar feliz; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Workbook 2 typos
I configured the calling
Is it just me or is Workbook 2 full of mistakes? For example, I'm in
lab 5 question 2.3. It says calls from phone button 2 should appear in
the history logs as +44 020 5943 2785.
The answer states to set the Subscriber Number Prefix to +44. That's
incorrect since the call comes in like
In workbook volume 2, lab 2, section 6.1 it says You must trust
markings from all endpoints and servers.
How do you trust the markings on FastEthernet ports on the HWIC-ESW?
mls qos commands are not available (even though the answer guide shows
them in there).
auto qos doen't do much.
Either answer yields the same results. 2+4+4=10. 2/10 = 20%.
20+40+40=100. 20/100 = 20%. So either answer is correct.
If the question doesn't mention shaping, I wouldn't add it. Does it ask
that you enable the priority queue?
John
-Original Message-
From:
...@onlinestudylist.com] On Behalf Of John McGaughey
(jomcgaug)
Sent: Sunday, September 04, 2011 12:57 PM
To: Ashraf Ayyash
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Strange Intercom/RSVP issue
OK. I'll work on getting this. I was also going to look at the traces as well
to see what
maximum sessions software 4
associate application SCCP
-Original Message-
From: Ashraf Ayyash [mailto:ash.ayy...@gmail.com]
Sent: Monday, September 05, 2011 2:11 PM
To: John McGaughey (jomcgaug)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Strange Intercom/RSVP
The bind didn't work either. Upgrading the IOS now.
-Original Message-
From: Ashraf Ayyash [mailto:ash.ayy...@gmail.com]
Sent: Monday, September 05, 2011 2:20 PM
To: John McGaughey (jomcgaug)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Strange Intercom/RSVP issue
IOS upgrade didn't help either. :(
-Original Message-
From: Ashraf Ayyash [mailto:ash.ayy...@gmail.com]
Sent: Monday, September 05, 2011 2:20 PM
To: John McGaughey (jomcgaug)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Strange Intercom/RSVP issue
and bind
Ayyash [mailto:ash.ayy...@gmail.com]
Sent: Monday, September 05, 2011 3:17 PM
To: John McGaughey (jomcgaug)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Strange Intercom/RSVP issue
catch it ,
CSCtg88293 Intercom: Talkback fails when phone is using RSVP mtp
intercom
Yes I have.
-Original Message-
From: Ashraf Ayyash [mailto:ash.ayy...@gmail.com]
Sent: Sunday, September 04, 2011 12:06 PM
To: John McGaughey (jomcgaug)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Strange Intercom/RSVP issue
have you configured the QOS section
OK. I'll work on getting this. I was also going to look at the traces as well
to see what port CUCM is telling the phone to send RTP to.
Thanks!
-Original Message-
From: Ashraf Ayyash [mailto:ash.ayy...@gmail.com]
Sent: Sunday, September 04, 2011 12:55 PM
To: John McGaughey (jomcgaug
Check your CCM group on the GW config in CUCM. Does the group contain
the Sub?
John
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of
michael.se...@compucom.com
Sent: Sunday, September 04, 2011 3:42 PM
To:
Hello,
I'm in workbook 1, lab 9A. I'm testing the Intercom feature between the
assistant and manager. It doesn't matter who initiates the intercom, I
get no audio to the target phone.
I pressed ?? on each phone, and the initiating phone I see packets going
out, and on the target phone I
Do SIP Rules support +Dialing? For example if I have the following SIP
Dial Rule:
Pattern - \+1212...
Timeout - 0
On a SIP phone missed calls shows +12123945001. If I try to dial this
number I have to wait for the interdigit timeout. It seems to ignore
the \+1212... pattern I
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