Hi,
How can i hardcode the RAS port that used by the UCM trunk to register with
GK
to the port number that i want instead of the dynamically choose of by the
system?
I know you may put in the device name (gk-trunk) under UCM service param,but
that's for port 1719 and 1720.
Any clue?
Shingei
_
Never mind. The bandwidth was configured on CUCM instead of the "WAN
interface".
On Tue, Jan 4, 2011 at 11:50 AM, Michael Luo wrote:
> I was doing ipexpert's CCIE Voice Volume 1 Lab 6A Task 6.1 - regarding
> Automated Alternate Route (AAR).
>
> In the "Verification" section, it says "Test AAR b
Hi Gents,
I'm working on Hunt group configuration on CUCM. I wanted the CallFwdNoAn and
CallFwdBusy settings shoud be honored for the user who forwarded the call
instead of the Hunt pilot/
Below My config:
One Hunt Group
One Hunt List
One Hunt Pilot (5000) and I've checked Use Personal prefere
Hi harooon,
Thank you for your help.
Unfortunately, RTMT returns this message "no files matches the date
range for node ..."
Is required tracing activated by default on UC to collect these files?
Thanks
Julien
Le 4 janv. 2011 à 11:21, haroon javed a
écrit :
dear Julien,
Use RTMT -->
Hi Roger,
MVA on a MGCP control gateway is possible.In fact,that is a coexisting of
both
MGCP and H323 on the same gateway,but you could not used a MGCP
control PRI for MVA.
you may refer to Netpro for own interest.
https://supportforums.cisco.com/thread/2005673
Shingei.
On Sun, Jan 2, 2011 at
Hi all,
I have a unity connection without LDAP integration. I would like to
monitor users who try to log in. Is there any tools or ways available
to monitor user login attempt/failed?
Julien
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For more information regarding industry leading CCIE
dear Julien,
Use RTMT --> trace and log center--> collect file --- > Role base
security. and use MLA on Unity Connection.
Regards
Haroon
On Tue, Jan 4, 2011 at 1:48 PM, Julien Krieger <
julien.krie...@ineo-gdfsuez.com> wrote:
> Hi all,
>
> I have a unity connection without LDAP integration.
The reason is that each DID is associated with a user's fax, which will end
up in their mailbox. It uniquely identifies them. My idea was to translate
it into another unique number, and then pass those digits to the hunt group
and into the fax.
But another colleague reminded me that SCCP controlle
I'm reviewing the video walk thro lab5.5. In the lab they set localized call
routing for hq and br1 with local partitions for pdt access. Why do they not
use local call routing group for a least 11 digit dialing?
11 digit
Set up a generic route pattern of \+91212.! and make use of lock route
Hi all,
I was wondering if there is anyone else struggling with PSTN configuration
via h323.
I have a lab built on dynamips, and am using one router as PSTN, and I have
voip dial-peers towards this router.
To be able to support this, I have all routers (BR1, BR2, HQ) configured as
CUBE, to pass c
Hi All ,
I have just started the QoS preparation and gone through books and SRND. I
don’t want to go through entire material and just want to concentrate on the
hot topic which is asked frequently . From the forum and friends ,i
concluded that MLP and FRF.12 is asked frequently . Want to d
Mark
Not sure if it'll help but when you are translating the DID to 4 digit on
CUCM, why don't you translate it directly to "4321" (HG number) as oppose to
translating it to "1234" first and then fwding it to HG number "4321"?
My 2 Cents..
- Basant
On Tue, Jan 4, 2011 at 5:16 PM, Mark Davis wr
Hi Jeff,
Thanks for your valuable suggestion.
It worked.
-S
_
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of
givemeccievoice2...@gmail.com
Sent: Sunday, January 02, 2011 2:50 PM
To: ccie_voice@onlinestudylist.com
Subject: Re
Hi Shrini,
If you follow the Features and Services Guide as mentioned before, you will
have success. You need to configure "hairpinning" for MGCP to work with
MVA.
The idea is that you will accept the call using 5999, but the MVA pilot
number will be a different number. You will have to
Hi,
In Unity Connection Voicemail.
When I listen the VM on HQ phone or BR1 Phone the time received in the
envelope is always the time on Unity Connection, where are HQ and BR1 are in
PST and EST respectively.
How can I get local time in the message envelope.
thanks
Shrini
__
Hi
if you want to use the caller phone preferences to route post hunt group
you need to use the Call Forward No coverage setting on their line.
thanks
On 3/01/2011 12:01 AM, ccie_voice-requ...@onlinestudylist.com wrote:
Send CCIE_Voice mailing list submissions to
ccie_voice@onlinestu
Thanks!
I will try these suggestions as well,
From: Roger Källberg [mailto:roger.kallb...@cygate.se]
Sent: Saturday, January 01, 2011 12:06 PM
To: Hough, Earl; ccie_voice@onlinestudylist.com
Subject: SV: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing
Hi Earl,
You might want
Hello People,
today i reviewed CUC and CUE integration with CME trying to analize different
scenario.I succesfully test both CUC and CUE integration Using SIP towards CME,
but i had some problem testing CUC integration using SCCP versus a CME.
The integration guide report that it should be possi
I'm viewing the video and have a question on route patterns.
In the video and books they go through localized call routing 5.5. They set up
local p
___
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www.ipexpert.com
You must also set a "call forward no coverage" (and the associated CSS) on the
dn that forwarded the call to the hunt pilot.
HTH
Sent from my Verizon Wireless Phone
- Reply message -
From: "Naoufal Kerboute"
Date: Sun, Jan 2, 2011 4:06 am
Subject: [OSL | CCIE_Voice] CUCM Huntgroup (No
In addition: Check if you configured CUPS on CUCM under application server
_
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of
shafqut.ha...@gmail.com
Sent: Sunday, January 02, 2011 11:28 PM
To: ccie_voice@onlinestudylist.com
Subject
How do I pass a called number through an ATA 187 analog connection? This is
in regards to the fax server on an old PBX that is currently connected using
two analog ports. There are multiple DIDs that translate to different
extensions that then get forwarded to a hunt group with the two analog
conne
After playing with codec settings on both sides of the GK/CUBE the
results I am experiencing are as follows. In each of these tests I am
able successfully call from a SCCP endpoint at BR2 to a SCCP endpoint at
BR1. And in each of these tests, regardless of the WAN codec, the SIP
endpoints are usi
Hi ShinGei , bkvalentine, Rogers et al
I remember it was successful last time when I configured it another lab when
HQ was h323.
Now I was confused around dial-peers hence had the question.
I will give a try now with MGCP + H323 on HQ and it should work.
Thanks all.
Shrini
From: ShinGei
Yeah, I was playing with the three options I see under “voice service voip ->
h323 -> call start”. The results are pasted below. Also, from the Cisco Voice
Gateways and Gatekeepers text, on page 58 of the H323 chapter:
“You do not have to configure Fast Start – it is used by default unless
Hi Experts,
"Wish you all a Happy and Prosperous New Year 2011"
First question this year :-)
HQ Site is MGCP.
When I call HQ Phone 5002 , HQ PSTN is ringing -- > all is good.
I have configured MVA number 5999 in service parameters and
Media Resources --> MVA --> 5999 / PT-INTERNAL / Eng
hi, set proper timezone for your user in CUC. hth
On Jan 2, 2011 12:02 PM, "Shrini" wrote:
> Hi,
>
> In Unity Connection Voicemail.
>
> When I listen the VM on HQ phone or BR1 Phone the time received in the
> envelope is always the time on Unity Connection, where are HQ and BR1 are
in
> PST and ES
Sorry, I don't fully understand your architecture (in your first post you
indicated you want to have different codecs on both CUBE call legs). But if
you want to do transcoding on CUBE you need to register transcoder on it
(via telephony services).
On Sat, Jan 1, 2011 at 15:36, Hough, Earl wrote:
+ enable Fast Start on CUBE and CUCM trunk (enable inbound fast start or sth
similar).
On Sat, Jan 1, 2011 at 15:49, Miron Kobelski wrote:
> Hi,
>
> SIP phones by default originate outgoing calls using SIP Early Offfer. When
> the next hop is H.323 trunk, you might want to try enabling Fast Star
I agree you need to ensure that fast start is enabled in order to facilitate
SIP Early Offer -> H323 Fast Start interworking. It was always my
understanding that H323 fast start was enabled by default on IOS platforms. In
fact, when you try to enable the command “call start fast” under voice s
I'm running out of ideas. But I think this symptoms could be related to
codec mismatch.
Check your dial-peers (+ remember voice class codec doesn't work with voice
register pool for some IOS versions).
Also you might want to have a look at DTMF settings. SIP uses RFC2833 and
H.323 out of band H.245
Correct…by default when you create a new GK-controlled trunk FastStart is NOT
enabled. You either need to enable it on the trunk, or change the IOS side to
slow start.
From: Miron Kobelski [mailto:findko...@gmail.com]
Sent: Saturday, January 01, 2011 10:18 AM
To: Hough, Earl
Cc: Ashar Si
I think it is 7.01
--
On Jan 1, 2011, at 11:41, George Goglidze wrote:
> Hi all,
>
> I was wondering if anyone knew the exact version in a lab.
> 7.0 ? 7.1 ?
>
> I know the blueprint says that any major version could be on a lab, but I was
> wondering if anyone knew any better.
>
> Many th
Hi,
SIP phones by default originate outgoing calls using SIP Early Offfer. When
the next hop is H.323 trunk, you might want to try enabling Fast Start on it
(voice services voip or voice class h323 -> call start fast). That might
solve your issue with H245 negotiation.
HTH
kobel
On Sat, Jan 1, 2
I have a problem with the calling number in lab 5A. I've configured the
calling party transformations on the HQ router but for some reason it only
shows the 4 digit extension when I dial 911. I have a CSS configured for
the transformation and the pattern set to 1XXX. I have the CSS assigned to
I was doing ipexpert's CCIE Voice Volume 1 Lab 6A Task 6.1 - regarding
Automated Alternate Route (AAR).
In the "Verification" section, it says "Test AAR by reducing the RSVP
BANDWIDTH" allocated to either the HQ or BR1 WAN interface down to 39kbps,
and then making a call from HQ Phone 2 to BR1 Pho
The architecture is as follows:
BR2 (CME) < --(WAN)- > HQ (GK/CUBE) < --(WAN)- > BR1
(CME)
There is a transcoder defined at both the BR2 and the HQ sites and registered
to the local instance of telephony-services, and I can verify that these are
fully functional
Review your dsp resource configuration for callmanager express.sccp should
be registered with the telephony service.
On Jan 1, 2011 12:50 PM, wrote:
> Send CCIE_Voice mailing list submissions to
> ccie_voice@onlinestudylist.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http:/
Hi all,
I was wondering if anyone knew the exact version in a lab.
7.0 ? 7.1 ?
I know the blueprint says that any major version could be on a lab, but I
was wondering if anyone knew any better.
Many thanks,
___
For more information regarding industry l
Hi Shafqut,
1. is the SIP trunk created in CUCM ?
2. Do you see the SIP trunk in CUPS?
3. Check if there is any firewall between CUPC and CUPS blocking port 5060.
--> probably this may be reason
4. incoming/outgoing ACL on CUPS
5. Restart CUP services.
6. Login to CUP cli and do telnet 5060 vic
If u r going from sip to sccp, definitely need transcoder on ur br1
gateway. Try it n let us know.
duy
ccie #27737 voice
tmobile g2
On Jan 1, 2011 10:39 AM, "Miron Kobelski" wrote:
> I'm running out of ideas. But I think this symptoms could be related to
> codec mismatch.
> Check your dial-peer
Restarting CTI manager on CUCM resolved this issue.
_
From: Shrini [mailto:linuxbos...@gmail.com]
Sent: Sunday, January 02, 2011 12:40 PM
To: 'Ashar Siddiqui'; 'haroon javed'
Cc: 'ccie_voice@onlinestudylist.com'
Subject: RE: [OSL | CCIE_Voice] cti ports not registered with UCCX
I have
Hi all,
Does any body has successfully deployed task no 2.4 & 2.5 from Lab 7 ??
Regards
___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
Shrini,
Thanks for your reply.
1. Yes SIP trunk is created in CUCM
2. Yes i can see the SIP trunk in CUPS and its published.
3. I tried by disabling both AntiVirus and Firewall
4. There are three incoming ACLs created automatically (AXL server ACL,
all, cups ACL)
5. Restarted b
I have configured UCCX atleast 20 times never had this problem.
Coincidently I am have the same exact problem now.
I reverted back the snapshot to clean UCCX , reconfigured step by step
(formally looking to the document)
ICD RPs are showing as unknown :-(
Were you able to figure out what the
Hello ShinGei,
Thanks for the info. In the end, MGCP will not make it for us [?]
Regards,
Roger Carpio.
On Sun, Jan 2, 2011 at 9:34 AM, ShinGei Yong wrote:
> Hi Roger,
>
> MVA on a MGCP control gateway is possible.In fact,that is a coexisting of
> both
> MGCP and H323 on the same gateway,but y
To add one thing more that everything is green on "System Troubleshooting" page
except the warnings for components which I haven't configured yet.
Regards,
Shamid
Sent from my BlackBerry® smartphone using my Telenor Persona connection
-Original Message-
From: shafqut hamid
Date: Mon, 3
Hi Anis,
What is Lab7 and what are tasks 2.4, 2.5..plz write it here briefly.
Also to make us help you efficiently please do include what you want to
do, what you have done so far, what is not working, what you have done
to make it work (debugs etc) and then we will be in a position to give
y
Hi Shrini,
What about your UCM configuration?
1.> is your H323 GW registered with UCM?
2.> what is your dialing behavior internally?4 or 10?if is 4,
then your in & outbound dp should be 4 digit patten as well
instead of 10.
Please provide more info
Shingei
On Sun, Jan 2, 2011 at 4:13 PM, Shrini
Thanks for this, good to know. I once had similar behaviour and fixed this
by enabling incoming Fast Start on CUCM. I though that it was also necessary
on originating gw.
On Sat, Jan 1, 2011 at 16:19, Hough, Earl wrote:
> ecent versions of the Cisco IOS use Fast Start for calls that RSVP
> init
Hello,
Did you add this gateway as H323 in CUCM as well? Features and Services
guide does not seem to mention MGCP for this feature.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmfeat/fsmobmgr.html#wp1206437
Were you specifically asked to have MVA and MGCP without H323?
Rega
Sorry, I missed the part where you mention these are 2 CMEs.
But regarding Fast Start - I've just checked that my router never shows this
command in running config. Both "call start fast" and "no call start fast"
don't show up. I can't figure out any other command to verify this at the
moment, but
Hi everyone!
I am trying to understand the right way to calculate the priority value in LLQ
with a RSVP configuration.
I have not been able to find documentation clarifying this.
So supposing HQ-BR1 4 calls g729
ip rsvp bandwitdh = 24*3 + 40 = 112
No problem with the rsvp bandwith, 3 calls wi
Hi Roig,
Each call in
Location based CAC :
g729 - 24k
g711 - 80k
RSVP:
g729 - 40k
g711 - 96k
Gatekeeper:
g729 - 16k
g711 - 128k (not sure 100%)
For your question
Lets take example of one g729 call rsvp
a= Worst case = (L3+PL)*8*100 = (40+10)*8*100 = 40k
b= One G729 = ((L2+L3)+PL)*50*8 = ((40)
Just found another archive straight from Vik's mail box. In more detail
http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg09933.html
Thanks
Shrini
On 1/5/2011 7:42 AM, Roig Borrell, Francesc Xavier wrote:
Hi everyone!
I am trying to understand the right way to calculate the prior
Hi Francesc,
A payload of 20 and 10 is not correct. RSVP and LLQ calculations are two
different things. For RSVP, you calculations are correct.
Correct Payloads (20 ms)
G711 - 160
G729 - 20
For example, FRF.12, G729, with compression:
IP/UDP/RTP - 2 bytes
G729 - 20 bytes
FRF.12
Hi Shrini,
Thank you for your answer. I don't see very clear how you take into
consideration L2 header
These values 40kbps (g729 10ms)/ 24kbps (g729 20ms) only consider
L3+UDP/RTP+Payload.
For 2 g729 calls ip rsvp bandwith= 24+40 =64 OK
But which value would you use for priority queue if you
Hi,
Take your example,my calculation for LLQ would be:
(3 calls x 26.8kbps) + (1 call x 40kbps)
= 120.4kbps ==>121kbps
I provision the first 3 calls in L2 bandwidth calculation,then i'll used
L3 bandwidth calculation for the 4th call,which is the worst case.
So i'll configure the PQ with above
In rsvp for one call 40k
if you set rsvp bandwidth to say 121k
3 g729 (3x40) calls will be allowed, even if you set 159 only 3 calls if
you increase it to 160 all 4 calls pass
It actually does not set the bandwidth, based one the number it
calculates the number of calls.
To test it you may t
Hi,
RSVP reservation and actual LLQ usage are 2 different things.
I think you should keep in mind, that there is no traffic in PQ before RSVP
reservation completes.
For RSVP calculation you only take into account L3. You have 2 possible
bandwidth values:
* standard (24kbps for G729/20ms) and
*
Hi Francesc,
As I noted before, the RSVP bandwidth calculation is different from the LLQ
bandwidth calculation.
For the scenario of 2 RSVP calls, you will need to calculate as follows:
40 + 24 = 64 (one worst case 10ms call and one normal 20 ms)
So under the serial interfaces you will c
Hi Jeff,
Great! Then we agree with the solution for this requirement. :)
Thank you very much!!
De: givemeccievoice2...@gmail.com [mailto:givemeccievoice2...@gmail.com]
Enviado el: miércoles, 05 de enero de 2011 20:53
Para: Roig Borrell, Francesc Xavier; 'Shrini'
CC: ccie_voice@onlinestudylist.c
I looked at the PG and they add in the calculation as I detailed in my most
recent email. However, I am totally with you. The RTP/LLQ is different from
the RSVP CAC and I would think that only a few extra Kbps would account for the
RSVP control traffic in the PQ.
Jeff
From: ccie_voice-
Definitely, Im sorry I didnt understand at first J
Happy studies!
Jeff
From: Roig Borrell, Francesc Xavier [mailto:francesc.ro...@tecnocom.es]
Sent: Wednesday, January 05, 2011 12:12 PM
To: givemeccievoice2...@gmail.com; 'Shrini'
Cc: ccie_voice@onlinestudylist.com
Subject: RE: [OSL |
I disagree... I would never include L2 in RSVP bandwidth calculations.
To see what values RSVP uses, check "show ip rsvp installed" in ringing and
connected states. it is 40 and 24 kbps for g729.
I'd say that RSVP overhead should constitute no more then 1kbps (only
several small messages during RS
After I just agreed with you! J
Below is not the RSVP calculation. That is the LLQ bandwidth calculations.
After I reviewed my notes and figured out the value necessary, I referred to
the PG. The PG calculates the PQ bandwidth by using 1 call at 10ms and 1 call
at 20ms. I am confused as
Hi guys,
Yes, thinking twice it doesn’t make a lot of sense consider the call with the
worst case payload (46.4) in order to adding RSVP signaling.
1 RSVP Request
Dec 17 18:47:58.630: RSVP 10.10.110.2_16548->10.10.110.1_17938[0.0.0.0]: start
requesting 40 kbps FF reservation for 10.10.110.2
2
Thanks for the details debugs Jeff.
Just wanted to double check with you that my examples are also correct ?
Thanks again
Shrini
On 1/5/2011 2:42 PM, Roig Borrell, Francesc Xavier wrote:
Hi guys,
Yes, thinking twice it doesn’t make a lot of sense consider the call
with the worst case paylo
Hi Shrini,
I believe you’re correct as well, but you were detailing the RSVP BW
calculation not the LLQ which the question was asking.
Jeff
From: Shrini [mailto:linuxbos...@gmail.com]
Sent: Wednesday, January 05, 2011 3:32 PM
To: Roig Borrell, Francesc Xavier
Cc: givemeccievoice2...@g
Hi Shrini,
The example you give was the RSVP calculation.In your example you calculated
all the calls
(total 4 calls)in worst case scenario,which is 40kbps per calls.
I'm not very agreed with your calculation, as per SRND pages 3-65 stated:
To provision 4 G729 streams:
(3*24) + 40 =112kbps
Only
I have 20 vouchers for sale if anyone interested.
Regads,
*--*
*Dew Swen
*
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