Is it must to give "codec G711ulaw" under "voice register pool 1"
I believe it Q dependent and also If I have G711ulaw under voice reg pool in
that case if I am calling say via GK from HQ into CME the transcoders will
be needed, but is there anythign else I am missing whcih will break if I
don't
Thx Roger, but is there in any debug or show command to verify the same, I
believe not, right ?
On Fri, Apr 30, 2010 at 3:29 PM, Roger Henderson wrote:
> This is overkill but... you could put an ACL on your relevant interfaces
> that looks something like: access-list 101 permit udp ho 1.1.1.1 any
The standard rule of a CCIE lab is that if a question doesn't specify
something, it doesn't matter (and hence we shouldn't care).
Roger
On Sat, May 1, 2010 at 10:17 AM, Sergio Polizer wrote:
> Hi,
>
> While doing the Vol 2 workbooks I have observed that when site Br1 (ISDN
> Pri/primary-ni) is
Hi,
While doing the Vol 2 workbooks I have observed that when site Br1 (ISDN
Pri/primary-ni) is configured as H323 the IOS set automaticlly the calling type.
If I change the switch-type to primary-net5 it does not happens e.g for
Br2. See bellow:
Primary-NI
Calling Party Number i = 0x2
This is overkill but... you could put an ACL on your relevant interfaces
that looks something like: access-list 101 permit udp ho 1.1.1.1 any eq 123
If 1.1.1.1 is your loopback you should see packet matches on that entry.
HTH
Roger
On Sat, May 1, 2010 at 4:40 AM, vccie2010 wrote:
> Angel, thx
I don't think a class codec is required as long as you have the codec g711
configured on the voip dial-peer
by default a voip dial-peer supports g.729 by default and CUE only supports
g.711 so we either need to force that call to be g711 somehow either using a
class codec command or just the codec
Hello all
Do you know if you can configure some kind of Multimedia queue? what i mean
is, when i have a caller in the queue and if he/se doesn't want to wait any
longer, they can press 1 to leave a voicemail(recorded message), instead of
sending that voicemail to a mailbox. i want that voicemail
Already done and calls to and from sip work fine. The problem as I see it, is
the dial-peer 222 acts as both incoming and outgoing dial-peer and IS hardcoded
for g711u. Therefore a transcoder will never be invoked#!! As both inbound
and outbound dial-peer is g711u so no codec mismatch no tran
Jeff, also *check *MTP Reguired box
On Fri, Apr 30, 2010 at 12:49 PM, vccie2010 wrote:
> ON your GK trunk - Uncheck H245 box
>
> On Fri, Apr 30, 2010 at 12:20 PM, Jeff Cotter wrote:
>
>>
>>
>> Station to Station calls work as expected between UCM and CME including
>> g729 call through gatekee
Vik, If we define "cuc7-pub" as domain name will it work , since CUE does
not have DNS server configured ? Per the IPX solution I am using "Unity IP
address" for the domain for unity location on cue.
On Thu, Apr 29, 2010 at 9:48 PM, Vik Malhi wrote:
> So the domain name defined in the unity
ON your GK trunk - Uncheck H245 box
On Fri, Apr 30, 2010 at 12:20 PM, Jeff Cotter wrote:
>
>
> Station to Station calls work as expected between UCM and CME including
> g729 call through gatekeeper to CME sip phone using g711. Transcoder is
> invoked properly and confirmed with show sccp connec
Station to Station calls work as expected between UCM and CME including g729
call through gatekeeper to CME sip phone using g711. Transcoder is invoked
properly and confirmed with show sccp connections. Call Routing and Transcoder
proven as functional with above.
However call fails immediate
Angel, thx for your response, what if Question specifically asks it , I am
not able to see it thouh in the show or debug ntp commands ?
Is thera away to verify it pls
On Fri, Apr 30, 2010 at 12:17 AM, Angel Perez wrote:
> Hi, is better to don't specify a source interface, "ntp source lo0" this
Hi all,
I was just wondering what is the importance of creating Voice class
codec on a H323 gateway?
I use to create this class like:
voice class codec
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
and then I would add it under default incoming voip dialpee
When I uploaded the updated vol2 lab 2, the PG was also uploaded I think
the solution for Vol 2 Q8.3 is correct having been over it again.
--
Vik Malhi CCIE #13890
Instructor - IPexpert, Inc.
Mailto: vma...@ipexpert.com
Telephone: +1.810.326.1444
Fax: +1.810.454.0130
Live Assistance, Please vi
CUPC STATUS MENU IS GRAYED-OUT
This usually means that CUPC failed to connect to the Presence Engine. This
could be caused by:
* Digest credential or Incoming ACL was not configured
* Proxy domain was not configured properly
* Network issue
Matthew Berry, C
Hi Matthew,
I'm encounter a very similar problem as you, which the CUPC status not shown
anything. From the "Show server Health" from CUPC, the process halt at the
"Presence" status, symptom is it keep connecting & disconnecting.
I've configured the proxy under the proxy domain, and i can login t
DNS is needed.
Roger Källberg
Från: Angel Perez [gorr...@hotmail.com]
Skickat: den 30 april 2010 09:24
Till: vccie2...@gmail.com; Vik Malhi
Kopia: osl osl
Ämne: Re: [OSL | CCIE_Voice] Lab 2 8.3 VPIM
Hi all:
Correct me if i'm wrong, Does CUC need dns to proper wor
I have emailed support, but no replies so far.
Ahmad Azeem
Date: Thu, 29 Apr 2010 10:03:49 -0700
From: vccie2...@gmail.com
To: tanner.ez...@gmail.com
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] IPX - please help.Trying to donwload my
config files and getting err
Hi all:
Correct me if i'm wrong, Does CUC need dns to proper working of VPIM? I know
that in CUE you can especify location domain as ip address or name, but is
this possible with CUC, I think it is not, maybe it's possible to add a static
binding for its name in the router... but not sure
Hi, vlan interface ip would be ok...
Date: Thu, 29 Apr 2010 17:54:43 -0700
From: gkr2...@yahoo.com
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] H323 Binding
Guys,
I'm building my own lab and not sure which ip address should I use
to add the BR1 router as S
Hi, is better to don't specify a source interface, "ntp source lo0" this way
all local interfaces are source of ntp, imagine that you have a routing issue
and you can't ping that interface... or that this interface is down, you would
be able to select another interface as source
hth
Da
I can confirm it too, what you are doing is just create and intercom betwen two
phones, the only thing that you have to remember is to add the speed dial
funcionality to the intercom numbers, in case of IPMA this is not required
becouse IPMA service do it for you when you add intercom lines to
Just tested this and yes it does work either way. The only thing I found of
interest here is that for the labels to change after altering the intercom
settings on the manager/assistant screens a restart of the IPMA service is
required.
Graham Hopkins
On 29 Apr 2010, at 20:27, vccie2010 wro
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