To All,
I apologize for distracting you from your studying.
I'm looking for some input from anyone with Avaya ACS experience.
I'm trying to provide CME to Avaya Partner ACS integration while we migrate
to CME.
I currently have VIC-4FXO and VIC2-FXO connected to extension ports on
the avaya.
Hi group,
I have my CUE configured with CUCM.
I can call into CUE voicemail and check messages, so I know my PIN is good.
I've subscribed my phone to the voiceview service and can get the
authentication url to show on the phone.
I login in with username scphX (thats the username) and the pin,
Hi All,
It's probably a silly question, but it's some I struggle with more often
than I like.
Can some clarify for me how a LOCAL and NATIONAL call generally need to
be dialed / presented outside the U.S.
In all the practice labs I've done, dialing international is usually
straight forward -
To All,
I'm hoping the group can help me understand the call flow for an MVA call.
I'm able to call into the MVA pilot number , have my remote destination
number recognized and be prompted for my PIN and to dial .
But I get the message Your call can not be completed as dialed for
anything I try
manager traces, you can easily check it.
From: ccielabrat ccielab...@gmail.com
Date: Tuesday, August 7, 2012 2:09 PM
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] MVA -Your call can not be completed as
dialed
To All,
I'm hoping the group can
(vipjinda) vipji...@cisco.com
*To:* ccielabrat ccielab...@gmail.com; ccie_voice@onlinestudylist.com
ccie_voice@onlinestudylist.com
*Sent:* Tuesday, August 7, 2012 2:29 PM
*Subject:* Re: [OSL | CCIE_Voice] MVA -Your call can not be completed as
dialed
It uses the re routing CSS on the remote
destination number.
If you check the call manager traces, you can easily check it.
From: ccielabrat ccielab...@gmail.com
Date: Tuesday, August 7, 2012 2:09 PM
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] MVA -Your call can not be completed
Just working on some CUE stuff today and stumbled across the option to
configure dtmf-relay rtp-nte under the sip subsystem of the CUE.
In general , I've always followed the consensus that you shoud use
RTP-NTE everywhere BUT when communicating with CUE which uses
SIP-NOTIFY.
I wanted to reach
I'm trying to use the newer builtin b-acd services in my lab setup.
Can anyone confirm my config and see if there is anything I'm missing?
I'm using the usual B-acd documentation as a template.
I'm getting the a problem where the call drops as soon as I call the pilot
number
application
_dt_prompt.au
param call-retry-timer 15
param voice-mail 4001
param max-time-call-retry 700
param service-name app-b-acd
!
dial-peer voice 4000 pots
service app-b-acd-aa
incoming called-number 4000
direct-inward-dial
On Sat, Dec 10, 2011 at 3:11 PM, ccielabrat ccielab...@gmail.com wrote
Point, as the RP just forwards calls, there are not RTP stream
ever to Route Point.
On Fri, Dec 9, 2011 at 8:23 AM, ccielabrat ccielab...@gmail.com wrote:
I need some help clarifying where I need transcoding for a UCCX where the
trigger DN is located across the WAN (Site B)
I have the MGCP GW
.
Let me test it in my LAB tommorow as well, and will infrom you.
On Fri, Dec 9, 2011 at 4:50 PM, ccielabrat ccielab...@gmail.com wrote:
So if that is the case, I guess it makes sense that the RP would need to
be in the HQ device pool , to force the call to g.729 based on the GW being
to match on the Globalized Calling numbers in your
calling transformation patterns.
The calling party transformation patterns need to be matched on the
pre-RP/RL digits. IE: 5XXX not +12127775XXX.
HTH,
Chris
On Wed, Dec 7, 2011 at 10:13 PM, ccielabrat ccielab...@gmail.com wrote:
To All,
I
to get the send the + globalized # to the gw and have it
trim off the +.
On Thu, Dec 8, 2011 at 8:50 AM, ccielabrat ccielab...@gmail.com wrote:
Thanks Chris,
I'll look into that now.
So , with this in mind, should I be able to check the use external #
mask within the calling party
I just wanted to do some testing on UCCX so I booted a vmware image of UCCX
that I've used before.
It's a fresh install with no integration.
When I log in, it says the JTAPI is out of sync.
I've fixed the JTAPI problem related to moving C:\windows\java files to
c:\winnt\java
It wants me to rerun
I can't get to the point to upload the license.
On Thu, Dec 8, 2011 at 2:07 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote:
do you have the License uploaded to the CCX , this can happen when you
have no license
Ash
On Thu, Dec 8, 2011 at 12:20 PM, ccielabrat ccielab...@gmail.com wrote:
I
, ccielabrat ccielab...@gmail.com wrote:
I can't get to the point to upload the license.
On Thu, Dec 8, 2011 at 2:07 PM, Ashraf Ayyash ash.ayy...@gmail.com
wrote:
do you have the License uploaded to the CCX , this can happen when you
have no license
Ash
On Thu, Dec 8, 2011 at 12:20 PM
I need some help clarifying where I need transcoding for a UCCX where the
trigger DN is located across the WAN (Site B)
I have the MGCP GW in Site B in a Device Pool (SiteB)
UCCX CTI-RP is in Device Pool (SiteB)
and CTI-Ports are all in Device Pool (HQ)
Per my region configuration, g.729 is
To All,
I just needed to check to see if anyone knows about a problem using calling
party transformations at the gateway level.
I have a setup where I am send a fully globalized called and calling # to
my gateways.
I wanted to make all needed adjustments just before it goes to the PSTN.
My
I have MVA configured on my h323 router, with the appropriate dial peers as
per the Cucm help pages.
I am able to call into the piloting dn and I can hear the MVA application
prompt me for my pin.
When I press any digits, they are not recognized.
I've tried to force a g711 codec and ensured
I ran into a problem the other day that has me confused.
I ran auto qos on the hq side, changed values as needed and pasted the
modified policy in the hq router.
I then took the same policy and pasted it into the SB router config and
bound it to the dlci.
All seemed to be ok until I tried to get
as well (without any bandwidth as it's not mentioned in
exam). Then itll take 1.5M by default.
Is there a command to verify that FRTS use 56k bandwidth because above
documents are very old.
Ken
On Sat, Dec 3, 2011 at 8:54 AM, ccielabrat ccielab...@gmail.com wrote:
To All,
I've been
will experience one way audio.
Vik Malhi – CCIE #13890
Managing Partner - IPexpert, Inc.
Telephone: +1.810.326.1444 ext 420
Fax: +1.810.454.0130
Mailto: vma...@ipexpert.com
On Dec 3, 2011, at 10:59 AM, ccielabrat wrote:
Hey everyone,
I can confirm after A LOT of testing, if you are given a QOS
at 2:53 PM, ccielabrat ccielab...@gmail.com wrote:
ok, I'm getting to understand this better.
I don't see any mention of a tcs failure though
See the output of debug h245 asn1 below. Where is the indication of a
failure?
Also, I have CUBE running with a Hw transcoder registered locally
To All,
I've been trying to figure the best/fastest way to get a WAN QOS
requirement completed on exam day.
I've become very comfortable with Auto-QOS and making the needed tweaks, so
Auto-QOS is the way I'm going to use.
The one piece of the strategy that I'm stilll wondering about is if WAN
in the FTP directory root ?
Ash
On Wed, Nov 30, 2011 at 1:03 PM, ccielabrat ccielab...@gmail.com wrote:
I'm trying to add a second language to an AIM-CUE.
I use the command software install add url ftp://x.x.x.x/xyz.pkg
and it seems to run without a problem but when it finishes processing
and see
how it will go
Ash
On Thu, Dec 1, 2011 at 8:02 AM, ccielabrat ccielab...@gmail.com wrote:
Hi Ashraf,
See below. Thank you!
ftp ls
200 Port command successful
150 Opening data channel for directory list.
cue-installer.nm-aim.7.0.1
cue-installer.nm-aim.7.0.6
cue-vm-en_GB
debugs won't show in this
case.
Regards,
Mohammed Al Baqari
Sent from my iPhone
On Dec 1, 2011, at 6:12 PM, ccielabrat ccielab...@gmail.com wrote:
I'm trying to setup a call from HQ CUCM via GK-Trunk to a Remote Gk
Zone.
I have the Gatekeeper configured with OutVia
I don't think this is an NDA topic.
If you think it is, please disregard.
Is there any restriction to create log files on the Lab pc you work off in
the lab?
I like to grab the default CUE config into a text file and modify then
paste back.
Just wondering if the PC is locked down that I can't
I'm trying to add a second language to an AIM-CUE.
I use the command software install add url ftp://x.x.x.x/xyz.pkg
and it seems to run without a problem but when it finishes processing the
file,
I get the follow message :
Language add-ons found on the system (1):
Installed SKUName
serial 0/0/0:23 or 15
interface?
Randall
Sent from my iPhone
On Nov 28, 2011, at 5:52 PM, ccielabrat ccielab...@gmail.com wrote:
I must be missing something easy.
I'm trying to get Calling Name to display on my PSTN phone when
receiving a call from a IP phone going through a H323
I must be missing something easy.
I'm trying to get Calling Name to display on my PSTN phone when receiving
a call from a IP phone going through a H323 gateway.
I've found many links online suggesting it's not supported but then others
suggesting it's possible.
Can someone point me to a good
Can someone help me understand what determines what gets displayed on the
phone display when calling outbound.
I have a setup where I have a h323 Gw and MGCP Gw in a single RL.
I create a route pattern of 9.2345678 and assign it to the RL.
If it goes to the H323 GW , I don't drop the 9 prefix in
Group,
Is there a service parameter to set for TFTP on CUCM to allow the TFTP
server to see new files uploaded without restarting the TFTP Service ?
___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
I need clarification about Gatekeepers using outvia to a CUBE zone.
I've always thought a CUBE config needed the underlying Telephony-service
config to be operational.
Is that the case?
I suppose if the call setup is using g.729 in/out of the CUBE , there is no
need to have anything but a
Voip ?
Any chance to reload the Funky CUE ?
Ash
On Sat, Oct 22, 2011 at 12:12 PM, ccielabrat ccielab...@gmail.com wrote:
I can't get CUE MWI working either.
This is my cue config for SIP
ccn subsystem sip
gateway address 10.1.131.1
mwi envelope-info
mwi sip unsolicited
end
I can't get CUE MWI working either.
This is my cue config for SIP
ccn subsystem sip
gateway address 10.1.131.1
mwi envelope-info
mwi sip unsolicited
end subsystem
I've tried all kinds of config on the CME router without success.
When running debug ccsip messages on the CME router , I don't
No takers?
-- Forwarded message --
From: ccielabrat ccielab...@gmail.com
Date: Sat, Sep 17, 2011 at 12:11 PM
Subject: Task Order in the lab
To: ccie_voice@onlinestudylist.com
To All,
I know this question has been asked 1000 times.
Now that I am ready to schedule my lab exam, I
To All,
I know this question has been asked 1000 times.
Now that I am ready to schedule my lab exam, I need to ask it again.
How are successful exam takers breaking down and grouping the tasks that
make up the lab
to allow for time savings ?
My natural approach so far is to take a sheet of
Anyone successfully get Phoneview working with CUCME?
I see express as an option in the group configuration with telnet://
as the protocol to use but I don't have any idea what would be needed
in the CUCME config.
Thanks
LabRat
___
For more information
Hi All,
I'm working on a better understanding of the options with MOH from CME.
I have one endpoint registered as SCCP and one as SIP.
Questions:
1.) Does the SIP phone configuration for MOH get defined in the
telephony-service area?
2.) And does multicast MOH to sip endpoints work the same as
I'm hoping i missed something simple.
I just got a 9971 registered on a CUCM 7.x server.
it works great but I noticed there are no available softkeys for hold,
park , etc during a call.
It's my first SIP phone I'm using, so is there something I'm missing
regarding supplemental services?
Right, thats why I included OT : (Off topic) in the subject line.
That way, for people who don't want to be bothered can simply ignore it.
On Wed, Mar 16, 2011 at 4:14 PM, Roger Carpio roger.car...@gmail.comwrote:
9971 are not evaluated in CCIE Lab as far as I know therefore, this is not
the
to ignore it if it is addressed to a specialized mailing
list. It also takes extra time to sort out the crap that does not belong
from the important emails.
I'm sure you will get some attention at the cisco support forums and 0
rejection.
El mié, 16-03-2011 a las 17:35 -0400, ccielabrat
Looks like I'm right back in the same mess again.
Now I can't seem to get the CIPC currently running SIP to run as a SCCP
phone.
I even changed the TFTP away from CME and pointed it to CUCM thinking that
would force the phone to register as an SCCP phone.
No good.
Is there a factory reset' type
Can anyone point me towards a good RTMT link that will give the basics of
how to collect a trace from CUCM 7.x ?
I'm looking into a GK related issue and what to see what messaging is
happening on the CUCM side.
I'm stumbling badly with RTMT.
Any help or links are greatly appreciated.
.
One more problem solved by endless hours of frustration ...
On Sat, Jul 31, 2010 at 7:09 PM, ccielabrat ccielab...@gmail.com wrote:
Maybe I'm missing a part of this process.
Should a factory reset be necessary?
On 7/31/10, Miron Kobelski findko...@gmail.com wrote:
You have your load
I'm running into a strange problem where an existing phone running sccp
firmware will not register using SIP firmware.
I've done the following
- Delete the ephone config (sccp)
- issue no create cnf (to delete the sccp cnf info)
- Created voice register pool with the mac address of the phone.
-
worked for me. But I know longer change firmware on CUCME itself -
it's so much easier and quicker by temporary registering the phone on CUCM.
regards
kobel
On Sun, Aug 1, 2010 at 12:13 AM, ccielabrat ccielab...@gmail.com wrote:
ete the ephone config (sccp)
- issue no create cnf (to delete
Assuming the following requirements for BR2 to HQ calling via Gatekeeper:
- Do not use tech prefix or default tech prefix
- Route calls to Sub and then to Pub if Sub unavailable.
So in this case a Static Alias configuration would be needed to allow the
call to route to CUCM without having the
Maybe a codec selection issue.
Do you have any restrictions on CUCM (Regions, Locations) that would
disallow the call setup?
On Sat, Jun 12, 2010 at 11:03 AM, jammer jones jammerjone...@gmail.comwrote:
What would cause the following issues with a gw configured as h323 to cucm
outbound calls
Just my 2 cents.
I had a very similar experience as you in terms of scoring. Some items I
know where working got no points.
It's my impression (and others I've spoken with) is that once the proctor
gets to -21 on your score, they simply stop any further testing in most
cases.
I could be wrong and
I've found a couple links about being able to get a console session to an IP
phone via the AUX port.Completely unsupported but apparently do-able.
Has anyone messed with this?
I have a 7970 that refuses to boot, it flashes but then goes slient.
Before I trash it, I figured I would try to find out
In terms of modifying the signaling DSCP value for phones and what is
generated by CallManager,
Are the entries noted below the only ones I need to check?
Enterprise Parameters
- DSCP for Cisco CallManager to Device Interface*
- DSCP for SCCP Phone Configuration*
Service Parameters:
- Voice
I think I have the VPIM Broadcast config straight now.But I want to confirm
I'm doing it the normal way.
CUE is configured with it's location VPIM-Broadcast ID as 852.
I'm creating a VPIM subscriber with an remote mailbox of 852 and assigning
it to the cue vpim location.
I'm creating a PDL
I just discovered a behavior that surprised me.
If I register only 3 voicemail ports in CM but have 8 port configured in
Unity, the mwi's behavior is
unpredictable.
I can only guess that Unity is attempting to use port 4-8 for MWI ,
regardless of there status within the system.
As soon as I
I'd like to be able to match a dial string with the format *400.I'd like to
match the beginning * and then strip it.
I can't seem to get a valid voice translation rule though.
I assumed I would have to escape the * by using \*
but I get an error when I try to use the following.
rule 1
Thanks for the quick reply.
Actually it is working , I fat fingered it in the config.
On Wed, Jun 24, 2009 at 3:30 PM, Cyrus cyrus@gmail.com wrote:
Hi,
It's not work with translation-rule 1
use
voice translation-rule 1
rule 1 /^\*\(400.\)/ /\1/
Cheers,
Cyrus
On Thu, Jun 25,
I'm testing VPIM delivery from Unity to CUE using a PDL with an extension of
666.
If I send a vm directly to PDL ext 666 , the message goes through and the
MWI light goes on for CUE extensions.
If I send a vm via Broadcast Manager to PDL 666 , the message doesn't go
through and the MWI light
The monitored line is a ephone-dn of 54xxx.
It's not assigned as a standard line to any phone, so it's ALWAYS idle.
So , by pressing the transfer softkey and then the button assigned as
monitor like it acts as a transfer + speed dial.
Strange thing is I wasn't able to get this to work with an
I haven't tried that.I thought if you have full-consult configured, you
would have to press transfer a second time to complete the transfer.
I'll test it on my setup which is a 3825 running 12.4.5b
On Wed, Apr 22, 2009 at 7:00 PM, Sergio Polizer spoli...@hotmail.comwrote:
Have you entered
I'm trying to setup a quick transfer to voicemail option.
I'd like to have a scenario as follows:
Extension 5001 receives a call.
A speed dial is configured on the second button of this ephone (*5002)
The user on 5001 presses transfer and the speed dial button for *5002.
Pressing Transfer again
Can someone tell me where to look for the VATS configuration info ?I know I
can do a search and find it but in case I have to go digging for it on lab
day, I need to know how to navigate to it.
Can someone help me understand what I'm missing regarding call-forward
pattern and transfer-pattern in CME.
I'd like to control where a call can be transferred to.
Either as a consult transfer or using CfwdAll .
Probably a very elementary question but it's never been clear to me.
Regarding how prompt files are referenced within an IPCC script:
How do it related to WHERE the prompt files are expected to be?
Example
QueuePrompt = SP[ICD\ICDQUEUE.wav]
vs
QueuePrompt = P[myprompt.wav]
Can someone point me towards info on how to approach EM to allow a user to
move between HQ and BR1 but make sure 911 calls go out only the local GW.
Alex,
Thanks for the reply.
I got it working.
On Thu, Mar 26, 2009 at 2:46 PM, Alex alex.arsen...@gmail.com wrote:
Have 911/9911 route patterns only in device CSS? Line CSS should not have
any such pattern.
Rgds
Alex
- Original Message -
*From:* ccielab...@gmail.com
*To:*
I think I'm running into a bug but wanted to check if I might be missing
something.
I have a call coming into CME via the HQ-rtr GK.
The call goes through fine if answered on the CME phone. (g729).
If the call go noan to the CUE, I get a fast busy on the HQ phone.
I also get a traceback msg on
I'm a little confused about how transcoding is invoked.
Assuming I have an endpoint that makes a call (g.729) to another endpoint
requiring g.711u.
Is it ALWAYS the g.711 side that must allocate the transcoder resource?
Yep, everything started working once I checked off requires MTP on the
trunk. .
But why? The call is coming in as g.729 and the device pool the trunk is in
is set to g.729
On Mon, Mar 23, 2009 at 11:54 AM, Chris Parker cpar...@cparker.us wrote:
If you are getting ring through when you
Any trick to getting local directory to work on CME?I have Service
local-directory configured and a handful of entries but can't get it to come
up on the phone.
I also have http server and http path configured.
I've re-created the cnf files for good mesaure but nothing works.
Outside of being told specifically to configure a h.225 trunk vs. an IC
Trunk, is there any reason to choose one over another in terms of connecting
to the BR2 CME ?
Any tricks to getting CUCM 7 working on ESXi ?My install keeps failing .
I haven't invested too much time yet, as I'm trying hard to pass with the
current blueprint :)
On Sat, Mar 21, 2009 at 1:15 PM, Arun Kumar arunv...@gmail.com wrote:
Hi
I'm running CUCM 7 and Unity Connection 7 on ESXi
Can someone confirm for me what the requirements are to have ephone-dn's in
CME register to a gatekeeper.
Is it dependent on using a dialplan command ?
I keep getting inconsistent results. (I think)
I also have run into the problem where I use No-Reg on the number
configuration and it still ends
I'm curious how people have approached the order of completing a lab.
What order do you use to ensure gathering the most points and build the lab
correctly without doubling back to adjust things as you get further into the
lab.
I know Vik and Mark have their own opinions on this, but I wanted to
Chris ,
Great layout and exactly what I was looking for.
Thanks
On Tue, Mar 3, 2009 at 10:27 PM, Chris Parker cpar...@cparker.us wrote:
I think everyone has their own way of doing it, but here's how I go about
it:
STEP ZERO - read the whole lab carefully. It's hard to do when you are
I'm working with call park.
If I place an internal call (phone to phone) , I can park the call and
pickup it up on another phone by dialing the call park DN.
If I place a call from PSTN into HQ via the 6608 , I can park the call and
pickup it up on another phone by dialing the call park DN.
If I
I'm trying to get a Poor mans hunt group working by using Unity Call
Handlers to ring a couple of extensions and then ultimately forward the call
out to a PSTN phone.I've gotten everything working but the last forward out
to a PSTN number (91408xxx)
I've checked and changed the default
Jose,
The CSS was set correctly, but I never reset the ports :)
It's all working now.
BTW: On a semi-related note, What is the significance of the CSS assigned
to the VM Pilot number?
I can't see how that DN would ever be leveraged for outbound
service.
- Scott
On Wed, Feb 18, 2009
Agreed, the CSS that is assigned to the VM port is used to determine what
dial patterns are available.
My question is the CSS that is assigned to the actual VM PILOT DN.
As I see it, the PILOT DN wouldn't be used for outbound calling.
- Scott
On Wed, Feb 18, 2009 at 11:55 AM, Cliff McGlamry
That would make sense but it doesn't seem to work that way.
I changed the CSS assigned to the VM PILOT to a CSS without any access to
the internal partitions.
All the VM ports were in an internal partition.
When I dial the VM pilot, it shouldn't work based on my changes, but it
does.
- Scott
I had been struggling with getting the IPIPGW scenario working (Task 4.9) in
the IPEXPERT workbook.
The specific problem was making an H.323 call from CME and have it delivered
via SIP Trunk to CM.
I found , although I had an MRGL assigned to the trunk, the MTP within the
MRGL wasn't in the same
Thanks for the reply Mark.I'll give it a try.
On Fri, Jan 9, 2009 at 1:16 PM, Mark Snow ms...@ipexpert.com wrote:
Well with all as you said, it 'should' have worked fine. That being said,
maybe at one time there was another DP assiged or a different R within one
of the DPs. Sometimes the UCM
I need to get a better understanding of how transcoding is invoked.
I've setup an IPIPGW on the HQ Router.
I'm trying to setup H.323(g711) - SIP (g729) calls to/from Hq/SiteC
I've configured a transcoder and registered it to the telephony service on
the HQ RT.
I've noticed in the IPExpert proctor
After some review, I see that the MTP being configured is a software MTP .
It is being assigned to the device pool that will be used for the ICT trunk
going to the IPIPGw.
-- Forwarded message --
From: ccielab...@gmail.com
Date: Sat, Jan 3, 2009 at 11:17 AM
Subject: Transcoding
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