[OSL | CCIE_Voice] PSTN Gatekeeper issues

2009-08-08 Thread Bill Hatcher
I have my own lab hardware so I have had to prot the configurations to my
equipment.  I am using 3825's for my HQ, PSTN, BR1 and BR2 gateways.  The
only issue I am having is trying to call that India number regiseterd on my
PSTN router from HQ through the Gatekeeper.  Looking at the debug h225 asn1
it seems that I get a location request from the HQ router for the number
916745738932.  The gatekeeper sends a location rejection.  Any ideas would
be a big help here.  I have been beating my head against it for a couple of
hours.  I am sure it is something little that I am missing.

Aug  9 09:38:27.359: RAS INCOMING ENCODE BUFFER::=
4A800815010580C49A78A6BC6540B5124A828B9030119288139D6EE1A7190019020A0AD2640901050045456C78335007000A0AD20B06B8211E0037003000430036004200370031003400300030003000300030003000300033000A0A6E0106B7178040070140010055005301800105
Aug  9 09:38:27.359:
Aug  9 09:38:27.359: RAS INCOMING PDU ::=
value RasMessage ::= locationRequest :
{
  requestSeqNum 2070
  destinationInfo
  {
dialedDigits : "916745738932"
  }
  nonStandardData
  {
nonStandardIdentifier h221NonStandard :
{
  t35CountryCode 181
  t35Extension 0
  manufacturerCode 18
}
data '828B9030119288139D6EE1A7190019020A0A...'H
  }
  replyAddress ipAddress :
  {
ip '0A0A6E01'H
port 1719
  }
  sourceInfo
  {
h323-ID : {"US"}
  }
  canMapAlias TRUE
  hopCount 6
}

Aug  9 09:38:27.363: H225 NONSTD INCOMING ENCODE BUFFER::=
828B9030119288139D6EE1A7190019020A0AD2640901050045456C78335007000A0AD20B06B8211E0037003000430036004200370031003400300030003000300030003000300033
Aug  9 09:38:27.363:
Aug  9 09:38:27.363: H225 NONSTD INCOMING PDU ::=
value LRQnonStandardInfo ::=
{
  ttl 6
  nonstd-callIdentifier
  {
guid '009288139D6EE1A7190019020A0AD264'H
  }
  gatewaySrcInfo
  {
e164 : "12123945002"
  }
  h225NonStdSrcCallSignalAddress h225NonStdIpAddress :
  {
ip '0A0AD20B'H
port 1720
  }
  h225NonStdSrcendpointIdentifier {"70C6B7140003"}
}

Aug  9 09:38:27.363: H225 NONSTD OUTGOING PDU ::=
value LRJnonStandardInfo ::=
{
  lrjterminationCause releaseCompleteCauseIE : '08028083'H
}

Aug  9 09:38:27.363: H225 NONSTD OUTGOING ENCODE BUFFER::= 510008028083
Aug  9 09:38:27.363:
Aug  9 09:38:27.363: RAS OUTGOING PDU ::=
value RasMessage ::= locationReject :
{
  requestSeqNum 2070
  rejectReason undefinedReason : NULL
  nonStandardData
  {
nonStandardIdentifier h221NonStandard :
{
  t35CountryCode 181
  t35Extension 0
  manufacturerCode 18
}
data '510008028083'H
  }
}

Aug  9 09:38:27.363: RAS OUTGOING ENCODE BUFFER::=
51081568B51206510008028083
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Re: [OSL | CCIE_Voice] 8.2 Park BLF retrieval not working

2009-08-08 Thread Aamir Panjwani
yep I had exactly the same issue when I tested few weeks ago

 

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Michael
Ciarfello
Sent: Sunday, 9 August 2009 4:24 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] 8.2 Park BLF retrieval not working

 

I can park (directed park) a call using the call park BLF, but can't
retrieve the call using the BLF as the proctor guide indicates.  It says
"Park Slot Unavailable" and get a fast busy tone.  I can retrieve the
parked call by dialing 80 and 8555 just fine.  Tried it on a 7960 and a
7970.

 

Found this post on Cisco Netpro that indicates you need another
(regular) speed dial as 808555 to retrieve it.  

http://forum.cisco.com/eforum/servlet/NetProf?page=netprof&forum=Unified
%20Communications%20and%20Video&topic=IP%20Phone%20Services%20for%20End%
20Users&topicID=.ee6c82d&CommCmd=MB%3Fcmd%3Dpass_through%26location%3Dou
tline%40^1%40%40.1ddbd838/2#selected_message
 

 

Also Feature and Services Guide DOESN'T say (that I can find) that you
can retrieve the directed parked call by using the Park BLF.  It only
says you can use the BLF to park the call.

 

The question doesn't say we have to retrieve the DCP from the DCP BLF.
Maybe it's just a verification section slip-up?

 

Anyone have any different expereinces or get this to work?

Thanks


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[OSL | CCIE_Voice] 8.2 Park BLF retrieval not working

2009-08-08 Thread Michael Ciarfello
I can park (directed park) a call using the call park BLF, but can't retrieve 
the call using the BLF as the proctor guide indicates.  It says "Park Slot 
Unavailable" and get a fast busy tone.  I can retrieve the parked call by 
dialing 80 and 8555 just fine.  Tried it on a 7960 and a 7970.

Found this post on Cisco Netpro that indicates you need another (regular) speed 
dial as 808555 to retrieve it.
http://forum.cisco.com/eforum/servlet/NetProf?page=netprof&forum=Unified%20Communications%20and%20Video&topic=IP%20Phone%20Services%20for%20End%20Users&topicID=.ee6c82d&CommCmd=MB%3Fcmd%3Dpass_through%26location%3Doutline%40^1%40%40.1ddbd838/2#selected_message

Also Feature and Services Guide DOESN'T say (that I can find) that you can 
retrieve the directed parked call by using the Park BLF.  It only says you can 
use the BLF to park the call.

The question doesn't say we have to retrieve the DCP from the DCP BLF.  Maybe 
it's just a verification section slip-up?

Anyone have any different expereinces or get this to work?
Thanks
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Re: [OSL | CCIE_Voice] 7.1 - inconsistent wording between WB and PG

2009-08-08 Thread Michael Ciarfello
And 8.1 (fails to mention cbarge)

From: ccie_voice-boun...@onlinestudylist.com 
[ccie_voice-boun...@onlinestudylist.com] On Behalf Of Michael Ciarfello 
[mciarfe...@iplogic.com]
Sent: Saturday, August 08, 2009 11:08 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] 7.1 - inconsistent wording between WB and PG

The WB fails to mention multicast MOH as a backup where the proctor guide does.

Hope that doesn't happen on the real exam.  lol.

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[OSL | CCIE_Voice] Great Website

2009-08-08 Thread Jason Granat
The author of the CUPS book I posted about earlier has a great
website. Lots of info about virtualization of UC components and lots
of other good stuff:

http://htluo.blogspot.com/


Sent while mobile



http://slash128.com
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[OSL | CCIE_Voice] BACD + Hunting

2009-08-08 Thread dave menk

Can I setup BACD so that I can hunt from one hunt group to another hunt group? 
I have tried this using various methods and no matter what I try the call is 
not released from the first hunt group until the alternate is reached.
This needs to on a RNA.

Hunt Group1 ---> Hunt Group2 ---> Hunt Group 3 ---> Voice Mail.

Current BACD Config
application
service callq flash:app-b-acd-2.1.2.2.tcl
param aa-hunt3 3303
param queue-len 10
param aa-hunt4 3304
param aa-hunt1 3301
param number-of-hunt-grps 4
param queue-manager-debugs 1
param aa-hunt2 3302
!
service aa flash:app-b-acd-aa-2.1.2.2.tcl
paramspace english index 0
param number-of-hunt-grps 2
param handoff-string aa
paramspace english language en
param max-time-vm-retry 2
param aa-pilot 3300
paramspace english location flash:
param second-greeting-time 60
param call-retry-timer 15
param max-time-call-retry 20
param voice-mail 3399
param service-name callq
! 
service aa2 flash:app-b-acd-aa-2.1.2.2.tcl
param voice-mail 3399
param handoff-string aa2
paramspace english language en
paramspace english index 0
param aa-pilot 3311
paramspace english location flash:
param service-name callq
param number-of-hunt-grps 1
param drop-through-option 3
!
service aa3 flash:app-b-acd-aa-2.1.2.2.tcl
paramspace english index 0
paramspace english language en
param drop-through-option 4
param voice-mail 3399
paramspace english location flash:
param handoff-string aa3
param aa-pilot 3312
param service-name callq
param number-of-hunt-grps 1
!
ephone-hunt 11 sequential
pilot 3301
list 3000, 3001, 3002
final 3311
timeout 3, 3, 3
ephone-hunt 12 sequential
pilot 3302
list 3100, 8125
final 3312
ephone-hunt 13 peer
pilot 3303
list 3002, 3001, 3000
final 3312
timeout 10, 10, 10
ephone-hunt 14 peer
pilot 3304
list 3001, 3000, 3002
final 3399
timeout 20, 20, 20

Hunt groups can be called directly and dial peers are configured for all 
service aa.

dial-peer voice 3 voip
destination-pattern 3300$
session target ipv4:142.42.65.254
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
!
dial-peer voice 4 voip
service aa
session target ipv4:142.42.65.254
incoming called-number 3300$
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
!
dial-peer voice 14 voip
destination-pattern 3311$
session target ipv4:142.42.65.254
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
!
dial-peer voice 15 voip
service aa2
incoming called-number 3311$
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
!
dial-peer voice 16 voip
destination-pattern 3312$
session target ipv4:142.42.65.254
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad 
!
dial-peer voice 17 voip
service aa3
incoming called-number 3312$
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad

Thanks,
Dave 
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[OSL | CCIE_Voice] 7.1 - inconsistent wording between WB and PG

2009-08-08 Thread Michael Ciarfello
The WB fails to mention multicast MOH as a backup where the proctor guide does.

Hope that doesn't happen on the real exam.  lol.

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Re: [OSL | CCIE_Voice] sip aar problem

2009-08-08 Thread Michael Ciarfello
Taking a look at the trace files and guessing what they mean since I am not a 
developer.
When I configured RSVP mandatory from HQ to HQ and have a HQ phone dial 911, 
the call fails.

Location says no need to reserve BW.  That's not a surprise.
Later on I see it try to allocate an RSVP MTP resource.
Then it says sendErrorOnRSVPResourceReq - RSVP not supported

I'm guessing becasue HQ to HQ location had mandatory, so it allocated it and 
RSVP over a PSTN is not supported???

Anyways, back to your question...
Did you try looking at the trace files?  Curious what the locations section is 
telling you.

Oh, did you do AAR group on the device or on the line level?
Something things you are in a different location than you are and only for the 
sip phone.  weird.


From: c george [cisco...@hotmail.com]
Sent: Saturday, August 08, 2009 1:44 PM
To: Michael Ciarfello; jonv...@gmail.com
Cc: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] sip aar problem

no dice. rsvp not configured.  but if it was I would have ben in shock as rsvp 
has to do nwith bandwidth wan calls.

so pstn direct or inbound calls should not be a issue do to rsvp


or am i missing something

Respectfully Charles George




From: cisco...@hotmail.com
To: mciarfe...@iplogic.com; jonv...@gmail.com
Date: Sat, 8 Aug 2009 16:50:10 +
CC: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] sip aar problem

i will check that. infact i believe i did do rsvp mandatory, but what would 
that have to do with a pstn call. i would think CAC only matters for WAN calls

Respectfully Charles George




From: mciarfe...@iplogic.com
To: cisco...@hotmail.com; jonv...@gmail.com
CC: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
Date: Sat, 8 Aug 2009 01:47:35 -0400
Subject: RE: [OSL | CCIE_Voice] sip aar problem


Did you ever get this solved?



I ran into something maybe similar.  I had the locations stuff configured 
wrong.  My HQ to HQ location also had RSVP mandatory and I was getting out of 
bandwidth messages when dialing, say, 911 from HQ—nothing too important.



From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of c george
Sent: Tuesday, August 04, 2009 2:28 PM
To: jonv...@gmail.com
Cc: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] sip aar problem



let me also add this is a h323 gw

and when in a low bandwidth state inbound call fail as well.

a debug voip ccapi inout show call fails because ccm is not responding

Respectfully Charles George






From: cisco...@hotmail.com
To: jonv...@gmail.com
Date: Tue, 4 Aug 2009 18:18:23 +
CC: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] sip aar problem

great idea!!

same results. sip phone 7961 cannot call out dialing a PSTN #. (9+#)

But AAR works. internal number shows reroting and dials over the PSTN.

is anyone else using a 7961 and checked sip aar for both internal and dialing a 
pstn number directly while in a low bandwidth state ?

if it worls, what sip firmware are you using and what ver cucm

Respectfully Charles George



> Date: Tue, 4 Aug 2009 12:35:11 -0500
> Subject: Re: [OSL | CCIE_Voice] sip aar problem
> From: jonv...@gmail.com
> To: cisco...@hotmail.com
> CC: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
>
> OK, put them both in the same CSS and see if the problem goes away
> (use the SCCP CSS for the SIP phone...)
>
> Then you know it is a RG issue...
>
> J
>
> On Tue, Aug 4, 2009 at 11:44 AM, c george wrote:
> > different css for sip and sccp
> >
> > everything is fine with sccp
> >
> > this is only a issue with a sip phone
> >
> > makes no sense to me because PSTN Calling should not have anything to do do
> > with how much bandwith you have. The call is going out its local gateway.
> >
> > Respectfully Charles George
> >
> >
> >
> >> Date: Tue, 4 Aug 2009 11:32:51 -0500
> >> Subject: Re: [OSL | CCIE_Voice] sip aar problem
> >> From: jonv...@gmail.com
> >> To: cisco...@hotmail.com
> >> CC: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
> >>
> >> Are you matching a pattern pointing at a local route group? Is the
> >> local route group down? Is it failing over to a remote RG? Are the CSS
> >> of the SIP and SCCP phones the same?
> >>
> >>
> >> Jonathan
> >>
> >> On Tue, Aug 4, 2009 at 11:08 AM, c george wrote:
> >> > 1st of this is only a Sip Problem
> >> >
> >> > 7961 phone  firmware Sip 41.8-5-2s
> >> > CUCM 7.1.2009
> >> >
> >> > AAR works
> >> > but when I call a PSTN NUMBER by dialing 9+PSTN the call fails stating
> >> > not
> >> > enough Bandwidth.
> >> >
> >> > But AAR works. When dialing a Internal number call reroute

[OSL | CCIE_Voice] QoS question Vol 2 Lab 1 Question 5.3

2009-08-08 Thread Steve Sarrick
Probably my weakest area is QoS - but I'm learning!

 

In the Vol 2 lab 1 there is a scenario where we want use LFI+FRTS+MLP
LFI between HQ and BR2.

 

I get the whole crtp and calculating our speed to allow 4 g729 calls.
What I'm not grasping is at the HQ site they use auto QoS voip trust
fr-atm and at BR2 they use auto qos voip fr-atm (no trust).  Am I
reading into this more than I should and I can use the trust or not
here, is it a time saver, or is there some other logic.  I'm not
grasping this piece and am interested if someone could help me get my
mind around this concept a bit.

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[OSL | CCIE_Voice] CUE in SRST...

2009-08-08 Thread Jonathan Charles
I posted this to other groups without adequate response...

So, I have a customer who wants to put in a CUE at every remote site
for VM, but use CUCM for call processing (so the CUE would work in
SRST, but be connected to CUCM via JTAPI).

We have demo gear from Cisco and we have the CUE7 working with CCM7,
in SRST, we have most functionality, but MWI does not work... (in ccm
mode, everything works...)

I have found lots of documentation saying it is supported, but none
saying how to do it...

>From the SRND:

"Although MWI changes are not supported in SRST mode with previous
releases of Cisco Unity Express, voice messages can be sent and
retrieved as during normal operation, but the MWI lamp state on the
phone remains unchanged until the phone registers again with Unified
CM. At that time, all MWI lamp states are automatically resynchronized
with the current state of the users' Cisco Unity Express voicemail
boxes. Cisco Unity Express 3.0 and later releases support MWI for SRST
mode"

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/6x/unityx.html

Anyone see this work?


Jonathan
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Re: [OSL | CCIE_Voice] sip aar problem

2009-08-08 Thread c george

no dice. rsvp not configured.  but if it was I would have ben in shock as rsvp 
has to do nwith bandwidth wan calls.

 

so pstn direct or inbound calls should not be a issue do to rsvp

 

 

or am i missing something

Respectfully Charles George


 


From: cisco...@hotmail.com
To: mciarfe...@iplogic.com; jonv...@gmail.com
Date: Sat, 8 Aug 2009 16:50:10 +
CC: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] sip aar problem



i will check that. infact i believe i did do rsvp mandatory, but what would 
that have to do with a pstn call. i would think CAC only matters for WAN calls

Respectfully Charles George


 


From: mciarfe...@iplogic.com
To: cisco...@hotmail.com; jonv...@gmail.com
CC: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
Date: Sat, 8 Aug 2009 01:47:35 -0400
Subject: RE: [OSL | CCIE_Voice] sip aar problem







Did you ever get this solved?
 
I ran into something maybe similar.  I had the locations stuff configured 
wrong.  My HQ to HQ location also had RSVP mandatory and I was getting out of 
bandwidth messages when dialing, say, 911 from HQ—nothing too important.
 


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of c george
Sent: Tuesday, August 04, 2009 2:28 PM
To: jonv...@gmail.com
Cc: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] sip aar problem
 
let me also add this is a h323 gw
 
and when in a low bandwidth state inbound call fail as well.
 
a debug voip ccapi inout show call fails because ccm is not responding

Respectfully Charles George


 



From: cisco...@hotmail.com
To: jonv...@gmail.com
Date: Tue, 4 Aug 2009 18:18:23 +
CC: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] sip aar problem

great idea!!
 
same results. sip phone 7961 cannot call out dialing a PSTN #. (9+#)
 
But AAR works. internal number shows reroting and dials over the PSTN.
 
is anyone else using a 7961 and checked sip aar for both internal and dialing a 
pstn number directly while in a low bandwidth state ?
 
if it worls, what sip firmware are you using and what ver cucm

Respectfully Charles George


 
> Date: Tue, 4 Aug 2009 12:35:11 -0500
> Subject: Re: [OSL | CCIE_Voice] sip aar problem
> From: jonv...@gmail.com
> To: cisco...@hotmail.com
> CC: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
> 
> OK, put them both in the same CSS and see if the problem goes away
> (use the SCCP CSS for the SIP phone...)
> 
> Then you know it is a RG issue...
> 
> J
> 
> On Tue, Aug 4, 2009 at 11:44 AM, c george wrote:
> > different css for sip and sccp
> >
> > everything is fine with sccp
> >
> > this is only a issue with a sip phone
> >
> > makes no sense to me because PSTN Calling should not have anything to do do
> > with how much bandwith you have. The call is going out its local gateway.
> >
> > Respectfully Charles George
> >
> >
> >
> >> Date: Tue, 4 Aug 2009 11:32:51 -0500
> >> Subject: Re: [OSL | CCIE_Voice] sip aar problem
> >> From: jonv...@gmail.com
> >> To: cisco...@hotmail.com
> >> CC: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
> >>
> >> Are you matching a pattern pointing at a local route group? Is the
> >> local route group down? Is it failing over to a remote RG? Are the CSS
> >> of the SIP and SCCP phones the same?
> >>
> >>
> >> Jonathan
> >>
> >> On Tue, Aug 4, 2009 at 11:08 AM, c george wrote:
> >> > 1st of this is only a Sip Problem
> >> >
> >> > 7961 phone  firmware Sip 41.8-5-2s
> >> > CUCM 7.1.2009
> >> >
> >> > AAR works
> >> > but when I call a PSTN NUMBER by dialing 9+PSTN the call fails stating
> >> > not
> >> > enough Bandwidth.
> >> >
> >> > But AAR works. When dialing a Internal number call reroutes as is
> >> > expected.
> >> > 3000 - 4375553000. Call works, terminates and completes.
> >> >
> >> > Respectfully Charles George
> >> >
> >> >
> >> >
> >> > 
> >> > Get back to school stuff for them and cashback for you. Try BingT now.
> >> > ___
> >> > For more information regarding industry leading CCIE Lab training,
> >> > please
> >> > visit www.ipexpert.com
> >> >
> >> >
> >
> > 
> > Get back to school stuff for them and cashback for you. Try BingT now.



Get back to school stuff for them and cashback for you. Try BingT now.



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Re: [OSL | CCIE_Voice] Performance of vmware in SSD Harddisk

2009-08-08 Thread Thomas Koch
UCM load time is 2-3 min running ESXi 3.5 on HP DL 380 G3 Raid 1+0 

Hope that helps.

 

Thomas J Koch
Owner/Consultant 
Digitones, LLC
Cell: 630-808-4910
E-mail: digito...@comcast.net

  _  

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of So Gwaai
Sent: Saturday, August 08, 2009 11:50 AM
To: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Performance of vmware in SSD Harddisk

 

Actually the loading time of UCM is ok for me (around 3 min from CLI appear
to GUI login page), but the time of gui of CUC is quiet long...

On Sat, Aug 8, 2009 at 11:44 PM, Tanner Ezell 
wrote:

Unlikely. In my VMware box (which is a Q6600 with 8gb ram and a standard
7200RPM hard disk) it takes maybe 2 minutes to bring UCM up.

 

How long does it take for the CLI to come up? And are you logging into the
CLI after it does.

 

As I recall, there is an issue with VMware UCM that prevents anything from
happening until after you log into the CLI.

On Sat, Aug 8, 2009 at 7:54 AM, So Gwaai  wrote:

Hi,

I'd just want to know if any significance improvement when I move my vmware
based (vmware workstation 6.5) CUC from the SATA harddisk to SSD harddisk?
Since I need to wait more than 10 min to get the GUI login prompt after the
CLI prompt appear... 

Thanks for any help

___
For more information regarding industry leading CCIE Lab training, please
visit www.ipexpert.com

 

 

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Re: [OSL | CCIE_Voice] Performance of vmware in SSD Harddisk

2009-08-08 Thread So Gwaai
Actually the loading time of UCM is ok for me (around 3 min from CLI appear
to GUI login page), but the time of gui of CUC is quiet long...

On Sat, Aug 8, 2009 at 11:44 PM, Tanner Ezell wrote:

> Unlikely. In my VMware box (which is a Q6600 with 8gb ram and a standard
> 7200RPM hard disk) it takes maybe 2 minutes to bring UCM up.
> How long does it take for the CLI to come up? And are you logging into the
> CLI after it does.
>
> As I recall, there is an issue with VMware UCM that prevents anything from
> happening until after you log into the CLI.
>
> On Sat, Aug 8, 2009 at 7:54 AM, So Gwaai  wrote:
>
>> Hi,
>>
>> I'd just want to know if any significance improvement when I move my
>> vmware based (vmware workstation 6.5) CUC from the SATA harddisk to SSD
>> harddisk? Since I need to wait more than 10 min to get the GUI login prompt
>> after the CLI prompt appear...
>>
>> Thanks for any help
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com


Re: [OSL | CCIE_Voice] sip aar problem

2009-08-08 Thread c george

i will check that. infact i believe i did do rsvp mandatory, but what would 
that have to do with a pstn call. i would think CAC only matters for WAN calls

Respectfully Charles George


 


From: mciarfe...@iplogic.com
To: cisco...@hotmail.com; jonv...@gmail.com
CC: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
Date: Sat, 8 Aug 2009 01:47:35 -0400
Subject: RE: [OSL | CCIE_Voice] sip aar problem







Did you ever get this solved?
 
I ran into something maybe similar.  I had the locations stuff configured 
wrong.  My HQ to HQ location also had RSVP mandatory and I was getting out of 
bandwidth messages when dialing, say, 911 from HQ—nothing too important.
 


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of c george
Sent: Tuesday, August 04, 2009 2:28 PM
To: jonv...@gmail.com
Cc: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] sip aar problem
 
let me also add this is a h323 gw
 
and when in a low bandwidth state inbound call fail as well.
 
a debug voip ccapi inout show call fails because ccm is not responding

Respectfully Charles George


 



From: cisco...@hotmail.com
To: jonv...@gmail.com
Date: Tue, 4 Aug 2009 18:18:23 +
CC: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] sip aar problem

great idea!!
 
same results. sip phone 7961 cannot call out dialing a PSTN #. (9+#)
 
But AAR works. internal number shows reroting and dials over the PSTN.
 
is anyone else using a 7961 and checked sip aar for both internal and dialing a 
pstn number directly while in a low bandwidth state ?
 
if it worls, what sip firmware are you using and what ver cucm

Respectfully Charles George


 
> Date: Tue, 4 Aug 2009 12:35:11 -0500
> Subject: Re: [OSL | CCIE_Voice] sip aar problem
> From: jonv...@gmail.com
> To: cisco...@hotmail.com
> CC: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
> 
> OK, put them both in the same CSS and see if the problem goes away
> (use the SCCP CSS for the SIP phone...)
> 
> Then you know it is a RG issue...
> 
> J
> 
> On Tue, Aug 4, 2009 at 11:44 AM, c george wrote:
> > different css for sip and sccp
> >
> > everything is fine with sccp
> >
> > this is only a issue with a sip phone
> >
> > makes no sense to me because PSTN Calling should not have anything to do do
> > with how much bandwith you have. The call is going out its local gateway.
> >
> > Respectfully Charles George
> >
> >
> >
> >> Date: Tue, 4 Aug 2009 11:32:51 -0500
> >> Subject: Re: [OSL | CCIE_Voice] sip aar problem
> >> From: jonv...@gmail.com
> >> To: cisco...@hotmail.com
> >> CC: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
> >>
> >> Are you matching a pattern pointing at a local route group? Is the
> >> local route group down? Is it failing over to a remote RG? Are the CSS
> >> of the SIP and SCCP phones the same?
> >>
> >>
> >> Jonathan
> >>
> >> On Tue, Aug 4, 2009 at 11:08 AM, c george wrote:
> >> > 1st of this is only a Sip Problem
> >> >
> >> > 7961 phone  firmware Sip 41.8-5-2s
> >> > CUCM 7.1.2009
> >> >
> >> > AAR works
> >> > but when I call a PSTN NUMBER by dialing 9+PSTN the call fails stating
> >> > not
> >> > enough Bandwidth.
> >> >
> >> > But AAR works. When dialing a Internal number call reroutes as is
> >> > expected.
> >> > 3000 - 4375553000. Call works, terminates and completes.
> >> >
> >> > Respectfully Charles George
> >> >
> >> >
> >> >
> >> > 
> >> > Get back to school stuff for them and cashback for you. Try BingT now.
> >> > ___
> >> > For more information regarding industry leading CCIE Lab training,
> >> > please
> >> > visit www.ipexpert.com
> >> >
> >> >
> >
> > 
> > Get back to school stuff for them and cashback for you. Try BingT now.



Get back to school stuff for them and cashback for you. Try BingT now.



Get your vacation photos on your phone! Click here.
_
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http://windowsliveformobile.com/en-us/photos/default.aspx?&OCID=0809TL-HM___
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Re: [OSL | CCIE_Voice] sip aar problem

2009-08-08 Thread c george

no i did not

Respectfully Charles George


 


From: mciarfe...@iplogic.com
To: cisco...@hotmail.com; jonv...@gmail.com
CC: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
Date: Sat, 8 Aug 2009 01:47:35 -0400
Subject: RE: [OSL | CCIE_Voice] sip aar problem







Did you ever get this solved?
 
I ran into something maybe similar.  I had the locations stuff configured 
wrong.  My HQ to HQ location also had RSVP mandatory and I was getting out of 
bandwidth messages when dialing, say, 911 from HQ—nothing too important.
 


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of c george
Sent: Tuesday, August 04, 2009 2:28 PM
To: jonv...@gmail.com
Cc: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] sip aar problem
 
let me also add this is a h323 gw
 
and when in a low bandwidth state inbound call fail as well.
 
a debug voip ccapi inout show call fails because ccm is not responding

Respectfully Charles George


 



From: cisco...@hotmail.com
To: jonv...@gmail.com
Date: Tue, 4 Aug 2009 18:18:23 +
CC: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] sip aar problem

great idea!!
 
same results. sip phone 7961 cannot call out dialing a PSTN #. (9+#)
 
But AAR works. internal number shows reroting and dials over the PSTN.
 
is anyone else using a 7961 and checked sip aar for both internal and dialing a 
pstn number directly while in a low bandwidth state ?
 
if it worls, what sip firmware are you using and what ver cucm

Respectfully Charles George


 
> Date: Tue, 4 Aug 2009 12:35:11 -0500
> Subject: Re: [OSL | CCIE_Voice] sip aar problem
> From: jonv...@gmail.com
> To: cisco...@hotmail.com
> CC: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
> 
> OK, put them both in the same CSS and see if the problem goes away
> (use the SCCP CSS for the SIP phone...)
> 
> Then you know it is a RG issue...
> 
> J
> 
> On Tue, Aug 4, 2009 at 11:44 AM, c george wrote:
> > different css for sip and sccp
> >
> > everything is fine with sccp
> >
> > this is only a issue with a sip phone
> >
> > makes no sense to me because PSTN Calling should not have anything to do do
> > with how much bandwith you have. The call is going out its local gateway.
> >
> > Respectfully Charles George
> >
> >
> >
> >> Date: Tue, 4 Aug 2009 11:32:51 -0500
> >> Subject: Re: [OSL | CCIE_Voice] sip aar problem
> >> From: jonv...@gmail.com
> >> To: cisco...@hotmail.com
> >> CC: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
> >>
> >> Are you matching a pattern pointing at a local route group? Is the
> >> local route group down? Is it failing over to a remote RG? Are the CSS
> >> of the SIP and SCCP phones the same?
> >>
> >>
> >> Jonathan
> >>
> >> On Tue, Aug 4, 2009 at 11:08 AM, c george wrote:
> >> > 1st of this is only a Sip Problem
> >> >
> >> > 7961 phone  firmware Sip 41.8-5-2s
> >> > CUCM 7.1.2009
> >> >
> >> > AAR works
> >> > but when I call a PSTN NUMBER by dialing 9+PSTN the call fails stating
> >> > not
> >> > enough Bandwidth.
> >> >
> >> > But AAR works. When dialing a Internal number call reroutes as is
> >> > expected.
> >> > 3000 - 4375553000. Call works, terminates and completes.
> >> >
> >> > Respectfully Charles George
> >> >
> >> >
> >> >
> >> > 
> >> > Get back to school stuff for them and cashback for you. Try BingT now.
> >> > ___
> >> > For more information regarding industry leading CCIE Lab training,
> >> > please
> >> > visit www.ipexpert.com
> >> >
> >> >
> >
> > 
> > Get back to school stuff for them and cashback for you. Try BingT now.



Get back to school stuff for them and cashback for you. Try BingT now.



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_
Get free photo software from Windows Live
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Re: [OSL | CCIE_Voice] Performance of vmware in SSD Harddisk

2009-08-08 Thread Jason Granat
Also, a thought on using VM, if they are indeed running it in the lab
for candidates then I would imagine that there shouldn't be any issues
preventing us from running it successfully, unless they have some
special tweaks that we can't get to :-)

Sent while mobile

On Aug 8, 2009, at 9:02, Jason Granat  wrote:

> I havent tries SSDs, but I tried 15K drives over 7200 and there was a
> noticeable difference, but not worth the $1k cost for 4 of them. I
> went back to 7200. I don't get the lag that you seem to experience. My
> VMs come up and are accessible in a few minutes.  The main lag I was
> trying to overcome is the response of my MS Exchange 07 auto
> attendant. Her voice gets choppy and perfmon shows the drives going
> crazy. The 15k drives improved this quite a bit but I don't have the
> problem with Unity so it wasn't worth it at this point. I run RAID5.
> For performance reference My VM server pull multi duty. Not only is
> this my voice lab server but for customer demos I also run CSAMC
> (Win2k3), CSM (Win2k3), NAC Appliance (1 manager and 1 server), LMS
> (Win2k3), MS Exchange 07, an XP Client, and a Linux server. All of
> these are VMs on the same host box and the only performance issue I
> have is the one noted above with choppy MS Exchange AA.
>
> Sent while mobile
>
> On Aug 8, 2009, at 8:47, Chris Parker  wrote:
>
>> It may help but you really need raid 1+0 to see a big improvement ssd
>> drives are really pricey I'm sure you could get the four drives you'd
>> need for raid for less money
>>
>> Sent from my iPhone
>>
>> On Aug 8, 2009, at 10:54 AM, So Gwaai  wrote:
>>
>>> Hi,
>>>
>>> I'd just want to know if any significance improvement when I move my
>>> vmware based (vmware workstation 6.5) CUC from the SATA harddisk to
>>> SSD harddisk? Since I need to wait more than 10 min to get the GUI
>>> login prompt after the CLI prompt appear...
>>>
>>> Thanks for any help
>>> ___
>>> For more information regarding industry leading CCIE Lab training,
>>> please visit www.ipexpert.com
>> ___
>> For more information regarding industry leading CCIE Lab training,
>> please visit www.ipexpert.com
>
>
>
> http://slash128.com
> ___
> For more information regarding industry leading CCIE Lab training,
> please visit www.ipexpert.com



http://slash128.com
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Re: [OSL | CCIE_Voice] Performance of vmware in SSD Harddisk

2009-08-08 Thread Jason Granat
I havent tries SSDs, but I tried 15K drives over 7200 and there was a
noticeable difference, but not worth the $1k cost for 4 of them. I
went back to 7200. I don't get the lag that you seem to experience. My
VMs come up and are accessible in a few minutes.  The main lag I was
trying to overcome is the response of my MS Exchange 07 auto
attendant. Her voice gets choppy and perfmon shows the drives going
crazy. The 15k drives improved this quite a bit but I don't have the
problem with Unity so it wasn't worth it at this point. I run RAID5.
For performance reference My VM server pull multi duty. Not only is
this my voice lab server but for customer demos I also run CSAMC
(Win2k3), CSM (Win2k3), NAC Appliance (1 manager and 1 server), LMS
(Win2k3), MS Exchange 07, an XP Client, and a Linux server. All of
these are VMs on the same host box and the only performance issue I
have is the one noted above with choppy MS Exchange AA.

Sent while mobile

On Aug 8, 2009, at 8:47, Chris Parker  wrote:

> It may help but you really need raid 1+0 to see a big improvement ssd
> drives are really pricey I'm sure you could get the four drives you'd
> need for raid for less money
>
> Sent from my iPhone
>
> On Aug 8, 2009, at 10:54 AM, So Gwaai  wrote:
>
>> Hi,
>>
>> I'd just want to know if any significance improvement when I move my
>> vmware based (vmware workstation 6.5) CUC from the SATA harddisk to
>> SSD harddisk? Since I need to wait more than 10 min to get the GUI
>> login prompt after the CLI prompt appear...
>>
>> Thanks for any help
>> ___
>> For more information regarding industry leading CCIE Lab training,
>> please visit www.ipexpert.com
> ___
> For more information regarding industry leading CCIE Lab training,
> please visit www.ipexpert.com



http://slash128.com
___
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Re: [OSL | CCIE_Voice] Performance of vmware in SSD Harddisk

2009-08-08 Thread Chris Parker
It may help but you really need raid 1+0 to see a big improvement ssd  
drives are really pricey I'm sure you could get the four drives you'd  
need for raid for less money

Sent from my iPhone

On Aug 8, 2009, at 10:54 AM, So Gwaai  wrote:

> Hi,
>
> I'd just want to know if any significance improvement when I move my  
> vmware based (vmware workstation 6.5) CUC from the SATA harddisk to  
> SSD harddisk? Since I need to wait more than 10 min to get the GUI  
> login prompt after the CLI prompt appear...
>
> Thanks for any help
> ___
> For more information regarding industry leading CCIE Lab training,  
> please visit www.ipexpert.com
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com


Re: [OSL | CCIE_Voice] Performance of vmware in SSD Harddisk

2009-08-08 Thread Tanner Ezell
Unlikely. In my VMware box (which is a Q6600 with 8gb ram and a standard
7200RPM hard disk) it takes maybe 2 minutes to bring UCM up.
How long does it take for the CLI to come up? And are you logging into the
CLI after it does.

As I recall, there is an issue with VMware UCM that prevents anything from
happening until after you log into the CLI.

On Sat, Aug 8, 2009 at 7:54 AM, So Gwaai  wrote:

> Hi,
>
> I'd just want to know if any significance improvement when I move my vmware
> based (vmware workstation 6.5) CUC from the SATA harddisk to SSD harddisk?
> Since I need to wait more than 10 min to get the GUI login prompt after the
> CLI prompt appear...
>
> Thanks for any help
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com


Re: [OSL | CCIE_Voice] My impression of V3

2009-08-08 Thread Arun Kumar
Congratulations... :)


On Fri, Aug 7, 2009 at 11:45 PM, Bill Hatcher  wrote:

>  Congratulations man!!!  Great job.
>
>
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Art Sandborgh
> *Sent:* Friday, August 07, 2009 7:50 PM
> *To:* ccie list
> *Subject:* [OSL | CCIE_Voice] My impression of V3
>
>
>
> All,
>
>
> First - please do not e-mail or contact me in any way to ask for any
> further information about the lab.  I will NOT respond - you've been
> warned.  I take the NDA very seriously and will not violate it.
>
> Second - a little background - this was my attempt #6 - twice on version 1,
> three times on ver 2.  I was not a big fan of V2, but that was my issue
> apparently as other folks seemed to pass it just fine.
>
>
> as to V3 -
>
> I thought it was a very fair (and challenging) test of technical knowledge
> of the UC7 software and hardware as listed on Cisco.com.
>
> I thought the questions were very clearly written and the goals set out
> attainable.
>
> I think that Ben and his team did an OUTSTANDING job re-writing the lab and
> I give them a standing ovation.
>
> There were items on the lab that I did not expect to see and I did leave
> some points on the table because I had not thoroughly enough prepared for
> all aspects of all the solution components (my bad).
>
> I did not get a personalized letter back from any of the proctors, just a
> form letter congratulating me for passing - I guess that will have to do.
>
> I want to thank IPExpert for the outstanding instruction and matrials and
> all of the contributors to this forum for the support along the way.
>
> Good luck to all - it is attainable, and worth it, even if you do have to
> go back again and again.
>
> Cheers!
>
> Art Sandborgh  #25081
>
>
>
>  --
>
> Express your personality in color! Preview and select themes for Hotmail®. Try
> it 
> now.
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
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[OSL | CCIE_Voice] Performance of vmware in SSD Harddisk

2009-08-08 Thread So Gwaai
Hi,

I'd just want to know if any significance improvement when I move my vmware
based (vmware workstation 6.5) CUC from the SATA harddisk to SSD harddisk?
Since I need to wait more than 10 min to get the GUI login prompt after the
CLI prompt appear...

Thanks for any help
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com


Re: [OSL | CCIE_Voice] UCXX 7.0 high Availability ?

2009-08-08 Thread Jonathan Charles
1 - failover is testable, but probably not during the first year...
2 - Go configure CUPS without AD and DNS, see what you can do it
ain't much...
3 - There is no DNS or AD...

None of the boxes support AD or DNS.


Jonathan

On Sat, Aug 8, 2009 at 2:01 AM, Azeem ahamed wrote:
> Hi there
>
> I am not sure whether this was discussed before in the forum but still
> wanted a confirmation for the below as i am building a home lab for voice
>
> 1. Is UCCX 7.0 to be configured for High Availability? In that case do we
> require to instance of UCCX 7.0 running ?
> 2. As per my understanding there is NO DNS as well AD for the ver 3.0, then
> how will we deploy CUPS?
> 3. If there is a windows based DNS as well as AD will it be on one of the UC
> aplliance or a seperate server ?
>
> Thanks
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
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www.ipexpert.com


Re: [OSL | CCIE_Voice] My impression of V3

2009-08-08 Thread Chris Parker
I asked the proctor about rebooting the rack and he said no that they  
do not. However they do verify that when the wan goes down that  
failover works per what the lab asks you to do.


Chris

Sent from my iPhone

On Aug 8, 2009, at 7:09 AM, "Aamir Panjwani" > wrote:



Good luck Sebastien with your result…let us know how you went J







From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice- 
boun...@onlinestudylist.com] On Behalf Of Sebastien Pouplin

Sent: Saturday, 8 August 2009 8:15 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] My impression of V3



Hi Folks,

I took once the V2 exam 3 months ago and failed, and very recently  
took the V3 versionbut did not have the result yet! :)


V2 in my opinion was asking for too many features and some 'basic'  
questions like Unity or IPCC integrations were very time consuming.  
So in the end, you were able to pass only by knowing in advance  
which exact types of questions will be asked...which is difficult to  
guess for your 1st attempt!


V3 is also time consuming but reasonably fair. I saw it much more  
realistic from the Voice Enterprise networking perspective. I could  
finish and complete all the tasks 30 minutes in advance and then  
test my dial plan.


By the way, when taking a CCIE exam I always reboot my rack before  
leaving, but does someone know if proctors reload your rack before  
grading ?


Now finger crossed...:)

Sebastien Pouplin ( CCIE #20409 )

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Re: [OSL | CCIE_Voice] My impression of V3

2009-08-08 Thread Jonathan Charles
Yeah, I too was surprised by how much time they give you... when you
flip through the 27 pages of lab, you think there is no way I am going
to have time for this... but you finish and realize you have 2 or 3
hrs left...

Not enough time to learn the stuff you didn't know, but enough time to
troubleshoot...

And since the four questions only take 3 minutes (30 minutes are way
too long... if you don't know it, you don't know it, and adding an
extra 25 minutes ain't gonna help...) so you get nearly an extra half
hour for the lab...



Jonathan

On Sat, Aug 8, 2009 at 5:14 AM, Sebastien Pouplin wrote:
> Hi Folks,
>
> I took once the V2 exam 3 months ago and failed, and very recently took the
> V3 versionbut did not have the result yet! :)
>
> V2 in my opinion was asking for too many features and some 'basic' questions
> like Unity or IPCC integrations were very time consuming. So in the end, you
> were able to pass only by knowing in advance which exact types of questions
> will be asked...which is difficult to guess for your 1st attempt!
>
> V3 is also time consuming but reasonably fair. I saw it much more realistic
> from the Voice Enterprise networking perspective. I could finish and
> complete all the tasks 30 minutes in advance and then test my dial plan.
>
> By the way, when taking a CCIE exam I always reboot my rack before leaving,
> but does someone know if proctors reload your rack before grading ?
>
> Now finger crossed...:)
>
> Sebastien Pouplin ( CCIE #20409 )
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com


Re: [OSL | CCIE_Voice] CME SRST for SIP phones

2009-08-08 Thread Jonathan Charles
I already know how to do SIP on CME and CUCM... it is no big deal...
however, when you see what you have to do for SIP for SRST and CCME,
it makes you wonder if Cisco really supports it, or it was a
last-minute tack on to the IOS...


J

On Sat, Aug 8, 2009 at 12:36 AM, Aamir
Panjwani wrote:
> Well we can whinge but SIP endpoints all day along, but that still doesn’t
> change the fact we need to study it for the exam…ehehe J
>
>
>
>
>
>
>
> From: ccie_voice-boun...@onlinestudylist.com
> [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Tanner Ezell
> Sent: Saturday, 8 August 2009 3:26 PM
> To: Jonathan Charles
> Cc: OSL Group; Michael Ciarfello
> Subject: Re: [OSL | CCIE_Voice] CME SRST for SIP phones
>
>
>
> Whoops, that's what I get for not reading ahead. Don't mind me :)
>
> On Fri, Aug 7, 2009 at 10:25 PM, Tanner Ezell 
> wrote:
>
> Another great point. A lot of the cool features SIP PBX/endpoints/whatever
> has, are proprietary to that vendor and not interoperable.. seems self
> defeating.
>
>
>
> On Fri, Aug 7, 2009 at 10:21 PM, Jonathan Charles  wrote:
>
> And their toe is being gnawed off, especially when they try to fax over
> it...
>
> You know, SIP might have a chance if there was a governing body saying
> "This is what you need to be able to do to be called 'SIP'... right
> now, just cuz one vendor says 'SIP' on it, doesn't mean that another
> vendor can talk to it...
>
>
> Jonathan
>
> On Fri, Aug 7, 2009 at 11:09 PM, Michael
> Ciarfello wrote:
>> No.  Not for endpoints.
>> Customers are getting into SIP trunking though, but only dipping their toe
>> in the water.
>>
>> There will be many advantages to SIP trunking.  Don't know about putting
>> 1000's SIP phones out.  People used to get caught up in the interoperability
>> of SIP and SIP phones will cost $10, etc.  They used to say vendor A, C, and
>> N have their own proprietary protocls.  But no one has seemed to care.  Put
>> a non-Cisco SIP phone on a CallManager and see what h*** it is to 1)
>> configure, 2), manage, etc.  The huge advantage of Cisco IP Phones (SCCP and
>> SIP) are you take it out of the box and plug it in. You don't have to touch
>> each phone like so many other PBX vendors.
>>
>> NOW, supporting SIP station side seems to be catching on for, say video
>> endpoints and other "weird" devices.  Catching on doesn't mean working well,
>> integrating well, feature rich, etc.
>>
>> Needs more time to simmer on the stove.
>>
>> 
>> From: ccie_voice-boun...@onlinestudylist.com
>> [ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie8340tx
>> [ccie834...@gmail.com]
>> Sent: Friday, August 07, 2009 11:49 PM
>> To: OSL Group
>> Subject: Re: [OSL | CCIE_Voice] CME SRST for SIP phones
>>
>> While we are on the topic of SIP:
>>
>> Has anyone really run into a customer/enterprise with a large SIP
>> implementation? If yes, Why ?
>>
>> Thanks
>> Padhu
>> - Original Message -
>> From: "Jonathan Charles" 
>> To: "Aamir Panjwani" 
>> Cc: 
>> Sent: Friday, August 07, 2009 10:36 PM
>> Subject: Re: [OSL | CCIE_Voice] CME SRST for SIP phones
>>
>>
>> Yep, it works fine... sorta...
>>
>>
>> http://www.cisco.com/en/US/products/sw/voicesw/ps2169/products_configuration_guide_chapter09186a0080557e86.html
>>
>>
>> J
>>
>> On Fri, Aug 7, 2009 at 8:34 PM, Aamir
>> Panjwani wrote:
>>> I can’t find any documentation which indicates whether CME SRST support
>>> SIP
>>> phones or not? I know normal SRST works fine for SIP phones.
>>>
>>>
>>>
>>> Can someone please point me in the right direction
>>>
>>>
>>>
>>> Thanks
>>>
>>>
>>>
>>>
>>>
>>> __
>>> This email has been scanned by the MessageLabs Email Security System.
>>> For more information please visit http://www.messagelabs.com/email
>>> __
>>>
>>> ___
>>> For more information regarding industry leading CCIE Lab training, please
>>> visit www.ipexpert.com
>>>
>>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
>
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Re: [OSL | CCIE_Voice] "Desk phone control" option missing from CUPS application menu

2009-08-08 Thread chikki venu
it is under menu option
 
application > Desktop control > User assingment  > find users
 
select the users >  Assign the user
 
Hope this helps
hapyy labbing
chikkis

--- On Sat, 8/8/09, Mike Thompson  wrote:


From: Mike Thompson 
Subject: Re: [OSL | CCIE_Voice] "Desk phone control" option missing from CUPS 
application menu
To: "jeremy co" 
Cc: "ccie_voice@onlinestudylist.com" 
Date: Saturday, August 8, 2009, 5:55 PM


Verify user - device association at all the appropriate locations

Sent from my phone, apologies for any typos.

On Aug 8, 2009, at 2:52 AM, jeremy co  wrote:

> Hi,
>
> could be very stupid question since I have no exp in CUPS.
>
> anybody can explain to me why this option is missing ?
>
> "Desk phone control" option missing from CUPS application menu
>
> How could I bring it over?
>
>
>
> Jeremy
> ___
> For more information regarding industry leading CCIE Lab training,  
> please visit www.ipexpert.com
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Re: [OSL | CCIE_Voice] "Desk phone control" option missing from CUPS application menu

2009-08-08 Thread Mike Thompson
Verify user - device association at all the appropriate locations

Sent from my phone, apologies for any typos.

On Aug 8, 2009, at 2:52 AM, jeremy co  wrote:

> Hi,
>
> could be very stupid question since I have no exp in CUPS.
>
> anybody can explain to me why this option is missing ?
>
> "Desk phone control" option missing from CUPS application menu
>
> How could I bring it over?
>
>
>
> Jeremy
> ___
> For more information regarding industry leading CCIE Lab training,  
> please visit www.ipexpert.com
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Re: [OSL | CCIE_Voice] UCXX 7.0 high Availability ?

2009-08-08 Thread Mike Thompson
No failover UCCX server from what I understand. The failover of UCCX  
means that it is integrated to pin and sub.

AD / DNS is testable, but won't be done for month (please correct if  
anyone knows anything different).

Some features for CUPS are AD dependant, but many are not.

Sent from my phone, apologies for any typos.

On Aug 8, 2009, at 3:01 AM, Azeem ahamed  wrote:

> Hi there
>
> I am not sure whether this was discussed before in the forum but  
> still wanted a confirmation for the below as i am building a home  
> lab for voice
>
> 1. Is UCCX 7.0 to be configured for High Availability? In that case  
> do we require to instance of UCCX 7.0 running ?
> 2. As per my understanding there is NO DNS as well AD for the ver  
> 3.0, then how will we deploy CUPS?
> 3. If there is a windows based DNS as well as AD will it be on one  
> of the UC aplliance or a seperate server ?
>
> Thanks
> ___
> For more information regarding industry leading CCIE Lab training,  
> please visit www.ipexpert.com
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Re: [OSL | CCIE_Voice] My impression of V3

2009-08-08 Thread Aamir Panjwani
Good luck Sebastien with your result...let us know how you went J

 

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Sebastien
Pouplin
Sent: Saturday, 8 August 2009 8:15 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] My impression of V3

 

Hi Folks,

I took once the V2 exam 3 months ago and failed, and very recently took
the V3 versionbut did not have the result yet! :)

V2 in my opinion was asking for too many features and some 'basic'
questions like Unity or IPCC integrations were very time consuming. So
in the end, you were able to pass only by knowing in advance which exact
types of questions will be asked...which is difficult to guess for your
1st attempt!

V3 is also time consuming but reasonably fair. I saw it much more
realistic from the Voice Enterprise networking perspective. I could
finish and complete all the tasks 30 minutes in advance and then test my
dial plan.

By the way, when taking a CCIE exam I always reboot my rack before
leaving, but does someone know if proctors reload your rack before
grading ? 

Now finger crossed...:)

Sebastien Pouplin ( CCIE #20409 )

__
This email has been scanned by the MessageLabs Email Security System.
For more information please visit http://www.messagelabs.com/email 
__


__
This email has been scanned by the MessageLabs Email Security System.
For more information please visit http://www.messagelabs.com/email 
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Re: [OSL | CCIE_Voice] "Desk phone control" option missing from CUPS application menu

2009-08-08 Thread Ravindra Lakpriya
Hi ,
Can you check the server health of the CUPC client ?? then we  can think of
something

On Sat, Aug 8, 2009 at 12:22 PM, jeremy co  wrote:

> Hi,
> could be very stupid question since I have no exp in CUPS.
>
> anybody can explain to me why this option is missing ?
>
> "Desk phone control" option missing from CUPS application menu
>
> How could I bring it over?
>
>
> Jeremy
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>


-- 
Ravindra Lakpriya
+94 773 532 094
___
For more information regarding industry leading CCIE Lab training, please visit 
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[OSL | CCIE_Voice] My impression of V3

2009-08-08 Thread Sebastien Pouplin
Hi Folks,

I took once the V2 exam 3 months ago and failed, and very recently took the
V3 versionbut did not have the result yet! :)

V2 in my opinion was asking for too many features and some 'basic' questions
like Unity or IPCC integrations were very time consuming. So in the end, you
were able to pass only by knowing in advance which exact types of questions
will be asked...which is difficult to guess for your 1st attempt!

V3 is also time consuming but reasonably fair. I saw it much more realistic
from the Voice Enterprise networking perspective. I could finish and
complete all the tasks 30 minutes in advance and then test my dial plan.

By the way, when taking a CCIE exam I always reboot my rack before leaving,
but does someone know if proctors reload your rack before grading ?

Now finger crossed...:)

Sebastien Pouplin ( CCIE #20409 )
___
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www.ipexpert.com


[OSL | CCIE_Voice] UCXX 7.0 high Availability ?

2009-08-08 Thread Azeem ahamed
Hi there

I am not sure whether this was discussed before in the forum but still
wanted a confirmation for the below as i am building a home lab for voice

1. Is UCCX 7.0 to be configured for High Availability? In that case do we
require to instance of UCCX 7.0 running ?
2. As per my understanding there is NO DNS as well AD for the ver 3.0, then
how will we deploy CUPS?
3. If there is a windows based DNS as well as AD will it be on one of the UC
aplliance or a seperate server ?

Thanks
___
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