Re: [OSL | CCIE_Voice] CCIE Voice Retirement

[Mann Chaddha]

Signed.



On Sun, Jun 2, 2013 at 6:03 PM, wrote:

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> Today's Topics:
>
>1. Re: CCIE Voice Retirement (Bill Lake)
>2. Re: Cisco: Provide a reasonable transition path from CCIE
>   Voice to CCIE Collaboration (m george)
>3. Re: CCIE Voice Retirement (Martin Sloan)
>
>
> --
>
> Message: 1
> Date: Sat, 1 Jun 2013 22:10:03 -0500
> From: Bill Lake 
> To: Martin Sloan 
> Cc: "ccie_voice@onlinestudylist.com" ,
> be...@cisco.com
> Subject: Re: [OSL | CCIE_Voice] CCIE Voice Retirement
> Message-ID:
> <
> cadpb93ofceaf47fknagbp1keksshu3v4+kwz0h4temdq8v-...@mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Dear Ben,
>
> In the last several years I have installed many Cisco Telepresence
> 3200/3000, 1300, 500 and 9200/9000 series systems.  I have done work with
> personal video conferencing systems on executive desktops. I have also been
> involved in many installations of VoIP systems from around 50 to over 3000
> phones at one location. I have also installed more software clients that
> support voice and video than I care to remember. I hope I am the very type
> of person that Cisco wants to earn their CCIE Voice.
>
> I earned that CCIE Voice this year to open doors for me and afford me some
> opportunities that were weren't there before.  Now just a short time after
> earning my CCIE with very relevant skills in today's market, Cisco has
> decided to close the door on CCIE Voice and introduce CCIE Collaboration.
>
> This clearly is within Cisco's right to do and as a name, it better fits
> what communications have evolved into.  That said, the underlying
> technology and ability to work on it have not changes.  Yes video has
> different requirements than voice but once you learn to provide good voice
> quality, you can leverage that to provide good video quality. The same is
> true of almost all of the "new" technologies, from dialing to software
> clients that are introduced in CCIE Collaboration.
>
> So when Cisco has decided in the past to retire a CCIE exam, it was due to
> the massive technology changes that in general left the CCIE track
> untenable.  If Cisco had decided to leave the name CCIE Voice and included
> the new tasks, it would still be relevant as it is with the new name. You
> can not say the same for CCIE ISP Dial, CCIE SNA/IP integration and so on.
> As a technology they are not dead but are completely untenable as a CCIE
> track but . Cisco can not logically take the same stance that Voice is a
> dead technology and is not integral to the new CCIE Collaboration.
>
> Since voice is so integral to the CCIE Collaboration, I would consider it
> to be more a change in name than technology.  In retiring CCIE Voice and
> introducing CCIE Collaboration, Cisco has punished CCIE Voice holders like
> never before. Even with their skills present and relevant to the CCIE
> track, they have been told that the only way to achieve the new CCIE
> Collaboration title is to pass the lab.  This is hard to believe as other
> tracks have changed far more over their lives and especially for those that
> passed CCIE Voice V3.  A perfect example of this is CCIE R&S from the early
> 2000's. During that exam candidates had to earn their strips on
> technologies like token ring, IPX and other similarly dead protocols. They
> are allowed to remain CCIE R&S by passing every 2 years a CCIE level
> written exam, any exam, so they don't even need to prove they are keeping
> current on R&S.  So it seems that Cisco is interested mostly in CCIE's
> keeping current in today's technology and not so much with your CCIE
> track.  That seems completely tossed out the window with CCIE voice. CCIE
> Voice is so integral to CCIE Collaboration that you can't logically argue
> that voice is a dead technology and you must earn your CCIE Collaboration
> by passing another Voice centric lab in the CCIE Collaboration.
>
> It is completely within Cisco's right to demand that anyone pass the CCIE
> Collaboration to earn the title.  It is however with great hope that the
> logical argument laid out here will help Cisco change paths on this and
> offer a different path to current holders of CCIE Voice.  Cisco could
> easily create or use the CCIE Collaboration written exam to ensure that
> people who have earned their CCIE Voice  continue to keep up with the ever
> changing technology.  Cisco cou

Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced

[Mann Chaddha]

I am one of those who cleared the exam 4 months back. And there will be
many who have cleared it in last 6 months whose efforts seem to have gone
in vain (not entirely though). I wonder what would be the state of mind of
guys who have the Lab scheduled in the next 90 days and have made the
payments.

I reckon Cisco will definitely have to think this one over. I believe they
respect the cert and hope that there will an alternative upgrade path for
IE Voice holders to gain Collab IE. Looking at the syllabus, this is
definitely not a game changer but a natural evolution of the voice domain.

I hope sense prevails.

Ciao
Mann Chaddha
CCIE Voice # 37926


On Wed, May 29, 2013 at 11:44 AM, wrote:

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> Today's Topics:
>
>1. Re: CCIE Collaboration officially announced (m george)
>2. Re: CCIE Collaboration officially announced   (Hesham
>   Abdelkereem) (Kamran Ahsanullah)
>3. Re: CCIE Collaboration officially announced (Karen Johnson)
>
>
> --
>
> Message: 1
> Date: Wed, 29 May 2013 09:16:01 +0500
> From: m george 
> To: Vik Malhi 
> Cc: OSL Group 
> Subject: Re: [OSL | CCIE_Voice] CCIE Collaboration officially
> announced
> Message-ID:
>  oybmklx5ysn8vvhgcurrzjr_vn2l_u_w35frdcduuc...@mail.gmail.com>
> Content-Type: text/plain; charset="windows-1252"
>
> This is quite ridiculous ! All other tracks (RS/SP) have gone through
> massive changes but they were retained. Even Security CCIE track has
> recently gone through 50% more overlap (ISE/WSA/ACS/WLC/AP & what not is
> new) but they didn't rename it & retire old one. If you look at CCIE
> Collaboration equipment list & topics, you won't find any significant
> different other than TP/Jabber/InterCluster stuff which is like 15%-20% new
> stuff.  It's so pathetic on cisco's part that they didn't value the years
> hardwork & effort of engineers to attain Voice CCIE. I know guys who sat
> lab like 7 times, some even 10 times to pass. & when they have finally
> passed this extremely tough lab, you are throwing their CCIE number in
> gutter by retiring a CCIE certification.  Will people go for CCIE Voice lab
> now ? Probably NOT & i bet this will be only track for which there won't be
> rush to complete certification.
>
> it's an extremely disappointing thing what Cisco has done. Cisco should
> protect investment made by tens of hundreds of engineers for years rather
> than giving them a retired track.  For a guy who passed lab on 7th attempt
> recently & is a Voice CCIE , will Cisco give him free vouchers 7 times to
> sit Collaboration CCIE now ? Morally , they should. Practically, they
> won't.
>
>  It doesn't make sense to me . Does it make sense to anyone among you ? If
> so, please explain how.
>
> On Wed, May 29, 2013 at 4:08 AM, Vik Malhi  wrote:
>
> > For my initial reaction read here:
> >
> > http://bit.ly/12MNK5t
> >
> >
> > Vik Malhi ? CCIE #13890
> > Managing Partner - IPexpert, Inc.
> >
> > Telephone: +1.810.326.1444 ext 420
> > Fax: +1.810.454.0130
> > Mailto: vma...@ipexpert.com
> >
> >
> >
> >
> > ___
> > For more information regarding industry leading CCIE Lab training, please
> > visit www.ipexpert.com
> >
> > Are you a CCNP or CCIE and looking for a job? Check out
> > www.PlatinumPlacement.com
> >
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> Message: 2
> Date: Wed, 29 May 2013 09:12:40 +0300
> From: Kamran Ahsanullah 
> To: ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] CCIE Collaboration officially
> announced   (Hesham Abdelkereem)
> Message-ID:
>  5dsje8qgqfv...@mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Hesham,
>
> if you have a Voice CCIE already or pass before Feb 2013,you will
> return your CCIE Voice. That will not be taken away from yo

[OSL | CCIE_Voice] At last.. Cleared :)

[Mann Chaddha]

Hey Guys

At last, I am CCIE (V) # 37926. So Relieved. I have learned a lot from all
you here on the forum. You all are true inspiration.

I will come back to the forum once it all settles in.

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

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

[OSL | CCIE_Voice] Help with Unified FX

[Mann Chaddha]

Hi Guys

I am facing some issues with Unified FX application to get Screenshots of
the phones. I can remote manage the phones alright but I am simply not able
to view their screens.

Here is what I have done:
1. Created an app user"*pvadmin*" with Server Monitoring, EM Authentication
& Tab Sync User Groups.
2. Created an end user "*pview*" with Standard CTI Enabled.
3. Associated all phones with pview end user.

Here is the error that I receive when I try to see screenshot:
*Command (Cmd:Screenshot) sent to device (MAC) using thread (0) with
response (XML Error response from phone)
*
I am using the Version 2.1.37 which is the latest one.

Please let me know if anyone has faced similar issues.

Thanks
Mann
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[OSL | CCIE_Voice] PL Servers Issue

[Mann Chaddha]

Hi Guys

Has anyone faced Servers not booting issue in the PL racks?

I am unable to see the servers on the switch and they don;t seem to be
coming up even after I tried to start/revert them from the session page.

Let me know if I am missing something here.

TIA
Mann
___
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Re: [OSL | CCIE_Voice] Prompt recording

[Mann Chaddha]

Hi

The best & the fastest way I have bee using is with the CUC Call Handlers.
Then I grant greeting admin privileges & change customer keypad mappings
for a user to record the same.

Do you see any error whilst looking at the greetings on the CH.

hth
Mann

On Mon, Jul 30, 2012 at 9:30 PM, wrote:

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> Today's Topics:
>
>1. Re: Prompt recording (Ken Wyan)
>
>
> --
>
> Message: 1
> Date: Mon, 30 Jul 2012 17:42:31 +0530
> From: Ken Wyan 
> To: ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] Prompt recording
> Message-ID:
> <
> capbg9bku7prygp3mc52bzc9kds70xp-pv1fslosmyewbdmn...@mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> If question ask to use Unity Connection , then use UC Media Master.
>
> >From candidate PC open UC media master & save your recording.
>
> Now from Candidate PC , open UCCX GUI & upload your recorded prompts.
>
> For all other tasks of UCCX , open UCCX GUI from UCCX server ( first RDP to
> UCCX server ) -- for best browser compatibility.
>
> HTH
>
>
>
> >   On Jul 29, 2012, at 9:27 PM, "Randall Crumm" 
> wrote:
> >
> >   Hello,
> > I recorded a greeting on a call handler. Then went to download it to my
> > computer, but the download menu item is grayed out.
> >
> > How can I download a greeting file(.wav) then upload it to UCCX
> >
> > Thanks,
> >
> > Cheers,
> > Randall
> >
> >  ___
> > For more information regarding industry leading CCIE Lab training, please
> > visit www.ipexpert.com
> >
> > Are you a CCNP or CCIE and looking for a job? Check out
> > www.PlatinumPlacement.com 
> >
> >
> > ___
> > For more information regarding industry leading CCIE Lab training, please
> > visit www.ipexpert.com
> >
> > Are you a CCNP or CCIE and looking for a job? Check out
> > www.PlatinumPlacement.com 
> >
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> End of CCIE_Voice Digest, Vol 77, Issue 53
> **
>
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Re: [OSL | CCIE_Voice] PLUS dial issue

[Mann Chaddha]

Hi Dan

Can you check the pre dots & digit manips on your \+ RP. I see you have
\+.1 RP. & you are dropping +.

Thanks
Mann

On Sat, Jul 7, 2012 at 9:30 PM, wrote:

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> Today's Topics:
>
>1. Re: Fwd: PLUS dial issue (Dan Quinlan (daquinla))
>
>
> --
>
> Message: 1
> Date: Sat, 7 Jul 2012 03:09:31 +
> From: "Dan Quinlan (daquinla)" 
> To: donny f 
> Cc: "ccie_voice@onlinestudylist.com" 
> Subject: Re: [OSL | CCIE_Voice] Fwd: PLUS dial issue
> Message-ID: 
> Content-Type: text/plain; charset="us-ascii"
>
> Are you doing manipulation at the Route Pattern level, at the Route List /
> Group level, or both?  If you are doing both, the Route List / Group
> manipulation takes precedence over the Route Pattern manipulation. Any
> manipulation at the Route Pattern will display on the phone, but if you
> also have manipulation at the Route List / Group, the Route List / Group
> manipulation will be done on the original dialed number ( +1909222 ).
>  If you are reusing a Route List, this might be the cause. The debug isdn
> q931 on the mgcp gateway should show what number is actually being passed
> in the DNIS information.
>
> DQ
> d...@cisco.com
>
> On Jul 4, 2012, at 8:29 PM, "donny f"  f.faraday...@gmail.com>> wrote:
>
> my missed call is :  +1909222.  And From DNA is show use the Route
> Pattern : \+.1!
>
> It should send to MGCP GW as 1909222, but did not.  (all Partition
> and CSS is good, verify with  DNA)
>
>
> When I tried to make normal call using 91909222, it send to MGCP GW as
>  1909222
>
>
>
> On Wed, Jul 4, 2012 at 5:28 PM, Bill Lake  whl...@gmail.com>> wrote:
> If your missed call is 9 digits it will not complete a 10 digit call.  You
> need to figure out when the number is being changed.
>
> On Wed, Jul 4, 2012 at 3:53 PM, donny f  f.faraday...@gmail.com>> wrote:
>
>
> -- Forwarded message --
> From: donny f mailto:f.faraday...@gmail.com>>
> Date: Wed, Jul 4, 2012 at 2:52 PM
> Subject: Re: [OSL | CCIE_Voice] PLUS dial issue
> To: Randall Crumm mailto:rrcr...@yahoo.com>>
>
>
> all,
>
> manipulation is performed correcly, because  RP : 91[2-9]x  , I
> also use "predot " and it works.
> and this is LD call, not Local call
>
> Any other possibilities?
>
> d
>
> On Wed, Jul 4, 2012 at 12:53 PM, Randall Crumm  rrcr...@yahoo.com>> wrote:
> Is that a valid number? looks like it is missing a digit and is it a local
> call since you are stripping the 1
> or should the 1 be sent?
>
>
> Cheers,
> Randall
>
> 
> From: donny f mailto:f.faraday...@gmail.com>>
> To: ccie_voice@onlinestudylist.com
> Sent: Wednesday, July 4, 2012 11:37 AM
> Subject: [OSL | CCIE_Voice] PLUS dial issue
>
> hi all,
>
>  I configure RP : \+1.!   , pt : pt-plus
>
> And have Phone  3001 , CSS_Plus (pt-plus) ti dial "MISSED CALL"
>  +1909345677
>
> - DNA check show it using the RP \+1.! , but the call failed and got
> message "Your call can't be completed as dialed, please consult your
> directory"
>
> Anybody know what missed ?I am using Cisco 7945.
>
> d
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
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> End of CCIE_Voice Digest, Vol 77, Issue 11
> *

[OSL | CCIE_Voice] NTP Master Dilemma

[Mann Chaddha]

Hi Experts

I am due for my next Lab attempt in 5 days time. I so far have never been
able to get marks for NTP section.

So I ask this question to the forum that if someone has ever scored 100% in
the Infra Section (without breaking the NDA of course) , can they lend some
light on this innocuous looking command, ntp master.

Here are some finding from my POD: [10.10.100.1 = PSTN RTR & 10.10.110.1=HQ
RTR]
1. With NTP Master Command

R1:
ntp source Loopback0
ntp master 10
ntp server 10.10.100.1 source Loopback0
!
clock timezone PST -8
clock summer-time PDT recurring

R2:
ntp server 10.10.110.1

Show commands:
R1:
HQ-R1#sh ntp status
Clock is unsynchronized, stratum 16, no reference clock
nominal freq is 250. Hz, actual freq is 250.0008 Hz, precision is 2**24
reference time is D37C1C18.AAB00668 (23:37:12.666 PDT Thu Jun 7 2012)
clock offset is 0. msec, root delay is 0.00 msec
root dispersion is 0.00 msec, peer dispersion is 0.00 msec
loopfilter state is 'CTRL' (Normal Controlled Loop), drift is -0.03145
s/s
system poll interval is 64, last update was 384 sec ago.
HQ-R1#
HQ-R1#
HQ-R1#sh ntp ass
HQ-R1#sh ntp associations

  address ref clock   st   when   poll reach  delay  offset
disp
x~127.127.1.1 .LOCL.   9 16 16   377  0.000   0.000
0.240
x~10.10.100.1.LOCL.   1 50 64   377  0.000 13194.1
3.293
 * sys.peer, # selected, + candidate, - outlyer, x falseticker, ~ configured


R2:
BR1-R2#sh ntp status
Clock is unsynchronized, stratum 16, no reference clock
nominal freq is 250. Hz, actual freq is 249.9990 Hz, precision is 2**24
reference time is . (16:00:00.000 PST Wed Dec 31 1899)
clock offset is 0. msec, root delay is 0.00 msec
root dispersion is 0.01 msec, peer dispersion is 0.00 msec
loopfilter state is 'CTRL' (Normal Controlled Loop), drift is 0.03885
s/s
system poll interval is 64, never updated.
BR1-R2#
BR1-R2#sh ntp ass

  address ref clock   st   when   poll reach  delay  offset
disp
 ~10.10.110.173.78.73.84 16 55 64   300  0.000 1325166
944.35
 * sys.peer, # selected, + candidate, - outlyer, x falseticker, ~ configured

CUCM:
admin:utils ntp status
ntpd (pid 6322) is running...

 remote   refid  st t when poll reach   delay   offset
jitter
==
*127.127.1.0 LOCAL(0)10 l   63   64  3770.0000.000
0.001
 10.10.110.1.INIT.  16 u   64  128  3701.897   -2.990
1087693


synchronised to local net at stratum 11
   time correct to within 12 ms
   polling server every 512 s

Current time in UTC is : Fri Jun  8 06:22:22 UTC 2012
Current time in America/Los_Angeles is : Thu Jun  7 23:22:22 PDT 2012



--

Scenario 2: Without NTP Master Command.
--
R1:
ntp source Loopback0
ntp server 10.10.100.1 source Loopback0
!
clock timezone PST -8
clock summer-time PDT recurring

R2:
BR1-R2#r | s ntp
ntp server 10.10.110.1

Show Commands:

HQ-R1#sh ntp status
Clock is synchronized, stratum 2, reference is 10.10.100.1
nominal freq is 250. Hz, actual freq is 250.0008 Hz, precision is 2**24
reference time is D37C21AB.D80D6753 (00:00:59.843 PDT Fri Jun 8 2012)
clock offset is -0.0078 msec, root delay is 0.00 msec
root dispersion is 0.01 msec, peer dispersion is 0.00 msec
loopfilter state is 'CTRL' (Normal Controlled Loop), drift is -0.03182
s/s
system poll interval is 64, last update was 264 sec ago.
HQ-R1#

HQ-R1#sh ntp ass

  address ref clock   st   when   poll reach  delay  offset
disp
*~10.10.100.1.LOCL.   1 11 64   377  0.000  -7.866
2.813
 * sys.peer, # selected, + candidate, - outlyer, x falseticker, ~ configured

HQ-R1#sh ntp ass d
10.10.100.1 configured, our_master, sane, valid, stratum 1
ref ID .LOCL., time D37C229E.73B36D17 (00:05:02.451 PDT Fri Jun 8 2012)
our mode client, peer mode server, our poll intvl 64, peer poll intvl 64
root delay 0.00 msec, root disp 0.47, reach 377, sync dist 0.00
delay 0.00 msec, offset -7.8662 msec, dispersion 2.81
precision 2**24, version 4
org time D37C22AE.D47DF91B (00:05:18.830 PDT Fri Jun 8 2012)
rec time D37C22AE.D7653258 (00:05:18.841 PDT Fri Jun 8 2012)
xmt time D37C22AE.D6E2248F (00:05:18.839 PDT Fri Jun 8 2012)
filtdelay = 0.000.000.000.000.000.000.000.00
filtoffset =   -0.01   -0.00   -0.00   -0.00   -0.00   -0.00   -0.00   -0.00
filterror = 0.000.000.000.000.000.000.000.00
minpoll = 6, maxpoll = 10


BR1-R2#sh ntp s
Clock is synchronized, stratum 3, reference is 10.10.110.1
nominal freq is 250. Hz, actual freq is 249.9990 Hz, precision is 2**24
reference time is D37C2310.A8751D42 (00:06:56.658 PDT Fri Jun 8 2012)
clock offset is 0.0034 msec, root delay is 0.00 msec
root dispersion i

[OSL | CCIE_Voice] MVA Behavior Question

[Mann Chaddha]

Hi Guys

Is it expected to receive a fast busy when you dial the DN from the Remote
Destination(RD) number?

I can dial other numbers but when I dial the number associated with RD, I
get a Fast busy. MVA otherwise is working fine. Also when I remove the Line
Association for the RD, calls start working again.

Just want to confirm if this is the expected behavior and is there any way
to avoid it.

Thanks
Mann
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[OSL | CCIE_Voice] the directories button display

[Mann Chaddha]

Hi All

I see that Cisco Data Dictionary for 7.0 defines the Priority Field as
follows:
"Priority from low to high defines where in a list a Service should appear:
1 = top of list, 50 (default) = middle, 100 = bottom"

I ran the following commands after deleting the Services from the CUCM
Admin,
run sql insert into telecasterservice
(pkid,Name,NameASCII,Description,URLTemplate,tkPhoneService,EnterpriseSubscription,Priority)
values('d0059763-cdcc-4be7-a2a8-bbd4aac73f63','Missed Calls','Missed
Calls','Missed Calls','Application:Cisco/MissedCalls',1,'f',*1*)

run sql insert into telecasterservice
(pkid,Name,NameASCII,Description,URLTemplate,tkPhoneService,EnterpriseSubscription,Priority)
values('a0eed443-c705-4232-86d4-957295dd339c','Placed Calls','Placed
Calls','Placed Calls','Application:Cisco/PlacedCalls',1,'f'*,2*)

run sql insert into telecasterservice
(pkid,Name,NameASCII,Description,URLTemplate,tkPhoneService,EnterpriseSubscription,Priority)
values('0061bdd2-26c0-46a4-98a3-48a6878edf53','Received Calls','Received
Calls','Received Calls','Application:Cisco/ReceivedCalls',1,'f',*3*)

run sql insert into telecasterservice
(pkid,Name,NameASCII,Description,URLTemplate,tkPhoneService,EnterpriseSubscription,Priority)
values('7eca2cf1-0c8d-4df4-a807-124b18fe89a4','Corporate
Directory','Corporate Directory','Corporate
Directory','Application:Cisco/CorporateDirectory',1,'f',*100*)

And here is the ouotput from the select command for the table:
admin:run sql select * from telecasterservice
pkid name
urltemplate  description
nameascii   tkphoneservice vendor version enterprisesubscription
enabled priority tkphoneservicecategory
 ===
 ===
=== == == === ==
===  ==
d0059763-cdcc-4be7-a2a8-bbd4aac73f63 Missed Calls
Application:Cisco/MissedCallsMissed CallsMissed
Calls1 f  t
50   0
a0eed443-c705-4232-86d4-957295dd339c Placed Calls
Application:Cisco/PlacedCallsPlaced CallsPlaced
Calls1 f  t
50   0
0061bdd2-26c0-46a4-98a3-48a6878edf53 Received Calls
Application:Cisco/ReceivedCalls  Received Calls  Received
Calls  1 f  t
50   0
7eca2cf1-0c8d-4df4-a807-124b18fe89a4 Corporate Directory
Application:Cisco/CorporateDirectory Corporate Directory Corporate
Directory 1 f  t
100  0
ca69f2e4-d088-47f8-acb2-ceea6722272e Voicemail
Application:Cisco/Voicemail  Voicemail
Voicemail   2 t
t   10
admin:


Still after re-subscribing the services to the phones, Corp Dir always
shows up at the top.

If anyone has been able to resolve this issue, please comment.


Thanks
Mann
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[OSL | CCIE_Voice] TFTP Issue on Hardware VPN to ProctorLabs

[Mann Chaddha]

Hi Guys

I am facing TFTP Timeout issue on my phones while using the Hardware VPN to
PL. I am using 7962 Phones and my IPSec session is established too. I have
tried to restart the CUCM TFTP Services, tried registering to the Router
TFTP (for PSTN Phone) but still no luck.

Is there anything required to allow TFTP traffic via the tunnel?

Thanks
Mann
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[OSL | CCIE_Voice] Question on Join Across Lines

[Mann Chaddha]

Hi Guys

Does anyone know what construct JAL uses while bridging 2 calls on different
Line Buttons?

I ask as I need to plan India specific dial plan which shall restrict
bridging of VoIP Calls to Local PSTN Calls. I went through Geolocations but
so far am not too convinced with its usability as a well constructed dial
plan shall never zero in on 2 IP Endpoints which are not allowed to converse
with each other in the first place.

Do advise.

Thanks
Mann
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Re: [OSL | CCIE_Voice] Passed CCIE#28970!!!!!!!

[Mann Chaddha]

Rogers

Mighty congrats on getting the number.

Also a relief to hear that people do clear from Bangalore. I have had 2 bad
stints :).

Mann

On Fri, May 20, 2011 at 9:30 PM, wrote:

> Send CCIE_Voice mailing list submissions to
>ccie_voice@onlinestudylist.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
>http://onlinestudylist.com/mailman/listinfo/ccie_voice
> or, via email, send a message with subject or body 'help' to
>ccie_voice-requ...@onlinestudylist.com
>
> You can reach the person managing the list at
>ccie_voice-ow...@onlinestudylist.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of CCIE_Voice digest..."
>
>
> Today's Topics:
>
>   1. MVA with Hairpinning (Michael Luo)
>   2. Re: Passed CCIE#28970!!! (Michael Luo)
>   3. Re: MVA with Hairpinning (George Goglidze)
>   4. Re: Passed CCIE#28970!!! (Hough, Earl)
>
>
> --
>
> Message: 1
> Date: Fri, 20 May 2011 10:09:54 -0500
> From: Michael Luo 
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] MVA with Hairpinning
> Message-ID: 
> Content-Type: text/plain; charset="iso-8859-1"
>
> I'm testing MVA with Hairpinning (
>
> http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmfeat/fsmobmgr.html#wp1126351
> )
>
> Call flow as below:
>
> PSTN Phone -> MGCP GW -> CUCM -> H323 GW -> MVA application
>
> I have no problem getting it worked if everything was in the same region
> (G711u was used).
>
> But if I put gateways into different region and try to use G.729.  The
> incoming call from PSTN just siliently disconnected.  I was expecting to
> hear the MVA welcome prompts.
>
> Where I can find more documents regarding codec handling in MVA hairpinning
> scenario?  I'd like to know if the VXML app on CUCM accepts G.729 or not.
> If not, how do we configure codec in incoming and outgoing dial-peers?
>
> H323 GW configuration as below.  2888 is the incoming DID.  2999 is the MVA
> number.
>
> dial-peer voice 2888 voip
>  service mva
>  incoming called-number 2888
>  dtmf-relay h245-alphanumeric
>  codec g711ulaw
>  no vad
> !
> dial-peer voice 2999 voip
>  destination-pattern 2999
>  session target ipv4:10.10.210.10 <-- CUCM Publisher
>  dtmf-relay h245-alphanumeric
>  codec g711ulaw
>  no vad
>
> voice service voip
>  allow h323 to h323
>
> application
>  service mva http://10.10.210.10:8080/ccmivr/pages/IVRMainpage.vxml
>
> Thanks!
> -- next part --
> An HTML attachment was scrubbed...
> URL:
> 
>
> --
>
> Message: 2
> Date: Fri, 20 May 2011 10:11:47 -0500
> From: Michael Luo 
> To: Rogers Ochieng 
> Cc: ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] Passed CCIE#28970!!!
> Message-ID: 
> Content-Type: text/plain; charset="iso-8859-1"
>
> Rogers,
>
> Congratulations!
>
> Very nice number by the way.  :)
>
> Michael
>
> On Fri, May 20, 2011 at 1:38 AM, Rogers Ochieng  >wrote:
>
> > Freshly minted! I took my exam yesterday in Bangalore and the good news
> is
> > here!!!
> >
> > Thanks to my study partners Michael, Fatai and Rahul!
> >
> > CCIE#28970 - Voice
> > Rogers Ochieng
> >
> > ___
> > For more information regarding industry leading CCIE Lab training, please
> > visit www.ipexpert.com
> >
> > Are you a CCNP or CCIE and looking for a job? Check out
> > www.PlatinumPlacement.com
> >
> -- next part --
> An HTML attachment was scrubbed...
> URL:
> 
>
> --
>
> Message: 3
> Date: Fri, 20 May 2011 16:34:41 +0100
> From: George Goglidze 
> To: Michael Luo 
> Cc: ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] MVA with Hairpinning
> Message-ID: 
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi Michael,
>
> Have you tried to put g729 on the dial-peer?
> Because if the h323 Gateway and MGCP gateway are in different regions and
> use g729 between them, and if on incoming dial-peer on the h323 gateway
> you're trying to negotiate g711, the negotiation will fail.
>
> Try putting:
>
> voice class codec 1
>  codec preference 1 g711u
>  codec preference 2 g729r
>
> dial-peer voice 2888 voip
>  service mva
>  incoming called-number 2888
>  dtmf-relay h245-alphanumeric
>  voice-class codec 1
>  no vad
> !
> dial-peer voice 2999 voip
>  destination-pattern 2999
>  session target ipv4:10.10.210.10 <-- CUCM Publisher
>  dtmf-relay h245-alphanumeric
>  voice-class codec 1
>  no vad
>
> Let me know if it works, as g729 should definitely work with MVA.
>
> Regards,
>
> On Fri, May 20, 2011 at 4:09 PM, Michael Luo  wrote:
>
> > I'm testing MVA with Hairpinning (
> >
> http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmfeat/fsmobmgr.html#wp1126351
> > )
> >
> > Call flow as below:
> >
> > PSTN Phone -> MGCP GW -> CUCM -> H323 GW -> MVA application
> >
> > I have no probl

[OSL | CCIE_Voice] Last Redirected Party vs Original Called Party

[Mann Chaddha]

Hi All

I have a requirement wherein I would like to present my VM System (Octel)
with the "Last Redirected Party" instead of the "Original Called Party" from
CUCM. By default CUCM sends the original called party in the RDNIS to the
VM. I know that its possible to manipulate this via Translation Profiles in
IOS.

But is there a way I can do this at per DN basis (not system-wide) in CUCM
itself?

TIA
Mann
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Re: [OSL | CCIE_Voice] Finally succeeded ..Got CCIE

[Mann Chaddha]

Congrat Srini. Its a grand achievement. Especially for all of us who are
striving for our numbers too.

Do share your experience with all of us.

Have a good day.


On Wed, May 4, 2011 at 7:53 PM, wrote:

> Send CCIE_Voice mailing list submissions to
>ccie_voice@onlinestudylist.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
>http://onlinestudylist.com/mailman/listinfo/ccie_voice
> or, via email, send a message with subject or body 'help' to
>ccie_voice-requ...@onlinestudylist.com
>
> You can reach the person managing the list at
>ccie_voice-ow...@onlinestudylist.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of CCIE_Voice digest..."
>
>
> Today's Topics:
>
>   1. Re: Finally succeeded ..Got CCIE (Bill Lake)
>   2. Re: Finally succeeded ..Got CCIE (Steve Denney (stdenney))
>   3. Re: Remote Gatekeeper Troubleshooting (Naoufal Kerboute)
>   4. Re: Finally succeeded ..Got CCIE (ccieid1ot)
>
>
> --
>
> Message: 1
> Date: Wed, 4 May 2011 05:53:38 -0500
> From: Bill Lake 
> To: Shrini 
> Cc: "ccie_voice@onlinestudylist.com" 
> Subject: Re: [OSL | CCIE_Voice] Finally succeeded ..Got CCIE
> Message-ID: 
> Content-Type: text/plain; charset=ISO-8859-1
>
> Congratulations, you earned it
>
> On Wed, May 4, 2011 at 3:02 AM, Shrini  wrote:
> > Hi Experts,
> >
> > Today I am officially announced as Voice CCIE.
> >
> > Thanks to one and all for your valuable suggestions and help throughout
> this
> > journey.
> >
> > Special thanks to Vik and IP Expert team for hosting this excellent
> mailer
> > list and helping us.
> >
> > Thanks again
> > Shrini
> >
> > ___
> > For more information regarding industry leading CCIE Lab training, please
> > visit www.ipexpert.com
> >
> > Are you a CCNP or CCIE and looking for a job? Check out
> > www.PlatinumPlacement.com
> >
>
>
> --
>
> Message: 2
> Date: Wed, 4 May 2011 07:11:49 -0500
> From: "Steve Denney (stdenney)" 
> To: "Shrini" , 
> Subject: Re: [OSL | CCIE_Voice] Finally succeeded ..Got CCIE
> Message-ID:
>
> Content-Type: text/plain; charset="us-ascii"
>
> Congrats Shrini!
>
>
>
> cheers, sd
>
>
>
> From: ccie_voice-boun...@onlinestudylist.com
> [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Shrini
> Sent: Wednesday, May 04, 2011 4:02 AM
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] Finally succeeded ..Got CCIE
>
>
>
> Hi Experts,
>
> Today I am officially announced as Voice CCIE.
>
> Thanks to one and all for your valuable suggestions and help throughout
> this journey.
>
> Special thanks to Vik and IP Expert team for hosting this excellent
> mailer list and helping us.
>
> Thanks again
> Shrini
>
> -- next part --
> An HTML attachment was scrubbed...
> URL:
> 
>
> --
>
> Message: 3
> Date: Wed, 4 May 2011 14:01:59 +
> From: Naoufal Kerboute 
> To: "ccie_voice@onlinestudylist.com" 
> Subject: Re: [OSL | CCIE_Voice] Remote Gatekeeper Troubleshooting
> Message-ID:
>
> Content-Type: text/plain; charset="us-ascii"
>
> No reply :s
>
> From: ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] On Behalf Of Naoufal Kerboute
> Sent: Wednesday, May 04, 2011 1:01 PM
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] Remote Gatekeeper Troubleshooting
>
> Hi guys,
>
> I feel that this mailing list is sleeping :)
> Any way I have a question and I hope that I get an answer from you guys.
> I'm playing with gatekeeper and I'm thinking of different way to
> troubleshoot the connection between my local gatekeeper and a remote one (I
> don't have access to remote GK). What is the best way to troubleshoot a
> broken connection between two gatekeeper?
> How can I check from debugs which prefix or default technology prefix the
> remote GK is using, or the bandwidth defined in the remote GK, let assume
> that the remote GK enable only g729 and I'm sending the call with g711, from
> the debug I'll see BRJ_INSUFFICIENT_RSC but how can I know which bandwidth
> is allocated for me so I can define a proper codec information.
>
> Waiting for your feedbacks
>
> Thanks a lot
> Naoufal
>
>
>
> *
> * This Communication is Private & Confidential. This message and any
> attachments may contain information that is privileged and / or confidential
> and is the property of MHD InfoTech LLC. *
> * It is intended solely for the person to whom it is addressed. If you are
> not the intended recipient, you are hereby notified that you are not
> authorized to read, print, retain copy, disseminate, distribute, or *

Re: [OSL | CCIE_Voice] UCM to CME via CUBE

[Mann Chaddha]

Sorry folks.

I got it working after I put the following commands on my CUBE:
!
voice service voip
 h323
  emptycapability
  h245 passthru tcsnonstd-passthru
!

Actually I found the reference in SRND on page 372.

Hth
Mann

On Thu, Apr 21, 2011 at 5:02 PM, Mann Chaddha wrote:

> 1 more thing.
>
> on the HQ GK Trunk:
> 1. MTP is enabled.
> 2. Wait for Far End H.245 Terminal Capability Set is UNCHECKED.
>
> Thanks
>
> On Thu, Apr 21, 2011 at 5:01 PM, Mann Chaddha wrote:
>
>> Hi All
>>
>> I am facing a strange issue fro the Vol2 Lab1. I have done this earlier
>> but today I'm observing a strange issues, where my CME phone stops receiving
>> & sending any RTP packets when supplementary services are invoked at the HQ
>> Phone.
>>
>> Call Flow:
>> HQ Phone --- GK Trunk (has MTP Enabled) --- HQ-RTR MTP --- HQ-CUBE 
>> HQ-XCoder (Registered locally to telephony-ser)    Br2 CME RTR --- CME
>> Phone.
>>
>> Supplementary services at CME phones work. I can hold unhold the call from
>> CME Phone (3002). But when I do the same from HQ Phone (5002), CME phone
>> stops getting & sending RTP.
>>
>> I have put the following commands on the HQ -RTR:
>> voice service voip
>>  allow-connections h323 to h323
>>  allow-connections h323 to sip
>>  allow-connections sip to h323
>>  supplementary-service h450.12
>>  h323
>>   emptycapability
>>   h225 id-passthru
>>   h225 connect-passthru
>>   h245 passthru tcsnonstd-passthru
>> !
>> !
>> gatekeeper
>>  zone local UCM cisco.com 177.1.254.1
>>  zone local VIA-ZONE cisco.com
>>  zone local UCME cisco.com outvia VIA-ZONE
>>  zone prefix UCM 1*
>>  zone prefix UCME 3*
>>  zone prefix UCM 5*
>>  gw-type-prefix 1#* default-technology
>>  bandwidth interzone zone UCM 32
>>  no shutdown
>> !
>> dial-peer voice 3000 voip
>>  incoming called-number 3...$
>>  dtmf-relay h245-alphanumeric
>>  no vad
>> !
>> dial-peer voice 3001 voip
>>  destination-pattern 3...$
>>  session target ras
>>  dtmf-relay h245-alphanumeric
>>  codec g711ulaw
>>  no vad
>>
>>
>> Here are my XCoder & MTP resources registration status:
>> HQ-R1#sh sccp
>> SCCP Admin State: UP
>> Gateway Local Interface: Loopback0
>> IPv4 Address: 177.1.254.1
>> Port Number: 2000
>> IP Precedence: 5
>> User Masked Codec list: None
>> Call Manager: 177.1.10.10, Port Number: 2000
>> Priority: N/A, Version: 7.0, Identifier: 3
>> Trustpoint: N/A
>> Call Manager: 177.1.10.20, Port Number: 2000
>> Priority: N/A, Version: 7.0, Identifier: 2
>> Trustpoint: N/A
>> Call Manager: 177.1.254.1, Port Number: 2000
>> Priority: N/A, Version: 7.0, Identifier: 1
>> Trustpoint: N/A
>>
>> Transcoding Oper State: ACTIVE - Cause Code: NONE
>> Active Call Manager: 177.1.254.1, Port Number: 2000
>> TCP Link Status: CONNECTED, Profile Identifier: 1
>> Reported Max Streams: 8, Reported Max OOS Streams: 0
>> Supported Codec: g711ulaw, Maximum Packetization Period: 30
>> Supported Codec: g711alaw, Maximum Packetization Period: 30
>> Supported Codec: g729ar8, Maximum Packetization Period: 60
>> Supported Codec: g729abr8, Maximum Packetization Period: 60
>> Supported Codec: g729r8, Maximum Packetization Period: 60
>> Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
>> Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
>> Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization
>> Period: 30
>>
>> MTP Oper State: ACTIVE - Cause Code: NONE
>> Active Call Manager: 177.1.10.20, Port Number: 2000
>> TCP Link Status: CONNECTED, Profile Identifier: 2
>> Reported Max Streams: 20, Reported Max OOS Streams: 0
>> Supported Codec: g729r8, Maximum Packetization Period: 60
>> Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
>> Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
>> Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization
>> Period: 30
>>
>>
>>
>> Any ideas if I'm missing something here.
>>
>> Thanks
>> Mann
>>
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
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Are you a CCNP or CCIE and looking for a job? Check out 
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Re: [OSL | CCIE_Voice] UCM to CME via CUBE

[Mann Chaddha]

1 more thing.

on the HQ GK Trunk:
1. MTP is enabled.
2. Wait for Far End H.245 Terminal Capability Set is UNCHECKED.

Thanks

On Thu, Apr 21, 2011 at 5:01 PM, Mann Chaddha wrote:

> Hi All
>
> I am facing a strange issue fro the Vol2 Lab1. I have done this earlier but
> today I'm observing a strange issues, where my CME phone stops receiving &
> sending any RTP packets when supplementary services are invoked at the HQ
> Phone.
>
> Call Flow:
> HQ Phone --- GK Trunk (has MTP Enabled) --- HQ-RTR MTP --- HQ-CUBE 
> HQ-XCoder (Registered locally to telephony-ser)    Br2 CME RTR --- CME
> Phone.
>
> Supplementary services at CME phones work. I can hold unhold the call from
> CME Phone (3002). But when I do the same from HQ Phone (5002), CME phone
> stops getting & sending RTP.
>
> I have put the following commands on the HQ -RTR:
> voice service voip
>  allow-connections h323 to h323
>  allow-connections h323 to sip
>  allow-connections sip to h323
>  supplementary-service h450.12
>  h323
>   emptycapability
>   h225 id-passthru
>   h225 connect-passthru
>   h245 passthru tcsnonstd-passthru
> !
> !
> gatekeeper
>  zone local UCM cisco.com 177.1.254.1
>  zone local VIA-ZONE cisco.com
>  zone local UCME cisco.com outvia VIA-ZONE
>  zone prefix UCM 1*
>  zone prefix UCME 3*
>  zone prefix UCM 5*
>  gw-type-prefix 1#* default-technology
>  bandwidth interzone zone UCM 32
>  no shutdown
> !
> dial-peer voice 3000 voip
>  incoming called-number 3...$
>  dtmf-relay h245-alphanumeric
>  no vad
> !
> dial-peer voice 3001 voip
>  destination-pattern 3...$
>  session target ras
>  dtmf-relay h245-alphanumeric
>  codec g711ulaw
>  no vad
>
>
> Here are my XCoder & MTP resources registration status:
> HQ-R1#sh sccp
> SCCP Admin State: UP
> Gateway Local Interface: Loopback0
> IPv4 Address: 177.1.254.1
> Port Number: 2000
> IP Precedence: 5
> User Masked Codec list: None
> Call Manager: 177.1.10.10, Port Number: 2000
> Priority: N/A, Version: 7.0, Identifier: 3
> Trustpoint: N/A
> Call Manager: 177.1.10.20, Port Number: 2000
> Priority: N/A, Version: 7.0, Identifier: 2
> Trustpoint: N/A
> Call Manager: 177.1.254.1, Port Number: 2000
> Priority: N/A, Version: 7.0, Identifier: 1
> Trustpoint: N/A
>
> Transcoding Oper State: ACTIVE - Cause Code: NONE
> Active Call Manager: 177.1.254.1, Port Number: 2000
> TCP Link Status: CONNECTED, Profile Identifier: 1
> Reported Max Streams: 8, Reported Max OOS Streams: 0
> Supported Codec: g711ulaw, Maximum Packetization Period: 30
> Supported Codec: g711alaw, Maximum Packetization Period: 30
> Supported Codec: g729ar8, Maximum Packetization Period: 60
> Supported Codec: g729abr8, Maximum Packetization Period: 60
> Supported Codec: g729r8, Maximum Packetization Period: 60
> Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
> Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
> Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization
> Period: 30
>
> MTP Oper State: ACTIVE - Cause Code: NONE
> Active Call Manager: 177.1.10.20, Port Number: 2000
> TCP Link Status: CONNECTED, Profile Identifier: 2
> Reported Max Streams: 20, Reported Max OOS Streams: 0
> Supported Codec: g729r8, Maximum Packetization Period: 60
> Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
> Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
> Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization
> Period: 30
>
>
>
> Any ideas if I'm missing something here.
>
> Thanks
> Mann
>
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[OSL | CCIE_Voice] UCM to CME via CUBE

[Mann Chaddha]

Hi All

I am facing a strange issue fro the Vol2 Lab1. I have done this earlier but
today I'm observing a strange issues, where my CME phone stops receiving &
sending any RTP packets when supplementary services are invoked at the HQ
Phone.

Call Flow:
HQ Phone --- GK Trunk (has MTP Enabled) --- HQ-RTR MTP --- HQ-CUBE 
HQ-XCoder (Registered locally to telephony-ser)    Br2 CME RTR --- CME
Phone.

Supplementary services at CME phones work. I can hold unhold the call from
CME Phone (3002). But when I do the same from HQ Phone (5002), CME phone
stops getting & sending RTP.

I have put the following commands on the HQ -RTR:
voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 supplementary-service h450.12
 h323
  emptycapability
  h225 id-passthru
  h225 connect-passthru
  h245 passthru tcsnonstd-passthru
!
!
gatekeeper
 zone local UCM cisco.com 177.1.254.1
 zone local VIA-ZONE cisco.com
 zone local UCME cisco.com outvia VIA-ZONE
 zone prefix UCM 1*
 zone prefix UCME 3*
 zone prefix UCM 5*
 gw-type-prefix 1#* default-technology
 bandwidth interzone zone UCM 32
 no shutdown
!
dial-peer voice 3000 voip
 incoming called-number 3...$
 dtmf-relay h245-alphanumeric
 no vad
!
dial-peer voice 3001 voip
 destination-pattern 3...$
 session target ras
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad


Here are my XCoder & MTP resources registration status:
HQ-R1#sh sccp
SCCP Admin State: UP
Gateway Local Interface: Loopback0
IPv4 Address: 177.1.254.1
Port Number: 2000
IP Precedence: 5
User Masked Codec list: None
Call Manager: 177.1.10.10, Port Number: 2000
Priority: N/A, Version: 7.0, Identifier: 3
Trustpoint: N/A
Call Manager: 177.1.10.20, Port Number: 2000
Priority: N/A, Version: 7.0, Identifier: 2
Trustpoint: N/A
Call Manager: 177.1.254.1, Port Number: 2000
Priority: N/A, Version: 7.0, Identifier: 1
Trustpoint: N/A

Transcoding Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 177.1.254.1, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 1
Reported Max Streams: 8, Reported Max OOS Streams: 0
Supported Codec: g711ulaw, Maximum Packetization Period: 30
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: g729ar8, Maximum Packetization Period: 60
Supported Codec: g729abr8, Maximum Packetization Period: 60
Supported Codec: g729r8, Maximum Packetization Period: 60
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization
Period: 30

MTP Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 177.1.10.20, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 2
Reported Max Streams: 20, Reported Max OOS Streams: 0
Supported Codec: g729r8, Maximum Packetization Period: 60
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization
Period: 30



Any ideas if I'm missing something here.

Thanks
Mann
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[OSL | CCIE_Voice] Two H323 Gateway related Questions

[Mann Chaddha]

Hi Experts

I have these 2 questions related to H323 behavior on Cisco VGs:

1. Can I trombone a call to the same gateway by using Dial-Peers (DP)? For
e.g. if I send the call to an egress VoIP DP & point it to the Loopback IP
Address. Then I use an ingress VoIP DP to receive the call. Is that
possible?

2. Can the same Voice Gateway be registered as a H323 Gateway to 2 separate
CUCM Clusters?

Any help will be greatly appreciated.

TIA
Mann
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Re: [OSL | CCIE_Voice] Rerouting CSS

[Mann Chaddha]

Erwan

The Rerouting CSS is used for dialing the SNR Device (like your cell phone).
This is used when someone calls your desk phone number eg 5002 which is
shared with your RDP.

The other CSS in the RDP is mainly used by MVA (DISA) functionality & when
you call from your Cellphone into the Office. This CSS will enable you to
hunt the RPs available to you when your call arrives at a Voice Gateway,
provided you changed the Service Parameter to this instead of Gateway's CSS.
You can find the service parameter under the Mobility Section

hth
Mann
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[OSL | CCIE_Voice] Help with SIP Phones for Vol2-Lab 8

[Mann Chaddha]

Hi All

I am doing Vol2-Lab 8. All phones are registered as SIP phones.

But I see that when I call from either 5002 or 1002 to each other, the phone
just keep waiting & then restarts. Call to same site phone, like 5002
calling 5001 works. I also tried to implement SIP Dial Rules, still no luck.
I have 7961 Phones so ideally they should not need Dial Rules.

Has anyone faced similar issues with SIP phones.

TIA
Mann
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[OSL | CCIE_Voice] Help with SIP Firmware

[Mann Chaddha]

Hi All

I got it working. Had to reconfigure SIP Registrar & register the phone ot
it. Then with username & password assigned to the phone, I could change the
network settings. Though the phone kept on showing the administrator
username only.

Good day.
Mann
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[OSL | CCIE_Voice] Help with SIP Firmware

[Mann Chaddha]

Hi All

I converted my 7961 phone to SIP Load for a Lab but am not able to revert
the Phone to SCCP Load. For changing any network settings on the phone, like
TFTP etc, the phone prompts for Authorization with Username & Password.

Has anyone faced this issue? Can anyone suggest how should I revert the
phone back to factory defaults or procedure to edit network settings?

A big thanks in advance.
.
Mann
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[OSL | CCIE_Voice] Changing Network Settings on SIP Firmware 7961

[Mann Chaddha]

Hi All

I am running Lab 9 & am facing this issue. My home phone 7961 has a SIP
firmware on it & is trying to register to the Br1-CME (Voice Register
Global). I need to edit the TFTP Server settings on the phone but itas
prompting for a username & password.

What are the defaults for these?

Thanks
Mann
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[OSL | CCIE_Voice] DISA Calling Issues

[Mann Chaddha]

Hi Experts

I am having the following issue with my MVA Configuration. I am working on
the IPX Lab7, I can login with the RDP & PIN to the MVA app. But when I dial
any number to call, like 5002 etc, the call just drops.

The BR2 GW hosting the MVA app is H323 GW. I have the TPs & RPs under the
RDPs CSS for routing calls to these DNs.

Also on the Service Params page, I have updated the CSS to RDP&Line CSS
instead of the Inbound GW CSS.

Do advise.

TIA
Mann
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[OSL | CCIE_Voice] Calling Name Question

[Mann Chaddha]

Hi Experts

I am doing IPX-Lab6 from Vol2. I am observing this behavior which I need
some clarity on.

I am sending CNAME from HQ & BR1 via MGCP GWs to the PSTN. I can see them on
the  Display IE on MGCP GWs with "deb isdn q931". But I don;t see the CNAMe
on the PSTN Phone whicle alerting & connected stage. Here is the debug isdn
q931 output from HQ Site:
-
Jan 22 06:02:32.819: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8  callref =
0x0001
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98383
Exclusive, Channel 3
Display i = 'Hq Ph2'
Calling Party Number i = 0x0081, '+12123945002'
Plan:Unknown, Type:Unknown
Called Party Number i = 0x80, '911'
Plan:Unknown, Type:Unknown
Jan 22 06:02:32.847: ISDN Se0/0/0:23 Q931: RX <- CALL_PROC pd = 8  callref =
0x8001
Channel ID i = 0xA98383
Exclusive, Channel 3
HQ-R1#
Jan 22 06:02:32.851: ISDN Se0/0/0:23 Q931: RX <- ALERTING pd = 8  callref =
0x8001
Progress Ind i = 0x8188 - In-band info or appropriate now available
-

At the same time, form my CME Site, the Display IE again show up in Q931 on
the GW and ALSO on the PSTN Phone. Is this the Expected behavior???

Here is  the output of deb q931from CME Site:

Jan 22 06:01:24.278: ISDN Se0/0/0:15 Q931: TX -> SETUP pd = 8  callref =
0x008C
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98383
Exclusive, Channel 3
Progress Ind i = 0x8183 - Origination address is non-ISDN
BR2-R3#
Display i = 'Br2-Ph2-3002'
Calling Party Number i = 0x0180, '+3545623002'
Plan:ISDN, Type:Unknown
Called Party Number i = 0x81, '112'
Plan:ISDN, Type:Unknown
Jan 22 06:01:24.306: ISDN Se0/0/0:15 Q931: RX <- CALL_PROC pd = 8  callref =
0x808C
Channel ID i = 0xA98383
Exclusive, Channel 3
Jan 22 06:01:24.310: ISDN Se0/0/0:15 Q931: RX <- ALERTING pd = 8  callref =
0x808C
Progress Ind i = 0x8188 - In-band info or appropriate now available
BR2-R3#
Jan 22 06:01:26.898: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8  callref
= 0x008C
Cause i = 0x8090 - Normal call clearing
---

Display IE is enabled on the MGCP GWs & on the CME, isdn ooutgoing diplay ie
is set at the interface level.

Please illuminate.

Thanks
Mann
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[OSL | CCIE_Voice] I Passed CCIE Voice #27803

[Mann Chaddha]

Hi Driss

Congrats on cracking the exam in the first attempt. Commendable feat.

Indeed I was so happy to hear the news as its been some time someone
shared their news of cracking the Lab. Helps us all.

Good luck
Mann
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[OSL | CCIE_Voice] Presence CTIGW AppUser

[Mann Chaddha]

Hi Guys

I somehow am not able to recall the Roles for the CTIGW User for
Presence Desktop Control. Don't even have a SERVER Infront of me.

This is the UCM Appuser. What are the roles we need to assign to it &
wht devices are associated to this?

Kindly respond at your earliest convenience as I have my Lab in 12 hrs.

Thanks
Mann
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Re: [OSL | CCIE_Voice] CUBE Early Offer - Transcoder vs MTP

[Mann Chaddha]

David

If my understanding is right, MTPs also have a DP that they can be
associated to ( I don't have a UCM in-front of me now). So I believe
you have your MTP in the HQ Region which is talking G729 to the CUBE
Trunk Region. And so with MTP its a G729 Call.

But with XCoder, which doesn't happen to have any DP, you seem to have
it support both G711 & G729 Codecs. UCM will always prefer a higher
quality Codec between 2 Endpoints, and so G711 is rightly being
negotiated between HQ & XCoder. But the other side is CME whose
incoming Dial Peer must be hardcoded to G729 & so your XCoder is
converting the media stream to G729 for that feed.

I hope this makes sense.

Good day.
Mann

On Sat, Nov 13, 2010 at 10:30 PM,
 wrote:
> Send CCIE_Voice mailing list submissions to
>        ccie_vo...@onlinestudylist.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
>        http://onlinestudylist.com/mailman/listinfo/ccie_voice
> or, via email, send a message with subject or body 'help' to
>        ccie_voice-requ...@onlinestudylist.com
>
> You can reach the person managing the list at
>        ccie_voice-ow...@onlinestudylist.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of CCIE_Voice digest..."
>
>
> Today's Topics:
>
>   1. CUBE Early Offer - Transcoder vs MTP (David A)
>
>
> --
>
> Message: 1
> Date: Sat, 13 Nov 2010 11:36:48 -0500
> From: David A 
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] CUBE Early Offer - Transcoder vs MTP
> Message-ID:
>        
> Content-Type: text/plain; charset=ISO-8859-1
>
> Hi All,
>
> I was doing a scenario where in I have HQ gateway setup as a CUBE.
>
> HQ = MGCP - phones in CUCM - 5002 - Region HQ with 729 to CUBE
> BR2 = CME - SCCP Phones
> CUBE trunk - Region g729 with all
>
> I am doing Early Offer on the CUBE with inbount and outbound faststart
> and it works fine
>
> My intial undersanding is that "mtp" is needed on HQ gateway with
> g729. Call works fine and both phones use g729.
>
> I however configured a "transcoder" with g711 and 729 and replaced the
> "mtp". Call works fine however in this case HQ phone uses g711 and CME
> uses g729 and I see 2 sessions on transcoder.
>
> All dialpeers are g729 (default voip)
>
> Can someone please help me understand why the codec used on HQ is g711
> in case of transcoder and g729 incase of MTP?
>
> Thanks in advance
>
> DA
>
>
> --
>
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>
> End of CCIE_Voice Digest, Vol 57, Issue 67
> **
>
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Re: [OSL | CCIE_Voice] Help with CUPS Server on PL

[Mann Chaddha]

Naoufal

I tried 3 different browsers & have reloaded the box too. I also tried
to bounce the tomcat service from the CLI. Still no luck.

Anything else that I must be doing here.

Thanks
Mann

On Sat, Nov 6, 2010 at 4:01 PM, Naoufal Kerboute  wrote:
> Check your browser or try to reload cups
>
> On Sat, Nov 6, 2010 at 9:31 AM, Mann Chaddha  wrote:
>>
>> Hi All
>>
>> I am facing issues with activating CUPS Services to day for the second
>> time on PL Racks (Different Ones). So I guess, I might be making some
>> mistake too.
>>
>> The moment I reach the Serviceability Page & click on either Service
>> Activation or Control Center, the page keeps refreshing but doesn't
>> show up anything.Doesn any one know how to fix this. I will appreciate
>> your help here.
>>
>> Thanks
>> Mann
>> ___
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>> visit www.ipexpert.com
>
>
>
> --
> Naoufal Kerboute
> Networks Service Manager
>
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[OSL | CCIE_Voice] Help with CUPS Server on PL

[Mann Chaddha]

Hi All

I am facing issues with activating CUPS Services to day for the second
time on PL Racks (Different Ones). So I guess, I might be making some
mistake too.

The moment I reach the Serviceability Page & click on either Service
Activation or Control Center, the page keeps refreshing but doesn't
show up anything.Doesn any one know how to fix this. I will appreciate
your help here.

Thanks
Mann
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[OSL | CCIE_Voice] Issues with Presence Server

[Mann Chaddha]

Hi Experts

I am on a PL Session and my CUPS Servers is not letting me login to
the Web Admin Utility. The CUPS Servers doesn't initialize properly. I
have rebooted the Server 2 times already but the DB Service Component
is not coming up every time. Its not even letting me login to the Web
Admin Page & throws the error ,"Database communication error ".

Do let me know if there is a fix available.

Here is the output from the "utils services list" which shows that "A
Cisco DB[STOPPED]  Component is not running".
-
admin:utils service list

Requesting service status, please wait...
System SSH [STARTED]
Cluster Manager [STARTED]
Service Manager is running
Getting list of all services
>> Return code = 0
A Cisco DB[STOPPED]  Component is not running
A Cisco DB Replicator[STARTED]
Cisco AMC Service[STARTED]
Cisco CDP[STARTED]
Cisco CDP Agent[STARTED]
Cisco CallManager Admin[STARTED]
Cisco CallManager Serviceability[STARTED]
Cisco CallManager Serviceability RTMT[STARTED]
Cisco Certificate Expiry Monitor[STARTED]
Cisco DRF Local[STARTED]
Cisco DRF Master[STARTED]
Cisco Database Layer Monitor[STARTED]
Cisco License Manager[STARTED]
Cisco Log Partition Monitoring Tool[STARTED]
Cisco RIS Data Collector[STARTED]
Cisco RTMT Reporter Servlet[STARTED]
Cisco Syslog Agent[STARTED]
Cisco Tomcat[STARTED]
Cisco Tomcat Stats Servlet[STARTED]
Cisco Trace Collection Service[STARTED]
Cisco Trace Collection Servlet[STARTED]
Cisco UP Client Profile Agent[STARTED]
Cisco UP Config Agent[STOPPED]  Component is not running
Cisco UP Intercluster Sync Agent[STARTED]
Cisco UP OAM Agent[STARTED]
Cisco UP Presence Engine Database[STARTED]
Host Resources Agent[STARTED]
MIB2 Agent[STARTED]
Native Agent Adapter[STARTED]
SNMP Master Agent[STOPPED]  Component is not running
SOAP -Log Collection APIs[STOPPED]  Component is not running
SOAP -Performance Monitoring APIs[STARTED]
SOAP -Real-Time Service APIs[STARTED]
System Application Agent[STARTED]
Cisco AXL Web Service[STOPPED]  Service Not Activated
Cisco Bulk Provisioning Service[STOPPED]  Service Not Activated
Cisco Serviceability Reporter[STOPPED]  Service Not Activated
Cisco UP Presence Engine[STOPPED]  Service Not Activated
Cisco UP SIP Proxy[STOPPED]  Service Not Activated
Cisco UP Sync Agent[STOPPED]  Service Not Activated
Service Manager is running
Getting list of all services
>> Return code = 0
A Cisco DB[STOPPED]  Component is not running
-

Thanks
Mann
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Re: [OSL | CCIE_Voice] Query on Soft-Pones Features

[Mann Chaddha]

Drew

As far as my understanding goes, IPBlue phones right now do NOT
support + Dialing. I had a word with their technical support team &
they said they may build this into the forthcoming version. I see that
the newer IP Blue Version has presence feature but I haven't tested
it.

And ofcourse, no Barge possible on either IPB or CIPC.

If anyone know more. do contribute.

Thanks
Mann

On Wed, Nov 3, 2010 at 5:07 PM, Dew Swen  wrote:
> me asking the same questions...
>
> which functions are not supported rather than "+ dialing" ?
>
> regards,
>
> --
> Dew Swen
> CCVP, CCDP, CCNP
>
>
>
>
> On Sat, Oct 30, 2010 at 8:12 PM, Mann Chaddha 
> wrote:
>>
>> Hi Everyone
>>
>> Can anyone tell me what all features we cannot test on the IPBlue &
>> CIPC Softphones wrt the Lab?
>>
>> I will appreciate if someone can share a handy list of these features.
>> I am a fortnight away from my Lab & have been practicing on Soft
>> Phones so far.
>>
>> Thanks
>> Mann
>> ___
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>> visit www.ipexpert.com
>
>
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[OSL | CCIE_Voice] Query on Soft-Pones Features

[Mann Chaddha]

Hi Everyone

Can anyone tell me what all features we cannot test on the IPBlue &
CIPC Softphones wrt the Lab?

I will appreciate if someone can share a handy list of these features.
I am a fortnight away from my Lab & have been practicing on Soft
Phones so far.

Thanks
Mann
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[OSL | CCIE_Voice] Is it possible to enable SIP Early Offer on a SIP Dial Peer?

[Mann Chaddha]

Hi Experts

I was wondering if it is possible to enable SIP Early Offer on a SIP
Dial Peer? I believe that this should be negotiated by an endpoint and
as Dial Peers are not endpoints in themselves, will they still support
Early Offer?

I went through Cisco's DOC but the method mentioned in the SIP IOS
Guide didn't work when I was working on the Lab 8 today.

Kindly advise.

Thanks
Mann
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[OSL | CCIE_Voice] Question Regarding + Dialing & Single Route Pattern

[Mann Chaddha]

Hi Guys

W.r.t Q 2.4 in Lab 3 (Vol2) ,where we have to apply TEHO Routing for
BR2 PSTN Calls. So all +34! calls should egress out from BR2 GW for HQ
& BR1 Sites.

I have a \+! RP & \+34! RP for TEHO. With TEHO, the requirement is to
have Local PRI as backup. So I pointed \+34! RP to a RL-TEHO-BR2 with
BR2 RTR as Primary & Std Local RG as  secondary. This all works fine
but I am not able to provide same DNIS on the Calling Phone. IT shows
the DNIS from the CAlled Party Xformations. So for BR2, I see
03214, & for LRG, I see 01134!. I would like to know how can we
standardize this output so that the end user doesn't know the
difference in Call Routing in the backend.

What exactly does the Phone show as DNIS ? I tried manipulating the RP
to apply called party XFors for 01134!, but incidentally Called Party
XFors are superseding that. I thought Called Party XFors don't
supersede RP Level Transformations, AM I Wrong here?

Any help will be appreciated.

Thanks
Mann
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Re: [OSL | CCIE_Voice] Assistance with IPBlue Multilab Version

[Mann Chaddha]

Got it resolved. It needs MAC without any dotted notation ( NO
.. but just ).

Sorry to bother.
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[OSL | CCIE_Voice] Assistance with IPBlue Multilab Version

[Mann Chaddha]

Hi

I recently purchased IPBlue Multilab Version and am unable to get them
registered with both the CME & UCM. It receives the "Failed to
retrieve configuration from TFTP Server XX.XX.XX.XX". I have been
running the demo version all this time & was able to run them without
these issues.

Is there anything specific we need to do with the multilab version.
Please advise at your earliest convenience as I am doing a Lab
currently & will appreciate your response.

Thanks
Mann
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[OSL | CCIE_Voice] Question on H323 Trunks & Media Negotiation

[Mann Chaddha]

Hi Everyone

I need some clarity on a call flow that uses multiple ICT/H323 Trunks.
Here is the scenario:

Cluster 1
Cluster 2
Cluster 3
   WAN
   WAN
PH 1 ---> GK Trunk (G729 Region) --> PH 2 (VM Roll Over)
---> ICT Trunk (G711 Region Only) > VM Ports
(VG248) to Octel

The call originates from Ph1 which uses a GK Trunk to arrive at
Cluster 2 Ph2. When the Ph2 call rolls over to VM, then the VM Pilot
points to a RP associated with an ICT with G711 Region only. My
understanding is that the ICT will still negotiate the calls G729 ( <
80 kbps) & forward it to Cluster 3 and it will not invoke any XCoder.
Cluster 3, I believe, will forward the call to VM Ports as G729 only.

I want to hard-code the VM Ports to talk only G711 but in this case,
even with appropriate Region Settings, I am unable to.

Is my understanding correct or am I missing something here?

Kindly suggest.

Thanks
Mann
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[OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

[Mann Chaddha]

Hi David

I reckon that by providing Voice Class Codec at the Outbound DP on
CME, you have allowed the call to proceed with G711 to the GK.
Ideally, if the Inbound DP (SIP Voice Pool in this case) and the
Outbound DP (DP to HQ/BR1) have been hard-coded to different codec
values, they should invoke a local XCoder. In your case, that doesn't
happen as your outbound DP has a Voice Class Codec assigned to it.

Why don't you hard-code the Outbound DP with G729 & Inbound DP with
G711 and then test the same?

HTH
Mann
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[OSL | CCIE_Voice] Proctorlabs Down?

[Mann Chaddha]

Even I got logged out :(

Mann
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[OSL | CCIE_Voice] Lab Documentation Related Question

[Mann Chaddha]

Hi Everyone

Just wanted to know how is the lab handed to the candidates.

   -  Is it in Softcopy or a hard binder?
   - Are we given the Vlan, Phone, PRI etc Info in a table similar to IPX
   Labs?
   - Are we provided a topology diagram & IP addresses similar to IPX Labs?

I hope these aren't part of the NDA. Please do share whatever is legit.

Thanks
Mann
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[OSL | CCIE_Voice] Vol2-Lab1: Can't call from PST to HQ-RTR

[Mann Chaddha]

Hi All

I am facing an issue with calling from my PSTN to HQ-RTR. I am doing the
Vol2-Lab1. This is the error I receive on the PSTN Router with ''Debug isdn
q931'':
*
ISDN Se0/3/0:23 **ERROR**: CCPCC_CallOrigination: SETUP timed-out (2nd T303)

*I don't even receive any setup request on the HQ-RTR. The ISDN is in
''Multiple Frame Established'' mode. I have enabled the debug isdn q931 but
I don't see anything.

Is there something that I'm missing in here.

Also, I could not find any phone connected to my Br1-RTR ESW Module.

Any help would be extremely helpful.

Thanks
Mann
*
*
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[OSL | CCIE_Voice] Lab11A : Unity Connection Related Simple Question

[Mann Chaddha]

Hi Guys

This is my first time with Unity Connection. I am working on the Lab 11A,
and have done the CUCM & CUC Integration configs. Just wanted to know
whether the VM POrts on CUCM show registered or unregistered when integrated
with CUC? I guess these are SCCP ports so should not require any User
association, right?

Please advise, still 2 hours to go for my proctor lab today.

thnx
Mann
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[OSL | CCIE_Voice] Local Route Group Related question

[Mann Chaddha]

Hi Everyone

I was wondering how will we manage dual carrier (with diff DNIS
requirements) with the LRG Approach. This is very much a standard at our
firm wherein we always have dual carriers for resiliency sake. With the old
school approach its pretty straightforward with 2 RG for each carrier &
putting them in the Site Specific RL.

Thnx
Mann
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[OSL | CCIE_Voice] Lab 5C Ex 5.2 Related Question

[Mann Chaddha]

Hi Guys

I have these following questions w.r.t. the exercise 5.2 in Lab 5C:
1. How does CUCM take SIP Calls from CME? We created a SIP Trunk from CUCM
to CME & SIP Dial-Peers at CME. Will it take calls from any Call Agent
(H323/SIP) without needing any configuration on itself ?
2. How will CUCM invoke Xcoder for CME SIP Calls to BR2 Phones (G729 only)?
Who will negotiate the MR requirements? Will SIP Trunk configs dictate
XCoding assignment here too?

Thanks
Mann
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[OSL | CCIE_Voice] Lab1A Final Configs

[Mann Chaddha]

Hi

I saw some old mails on non availability of Lab1A Initial Configs. I had to
load the final configs to access all the Servers. But I was able to access
other devices with Initial Configs too.

How did others approach Lab1A? Please let me know.

Thanks
Mann.
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[OSL | CCIE_Voice] Codec Negotiation for CTI Ports for IPCC Mobile Agents

[Mann Chaddha]

Hi

As per Cisco's Doc, LCP/RCP CTI Ports which are required for Mobile
Agents CANNOT negotiate Codec and need to have same Codec for Inbound
& Outbound Calls. I have a scenario where the remote users (Mobile
Agents) reside on another CUCM Cluster. The calls over the 2 clusters
shall be G729 as they are over the WAN. The Clusters have H323 Trunks
with GK for Routing purpose.

Has anyone of you faced a similar limitation & how did you avoid it?
Is there a way you can invoke Xcoders for CTI Ports ( Jtapi Gateway on
PG does patch the media between Inbound & Outbound Call Legs)?

I think this is great forum for this question, though IPCCE ain't on
CCIE blueprint.

Thanks
Mann
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[OSL | CCIE_Voice] CCIE Voice Written Exam

[Mann Chaddha]

Hi

I'm looking forward to take the CCIE Written exam in July. Can someone
share the information on how different that would be from the CCVP
examinations? What all content need to be stressed upon which is not
part of the CCVP examinations?

I see that the topics on the BP that are different than in CCVP are
UCE, SIP, Unity, IPCC Express, IPIVR/CRS. Can someone share their
experience of preparing for the written exam?

Also what is the passing requitement for the exam. I heard its 75%
instead of 85% for the 642 series exams.

Thanks
Mann.

Re: [OSL | CCIE_Voice] FAC on Internal DN

[Mann Chaddha]

Thank you Cliff & Anil.

This should be good for me to dig further.

Good day guys.
Mann.

On Thu, Apr 2, 2009 at 3:04 PM, Cliff McGlamry  wrote:
> Add the following command to your gateway:
>
> voice service voip
> allow h323 to h323
>
>
> That should fix it.
>
> Cliff
>
> - Original Message -
> From: "Mann Chaddha" 
> To: "Cliff McGlamry" 
> Cc: 
> Sent: Thursday, April 02, 2009 12:13 AM
> Subject: Re: [OSL | CCIE_Voice] FAC on Internal DN
>
>
> Cliff
>
> That is certainly possible but still there is a requirement of some
> authentication before patching he call to the system.
>
> And I guess a vanila H323 GW also will not hairpin as both the call
> legs are VoIP. We may need IPIPGW to achieve that.
>
> Do you have any idea of how to reoriginate the call from one Dial Peer
> to another?
>
> Thanks
> Mann
>
> On Thu, Apr 2, 2009 at 12:31 AM, Cliff McGlamry  wrote:
>> Why not just put it in a partition and limit who can dial it by only
>> including it in a limited set of Call Search Spaces?
>>
>>
>> - Original Message -
>> From: "Mann Chaddha" 
>> To: "Cliff McGlamry" 
>> Cc: 
>> Sent: Wednesday, April 01, 2009 2:15 AM
>> Subject: Re: [OSL | CCIE_Voice] FAC on Internal DN
>>
>>
>> Thanks Cliff
>>
>> Basically we are interfacing a Public Announceement System(PAS) with
>> an analog port on a VG248. The PAS doesn't have any builtin
>> authentication & so I need to enforce FAC on the DN itself.
>>
>> Mann.
>> 
>>
>> On Tue, Mar 31, 2009 at 10:35 PM, Cliff McGlamry 
>> wrote:
>>> MGCP can make it work only if it's going to be hairpinned via the PSTN.
>>>
>>> If you use H323, you can run it in and out on VOIP dial-peers.
>>>
>>> H323 would likely be a better way to go if you must do this. That said,
>>> this is a pretty strange requirement. I would ask some hard questions
>>> about
>>> exactly WHY this needs to be done. It might be more appropriate to do
>>> something where you are working with CDR after the fact and billing based
>>> on
>>> that back to the originating station.
>>>
>>>
>>> - Original Message -
>>> From: "Mann Chaddha" 
>>> To: 
>>> Sent: Tuesday, March 31, 2009 12:55 PM
>>> Subject: [OSL | CCIE_Voice] FAC on Internal DN
>>>
>>>
>>> Hi
>>>
>>> We’ve come across a typical requirement wherein we need to enable
>>> FAC(Forced Auth Code) on an Internal DN.
>>>
>>> I was thinking of creating a RP & hair pinning it back from a GW to
>>> the required DN.This DN could be only in this GW's CSS.
>>>
>>> Is this possible with MGCP or I need to have only H.323 GW? What will
>>> be the required config on the Gateway?
>>>
>>> Thanks
>>>
>>> Mann
>>>
>>>
>>>
>>
>>
>>
>
>
>

Re: [OSL | CCIE_Voice] FAC on Internal DN

[Mann Chaddha]

Cliff

That is certainly possible but still there is a requirement of some
authentication before patching he call to the system.

And I guess a vanila H323 GW also will not hairpin as both the call
legs are VoIP. We may need IPIPGW to achieve that.

Do you have any idea of how to reoriginate the call from one Dial Peer
to another?

Thanks
Mann

On Thu, Apr 2, 2009 at 12:31 AM, Cliff McGlamry  wrote:
> Why not just put it in a partition and limit who can dial it by only
> including it in a limited set of Call Search Spaces?
>
>
> - Original Message -
> From: "Mann Chaddha" 
> To: "Cliff McGlamry" 
> Cc: 
> Sent: Wednesday, April 01, 2009 2:15 AM
> Subject: Re: [OSL | CCIE_Voice] FAC on Internal DN
>
>
> Thanks Cliff
>
> Basically we are interfacing a Public Announceement System(PAS) with
> an analog port on a VG248. The PAS doesn't have any builtin
> authentication & so I need to enforce FAC on the DN itself.
>
> Mann.
> 
>
> On Tue, Mar 31, 2009 at 10:35 PM, Cliff McGlamry  wrote:
>> MGCP can make it work only if it's going to be hairpinned via the PSTN.
>>
>> If you use H323, you can run it in and out on VOIP dial-peers.
>>
>> H323 would likely be a better way to go if you must do this. That said,
>> this is a pretty strange requirement. I would ask some hard questions
>> about
>> exactly WHY this needs to be done. It might be more appropriate to do
>> something where you are working with CDR after the fact and billing based
>> on
>> that back to the originating station.
>>
>>
>> - Original Message -
>> From: "Mann Chaddha" 
>> To: 
>> Sent: Tuesday, March 31, 2009 12:55 PM
>> Subject: [OSL | CCIE_Voice] FAC on Internal DN
>>
>>
>> Hi
>>
>> We’ve come across a typical requirement wherein we need to enable
>> FAC(Forced Auth Code) on an Internal DN.
>>
>> I was thinking of creating a RP & hair pinning it back from a GW to
>> the required DN.This DN could be only in this GW's CSS.
>>
>> Is this possible with MGCP or I need to have only H.323 GW? What will
>> be the required config on the Gateway?
>>
>> Thanks
>>
>> Mann
>>
>>
>>
>
>
>

Re: [OSL | CCIE_Voice] FAC on Internal DN

[Mann Chaddha]

Thanks Cliff

Basically we are interfacing a Public Announceement System(PAS) with
an analog port on a VG248. The PAS doesn't have any builtin
authentication & so I need to enforce FAC on the DN itself.

Mann.


On Tue, Mar 31, 2009 at 10:35 PM, Cliff McGlamry  wrote:
> MGCP can make it work only if it's going to be hairpinned via the PSTN.
>
> If you use H323, you can run it in and out on VOIP dial-peers.
>
> H323 would likely be a better way to go if you must do this.  That said,
> this is a pretty strange requirement.  I would ask some hard questions about
> exactly WHY this needs to be done.  It might be more appropriate to do
> something where you are working with CDR after the fact and billing based on
> that back to the originating station.
>
>
> - Original Message -
> From: "Mann Chaddha" 
> To: 
> Sent: Tuesday, March 31, 2009 12:55 PM
> Subject: [OSL | CCIE_Voice] FAC on Internal DN
>
>
> Hi
>
> We’ve come across a typical requirement wherein we need to enable
> FAC(Forced Auth Code) on an Internal DN.
>
> I was thinking of creating a RP & hair pinning it back from a GW to
> the required DN.This DN could be only in this GW's CSS.
>
> Is this possible with MGCP or I need to have only H.323 GW? What will
> be the required config on the Gateway?
>
> Thanks
>
> Mann
>
>
>

[OSL | CCIE_Voice] FAC on Internal DN

[Mann Chaddha]

Hi

We’ve come across a typical requirement wherein we need to enable
FAC(Forced Auth Code) on an Internal DN.

I was thinking of creating a RP & hair pinning it back from a GW to
the required DN.This DN could be only in this GW's CSS.

Is this possible with MGCP or I need to have only H.323 GW? What will
be the required config on the Gateway?

Thanks

Mann