Common guys, no one know this answer of this?
Come out of CCIE lab questions for a change (we had enough of them
already) and see if you can find a way out of this one?
Ash
On 11/03/2011 22:15, Ashar Siddiqui wrote:
Hello,
I would be glad if someone can help in this. I am working on a CME
Hi,
Don't be freaked out and don't lose hope. Passing in first attempt
doesn't mean you know everything and technically speaking there is no
difference in which attempt you clear your lab as long as you pass it.
My study partner, an individual with tremendous knowledge and experience
in every
Hello,
I would be glad if someone can help in this. I am working on a CME
hunting setup for a customer where when a call is not answered/busy on
any phone, it will start ringing a hubt listcustomer has reported
that although hunting is working perfectly fine but the problem is when
it sta
Congrats!
Don't get carried away :)
Ash>
On 06/03/2011 08:38, voicemail voicemail wrote:
YAH
I am so happy to pass.. CCIE VOICE:))
So here is the updates guys :) :) :)
Well NDA so ping me if u need it
I got the result on friday ev
Is this a MGCP gateway?
Make sure the L3 binding has been tear down. (no isdn bind-l3 ccm-manager)
Also check if the gateway is registering with redundant ccm, if it doesn
then issue no ccm-manager redundant-host x.x.x.x.
Ash>
CCIE#26244
http://ccieash.wordpress.com/
From: ccie_v
Yes incoming called-number 1999.. Will work for MWI.
From: Mritunjay Kumar [mailto:mjs...@gmail.com]
Sent: 20 January 2011 13:48
To: Ashar Siddiqui
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] how to set mwi on/off number in unity
express
Hi Ashar ,
thanks for reply
You will have to create mwi on and mwi off ephone dn. Let's say:
ephone-dn 8
number 1990
mwi on
!
!
ephone-dn 9
number 1991
mwi off
!
!
If you want to set MWI on, dial 19902601 and it should turn the light on
(2601 being any ext.)
Ash>
From: ccie_voice-boun.
You can do it either way but I would prefer the access port mode as trunk
method is normally used on old switches where access mode doesnt work.
Ash>
CCIE#26244
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roger Källberg
Sen
You dont need MWI on/off dialpeers for CUE-CUCM integration. They are
already there.
This is the procedure I practically use to perform and never had CUE-CUM
integration issues.
http://ccieash.wordpress.com/2010/07/17/cisco-unity-express-setup-cue-cucm/
Thanks
Ash
From: ccie_vo
Hello all,
For those of you who are not aware of my blog, please visit the following
link for CCIE Voice lab strategy. I thought it will be helpful for most of
you gearing up for lab.
http://ccieash.wordpress.com/2010/07/01/ccie-voice-lab-strategy/
While Matthew has already pointed o
This always worked for me.
http://ccieash.wordpress.com/2010/07/10/ip-phone-background-image-cme/
Ash>
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Shrini
Sent: 11 January 2011 18:29
To: Rahul Kapor
Cc: ccie_voice@onlinestu
Make sure under telephony-service or call-manager-fallback you add voicemail
command with complete PSTN voicemail number.
There must be a dialpeer to handle VM routing over PSTN.
HTH
Ash
On Wed, Jan 12, 2011 at 9:50 AM, Shrini wrote:
> Hi Experts,
>
> How can I get the Vmail speeddial button
Are you doing http://ccm-addres/ccmadmin
OR
https://ccm-address/ccmadmin
Thanks
Ash
On Wed, Jan 12, 2011 at 11:02 AM, Rashid Khan wrote:
> Thanks Ashar for your prompt Response..
>
> Tomcat service is started... and I also restarted that Cisco Tomcat
> Service... but stil
Check if tomcat service is running by doing an SSH to CCM and running
utils service list.
also CCM on VM always takes sometime to show webpage. Give like 8-10
minutes and check the service I mentioned above.
You can restart the service as well from SSH... *utils service restart
Cisco Tomcat.
A
Hi Anis,
What is Lab7 and what are tasks 2.4, 2.5..plz write it here briefly.
Also to make us help you efficiently please do include what you want to
do, what you have done so far, what is not working, what you have done
to make it work (debugs etc) and then we will be in a position to give
y
If you have changed anything in CTI ports at CUCM after creating them at
UCCX then this issue may occur.
Delete all CTI ports from CUCM, go into Route plan report and delete the
numbers assigned to them as well.
Do an IP telephony resync thru IPCC and I hope this may resolve the issue.
Ash>
On
I would agree with John here as Software MTP would not support G729.
This might be the issue with your case.
An MTP needs to be pre-allocated for early-offer calls, you must
configure an external MTP or transcoder device to use this feature. The
software MTP does not support G.729 over SIP trunk
Hello John,
You will be most welcome but you sure you are from UK?
Thanks
On 22/12/2010 16:32, Arun Kumar wrote:
John, indeed this is great forum. Shot your question here and welcome.
On Wed, Dec 22, 2010 at 9:16 PM, John Smith
mailto:john.smith.cc...@gmail.com>> wrote:
HI Team,
I
Guys,
I have setup an ephone hunt group in CME 7.x where these five extension
numbers if busy/NA forwards calls to another number which then forwards to a
company wide Hunt group (some 19 extensions in there) which has final number
as Voicemail Pilot 8000.
I was under the impression that if a cal
s well for FRF.12.
On Thu, Sep 30, 2010 at 8:32 PM, Ashar Siddiqui <mailto:siddas...@gmail.com>> wrote:
Help yourself .. it would then be worth being CCIE..
Ash>
voice-gang voice-gang wrote:
Lab in 5 days if anyone can help that would be appreciated
On
Help yourself .. it would then be worth being CCIE..
Ash>
voice-gang voice-gang wrote:
Lab in 5 days if anyone can help that would be appreciated
On Thu, Sep 30, 2010 at 4:12 PM, voice-gang voice-gang
mailto:mgcptroubleshoot...@gmail.com>>
wrote:
8.1 Switch QoS
2) On port fa 1/0/1
Well done! Congrats..
Ash>
CCIE#26244 (Voice)
groganhockey wrote:
26966!
I'm not sure how often it'll come up in everyday life, but there it is.
I took the lab Monday in RTP and finally got my score report last
night.
IPExpert/Vik/Amy: Thank you for the excellent study guides,
walk
Follow this process and you will never have a miss.
At CME router do the following:
Ping the tftp server and check connectivity
Check if there is a directory on flash like
Desktops/320x196x4/List.xml if not then make a directory
Create directory in flash “mkdir
flash:/Desktops/320x196x4
I am also waiting for this book to be released.
Not out yet!
Ash>
satpal yadav wrote:
hi this is satpal yadav can anybody provide me this book
Troubleshooting
Cisco IP Telephony 2nd Edition by Anne Smith .
___
For more information regarding i
Well done Matt - I feel really happy for you - I think your years of
learning, bootcamps and hard work made you nail it first attempt. Attempts
doesn't matter as long as you get that number in the end. Enjoy!
Ash
CCIE#26244 (Voice)
-Original Message-
From: ccie_voice-boun...@onlinestudyli
Count me in too.
I am also trying to find a way to upload files from CLI but still could not
find any command for that.
Anyone out there who knows if it’s possible at all? I think it’s not possible.
Ash>
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinest
CME 7.0 won't display '+' sign at the top with incoming calling number.
If you have properly applied translation rules then you will see '+' at
the bottom of the phone screen.
Ash>
CCIE Voice GMAIL wrote:
Hi
everyone,
I’m
trying prepend a ‘+’ at the beginning of
the call
What is your gatekeeper config?
What prefix you matching at GK?
Ash>
-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Matthew Berry
Sent: 11 August 2010 05:21
To: CCIE Voice OSL
Subject: [OSL | CCIE_Voice] Calling
ylist.com
When replying, please edit your Subject line so it is more specific
than "Re: Contents of CCIE_Voice digest..."
Today's Topics:
1. Re: MGCP Debug Packets (Ashar Siddiqui)
2. RTMT for Dummies (ccielabrat)
3. Re: RTMT for Dummies (Miron Kobelski)
4. Re: RTMT for
All explained here:
http://ccieash.wordpress.com/2010/06/22/mgcp-call-preservation/
Miron Kobelski wrote:
Hi,
AFAIR, you can see this better with "debug ccm-manager backhaul".
regards
kobel
On Fri, Aug 6, 2010 at 7:43 PM, Edwin Dotson
wrote:
I
I am glad that the solution proposed by Cisco is exactly what I did months
back after trying different solutions.
Ash.
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of CCIE Voice
GMAIL
Sent: 06 August 2010 03:13
To: ccie_voice@onlines
Hello,
I have come across a Cisco Attendant console issue where call comes in to
Pilot point 8800 and then it is broadcast to three different sites.
60% of times their calls are hung.
I have reset CTI manager, Cisco attendant console service and CCM service
but the issue is still there.
Furt
There is a Bug in CME 7.0 which will not show + at the Top.
You can only see at the bottom.
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of MARSHALL, JODY
C (ATTBCS)
Sent: 04 August 2010 12:55
To: ccie_voice@onlinestudylist.com
Subject:
ort, sometimes it was blocked and other times
it wasn't. This is why it was not affecting everyone.
HTH
-Adam
On Fri, Jul 30, 2010 at 11:40 AM, Ashar
Siddiqui <siddas...@gmail.com>
wrote:
Has anyone come across this?
One of my customer is having few 7941s
Has anyone come across this?
One of my customer is having few 7941s which is just showing two ringtones
Chirp 1 and Chirps 2.
A message on the phone says 'ring list unavailable'
Could it be a firmware issue?
CUCM 6.1.2.1000-13
Some details from the phone:
Host Name SEP00215554
Pho
full
report in my opinion.
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ashar Siddiqui
Sent: Thursday, July 29, 2010 11:25 AM
To: Ohamien Uhakheme
Cc: OSL Group
Subject: Re: [OSL | CCIE_Voice] First attempt
I am sure you will figure out wh
First give it a try, if it works then I will tell you the logic.
Ash>
Erwan Erwan wrote:
ok wll try incoming, called 9 in my sesson
i use sec dial tone 9 , but no effect.
so what is the logic by using above solution , As
I am sure you will figure out what mistakes you made which resulted in
0%.
I know its very hard to find out when you are sure your solution is
100% but believe me I have been through this and you will come to know
how a tiny mistake in that particular section or may be in some other
section res
Try this in 2nd dial peer:
dial-peer voice 9 pots
incoming called-number 9
destination-pattern 9[2-9]..$
port 0/0/0:23
forward-digits 7
also are you using under telephony-service 'secondary-dialtone 9'
Ash>
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-bo
7936 being at an older
firmware
version.
-Monica
From:
ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ashar
Siddiqui
Sent: Wednesday, July 28, 2010 5:19 AM
To: 'CCIE Voice'; 'Shady Hasan'
Cc: ccie_voice
more thing, other 7936 phones hooked into the same port works fine.
I think if somehow I can upgrade the firmware, I may resolve the issue.
Thanks
Ash
From: CCIE Voice [mailto:cc...@corb.net]
Sent: 28 July 2010 13:04
To: Shady Hasan
Cc: Ashar Siddiqui; ccie_voice@onlinestudylist.com
Subject
Hello,
I am trying to make one 7936 conference phone for a customer to register
with call manager.
No matter what I do, it is not getting registered. It says ' TFTP server not
found'.
I can ping the IP address of the phone from Call manager and from phone I
can ping rest of the network. All
:09, Berry, Matthew J. wrote:
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..
Akbar Sahab,
I cannot see your calling number over the ISDN?
Cause i = 0x829F - Normal, unspecified
Telco of called party end is rejecting the call.
Ash>
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Akbar Ali
Sent: 22 J
I will leave that to you to play with.
If you do get PSTN caller to join "meetme" conference thru Join Key then do
let us know J .. It would be worth sharing..
Happy Labbing...
Ash>
From: jeremy co [mailto:jeremy.coo...@gmail.com]
Sent: 22 July 2010 09:50
To: Asha
onference parties only
dial the conference number to join the conference. This soft key must be
configured before you can initiate meet-me conferences.
Join key can be used in normal conferencing.
Ash>
From: jeremy co [mailto:jeremy.coo...@gmail.com]
Sent: 22 July 2010 09:19
To: Ashar
I don't think it would work that way.
All participants must come through by dialling meetme number to join a
meetme conf (this is what I think) otherwise it won't work.
Ash>
From: jeremy co [mailto:jeremy.coo...@gmail.com]
Sent: 22 July 2010 08:51
To: Ashar Siddiqui
C
Sorry didn't understand this bit "Ph1 create meetme conference and ph2 and
ph3 joins to conference. PSTN call 4013 and ph1 picks up the call".
PSTN should call DDI of meet me number like XX4200 and I believe it will
go directly in conference. Why PSTN is dialling 4013??
Ash>
From: c
Hello all,
I can see you guys have referred to my blog regarding this issue and I
can assure you it worked for me quite well with that config.
Mark - Please check when in SRST, your Conference bridge has registered
to CME IP (192.168.1.254)...if not then enter a null route to Pub and
Subscrib
2 g711alaw
codec pref 3 g711ulaw
Thanks
Ash>
http://ccieash.wordpress.com
Shady Hasan wrote:
If you r using NM card to connect to the ISDN:
under the voice-card configuration, try this command "local-bypass"
which allow the hairpinning the call without using DSP.
Regards,
Shad
so it is more specific
>than "Re: Contents of CCIE_Voice digest..."
>
>
>Today's Topics:
>
> 1. Calls transferred to mobile phone drops (Ashar Siddiqui)
> 2. Re: Calls transferred to mobile phone drops (Shady Hasan)
>
>
>
Let me give it a go and I will come back to you.
Thanks.
From: Shady Hasan [mailto:shady@gmail.com]
Sent: 20 July 2010 12:39
To: Ashar Siddiqui
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Calls transferred to mobile phone drops
If you r using NM card to connect
Hi all,
I am working on a issue for a customer where he cannot transfer external
calls to a mobile phone.
His CSS has all privileges to make calls to mobile numbers and he can make
calls to mobile number fine but when he transfers a call it gets dropped.
An external call comes in to 214
One of my customer have come across an issue with
the UCCX servers. They have two servers, 10.10.100.64 and 10.10.100.65.
The .64 is Master and .65 is Stand by.
Last week, one of their IT guy added some work flows and Wrap up codes.
He also assigned some agents to proper teams from Cisco Deskto
I was just thinking why would you do that? I mean if you are in a region
where its E1 you will have both as E1 and if you are in US/NA then it would
be T1.
Why using both?
I don't know if it's possible.
Ash>
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@online
Upload all of them on CUCM TFTP then go to router and run these commands:
Router # copy tftp://ccm-ip-address//List.xml flash:
Do the same for image files.
Remember to keep the same path for files under flash which you have
mentioned in List.xml.
Ash>
-Original Message-
From: ccie_vo
Hi Matt,
The IP phone to IP phone issue is proctor lab issue.
You must hear music if you put PSTN phone on hold and also you should see
the output from sh ccm-manager mus
As I am at work, will not be able to explain further but I have explained
all possible MOH issues on my blog. Go into Media
I was thinking the same Marty, infact I went back to customer and told
them that this is not possible and you will get your Main Site # as
calling party number.
Thanks
Ash
Marty Beutler wrote:
Hi
Ash,
That is normal behavior for CFWall. Mobile Connect (Single
Number Reach) will al
Hi all,
Cfwdall on the IP phone to mobile is not showing the external calling party
number.
When an external PSTN caller calls customer, he can see calling party number
fine on his IP phone but if he CFWdall his IP phone to mobile phone then he
gets their office main number as calling numbe
Did you not delete the Subscriber from CUCM group and re-added it again
before the rebuilt?
Ash>
Duncan Hamilton-Walker wrote:
Hi Matt,
Yes i’m
using my own lab..
thinking about it..the SUB has been rebuilt, due to an issue with the
DB.. So
im thinking that the PUB think
It will not.
You are not configuring ephones, you are just configuring privacy thing.
Ash>
From: kobel [mailto:findko...@gmail.com]
Sent: 02 July 2010 16:00
To: Ashar Siddiqui
Cc: sean hurricane; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] SRST
hi,
doesn
Sean,
Do srst auto-prov none and then just create ephones (as many as required)
and put the following in there:
Ephone 1
No privacy
!
Ephone 2
No privacy
!
You will need to do all the basic requirements for Cbarge like conference
hardware, sdspfarm units etc and configuring dspfar
Mgcp / no mgcp is a very important command when dealing with MGCP gateways!
If not using trombone then always do no mgcp/mgcp for any changes you make
at gateway or call manager to come into effect.
Ash>
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylis
There is no drawback in first option but I would like to be as specific as
possible.
The 2nd rule should be like this:
voice translation-rule 1
rule 1 /^3214\(3...\)$/ /\1/
This way only calls for your DDI range will be entertained by the gateway
and rest will be rejected.
At
Hi all,
I came across an interesting situation today for a customer and thought to
take input from you guys.
Customer has several sites which connects to Central CUCM cluster. Each site
has SRST enabled.
During Out of hours the main line is forwarded to a Hunt group which has
three interna
Exactly what Roger said below...I will never change Fragment Size.
Ash>
Roger Källberg wrote:
You
shouldn't change the fragment size. Reason being that you want the
fragment to be of a size that would give you a 10ms transmit delay in
the event of congestion.
Brgds,
des
Regards,
Joaquim Fernandes
--- On Sat, 6/26/10, Ashar Siddiqui
wrote:
From: Ashar Siddiqui
Subject: Re: [OSL | CCIE_Voice] File Transfer from CUCM to CUCME
To: "Joaquim Fernandes" ,
"ccie_voice@onlinestudylist.com"
Date: Saturday,
Correct behaviour!
Ash>
Unified Communications wrote:
Hi Experts,
I configured SNR in my lab. But when I answer the call from the PSTN
phone, I don't see "In use Remote" on the IP phone.
But when I select the line which shows red, by pressing the line
button, I do see the "In Use Remot
Load the .png files at CUCM - OS Admin > TFTP.
At CME Router >
# copy tftp://cucm-ip-address/whateverthefile.png flash:
Ash>
Joaquim Fernandes wrote:
HI Team,
Just needed to know how to download .png files from CUCM to CUCME
without using a
?
Bo
On Sat, Jun 26, 2010 at 5:01 AM, Ashar
Siddiqui <siddas...@gmail.com>
wrote:
You can Strip off '+' from a calling number when in SRST by using
translation rules but any such rule will also affect your normal
operation (not in SRST).
I would advise if y
You can Strip off '+' from a calling number when in SRST by using
translation rules but any such rule will also affect your normal
operation (not in SRST).
I would advise if you want to take off '+' when in SRST, configure
telephony service as srst mode auto prov all and then edit the
ephone-
There you go! You now know the issue.
Share with us when you resolve it.
Ash>
naoufal kerboute wrote:
Hi Ashar,
I need to play in case to forget
the stress of the lab, it's coming soon.
I think you are right the problem
on the UCM side, I will try to restart the Med
First of all, why are you playing? You should be
labbing properly as voice field is not a playground..
Joke aside... ;)
What is your "‘show
ccm-manager music-on-hold" ?
If no MoH
streams are shown by this command then
CCM has failed to provide the gateway with MoH
Also keep in mind Tone on
few checks :
- Codec under voicemail Voip dialpeer should be g711ulaw
- add incoming-called number at CUE Voip dialpeer (your MWI range)
- Remove voice class codec 1 - if you are using it under a default
incoming voip dialpeer
- Make sure you edit MWI application at CUE with your MWI ON and MW
Hello all,
I am having an issue while accessing GUI page of CUCM for one of the
customer.
CUCM version is 6.1.1.3101-1.
When I enter username and password, the call manager accepts the credentials
and then sits in "Loading please wait" state for indefinite time.
Searched everywhere but co
Discussed so many times..
do this..
voice service voip
allow-connections h323 to h323
h323
emptycapability
h225 id-passthru
h245 passthru tcsnonstd-passthru
on trunk in gatekeeper make sure to uncheck "wait for far end h.245
terminal capability set"
Ash>
CCIE#26244
naoufal.kerbo
Title: RE : [OSL | CCIE_Voice] RE : GK CUBE behaviour
Whats in the debugs?
deb cch323 h225
Ash>
naoufal.kerboute wrote:
Hi Ashar,
It's included in my config but not resolved the problem.
Message d'origine----
De: Ashar Siddiqui [
REG_MOH must have G711 with all other regions. That region will go in
DP_MOH.
You are missing ip source address command under fallback.
Also, like I said before, do a sh eph summ and let us know what you are
getting.
Ash>
Mark wrote:
Thanks guys, in this case I am not using CME. BR2 is p
On the gateway make sure you have ccm-manager music-on-hold.
Under telephony-service, there should be max-ephone and max-dn.
Also do show ephone summary and check if MOH has been enabled.
If all this doesn't work, send us your gateway configs.
Ash>
Mark wrote:
If PUB is configured for mul
I would put all my internal numbers in NONE partition while patterns for HQ
and BR1 under PT-US while BR2 into PT-BR2. CSS for HQ and CSS for BR1 will
have PT-US while CSS-BR2 will only have PT-BR2. You will never come across
Interdigit timeout. Also, I would recommend changing the t302 to 5 sec.
Voice translation-rule 1
rule 1 // /+\0/
!
Voice translation-profile ANI-PLUS
translate calling 1
Then under International dialpeer call this translation profile
~
translation-profile outgoing ANI-PLUS
Ash>
Bo Gao wrote:
Hi everyone,
How do you configure voice translation rule in h3
ion memory.
System fpga version is 250027
System readonly fpga version is 250025
Option for system fpga is 'system'.
62720K bytes of ATA Slot0 CompactFlash (Read/Write)
From: Kumar Vishal [mailto:kvis...@uebiz.com]
Sent: 21 June 2010 15:26
To: Ashar Siddiqui
Subject: RE: [OSL
Hello all,
I have a customer with an analogue line which does not appear to be sending
DTMF tones correctly. This is causing issues when users attempt to log into
BT meetme conferences as they are unable to enter partiticpant passcodes /
chairperson passcodes etc. When they do, it does not reco
rty transformation mask
PreDot
972 (city code)
National, Isdn
try it and let me know if u have
any question
On Sun, Jun 20, 2010 at 6:49 AM, Ashar
Siddiqui <siddas...@gmail.com>
wrote:
Did you also try what I suggested? masking Called party at RL det
on my lab which seriously I don't want to do for a week at least
lol.
Ash>
Ashar Siddiqui wrote:
Sorry Ignore my last post, I thought you are asking about Calling party
number (ANI).
The one Angel mentioned is a possible solution or try this one...make
one route pattern, Create two R
Did you also try what I suggested? masking Called party at RL detail
level!
cisco voip wrote:
I tried this just now. and it is not working,
So what i was thinking is correct, it can match only one route pattern
and call cannot come back.
Is there any other way anyone would think of?
Try this:
This will block international calls for all users.
dial-peer cor custom
name International
name block-int
!
!
dial-peer cor list InternationalCalls
member International
!
dial-peer voice 9011 pots
corlist outgoing InternationalCalls
translation-profile outgoing int-pstn
desti
Sorry Ignore my last post, I thought you are asking about Calling party
number (ANI).
The one Angel mentioned is a possible solution or try this one...make
one route pattern, Create two RG in the RL, then place mask under
Called party like XXX and XX under Route list detail level.
I
Do it at the Route list detail level. Turn on the mask and set the mask
accordingly for Calling party number. Do not do anything at the Route
Pattern.
Ash>
cisco voip wrote:
Hi Experts,
If I have two MGCP gateways BR1 and BR2, and call should go thru BR1
and if it fails it should go thru B
I would go ahead and set native vlan to Data vlan if it's not specified.
Ash>
Angel Perez wrote:
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
You just took my words Moataz...without a doubt this month has proved
"CCIE Voice month " of the Year unless the record gets broken in coming
months.
If I am not wrong, we already have "eight" CCIEs so far in this month
and the month has still 10 days to go ;)
Ash>
Moataz Mamdouh wrote:
Congrats Salman! looks like Cisco is keeping track of our lucky numbers
as well ;)
Have a party :)
Ash>
Wayne Lawson wrote:
Congratulations!
Regards,
Wayne A. Lawson II - CCIE #5244 (R&S)
Founder, President & CEO - IPexpert, Inc., Proctor Labs,
Inc. & Platinum Solutions Grou
back in December 2009 but now I have decided to stick with it as I
won't find such a nice bunch of people anywhere..
N.B: Above all, I loved my number..Digit '4' is my lucky number and
Cisco made sure that I have enough of them.. :)
Thank you all. It's party time now ;)
Ashar
Install FileZilla on UCCX box if there is no FTP server already there.
Your error suggests there is no FTP server running on the box.
Ash>
Roger Källberg wrote:
What
config have you loaded? The ftp server on the UCCX aren't setup in all
labs, use UCCX config from volume 2 or 1 wee
MALMÖ
Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se
Från: Ashar
Siddiqui [siddas...@gmail.com]
Skickat: den 14 juni 2010 11:38
Till: Roger Källberg
Kopia: kobel; wolfsrudel; ccie_voice@onlinestudylist.com
Ämne: Re: [OSL | CCIE_Voice] Spanning-tree portfast
part of the de facto port config configuracion, unless were have
specific reasons not to do so. imho
hth
On 6/13/10, Ashar Siddiqui <siddas...@gmail.com> wrote:
> Hi,
>
> In Proctor lab HW-Switch I can see this command:
>
> interface FastEthernet1/0/2
> switch
Don't forget to add max-ephone and max-dn under call-manager-fallback.
Ash>
ccie rs wrote:
ccm-manager music-on-hold
call-manager-fallback
moh music-on-hold.au
multicast moh 239.1.1.1 port 16384 route 10.10.201.1 10.10.110.2
This is a H323 GW on BR1 registered to CUCM
efore if it is asked to
define port fast then we do not trunk the port. It is assumed that when
asked for portfast, Switch will not be an XL or EtherSwitch
module.
Ashar Siddiqui wrote:
Yes I know they are configured as Trunk ports but the phone port has
nothing to do with Route Bridge Ele
).
it's part of the de facto port config configuracion, unless were have
specific reasons not to do so. imho
hth
On 6/13/10, Ashar Siddiqui <siddas...@gmail.com> wrote:
> Hi,
>
> In Proctor lab HW-Switch I can see this command:
>
> interface FastEthernet1/0
Hi,
In Proctor lab HW-Switch I can see this command:
interface FastEthernet1/0/2
switchport access vlan 10
switchport mode access
switchport voice vlan 20
spanning-tree portfast
But "Spanning-tree portfast" is not used on BR1/BR2 ports where phones
are connected. Any specific reason? I
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