Hi Mark:
If pt-br2 is not part of hq and br1 css you won't have to wait for interdigit
timeout, then create a new pt only for phones ie pt-phones and add it to all css
hth
> From: m...@markholloway.com
> Date: Mon, 21 Jun 2010 22:53:39 -0700
> To: m...@markholloway.com
> CC: ccie_voice@
, 2010 at 2:20 PM, Berry, Matthew J.
wrote:
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
To: Angel Perez
Cc: osl osl
Sent: Mon Jun 21 16:04:44 2010
Subject: Re:
Hi:
Unicast is not permited beetween sccp phones for CME (thanks Amy), so no need
for Whireshark :) you can only test uni from pstn
thx
> From: ghopk...@wolf-rock.co.uk
> Date: Mon, 21 Jun 2010 18:11:54 +0100
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] CUCME Unica
Please paste your config
Subject: RE : [OSL | CCIE_Voice] SIP phones for CME
Date: Mon, 21 Jun 2010 13:54:33 +0100
From: naoufal.kerbo...@cbi.ma
To: gorr...@hotmail.com; ccie_voice@onlinestudylist.com
No I'm working on PL vRack
Message d'origine----
De: A
Are you working on your own gear?
If so check that your phones have the correct fw
hth
Date: Mon, 21 Jun 2010 13:19:36 +0100
From: naoufal.kerbo...@cbi.ma
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] SIP phones for CME
Hi guys,
I'm working on lab9 Vol2, and I hav
voice@onlinestudylist.com
Awesome I will give it a try and let you know.
Thx Mike
On Mon, Jun 21, 2010 at 7:05 AM, Angel Perez wrote:
Mike:
This have been discussed previously in the list (make a search), although
phones config is not showed with srst auto none, you can add ephones if you
Sun, 20 Jun 2010 17:28:59 +0530
Subject: Re: [OSL | CCIE_Voice] Connected number display
From: voip.ccieci...@gmail.com
To: gorr...@hotmail.com
CC: siddas...@gmail.com; ccie_voice@onlinestudylist.com
i tested bot the RP first.. then i did a no mgcp command on GW1
On Sun, Jun 20, 2010 at 4:52 PM
How
can you add the "privacy off" command to the ephones, if the ephone
configuration is not in the config ?
Regards,
Mike
On Mon, Jun 21, 2010 at 3:40 AM, Angel Perez wrote:
Hi:
For srst mode auto none, just add the following
ephone 1
privacy off
ephone 2
privacy off
h
Hi:
For srst mode auto none, just add the following
ephone 1
privacy off
ephone 2
privacy off
hth
Date: Sun, 20 Jun 2010 20:53:20 -0400
From: 2xcci...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Privacy - SRST Mode Auto Provision None
I hav
come back.
Is there any other way anyone would think of??
On Sun, Jun 20, 2010 at 12:00 AM, Angel Perez wrote:
Hi Ash, I think that to change calling number at phone display you may do
transformation at rp level, correct me if i'm wrong
thx
Date: Sat, 19 Jun 2010 12:34:08
like
XXX and XX under Route list detail level. I have not tested it so
give it a try and let us know how it works.
Ash>
Angel Perez wrote:
Hi:
The only way I can imagine to make this work is with to different route
patterns, instead with one route pattern and a route list with two
, they're all getting trunked. If
I fat-finger a vlan number...
Not that any of that should be hard to troubleshoot, but on test day, I just
don't want any extra, self-induced stress.
Just my two cents!
Scott
http://ccie.meganandscott.com
Blogging my way to my 8/16/2010 lab exam da
Well done Ash :) Very good job
Date: Fri, 18 Jun 2010 19:46:07 +0100
From: siddas...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] CCIE Voice #26244
Hello all,
I went to Brussels yesterday and just an hour before learned that I am now
officially CCIE Voice. It
In case of mgcp and h323 gw as backup one rp would be enough becouse h323 gw
calls would display the number as it egress the UCM to h323 gw, so in this case
you would set DDI at route pattern to meet mgcp gw "phone display"
requirements, then use rl detail to meet h323 gw "phone display" requir
Hi check Amy reply:
http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg16368.html
hth
From: engnasse...@hotmail.com
To: ccie_voice@onlinestudylist.com
Date: Sat, 19 Jun 2010 13:43:23 +0300
Subject: [OSL | CCIE_Voice] Ver.2 Lab 3 Messaging
Hello Everyone,
I am trying to
Hi:
The only way I can imagine to make this work is with to different route
patterns, instead with one route pattern and a route list with two options,
something like this:
rp1: 91[2-9]XX.[2-9]XX DDI PREDOT, PT=br1-local-first-option
rp2: 91.[2-9]XX[2-9]XX DDI PREDOT, PT=br1
Hi:
In case that it's not specified, would you set the native vlan? And would you
set it for data or for servers vlan in case of hq?
Or simply would you let the vlan1 to be the native vlan?
Thanks
__
Hi, you can avoid this behaviour configuring a voice translation rule at h323
gw at incoming dial-peer from ucm route pattern. Transform calling number to rd
number. Also you can do this at route pattern with a mask.
hth
Date: Fri, 18 Jun 2010 18:11:06 +0200
From: findko...@gmail.com
To:
Hi:
from CUE:
voicemail callerid
hth
Date: Thu, 17 Jun 2010 07:24:17 +1200
From: dberlin...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Lab 3 Volume 2 > SRST CUE Unknown caller
Hello
In the following scenario: Phone 1002 rings 3002 in SRST m
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 specifi
Hi:
dial-peer voice n pots
clid strip ! no calling number
clid strip name ! no calling name
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 thi
Sent from my phone
On Jun 15, 2010, at 7:26 AM, Angel Perez wrote:
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 u
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 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 s
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 trie
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 becous
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; ciscovoiceg...
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: ccie_voice@onlinestudyl
9
1st Lab Attempt: Aug 16, 2010
On 6/15/2010 3:37 AM, Angel Perez wrote:
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
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
here is no need for srst auto prov all and dialpeer hunt 3 etc...
hth
On Mon, Jun 14, 2010 at 3:20 PM, Angel Perez wrote:
Hi:
Did you manage to make this work?
Finally I got some time to relab it, if you are interested let me know and I'll
post my working config
thx
Hotmail: Fr
Hi:
Did you manage to make this work?
Finally I got some time to relab it, if you are interested let me know and I'll
post my working config
thx
_
Hotmail: Free, trusted and rich
Hi:
No, to view presence status of your contacts:
Add ippm service at ucm
Subscribe to desired phones
>From phone access ippm service and finally add contacts from there (you will
>see the option in the menu)
Or better integrate with ad, search from upc and double click on the conta
the smtp domain name in ucon and rebooting it.
Sent from my phone
On Jun 11, 2010, at 7:49 AM, Angel Perez wrote:
Hi:
I've the following DNS configuration:
cue-> cue.cisco.lab
cuc-> cuc.cisco.lab (ip address 150.200.30.13)
Both cue and cuc have properly dns address
Hi:
I've the following DNS configuration:
cue-> cue.cisco.lab
cuc-> cuc.cisco.lab (ip address 150.200.30.13)
Both cue and cuc have properly dns address and domain configured
Cue config:
network location id "440"
email domain cue.cisco.lab
name "cue"
end location
Very very good job Roger :)
From: roger.kallb...@cygate.se
To: ccie_voice@onlinestudylist.com
Date: Fri, 11 Jun 2010 14:13:23 +0200
Subject: [OSL | CCIE_Voice] I passed CCIE Voice (# 26199)
I took my lab yesterday, first attempt, just got the score report. I passed :-)
I will write down
maximum sessions 19
associate application SCCP
Cheers
On Thu, Jun 10, 2010 at 8:12 PM, Angel Perez wrote:
Hi:
You need software mtp from ios not from ucm, make sure that ios mtp are
configured and registered, to be sure that mtp is working verify it with sh
sccp or from ucm.
Once you have ios
n SCCP
Cheers
On Thu, Jun 10, 2010 at 8:12 PM, Angel Perez wrote:
Hi:
You need software mtp from ios not from ucm, make sure that ios mtp are
configured and registered, to be sure that mtp is working verify it with sh
sccp or from ucm.
Once you have ios mtp registered add a mrg and
Hi:
Your mixing traffic shapping with link fragmentation and interleaving
The first one (frame relay traffic shapping or cb traffic shapping) limits the
amount of bandwith that you send to a link for example to avoid service
provider policing to this traffic.
The second one is a lin
The better one is the one in wich you pass :)
From: engnasse...@hotmail.com
To: lakpr...@gmail.com; cci...@gmail.com
Date: Thu, 10 Jun 2010 15:33:48 +0300
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Better Voice Lab Locations
Yes sure,
Now Dubai is available
Reg
+0200
Subject: Re: [OSL | CCIE_Voice] UCCX CSQ hunting order issue
From: findko...@gmail.com
To: gorr...@hotmail.com
CC: wolfsru...@gmail.com; ccie_voice@onlinestudylist.com
VMWare ESXi 4.0.0
UCCX 7.0(1)_Build168
On Thu, Jun 10, 2010 at 10:33 AM, Angel Perez wrote:
Hi:
Wich vmware version do
Hi:
Wich vmware version do you have installed?
I'm working with esxi and I've never seen this
Date: Thu, 10 Jun 2010 09:28:32 +0200
From: findko...@gmail.com
To: wolfsru...@gmail.com
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] UCCX CSQ hunting order issue
I woul
Hi:
You need software mtp from ios not from ucm, make sure that ios mtp are
configured and registered, to be sure that mtp is working verify it with sh
sccp or from ucm.
Once you have ios mtp registered add a mrg and include all ucm software mtp and
cnf, then do not include this mrg to
...@hotmail.com
CC: ccie_voice@onlinestudylist.com
Thanks Angel,
Thats exactly what I was after.
Clare
On Wed, Jun 9, 2010 at 6:37 PM, Angel Perez wrote:
Hi:
For pstn and wan connectivity you dont need any connection to sw.
But if you are planning to use pstn gw as a remote gk, ntp, etc
Hi:
For pstn and wan connectivity you dont need any connection to sw.
But if you are planning to use pstn gw as a remote gk, ntp, etc you can use one
fast eth to connect to hq sw at servers vlan.
Then at the other fast eth port on your pstn gw you can plug pstn phone
directly, config
Hi:
I tested it some time ago an it didn't works... so I needed to use voice
translation...
I think that other people had problems with this also
Give it a try a let us know
hth
Date: Tue, 8 Jun 2010 20:36:49 -0500
From: ciscovoiceg...@gmail.com
To: ccie_voice@onlinestudylist.com
Hi:
In real live thats depend on the timezone, for US time zones (PDT, EDT, ...) is
not necessary becouse the default has the correct date, but for example at
Europe summer time start at different week depending on the zone so you should
manually configure.
In the lab I suppose that yo
Hi all:
Anybody knows if .wav files recorded with CUE prompt manager (aka TUI) are
valid for bacd tcl scripts?
BACD prompts are .au files but it think that .wav are also valid, anybody can
clarify this?
Thanks
___
Jun 7 22:07:44.928: ////GK/gk_process: QUEUE_EVENT
(minor 0) wakeup
Jun 7 22:07:44.936: ////GK/gk_process: QUEUE_EVENT
(minor 0) wakeup
HQ-R1#
HQ-R1#do u all
On Mon, Jun 7, 2010 at 6:19 AM, Angel Perez wrote:
Hi:
Can you paste the following:
ge Newcall Softkey and then within a fraction of a second it disappears
with the new call softkey template and I get a dial tone.
Router has been rebooted as well.
did you give it a go Angel?
cv
On Mon, Jun 7, 2010 at 10:37 AM, Angel Perez wrote:
Hi, you are partially right:
When y
on managing the list at
ccie_voice-ow...@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. Setting up Voicemail to send Email - CME/CUE 7.0 (Ashar Siddiqui)
2. Re: S
Hi:
Did you add:
ip http server
ip http server path...
telephony-servi
web admin system name ...
For your question I'm not sure becouse I haven't test email notification, it
but if you want to send voicemail to email inbox you can do it with imap
configuration is not d
't help me as well
any other idea ???
Thx
Dani
On Mon, Jun 7, 2010 at 4:44 AM, Angel Perez wrote:
Hi:
The outvia and invia comands are for ip to ip gw and the show gateke calls
doesn't show an ip2ip gw call...
Date: Sun, 6 Jun 2010 19:48:27 -0400
From: daniyal.
Hi, CUC will be added by default to ucm once you active cuc services, the same
happend with cups, once you activate cups services it will be added
automatically to ucm
Try it a let us know
Date: Sun, 6 Jun 2010 21:59:12 -0500
From: ciscovoiceg...@gmail.com
To: ccie_voice@onlinestudylis
AM, ccie voice wrote:
Thanks Angel.
I am at work but will give it a go today.
I am just wondering that everytime when phones will go in SRST..do I have to go
into ephones and add the button? this does not look practical.
I will give it a go anyways.
vc
On Mon, Jun 7, 2010 at 9:39 AM,
do I have to go
into ephones and add the button? this does not look practical.
I will give it a go anyways.
vc
On Mon, Jun 7, 2010 at 9:39 AM, Angel Perez wrote:
Hi:
Once the ephones have registered to srst add privacy button under ephone.
Let us know, hth
Date: Sun, 6 Jun 2010 21:4
Hi:
The outvia and invia comands are for ip to ip gw and the show gateke calls
doesn't show an ip2ip gw call...
Date: Sun, 6 Jun 2010 19:48:27 -0400
From: daniyal.vo...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Gatekeeper 4 Digit call
I checked c
Hi:
Once the ephones have registered to srst add privacy button under ephone.
Let us know, hth
Date: Sun, 6 Jun 2010 21:46:26 +0100
From: cci...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] CBarge in SRST ~(Again)
Hi,
Sorry for the incomplete email earlier
Hi:
Do you have privacy on at any of the phones before going to srst?
Also sometimes you have to reload the gw with cme srst to make it works properly
hth
> From: 1.matt.h...@gmail.com
> Date: Sat, 5 Jun 2010 22:55:23 -0500
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE
Hi:
Sometimes you have to reload the gw to make presence works
hth
Date: Sat, 5 Jun 2010 12:18:43 +0200
From: findko...@gmail.com
To: salman.shaik...@gmail.com
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CME 7.0 Presence caller-list is not working ...
and maybe
answer that as I’ll be
breaking NDA
I think NDA would apply to those who’ve attempted or passed the lab. Others
have not agreed to any NDA
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Angel Perez
Sent: Friday, June 04, 2010 9
, 5 Jun 2010 03:19:11 +0900
Thanks, and sorry didn’t really mean to ask contents, more of a rough info.
question as in blueprints don’t say it, so was pretty much curious.
Thanks for the heads up
From: Angel Perez
Sent: Saturday, June 05, 2010 2:25 AM
To: siddas...@gmail.com ; jon1...@ho
Hi, you are correct, with mgcp there are no h323 dial-peers (from the incoming
call perspective) and signaling is backhauled to ccm directly
From: salman.shaik...@gmail.com
Date: Fri, 4 Jun 2010 13:00:46 -0400
To: siddas...@gmail.com
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE
Hi Jon, you can't ask anything about exam contents, sorry
From: siddas...@gmail.com
To: jon1...@hotmail.com; ccie_voice@onlinestudylist.com
Date: Fri, 4 Jun 2010 15:39:27 +0100
Subject: Re: [OSL | CCIE_Voice] Lab and Language settings
No, I don’t think so..
As a rule of thumb just select
Hi:
If the vlan.dat file is deleted you will get this result
Make sure that the vlan exists and also that it is active:
vlan 130
create
name data
status active
vlan 240
create
name voice
status active
hth
> Date: Thu, 3 Jun 2010 19:08:22 -0400
> From: amccar...@cciequ
Hi:
If you have a tftp server program for windows you will be able to upload the
files directly to router flash from your pc, this will be shorter :)
hth
From: salman.shaik...@gmail.com
Date: Thu, 3 Jun 2010 12:54:39 -0400
To: chik...@yahoo.com
CC: ccie_voice@onlinestudylist.com
Subje
ing locations
throughout the United States, Europe, South Asia and Australia. Be sure to
visit our online communities at www.ipexpert.com/communities
<http://www.ipexpert.com/communities> and our public website at
www.ipexpert.com <http://www.ipexpert.com/>
From: Angel Perez
Hi Daniyal:
Check question 10 and answer 11 from Ben at may 17
https://learningnetwork.cisco.com/message/68646#68649
Date: Thu, 3 Jun 2010 10:06:03 -0400
From: daniyal.vo...@gmail.com
To: earl.ho...@pcmall.com
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CME
/guide/lsw_hwic_ethsw_ic_ps6441_TSD_Products_Configuration_Guide_Chapter.html#wp1051730
- Original Message -
From: Peter Farkas
To: Angel Perez
Cc: osl osl
Sent: Wednesday, June 02, 2010 8:36 PM
Subject: Re: [OSL | CCIE_Voice] show vlan-s brief
I also cheked mine. Further
South Asia and Australia. Be sure to
visit our online communities at www.ipexpert.com/communities
<http://www.ipexpert.com/communities> and our public website at
www.ipexpert.com <http://www.ipexpert.com/>
From: Angel Perez
Date: Wed, 2 Jun 2010 17:21:42 +
To: osl osl
Su
Hi:
When I configure the swich port of my hwic-esw with the "old" method:
interface range fas 0/3/0 - 3
swicht mode trunk
swicht trunk encap dot1q native vlan 200
swicht voice 300
I get the following result:
sh vlan-s bri
VLAN Name Status
Hi:
When I call mva number from pstn, the rd number is matched so I enter de pin
12345 # then 1 # for call and finally the number I want to call 911 #
The problem I have is that between the prompts there is a silence of 5 - 7 sec,
sometimes the prompt doesn't sounds, but if I press the c
Angel Perez
Sent: Monday, May 31, 2010 9:16 PM
To: osl osl
Subject: [OSL | CCIE_Voice] ip rsvp bandwith
Hi all:
For two g729 calls, how much band would you set at ip rsvp bandwith
These are the two options:
1: ip rsvp bandwith 64 (40 + 24)
or
2: ip rsvp bandwitn 80 (40 + 40)
The first one
Hi all:
For two g729 calls, how much band would you set at ip rsvp bandwith
These are the two options:
1: ip rsvp bandwith 64 (40 + 24)
or
2: ip rsvp bandwitn 80 (40 + 40)
The first one looks find becouse once the call is completed the rsvp bandwith
is reduced to 24 and a
29 May 2010, at 19:08, Brian Valentine wrote:
You should try "debug ccsip messages" on the PSTN or CUBE router. It will show
you the codec negotiation.
On May 29, 2010 1:55 PM, "Angel Perez" wrote:
Hi:
I have a sip trunk to my pstn router I'm trying to check th
Hi:
I have a sip trunk to my pstn router I'm trying to check the codec that the
call is using but I can't this info at ucm traces or pstn gw debugs.
I have try sip stack traces at ucm and also deb ccsip all at pstn, but I can't
this info
Any suggestion?
Hi:
Do you have any called party transformation in the gw called party
transformation calling search space?
hth
Date: Fri, 28 May 2010 11:35:42 -0500
From: tamnhu...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] H323 Gateway - Called Party Number Type: Unkno
Hi again:
In my case is working like this:
br2#sh flash:
-#- --length-- -date/time-- path
86 0 Apr 15 2010 17:14:52 Desktops
87 0 Apr 15 2010 17:14:56 Desktops/320x196x4
88 647 Apr 15 2010 17:15:00 Desktops/320x196x4/logo.png
89 239 Apr 15 201
Hi:
Take a look to DNA on UCM, maybe you can have some information there
hth
Date: Thu, 27 May 2010 17:48:59 -0700
From: lme...@signal.ca
To: bga...@gmail.com
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] MVA issue
Checked that..
I got TAC involved.. They do n
Hi:
Your problem is here:
flash:/Desktops/320x196x4/
flash file should look like this:
flash:Desktops/320x196x4/
hth
Date: Thu, 27 May 2010 19:46:06 -0400
From: salman.shaik...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] CME Background Image Iss
For all these extension wouldn't be scalable...
I think that this behaviour could be changed system wyde but I can't remember
how
From: siddas...@gmail.com
To: r.ochi...@mfient.com
Date: Wed, 26 May 2010 12:03:46 +0100
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE
Hi:
You can try this:
Add a cue aa with a dummy ext for example 2999, then at this aa just transfer
to 2904 vm, then at 2905 phone add a speed dial to 2999 with a label like 2904
VM, this way when 2905 wants to transfer to 2904 vm the user should press
transfer + speed dial + transfer
Hi Matthew:
Did you set calling name and epnm at line from the RDP? (Not from the phone)
Date: Tue, 25 May 2010 20:11:49 -0500
From: ciscovoiceg...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Vol 2 Lab 4 Section 3.1 - Mobile Connect Question
Fellow nerds,
I am
: ccie_voice@onlinestudylist.com
have you added the ac application user to the Standard CTI Allow Call Park
Monitoring group?
2010/5/25 Angel Perez
Hi:
Yesterday I was trying to create an AC pilot point, but the pilot didn't get
registered, I created a user with cti rights and with p
Hi:
Yesterday I was trying to create an AC pilot point, but the pilot didn't get
registered, I created a user with cti rights and with phones controlled, then I
configured this user as the ac user...
I tried to reset the ac service but with no luck
Any other suggestion?
Thanks
Hi:
You can use incoming called number 7771234 at dial-peer range 7771000-7771005
and 7771235 at dial-peer range 7771006-77710010
hth
> Date: Mon, 24 May 2010 16:59:08 +1000
> From: vip...@gmail.com
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] CME - direct
hi, it´s a cme 7 version issue, please read this:
https://learningnetwork.cisco.com/message/5
hth
Date: Sat, 22 May 2010 14:17:57 -0400
From: 2xcci...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] CME Globalization ?
Is it possible to globalize a "calli
Hi:
Dev Mob is working as expected (the phone is taken romain sensitive setting
from the roaming device pool) but when I click on View Current Device Mobility
Settings link the roaming settings are not shown at the pop up windows... but
the phones has the correct settings...
The same th
/authentication/authenticate.do
is correct
On Thu, May 20, 2010 at 6:51 PM, Angel Perez wrote:
Hi:
Wich is the correct one?
This one?:
url authentication http://cue/voiceview/authentication/authenticate.do
thx
Date: Thu, 20 May 2010 18:01:55 +0530
Subject: Re: [OSL | CCIE_Voice] CME-CUE
.do
is correct
On Thu, May 20, 2010 at 6:51 PM, Angel Perez wrote:
Hi:
Wich is the correct one?
This one?:
url authentication http://cue/voiceview/authentication/authenticate.do
thx
Date: Thu, 20 May 2010 18:01:55 +0530
Subject: Re: [OSL | CCIE_Voice] CME-CUE VOICEVIEW
Fr
authentication URL
On Thu, May 20, 2010 at 4:51 PM, Angel Perez wrote:
Hi all:
I've the following problem with voiceview and CME:
I've sucsesfully configure the voiceview service for phones, I access the
service (no pin asked) but once I see the menu options (1 inbox, 2 Send
Messa
Hi all:
I've the following problem with voiceview and CME:
I've sucsesfully configure the voiceview service for phones, I access the
service (no pin asked) but once I see the menu options (1 inbox, 2 Send
Messages, 3 etc) I can't select any of the options neither logout with Logout
butt
Hi, start with this :)
http://pushkarbhatkoti.wordpress.com/category/cue-voicemail-vpim-networking-cue-to-unity-in-10-minutes/
http://www.brainbump.net/2009/04/easy-approach-for-configuring-and-setting-up-cisco-unity-express/#more-503
http://www.cisco.com/en/US/docs/voice_ip_comm/unity
Hi Matthew:
If you set br1 device pool and hub-none loc at gw settings and this device
pool has location br1, in this case, dp general configuration will overwrite gw
specific configuration, hub-none location is an exception to the general rule.
Check the first 3 paragraphs of this pos
Hi, the bandwith assigned to a location affects incoming and outgoing call
to/from this location, this is way is only valid for a hub and spoke topology
(all the calls go throw the hub). If your topology is not hub and spoke you
should use rsvp wich match one to one locations (you specify res
Hi, check this topic:
http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg16572.html
hth
From: wormh...@sch.hu
To: ccie_voice@onlinestudylist.com
Date: Tue, 18 May 2010 20:24:30 +0200
Subject: [OSL | CCIE_Voice] cannot dial from MVA
Gents,
I have an issue with MVA. MVA
Ummm, did you add
sip
bind all source interface
???
Date: Tue, 18 May 2010 20:01:06 +0300
Subject: Re: [OSL | CCIE_Voice] CUBE Issue - Vol2 Lab 8 - BR2 Calls to CUCM
From: waelag...@gmail.com
To: gorr...@hotmail.com
CC: ccie_voice@onlinestudylist.com
Done, but still the same :(
B
Hi add this under
telephony service
max-dn 1
max-phone 1
ip source add 10.10.202.1
hth
Date: Tue, 18 May 2010 19:52:42 +0300
From: waelag...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CUBE Issue - Vol2 Lab 8 - BR2 Calls to CUCM
All config seems fine:
Hi:
Make sure that you have all the neccesary commands under telephony-service
max-dn
max-phone
source add
Of course you also need sdspfarm related commands you may already have
hth
Date: Tue, 18 May 2010 19:42:29 +0300
Subject: Re: [OSL | CCIE_Voice] CUBE Issue - Vol2 La
Yes it's possible I made it
From: naoufal.kerbo...@cbi.ma
To: akashapa...@yahoo.com
Date: Mon, 17 May 2010 23:55:49 +
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE Voice Schedule
90 days before the exam date if you Will pay by wire transfer, you CAN schedule
Hi:
Add the following:
dspfarm profile 1 trans
shut
codec g729r8
no shut
By default g729r8 is not configured
Let us know
Date: Tue, 18 May 2010 19:21:24 +0300
Subject: Re: [OSL | CCIE_Voice] CUBE Issue - Vol2 Lab 8 - BR2 Calls to CUCM
From: waelag...@gmail.com
To: gorr.
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