Didn't work. You are shooting blanks. lol
From: ccie_voice-boun...@onlinestudylist.com
[ccie_voice-boun...@onlinestudylist.com] On Behalf Of anil kumar
[vccie2...@gmail.com]
Sent: Sunday, October 04, 2009 2:28 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL
Don't need it. I think a PG typo. According to the SRST SIP documentation
it's for invoking an IVR type application, like BACD but not BACD. Have no
idea where the app is. It's not built-in either.
From: ccie_voice-boun...@onlinestudylist.com
[ccie_voice-boun
Seemed if I didn't have the h323 gateway voip bind command on one of the
interfaces, I got the same symptoms. The lab (volume2, lab 3) was using all
MGCP (I think) and didn't put the h323 bind command on. That seemed to make a
difference.
Other than that, what does ccapi show? And what is y
___
For more information regarding industry leading CCIE Lab training, please visit
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Hi Brett,
I will have to go back to my Globalization lab and try it again. I took some
notes, but not as detailed as your explanation, so I don't remember exactly
what I saw in all instances. Seemed whatever number showed up in the popup box
while the CUPC softphone was ringing got stored in
1. Depends on your methodology and what the questions are asking. Go through
the dial-plan chapter in the SRND. Spend a lot of time there trying the things
they talk about. I spent a week in that chapter reading, configuring then
documenting in my notes and diagrams. You may be able to get a
Can you let us know your outputs for ..
1. debug isdn q921
2. sh isdn status
--- On Sun, 10/4/09, Erwan Erwan wrote:
From: Erwan Erwan
Subject: [OSL | CCIE_Voice] BR-2 E1 as MGCP gateway
To: ccie_voice@onlinestudylist.com
Date: Sunday, October 4, 2009, 4:32 AM
hi,
I tried to run
Erwan, Few things you may do pls...
1. Check what DNIS number you are getting "debug isdn q931"
2. you may test the voice translation rule :
"test voice-translation rule 60 312301" and see it gets translated to
or not.
HTH...
--- On Sun, 10/4/09, Erwan Erwan wrote:
From: Erwa
I am having some issues with MVA .
When I dial in after keying in my pin number I can turn on or off mobility that
is working fine but I can’t dial out any number I try dialling internal numbers
i.e 3001, 5001 and well as PSTN number as soon as finish dialling the number
after pressing # the cal
It's my understanding that the older gen phones simply do not support
display of the + character so UCM strips it before handing the call to
them. Regardless of where the logs were to be stored, the content is xml
(data is data) in either case is it not? H.
Anywho, re: CUPC, it supports + di
I have a few questions about manipulating CLID and number type.
1) What is the best method for setting "called party" local calls to
subscriber? Transformation Pattern, Called number Transformation Pattern (DP or
GW), or Route Pattern?
2) With the calling party ID set to 10 digits (ex. 3
Excellent. Thanks for the explanation.
From: Brett [brett.sal...@gmail.com]
Sent: Saturday, October 03, 2009 6:37 PM
To: Michael Ciarfello
Cc: ABIOLA ADEFILA; manishankar pandey; OSL Group
Subject: Re: [OSL | CCIE_Voice] conference bridge regisration
'version' use
I had another look at this.
Deleted the users in CCM, re-added them (different user names,) re-associated
everything and it worked. Was able to add contacts in CUP's ccmuser pages and
see presence status, etc. Weird.
From: ccie_voice-boun...@onlinestudy
hi,
call from PSTN to Br1 success, and it supposed to trigger the "Translation
Rule" in Br1, however can't see it.
BR1 config
voice translation-rule 60
rule 1 /^312301\(1...\)/ /\1/
!
!
voice translation-profile In-DNIS
translate called 60
!
voice-port 0/0/0:23
translatio
hi,
I tried to run BR-2 E1 as MGCP gateway, but can't bring the isdn up , any
advice appreciated. tks
Here is config
-
BR-2
controller E1 1/0/0
pri-group timeslots 1-3,16 service mgcp
!
interface Serial1/0/0:15
no ip address
encapsulation hdlc
isdn switch-type primary
'version' used to be an optional command which I believe defaulted to 3.1 in
days past. As of 12.4(22)T the version is now required. The version numbers
themselves correlate to feature support, though I do not have an exhaustive
list for each. Ad Hoc/MeetMe hardware conferencing in cme requires th
Can you (or anyone else) verify if CUPC suffers the same fate? Seemed to in
my testing. Just want to make sure I am correct.
Interesting that there is a difference between softphone mode and deskphone
mode. SO I would watch out for that!!!
From: Mark
10030814q46d453e7v21298d9f67b5c...@mail.gmail.com<mailto:3082f9d40910030814q46d453e7v21298d9f67b5c...@mail.gmail.com>>
Content-Type: text/plain; charset="iso-8859-1"
Hello,
Trying to answer question 6.2 of lab 1 on the procttor lab
After configuring the meetme, dialing t
OK, let's take one direction at a time. CCM to CCME. (5002 to 3002)
It looks like from the GK output, your got your ARQ. Let's debug voip ccapi on
the CCME router and see what we get.
Does the phone ring on the other side?
Where are you getting that message? A voice message from the annuncia
Should the version be 7.0+ ???
sccp ccm 10.10.110.10 identifier 2 version 5.0.1
What does this command actually do under the hood?
From: ccie_voice-boun...@onlinestudylist.com
[ccie_voice-boun...@onlinestudylist.com] On Behalf Of ABIOLA ADEFILA
[adefilabi...@gm
thanks a lot, I will Try this
Akash Patel
Presales Consultant
On Oct 3, 2009, at 4:14 PM, Michael Ciarfello wrote:
tftpd32 program for windows has a quick DHCP server. Also a good one for
tftping files to router flash.
You will need a power brick or a real hub. Connect the phone, your com
I'll have to try a CCME music file again. Thought I tried uploading it once
last night and ran into the same problem. Told me invalid header or something.
From: ABIOLA ADEFILA [adefilabi...@gmail.com]
Sent: Saturday, October 03, 2009 3:56 AM
To: Michael Ciarfello
Isn't that a legacy command from the NM-HD-FARM module and IOS's before,
something like 12.2 or 12.1 or maybe earlier. Can't remember.
From: ccie_voice-boun...@onlinestudylist.com
[ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kumar, Narinder
[narinder.k
tftpd32 program for windows has a quick DHCP server. Also a good one for
tftping files to router flash.
You will need a power brick or a real hub. Connect the phone, your computer
with tftpd32 and the hub on an isolated network. Or the phoe with power brick
via crossover cable to your PC wit
Hello,
i have configured gatekeeper and trunk to call between the hq (5002) and
br2(3002)
calls from either side will give you the message '' the person you are
trying to call is not available''
below is the debug
anyone with an idea?
rgd
Q-RTR#debug gateke
HQ-RTR#debug gatekeeper main 10
HQ-RT
was scrubbed...
> URL:
> http://onlinestudylist.com/pipermail/ccie_voice/attachments/20091004/82d5011a/attachment-0001.htm
>
> ------
>
> Message: 2
> Date: Sat, 3 Oct 2009 16:14:18 +0100
> From: ABIOLA ADEFILA
> Subject: [OSL | CCIE_Voice] Me
SIP SRST is new to me, having trouble figuring out part of the syntax (the
Proctor Guide doesn't really discuss it). On BR1, when configuring voice
register pool 1, there's a line saying "application sip.app", I'm assuming
refering to a file on flash that needs to be invoked in an SRST scenario.
C
iguring the meetme, dialing the number . i get the message,
cant reach number
i remembered configuring it on a local lab with just one callmanager and two
phones, it worked then
Any one with an idea?
-- next part -
Hello,
Trying to answer question 6.2 of lab 1 on the procttor lab
After configuring the meetme, dialing the number . i get the message,
cant reach number
i remembered configuring it on a local lab with just one callmanager and two
phones, it worked then
Any one with an idea?
___
Darren,
I didn't understand your topology, try to put on a piece of paper but didn't
make sense, can you pls try to explain again? Thanks Narinder
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Darren Manners
Sent: Wednesday, 30 Sept
In your config .
voice-card 0
no dspfarm-Why you have no dspfarm ??? change it to dspfarm , reset
your conf bridge.
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ABIOLA ADEFILA
Sent: Sunday, 4 October 2009 12:06 AM
To: manish
Sent via BlackBerry from T-Mobile
-Original Message-
From: bkvalent...@gmail.com
Date: Sat, 3 Oct 2009 14:34:23
To: ABIOLA ADEFILA
Subject: Re: [OSL | CCIE_Voice] conference bridge regisration
Looks like the sccp ccm lines aren't pointing to the correct ip addresses.
Try 10.10.210.1
Hello,
i checked my config the bind was there but still the conference bridge is
not registering
i did
no sccp
sccp
the command is as shown below
hostname BR1-RTR
!
boot-start-marker
warm-reboot
boot-end-marker
!
logging message-counter syslog
!
no aaa new-model
memory-size iomem 20
network-clock-
Hello,
i deleted and copy again and it worked well.
am to make sure that the phones at hq and branch 1 gets there music-on-hold
from the routers and not the PUB or Subscriber
Debug ephone moh was fine on the two routers
but i hear a tone when user's are place on hold at both branches
Regards
On
Hello,
I think i did not put the bind command, will fix it and get back to you
Regards
On Sat, Oct 3, 2009 at 2:59 AM, manishankar pandey wrote:
> Pl check that the name used on the Router config and on the CCM. Both
> needs to match. Yes, it wil be interesting to see your config on route
Hi Akash,
This is a common issue with the newer model of IP phones, you require to
have a DHCP server available after a factory reset, the DHCP server must
point to a TFTP server where the phones firmware files are located.
Example CME config:
P3-R3-2811-163(dhcp-config)#ip dhcp pool Test
P3-R
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