Re: [OSL | CCIE_Voice] calling out to FXS port via SIP trunk-nocaller-id?

[Jane Ryer (jryer)]

Vik,

 

Calling name was working for me coming in from CME to CCM over the SIP
trunk - it did display the name I had set with the station-id command on
the voice-port for the FXS port.  I couldn't tell about the other
direction because the battery was dead in my analog phone and so it
didn't show me caller-id.  It was "connected name" that didn't seem to
work right for me when I called from CCM out to the analog phone on CME.

 

As I work on practice labs over the next five days, I will pay attention
to what happens with calling name and connected name on a SIP trunk
between CCM and CME to IP phones.  I will post my findings here sometime
in the next week.

 

Jane

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vik Malhi
Sent: Tuesday, June 10, 2008 6:32 PM
To: 'OSL CCIE Voice Lab Exam'
Subject: Re: [OSL | CCIE_Voice] calling out to FXS port via SIP
trunk-nocaller-id?

 

I persevered a while back and never got Calling Name to function from
CCM to CME via the SIP trunk. So at the moment it appears that "this is
the way it works" unless somebody has any further insight.

 

BTW- I expect the CME phone to function in the same way as an FXS
portsee below for the reason why.

 

P27-BR2-RTR#sh telephony-service voice-port 

 

voice-port 50/0/1
 station-id number 3001

 station-id name br2 phn2
 timeout ringing 12
!
P27-BR2-RTR#sh telephony-service dial-p

 

dial-peer voice 20001 pots
 destination-pattern 3001$
 huntstop
 progress_ind setup enable 3
 port 50/0/1

 

Vik Malhi - CCIE #13890 
Senior Technical Instructor - IPexpert, Inc. 

Telephone: +1.810.326.1444 
Fax: +1.810.454.0130 
Mailto: [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>  

Join our free online support and peer group communities: 
http://www.IPexpert.com/communities 

IPexpert - The Global Leader in Self-Study, Classroom-Based,
Video-On-Demand and Audio Certification Training Tools for the Cisco
CCIE R&S Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice
Lab and CCIE Storage Lab Certifications.

 

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jane Ryer
(jryer)
Sent: Tuesday, June 10, 2008 3:26 PM
To: OSL CCIE Voice Lab Exam
Subject: Re: [OSL | CCIE_Voice] calling out to FXS port via SIP trunk
-nocaller-id?

Hey, Onur,

 

I will try it tomorrow when I go in to the office and post results here.

 

Jane

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Onur
Tufekci
Sent: Tuesday, June 10, 2008 4:08 PM
To: OSL CCIE Voice Lab Exam
Cc: 
Subject: Re: [OSL | CCIE_Voice] calling out to FXS port via SIP trunk -
nocaller-id?

 

Can you please try:

 

Sip-ua

Remote-party-id

 

I am curious if it works.


On Jun 9, 2008, at 11:11 PM, "Jane Ryer (jryer)" <[EMAIL PROTECTED]>
wrote:

I set up a SIP trunk from Call Manager to a router with an FXS
port.  When I call from the analog phone attached to the FXS port to an
IP Blue phone registered to Call Manager, I do see the name and number
for the FXS port (as set via station-id commands on the voice-port for
the FXS port).  However, if I call out from the IP Blue phone to the
analog phone, all I see on the IP Blue phone is the number I dialed
(4001) - no name.  Is this to be expected with SIP trunks?

 

Here is the relevant portion of my router config:

 

voice-port 0/2/1

 station-id name Analog Phone

 station-id number 2122214001

 caller-id enable   (not sure whether this accomplished anything
or not - didn't work differently with or without it)

!

dial-peer voice 4000 voip

 session protocol sipv2

 session target ipv4:10.x.x.x   (IP address of my CCM)

 incoming called-number 4...

 dtmf-relay rtp-nte   (just realized that I put this command on
this dial peer but not the one to CCM)

 codec g711ulaw

 no vad

!

dial-peer voice 4001 pots

 destination-pattern 4001

 port 0/2/1

!

dial-peer voice 1000 voip

 destination-pattern 1...

 session protocol sipv2

 session target ipv4:10.x.x.x   (IP address of my CCM)

 codec g711ulaw

 no vad

!

 

Any insight would be appreciated.  Is this supposed to work or
not?  Is it just a limitation of SIP?  Or am I missing some
configuration that is needed to pass the called name back?

 

Thanks,

Re: [OSL | CCIE_Voice] calling out to FXS port via SIP trunk - nocaller-id?

[Jane Ryer (jryer)]

Onur,

 

Turns out that that's the default.  I did try disabling it (and it then
showed up in the router's running config as "no remote-party-id"), still
did not see connected name when I called from IP Blue phone (registered
to CCM) to the analog phone (on router FXS port).

 

I also played around with the "calling-info pstn-to-sip remote-party-id
name set " and "calling-info sip-to-pstn name set " commands,
but those only seem to affect calling name, not connected name.

 

I'm done experimenting.  I'm going to trust that not having the
connected name show up in this type of scenario will not cost me points
in the real lab.  Even if it does, I've burned enough of my study time
investigating it.  I just wanted to be sure I wasn't missing something
obvious.

 

Jane

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Onur
Tufekci
Sent: Tuesday, June 10, 2008 4:08 PM
To: OSL CCIE Voice Lab Exam
Cc: 
Subject: Re: [OSL | CCIE_Voice] calling out to FXS port via SIP trunk -
nocaller-id?

 

Can you please try:

 

Sip-ua

Remote-party-id

 

I am curious if it works.


On Jun 9, 2008, at 11:11 PM, "Jane Ryer (jryer)" <[EMAIL PROTECTED]>
wrote:

I set up a SIP trunk from Call Manager to a router with an FXS
port.  When I call from the analog phone attached to the FXS port to an
IP Blue phone registered to Call Manager, I do see the name and number
for the FXS port (as set via station-id commands on the voice-port for
the FXS port).  However, if I call out from the IP Blue phone to the
analog phone, all I see on the IP Blue phone is the number I dialed
(4001) - no name.  Is this to be expected with SIP trunks?

 

Here is the relevant portion of my router config:

 

voice-port 0/2/1

 station-id name Analog Phone

 station-id number 2122214001

 caller-id enable   (not sure whether this accomplished anything
or not - didn't work differently with or without it)

!

dial-peer voice 4000 voip

 session protocol sipv2

 session target ipv4:10.x.x.x   (IP address of my CCM)

 incoming called-number 4...

 dtmf-relay rtp-nte   (just realized that I put this command on
this dial peer but not the one to CCM)

 codec g711ulaw

 no vad

!

dial-peer voice 4001 pots

 destination-pattern 4001

 port 0/2/1

!

dial-peer voice 1000 voip

 destination-pattern 1...

 session protocol sipv2

 session target ipv4:10.x.x.x   (IP address of my CCM)

 codec g711ulaw

 no vad

!

 

Any insight would be appreciated.  Is this supposed to work or
not?  Is it just a limitation of SIP?  Or am I missing some
configuration that is needed to pass the called name back?

 

Thanks,

Re: [OSL | CCIE_Voice] calling out to FXS port via SIP trunk - nocaller-id?

[Jane Ryer (jryer)]

Hey, Onur,

 

I will try it tomorrow when I go in to the office and post results here.

 

Jane

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Onur
Tufekci
Sent: Tuesday, June 10, 2008 4:08 PM
To: OSL CCIE Voice Lab Exam
Cc: 
Subject: Re: [OSL | CCIE_Voice] calling out to FXS port via SIP trunk -
nocaller-id?

 

Can you please try:

 

Sip-ua

Remote-party-id

 

I am curious if it works.


On Jun 9, 2008, at 11:11 PM, "Jane Ryer (jryer)" <[EMAIL PROTECTED]>
wrote:

I set up a SIP trunk from Call Manager to a router with an FXS
port.  When I call from the analog phone attached to the FXS port to an
IP Blue phone registered to Call Manager, I do see the name and number
for the FXS port (as set via station-id commands on the voice-port for
the FXS port).  However, if I call out from the IP Blue phone to the
analog phone, all I see on the IP Blue phone is the number I dialed
(4001) - no name.  Is this to be expected with SIP trunks?

 

Here is the relevant portion of my router config:

 

voice-port 0/2/1

 station-id name Analog Phone

 station-id number 2122214001

 caller-id enable   (not sure whether this accomplished anything
or not - didn't work differently with or without it)

!

dial-peer voice 4000 voip

 session protocol sipv2

 session target ipv4:10.x.x.x   (IP address of my CCM)

 incoming called-number 4...

 dtmf-relay rtp-nte   (just realized that I put this command on
this dial peer but not the one to CCM)

 codec g711ulaw

 no vad

!

dial-peer voice 4001 pots

 destination-pattern 4001

 port 0/2/1

!

dial-peer voice 1000 voip

 destination-pattern 1...

 session protocol sipv2

 session target ipv4:10.x.x.x   (IP address of my CCM)

 codec g711ulaw

 no vad

!

 

Any insight would be appreciated.  Is this supposed to work or
not?  Is it just a limitation of SIP?  Or am I missing some
configuration that is needed to pass the called name back?

 

Thanks,

Re: [OSL | CCIE_Voice] Cisco CUE Module

[Jane Ryer (jryer)]

Is this on a Proctor Labs rack, or your own lab?

 

You need an "ip http server" command in the CME router config.  It is
already there in all of the Proctor Lab routers, but if you are setting
up your own lab, then you may need to enter it.

 

Jane

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul and
Bobs
Sent: Tuesday, June 10, 2008 3:41 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Cisco CUE Module

 

Hi Guys

Struggling on something, the Cue Module. I setup the CME with all the
setting required including the "web admin system name admin password
cisco" and have reset the cisco CUE to factory defaults setting the new
account details to admin and cisco in CUE. When I get to the first setup
page on the web page I get to  login where it says " first time only"
and then when trying to put in the username and password to integrate
with CME I keep getting error to check username. I cant seem to get the
two to synch up.

Does anyone have any ideas on this?

Thanks

Paul

[OSL | CCIE_Voice] calling out to FXS port via SIP trunk - no caller-id?

[Jane Ryer (jryer)]

I set up a SIP trunk from Call Manager to a router with an FXS port.
When I call from the analog phone attached to the FXS port to an IP Blue
phone registered to Call Manager, I do see the name and number for the
FXS port (as set via station-id commands on the voice-port for the FXS
port).  However, if I call out from the IP Blue phone to the analog
phone, all I see on the IP Blue phone is the number I dialed (4001) - no
name.  Is this to be expected with SIP trunks?

 

Here is the relevant portion of my router config:

 

voice-port 0/2/1

 station-id name Analog Phone

 station-id number 2122214001

 caller-id enable   (not sure whether this accomplished anything or not
- didn't work differently with or without it)

!

dial-peer voice 4000 voip

 session protocol sipv2

 session target ipv4:10.x.x.x   (IP address of my CCM)

 incoming called-number 4...

 dtmf-relay rtp-nte   (just realized that I put this command on this
dial peer but not the one to CCM)

 codec g711ulaw

 no vad

!

dial-peer voice 4001 pots

 destination-pattern 4001

 port 0/2/1

!

dial-peer voice 1000 voip

 destination-pattern 1...

 session protocol sipv2

 session target ipv4:10.x.x.x   (IP address of my CCM)

 codec g711ulaw

 no vad

!

 

Any insight would be appreciated.  Is this supposed to work or not?  Is
it just a limitation of SIP?  Or am I missing some configuration that is
needed to pass the called name back?

 

Thanks,

Jane

 

[OSL | CCIE_Voice] review of Volume 3 of new "Box Set" or "Blended Learning Solutin"

[Jane Ryer (jryer)]

I placed my order for Volume 3 of the new workbooks/box set/Blended
Learning Solution on Monday, May 12th.  The .PDF files were loaded onto
my Ipexpert account on Wednesday, May 14th.  (There is one .PDF file for
the ten lab scenarios, and then ten .PDF files for the Proctor Guides
for each of the ten labs.)  At my request, Mike Down also loaded the
updated Workbook volumes 1 and 2 into my account at the same time (I
believe that the upgrade for those is free if you have previously
purchased the old v4 workbook from Ipexpert, but check with the Ipexpert
sales guys on that.)  However, I have not even opened those files yet -
I was more interested in the new practice labs in Volume 3.

 

I was warned at the time that it could be up to three weeks before I
received the hard drive with the video walkthroughs, and that proved to
be precisely correct - I received the hard drive two days ago on Monday,
June 2nd.  The lead time may be less now that the process is worked out
- if time is critical to you (like if you have a lab date scheduled for
late June or early July), then ask before you submit your order.  I was
at a point that having the labs themselves and the proctor guides was as
important to me as the video walkthroughs (my last attempt at the Voice
lab exam was May 8th and my next attempt is in two weeks on June 18th).
So I went ahead and ordered, knowing that I might only have two weeks
with the video walkthroughs before my next attempt.

 

First, an overview - there are ten labs.  The first five are MGCP (for
the BR1 gateway) and Mark does the video walkthrough.  Labs six through
ten are H.323 for the BR1 gateway and Vik does the video walkthrough.
There is a good mixture of all the possible scenarios you might face for
the other elements (various Gatekeeper zones and restrictions, different
multicast or unicast MoH scenarios, different IPCC Express scripts,
IPMA, B-ACD, different QoS scenarios, etc. etc.)

 

During the last three weeks, I have done five of the ten labs.  Four of
them were during 7:45 hour Proctor Labs vRack sessions (which actually
work out to be about seven hours of actual work time if you take breaks
and eat a meal).  I found them to be remarkably similar to the actual
lab exam experience - it was certainly a challenge to complete all the
tasks in seven hours, but it was possible to do so.  For the fifth lab I
did, I wanted to nail QoS, so I booked a double session (fifteen hours
and 45 minutes).  I used about four hours to configure the lab up
through Dial Plan, Media, and High Availability, and then spent a good
eight hours going through every QoS question in all ten labs.  I would
do all the QoS configuration for one lab, go look at the proctor guide
for that lab to check how I did, wipe it all out, and then move on to
the QoS questions in another lab.  That twelve-hour day may be the key
in whether I pass my next attempt at the lab exam or not - QoS has
always taken way too much time for me and I wasn't sure I had it right -
I feel much more confident now.

 

Since I received the hard drive two days ago, I have made it a point to
view the video walkthroughs for the questions that I was not 100%
positive I had configured correctly while working on the five labs that
I've been through.  The menu is very easy to use - you pull up the video
walkthroughs by question (see the demo), so you can just easily just
watch what you are currently working on.  For those of you who have
taken the Ipexpert Boot Camps, this is very similar to the solutions
that were provided on the USB key, except that you have audio as well as
video, and there is a video file (with sound) for every single question,
where the USB keys often just had a text file for router configuration.
The explanations are very thorough and clear - again, those of you have
who have taken the Boot Camps, or even just watched the live vLectures,
know what a gift both Mark and Vik have for teaching.  This is like
having Boot Camp broken up into little snippets so that you can go
straight to the piece you need when you need it, and repeat as many
times as necessary until you throughly understand it.

 

I have two more weeks to do the other five labs before my next attempt
at the actual CCIE Voice lab exam.  If I don't pass this time, it will
not be due to a lack of exposure to all the right material - I can't
imagine what they could possibly put on the lab exam that isn't covered
in these ten labs!  If you discipline yourself to do these labs in less
than eight hours, they also offer a good simulation of the time pressure
of the actual lab exam.  

 

I'll keep you posted - if I do actually pass in two weeks, then
obviously this testimonial will be more credible.  But with my three
weeks of exposure to the labs and proctor guides, and two days of
exposure to the video walkthroughs on the hard drive, I honestly believe
that the $700 I spent for Volume 3 is the best investment I've made yet
in my preparation for the CCIE Voice lab

Re: [OSL | CCIE_Voice] IPexpert Blended Learning Solution (BLS) - FreeLab Download

[Jane Ryer (jryer)]

Hey, Mark,

 

The demo web page says that what you can watch for free is "Lab 2 Media
Resources Task 2.24", but what actually comes up when you click on the
link is task 2.42, which is B-ACD.  (It looks like whoever developed the
web page just transposed digits.)  From a demo perspective, that's
probably not as good because it doesn't show off both Call Manager
config and IOS config, just IOS.  

 

Also, when I received my hard drive two days ago, I couldn't watch the
videos until I downloaded the current version of Flash software from the
link given in the "Help" section of the hard drive:

http://www.adobe.com/shockwave/download/download.cgi?P1_Prod_Version=Sho
ckwaveFlash

 

So, if anyone is trying to watch the demo and nothing happens, try
installing that Flash software first.

 

Jane

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Snow
Sent: Wednesday, June 04, 2008 12:30 PM
To: CCIE Voice Maillist
Subject: [OSL | CCIE_Voice] IPexpert Blended Learning Solution (BLS) -
FreeLab Download
Importance: High

 

Hey folks,

 

Most of you that ordered your HDD (either Box Set or the full BLS)
should have received your HDD's by now.

For those of you that have not ordered yours yet and haven't heard any
feedback from others that have yet (most likely due to the overwhelming
amount of information contained therein :) - can go to this link (below)
to get an idea for the custom designed interface that accompanies the
Hard Drive for easy access of all the material - and also to get an
entire lab (Lab 2) WITH Lab 2 Proctor Guide for free - and 1 of the
video's for Lab 2, along with a few demos of the current Audio and Video
on Demand products.

 

Check out the demo:

IPexpert Voice Blended Learning Solution
 

 

 

Have Fun!

 

p.s. Those of you who have received your HDDs - holler back - tell us
what you think!

 

-- 

Mark Snow

CCIE #14073 (Voice, Security)

 

Senior Technical Instructor - IPexpert, Inc.

 

Telephone: +1.810.326.1444

Fax: +1.309.413.4097

Mailto: [EMAIL PROTECTED]

--

Join our free online support and peer group communities:
http://www.IPexpert.com/communities

--

IPexpert - The Global Leader in Self-Study, Classroom-Based,
Video-On-Demand and Audio Certification Training Tools for the Cisco
CCIE R&S Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice
Lab and CCIE Storage Lab Certifications.

--

 

Re: [OSL | CCIE_Voice] New Study Guide

[Jane Ryer (jryer)]

I just received the hard drive.  (So, the three week estimate was pretty
accurate.)  I will post a review later this week when I've had a chance
to watch some of the video clips.

 

The PDF files for the ten new labs and solutions were posted to my
Ipexpert account several weeks ago (soon after I previously posted).  I
have done four of the ten labs, with plans to do a fifth one during a
4:00 - midnight EDT vRack session today.  The labs are excellent, even
without the video solutions.  I found them to be similar to the two
BootCamp labs (which I had previously been using for my preparation for
the lab exam), but a little more achievable in eight hours.  I have made
two attempts at the Voice lab exam, and in my opinion the ten new labs
are very similar to the amount of configuration that I saw in the actual
lab exams.  Especially the first BootCamp lab was considerably more
complex/detailed and tough to finish in one eight-hour (well, actually
7:45 minus meals/breaks) vRack session.

 

Jane

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jane Ryer
(jryer)
Sent: Tuesday, May 13, 2008 4:45 PM
To: Onur Tufekci; CCIE Voice
Subject: Re: [OSL | CCIE_Voice] New Study Guide

 

Hi, Onur,

 

I've ordered Volume 3.  It's not shipping yet - the sales guys said it
could be up to three weeks before the hard drives are shipped, but they
are trying to make it happen sooner.  I'll be working my way through the
ten labs soon after I receive them and I'll post a review to this list.

 

Based on the video clips that were provided on the USB key from the
Ipexpert Voice boot camp, I am expecting that these ten new labs will be
extremely helpful in preparation for the lab exam.

 

There will also be some updates to the workbook and proctor guide that
you already have (now known as volumes 1 and 2).  Contact IPexpert
sales, because my understanding is that electronic updates will be
provided free of charge to previous customers.

 

Jane

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Onur
Tufekci
Sent: Tuesday, May 13, 2008 4:13 PM
To: CCIE Voice
Subject: [OSL | CCIE_Voice] New Study Guide

 

Hello Everyone,

Did anybody buy the new study guide yet? I am interested in the section
3 actually. Are there any feedbacks on it from any one of you? I already
have the V4 workbook kit.

Regards,

Onur.

Re: [OSL | CCIE_Voice] Thanks. I finally passed.

[Jane Ryer (jryer)]

Congratulations, Scott!   It's great to know that there is light at the
end of the tunnel.  I will make my third attempt on June 18th.

 

Someone else asked about how to get into the lab every 30-40 days.
Check the scheduling site multiple times every day - slots do open up,
often at the 28-day mark when people reach the cancellation deadline.
As a matter of fact, slots for June 5th, 6th, and 10th in SJ sat
unclaimed for at least three or four days in the last two weeks.  

 

Scott mentioned that he had heavily used the Proctor Labs vRacks in his
preparation.  I have relied solely on Proctor Labs vRacks in my
preparation over the last six months or so - the only equipment I have
at home is one router to set up the EasyVPN tunnel, and four IP phones.
I think I've used somewhere around 60 eight-hour sessions at this point,
not counting the two weeks of BootCamp access.  Since we are on the
topic of saying "thanks" to Ipexpert today, I wanted to add my thanks to
Glenn Champine and Drew LePla, the two main tech support guys for the
Proctor Labs racks.  They have responded many times to my after-hours
pages to reset various pieces of hardware, and are always responsive and
easy to work with.  It can't be easy for CCNP-level engineers to support
a bunch of engineers trying to achieve CCIE certifications, but they
have been consistently polite to me, honest about their limitations, and
willing to work with me when I make suggestions about how to fix
something I think is broken.  Mark and Vik are the public face of
Ipexpert (on the Voice side) and I also thank them for all their
expertise and guidance.  I just wanted to also acknowledge how important
the "behind the scenes" guys are to my preparation for the lab exam.

 

Jane

 

Jane Ryer, CCIE #   (R&S)

Network Consulting Engineer

Cisco Systems Advanced Services team

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of IPheaders
Sent: Friday, May 23, 2008 1:46 PM
To: CCIE Voice
Subject: [OSL | CCIE_Voice] Thanks. I finally passed.

 

I just wanted to take a moment to says thanks to Mark, Vik, and everyone
else at IPExpert for the countless hours they put in to develop a
quality product and going the extra mile to help moderate this forum and
to help keep us straight. I also wanted to thank everyone that
participates in this forum as well. I have never engaged in any type of
forum in the past but found myself thorougly enjoying my experience with
this one.

 

I did just recently pass my voice IE lab last week. I'm sharing this not
to boast, but rather to encourage everyone to keep working hard and to
promote the affectiveness of this forum and IPExpert's product. I wish
the best of luck to everyone and I will continue to check in on this
forum and help contribute whenever I can.

 

Cheers,

Scott - CCIE #20903

Re: [OSL | CCIE_Voice] Problem with ISDN layer not coming up

[Jane Ryer (jryer)]

One other thing to look at - you would have had to shut down the voice port in 
order to remove the "service mgcp" from the controller.  Did you remember to 
"no shut" it after the reconfiguration?

Once it comes up "Multiple Frame Established" without mgcp, then you know the 
connection to the PSTN router is good, and you can start trying to re-introduce 
mgcp.  I've never used ccm-manger config (I just always do it manually), so I'm 
not sure what issues you might run into with that.

Jane


-Original Message-
From: Clawson, Von [mailto:[EMAIL PROTECTED] 
Sent: Friday, May 16, 2008 9:03 AM
To: Jane Ryer (jryer)
Cc: ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] Problem with ISDN layer not coming up

Um, yeah I tried contacting live support and got nothing from them.  They told 
me to come here.


Von Clawson
Manager, Nexus IS Technical Assistance Center
Nexus IS, Inc.
Office: :480.517.6672 Mobile: :480.200.1918 Fax:  480.517.6673 
[EMAIL PROTECTED]


-----Original Message-
From: Jane Ryer (jryer) [mailto:[EMAIL PROTECTED] 
Sent: Friday, May 16, 2008 9:00 AM
To: Clawson, Von
Cc: ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] Problem with ISDN layer not coming up

Leave it with just pri-group timeslots 1-3 (no mgcp), and reboot your PSTN 
router (there's a button near the bottom of the Proctor Labs web page).

I've had to do that once or twice in the past.

If that doesn't work, use the troubleshooting menu on the PSTN router (telnet 
to 10.x.200.2) to see what's being reported on that side and contact the 
Proctor Labs support guys.

Jane

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Clawson, Von
Sent: Friday, May 16, 2008 8:55 AM
To: Tyson Scott
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Problem with ISDN layer not coming up

I removed service mgcp and just brought it up with pri-group timeslots 1-3.
Still only get TEI_Assigned


Von Clawson
Manager, Nexus IS Technical Assistance Center
Nexus IS, Inc.
Office: :480.517.6672 Mobile: :480.200.1918 Fax:  480.517.6673 
[EMAIL PROTECTED]


-Original Message-
From: Tyson Scott [mailto:[EMAIL PROTECTED] 
Sent: Friday, May 16, 2008 8:50 AM
To: Clawson, Von
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Problem with ISDN layer not coming up

Von,
Have you tested to make sure the ISDN comes up before you register it
with callmanager? I recommend removing your mgcp configuration from
the trunk and get the trunk to come up with just the basic
configuration.  Once you have this add the mgcp configuration.  It
will help you narrow down where the problem is occuring.

On Fri, May 16, 2008 at 10:36 AM, Clawson, Von <[EMAIL PROTECTED]> wrote:
> I've now had two sessions in which I could not get ISDN layer 3 up on the
> BR1 router.  The gateway registers fine.  The config is below.  Can anyone
> help me figure out what I'm doing wrong?
>
> By the way, I'm in the middle of a session now so any help would be
> appreciated.
>
>
>
> Current configuration : 3544 bytes
>
> !
>
> version 12.4
>
> service timestamps debug datetime msec
>
> service timestamps log datetime msec
>
> no service password-encryption
>
> !
>
> hostname P4-BR1-RTR
>
> !
>
> boot-start-marker
>
> boot system flash:c2800nm-adventerprisek9_ivs-mz.124-3g.bin.bin
>
> boot-end-marker
>
> !
>
> !
>
> no aaa new-model
>
> !
>
> resource policy
>
> !
>
> network-clock-participate wic 0
>
> network-clock-select 1 T1 0/0/0
>
> ip subnet-zero
>
> !
>
> !
>
> ip cef
>
> !
>
> !
>
> !
>
> isdn switch-type primary-ni
>
> !
>
> voice-card 0
>
>  no dspfarm
>
> !
>
> !
>
> !
>
> !
>
> !
>
> !
>
> !
>
> !
>
> !
>
> !
>
> !
>
> !
>
> !
>
> !
>
> !
>
> application
>
>  global
>
>   service alternate default
>
>  !
>
> !
>
> !
>
> !
>
> controller T1 0/0/0
>
>  framing esf
>
>  linecode b8zs
>
>  pri-group timeslots 1-3,24 service mgcp
>
> !
>
> !
>
> !
>
> !
>
> !
>
> interface Loopback0
>
>  ip address 172.4.101.1 255.255.255.255
>
>  ip ospf network point-to-point
>
> !
>
> interface FastEthernet0/0
>
>  no ip address
>
>  duplex auto
>
>  speed auto
>
> !
>
> interface FastEthernet0/1
>
>  no ip address
>
>  duplex auto
>
>  speed auto
>
> !
>
> interface Serial0/0/0:23
>
>  no ip address
>
>  isdn switch-type primary-ni
>
>  isdn incoming-voice voice
>
>  isdn bind-l3 ccm-manager
>
>  no cdp

Re: [OSL | CCIE_Voice] Problem with ISDN layer not coming up

[Jane Ryer (jryer)]

Leave it with just pri-group timeslots 1-3 (no mgcp), and reboot your PSTN 
router (there's a button near the bottom of the Proctor Labs web page).

I've had to do that once or twice in the past.

If that doesn't work, use the troubleshooting menu on the PSTN router (telnet 
to 10.x.200.2) to see what's being reported on that side and contact the 
Proctor Labs support guys.

Jane

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Clawson, Von
Sent: Friday, May 16, 2008 8:55 AM
To: Tyson Scott
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Problem with ISDN layer not coming up

I removed service mgcp and just brought it up with pri-group timeslots 1-3.
Still only get TEI_Assigned


Von Clawson
Manager, Nexus IS Technical Assistance Center
Nexus IS, Inc.
Office: :480.517.6672 Mobile: :480.200.1918 Fax:  480.517.6673 
[EMAIL PROTECTED]


-Original Message-
From: Tyson Scott [mailto:[EMAIL PROTECTED] 
Sent: Friday, May 16, 2008 8:50 AM
To: Clawson, Von
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Problem with ISDN layer not coming up

Von,
Have you tested to make sure the ISDN comes up before you register it
with callmanager? I recommend removing your mgcp configuration from
the trunk and get the trunk to come up with just the basic
configuration.  Once you have this add the mgcp configuration.  It
will help you narrow down where the problem is occuring.

On Fri, May 16, 2008 at 10:36 AM, Clawson, Von <[EMAIL PROTECTED]> wrote:
> I've now had two sessions in which I could not get ISDN layer 3 up on the
> BR1 router.  The gateway registers fine.  The config is below.  Can anyone
> help me figure out what I'm doing wrong?
>
> By the way, I'm in the middle of a session now so any help would be
> appreciated.
>
>
>
> Current configuration : 3544 bytes
>
> !
>
> version 12.4
>
> service timestamps debug datetime msec
>
> service timestamps log datetime msec
>
> no service password-encryption
>
> !
>
> hostname P4-BR1-RTR
>
> !
>
> boot-start-marker
>
> boot system flash:c2800nm-adventerprisek9_ivs-mz.124-3g.bin.bin
>
> boot-end-marker
>
> !
>
> !
>
> no aaa new-model
>
> !
>
> resource policy
>
> !
>
> network-clock-participate wic 0
>
> network-clock-select 1 T1 0/0/0
>
> ip subnet-zero
>
> !
>
> !
>
> ip cef
>
> !
>
> !
>
> !
>
> isdn switch-type primary-ni
>
> !
>
> voice-card 0
>
>  no dspfarm
>
> !
>
> !
>
> !
>
> !
>
> !
>
> !
>
> !
>
> !
>
> !
>
> !
>
> !
>
> !
>
> !
>
> !
>
> !
>
> application
>
>  global
>
>   service alternate default
>
>  !
>
> !
>
> !
>
> !
>
> controller T1 0/0/0
>
>  framing esf
>
>  linecode b8zs
>
>  pri-group timeslots 1-3,24 service mgcp
>
> !
>
> !
>
> !
>
> !
>
> !
>
> interface Loopback0
>
>  ip address 172.4.101.1 255.255.255.255
>
>  ip ospf network point-to-point
>
> !
>
> interface FastEthernet0/0
>
>  no ip address
>
>  duplex auto
>
>  speed auto
>
> !
>
> interface FastEthernet0/1
>
>  no ip address
>
>  duplex auto
>
>  speed auto
>
> !
>
> interface Serial0/0/0:23
>
>  no ip address
>
>  isdn switch-type primary-ni
>
>  isdn incoming-voice voice
>
>  isdn bind-l3 ccm-manager
>
>  no cdp enable
>
> !
>
> interface Serial0/1/0
>
>  no ip address
>
>  encapsulation frame-relay IETF
>
>  no fair-queue
>
>  frame-relay lmi-type ansi
>
> !
>
> interface Serial0/1/0.1 point-to-point
>
>  ip address 162.4.101.2 255.255.255.0
>
>  ip ospf mtu-ignore
>
>  frame-relay interface-dlci 101
>
> !
>
> interface FastEthernet1/0
>
>  switchport trunk native vlan 140
>
>  switchport mode trunk
>
>  switchport voice vlan 240
>
> !
>
> interface FastEthernet1/1
>
>  shutdown
>
> !
>
> interface FastEthernet1/2
>
>  shutdown
>
> !
>
> interface FastEthernet1/3
>
>  shutdown
>
> !
>
> interface FastEthernet1/4
>
>  shutdown
>
> !
>
> interface FastEthernet1/5
>
>  shutdown
>
> !
>
> interface FastEthernet1/6
>
>  shutdown
>
> !
>
> interface FastEthernet1/7
>
>  shutdown
>
> !
>
> interface FastEthernet1/8
>
>  switchport trunk native vlan 140
>
>  switchport mode trunk
>
>  switchport voice vlan 240
>
> !
>
> interface FastEthernet1/9
>
>  shutdown
>
> !
>
> interface FastEthernet1/10
>
>  shutdown
>
> !
>
> interface FastEthernet1/11
>
>  shutdown
>
> !
>
> interface FastEthernet1/12
>
>  shutdown
>
> !
>
> interface FastEthernet1/13
>
>  shutdown
>
> !
>
> interface FastEthernet1/14
>
>  shutdown
>
> !
>
> interface FastEthernet1/15
>
>  shutdown
>
> !
>
> interface Vlan1
>
>  no ip address
>
>  shutdown
>
> !
>
> interface Vlan240
>
>  ip address 10.4.201.1 255.255.255.0
>
>  ip helper-address 10.4.200.21
>
> !
>
> router ospf 1
>
>  log-adjacency-changes
>
>  network 10.4.101.0 0.0.0.255 area 0
>
>  network 10.4.201.0 0.0.0.255 area 0
>
>  network 162.4.101.0 0.0.0.255 area 0
>
>  network 172.4.101.0 0.0.0.255 area 0
>
> !
>
> ip classless
>
> !
>
> !
>
> ip http server
>
> no ip http secure-server
>
> !
>
> !
>
> !
>
> !
>
> !
>
> control-plane
>
> !
>
> !
>
> !
>
> voice-port 0/0/0:23
>
> !
>
> 

Re: [OSL | CCIE_Voice] New Study Guide

[Jane Ryer (jryer)]

Hi, Onur,

 

I've ordered Volume 3.  It's not shipping yet - the sales guys said it
could be up to three weeks before the hard drives are shipped, but they
are trying to make it happen sooner.  I'll be working my way through the
ten labs soon after I receive them and I'll post a review to this list.

 

Based on the video clips that were provided on the USB key from the
Ipexpert Voice boot camp, I am expecting that these ten new labs will be
extremely helpful in preparation for the lab exam.

 

There will also be some updates to the workbook and proctor guide that
you already have (now known as volumes 1 and 2).  Contact IPexpert
sales, because my understanding is that electronic updates will be
provided free of charge to previous customers.

 

Jane

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Onur
Tufekci
Sent: Tuesday, May 13, 2008 4:13 PM
To: CCIE Voice
Subject: [OSL | CCIE_Voice] New Study Guide

 

Hello Everyone,

Did anybody buy the new study guide yet? I am interested in the section
3 actually. Are there any feedbacks on it from any one of you? I already
have the V4 workbook kit.

Regards,

Onur.

Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 27, Issue 21

[Jane Ryer (jryer)]

I found it easiest to just memorize the url, rather than worrying about
whether the documentation would be available in the lab.

 

Jane

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vladimir
Gotsev
Sent: Friday, May 09, 2008 11:33 PM
To: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 27, Issue 21

 

Can you share ideas on how to find or memorize the IPPA service URL?

On Fri, May 9, 2008 at 1:43 AM, <[EMAIL PROTECTED]>
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
   [EMAIL PROTECTED]

You can reach the person managing the list at
   [EMAIL PROTECTED]

When replying, please edit your Subject line so it is more specific
than "Re: Contents of CCIE_Voice digest..."


Today's Topics:

  1. qos marking ideas !!! (Djokic Sinisa)


--

Message: 1
Date: Fri, 9 May 2008 09:42:51 +0200
From: "Djokic Sinisa" <[EMAIL PROTECTED]>
Subject: [OSL | CCIE_Voice] qos marking ideas !!!
To: 
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"



hi team..

i'm new on this list and have some concerns about QoS..

so, maybe someone can help..

so, the thing is that i want to mark signalig traffic ( h323, sccp,
mgcp,
ras, sip ) on HQ-RTR, RSB-RTR and RSC-RTR, and NOT trust markings on the
switches or to remark on them..



so, this is idea how to do it, but i have some concerns as you would see
and
doubts about it..



so, if anyone has idea how to do it i'd appreciated it..



so here it is..







for HQ-RTR



!

ip access-list extended CONTROL-HQ

 permit tcp any range 2000 2002 any
ccm-to-phones   we need cover both directions i
think - it all goes over the same subinf on HQ-RTR

 permit tcp any any range 2000 2002
phones-to-ccm   we need cover both directions i
think - it all goes over the same subinf on HQ-RTR

 permit tcp any eq 2428 any
ccm-to-mgcp-gw

 permit tcp any any eq 2428
mgcp-gw-to-ccm6608-to-ccm, for RSB-RTR-to-ccm it
goes on RSB-RTR

 permit udp any eq 2427 any
ccm-to-mgcp-gw

 permit udp any any eq 2427
mgcp-gw-to-ccm6608-to-ccm, for RSB-RTR-to-ccm it
goes on RSB-RTR

 permit tcp any any eq 1720

 permit udp any eq 1719 any
ccm-to-gk  vice-versa gk-to-ccm i'm not
shure should i do it and how to do it


 permit tcp any any eq 1718
ccm-to-gk

 permit udp any any eq 5060
ccm-to-sip-gwvice versa handles ip qos
command
under dial-peer

 permit tcp any any eq 5060
ccm-to-sip-gwvice versa handles ip qos
command
under dial-peer

!

class-map match-any CONTROL-HQ

 match access-group name CONTROL-HQ

!

policy-map MARK

class CONTROL-HQ

 set dscp cs3

 class class-default
I SUPPOSE IT MUST be default one as well, beacuse if not we may have
undesired data traffic with "wrong" marking traversing our WAN

 set dscp default
but it so, there must be class for RTP as well or it would be remarked
and
that's bad

!

interface FastEthernet0/0.XY - VOICE ONE


 service-policy input MARK

!

interface FastEthernet0/0.XY  - DATA ONE
SHOULD i put in on data subinterface as well - the same reason as above,
should i take care of potentially uneamted traffic form data vlan

 service-policy input MARK





this makes sence to put only in HQ ethernet ingress subinterfaces -
input..

i'm not shure should it be put on data as well..i think yes..





for RSB-RTR





ip access-list extended CONTROL-RSB

 permit tcp any any range 2000 2002
phones-to-ccm

!

class-map match-any CONTROL-RSB

 match access-group name CONTROL-RSB

!

policy-map MARK

class CONTROL-RSB

 set dscp cs3

 class class-default
I SUPPOSE IT MUST be default one as well, beacuse if not we may have
undesired data traffic with "wrong" marking traversing our link

 set dscp default



the only place i can think of this have sense to put is in input
direction
on "interface Vlan XY '( voice ) as well as in input direction on
"interface
Vlan XY" ( data )..







mgcp ip qos dscp cs3 control
handles mgcp originating from router

no H323 from-and-to-RSB
when srst is working, then wan is down

no RAS from-and-to RSB

no SIP from-and-to RSB

!



for RSB-RTR



ip access-list extended CONTROL-RSB

 permit tcp any any range 2000 2002
phones-to-ccm   ALTHOUGH i don't se point to mark SCCP on RSC since
doesn't
traverse WAN

!

class-map match-any CONTROL-RSB

 match access-group name CONTROL-RSB

!

policy-map MARK

class CONTROL-RSB

 set dscp cs3

 class class-default
I SUPPOSE IT MUST be default one as well, beacuse if not we may have
undesired data traffic with "wrong" marking 

Re: [OSL | CCIE_Voice] Troubleshoot Multicast MOH

[Jane Ryer (jryer)]

Base multicast address of 239.1.1.3 is for a G711 stream, then it goes
up by one for the other types of codecs.  Since G729 is the third
possibility of the four, the address you want is 239.1.1.5 in your
configuration.

 

A more typical configuration would be to set .1 as the base multicast
address, then your router config (.3 for G729) would be correct.

 

Also on the router, did you specify the IP addresses for which
interfaces to send the multicast stream out?  You would want the Voice
VLAN interface address and the Loopback address (so that calls from the
PSTN would hear it).

 

Good luck!

Jane

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ccievoice1
Sent: Friday, May 09, 2008 9:14 PM
To: CCIE Voice Maillist
Subject: [OSL | CCIE_Voice] Troubleshoot Multicast MOH

 

Hi all,

Not able to get Multicast MOH to work for remote site using g729.
Instead, hearing Tone on Hold only. No problem with HQ site Multicast
MOH.

I configured:
1. ip multicast routing and ip pim dense-mode in router
2. ccm-manager music-on-hold | moh music-on-hold.au | multicast moh
239.1.1.3 port 16384 configured in router
3. Enable Multicast Audio Sources on this MOH Server
4. Base Multicast IP Address = 239.1.1.3
5. Base Multicast Port Number = 16384
6. Increment Multicast on = IP Address
7. Max Hops = 5
8. MOH Audio Source:  SampleAudioSource (1) = Allow Multicasting
9. In Media Resource Group =  Use Multicast for MOH Audio (requires at
least one multicast MOH resource)
10. Cisco IP Voice Media Streaming App = 711 mulaw and 729 Annex A
selected.
11. Media Resource Group List configured in Device Pool level with
appropriate MRG.
12.MOHMulticastResourceActive counter got incremented when put the call
On-Hold.
13. Reboot all the devices
Perhaps, anyone know some good steps in troubleshooting multicast moh
issue?

Thanks.

Re: [OSL | CCIE_Voice] Configure BR1 Router as H323 Gateway

[Jane Ryer (jryer)]

Make sure that you are not stripping the "9" in the BR1 route group
within the route list on CCM.   (in other words, don't select "predot")

 

Jane

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Devildoc
Sent: Friday, April 25, 2008 3:09 PM
To: Justin Steinberg
Cc: CCIE Voice Online Study List
Subject: Re: [OSL | CCIE_Voice] Configure BR1 Router as H323 Gateway

 

I guess i forgot that part.  That makes sense.  However, that part of
configuration is only for incoming calls from the PSTN.  How about the
outgoing calls to the PSTN?  I couldn't make any outboudn call to the
PSTN.  I created a route pattern with a route list containing BR1 in CCM
but I couldn't make the call.
 
JD





Date: Fri, 25 Apr 2008 16:55:47 -0400
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [OSL | CCIE_Voice] Configure BR1 Router as H323 Gateway
CC: ccie_voice@onlinestudylist.com

you need to point your BR1 DID range to the callmanager IP's. 

h323 is peer to peer.  it doesn't know about callmanager unless you tell
it.


dial-peer voice 2000 voip
destination-pattern 6175212...
session target ipv4:10.21.200.20  

dtmf-relay h245-alphanumeric
!
dial-peer voice 2001 voip
preference 1
destination-pattern 6175212...
session target ipv4:10.21.200.21  

dtmf-relay h245-alphanumeric

then on callmanager you need to accept the 10 digit DID, and translate
that to your DN.  either via the sig digits gateway param, xlate
pattern, or 10 digit dn.



On Fri, Apr 25, 2008 at 4:49 PM, Devildoc <[EMAIL PROTECTED]>
wrote:

Hello,
 
Has anyone successfully configured BR1 router as an H323 Gateway in CCM
for making calls to and receiving calls from the PSTN?  I couldn't get
it to work.  Here is what I did.
 
1. Create an H323 gateway in CCM with BR1's Loopback IP address as the
name of the gateway.
2. And here is the partial configuration on the BR1 router that i think
is relevant to the H323 configuration.
 
BR1 Router Configuration

 
network-clock-participate wic 0
 
isdn switch-type primary-ni
 
controller T1 0/0/0
 framing esf
 linecode b8zs
 pri-group timeslot 1-3
 
voice service voip
 allow-connections h323 to h323
 
 
int loopback0
 ip address 172.21.101.1   255.255.255.0
 
 h323-gateway voip interface
 h323-gateway voip bind srcaddr 172.21.101.1  
 
interface Serial0/0/0:23
 no ip address
 isdn switch-type primary-ni
 isdn incoming-voice voice
 isdn outgoing display-ie
 isdn outgoing ie redirecting-number
 isdn bchan-number-order ascending
 no cdp enable
 
dial-peer voice 1 pots
 incoming called-number .
 port 0/0/0:23
 direct-inward-dial
 
dial-peer voice 10 pots
 destination-pattern 911
 port 0/0/0:23
 no digit-strip
 
dial-peer voice 20 pots
 destination-pattern 91[2-9]..[2-9]..
 port 0/0/0:23
 forward-digit 11
 
dial-peer voice 100 voip
 incoming called-number .
 dtmf-relay h245-alphanumeric
 
 
I did a show isdn status and it showed active layer 1 and 2 and
multiple_frames_established.  But when i made a call from BR1 phone 3 to
911, it didn't go through.  I got that announcement "your call cannot be
completed as dial...".  Am i missing something in my configuration to
set up BR1 router as a proper H323 gateway?  Does anyone have any idea?
Thanks for any info.
 
JD
 
 
 



Spell a grand slam in this game where word skill meets World Series. Get
in the game.
 

 

 



Back to work after baby- how do you know when you're ready?
 

Re: [OSL | CCIE_Voice] UniverCD is toast...

[Jane Ryer (jryer)]

I attempted the lab in RTP on March 11th.  The file was definitely the
CAD installation guide, and I'm pretty sure that it was version 6.1 (for
IPCC Express Release 4.0).

 

Jane



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Santry,
Ryan
Sent: Wednesday, April 23, 2008 10:51 AM
To: Scott Monasmith; Edward French
Cc: CCIE Maillist
Subject: Re: [OSL | CCIE_Voice] UniverCD is toast...

 

Do you know what version of the document was on the desktop?

 

Thanks

 

Ryan Santry

Business Communications 

Senior Technical Support

Sentinel Technologies, Inc

2550 Warrenville Road

Downers Grove, IL,60515

SNR: 630-769-4394

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott
Monasmith
Sent: Monday, April 21, 2008 11:07 AM
To: Edward French
Cc: CCIE Maillist
Subject: Re: [OSL | CCIE_Voice] UniverCD is toast...

 

In RTP, I noticed an IPCC admin .PDF on the desktop now.

On Sun, Apr 20, 2008 at 10:02 AM, Edward French
<[EMAIL PROTECTED]> wrote:

My problem was the url for the phone agent

- Original Message 
From: Jonathan Charles <[EMAIL PROTECTED]>

To: Edward French <[EMAIL PROTECTED]>
Cc: CCIE Voice 
Sent: Sunday, April 20, 2008 10:45:43 AM
Subject: Re: [OSL | CCIE_Voice] UniverCD is toast...

Well, um... there is a probability that we would need that too...



Jonathan

On Sat, Apr 19, 2008 at 7:36 PM, Edward French
<[EMAIL PROTECTED]> wrote:
>
> Friday I had access to everything I needed except IPCCX
>
>
> - Original Message 
> From: Jonathan Charles <[EMAIL PROTECTED]>
> To: CCIE Voice 
> Sent: Saturday, April 19, 2008 11:03:35 AM
> Subject: [OSL | CCIE_Voice] UniverCD is toast...
>
>  So, what, if anything, do we get to use on the lab?
>
>
>
> Jonathan
>
>

 




-- 
"There are only 10 types of people in the world: Those who understand
binary, and those who don't" 

[OSL | CCIE_Voice] take exam in RTP or SJ ?

[Jane Ryer (jryer)]

Be sure to consider time zone changes when you make your decision about
whether to schedule your lab exam in RTP or in SJ.

 

I live in Denver (Mountain Time Zone) and scheduled my first lab attempt
on Tuesday, March 11th in RTP.  I didn't even consider the two-hour time
difference between MST and EST when choosing whether to take the lab in
RTP or SJ.  I grew up in North Carolina, and make 3-4 trips a year to
the NC/SC area, and rarely have any issues shifting gears when I travel
there.

 

BUT, I got caught by two things I did not anticipate.  The daylight
savings time shift was early this year - on Saturday, March 8th.  That
turned my two-hour time difference into a three-hour time difference.  I
had not checked what time the lab started until the week before I was
scheduled to take the test.  It turns out that the exam starts at 7:15
in RTP but not until 8:00 in SJ.  So, with the now three-hour time
difference, I started my lab attempt at 4:15 a.m. body clock time.  I
also had not realized until the night before that my hotel didn't start
serving breakfast until 6:30 a.m., so I rushed to eat breakfast and
still get to the lab by 7:15 a.m.


I wouldn't have passed on my first attempt anyway, but I certainly
didn't do myself a favor by scheduling that attempt in RTP rather than
SJ.

 

My advice is that if you plan to travel to RTP from any time zone other
than Eastern, it might not hurt to arrive two days before your scheduled
lab date, rather than the day before.  Be aware that you need to be at
Cisco at 7:00 a.m., and plan ahead about breakfast.

 

Good luck to all the rest of you who are attempting the lab!

 

Jane

Re: [OSL | CCIE_Voice] QoS SRND in lab

[Jane Ryer (jryer)]

Hi, Mark,

Here's what I'm looking at:

QoS SRND version 3.3 - page 2-5 into 2-6

Call Signaling Ports
In this design chapter, only Skinny Call Control Protocol (SCCP) ports
(TCP Ports 2000-2002) are used to identify call signaling protocols to
keep the examples relatively simple.  However, SCCP is by no means the
only call signaling protocol used in IP telephony environments.  Cisco
recommends including all relevant call signaling ports required for a
given IPT environment in the access lists that identify call signaling
protocols.  Firewalls protecting CallManagers should also allow
additional ports to provide the supplementary services that CallManagers
provide or require.

And then all their call-signaling ACL examples only include this line:

permit tcp 10.1.110.0 0.0.0.255 any range 2000 2002

not the other lines shown in the Ipexpert Bootcamp slide titled
"Classification for Server" (which is slide 259 in my printed copy of
the bootcamp slides.

I'd like to find an accurate list of TCP/UDP ports to memorize since it
is not in the SRND any more.  I made a written note that the bootcamp
printed slide was incorrect - the second line (which I noted was for GK
discovery) should have been udp port 1718 instead of tcp port 1718.  I
also made a note on that slide that Mark specifically mentioned cutting
and pasting this particular ACL from the SRND, which is not possible
anymore.

Not a big deal, as someone else mentioned it's a good idea to know all
these TCP and UDP ports anyway.  I just would prefer not to spend my
time reinventing the wheel (developing the list of TCP and UDP ports to
memorize) if it's accurately documented somewhere.

Jane

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Snow
Sent: Thursday, March 27, 2008 12:08 PM
To: Jane Ryer (jryer)
Cc: CCIE Voice Online Study List
Subject: Re: [OSL | CCIE_Voice] QoS SRND in lab

Which page of which version of the QoS SRND are you referring to?
Just so I can look it up and be on the same page (pun intended :) with  
you to accurately address your concern/question.

-- 
Mark Snow
CCIE #14073 (Voice, Security)
CCSI #31583

Senior Technical Instructor - IPexpert, Inc.
A Cisco Learning Partner - We Accept Learning Credits!

Telephone: +1.810.326.1444
Fax: +1.309.413.4097
Mailto: [EMAIL PROTECTED]
--
Join our free online support and peer group communities:
http://www.IPexpert.com/communities
--
IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On- 
Demand and Audio Certification Training Tools for the Cisco CCIE R&S  
Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and  
CCIE Storage Lab Certifications.
--

On Mar 27, 2008, at 2:01 PM, Jane Ryer (jryer) wrote:
> Hello, Mark,
>
> I'll forward below the previous post that describes the ACL that we're
> concerned about.  I can't find it in the QoS SRND v3.3.  That  
> version of
> the SRND instead lists the first line (tcp any range 2000 2002 any)  
> and
> then makes a comment about possibly needing other TCP and UDP ports.
>
> Unless I can find this in the documentation that is available in the
> lab, I intend to memorize the list of TCP and UDP ports before my next
> lab attempt.
>
> Jane
>
> 
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Edward
> French
> Sent: Sunday, March 16, 2008 5:01 PM
> To: CCIE Maillist
> Subject: [OSL | CCIE_Voice] QOS 6500
>
> Lab 25 task 46 says to
> Configure the Catalyst 6500 to mark all VOIP control traffic from the
> CallManager as AF31.
>
> Which would be:
> set port qos 3/23 port-based
> set qos acl ip SERVER dscp 26 tcp any range 2000 2002 any
> set qos acl ip SERVER dscp 26 tcp any any range 11000 11999
> set qos acl ip SERVER dscp 26 tcp any any range 1024 4999
> set qos acl ip SERVER dscp 26 tcp any any range 1719 1720
> set qos acl ip SERVER dscp 26 udp any eq 2427 any
> set qos acl ip SERVER dscp 26 tcp any eq 2428 any
> commit qos acl SERVER
> set qos acl map SERVER 3/23
>
>
> And in the DVD and Audio it says the best thing to do is copy the QOS
> commands from the SRND. However the above commands are not in the  
> SRND.
> The SRND say do the following
>
>
> set port qos 3/23 trust trust-dscp
>
> or on a 2Q2T port
>
> set qos acl ip TRUST-DSCP trust-dscp any
> commit qos acl TRUST-DSCP
> set qos acl map TRUST-DSCP 3/23
>
> My question is do the commands from the SRND meet the requirements of
> the task? And if not is there an easier way than the first solution,
> since this ACL is not in the SRND? Also given that the first  
> solution is
> an older solution is this what I should expect on the lab?
>
> Basically what is the easiest way to find and copy the QOS statements
> for the 6500? I do not have access to one outside of the lab so I  
> would
> appreciate any tricks or tips rather than memorization.
>
> Thanks
>
> Ed

Re: [OSL | CCIE_Voice] QoS SRND in lab

[Jane Ryer (jryer)]

Hello, Mark,

I'll forward below the previous post that describes the ACL that we're
concerned about.  I can't find it in the QoS SRND v3.3.  That version of
the SRND instead lists the first line (tcp any range 2000 2002 any) and
then makes a comment about possibly needing other TCP and UDP ports.  

Unless I can find this in the documentation that is available in the
lab, I intend to memorize the list of TCP and UDP ports before my next
lab attempt.

Jane


From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Edward
French
Sent: Sunday, March 16, 2008 5:01 PM
To: CCIE Maillist
Subject: [OSL | CCIE_Voice] QOS 6500

Lab 25 task 46 says to 
Configure the Catalyst 6500 to mark all VOIP control traffic from the
CallManager as AF31.

Which would be:
set port qos 3/23 port-based
set qos acl ip SERVER dscp 26 tcp any range 2000 2002 any
set qos acl ip SERVER dscp 26 tcp any any range 11000 11999
set qos acl ip SERVER dscp 26 tcp any any range 1024 4999
set qos acl ip SERVER dscp 26 tcp any any range 1719 1720
set qos acl ip SERVER dscp 26 udp any eq 2427 any
set qos acl ip SERVER dscp 26 tcp any eq 2428 any
commit qos acl SERVER
set qos acl map SERVER 3/23


And in the DVD and Audio it says the best thing to do is copy the QOS
commands from the SRND. However the above commands are not in the SRND.
The SRND say do the following 


set port qos 3/23 trust trust-dscp 

or on a 2Q2T port

set qos acl ip TRUST-DSCP trust-dscp any
commit qos acl TRUST-DSCP
set qos acl map TRUST-DSCP 3/23

My question is do the commands from the SRND meet the requirements of
the task? And if not is there an easier way than the first solution,
since this ACL is not in the SRND? Also given that the first solution is
an older solution is this what I should expect on the lab?

Basically what is the easiest way to find and copy the QOS statements
for the 6500? I do not have access to one outside of the lab so I would
appreciate any tricks or tips rather than memorization.

Thanks

Ed

Re: [OSL | CCIE_Voice] Gatekeeper question

[Jane Ryer (jryer)]

Sorry, correcting my typos - the middle column should of course be route
group (RG), not route list (RL).

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jane Ryer
(jryer)
Sent: Monday, March 24, 2008 9:11 AM
To: Daniel Dellinger; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Gatekeeper question

 

Hi, Daniel,

 

I have also been working my way through Exam II recently, and came to
the same conclusion that you did - that the solution is wrong.  Here's
what I ended up with, which seems to work:

 

RL-GK-ADD-PREFIX  RG_GK  predot + add prefix 1#

RL-GK-NO-PREFIXRG_GK  predot

 

9.01134!PT-HQ-INT RL-GK-NO-PREFIX

9.01134!#  PT-HQ-INT RL-GK-NO-PREFIX

9.01134!PT-BR1-INT   RL-GK-NO-PREFIX

9.01134!#  PT-BR1-INT   RL-GK-NO-PREFIX

 

9.011!   PT-HQ-INT RL-GK-ADD-PREFIX

9.011!#  PT-HQ-INT RL-GK-ADD-PREFIX

9.011!   PT-BR1-INT   RL-GK-ADD-PREFIX

9.011!#  PT-BR1-INT   RL-GK-ADD-PREFIX

 

The things I changed from the provided solution were:

a)   I reversed which dial patterns prefixed the 1#  (as you noted)

b)   I corrected the partition error (PT-HQ-INT was listed for both
9.01134 rows, and PT-BR1-INT was listed for both 9.011 rows)

c)   I moved the digit manipulation to the route list instead of the
dial pattern

 

Mark or Vik, can you confirm that Daniel and I are taking the correct
approach?

 

Thanks,

Jane

 

Jane Ryer, R&S CCIE # 

Network Consulting Engineer

Cisco Systems Advanced Services

 

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel
Dellinger
Sent: Sunday, March 23, 2008 7:48 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Gatekeeper question

 

I'm working through Exam II from recent Ipexpert bootcamp (which was
excellent by the way).  I'm not sure if there is an error the solutions
guide for  a question, or I'm just not understanding.

 

On question 19 (exam II) it states that " International calls from both
Call Manager sites to Germany (country code 34) should be sent to the
gatekeeper and out to the remote PSTN-WAN gatekeeper.  The remote
gatekeeper is not expecting any tech-prefix- just the international
prefix plus the remainder of the digits.  All other International calls
(e.g. 0119876543 from Callmanager should be sent out of the BR2 gateway
via the local gatekeeper (hq)".

 

The solutions guide has the below route patterns:

 

9.01134! (Predot + prefix 1#)   

9.01134!# (Predot + prefix 1#)

9.011! (Predot)

9.011!# (Predot)

 

Below is Gatekeeper config in solutions guide:

 

gatekeeper

 zone prefix HQ-RTR1 3... gw-priority 10 CME

 zone prefix HQ-RTR1 011* gw-priority 10 CME

 zone prefix PSTN-WAN 01134*

 

Now the points I don't understand:

 

1. Why we would drop the predot and then prepend a 1# for calls
that are destined for the PSTN-Gatekeeper (9.01134! and 9.01134!#) if
the PSTN-Gate is not expecting tech-prefix (and a tech-prefix is not
required to send LRQ).  I believe the tech-prefix will be striped prior
to LRQ so doesn't matter, but want to understand why we would send a
tech prefix if it isn't needed for analysis.

2.And since question states that all other international calls
should be sent out BR2 (so route patterns 9.011! and 9.011!#) why are we
not dropping the pre-dot and then prefixing a 1# since tech-prefix is
required (as default technology prefix was not permitted).

 

Seems to me that the situation should be reversed from a tech-prefix
prepend perspective.   Is the solutions guide incorrect for this
question or am I missing something?

 

Thanks

Daniel

 

 

 

 

 

Re: [OSL | CCIE_Voice] Whats Missing

[Jane Ryer (jryer)]

Hi, Matt,

 

Any update on when the new workbooks (and other materials) might be
available for purchase?  The ipexpert.com web site has not changed since
the last week of February.  Many of us were expecting that since the
special on purchasing the old material ended on Feb. 29th that the new
materials might be available soon after that.

 

My next CCIE Voice lab attempt is scheduled for May 8th.   Will the
workbooks be available in time for me to use them in my preparation?  

 

Thanks,

Jane

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Brooks
at IPexpert
Sent: Friday, March 21, 2008 10:10 AM
To: Jonathan Charles
Cc: Mark Snow; CCIE Maillist; Edward French; Juan Lopez Hernandez -X
(jlopezhe - IBM - INS at Cisco)
Subject: Re: [OSL | CCIE_Voice] Whats Missing

 

Jon, 

I will have one of our Training Advisors contact you to answer your
questions. 

Thanks!

-- 
Matt Brooks

Vice President - IPexpert, Inc.

Telephone: +1.810.326.1444 x101
Cell: +1.810.434.7447
Fax: +1.810.454.0130
Mailto: [EMAIL PROTECTED]
--
Join our free online support and peer group communities:
http://www.IPexpert.com/communities
--
IPexpert - The Global Leader in Self-Study, Classroom-Based,
Video-On-Demand and Audio Certification Training Tools for the Cisco
CCIE R&S Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice
Lab and CCIE Storage Lab Certifications.
--



On Thu, Mar 20, 2008 at 9:42 PM, Jonathan Charles <[EMAIL PROTECTED]>
wrote:

Dumb question, but I just bought the CCIE Voice kit (DVD, Audio,
Proctor, Workbook thingy) how do we get the upgrade, when it is
released? Or is it an upgrade?



Jonathan

On Thu, Mar 20, 2008 at 6:59 AM, Mark Snow <[EMAIL PROTECTED]> wrote:
>
> Using the DocCD is a perfectly legitimate technique for all but maybe
the
> IPMA - I would know that one cold since it so crucially affects so
many
> point areas - your dial plan, your CoR, etc. You need to be able to
complete
> the lab in 6 hours flat while studying outside of the real lab-
because you
> need to factor in at least 1hour used up in the lab due to
nervousness,time
> pressure,fact u are paying for it out of pocket and may not be able to
> schedule again for months,etc. That leaves you with 1hour left in the
real
> lab for troubleshooting and anything small that you may not have ever
seen
> before (they do change the lab from time to time).
>
> Aside from that - we are releasing in the next weeks our newest labs
> complete with video walkthroughs that I think Wayne mentioned about a
month
> back on this mailer. I would get that WB (it is by far the most
complex and
> also detailed walkthrough to date that I have ever seen or helped
create -
> with some labs having over 20 hours of video for solutions
walkthroughs).
>
> HTH some,
>
> Mark Snow
> Sr Technical Instructor
> IPexpert, Inc.
>
> Sent from my iPhone
>
>
> On Mar 20, 2008, at 7:13 AM, Edward French <[EMAIL PROTECTED]>
wrote:
>
>
>
> Exactly. The Lab fee is out of my pocket so I am just looking to be as
> prepared as possible.
>
> - Original Message 
> From: Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco)
> <[EMAIL PROTECTED]>
> To: ccievoice1 <[EMAIL PROTECTED]>
> Cc: CCIE Voice Online Study List 
> Sent: Thursday, March 20, 2008 6:41:02 AM
> Subject: Re: [OSL | CCIE_Voice] Whats Missing
>
>
> of course, but reading some feedback on people who've tried already
might be
> helpful in my opinion - it's a personal matter of course - each and
everyone
> of us does these exercises on it's own terms - but it's to get a
general
> idea before we even try the lab :)
>
>  
>  From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
ccievoice1
> Sent: Thursday, March 20, 2008 11:36 AM
> To: Edward French
> Cc: CCIE Maillist
> Subject: Re: [OSL | CCIE_Voice] Whats Missing
>
>
> Give the lab a try!! And you will know what is missing.
> Well, i personally think it might be better to complete within 6 - 7
hrs so
> you can have time for troubleshooting.
>
>
> On Thu, Mar 20, 2008 at 6:32 PM, Edward French
<[EMAIL PROTECTED]>
> wrote:
>
> >
> >
> >
> > I have seen several comments about people who have attempted the
test in
> the past couple of weeks. How well did the Proctor labs ultimate lab
guide
> prepare you for the test? What areas did you find the most difficult?
I can
> quickly and without reference perform all tasks in the books with the
> exception of: IPMA, Fast/Quick Dial, EM, Fax, BACD, QOS on 6500, QOS
on
> FR,and sometimes Gatekeeper gets me. I can quickly find the IPMA,
Fast/Quick
> Dial, EM, Fax and BACD on the univercd or other available source. and
I can
> usually complete the full lab scenarios in th 7:45 proctor lab
session.
> Based on your experience with the lab and my above statements do you
think I
> am ready to take the lab? Additionally I have been working in voice
for 21
> years, I have been a CCNA for I think 10 years and I have been
Microsoft
>

Re: [OSL | CCIE_Voice] Gatekeeper question

[Jane Ryer (jryer)]

Hi, Daniel,

 

I have also been working my way through Exam II recently, and came to
the same conclusion that you did - that the solution is wrong.  Here's
what I ended up with, which seems to work:

 

RL-GK-ADD-PREFIX  RL_GK  predot + add prefix 1#

RL-GK-NO-PREFIXRL_GK  predot

 

9.01134!PT-HQ-INT RL-GK-NO-PREFIX

9.01134!#  PT-HQ-INT RL-GK-NO-PREFIX

9.01134!PT-BR1-INT   RL-GK-NO-PREFIX

9.01134!#  PT-BR1-INT   RL-GK-NO-PRERIX

 

9.011!   PT-HQ-INT RL-GK-ADD-PREFIX

9.011!#  PT-HQ-INT RL-GK-ADD-PREFIX

9.011!   PT-BR1-INT   RL-GK-ADD-PREFIX

9.011!#  PT-BR1-INT   RL-GK-ADD-PREFIX

 

The things I changed from the provided solution were:

a)   I reversed which dial patterns prefixed the 1#  (as you noted)

b)   I corrected the partition error (PT-HQ-INT was listed for both
9.01134 rows, and PT-BR1-INT was listed for both 9.011 rows)

c)   I moved the digit manipulation to the route list instead of the
dial pattern

 

Mark or Vik, can you confirm that Daniel and I are taking the correct
approach?

 

Thanks,

Jane

 

Jane Ryer, R&S CCIE # 

Network Consulting Engineer

Cisco Systems Advanced Services

 

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel
Dellinger
Sent: Sunday, March 23, 2008 7:48 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Gatekeeper question

 

I'm working through Exam II from recent Ipexpert bootcamp (which was
excellent by the way).  I'm not sure if there is an error the solutions
guide for  a question, or I'm just not understanding.

 

On question 19 (exam II) it states that " International calls from both
Call Manager sites to Germany (country code 34) should be sent to the
gatekeeper and out to the remote PSTN-WAN gatekeeper.  The remote
gatekeeper is not expecting any tech-prefix- just the international
prefix plus the remainder of the digits.  All other International calls
(e.g. 0119876543 from Callmanager should be sent out of the BR2 gateway
via the local gatekeeper (hq)".

 

The solutions guide has the below route patterns:

 

9.01134! (Predot + prefix 1#)   

9.01134!# (Predot + prefix 1#)

9.011! (Predot)

9.011!# (Predot)

 

Below is Gatekeeper config in solutions guide:

 

gatekeeper

 zone prefix HQ-RTR1 3... gw-priority 10 CME

 zone prefix HQ-RTR1 011* gw-priority 10 CME

 zone prefix PSTN-WAN 01134*

 

Now the points I don't understand:

 

1. Why we would drop the predot and then prepend a 1# for calls
that are destined for the PSTN-Gatekeeper (9.01134! and 9.01134!#) if
the PSTN-Gate is not expecting tech-prefix (and a tech-prefix is not
required to send LRQ).  I believe the tech-prefix will be striped prior
to LRQ so doesn't matter, but want to understand why we would send a
tech prefix if it isn't needed for analysis.

2.And since question states that all other international calls
should be sent out BR2 (so route patterns 9.011! and 9.011!#) why are we
not dropping the pre-dot and then prefixing a 1# since tech-prefix is
required (as default technology prefix was not permitted).

 

Seems to me that the situation should be reversed from a tech-prefix
prepend perspective.   Is the solutions guide incorrect for this
question or am I missing something?

 

Thanks

Daniel

 

 

 

 

 

Re: [OSL | CCIE_Voice] More DocCD changes!!!

[Jane Ryer (jryer)]

Hi, Scott,

 

I took the lab in RTP last Tuesday (March 11th), and there was a PDF of
the Cisco CAD Installation Guide (CAD 6.1 for IP Contact Center Express
Edition Release 4.0) on the desktop, along with the QoS SRND and the
Call Manager SRND.  They have presumably added this since the links are
broken if you try to navigate to it from DocCD.

 

But, I agree that part of lab preparation should be knowing how to get
to navigate to various topics from the main level of
http://www.cisco.com/univercd .There is no extra time during a lab
attempt to be figuring it out.

 

Jane Ryer

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott
Monasmith
Sent: Tuesday, March 18, 2008 9:04 AM
To: CCIE Maillist
Subject: [OSL | CCIE_Voice] More DocCD changes!!!

 

Those of you that plan on utilizing the DocCD during the lab exam as
part of your test taking strategy, I can not stress enough that you need
to continually monitor the status of the links that you plan on using
for the lab. The ENTIRE Contact Center portion of the DocCD has now been
moved off of www.cisco.com/univerCD. Granted, redirects from DocCD to a
particular page should work during the exam, but I don't believe this
particular move regarding Contact Center will work since you are not
being redirected to a particular .PDF or web page but rather the
Cisco.com support home page.

 

Cheers,

Scott

Re: [OSL | CCIE_Voice] IPCC Phone Agent Service URL - movedfromUnivercd ??

[Jane Ryer (jryer)]

I sat the lab for the first time in RTP this past Tuesday, March 11th.
(I did not pass - not an unexpected outcome, but still disappointing.)

In addition to the CCM and QoS SRND's on the desktop, there is now also
a .PDF version of the Cisco CAD Installation Guide, CAD 6.1 for IP
Contact Center Express Edition Release 4.0.  I am pretty sure that it
was the file found at this url:
http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_cente
r/crs/express_4_0/installation/for_cad/cad611ig.pdf

It does include the url for the IP Phone Agent service.

The redirect links worked fine for everything else I tried.

Jane Ryer
R&S CCIE # 
Network Consulting Engineer
Cisco Systems Advanced Services team


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Snow
Sent: Tuesday, February 26, 2008 7:10 PM
To: Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco)
Cc: ccie_voice@onlinestudylist.com; Mike Prestidge
Subject: Re: [OSL | CCIE_Voice] IPCC Phone Agent Service URL -
movedfromUnivercd ??

CME is still here:
http://www.cisco.com/univercd/cc/td/doc/product/voice/its/index.htm
Just have to poke around to find it.

But the UCCX - That is disturbing.
I am looking into it.


Mark Snow
CCIE #14073 (Voice, Security)
CCSI #31583
Senior Technical Instructor - IPexpert, Inc.
A Cisco Learning Partner - We Accept Learning Credits!
Telephone: +1.810.326.1444
Fax: +1.309.413.4097
Mailto: [EMAIL PROTECTED]

IPexpert - The Global Leader in Self-Study, Classroom-Based, Video On  
Demand and Audio Certification Training Tools for the Cisco CCIE R&S  
Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and  
CCIE Storage Lab Certifications.


On Feb 26, 2008, at 5:29 PM, Juan Lopez Hernandez -X (jlopezhe - IBM -  
INS at Cisco) wrote:

> The same goes for all CME related docs :-( , since last night...
>
> Juan
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Mike
> Prestidge
> Sent: Tuesday, February 26, 2008 11:23 PM
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] IPCC Phone Agent Service URL - moved
> fromUnivercd ??
>
> It seems that the (annoying) people moving the documents off Univercd
> have now also moved the documents with the URL for IP Phone agents!!
>
> I used to be able to find this by browsing via the following:
>
> Univercd > Customer Contact Software > IPCC Express and IP IVR > CRS
> 5.0(x) > English > Documentation for Cisco IP Agents > Cisco CAD
> Installation Guide 6.4
>
> Now this documentation has also been moved to a link that is not
> available in the lab.  Does anyone know an alternative location to  
> find
> the URL within Univercd?
>
> Mike
>
> This communication, including any attachments, is confidential. If you
> are not the intended recipient, you should not read it - please  
> contact
> me immediately, destroy it, and do not copy or use any part of this
> communication or disclose anything about it. Thank you. Please note  
> that
> this communication does not designate an information system for the
> purposes of the Electronic Transactions Act 2002.
>

[OSL | CCIE_Voice] CUE MWI message picked up by IPIPGW incoming sip dial peer

[Jane Ryer (jryer)]

This dial peer existed on my BR2 router, left over from workbook task
4.9 to setup an IPIPGW:

 

dial-peer voice 10 voip

incoming called-number 3...

session protocol sipv2

dtmf-relay rtp-nte

 

I setup CUE on BR2, which resulted in this ephone-dn:

 

ephone-dn 36

number 3999...

mwi on

 

If I directly dialed 39993003, then the mwi did come on for BR2 phone 3.

 

However, if I left a voicemail message for that phone, the MWI did not
come on.  A debug showed me that it was matching the IPIPGW dial peer
instead, which in retrospect makes sense because CUE is using SIP to
communicate with the CME router.  

 

I tried specifying the session target on dial-peer 10 to be the IP
address for HQ-RTR, thinking that that might limit use of that dial-peer
to only that session, but that did not work.  Since I was focusing on
CUE at the time, I just deleted dial-peer 10 so that my mwi lights would
light, and decided to come back and think about it some more today.

 

What would have been my best option so that the IPIPGW would have still
worked, and the mwi messages would have matched the ephone-dn instead of
dial-peer 10?  Would it have worked to specify interdigit timeout at the
end of the incoming called-number?  (i.e. incoming called-number 3...T
?)

 

Thanks,

Jane

 

 

[OSL | CCIE_Voice] issues with two remote phones logging in to IP Phone Agent

[Jane Ryer (jryer)]

I was working on IPCC Express scripting yesterday, and had problems with
getting two phones logged in to IP Phone Agent so that I could test out
my script.

 

I was connected to the Proctor labs rack through EasyVPN, with 7961
phones at home.  I created one-button-login service for phone 3 at both
my HQ and BR1 sites, specifying the agent1 user for the HQ phone and
agent2 user for the BR1 phone.  If I login just one of the two phones,
everything is fine.  But if I log them both in, they migrate to being
the same user (either agent1 or agent2, I think it depends on the
sequence I log them in).  At least once, I got some sort of error while
logging in about a duplicate IP address, which is when I realized that
Call Manager does see both these phones as the same IP address. 

 

I am pretty sure that I will have the same issue with IP Blue software
phones while I am connected via EasyVPN.  Would this work if I instead
use the VPN client to connect to the Proctor labs rack, and then bring
up two IP Blue software phones?  It's been a while since I've done that,
and I just don't remember now how the IP addresses showed up.  This is
the first exercise where I've run into issues with the NAT/PAT way of
providing connectivity between the phones and the rack.

 

Thanks,

Jane

[OSL | CCIE_Voice] bringing up ISDN at BR1

[Jane Ryer (jryer)]

I have had repeated issues (on different pods) getting the PRI in the
BR1 routers to reach an ISDN layer 2 status of "multiple_frame_
established".  My usual tactic to get the link up is to reload the BR1
router, but yesterday even that didn't work on Pod32.  I was finally
successful by hitting the power button (on the Proctor labs web page)
for my PSTN router.  (My guess is that probably just does a shut/no shut
on the controller on the PSTN router; I doubt that they would actually
power down a router which is shared between pods.)  When I "powered back
on" my PSTN router, the link came right up.

 

My usual process is to issue all the necessary IOS commands on the BR1
router (per task 4.2 in the Ipexpert workbook), then sometime later (in
some cases it might be an hour or two later) to get to the Call Manager
configuration.  When I finish the Call Manager configuration, I do a
"reset" on the gateway on Call Manager, then go to the router and issue
"no mgcp" followed by "mgcp".  The PRI almost never comes up at that
point, so then I try combinations of shutting down mgcp on the router,
completely restarting Call Manager on both Pub and Sub, doing a "no
shut" on mgcp.  Sometimes I am successful with some of those attempts,
but more typically I then reload the BR1 router and after the router
reloads, the link comes right up.

 

I am concerned whether I am going to run into this in the actual lab,
without any access to the "PSTN router" side of things.  I guess if that
happened, I could run some isdn debugs and ask the proctor to reset the
other side, but I really need to understand better what's happening
before I do that.  It's been a VERY long time since I studied ISDN
q921/q931 protocols.  I did do a few isdn debugs yesterday, but didn't
know what I was looking at, and managed to lock up the router by issuing
"debug isdn all" or something like that.  

 

Is there a better process/procedure for setting this up?  Should I not
do the "isdn bind-l3 ccm-manager" command on the serial 0/0/0:23
interface until Call Manager is set up?  But won't the "service mgcp" at
the end of the pri-group command on the controller keep it from coming
up anyway until Call Manager is responding?  By the way, a "sho
ccm-manager" command was showing that MGCP was registered to my
subscriber, with a backup of my publisher. 

 

Has anyone else had similar issues?  What am I missing here?  Do I need
to dig more deeply into q921/q931?  (There are so many other areas that
I need to be studying  . . .sigh!!)

 

Thanks,

Jane

[OSL | CCIE_Voice] problem with PSTN-WAN router in pod32

[Jane Ryer (jryer)]

I am working with rack 32 today, and having a problem getting
international dialing to work through my HQ-RTR gatekeeper.  I used the
available debugging commands on my PSTN router (10.32.200.2), and found
that when I place a call to the international number (starting with
011), it is matching dial peer 10 on the PSTN router.  Here is the
output of "sho dial-peer summary" (option 3 on the menu):

CEnter your option and press Return:3
dial-peer hunt 0
 ADPRE PASS
OUT 
TAGTYPE  MIN  OPER PREFIXDEST-PATTERN  FER THRU SESS-TARGET
STAT PORT
1  pots  up   up0
down 
2  voip  up   up 8911   0  syst
ipv4:192.20.200.30  
7  pots  up   up 211... 0
up   0/3/0:23
8  pots  up   up 6175222... 0
up   0/3/1:23
9  pots  up   up 3313223... 0
up   0/2/0:0
10 pots  up   up   3313253   0113313223...  0
up   0/2/0:0
3  voip  up   up 8999   0  syst
ipv4:192.20.200.30  
4  voip  up   up 617522 0  syst
ipv4:192.20.200.30  
5  voip  up   up 21 0  syst
ipv4:192.20.200.30  
6  voip  up   up 331322 0  syst
ipv4:192.20.200.30  
110voip  up   up 0119876543 0  syst
ipv4:192.20.200.30  
911voip  up   up 9110  syst
ipv4:192.20.200.30  
9876   voip  up   up0  syst


I think that prefix on dial-peer 10 is incorrect - shouldn't it be
3313223 instead?  From my PSTN phone, I can successfully call my BR2
phone using 3313223003 but 3313253003 does not work.

Vic or Mark, would it be possible to get this corrected?  I have pod 32
booked for a double session today (until midnight EST), and I'd like to
have international calling working.

Thanks,
Jane Ryer

Re: [OSL | CCIE_Voice] VT Advantage

[Jane Ryer (jryer)]

In the interest of fully understanding what documentation is available
(both while in the lab exam and more generally) . . . 

 

Vic, exactly which document are you referring to as "the Video Telephony
SRND"?

 

If I go to http://www.cisco.com/go/srnd and click on the link for "View
All Documents", the only document I find with "video" in the title is:

Cisco Unified Videoconferencing Solution Reference Network Design

 

I pulled up the full .pdf version of that document and searched through
it and could not find the table that Ed is referring to.

 

I also pulled up the full "Cisco Unified Communications SRND based on
Cisco Unified CallManager v4.x" and downloaded the chapter titled "IP
Video Telephony" and still do not find the chart.

 

I am wondering if the chart that you are referring to is in an earlier
version (no longer available on CCO) of one of the documents.


Thanks,
Jane

 

Jane Ryer, R&S CCIE # 

Network Consulting Engineer

Cisco Systems Advanced Services team

 

 

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vik Malhi
Sent: Monday, February 04, 2008 2:57 PM
To: 'Edward French'; 'CCIE Maillist'
Subject: Re: [OSL | CCIE_Voice] VT Advantage

 

Page 33 of the Video Telephony SRND.

 

Vik Malhi
CCIE Voice Instructor / Developer - IPexpert, Inc.
CCIE Voice #13890 CCSI #31584
URL: http://www.IPexpert.com  
Toll Free: +1.866.225.8064
International: +1.810.326.1444

 

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Edward
French
Sent: Friday, February 01, 2008 6:23 PM
To: CCIE Maillist
Subject: [OSL | CCIE_Voice] VT Advantage

In the Proctor Guide for  Section 2 on page 37 there is a chart with the
supported audio and video codecs and associated bandwiths for the VT
advantage camera. I have searched CCO for over an hour trying to find
the document that this chart is in. Does anyone know where to find this?

Ed