You should be fine.
But if you feel you must touch the gear - find a local Cisco Office and see
if they don't have the gear you need to work with.
On Thu, Apr 23, 2009 at 11:19 PM, junk cisco wrote:
> hi,
>
> I need your advice before taking the lab in US
>
> I have studied 8 months with Re
hi,
I need your advice before taking the lab in US
I have studied 8 months with Remote lab from Internetworkexpert , do i need
to rent real lab (physical lab) before sitting the lab exam ? I am
planning to take the lab in May.
Need suggestion please? If yes, where can rent for good price ?
Norma:
Would you like to share with the group what the issue was. I'm sure that you
are not the first or last to have this happen :-)
Thanks
Larry Hadrava
CCIE #12203 CCNP CCNA
Sr. Support Engineer – IPexpert, Inc.
URL: http://www.IPexpert.com
On Thu, Apr 23, 2009 at 7:03 PM, Norma Exel wrote:
nevermind. figured it out. i apologize as this is too far out of the
scope of the lab study. thanks anyway. -N
- Original Message -
From: "Norma Exel"
To: "Norma Exel" , ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] SIP Trunk Issue
Date: Wed, 1 Apr 2009 23:30
Cliff, You are right!
If I press hold during a call to BR1 two sessions of hw-xcoder at HQ is
allocated.
But, The MTP checkbox is UNchecked. I do not understanding how it are beeing
requested.
I just have one trunk, so there is no option to be other trunk. I restarted the
services and all th
I don't see how that could work without an MTP. Suggest you reset CCM services
and retry. I think you have it right, but you need the MTP for supplementary
services.
- Original Message -
From: Sergio Polizer
To: ccie_voice@onlinestudylist.com ; cl...@mcglamry.net
Sent: Thurs
Two match dial-peers is right, we 1st one will be used if its status is up and
remote end able to receive it, otherwise 2nd one kick in.
After dial-peer matched, codec/dtmf-relay come in.
James
> Message: 1
> Date: Thu, 23 Apr 2009 10:24:22 -0700
> From: Nara Shikamaru
> Subject: Re: [OSL |
I found one solution:
For G729 calls between CME to HQ and Br1 phones through a GK WITH supplementary
services, I used:
- H225 GK Controlled (MTP not checked)
- With DP Br2 (G729 with all other regions) and with HW xcoder at HQ DP.
From: spoli...@hotmail.com
To: cl...@mcglamry.net; ccie_voic
Cliff, Thanks for your input. See the answers bellow.
If you're getting a fast busy on the transfer, you
probably have an issue with your Region set up. Things to look
at:
A) No, It is when I answer the call at BR1 phone.
What codec is defined between the GKTrunk and the BR1
region? If i
If you're getting a fast busy on the transfer, you probably have an issue with
your Region set up. Things to look at:
What codec is defined between the GKTrunk and the BR1 region? If it's defined
as G711, but you have limited bandwidth, the call will fail.
Do you have a hardware transcoder
Why would you want to install CUC on a DL 380? You can install all of the
platforms on VMware (Intel only right now, not AMD yet.) I have CUCM, CRS,
CUC, and CUPS on a dell desktop with 4 gb ram.
On Tue, Apr 21, 2009 at 11:28 PM, Jonathan Charles wrote:
> They are cheap on Ebay...
>
> I have a
HI,
I have a call from BR2 to HQ/BR1 in a ICT_GK with MTP checked to provide
supplementary services like hold/transfer.
My ICT have a BR2 DP that speak G729 with all others and a HW transcoding that
have HQ DP.
When I call from CME to HQ the call goes to HQ transcoder at G729 and connect a
H
>From the trace, it would appear that the most likely issue is that the host
>you have defined in the MX record doesn't exist.
Within an MX record (in Microsoft DNS), there is a spot where you put in the
host name. This should be the fully qualified domain name of the host, and
MUST be DNS r
It's set to 4 digits on the trunk. That's not the problem here.
On Thu, Apr 23, 2009 at 10:14 AM, Rick Grimes wrote:
> Change your destination-pattern to 4 digits or make sure that significant
> digits is set to 4 in your sip trunk.
>
>
>
>
>
>
>
> *From:* ccie_voice-boun...@onlinestudylist.com
Change your destination-pattern to 4 digits or make sure that significant
digits is set to 4 in your sip trunk.
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Nara Shikamaru
Sent: Thursday, April 23, 2009 11:26 AM
To: Jiahong - tobecci
Hi,
I configured vpim, but seems cue cannot connect to unity
com.cisco.aesop.smtp.SmtpService : findLocation: ret 1
5410 01/20 21:36:29.167 netw dbug 1 com.cisco.aesop.smtp.SmtpService :
findLocation: ret 1
5410 01/20 21:36:29.168 netw dbug 1 com.cisco.aesop.smtp.SmtpService :
remotesLocations si
James,
If there are two dial peers that are identical except for the
preference setting, is it normal for the debug to reflect matching on both?
I'm wondering if the problem is occuring before the dial-peer has an
oppurtunity to do anything with the call. Codec and dtmf-relay settings are
rel
BlankI have been googling this for sometime now, but no luck yet. I know that
there is a digit manipulation file in UNITY server just like the num-exp in
CME. The only difference is that, it is an editable flat file residing in one
of the directories in UNITY server.
For some reasons I cannot
OK you match the outbound dial-peer properly. You may have problem with
dtmf-relay or codec negotiation.
Copy/paste your 'voice class codec 1' config. Also add 'dtmf-relay sip-notify
rtp-nte' in both dial-peers
James F.
> Message: 1
> Date: Wed, 22 Apr 2009 22:48:04 -0700
> From: Nara Shikam
No, there isn't.
This is where VPIM and the exchange voice connector comes in. There should be
a user in Unity defined with the primary extension of 1001. But no user named
1001.
Cliff
- Original Message -
From: Scott ODonnell
To: OSL Group
Sent: Thursday, April 23, 2009 9
Hi Vik,
are there a reason to not allow a phone remote control(voip integrations, free
on internet).?, that could be great.
you can try by your own. on my ccm work fine. do you know if is a port
restriccion or something like that?. can you suggest it to ip-expert?. with a
phone remote co
Just reading through some threads on VPIM problems.I wanted to confirm
something.
If I create a subscriber on UNITY , called H Phone1 with extension 1001, the
underlying domain account is HPhone1 with a smtp address of
hpho...@voip.lab(or whatever domain is configured).
I just want to make sure t
22 matches
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