I'm sorry, that pattern would be used in UCM. It appears that your
configuration is correct.
From: C-note
To: ccieid1ot ; ayman labib
Cc: OSL Group
Sent: Mon, October 11, 2010 7:58:39 PM
Subject: Re: [OSL | CCIE_Voice] Calling and Called info
voice transl
Only in the original matching portion. So if to match a + in the number :
voice translation-rule 70
rule 1 /\+\(.*$\)/ /\0/ type any international plan any isdn
If adding a +, you only need to put it at the beginning:
rule 1 /^9\(.*$\)/ /+\1/ type any international plan any isdn
If y
voice translation-rule 70
rule 1 /\(^3...\)/ /+1617303\1/
The + sign in +1617 needs to be preceded with \+ to treat it as the literal
character '+' and not the wild card character + as such:
voice translation-rule 70
rule 1 /\(^3...\)/ /\+1617303\1/
From: c
Hi All,
I did the debug voip ipipgw and H.323 - SIP Networking is Working but SIP to
H323 Networking is failing
SITE-C#
000943: Oct 11 23:11:30.852 CDT: //112/8032C21C0100/H323/setup_ind: Receive
bearer cap infoXRate 16, rateMult 0
000944: Oct 11 23:11:30.896 CDT:
//112/8032C21C0100/H323/cch32
Hi All,
I was doing Lab4 and everything works as expected but I was very curious to
understand why the SIP Phones registered to CME cannot call the SCCP Phone in
HQ? So I started with the below options:-
1.Change the Codec of the SIP Phone to G.729 and place a call from 5001 (HQ) -
This call
I c nothing wrong with it.
On Mon, Oct 11, 2010 at 3:42 PM, ayman labib wrote:
> Greetings.
>
> Any ideas about my question below. I need to be 100% sure this time. I
> booked for my second attempt in December, and I would like to be well
> prepared this time. My 1st attempt was back in June.
Greetings.
Any ideas about my question below. I need to be 100% sure this time. I booked
for my second attempt in December, and I would like to be well prepared this
time. My 1st attempt was back in June. Any input is greatly appreciated.
Thanks
From: ay
Hi,
Thats what was missing from my logic. You can't allocate bandwidth to the
priority queue. It empties the queue as long as there are packet enqueued. (I
don't know how many times I read that, but didn't absorb it). So if you apply
priority-queue out, then you over ride both the queue 1 shape
This will work too.
voice translation-rule 40
rule 1 // // type any international plan any isdn
voice translation-rule 70
rule 1 /\(^3...\)$/ /+1617303\1/
On Mon, Oct 11, 2010 at 7:57 AM, ayman labib wrote:
> Hello Experts.
>
> Just wondering, what would be the best way to insert calling
If you set the interface to 'priority-q out'; It's using 100% of the
bandwidth so in essence no sharing. Leave the priority-q out command.
On Mon, Oct 11, 2010 at 2:56 PM, Ohamien Uhakheme wrote:
> That's mostly right, but I think you are missing the priority queue.
>
> So taking your example (
I stumbled upon this Ebay listing. Not sure if it's a decent deal just
thought I would pass it along to the list.
http://cgi.ebay.co.uk/ws/eBayISAPI.dll?ViewItem&item=290483159188&ssPage
Name=STRK:MESELX:IT
-Damon
___
For more informatio
That's mostly right, but I think you are missing the priority queue.
So taking your example (with modifications):
srr-queue bandwidth share 0 40 30 30
srr-queue bandwidth shape 10 20 0 0
priority-queue out
This is the way it would appear in the config, and it works bottom up.
The priority-q
You can create your own. I normally use Gimp software to resize my images for
large and TN
Sent from my iPhone
On Oct 11, 2010, at 11:45 AM, Pithog Oil wrote:
> Can some one send a phone back-ground image to me.
>
> Pithog oil
>
> ___
> For more
Hello Prashant,
Well it WORKS !!!, I figured out that there are two ways to make the job done :
1- Either by creating Alternative extension for the assistant which is 5002 in
my case and create MWI Extension 1080 as well for the assistant user to perform
any updates for MWI to be send to th
I am trying to get LiveRecord to work on a CUE that is registered to a UCM
box, and having a little trouble doing so. I have tried a few different ways
of getting it to work, but nothing is proving to be successful. I have been
unable to find a document that properly describes a process, so I have
no with physical phones you need to get a Cisco POE Switch and A Cisco 8xx,
18xx, 26xxXM, or 28xx series router running 12.4.20 (or later) Advanced IP
Services IOS.
*Please Note*: 12.4(20) (or later) IS *Required*
then you will copy paste the configuration to the router and switch. and
you're goo
Can some one send a phone back-ground image to me.
Pithog oil
___
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Hi Amr,
Usually when I create a VM profile the mask on it is for the 4 digit
extension. Now in case of IPMA if the Manager DN (1080) does not have any
mailbox or we need to send the voicemail to the Assistant mailbox, we cannot
use the mask. We need to send the Assistant DN ie in your ca
So, with real phones, do the phones support VPN themselves? (with VPN
client in them?)
Or do we configure them in other way?
--
Dew Swen
CCVP, CCDP, CCNP
On 10.10.2010 22:09, Cristobal Priego wrote:
you can run multiple instances of softphones IP- BLue
however you can't test a few features
Hello,
I'm a bit confused about srr-queue bandwidth settings for QoS. I was hoping
someone could confirm my understanding below. I want to make sure I have this
straight.
srr-queue bandwidth concepts
input queues only support shared mode
output queues support both shared and shaped.
shaped mode
Last 2 weeks I took the IPX 2 weeks bootcamp in Columbus.
I want to thank Amy for being such a great instructor and /.*IPX\(Vik…\)$/
/\1/ for all the training content and well done arrangements! This is was a
great tech boost, plus great self readiness assessment for me and I believe
all the folk
Hello Experts.
Just wondering, what would be the best way to insert calling and called info at
remote sites. Assuming that SiteB is an SRST H323. Site C is a CME site.
/
Here's an example of what I'm using to insert the calling and called info. Can
anyone please validate and offer suggestio
I just turn off auto registration and manually define the phone with the
protocol I want to use then let it register. New image gets installed on phone.
Tim depends upon link speeds and how many you are doing at once but no more
than 10-15 minutes for my whole lab
On 11 Oct 2010, at 09:13, Pi
And I add a null route to CUCM subnet.
On Mon, Oct 11, 2010 at 08:00, Mark Holloway wrote:
> I always shut down the Serial interface of the Frame Relay WAN link.
>
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www
Hi Experts,
As per the requirement in this question for adding a customer on the queue for
60 seconds, This could be acheived from the "DelayedWhileQueue" parameter on
the icd.aef script ,but does the 60 includes the seconds that the nice lady is
talking and instructing me to hold on queue
You can also modify the redirecting number on the terminating router with a
"translate redirect-called" in a translation profile. One thing I've noticed is
that when the "for" number is manipulated the "by" number changes to unknown -
anyone know why this is ?
Graham
On 9 Oct 2010, at 17:
Is there a better way to get it done that i dont know about, what is the
estimated time for this conversion, i really want to know if its a good idea
to use the BAT SCCP to SIP migrate tool in the lab, can you give time
estimates in figures.
--- On Mon, 10/11/10, Tanner Ezell wrote:
From: T
Hi Prashant
Yes you are right , I have created MWI extension on hqph2 user and added a new
VM profile to the IPMA ext. and the Device Profile as well , when calling from
any phone to the manager directly it gives me that "hqph2 is not availabe" ,so
after leaving a message it suppose to be sen
Depends on a lot of factors. Phone upgrades aren't fast even on a LAN.
On Mon, Oct 11, 2010 at 12:41 AM, Pithog Oil wrote:
> The last time i tried to convert a SCCP phone to SIP on CUCM using BAT
> migrate tool, it took forever, please how long does this process take when
> you do it.
>
> Could
The last time i tried to convert a SCCP phone to SIP on CUCM using BAT migrate
tool, it took forever, please how long does this process take when you do it.
Could it be that i was doing something wrong.
Pithog oil
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