You will probably need to make a xml file with the services needed, in the
order asked...
-Chase Mergenthal
--
If winners never quit and quitters never win, then who coined the phrase, "Quit
whi
Hi,
Have cleared my attempt and just want to sell the license so that i can
recover my money which i have paid.
Let me know if anyone is interested in the same so that i can ask for
transfer of my license.
I paid it from CK here is the link
http://www.certknowledge.com/ipb/topic/24-ccievoicelab
You need to go the application user rm and associate the agent phones to this
user.
--ms
Michael Sears
CCIE (V) 38404
Message: 2
Date: Wed, 06 Mar 2013 22:10:21 +0530
From: "singh"
To: ccie_voice-requ...@onlinestudylist.com;
ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] On
I have to ask. Am I the only one that thinks the requirement of "There can be 4
concurrent calls. G711 CODEC to be used for multi-directional audio." is odd
when an earlier requirement states you should use G729 between sites?
--
William Bell
blog: http://ucguerrilla.com
twitter: @ucguerrilla
Maybe, device association for the RmCm subsystem user id? That user id in
CUCM should be associated with the agent phones.
On Wed, Mar 6, 2013 at 11:40 AM, singh wrote:
>
> I have one button login set for my uccx agents and have verified that
>
> - the agent id and password for the users
>
> -
Hello,i am trying to remove a directory services from a phone execpt Voicemail.
i follow the technic provided by IPexpert work book the problem i am having the
directories are not line up the way i wanted like first Missed Call, Received
Call etc. Can someone shows me how to do that ? i will ap
Not sure it makes any difference in this situation, but I never use codec
pass-through on my configuration. I've never had any issues.
On Wed, Mar 6, 2013 at 12:32 PM, wrote:
> --MJ
>
> Your problem is a misconfigured location somewhere in CUCM.
>
> Your configuration on the gateways is correc
In your SCCP gatewway config are you using SCCP version 5+ or later? must
be 5 or later for RSVP to work.
i do not believe you will want to use the pass-through codec either.
can you verify those wan interfaces are correct> maybe you have a
subinterface and are specifying the RSVP bandwidth on the
--MJ
Your problem is a misconfigured location somewhere in CUCM.
Your configuration on the gateways is correct to allow 4 calls using RSVP based
CAC. In my experience the issue your running into is not going to be an issue
with the configuration on your gateways (use show SCCP on gateways to v
I have one button login set for my uccx agents and have verified that the agent
id and password for the users association of rmuser with the phones resource
group contains the agentshowever I am seeing the following error on one button
login..."Unable to log you in due to conf error ( phone is n
I am a little confused by the question. The way I read it you have:
DEFAULT Intra-site codec should be G711
DEFAULT inter-site codec should be G729
For HQ to Branch_X use G711 codec and RSVP
So, in other words the question gives you permission to not use G729 for calls
between HQ and Branch_X. A
hi Guys,
I have to Configure IP Phones and gateways in such as way that all calls
within same site
should use G711 Codec. Also, all calls between the sites to remote IP
phones and
gateways should use G729 Codec.
RSVP Call Admission Control (CAC) between HQ and branch site based on
bandwidth limita
To make VM work in SRST one method is to mask the calling number, i.e., on the
hunt pilot to . At times this method may conflict with other requirements.
You can also utilize alternate extensions in UC if the question does not say
you can't use alternate extensions to accomplish this.
Ho
VM in SRST (CISCO CCIE VOICE:
There are two ways of doing this ...
1. Alternate Extension
2. Calling Number transformation.
For alternation Extension ..
1. issue debug ISDN q931 on HQ router
2. Press the voice mail button on the SB phone
3. Check the debug output on HQ router for the Calling nu
Hi
William has right !
Regards
Chrysostomos
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of William Bell
Sent: Τετάρτη, 6 Μαρτίου 2013 2:50 μμ
To: Pixar Perfect
Cc: CCIE Voice OSL
Subject: Re: [OSL | CCIE_Voice] Phone NTP Reference Vs N
I had the same impression as Bill. It would be very interesting if that was
the case.
Sent from my iPhone
On Mar 6, 2013, at 7:49 AM, William Bell wrote:
> Pixar,
>
> Are you certain about the Phone NTP reference and CUPC? I have not heard that
> before. I was under the impression that CU
Pixar,
Are you certain about the Phone NTP reference and CUPC? I have not heard that
before. I was under the impression that CUPC would use the clock of the
underlying OS.
-Bill
--
William Bell
blog: http://ucguerrilla.com
twitter: @ucguerrilla
On Mar 6, 2013, at 12:15 AM, Pixar Perfect wrot
HI All
When the Branch-1 is in SRST Proper Mode (call-manager-fallback) When i am
trying t press voice mail button its playing system greeting not the user
greeting ,its there any thing i need do on CUCM in order to play user
greeting when the phones are in SRST..
Thanks
___
Phone NTP reference is used for phones that uses SIP protocol to get there
time ...
On Tue, Mar 5, 2013 at 1:15 PM, wrote:
> Send CCIE_Voice mailing list submissions to
> ccie_voice@onlinestudylist.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://onlin
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