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Hi:
I am testing the B-ACD TCL in BR2, the connection between BR2 (CME) and the CCM
is trough a gatekeeper and when I called to the pilot number of the script is
not working. The local calls from the IP Phones registered to the CME are
working and the script is ok, the calls coming from the P
Jonathan:
What Mark said is, you don´t have to specify the D channel, the IOS of the
router does it by default, I mean if you use the command pri-group timeslots
1-8 for example you will see the config timeslots 1-8,24, and the example shows
you a fractional T1 or E1.
Regards,
Jose> Date:
Hi Erin:
I think you are wright, if the call are coming from H323 to SIP you don´t need
to strip the DTMF tones, this is just for calls from H323 to SIP where you
strip the DTMF tones in the SIP incoming dial-peer.
Regards,
Jose
From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Wed, 26 Mar 200
Hi Gustavo:
The recommendation would be the b) option, actually you can round up to 30
kbps, and to make the life easier you have the table of this values in the SRND
guide page 1-15.
Regards,
Jose
Date: Thu, 20 Mar 2008 18:12:57 -0500From: [EMAIL PROTECTED]: [EMAIL
PROTECTED]: [OSL | C
Take a backup of the clean configuration with BARS and every time you need it
do a resote.
Date: Thu, 20 Mar 2008 14:24:56 -0400From: [EMAIL PROTECTED]: [EMAIL
PROTECTED]: [EMAIL PROTECTED]: Re: [OSL | CCIE_Voice] Clear Database
Only easy way I know is to reinstall the CCM. I am sure you alrea
Hi Fernando:
In your debug I am not seen the tech prefix of prepended to the number you
dialed, make sure you are adding this prefix in CCM to this number.
Regards,
Jose
Date: Thu, 20 Mar 2008 17:19:50 +0100From: [EMAIL PROTECTED]: [EMAIL
PROTECTED]; [EMAIL PROTECTED]: [OSL | CCIE_Voice]
Also, if you are using SIP as one of the call legs, use the bind command under
voice service voip, and sip.
Date: Thu, 20 Mar 2008 09:02:52 -0700From: [EMAIL PROTECTED]: [EMAIL
PROTECTED]; [EMAIL PROTECTED]: Re: [OSL | CCIE_Voice] RTP stream addresses used
On your gateways your h323-gatewa
Hi Scott:
Yes, the name hast to be the same in this type of configuration, IOS Enhanced,
when you have just IOS is the MTP plus the MAC address.
Regards,
Jose> Date: Wed, 19 Mar 2008 18:33:11 -0700> From: [EMAIL PROTECTED]> To:
ccie_voice@onlinestudylist.com> Subject: [OSL | CCIE_Voice] Tr
Hi Scott:
I did the lab exam yesterday, I failed :(, but talking to the proctor they are
considering changes for the lab exam but not for this year.
Regards,
Jose
Date: Wed, 19 Mar 2008 13:59:41 -0700From: [EMAIL PROTECTED]: [EMAIL
PROTECTED]: [OSL | CCIE_Voice] CCVP split = CCIE split?
,
Presumably you can do it by assigning a non-dialable number (e.g. A0001) to the
voicemail pilot?
Never tried it myself though...
Rgds
Alex
On 14/03/2008, Jose Linero Welcker <[EMAIL PROTECTED]> wrote:
Hi Mark: I am just curious to see if it can be done, I will have the lab exam
next T
na numbers ...
Mark Snow
Sr Technical Instructor
IPexpert, Inc.
Sent from my iPhone
On Mar 13, 2008, at 10:06 PM, Jose Linero Welcker <[EMAIL PROTECTED]> wrote:
Hi Senthil: Thanks for the reply, I tried applying a cor list to the dial-peer,
however this blocks everything, including the call w
, Mar 13, 2008 at 7:30 PM, Jose Linero Welcker <[EMAIL PROTECTED]> wrote:
Hi: Is there any way to block direct calls to CUE, I mean, just permit a call
into the CUE when you press the message button, or when a call is redirect for
busy or no answer reason. The idea is to block the call when
Hi:
Is there any way to block direct calls to CUE, I mean, just permit a call into
the CUE when you press the message button, or when a call is redirect for busy
or no answer reason. The idea is to block the call when a user dials the
dial-peer number directly just as we can do in Unity with
.
Using CCM 4.2(3), IPCC Express 4.0(4), IOS version 12.4(8) Advance IP Services
Please find the attched debugs and config file.
regards,
Syed Khalid Ali
- Original Message -
From: Jose Linero Welcker
To: Syed Khalid Ali ; ccie_voice@onlinestudylist.com
Cc: [EMAIL PROTECTED]
Sent
Hi Gustavo:
The SRND is talking about class based FRTS not MLPoFR, as JD said in MLPoFR you
need frame-relay traffic-shapping in the main interface.
Regards,
Jose
Date: Tue, 11 Mar 2008 07:25:15 -0500From: [EMAIL PROTECTED]: [EMAIL
PROTECTED]; [EMAIL PROTECTED]: Re: [OSL | CCIE_Voice] QO
Hi:
Are you using B-ACD TCL?, could you post your configuration?.
Regards,
Jose
From: [EMAIL PROTECTED]: [EMAIL PROTECTED]; [EMAIL PROTECTED]: Tue, 11 Mar 2008
14:25:26 +0500CC: [EMAIL PROTECTED]: [OSL | CCIE_Voice] Call on PRI Line are
not transfered to IP-AA
Greetings,
I am havin
Hi:
If we configure the manager in Branch Office one and the assistant in HQ, and
there is not enough bandwidth, the using AAR is there any limitation to use
IPMA and AAR?, what I am seeing is, when the manager can receive the calls,
there is no problem, however when divert all is activated I
Hi Jonathan:
According to CCO, this is the information:
CCM version 4.1(3) with latest SR
Unity version 4.0(5)
CRS version 4.0(1)
Yes the IPCC and the CCCM are coresident, but I think there is a cluster with
Pub and Sub, and the Unity is in a single server.
Regards,
Jose> Date: Mon, 10
Hi Juan:
Actually the 6500 based the QoS in an internal DSCP, that is why you have to
map the COS value to a DSCP and when the packet goes out the switch, you have a
DSCP to COS value map, based on this map you have a behaviour in the transmit
queue.
Hope this help.
Regards,
Jose
Date
Hi:
I fixed the MWI problem, I did not configure the IPMA partition in MWI CSS and
verify that the manager partition were on top.
Know the problem is with AAR, any restriction to make work IPMA in an AAR
situation?
Thanks and regards,
Jose
From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Sa
Hi all,
I am testing IPMA and it is working without problems, however when I left a
voicemail to the manager, the MWI is not working, the configuration I have for
MWI for the phones is this:
MWI Numbers
Partition: MWI-PT
CSS: MWI-CSS --> MWI-PT, Internal-PT, Manager-PT
In all phones the
Hi Priyank
No I have not, but did you try to do a factory reset in CUE?
Regards,
Jose
Date: Thu, 6 Mar 2008 09:24:38 -0600From: [EMAIL PROTECTED]: [EMAIL PROTECTED]:
[OSL | CCIE_Voice] CUE error - Web & JTAPI Login Failed
Hey guys,
I have been getting this error on CUE when I try to int
Hi Edward:
One way could be registering the CCM with a technology prefix, it could be 1#,
2# wichever number, and do the same with the CME, the CME tech prefix could be
the same as the CCM, if you decide to register the CCM and the CME with
different tech prefix the other site hast to append
Hi JD:
Actually I tested and worked, what I had was:
CCM -- h323 G729 (GK Controlled) --- CME -- H323 Dial-peer G711 to B-ACD
I just configured the transcodec resources and the IPIPGW feature, do you have
the allow H323 to H323 command under the voice service?
Do a debug voip dialpeer to
Test, please ignore
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Hi:
I found the problem, was the call-forward pattern.
Regards,
Jose
From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: RE: CUE FNA FB AgainDate: Wed, 27
Feb 2008 03:05:57 +
Hi: I was doing some tests and I realized that due I configured the CME as an
IPIPGW then the gatekeeper trunk conf
Hi:
I was doing some tests and I realized that due I configured the CME as an
IPIPGW then the gatekeeper trunk configuration should have the same settings as
the trunk we configure for IPIPGW, that is with the Outbound faststart and the
inbound faststar checked and also with the MTP checked,
if you were to uninvoke your transcoder, will calls still work coming in
from the PSTN? could you try?
Chad
On Tue, Feb 26, 2008 at 1:26 PM, Jose Linero Welcker <[EMAIL PROTECTED]> wrote:
Hi all: Again I have problems with CFNA and CFB, locally in CME when any of the
users call each oth
Hi all: Again I have problems with CFNA and CFB, locally in CME when any of the
users call each other can leave a voice mail, the MWI is working without
problems, when a PSTN caller calls the phone it can be redirected to a voice
mail and all works, I have a problem when the call is coming from
Hi all,
Quick question, having the connection between CME and CCM trough an IPIPGW
where the CME leg is SIP and the CCM leg is H323, and the call flow is from CME
to CCM do we need always the DTMF relay:
dtmf-relay rtp-nte digit-drop
or can we use just the command:
dtmf-relay rtp-nte
T
PROTECTED]: Re: [OSL | CCIE_Voice] ATA Upgrade to H323
Jose,Did you copy the H323 .zup file to the TFTP directory?
- Original Message From: Jose Linero Welcker <[EMAIL PROTECTED]>To:
[EMAIL PROTECTED]: Friday, February 22, 2008 6:17:02 PMSubject: [OSL |
CCIE_Voice] ATA Upgrade to H3
Hi,
Sorry, maybe this is a stupid question but you know sometimes studying we get
stuck, I am trying to upgrade a ATA with SCCP image to H323, the ATA is
registered to CCM I download the H323 .zup file, and I am filling the field
"Phone Load Name" with the file name, I reset the ATA but it do
rtification Training Tools for the Cisco CCIE R&S Lab, CCIE Security Lab,
CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications.
From: Devildoc [mailto:[EMAIL PROTECTED] Sent: Thursday, February 21, 2008 1:39
PMTo: [EMAIL PROTECTED]; 'Jose Linero Welcker';
n Training Tools for the Cisco CCIE R&S Lab, CCIE
Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab
Certifications.
From: Jose Linero Welcker [mailto:[EMAIL PROTECTED] Sent: Wednesday, February
20, 2008 2:01 PMTo: [EMAIL PROTECTED]; [EMAIL PROTECTED]: RE: [OSL
Hi Vik: Yes it is just for the calls coming from the WAN, What I am seeing is
the CME sends a SIP 302 message to the IPIPGW, and the IPIPGW shoud send a
re-invite to the number of the CUE, however the IPIPGW sends an ACK and I heard
a busy tone, attached are the configurations of the IPIPGW and
do, try to configure the sip incoming dial-peer with "codec g711ulaw"
On Wed, Feb 20, 2008 at 3:29 PM, Jose Linero Welcker <[EMAIL PROTECTED]> wrote:
Hi JD: Actually I have it: dial-peer voice 4 voip session protocol sipv2
incoming called-number 3... dtmf-relay rtp-nte Any ideas? Rega
Hi JD:
Actually I have it:
dial-peer voice 4 voip session protocol sipv2 incoming called-number 3...
dtmf-relay rtp-nte
Any ideas?
Regards,
Jose
From: [EMAIL PROTECTED]: [EMAIL PROTECTED]; [EMAIL PROTECTED]: Wed, 20 Feb 2008
13:22:47 -0800Subject: Re: [OSL | CCIE_Voice] CUE FNA FB Pr
Hi all:
I have this configuration:
CM --- H323 Trunk --- IPIPGW --- SIP Trunk --- CME
The H323 is using G711 and the the SIP Trunk is using G729, the calls between
the phones in CCM and CME are working without problems, however when I am going
to forward the call to CUE due CFNA or CFB I r
e list, route group, gateway and h323' dial-peer are
configured correctly.Only when I assign H323 and 6608 in a single route list,
the call is not able to failover from h323 to 6608Thanks.
On Feb 16, 2008 9:25 PM, Jose Linero Welcker <[EMAIL PROTECTED] <mailto:[EMAIL
PROTECTED]> >
Jose Linero Welcker <[EMAIL PROTECTED]> wrote:
Hi: No there is not an specific requirement, where are you doing the digit
manipulation, in the RP or in the RG? Regards, Jose
Date: Sat, 16 Feb 2008 19:38:12 +0800From: [EMAIL PROTECTED]: [EMAIL
PROTECTED]: [OSL | CCIE_Voice] Route Gro
Hi:
No there is not an specific requirement, where are you doing the digit
manipulation, in the RP or in the RG?
Regards,
Jose
Date: Sat, 16 Feb 2008 19:38:12 +0800From: [EMAIL PROTECTED]: [EMAIL
PROTECTED]: [OSL | CCIE_Voice] Route Group FailoverHi all,I have created 2
Route List: RL_H
13293001.
I took a look at the PSTN WAN but it won't let me do a show run. I assume its
abad prefix going out.
Thanks,
Chad
On 2/9/08, Patel, Mrugesh <[EMAIL PROTECTED]> wrote:
Yes.
Do you see something wrong?
Thanks
From: Jose Linero Welcker [mailto:[EMAIL PROTECT
Hi Patel:
Is your HQ Transcodec registered?
Regards,
Jose
Date: Sat, 9 Feb 2008 09:52:08 -0600From: [EMAIL PROTECTED]: [EMAIL PROTECTED]:
[OSL | CCIE_Voice] gatkeeper issue again
Can anyone please tell me what am I doing wrong here?
I am sending a call from HQ to BR2 via HQ-gatekeep
Hi Mark and all:
If we have a publisher and subsrciber, and the Phones are registered first
to the subscriber having the publisher as a backup, if we want to change the
station keepalive to the Publisher, I think we have to modify the parameter
"Station and Backup Server Keepalive Interval" be
Hi Mark:
I reset the module, reset the switch and the only thing that worked was take
out the car and inserted it again.
Thanks for your help.
Regards,
Jose
From: Mark Snow <[EMAIL PROTECTED]>
To: Jose Linero Welcker <[EMAIL PROTECTED]>
CC: ccie_voice@onlinestudylist.com
Session ID = 163
Total Allocated ISDN CCBs = 0
Actually I was working well with it before I erased the configuration to
begin again.
Thanks for your help.
Regards,
Jose
From: Mark Snow <[EMAIL PROTECTED]>
To: "Jose Linero Welcker" <[EMAIL PROTECTED]>
CC: ccie_voi
Hi:
I begun doing the challenges labs of the workbook, I erased all the
equipments configuration in order to do the lab. I configured the 6608 of
the 6500 and found an issue, the port is registered with callmanager but the
status is notconnect, I check the E1 in the PSTN gateway simulator and
as I did this the
integration worked without problems.
Thanks for your answers anyway.
Regards,
Jose
From: Edward French <[EMAIL PROTECTED]>
To: Jose Linero Welcker
<[EMAIL PROTECTED]>,ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CUE integration problem
Date:
Hi all:
I am doing the integration of CUE with CME, I have a problem I tried several
things including reinstalling the CUE software but I get the same error, the
configuration I have in CME for the integration is:
ip http path flash:
interface Service-Engine1/0
ip unnumbered GigabitEthernet0
Hi:
I am working the point 9 of section 12, I configure the policer for 6500
with a rate of 32k and a burst of 8k according to the Documentation CD,
however looking the solution I am seeing the rate of 32k and the burst of
13. Could anybody explain why this burst value?.
Thanks in advance.
Hi:
I am doing the pint 16 of Section 11 and I have the H323 client (I am using
Netmeeting) registered in the Gatekeeper, also I configure the H323 client
in CCM and I have the CCM registered in Gatekeeper in the zone I define for
this exercise, VIDEO-GK, The configuration I have for this is:
Hi:
Is there any way to get the agent in the ready state when he logs in without
make him to get ready.
Regards,
Jose
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Hi:
If we need to make the setup of the call 3 seconds quicker than default in
R2 we can use the cas-custom command dnis-digits according to its
explanation:
If the AS5x00 doesnt know the number of DNIS digits beforehand, it has to
rely on a timeout mechanism (3 seconds) to detect the end o
Hi:
I am doing the lab section 4 guide, I am configuring the R2 variations it
asks, however it has been difficult for me to find the documentation in the
documentation CD regarding the CAS-custom configuration specific for the
12.4 IOS version for ISR. I have an idea to change the configuratio
Hi Mark:
Could you tell me how can I do that?
Regards,
Jose
From: Mark Snow <[EMAIL PROTECTED]>
Reply-To: <[EMAIL PROTECTED]>
To: "'Jose Linero Welcker'"
<[EMAIL PROTECTED]>,
Subject: RE: [OSL | CCIE_Voice] ICD Agent Ready
Date: Sun, 4 Jun 2006 08:24:
Hi all,
When we are using an agent in CRA ICD the agent has to log in and after push
the ready button to become available to the application, is there any form
that just entering the log in information the agent becomes ready without
pushing the ready button on the phone.
Regards,
Jose
___
Hi Mark:
Yes I did, any idea what is happening. Could be a problem with the database?
Regards,
Jose
From: Mark Snow <[EMAIL PROTECTED]>
Reply-To: <[EMAIL PROTECTED]>
To: "'Jose Linero Welcker'"
<[EMAIL PROTECTED]>,<[EMAIL PROTECTED]>
CC
.
Regards,
Jose
From: Mark Snow <[EMAIL PROTECTED]>
Reply-To: <[EMAIL PROTECTED]>
To: "'Jose Linero Welcker'"
<[EMAIL PROTECTED]>,<[EMAIL PROTECTED]>
CC:
Subject: RE: [OSL | CCIE_Voice] CRA JTAPI Subsystem Problem
Date: Fri, 14 Apr 2006 14:35:30
MAIL PROTECTED]>
To: "'Jose Linero Welcker'" <[EMAIL PROTECTED]>
CC:
Subject: RE: [OSL | CCIE_Voice] CRA JTAPI Subsystem Problem
Date: Thu, 13 Apr 2006 12:41:51 -0700
Enable debug tracing and restart the engine. Check the trace file and
look for any errors with the
rtification
Training Tools for the Cisco CCVP, CCIE R&S, Security, SP, and Voice
Labs!"
-Original Message-
From: Jose Linero Welcker [mailto:[EMAIL PROTECTED]
Sent: Wednesday, April 12, 2006 3:45 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] CRA JTAPI Subsystem Prob
Hi:
I am configuring the cra for the call manager in a lab environment, the
Jtapy subsystem is in service partial state, the reason is when I am trying
to associate the CTI ports in CTI port group configuration when I use the
Associate CTI Ports no CTI ports appears in the search web page, the
Hi, my lab date is June 8th
From: "Andrew Riley" <[EMAIL PROTECTED]>
To:
Subject: Re: [OSL | CCIE_Voice] CCIE Voice Lab Date
Date: Wed, 12 Apr 2006 09:48:42 +1000
I have mine set for April 28th
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wayne Lawson
Sent: Wedne
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