Hi:
I was wondering how can you add privacy/privacy off to the ephone if you are
setting srst auto none?
The only way I can imagine is changing from srst auto all to auto none once the
ephone are configured.
Correct me if i'm wrong
thanks
Date: Mon, 14 Jun 2010 18:06:15
Hi all,
I was going through the different ways we configure FRF.12 and MLPP on
serial interfaces and came across few findings.
When we are configuring FRF.12, we will use the command auto qos voip
trust under the DLCI. While when we are configuring MLPP then we will be
using auto qos voip trust
Also, after the Auto QOS generates a lot of classes etc. We do edit few
things here and there. Just wanted to confirm that is it a good practice to
remove rtp header compression?
I use to remove it always but now I am getting conflicting feedback that
should we remove it or not?
interface
Hi,
I'm trying to do a real time tracing of ip phone activity, for example; when
the phone goes off hock, the line seized, CUCM sending the signaling and
tone etc...
I'm using RTMT -- Real time tracing -- View real time data to do so but
unsuccessful.
Anyone know which services should be select
Kobel,
In my opinion, you should only retain the frame-relay ip rtp
header-compression under the frame-relay DLCI if you are asked to
compress the rtp packets. Because we're dealing with a slow-speed link,
auto qos tries to be helpful by adding in this command.
My general stance when it
Hi:
Just to add something to Matthew's reply, be sure that you set the correct
compression method either frame relay (activated by default with auto qos voip
trust in links with 768k bandwith or less) or class based (compress header ip
rtp at desired class) .
You can't have both at the
I would turn on detailed tracing through CUCM Serviceability and then
monitoring the SDL or SDI traces (I always forget which one) through the
CUCM CLI. It's the best way I know how.
file tail activelog /cm/trace/cmi/sdl
file tail activelog /cm/trace/cmi/sdi
*Matthew Berry*
/A+, CCENT,
Angel -
I think you are right. The only way I can see of configuring privacy
on/off would be through the ephone section itself. Privacy isn't an
option with an ephone-template, otherwise you could have set it there.
You could possibly set no privacy under telephony-service, but that
would
My vote on this goes to Matthew. If not clearly asked for, better to remove any
unwanted parts that quto qos putts in.
Brgds,
Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB
Eric Perssons väg 21, SE-217 62 MALMÖ
Direkt: +46108787498
Växel: +46108787400
Just a side note - I also use those two commands, but with a little
modification:
file tail activelog /cm/trace/cmi/sdi recent
this makes the CLI to choose the most recent file (no need to type in the
filename yourself).
RTMT is such a waste of time, when it comes to traces ;)
BTW, the most
Also if you would have class based shaping, stay away from auto qos. Auto will
configure FRTS, ie not class based shaping. There is an example on how to setup
cb shape in the QoS SRND. Stick with that one.
Brgds,
Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB
Eric Perssons väg 21,
Hi:
These are my observations for srst and cbarge
First of all you will need a cnf bridge configured, the best way is adding srst
ip add as a third option in sccp ccm group, once your cnf bridge is registered
to srst router (it take some more times than phones) you will need a dn octo
When configuring call preservation for an H.323 gateway, I am using the
following command:
*voice service voip
h323
call-preserve*
As soon as I hit ENTER, the IOS spits back this warning/notice to me:
*Warning: Configuring media inactivity detection to avoid hung calls is
highly
I am getting an odd OSPF error after having configured my service-engine
for the CUE module:
*Jun 14 05:46:22.401: %OSPF-4-NO_IPADDRESS_ON_INT: No IP address for
interface Service-Engine0/0*
Everything appeared to function properly even with this error being
reported. Below is my example
For h323 call preservation adding
voice service voip
h323
call-preserve
And: Allow Peer to Preserve H.323 Call at ucm call manager service param
advanced
would be enough
hth
Date: Tue, 15 Jun 2010 06:45:33 -0500
From: ciscovoiceg...@gmail.com
To:
Hi:
This is becouse you are setting ip unnumbered, there is another method with ip
address, with it you won't get this error
But the error it's just cosmetic
hth
Date: Tue, 15 Jun 2010 06:48:58 -0500
From: ciscovoiceg...@gmail.com
To: ccie_voice@onlinestudylist.com;
Thanks, Angel!
*Matthew Berry*
/A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/
*_Vitals:_*
*GVoice: *+1.612.424.5044
*Gmail*: ciscovoiceg...@gmail.com
*Skype*: ciscovoiceguru
*Twitter*: ciscovoiceguru
*_Cert Stats:_*
Cisco Cert Journey Began: Jan 1, 2009
1st Lab Attempt: Aug 16,
I figured as much, but it's always better to run it by your fellow egg
heads before assuming.
Thanks, brutha'
*Matthew Berry*
/A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/
*_Vitals:_*
*GVoice: *+1.612.424.5044
*Gmail*: ciscovoiceg...@gmail.com
*Skype*: ciscovoiceguru
*Twitter*:
Call preservation is enabled as soon as you enter call-preserve, the warning is
for a best practis/recommendation for additional configuration. Look at this,
http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/cisco_ios_h323_configuration_guide/old_archives_h323/4gwconf.html#wp1150861
Regards
Roger
Search the OSL. Has been posted before. This is an informal message, nothing to
pay any attention to.
Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB
Eric Perssons väg 21, SE-217 62 MALMÖ
Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.semailto:roger.kallb...@cygate.se
Hi:
There are certain services: em, ipma, ac, axl or even dhcp and tftp that you
can activate at pub or sub.
If it is not specified you can doubt if you may activate it at pub, sub or
both, my question is what do you think is the best practice to use pub or sub,
or it is the same
Hi guys,
I was working on vol2 lab6 question 3.1, And I cannot get the Mobile
number ring on br1 phones or br2 phones as if it's coming from hq ph2,
however when I've called from Mobile PSTN to hq phones I can see that
the number is from HQ Phone2 (4002).
I know that is a problem with the
CUCM don't provide redundancy for EM.
For IPMA you can activate the service on sub or on pup also both if u
want redundoncy
On 06/15/2010 12:59 PM, Angel Perez wrote:
Hi:
There are certain services: em, ipma, ac, axl or even dhcp and tftp
that you can activate at pub or sub.
If it is not
Hi:
I was trying to setup CUE to say voicemail user name instead of phone number
when somebody left a message at voicemail, (like in CUC) but the most i can do
is just to hear phone number (voicemail callerid), after some tests my
conclusions is that it is not possible
Anybody has
Hi:
Are you sure? I'm logged right know to UCM cluster and I can activate the
service at both pub and sub...
Anyway for ipma example if redundancy is not required, would you use pub or sub
when adding the service url... that is the big question
thanks
Date: Tue, 15 Jun 2010
I dont think There is a way to configure redundancy for em. You can
activate on pub/ sub but only use one of tgem.
Let me know if i am mistaken.
Sent from my phone
On Jun 15, 2010, at 7:26 AM, Angel Perez gorr...@hotmail.com wrote:
Hi:
Are you sure? I'm logged right know to UCM cluster and
Hi:
You can't configure redundancy like with tftp but you can configure two
services one with pub ip address and other one with sub ip address, this way if
pub is down you can let the user to activate em from sub service
thanks
From: pav.c...@gmail.com
To: gorr...@hotmail.com
you can gen em redundancy by means of slb in the gateway, but it's out
of the scope of the current blueprint.
don't have any link right know but can be looked up easily @ cisco.
hth
On 6/15/10, Pavan pav.c...@gmail.com wrote:
I dont think There is a way to configure redundancy for em. You can
AFAIK, there is also another solution - round robin resolving on DNS server.
but it's also out of the scope probably.
I can't point to any document right now, but AFAIR to configure phone
services for both EM on Pub and Sub and let user manually select the working
one.
On Tue, Jun 15, 2010 at
I think your problem can occur because of SIP Gateway in CUCME config (dual
config SCCP and SIP).
Please check your CUE config section:
ccn subsystem sip
gateway address 10.10.202.1 -- sip bind.
mwi envelope-info
mwi sip outcall sub-notify
end subsystem
In the gateway address must appear
Hi could some one pls help to resolve this issue
in CME i don't want send the Calling Name on specific dial-peer but Number
suppose to go
under D channel i have configured Isdn out display ie that affecting on all
calls
but requirement is that i just want to block or restrict one
person/dial-peer
on dial-peer config: clid strip name
hth
On Tue, Jun 15, 2010 at 1:56 PM, ccievoice daniyal.vo...@gmail.com wrote:
Hi could some one pls help to resolve this issue
in CME i don't want send the Calling Name on specific dial-peer but Number
suppose to go
under D channel i have configured Isdn
Dani,
Here is an example of a dial-peer that will strip name but send correct ANI.
Doing it this way will only affect calls to 999. All other calls will send
name if you have it configured properly on the Serial Interface for the PRI.
dial-peer voice 999 pots
translation-profile outgoing
I think it's clip strip under the dial peer.
At the movies so I can't verify on my lab unless there's a killer IPX
vRack app for my iPhone. ;)
Matthew Berry
**Sent from my iPhone**
Skype/Twitter: ciscovoiceguru
Google Voice: +1 612 424 5044
On Jun 15, 2010, at 11:56 AM, ccievoice
I guess that you won't forget this one :)
Date: Tue, 15 Jun 2010 12:56:46 -0400
From: daniyal.vo...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] CME Calling Name
Hi could some one pls help to resolve this issue
in CME i don't want send the Calling Name on
yeah it's working now thx ...
On Tue, Jun 15, 2010 at 1:13 PM, Angel Perez gorr...@hotmail.com wrote:
I guess that you won't forget this one :)
--
Date: Tue, 15 Jun 2010 12:56:46 -0400
From: daniyal.vo...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject:
If we are required to do the following.
while in SRST phones should appear like they do when they are registered to
CUCM except for the message fallback displayed at the bottom of the
phones.
what items make a phone in srst mode appear exactly like in cucm
i can think of these
date/time
OK I have read everything possible on LAN QOS and here is what i am trying
to understand
if i am asked following
1. to put a certain cos value say 4 in the priority queue in one question
and then in another question i am asked
2. to guarantee queue 1 for for 20% of the traffic and the
In Lab 8, you are asked to configure iLBC between HQ and BR1 with RSVP
CAC on top of that.
The Proctor Guide tells me to set the Link Loss Type under CUCM SYSTEM
LOCATIONS to Lossy.
However, all of my testing to date seems to demonstrate that the lossy
setting does not affect whether iLBC
In the Proctor Guide for Lab 8, I am directed to setup a hardware
transcoder for the purpose of facilitating a 10-party MeetMe
conference. However, I seemed to be able to get everything setup
properly without a hardware conferencing resource.
In fact, I have done many customer
7962/5 phones support iLBC. Are you testing with these phones?
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
Without the hw cfb you will not be able to add conference participants
that use a low bit rate codec such as g729. Unless you are using a sw
cfb with a transcoder.
Vik Malhi - CCIE#13890
Senior Technical Instructor - IPexpert Inc
Telephone: +1.810.326.1444
Fax: +1.810.454.0130
Mailto:
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