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,
Thanks
- Original Message
From: Jonathan Charles <[EMAIL PROTECTED]>
To: Edward French <[EMAIL PROTECTED]>
Cc: ccie_voice@onlinestudylist.com
Sent: Tuesday, February 26, 2008 9:48:25 PM
Subject: Re: [OSL | CCIE_Voice] SQL replication check\fix Q.
It
is
a
Cisco
trace
file
translati
It is a Cisco trace file translation tool...
Check it out, it was stolen from TAC...
http://www.employees.org/~pgiralt/TranslatorX/
Jonathan
On Tue, Feb 26, 2008 at 8:41 PM, Edward French <[EMAIL PROTECTED]> wrote:
>
> Ok I have been ccm for 4 years now and have never heard of translaterX, wh
Just a question, with Cisco working hard to move the docs from UniverCD to
Cisco support page, can we still rely on UniverCD during our lab? Or perhaps
we have to look for alternate method?
Thanks.
On Wed, Feb 27, 2008 at 10:09 AM, Mark Snow <[EMAIL PROTECTED]> wrote:
> CME is still here:
> http
CME is still here:
http://www.cisco.com/univercd/cc/td/doc/product/voice/its/index.htm
Just have to poke around to find it.
But the UCCX - That is disturbing.
I am looking into it.
Mark Snow
CCIE #14073 (Voice, Security)
CCSI #31583
Senior Technical Instructor - IPexpert, Inc.
A Cisco Learning
Well, below configuration will block DND function on ephone 2.
!
ephone 2
button 1f2
no dnd feature-ring
!
So, back to my question, Just wondering is it possible when ephone 1 calling
ephone 2, and ephone 1 will hear only silence when ephone 2 pressed DND
(well, which no DND block on ephone 2)?
I think this is more a terminal services issue than anything, I installed the
agent on both Pub And Sub had agent 1 log onto pub agent 2 log on to sub and it
worked fine
- Original Message
From: Jane Ryer (jryer) <[EMAIL PROTECTED]>
To: ccie_voice@onlinestudylist.com
Sent: Tuesday, Febr
Hi,
I was trying to configure CME such that 2 users can intercom to each
other, as per exercise 14.3. It's specifically asked to configure the
intercoms such that both parties can have an immediate 2-way
communication - apart from the barge-in functionality. But apparently, I
have to choose betwe
The same goes for all CME related docs :-( , since last night...
Juan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Prestidge
Sent: Tuesday, February 26, 2008 11:23 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] IPCC Phone Agent
It seems that the (annoying) people moving the documents off Univercd have now
also moved the documents with the URL for IP Phone agents!!
I used to be able to find this by browsing via the following:
Univercd > Customer Contact Software > IPCC Express and IP IVR > CRS 5.0(x) >
English > Docu
Hi Chad:
Thanks for your answer, actually the calls coming from the PSTN are working
with or without the Transcodec for both CUE and B-ACD, the problem is when I
called from CCM trough a Gatekeeper, nor the forward to CUE works nor the B-ACD.
The transcodec is registered:
Feb 26 22:03:08.0
Jose 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 eac
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
The best way to resolve this problem is by placing the incoming dialpeer for
MWI under the same dialpeer for outbound to CUE.
dial-peer voice 3600 voip
incoming called-number 399[89]
destination-pattern 3600
session protocol sipv2
sesion target ipv4:10.5.202.2
dtmf-relay sip-notify
co
You only need digit-drop for incoming sip dialpeer. Other than that, you don't
need it.
JD
From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Tue, 26 Feb 2008 02:16:01
+Subject: [OSL | CCIE_Voice] IPIPGW RTP-NTE
Hi all, Quick question, having the connection between CME and CCM trough an
IPI
Hi all,
Just wondering is it possible when 2111 calling , and pressed DND
then 2111 will hear only silence?
!
ephone-dn 2
number
!
ephone 2
button 1f2
no dnd feture-ring <-- and exactly what this command do?
!
Thanks.
This dial peer existed on my BR2 router, left over from workbook task
4.9 to setup an IPIPGW:
dial-peer voice 10 voip
incoming called-number 3...
session protocol sipv2
dtmf-relay rtp-nte
I setup CUE on BR2, which resulted in this ephone-dn:
ephone-dn 36
number 3999...
mwi on
I
possibly - can you try IP Blue as a temporary test?
Mark Snow
CCIE #14073 (Voice, Security)
CCSI #31583
Senior Technical Instructor - IPexpert, Inc.
A Cisco Learning Partner - We Accept Learning Credits!
Telephone: +1.810.326.1444
Fax: +1.309.413.4097
Mailto: [EMAIL PROTECTED]
IPexpert - The Glo
I was working on IPCC Express scripting yesterday, and had problems with
getting two phones logged in to IP Phone Agent so that I could test out
my script.
I was connected to the Proctor labs rack through EasyVPN, with 7961
phones at home. I created one-button-login service for phone 3 at both
Here are the details,
AAR Group: HQ and BR1, from HQ's perspective prefix 91 and same from BR1
side.
Location bandwidth was set to 23 on BR1 side
Yes, external mask was specified.
CCM service parameter for AAR was set to TRUE and restarted CCM service.
Created seperate partitions and calling s
Please include a few more specifics:
- AAR Group and what the Prefix is for both Groups (Groups should/must
differ from Phone in SiteA and Phone in SiteB)
- Please note what Location bandwidth you have for both Phones (only
one location must be set to 23 or less)
- Please note if you have any
http://cisco.com/en/US/products/sw/voicesw/ps5640/products_configuration_guide_chapter09186a00806b0a0f.html
Configuring DTMF Relay Digit-Drop on an IP-to-IP Gateway with Cisco
Unified CallManager
To avoid sending both in-band and out-of band tones to the outgoing
leg when sending IPIPGW cal
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