It’s difficult to get back to the study routine, after having failed the exam & then taking a 2month break for no apparent reason.
A time that should have ideally been used to study and practice, has been wasted on work related activities, which don’t add any benefit to the CCIE journey. Now that’s a real waste of our precious time.
Remember that every minute of every waking hour you spend on practicing for the lab, is a dollar invested in yourself, regardless of your exam result!
I finally worked up the courage after a lot of hesitation and went online to http://www.cisco.com/web/go/ccie to book my lab exam worth US$1400 (that’s not chump change in my part of the world).
After putting in my written exam information, I had to make the difficult choice of selecting a date. It had to be something reasonable, meaning not 1 month (too little time) or 6 months (too much time –> you end up slacking). In the end, I decided to go for the 90 day payment deadline and hoped that it would be enough to practice, revise, practice & practice some more & then some. I settled on a date in February and hit the Pay Now button!
* This was a post in the past, that I couldn’t bring myself to post. So, it’s official now, published online.
Scenario – Getting NTP to synchronize with the right source
People keep complaining how their NTP clock doesn’t synchronize. Here is hopefully a scenario that will cover and resolve any issues.
1. Determine master NTP site and configure it as below:
ntp master
2. Determine NTP main peer and configure as below:
ntp source [interface]
ntp server [master NTP address]
3. Determine NTP slave and configure as below:
ntp server [main peer NTP address]
Note:
ntp clear drift
show ntp status –> Should show Clock is Synchronized
show ntp associations –> Should show * with the NTP peer/master – confirms synchronization
show ntp associations detail
Sample Output
Branch-Router#sh ntp status
Clock is synchronized, stratum 10, reference is 192.168.1.254
nominal freq is 250.0000 Hz, actual freq is 249.9999 Hz, precision is 2**24
reference time is CFB54ACA.D0B73A56 (07:14:34.815 PAK Sun Jun 6 2010)
clock offset is 0.0227 msec, root delay is 0.02 msec
root dispersion is 0.05 msec, peer dispersion is 0.00 msec
loopfilter state is ‘CTRL’ (Normal Controlled Loop), drift is 0.000000319 s/s
system poll interval is 128, last update was 135 sec ago.
Branch-Router#sh ntp assoc
address ref clock st when poll reach delay offset disp
*~192.168.1.254 192.1.1.100 9 64 128 377 0.000 22.769 4.537
* sys.peer, # selected, + candidate, – outlyer, x falseticker, ~ configured
References:
Verifying NTP Status with the show ntp associations Command
We will be covering the SIP phones config on CME in this post. And giving a comparison on how it compares with the SCCP configuration. Steps of configuration for SCCP would be covered in the next post.
Note: We can’t use telephony-service setup for this in the lab, as we risk getting our SIP phones registered as SCCP phones, reverting back to SIP is time-consuming. Also, keep in mind that the telephony-service setup command is being phased out, so better to get used to the CLI configuration or build yourself some templates.
Below are the steps one by one:
1. VoIP Services Configuration
voice service voip
allow-connections sip to sip ! Allows SIP connections
sip
registrar server expires max XXX min XXX ! mandatory – defines registration timer in milliseconds
2. Voice Register Global (think telephony-services)
voice register global
mode cme
source-address 192.168.1.1 port 5060
max-dn 2
max-pool 2
tftp-path flash:
create profile
date-format m/d/y
time-format 12
3. Voice Register DN XX (think ephone-dn)
voice register dn 1
number 5681
voice register dn 2
number 5682
4. Voice Register Pool (think ephone)
voice register pool 1 ! for first phone
id mac AAAA.BBBB.CCCC
type 7961 ! mandatory – defines type of config file for phone
username 5681 password cisco
number 1 dn 1
dtmf-relay rtp-nte
!!!! If asking for DTMF RFC 2833 use rtp-nte, otherwise use sip-notify !!!!
codec g711ulaw ! SCCP phones create FXS & Pots Dial-peers, SIP creates VoIP Dial-peers –> default codec is G.729
description 97236725681 ! gives the header bar display
voice register pool 2 ! for second phone
!!! same exercise as above
5. Dial-plan & template ( avoid pressing ‘Dial’ on non-KPML phones – 7940/60)
voice register dialplan 1
type 7940-7960
pattern 1 5… timeout 0
!!!!! for phones that DO NOT SUPPORT KPML, digits sent to CME without Dial key !!!!
voice register template 1
dialplan 1
6. Voice Register Global ( apply changes & reset)
voice register global
create profile
reset
References:
Configuring CME for SIP to Allow Phones to Register
CUCME – Basic SIP endpoints configuration
CUCME – System Administration Guide
Cisco started offering the G.722 Wideband audio codec a few years back. Our first experience with it was with the Cisco Unified CallManager 5. In the initial days it was possible to request the wideband handset for the old phones 7941/60/70/71 as an accessory. On the newer models 7942/62/75 these wideband handsets are provided by default. More technical details can be found in the references at the bottom of this post.
The thing to note here is when you have two phones that support the G.722 codec, by default, the codec negotiated for them would be the G.722, not G.711. Why you might ask? Well, if the Region setting (under Device Pool & Region) is set to use G.711 then any codec with the equivalent amount of bandwidth or less could be used. Since the G.722 gives better quality, so it is preferred over the G.711.
Under certain conditions you might want to force the phones to use G.711 and would therefore need to disable the advertisement and selection of G.722 code. This can be carried out in the following easy steps below.
How to Disable G.722 Codec in CUCM 7
1. Browse to System>Enterprise Parameters
2. Search for Advertise G.722 Codec
3. Disable the Advertisement of G.722 Codec
4. Browse to System>Service Parameters
5. Select the CUCM server and the CallManager Service
6. Search for G722 Codec Enabled and disable it
References:
Wideband Audio and IP Telephony
Hello everyone.
This is our first post on the way to (hopefully) becoming Certified Cisco Voice Expert (CCIE-Voice). The focus for this blog is to create blog posts that can be used as references at a future time for quick revision of selected sections or sub-technologies.
