from:"David A"

Hello,

Just wondering if it's just me.  I'm trying from 2 different PCs and cannot
access the webpage...

Thanks,

-Dave
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Re: [OSL | CCIE_Voice] Can anyone access www.proctorlabs.com ???

Does anyone remember the access server IP?  The EZVPN is working, but the
infrastructure is blank, so nothing is accessible...

Tyson - not sure if you have some way to get hold of Proctor Labs support...


Thanks.

On Sat, Aug 14, 2010 at 9:38 AM, Brian Valentine bkvalent...@gmail.comwrote:

 No... I'm having the same issue.

 On Sat, Aug 14, 2010 at 9:35 AM, David Lee d16...@gmail.com wrote:
  Hello,
  Just wondering if it's just me.  I'm trying from 2 different PCs and
 cannot
  access the webpage...
  Thanks,
  -Dave
 
  ___
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  visit www.ipexpert.com
 
 

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Re: [OSL | CCIE_Voice] Can anyone access www.proctorlabs.com ???

I just got connected now.

-Dave

On Sat, Aug 14, 2010 at 10:46 AM, Scott Newberry sc...@meganandscott.comwrote:

 FYI, got an email from Viking. Looking into it.

 Sent from my mobile phone.  Please excuse my brevity and any spelling
 errors.

 On Aug 14, 2010 9:42 AM, David Lee d16...@gmail.com wrote:
  Does anyone remember the access server IP? The EZVPN is working, but the
  infrastructure is blank, so nothing is accessible...
 
  Tyson - not sure if you have some way to get hold of Proctor Labs
 support...
 
 
  Thanks.
 
  On Sat, Aug 14, 2010 at 9:38 AM, Brian Valentine bkvalent...@gmail.com
 wrote:
 
  No... I'm having the same issue.
 
  On Sat, Aug 14, 2010 at 9:35 AM, David Lee d16...@gmail.com wrote:
   Hello,
   Just wondering if it's just me. I'm trying from 2 different PCs and
  cannot
   access the webpage...
   Thanks,
   -Dave
  
   ___
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 please
   visit www.ipexpert.com
  
  
 

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[OSL | CCIE_Voice] Unable to reload the CUE license file

Hello,

I got the error Online install/download is not allowed due to insufficient
FLASH capacity when attempting to update the CUE license file.  I tried to
do a software reinstall via bootflash, thinking that it would clear the
flash, but the error still remained.

Can anyone advise what else can be done in this case?

Thanks,

-Dave
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[OSL | CCIE_Voice] ATA in SRST

2010-07-17 Thread David Lee

Hello,

I was wondering if anyone has experienced ATA's not registering to the SRST
router during fallback...  They are on the same DP as the phones, and the
phones register, but no registration on the ATAs...

Thanks,

-Dave
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[OSL | CCIE_Voice] Question on globalized dial plan

2010-07-13 Thread David Lee

Hello,

This is a question for a real implementation for all experts (and us
wannabes :) ).

The PSTN dial plan is 10-digit for local and 11-digits for North American
LD.  However, neighboring cities have the same area code.  So for example,
(905) ABC-1234 is a local call, and (905) XYZ-6789 could be rejected because
it is a long distance call, requiring the country code (1).

From this site (www.localcallingguide.com), I can determine which NXX (the 3
digits prefix after the area code) are local to the area.  So I can brute
force config which RP I strip +1, and which I only strip the +.  I was just
wondering if there is a more elegant way...

Thanks,

-Dave
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[OSL | CCIE_Voice] Vol2 Lab1, 4.2 Supplementary services

Hello,

I am at a lost.  I got most of this section working.  I can resume a call if
the hold was initiated by an UCM phone or the CME SCCP phone.  However, I
cannot resume if the hold was initiated by the CME SIP phone.   Any one have
ideas what can be looked at to troubleshoot?  (Software MTP is configured
and active during the call.  Codecs are also right.)

Thanks,

-Dave
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Re: [OSL | CCIE_Voice] Vol2 Lab1, 4.2 Supplementary services

Let me clarify.  I am using IOS Software MTP, not the UCM software MTP.



On Sat, Jun 26, 2010 at 8:04 PM, Pavan pav.c...@gmail.com wrote:

 You will need a hardware mtp (i.e dsp / ios sw ) not the one provided by
 ucm

 Sent from my phone

 On Jun 26, 2010, at 18:47, David Lee d16...@gmail.com wrote:

  Hello,
 
  I am at a lost.  I got most of this section working.  I can resume a call
 if the hold was initiated by an UCM phone or the CME SCCP phone.  However, I
 cannot resume if the hold was initiated by the CME SIP phone.   Any one have
 ideas what can be looked at to troubleshoot?  (Software MTP is configured
 and active during the call.  Codecs are also right.)
 
  Thanks,
 
  -Dave
 
 
  ___
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 visit www.ipexpert.com

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Re: [OSL | CCIE_Voice] Vol2 Lab1, 4.2 Supplementary services

Hi Daniel,

With your helpful post, consultative transfer works now. :)

The hold/resume still DOES NOT WORK when initiated from the BR2 SIP
phone...  Weird.

Thanks,

-Dave



On Sat, Jun 26, 2010 at 8:22 PM, Daniel Berlinski dberlin...@gmail.comwrote:

 David please check the link below
 http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg17219.html

 These were the troubleshooting  I completed  with the help of other members
 of this list to get it working.  I got it working by using IOS software MTP,
 adding some h323 commands under the CUBE, and unchecking wait for
 capabilities exchange from the h225 controlled trunk

 Cheers


 On Sun, Jun 27, 2010 at 12:18 PM, David Lee d16...@gmail.com wrote:

 Let me clarify.  I am using IOS Software MTP, not the UCM software MTP.




 On Sat, Jun 26, 2010 at 8:04 PM, Pavan pav.c...@gmail.com wrote:

 You will need a hardware mtp (i.e dsp / ios sw ) not the one provided by
 ucm

 Sent from my phone

 On Jun 26, 2010, at 18:47, David Lee d16...@gmail.com wrote:

  Hello,
 
  I am at a lost.  I got most of this section working.  I can resume a
 call if the hold was initiated by an UCM phone or the CME SCCP phone.
  However, I cannot resume if the hold was initiated by the CME SIP phone.
 Any one have ideas what can be looked at to troubleshoot?  (Software MTP is
 configured and active during the call.  Codecs are also right.)
 
  Thanks,
 
  -Dave
 
 
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 please visit www.ipexpert.com



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Re: [OSL | CCIE_Voice] Vol2 Lab1, 4.2 Supplementary services

I only have IOS Software MTP MRG in the MRGL.  It is engaged.

HQ-RTR#sh sccp conn
sess_idconn_idstype mode codec   ripaddr rport sport

33555465   33554541   mtp   sendrecv g729192.168.10.13   20536 17736
33555465   33554540   mtp   sendrecv g72910.10.112.2 17962 19012




On Sat, Jun 26, 2010 at 8:32 PM, Daniel Berlinski dberlin...@gmail.comwrote:

 Sorry, forgot to mentionto make sure your ios software MTP MRG is on top of
 the Hardware IOS xcoder to ensure you are invoking it first choice
 On Sun, Jun 27, 2010 at 11:47 AM, David Lee d16...@gmail.com wrote:

 Hello,

 I am at a lost.  I got most of this section working.  I can resume a call
 if the hold was initiated by an UCM phone or the CME SCCP phone.  However, I
 cannot resume if the hold was initiated by the CME SIP phone.   Any one have
 ideas what can be looked at to troubleshoot?  (Software MTP is configured
 and active during the call.  Codecs are also right.)

 Thanks,

 -Dave



 ___
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 visit www.ipexpert.com



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[OSL | CCIE_Voice] cisco.com link

2010-06-17 Thread David Lee

A thread-jack / follow up question regarding the cisco.com link: do people
know if the Configuration Examples and Tech notes are accessible during the
exam?

Thanks,

-Dave
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Re: [OSL | CCIE_Voice] translation rule

2010-05-21 Thread David Holman

I keep this link handy for voice translation questions:

http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080325e8e.shtml

On Fri, May 21, 2010 at 4:15 PM, Wael Agina waelag...@gmail.com wrote:

 Dear Ashar,

   The ^$ is catching null, which could be used to catch calls from unkown.
 example usage, drop any calls from PSTN that has ANI of unkown type.
 On H323 you could use following rule to do this

 voice translation-rule 1
  rule 1 reject /^$/

 voice translation-profile Drop-Unknown
   translate calling 1

 dial-peer voice 1 pots
 direct-inward-dial
 incom called .
 *call-block translation-profile incoming Drop-Unknown*

 For you example may be it i setting unknown ANI to be 42000 for example,
 bu not sure, need to be tested.

 Regards,
 Wael Agina

 On Fri, May 21, 2010 at 11:02 PM, Ashar Siddiqui siddas...@gmail.comwrote:

  Hi,

 I know I may sound stupid to some but I really want to know the purpose of
 ^$ in a translation rule for e.g:

 voice translation-rule 100
  rule 1 /^$/ /42000/
 !


 ^$ is null...what does it mean? what is a null number?

 Ash

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 --

 Thanks and Best Regards,
 Wael Agina

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[OSL | CCIE_Voice] Lab5A -- RL/RG question

2010-04-10 Thread David Lee

Hello,

Lab 5A, step 5.8 specifies the creation of a RL, with the SIP trunk to HQ-GW
as primary RG, and BR1 RTR MGCP GW as secondary RG.  According to the PG,
once the voice-ports on HQ-RTR are shut, calls should go to BR1 RTR.  (I
know the call won't succeed because the called number would be different.)
 However, I don't see the call coming to BR1 RTR at all from debug isdn
q931!  The DNA says that the BR1 RG is selected.

Can anyone shed any light on this?

Thanks,

-Dave
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Re: [OSL | CCIE_Voice] Lab5A -- RL/RG question

2010-04-10 Thread David Lee

Hi Ash,

I don't think the call is even making it to BR1 RTR...  According to DNA,
the BR1 RG is an option.  If the RL only has BR1 RG, the call works.  When
the RL contains both HQ RG and BR1 RG, and the voiceports on HQ RTR are
shut, the call does not go to BR1 RG.

Is there away to see what digits (if any) are sent to an MGCP gateway?
 There is no result on debug isdn q931...

Thanks

-Dave


On Sat, Apr 10, 2010 at 6:53 PM, Ashar Siddiqui siddas...@gmail.com wrote:

  Are you talking about the National dialing backup from HQ router? I don't
 know much what's in the proctor guide but this is how you gonna do it.

 Just make pattern 91.[2-9]XX[2-9]XX  PreDot and insert a route List
 with HQ-GW first and BR1-GW as second choice. At HQ-GW RL details, prefix
 91, NANP preDot..set other ANI requirements if it has been asked. At Br1-GW,
 prefix 1, NANP etc and it should work.

 Ash



 On 10/04/2010 22:57, David Lee wrote:

 Hello,

  Lab 5A, step 5.8 specifies the creation of a RL, with the SIP trunk to
 HQ-GW as primary RG, and BR1 RTR MGCP GW as secondary RG.  According to the
 PG, once the voice-ports on HQ-RTR are shut, calls should go to BR1 RTR.  (I
 know the call won't succeed because the called number would be different.)
  However, I don't see the call coming to BR1 RTR at all from debug isdn
 q931!  The DNA says that the BR1 RG is selected.

  Can anyone shed any light on this?

  Thanks,

  -Dave


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 --
 Thanks,
 Ashar Siddiqui


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Re: [OSL | CCIE_Voice] Lab5A -- RL/RG question

2010-04-10 Thread David Lee

Thanks, Ash.

The problem was not with the dial-plan, but with the CM Service Parameter
Stop Routing on Unallocated Number Flag.  Once set to False (default is
True), it goes to the next RG in the RL.

That only took 3 hours. :)  Sad thing is I was burned on something like this
6 months ago on a project, and I couldn't remember exactly which parameter
TAC told me to change...

On Sat, Apr 10, 2010 at 7:52 PM, Ashar Siddiqui siddas...@gmail.com wrote:

  David,

 You sure you are sending the correct digits to BR1-RTR as required by Br1
 PSTN?
 What Pattern you are using and what changes you are making at RL details
 (if any).
 I think you are doing something wrong at RL.



 On 11/04/2010 00:46, David Lee wrote:

 Hi Ash,

  I don't think the call is even making it to BR1 RTR...  According to DNA,
 the BR1 RG is an option.  If the RL only has BR1 RG, the call works.  When
 the RL contains both HQ RG and BR1 RG, and the voiceports on HQ RTR are
 shut, the call does not go to BR1 RG.

  Is there away to see what digits (if any) are sent to an MGCP gateway?
  There is no result on debug isdn q931...

  Thanks

  -Dave


  On Sat, Apr 10, 2010 at 6:53 PM, Ashar Siddiqui siddas...@gmail.comwrote:

 Are you talking about the National dialing backup from HQ router? I don't
 know much what's in the proctor guide but this is how you gonna do it.

 Just make pattern 91.[2-9]XX[2-9]XX  PreDot and insert a route List
 with HQ-GW first and BR1-GW as second choice. At HQ-GW RL details, prefix
 91, NANP preDot..set other ANI requirements if it has been asked. At Br1-GW,
 prefix 1, NANP etc and it should work.

 Ash



 On 10/04/2010 22:57, David Lee wrote:

  Hello,

  Lab 5A, step 5.8 specifies the creation of a RL, with the SIP trunk to
 HQ-GW as primary RG, and BR1 RTR MGCP GW as secondary RG.  According to the
 PG, once the voice-ports on HQ-RTR are shut, calls should go to BR1 RTR.  (I
 know the call won't succeed because the called number would be different.)
  However, I don't see the call coming to BR1 RTR at all from debug isdn
 q931!  The DNA says that the BR1 RG is selected.

  Can anyone shed any light on this?

  Thanks,

  -Dave


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 visit www.ipexpert.com



 --
 Thanks,
 Ashar Siddiqui




 --
 Thanks,
 Ashar Siddiqui


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[OSL | CCIE_Voice] Ben Ng's poll - Should the Unity Connection Remote Port Status Monitor be offerred as a tool in the CCIE Voice lab?

2010-04-01 Thread David Lee

TAC recently showed me this tool, and I find it having a nicer interface
than RTMT for what it does. Anyways, I asked on the CCIE Voice Study Group
at the Cisco Learning Network, and Ben opened it up to a vote. He'll
consider having it installed on Candidate PCs if there are 100 yes votes.

Link to the thread:
https://learningnetwork.cisco.com/message/59281#59281

Link to poll:
https://learningnetwork.cisco.com/poll.jspa?poll=1094

On Thu, Apr 1, 2010 at 7:41 AM, ccie_voice-requ...@onlinestudylist.comwrote:

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: Priority Queue on UCCX 5.0.2 (Otto Sanchez)
2. Re: What Documentation is Accessible in Lab (Ohamien Uhakheme)
3. Re: Bandwidth Per Call (Angel Perez)

Message: 1
Date: Thu, 1 Apr 2010 06:58:22 -0430
From: Otto Sanchez o...@ipexpert.com
Subject: Re: [OSL | CCIE_Voice] Priority Queue on UCCX 5.0.2
To: Cristobal Priego cristobalpri...@gmail.com
Cc: ccie_voice@onlinestudylist.com
Message-ID:
u2wbe6b39441004010428l2aa3ae75w3d49fb1ff251e...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1

Hi Cris,

Please take a look to the following docs:

http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_5_0/programming/script_devp/gs_with_scripts/crs501gs.pdf
Cisco CRS Scripting and Development Series: Volume 1, Getting Started with
Scripts 5.0(1) , Chapter 17, shows an example on how to set up the set
priotity step

http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_5_0/programming/script_devp/editor_step_ref/crs501sr.pdf
Cisco CRS Scripting and Development Series: Volume 2, Editor Step Reference
5.0(1) , show the set priority step properties

Also make sure you are not using standard licenses with your uccx
deployment,

On Wed, Mar 31, 2010 at 7:09 PM, Cristobal Priego
cristobalpri...@gmail.com
wrote:

Hello all,

I'd like to know if you have any documentation or if you could point me
on
the proper track. I'd like to set up a priority queue on my script. I'm
using the priority step, but it doesn't seem to be working properly. how
do
i do it

thanks

Cris

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--
Regards,

Otto Sanchez
CCIE #25592 (Voice)
Support Engineer - IPexpert, Inc.
URL: http://www.IPexpert.com
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Message: 2
Date: Thu, 1 Apr 2010 07:40:18 -0400
From: Ohamien Uhakheme oham...@gmail.com
Subject: Re: [OSL | CCIE_Voice] What Documentation is Accessible in
Lab
To: Mike Brooks 2xcci...@gmail.com
Cc: ccie_voice@onlinestudylist.com
Message-ID:
u2n64cae69e1004010440uaa5a7093sb2b286643014...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1

What about during the OEQs? Do we have access to anything during that
time?

Ohamien

On Wed, Mar 31, 2010 at 7:33 PM, Mike Brooks 2xcci...@gmail.com wrote:

According to Ben Ng:

We have four SRND documents ready to be opened, also you have the online
Cisco document page.
1. UC 7 SRND
2. CUCME 7 SRND
3. UCCX 7 SRND
4. Enterprise QoS SRND 3.3

I think thats it.

Mike

On Wed, Mar 31, 2010 at 7:23 PM, Bryan Brooks ccieiwi...@gmail.com
wrote:

Hi Everyone,

I was curious if someone could provide a list of documentation that is
available when sitting the lab. I want to become familiar with the
documentation just in case I need to find something. I tried searching
the
archive but was not successful. Thanks in advance for any info
provided.

Thanks
Bryan Brooks

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Message: 3
Date: Thu, 1 Apr

Re: [OSL | CCIE_Voice] CUPC DIABLED

2010-03-05 Thread David



-Original Message-
From: Steve Sarrick ssarr...@drsllc.net
Sent: Friday, March 05, 2010 7:52 AM
To: J Hogan j.jho...@gmail.com; Ashar Siddiqui siddas...@gmail.com
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CUPC DIABLED



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[OSL | CCIE_Voice] New Workbook Lab Access

Have all the of the new labs been posted online? I am only able to see
Volume 2 labs 1 - 4 when I login.

TIA
Dave
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Re: [OSL | CCIE_Voice] SRST on BR2 not working

Did you check your SRST reference in CUCM?

Dave

On Feb 8, 2010, at 11:08 PM, vccie2010 wrote:

 Hi
  
 My BR2 site phones don't fall back to SRST.
  
 call-manager-fallback
  max-conferences 8 gain -6
  transfer-system full-consult
  ip source-address fa voice subinterface port 2000
  max-ephones 10
  max-dn 5 dual-line
  voicemail 3600 call-forward pattern .T
  call-forward busy 3600
  call-forward noan 3600 timeout 10
  mwi relay
 sip-ua
  mwi-server ipv4:cue IP addr  expires 3600 port 5060 transport udp 
 unoslicited
  
 Am I missing anythign here pls...
  
 -ak
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Re: [OSL | CCIE_Voice] SRST on BR2 not working

Do you have

Application
Global
Service alternate default 

In the config ?

Dave
On Feb 8, 2010, at 11:34 PM, vccie2010 wrote:

 Dave,
  
 Here is the SRST related config, do you need the comlete GW configs pls...
  
 call-manager-fallback
  max-conferences 8 gain -6
  transfer-system full-consult
  ip source-address fa voice subinterface port 2000
  max-ephones 10
  max-dn 5 dual-line
  voicemail 3600 call-forward pattern .T
  call-forward busy 3600
  call-forward noan 3600 timeout 10
  mwi relay
 sip-ua
  mwi-server ipv4:cue IP addr  expires 3600 port 5060 transport udp 
 unoslicited
  
 -ak
 
 On Mon, Feb 8, 2010 at 9:08 PM, vccie2010 vccie2...@gmail.com wrote:
 Hi
  
 My BR2 site phones don't fall back to SRST.
  
 call-manager-fallback
  max-conferences 8 gain -6
  transfer-system full-consult
  ip source-address fa voice subinterface port 2000
  max-ephones 10
  max-dn 5 dual-line
  voicemail 3600 call-forward pattern .T
  call-forward busy 3600
  call-forward noan 3600 timeout 10
  mwi relay
 sip-ua
  mwi-server ipv4:cue IP addr  expires 3600 port 5060 transport udp 
 unoslicited
  
 Am I missing anythign here pls...
  
 -ak
 
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Re: [OSL | CCIE_Voice] SRST on BR2 not working