Re: [OSL | CCIE_Voice] Calls to CUE fail over WAN

2011-12-27 Thread bkvalent...@gmail.com
Did you enable cube? 

Sent from my Verizon Wireless Phone

- Reply message -
From: John McGaughey (jomcgaug) jomcg...@cisco.com
Date: Tue, Dec 27, 2011 7:27 pm
Subject: [OSL | CCIE_Voice] Calls to CUE fail over WAN
To: ccie_voice@onlinestudylist.com

Hello,

 

I'm in Workbook 2, Lab 8, question 4.5.  I have CUE setup properly and
calls between CME phones roll successfully to voicemail.  However, they
fail to roll to voice mail when an HQ phone calls a BR2 phone.

 

The answer guide says to configure an xcoder on the BR2 router.  Here's
my config.

 

sccp local Vlan400

sccp ccm 10.10.202.1 identifier 1 version 7.0

sccp

!

sccp ccm group 1

associate ccm 1 priority 1

associate profile 1 register br2-xcoder

!

dspfarm profile 1 transcode

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec g729r8

maximum sessions 4

associate application SCCP

telephony-service

sdspfarm units 1

sdspfarm transcode sessions 4

sdspfarm tag 1 br2-xcoder

authentication credential admin cisco

xml user pvadmin password cisco 15

max-ephones 10

max-dn 10 no-reg both

ip source-address 10.10.202.1 port 2000

url services http://10.10.202.2/voiceview/common/login.do

url authentication http://10.10.202.1/CCMCIP/authenticate.asp pvphone
cisco

voicemail 3600

max-conferences 12 gain -6

web admin system name admin password cisco

transfer-system full-consult

create cnf-files version-stamp 7960 Dec 28 2011 00:02:05

 

Here's the dial-peer to CUE.

 

dial-peer voice 3600 voip

destination-pattern 3600

session protocol sipv2

session target ipv4:10.10.202.2

incoming called-number 3600

dtmf-relay sip-notify

codec g711ulaw

no vad

 

A sh sccp shows the xcoder registered.

 

BR2-RTR#sh sccp

SCCP Admin State: UP

Gateway Local Interface: Vlan400

IPv4 Address: 10.10.202.1

Port Number: 2000

IP Precedence: 5

User Masked Codec list: None

Call Manager: 10.10.202.1, Port Number: 2000

Priority: N/A, Version: 7.0, Identifier: 1

Trustpoint: N/A

 

Transcoding Oper State: ACTIVE - Cause Code: NONE

Active Call Manager: 10.10.202.1, Port Number: 2000

TCP Link Status: CONNECTED, Profile Identifier: 1

Reported Max Streams: 8, Reported Max OOS Streams: 0

Supported Codec: g711ulaw, Maximum Packetization Period: 30

Supported Codec: g711alaw, Maximum Packetization Period: 30

Supported Codec: g729ar8, Maximum Packetization Period: 60

Supported Codec: g729abr8, Maximum Packetization Period: 60

Supported Codec: g729r8, Maximum Packetization Period: 60

Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30

Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30

Supported Codec: inband-dtmf to rfc2833 conversion, Maximum
Packetization Period: 30

 

A trace on the router shows this.

 

Dec 28 00:00:52.604: //-1//SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:3600@10.10.202.1:5060 SIP/2.0

Via: SIP/2.0/UDP 10.10.200.3:5060;branch=z9hG4bK2AB55

Remote-Party-ID: HQ Phone 2
sip:5002@10.10.200.3;party=calling;screen=yes;privacy=off

From: HQ Phone 2 sip:5002@10.10.200.3;tag=4F3F96E8-171B

To: sip:3600@10.10.202.1

Date: Wed, 28 Dec 2011 00:00:52 GMT

Call-ID: D7867EA6-301D11E1-80CE8486-37477DE6@10.10.200.3

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 3615875710-807211489-2160690310-927432166

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Max-Forwards: 70

Timestamp: 1325030452

Contact: sip:5002@10.10.200.3:5060

Diversion:
sip:3002@10.10.200.3;privacy=off;reason=no-answer;counter=1;screen=no

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 267

 

v=0

o=CiscoSystemsSIP-GW-UserAgent 1283 3954 IN IP4 10.10.200.3

s=SIP Call

c=IN IP4 10.10.200.3

t=0 0

m=audio 17098 RTP/AVP 18 101

c=IN IP4 10.10.200.3

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

 

Dec 28 00:00:52.612: //-1//SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 488 Not Acceptable Media

Via: SIP/2.0/UDP 10.10.200.3:5060;branch=z9hG4bK2AB55

From: HQ Phone 2 sip:5002@10.10.200.3;tag=4F3F96E8-171B

To: sip:3600@10.10.202.1;tag=3003BC-1D3E

Date: Wed, 28 Dec 2011 00:00:52 GMT

Call-ID: D7867EA6-301D11E1-80CE8486-37477DE6@10.10.200.3

Timestamp: 1325030452

CSeq: 101 INVITE

Allow-Events: telephone-event

Warning: 304 10.10.202.1 Media Type(s) Unavailable

Reason: Q.850;cause=65

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

 

 

Dec 28 00:00:52.628: //-1//SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:3600@10.10.202.1:5060 SIP/2.0

Via: SIP/2.0/UDP 10.10.200.3:5060;branch=z9hG4bK2AB55

From: HQ Phone 2 sip:5002@10.10.200.3;tag=4F3F96E8-171B

To: sip:3600@10.10.202.1;tag=3003BC-1D3E

Date: Wed, 28 Dec 2011 

Re: [OSL | CCIE_Voice] ANI Translation

2011-12-24 Thread bkvalent...@gmail.com
You are correct.  The route list translations will be used whereever the route 
list is used.

You can assign translations to ANI using a Translation profile.  Translation 
profiles match on called number, but allow manipulations to called or calling 
numbers.

So your user's css points to the TP.  It does manipluations in ANI and then 
uses its own css to send the call out the appropriate route pattern. 

Also, keep in mind if you send the calls out an h323 or SIP gateway, you can do 
the ANI translations in the outbound dialpeers.  So, no cucm based 
manipulations might benecessary.

Also, keep in mind you can use calling party transformations... Perhaps on the 
outbound gateway.

Brian


- Reply message -
From: datucha123 datucha123 datucha...@gmail.com
Date: Sat, Dec 24, 2011 11:42 am
Subject: [OSL | CCIE_Voice] ANI Translation
To: ccie_voice@onlinestudylist.com

Hello,

I am a little confused about the ANI translation.

Imagine that we have several Route Patterns:

1. Local Calls Pattern
2. Long Distance Calls Pattern
3. International Calls Pattern

Actual Pattern Strings does not matter.

All Patterns point to the same Route List (there is only one Route List
defined)

Now if we want to provide different ANI based on the Called Number, we have
to use the Route Pattern Translation options so that when the call
goes through Local Calls Pattern, we can set ANI as X.
But when the call goes through the Long Distance Calls Pattern we can set
the ANI to Y
And so on, for International Calls Pattern as well.

As and I guess it is NOT POSSIBLE  to use the Route List ANI Translation
Options for ANI translation based on the Called Number, as all Patterns
point to the same Route List, and Route List cannot differentiate which
Route Pattern has been called initially. So it will Translate ANI for all
calls, without differentiating which Pattern has been called.

 Am I right? Or I am missing something, and do not fully understand the
Route Lists?
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Re: [OSL | CCIE_Voice] RFC2833 with MTP

2011-12-17 Thread bkvalent...@gmail.com
I wonder if you sent it back through the Lo0 interface if it would work.


- Reply message -
From: Mohd Baqari baqari.voic...@gmail.com
Date: Sat, Dec 17, 2011 6:16 pm
Subject: [OSL | CCIE_Voice] RFC2833 with MTP
To: datucha123 datucha123 datucha...@gmail.com
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com


Ya .. Tested this and same result

Regards,
Mohammed Al Baqari

Sent from my iPhone

On Dec 16, 2011, at 4:44 PM, datucha123 datucha123 datucha...@gmail.com wrote:

 You are also  experiencing  the same for MTP on CUBE?
  
 
  
 On Fri, Dec 16, 2011 at 3:37 PM, Mohd Baqari baqari.voic...@gmail.com wrote:
 I look into it as well in my IOS to confirm the same ... Thx for sharing the 
 info
 
 
 Regards,
 Mohammed Al Baqari
 
 Sent from my iPhone
 
 On Dec 16, 2011, at 2:47 PM, datucha123 datucha123 datucha...@gmail.com 
 wrote:
 
 I have found out the following.
  
 The Router IOS MTP (Not DSP, well DSP also does support it) supports 
 RFC2833, I have tested it and the RFC2833 from a SIP Trunk are translated 
 succesfuly for UCCX IVR.
  
 But when the IOS or DSP MTPs are located on the same Router as the CUBE, 
 then the MTP does not make RFC2833.
  
 As soon as I have moved MTP to other Router, Branch1 Router, the RFC2833 was 
 succesfuly, the MTP did the Translation between RFC2833 and Out of Band 
 DTMF., 
  
 I think this is a BUG, because the same problem is observer of CCM  modules 
 when the MTP located on it and the solution is to move the MTP to ISR 
 Routers.
  
 So bear in mind, that the RFC2833 may not work for MTP, when that MTP 
 resource is located on the CUBE Router.
  
 
 
  
 On Thu, Dec 15, 2011 at 11:38 PM, datucha123 datucha123 
 datucha...@gmail.com wrote:
 I see now. Great info.
  
 Also in that case, only DSP MTP and CUCM Software MTP can translate RFC2833 
 and that would be only for G711 calls, as CUCM and DSP MTPs support only 
 G711, am I right? And if we want to use other codec for MTP and we have to 
 configure the IOS MTP (max sessions software) which does not support RFC2833 
 in this case.
  
 Also the Transrating (different packetization period codec) is not supported 
 on max-session software MTP, and we need to use either CUCM Software or DSP 
 MTP, that support only G711.
  
  
  
 On Thu, Dec 15, 2011 at 10:43 PM, Mohammed Al Baqari 
 baqari.voic...@gmail.com wrote:
 Ok since you are using MTP on gateway you need to make it as hardware MTP to 
 do inband to OOB DTMF translation.
 
  
 
 Regards,
 
 Mohammed Al Baqari
 
  
 
 From: datucha123 datucha123 [mailto:datucha...@gmail.com] 
 Sent: Thursday, December 15, 2011 10:09 PM
 To: Mohd Baqari
 Cc: ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] RFC2833 with MTP
 
  
 
 MTP is invoked, and the show sip-ua calls shows that the RFC2833 is used for 
 both calls legs. Even the debug voip rtp session named-events shows that the 
 DTMF tones are traversing. But the UCCX does not undertand them correclty.
 
 On Thu, Dec 15, 2011 at 9:03 PM, Mohd Baqari baqari.voic...@gmail.com 
 wrote:
 
 Make sure that mtp is invoked during the call and verify the dtmf method 
 used on both call legs using show call active voice command.
 
 Regards,
 Mohammed Al Baqari
 
 Sent from my iPhone
 
 
 On Dec 15, 2011, at 6:33 PM, datucha123 datucha123 datucha...@gmail.com 
 wrote:
 
  Hello,
 
  I have a problem with MTP and RFC2833.
 
  When calling from BR2 CCME to UCCX at HQ Site through the SIP CUBE, which 
  has RFC2833 as the DTMF method, the UCCX IVR does not understand the DTMF 
  tones correctly when the Router IOS MTP is used on a SIP Trunk in CUCM.
  But as soon as I assing the CUCM Native Software MTP to use on a CUCM SIP 
  Trunk, the RFC2833 DTMF digits are recognized correctly in UCCX.
 
  What do I mean under incorrect DTMF inderstanding, is that when the Caller 
  at BR2 CUCME presses any digits in UCCX IVR, he is transferred to operator 
  always, and it does not matter what digits he pressed. Until I chenge the 
  Router IOS MTP to CUCM Native Software MTP, and then the BR2 Caller can 
  press the digits and even transfer to required extension from UCCX IVR.
 
  ___
  For more information regarding industry leading CCIE Lab training, please 
  visit www.ipexpert.com
 
  Are you a CCNP or CCIE and looking for a job? Check out 
  www.PlatinumPlacement.com
 
  
 
 
 
 
___
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Re: [OSL | CCIE_Voice] lab - 6

2011-11-22 Thread bkvalent...@gmail.com
Trolls.  Don't feed them and maybe they will go away.

- Reply message -
From: datucha123 datucha123 datucha...@gmail.com
Date: Tue, Nov 22, 2011 1:38 pm
Subject: [OSL | CCIE_Voice] lab - 6
To: Ashraf Ayyash ash.ayy...@gmail.com
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com, Ken 
Wyan kew...@gmail.com


Can anybody explain me, what does the LAB 6 and etc means?

Why people are mentioned LAB 3,4,5-6?



On Tue, Nov 22, 2011 at 8:45 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote:

 Hello Ken , All ,

 Lets clarify all this in one shot to make it clear to everyone ,

 First , This is a clear NDA violation on a public alias , can you
 please be more Professional ?

 Second , i have discussed this with someone else who came couple of
 months ago and said he got lab 5 and i finally i shared in the alias
 that those people is trying to do commercial to DOC they created
 themselves and called it Lab 5 or Lab 6 and they are trying to
 manipulate people under pressure and we agree that this is ridiculous
 and if anyone wanna share something to help he must
 chose his words and be professional ,

 Now in regard of the DHCP i think this Question have been answered
 correctly in case you will get it in the lab ,

 The CCIE Lab is not design to fit to the real implementations , its
 set to test the knowledge in certain areas no matter if its against
 the real life design or not ,

 To answer Ken's Question ,

 The CCM MTP is not capable to do G729 codec and so the HW MTP when you
 configure max session hardware under the dsp profile
 those 2 guys can do G711 only

 So if you have g729 Call and you want do EO call in the CCM version in
 the lab , you have to have SW IOS MTP register with the CCM and have
 Codec G729 under it dsp profile and make sure you have it in the MGR
 available to the SIP Trunk ,

 Thanks
 Ash

 On Tue, Nov 22, 2011 at 7:24 AM, Peter Jeff peterjeff2...@yahoo.com wrot
   H a H a
 
  R eal life experience and ccie is 2 different thing so whoever is
 thinking
  and trying to match that it is stupidity
 
  I know 15 to 18 years real life experts fail lab but i know fresher pass
 it
  so next match it
 
  and whoever match it is stupidity
 
 
  From: Julien Krieger krieger.jul...@gmail.com
  To: Randall Crumm rrcr...@yahoo.com
  Cc: Peter Jeff peterjeff2...@yahoo.com; Mark Reed marklr...@gmail.com
 ;
  ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com; Ken
 Wyan
  kew...@gmail.com
  Sent: Tuesday, November 22, 2011 2:09 PM
  Subject: Re: [OSL | CCIE_Voice] lab - 6
 
  Hi Randall,
 
  I agree with you don't want to go this road. Certainly Voice Lab is about
  doing many things you would never do in real life 
  I guess that is why many of us get frustrated when they fail...  :(
 
  Julien
 
  2011/11/22 Randall Crumm rrcr...@yahoo.com
 
  Thanks  Mark!
 
  The real question is why is a question like this in the CCIE Voice lab in
  the first place.
  Randall
 
  From: Mark Reed marklr...@gmail.com
  To: Peter Jeff peterjeff2...@yahoo.com
  Cc: Ken Wyan kew...@gmail.com; ccie_voice@onlinestudylist.com
  ccie_voice@onlinestudylist.com
  Sent: Tuesday, November 22, 2011 3:22 AM
  Subject: Re: [OSL | CCIE_Voice] lab - 6
 
  The answer is in this document.
 
 
 http://www.cisco.com/en/US/docs/ios/ipaddr/configuration/guide/iad_dhcp_svr_cfg_ps6441_TSD_Products_Configuration_Guide_Chapter.html#wp1117976
 
  I tried it last night in my lab.  Here is my config using the static
  instructions in that doc.
 
  ip dhcp pool SCDHCP
 origin file flash:SCDHCP
 option 150 ip 142.100.64.12 142.100.64.11
 default-router 142.102.66.254
  BR2-RTR# more flash:SCDHCP
  *time* Nov 21 2011 07:52 AM
  *version* 2
  !IP addressTypeHardware address Lease expiration
  142.102.66.21 /24   id   0100.1B54.951C.D9   Infinite
  142.102.66.22 /24   id  0100.1AA1.93E0.F5 Infinite
  *end*
 
  BR2-RTR#sh cdp nei det
  Device ID: SEP001B54951CD9
  Entry address(es):
IP address: 142.102.66.21
  ...
  Device ID: SEP001AA193E0F5
  Entry address(es):
IP address: 142.102.66.22
 
  I created the SCDHCP doc in notepad.  Uploaded it to CUCM, then TFTPed
 it to
  my router.  Once the file is in flash the since dhcp pool looks up the
  static IPs in the file then assigns it to the appropriate phone.
  On Mon, Nov 21, 2011 at 2:14 PM, Peter Jeff peterjeff2...@yahoo.com
  wrote:
  The real question is why is a question like this in the CCIE From: Mark
 Reed
  marklr...@gmail.com
  To: Peter Jeff peterjeff2...@yahoo.com
  Cc: Ken Wyan kew...@gmail.com; ccie_voice@onlinestudylist.com
  ccie_voice@onlinestudylist.com
  Sent: Tuesday, November 22, 2011 3:22 AM
  Subject: Re: [OSL | CCIE_Voice] lab - 6
 
  The answer is in this document.
 
 
 http://www.cisco.com/en/US/docs/ios/ipaddr/configuration/guide/iad_dhcp_svr_cfg_ps6441_TSD_Products_Configuration_Guide_Chapter.html#wp1117976
 
  I tried it last night in my lab.  Here is my config using the static
  instructions 

Re: [OSL | CCIE_Voice] SRST keepalive time

2011-11-08 Thread bkvalent...@gmail.com
They need to apply QoS, especially on the WAN.  There is no other solution -- 
only several options of bandages.  What you need is to stop the bleeding.  I 
doubt very highly that Cisco would support the configuration without QoS in 
place.

2mbps sounds like it could be an Ethernet handoff?  If so, you'll need to shape 
the link down to 2mbps.  I'd use a nested policy that applies an LLQ policy 
inside a shaping policy.
Need to do the same on both sides.



- Reply message -
From: Emanuel Damasceno aedamasc...@gmail.com
Date: Tue, Nov 8, 2011 8:18 pm
Subject: [OSL | CCIE_Voice] SRST keepalive time
To: ccie_voice@onlinestudylist.com

Thanks for the replies.

It didn't work. The customer has a centralized solution with 2 sites. His
two sites use MGCP, but unfortunately his environment is a screwy solution.
He has EVERYTHING on VLAN 1. I explained to him many times he needed to
segregate his VLANs and use QoS for his WAN. But we have to work with this
CRAPPY solution for the time being...

This is what he needs (I'll try to be as specific as I can).

His screwed up WAN link is a 2MBps (I highly doubt this) without any QoS
applied (neither in his side or his SP). The link is very unstable and his
phones on the remote site keeps coming back and forth from SRST. Since it
keeps doing it for so long, the phones keep just being out of its service,
because it registers, then there is a problem with the link and all his
phones in the remote site tries to go to SRST mode, but then all of the
sudden the link is back and the phones try to go back to CUCM.

He was asking me to put a 12 hour (yes, who doesn't have a customer like
this...) before the phones tried to connect with CUCM again. In other
words, he wanted to let the phones in SRST mode for a few hours until he
fixes his problem with the link. So, I tried the command ccm-manager
switchback uptime-delay minutes, and  we simulate a WAN outage. The
phones entered SRST (well, at least he said it was. I am the remote
support), but when he turned on the CUCM back, the phones registered right
away. Even after setting the time for two hours.

So, we went on Enterprise Parameters and changed the connection duration
monitor but I didn't do it clusterwise. I did it on the device pool on the
remote site. Now, the phones didn't register to SRST so  none of the phones
were making any calls or receiving it (that's why I wonder his SRST
solution never worked in the first place). So I rolled back to the standard
settings and I told him I'd study his solution and give him back an answer.
Honestly, knowing what he has, I think this won't work, but can you guys
give me any ideas?

I appreciate it.

*Antonio Emanuel Damasceno*
CCNA, CCNA Voice, CCNP Voice, CCIE Voice (written)
CompTIA Network+




On Tue, Nov 8, 2011 at 2:44 PM, Emanuel Damasceno aedamasc...@gmail.comwrote:

 Hello experts,

 I currently have a customer who has CUCMBE in his environment with SRST
 enabled. His voice gateway is over a WAN link, and the link is unstable.
 His phones keep registering back and forth, and now he wants to keep his
 phones in SRST mode for a little longer than usual. His configs are
 call-manager-fallback based (no CME as SRST)... How can I achieve this? Is
 it through CUCMBE or his CME? His gateway is MGCP.

 Thanks
 *Antonio Emanuel Damasceno*
 CCNA, CCNA Voice, CCNP Voice, CCIE Voice (written)
 CompTIA Network+



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Re: [OSL | CCIE_Voice] CUCM and UCCX - JTAPI issue

2011-11-05 Thread bkvalent...@gmail.com
See if this helps.

http://www.onlinestudylist.com/archives/ccie_voice/2009-February/057713.html

Brian

- Reply message -
From: Duncan Hamilton-Walker dun...@rosethorn.plus.com
Date: Sat, Nov 5, 2011 7:21 pm
Subject: [OSL | CCIE_Voice] CUCM and UCCX - JTAPI issue
To: 'Bill Lake' whl...@gmail.com
Cc: 'CCIE Study' ccie_voice@onlinestudylist.com


I don't know yet as i have just finished building the server for the next
part of my studies.. but every time i launch the web interface and log in i
get the same.. ?

 

 

 

From: Bill Lake [mailto:whl...@gmail.com] 
Sent: 05 November 2011 23:40
To: Duncan Hamilton-Walker
Cc: CCIE Study
Subject: Re: [OSL | CCIE_Voice] CUCM and UCCX - JTAPI issue

 

I believe this is a bug, does it work correctly if you continue or relaunch?




On Sat, Nov 5, 2011 at 2:47 PM, Duncan Hamilton-Walker
dun...@rosethorn.plus.com wrote:

Dear All,

 

Can anyone please shine some light on my problem message below;

I have resync the JTAPI client a few times with the same result.. 

 

Cisco JTAPI Client

An error has occurred while trying to resynchronize the Cisco JTAPI Client. 

Please try Cisco JTAPI Resync in Cisco Unified CCX Administration again.

 

System versions below;

 

Cisco Unified CM Administration

System version: 7.1.5.33900-10

 

Cisco Application Administration - 7.0(1)SR05_Build504 

Package: Cisco Unified CCX Premium

Cluster: default

 

Thanks

D

 

 


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  _  

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Re: [OSL | CCIE_Voice] [How to make an ISO from DVD's]

2011-11-01 Thread bkvalent...@gmail.com
There must be a million ways to do this.  Google is your friend.  I like Infra 
Recorder, mainly because it is light and open source.

http://infrarecorder.org

- Reply message -
From: michael.se...@compucom.com
Date: Tue, Nov 1, 2011 6:37 pm
Subject: [OSL | CCIE_Voice] [How to make an ISO from DVD's]
To: ccie_voice@onlinestudylist.com

Greetings,
I've got a set of CUCM CUWL Disks and I want to turn them into an .iso image so 
I can burn them to a VM.  Any suggestions or knowledge share would be 
appreciated.  -Thanks.
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Re: [OSL | CCIE_Voice] Home lab PSTN phone woes

2011-09-29 Thread bkvalent...@gmail.com
On the PSTN router, did you do create cnf-files under telephony service?

- Reply message -
From: Voiper datapack...@gmail.com
Date: Thu, Sep 29, 2011 8:41 am
Subject: [OSL | CCIE_Voice] Home lab PSTN phone woes
To: ccie_voice@onlinestudylist.com

Ahoy one and all:

I am a struggling voice aspirant. Got my home lab setup and started IPE
volume I workbook. I have two 3550 switches, one for HQ and the other for
BR1.My pstn phone is connected to the BR1-3550. The pstn configuration is
similar to the IPE pstn config (Vol 1, lab4). I am able to ping all
interfaces from my pstn router(2811). However, the pstn phone (7960) is not
able to get any ip address; show 'registration rejected' message.

Here are the things I undertook to try to resolve the issue:

1) verified the dhcp configuration was correct on the pstn router.
ip dhcp excluded-address 10.10.100.1 10.10.100.49
ip dhcp excluded-address 10.10.100.70 10.10.100.254
!
ip dhcp pool PSTN
   network 10.10.100.0 255.255.255.0
   default-router 10.10.100.2
   option 150 ip 10.10.100.2

2) Created Vlan for pstn on BR1-3550
BR1-3550#svb
VLAN Name StatusPorts
  -
---
1default  activeFa0/4, Fa0/5, Fa0/6, Fa0/7,
Fa0/8, Fa0/9, Fa0/10, Fa0/11, Fa0/12, Fa0/14, Fa0/15
Fa0/16, Fa0/17, Fa0/18,
Fa0/19, Fa0/20, Fa0/21, Fa0/22, Fa0/24, Gi0/1, Gi0/2
10   DATA active
20   PHONES   active
30   SERVERS  active
100  PSTN activeFa0/13
130  BR1-DATA activeFa0/2, Fa0/3
240  BR1-PHONES   activeFa0/2, Fa0/3
1002 fddi-default act/unsup

3) Configured port for pstn on BR1-3550
interface FastEthernet0/13
 description Connected to PSTN phone
 switchport access vlan 100 -- (I also tried using access vlan 200 and
voice vlan 400)
 switchport mode access
4) Also created interface vlan on BR1-3550, not sure if needed, was getting
desperate :(
interface Vlan100
 ip address 10.10.100.2 255.255.255.0

5) Verified ephone configuration on pstn router. Correct mac-add
ephone  1
 device-security-mode none
 mac-address 0011.2189.C2A7
 type 7960
 button  1:1 2:2 3:3 4:4
 button  5:5 6:6

5) Power cycled pstn phone. Factory reset 123456789*0#.

6) Connected the pstn phone to HQ so that it would get the tftp
configuration from the Call Manager. Also tried to use option 150 to point
to CUCM on the pstn router. I even changed the phone with another 7960.

7) Tried to download the 7960 load file on the pstn router, but was not
successful as the pstn is not able to ping my Windows 7 desktop
PSTN-WAN#dir flash:
Directory of flash:/
1  -rw-3395  Mar 17 2008 15:23:58 -04:00  bgp-config
2  -rw-5814  Mar 20 2008 16:27:56 -04:00  fr-config
3  -rw-1646  Feb 23 2006 00:31:02 -05:00  sdmconfig-2811.cfg
4  -rw-  812032  Feb 23 2006 00:32:00 -05:00  es.tar
5  -rw- 1007616  Feb 23 2006 00:32:28 -05:00  common.tar
6  -rw-1038  Feb 23 2006 00:32:50 -05:00  home.shtml
7  -rw-  113152  Feb 23 2006 00:33:10 -05:00  home.tar
8  -rw-  511939  Feb 23 2006 00:33:32 -05:00  128MB.sdf
9  -rw-59846440  Jul 20 2009 21:08:26 -04:00
c2800nm-adventerprisek9_ivs-mz.124-24.T.bin

As you see I ran out of options. I would be thankful for any suggestion.

Best
Voiper
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Re: [OSL | CCIE_Voice] I Pass CCIE voice finally

2011-09-09 Thread bkvalent...@gmail.com
Congrats

Sent from my Verizon Wireless Phone

- Reply message -
From: Ahmed Ellboudy ahmed_ellbo...@rayacorp.com
Date: Fri, Sep 9, 2011 11:57 am
Subject: [OSL | CCIE_Voice] I Pass CCIE voice finally
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com

HI ALL ,



I Pass MY exam after 2nd attempt , finaly I did it my CCIE #30079 ,Thanks for 
IPX document it was a good documents for your preprations,

Thanks,



Ahmed Ellboudy, CCIE-voice # 30079




From: ccie_voice-boun...@onlinestudylist.com 
[ccie_voice-boun...@onlinestudylist.com] on behalf of Kshitij Singhi 
[martinian.ksin...@gmail.com]
Sent: Friday, September 09, 2011 3:43 PM
To: Ken Wyan
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] clock summer-time

Hi Ken,

Whatever I had sent across in my email is verified. The theory is vast with 
respect to NTP and the rest of the time zone stuff, but the configuration which 
I had sent across works as required for the exam.

The lab is a test of how much you know with respect to NTP etc, and not whether 
you are aware of the different time zones all over the world and the date when 
DST is active etc. For that, a Geography test would be more appropriate (and 
there is no way I would be taking THAT test again) ;) :)

On Fri, Sep 9, 2011 at 6:35 PM, Ken Wyan 
kew...@gmail.commailto:kew...@gmail.com wrote:
Dear Kshitiji,

Earlier I also thought your same way after referring to cisco docs for NTP.

But when we check time zones available in CUCM Date/Time groups ; some of the 
non-US locations have daylight saving times. Eg: North Asia East Standard / 
Daylight time ( GMT + 08:00 ). Therefore CUCM has pre-defined location based 
timezones ( DST for non US as well ).

Routers don't have such pre-defined time zones. PST EST PDT are just labels 
only. if we put clock summer-time command clock goes one hour ahead during US 
DST period. Obviously DST start  end dates differ in different locations in 
the world .(even same standard timezone can have different DST start  end 
dates based on country ; then how does Router determine DST start  end date 
from offset we configure in clock-timezone command?)

Some previous posts suggested to match router time with phone time (CUCM time)  
by putting or not putting summer-time command (in Router - Trial  error). But 
this will not help if we are outside Routers DST window , but inside CUCM DST 
window.

Below cisco docs confirm that clock summer-time can be used for US without 
start  end dates.

Anybody has more verified details on this?


Configure Summer Time (Daylight Savings Time)

To configure summer time (daylight savings time) in areas where it starts and 
ends on a particular day of the week each year, use the following command in 
global configuration mode:

Command

Purpose


clock summer-time zone recurring [week day month hh:mm week day month hh:mm 
[offset]]


Configure summer time.



If summer time in your area does not follow this pattern, you can configure the 
exact date and time of the next summer time events by using one of the 
following commands in global configuration mode:

Command

Purpose


clock summer-time zone date month date year hh:mm month date year hh:mm [offset]

or

clock summer-time zone date date month year hh:mm date month year hh:mm [offset]


Configure summer time.



For an example of configuring summer time, see the section Clock, Calendar, 
and NTP Configuration 
Exampleshttp://www.cisco.com/en/US/docs/ios/12_0/configfun/configuration/guide/fcgenral.html#wp4739
 at the end of this chapter.

Clock, Calendar, and NTP Configuration Examples

In the following example, a router with a system calendar has server 
associations with two other systems, transmits broadcast NTP packets, 
periodically updates the calendar, and redistributes time into VINES:

clock timezone PST -8


clock summer-time PDT recurring


ntp update-calendar


ntp server 192.168.13.57


ntp server 192.168.11.58


interface Ethernet 0/0


 ntp broadcast


vines time use-system










On Fri, Sep 9, 2011 at 4:28 PM, Kshitij Singhi 
martinian.ksin...@gmail.commailto:martinian.ksin...@gmail.com wrote:
Here is what needs to be done:

Use clock summer-time TIMEZONE offset from GMT

Hence, as an example, for PST/CST/EST it will be:

PST - - - clock time-zone PST -8
CST - - - clock time-zone CST -6
EST - - - clock time-zone EST -5

For ALL US sites, set the Daylight Savings to recurring. Hence, continuing from 
the above:

PST - - -  clock summer-time PDT recurring
CST - - - clock summer-time CDT recurring
EST - -  -clock summer-time EDT recurring

Don't worry about the actual date for DST. Just make sure that you choose a 
Date/Time Group on CUCM which has Daylight Savings Time along with the 
appropriate time zone. Also, try to ensure that the phones and the local 
routers have the same time.

For all non-US sites, there is no need to set DST. Just set the clock time-zone 
command.





[OSL | CCIE_Voice] Do not feed the trolls

2011-08-26 Thread bkvalent...@gmail.com


Remember.  Don't feed the trolls.___
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Re: [OSL | CCIE_Voice] UCCX Agent in ready state after login

2011-08-20 Thread bkvalent...@gmail.com
Did you try setting it in Desktop Administrator?

Sent from my Verizon Wireless Phone

- Reply message -
From: Wael Agina waelag...@gmail.com
Date: Sat, Aug 20, 2011 6:20 am
Subject: [OSL | CCIE_Voice] UCCX Agent in ready state after login
To: ccie_voice@onlinestudylist.com

Dears,

   Any one tried before to get ipccc agents in the ready state after login .
Normally agents went to Not Ready state and have to manually change it to
ready after they login from the service button.
I tried a lot in my home lab, but always agents login in Not Ready state.

Any idea ?


-- 

Thanks and Best Regards,
Wael Agina
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Re: [OSL | CCIE_Voice] 4-port HWIC-ESW - Phone can't power-on

2011-08-19 Thread bkvalent...@gmail.com
Do you have a POE power supply in the router?

- Reply message -
From: Bill Lake whl...@gmail.com
Date: Fri, Aug 19, 2011 8:25 pm
Subject: [OSL | CCIE_Voice] 4-port HWIC-ESW - Phone can't power-on
To: Ramcharan Arya ramcharan.a...@gmail.com
Cc: ccie_voice@onlinestudylist.com


what is the output of the following command
#show power inline



On Fri, Aug 19, 2011 at 4:41 PM, Ramcharan Arya ramcharan.a...@gmail.comwrote:

 Hi,

 I have cisco 2821 router and 4-port HWIC-WSP PoE card. Ip phone connected
 to one of the port on switch module but it is unable to power on.

 Here is module information:

 Router#sh inv | inc 4
 NAME: 4 Port FE Switch on Slot 0 SubSlot 0, DESCR: 4 Port FE Switch
 PID: HWIC-4ESW , VID: V01 , SN: FOC122434S2
 NAME: WIC/VIC/HWIC 0 Power Daughter Card, DESCR: 4-Port HWIC-ESW Power
 Daughter Card
 PID: ILPM-4, VID: V01 , SN: FOC122710SV


 Can you someone please share your configure of your router for comparison.

 Thanks,
 Ramcharan Arya


 ___
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Re: [OSL | CCIE_Voice] FXO lines !

2011-08-06 Thread bkvalent...@gmail.com
CME doesn't have css/partitions... instead, use COR.

Now, how exactly is this a ccie lab exam study question?  There are no FXO 
ports on the lab exam. 

- Reply message -
From: Shabeeb Mohammed shabeebc...@gmail.com
Date: Sat, Aug 6, 2011 9:40 am
Subject: [OSL | CCIE_Voice] FXO lines !
To: Cisco Voip cisco_new...@yahoo.com
Cc: ccie_voice@onlinestudylist.com


configure 2 different line to use the 2 fxo line separately... i mean each
fxo line will be under separate partition and css...


On Sat, Aug 6, 2011 at 1:34 PM, Cisco Voip cisco_new...@yahoo.com wrote:

 Hi all.

 I am using cisco 2801 with 2 fxo cards. Running CME version 7. I am using
 7940 phone. Now is there anyway that when the user of 7940 press line 1, he
 gets FXO line 1, and when he press 2 he gets FXO line 2.

 Is this possible or do i need to define outside access code for both these
 lines ?

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Re: [OSL | CCIE_Voice] SRST voice mail

2011-07-30 Thread bkvalent...@gmail.com
Build a did and use call routing rule for subscriber sign in, as opposed to 
attempt sign in.

Srst voicemail points to the DID.


Sent from my Verizon Wireless Phone

- Reply message -
From: Victor Malyuga victor_maly...@yahoo.com
Date: Sat, Jul 30, 2011 3:21 pm
Subject: [OSL | CCIE_Voice] SRST voice mail
To: ccie_voice@onlinestudylist.com

Suppose phones need to access voice mail in SRST mode and hear subscriber 
sign-in greetings when dial VM directly. VM located in a central site. We can 
not use alternate extension in VM.

I can use EPNM and calling party transformation mask  on VM Hunt Pilot to 
get this functionality.

However, if there is a requirement to play PSTN caller ANI in new messages, the 
transformation mask will cut the PSTN ANI to 4 digits.

How to overcome that?

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Re: [OSL | CCIE_Voice] [CME-DHCP PROBLEM]

2011-07-30 Thread bkvalent...@gmail.com
Can you wipe the phone?  Seems to be holding into an old ip address.

- Reply message -
From: CCIEVoiceKP ccievoic...@gmail.com
Date: Sat, Jul 30, 2011 3:29 pm
Subject: [OSL | CCIE_Voice] [CME-DHCP PROBLEM]
To: michael.se...@compucom.com michael.se...@compucom.com
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com, 
michael.se...@compucom.com michael.se...@compucom.com


Do a sh ip dhcp bind and see if your phones have IPs ... Then check the IP of 
your option 150 ... Is the 201 subnet correct?

Kevin Prater
907.301.6519

Sent from my iPhone and I have big thumbs ... So please excuse the typos.

On Jul 30, 2011, at 11:22 AM, michael.se...@compucom.com wrote:

 REQUESTED CONFIGURATION:
 
 Assign host addresses for the CME site starting at .50 and ranging to .69?
 
  
 
 Configuration:
 
  
 
 ip dhcp excluded-address 10.10.202.1 10.10.202.49
 
 ip dhcp excluded-address 10.10.202.70 10.10.202.254
 
  
 
 ip dhcp pool VOICE
 
network 10.10.202.0 255.255.255.0
 
default-router 10.10.202.1
 
option 150 ip 10.10.201.1
 
  
 
 PHONES WON’T REGISTER:
 
 DEBUG:
 
 BR2-RTR#
 
 Jul 30 19:05:53.918:   DHCPD: Sending notification of DISCOVER:
 
 Jul 30 19:05:53.922:   DHCPD: htype 1 chaddr 001b.d4c6.16a0
 
 Jul 30 19:05:53.922:   DHCPD: remote id 020a0a0aca0103000190
 
 Jul 30 19:05:53.922:   DHCPD: circuit id 
 
 Jul 30 19:05:53.922:   DHCPD: Seeing if there is an internally specified pool 
 class:
 
 Jul 30 19:05:53.922:   DHCPD: htype 1 chaddr 001b.d4c6.16a0
 
 Jul 30 19:05:53.922:   DHCPD: remote id 020a0a0aca0103000190
 
 Jul 30 19:05:53.922:   DHCPD: circuit id 
 
 Jul 30 19:05:53.922:   DHCPD: subnet [10.10.202.1,10.10.202.254 in address 
 pool VOICE is empty.
 
 Jul 30 19:05:53.922:   DHCPD: Sending notification of ASSIGNMENT FAILURE:
 
  
 
 When I change the configuration to the following everything works but it 
 doesn’t meet the requirements:
 
  
 
 Configuration:
 
  
 
 ip dhcp excluded-address 10.10.202.1 10.10.202.10
 
  
 
 ip dhcp pool VOICE
 
network 10.10.202.0 255.255.255.0
 
default-router 10.10.202.1
 
option 150 ip 10.10.201.1
 
  
 
 Any input would be appreciated, I think it’s a bug, but haven’t done a scrub 
 yet because I’m in the middle of doing a lab.
 
  
 
 Michael Sears
 
 CompuCom
 
 US West, Professional Services
 
 Senior IPT Engineer
 
 Office:+1 720.344.6833
 
 Mobile:   +1 303.328.5590
 
 Fax:+1 978.863.0740
 
 Conf:  +1 866-389-9494 #6159225  Host Code: 879 7896
 
 E-Mail:   michael.se...@compucom.com
 
 Web:   http://www.compucom.com
 
  
 
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Re: [OSL | CCIE_Voice] [CME-DHCP PROBLEM]

2011-07-30 Thread bkvalent...@gmail.com
Seriously, forget it.  Who cares?  Move on.  Your config was good.

Sent from my Verizon Wireless Phone

- Reply message -
From: Леонид Архипов leoni...@gmail.com
Date: Sat, Jul 30, 2011 7:39 pm
Subject: [OSL | CCIE_Voice] [CME-DHCP PROBLEM]
To: michael.se...@compucom.com michael.se...@compucom.com
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com, 
michael.se...@compucom.com michael.se...@compucom.com


In this case dhcp_srv has compiled excluded ranges producing a contiguous 
range. It's seen in the penultimate debug message.
 I'm new in this community, so greetings to all.

Отправлено с iPhone

31.07.2011, в 6:22, michael.se...@compucom.com написал(а):

 REQUESTED CONFIGURATION:
 
 Assign host addresses for the CME site starting at .50 and ranging to .69?
 
  
 
 Configuration:
 
  
 
 ip dhcp excluded-address 10.10.202.1 10.10.202.49
 
 ip dhcp excluded-address 10.10.202.70 10.10.202.254
 
  
 
 ip dhcp pool VOICE
 
network 10.10.202.0 255.255.255.0
 
default-router 10.10.202.1
 
option 150 ip 10.10.201.1
 
  
 
 PHONES WON’T REGISTER:
 
 DEBUG:
 
 BR2-RTR#
 
 Jul 30 19:05:53.918:   DHCPD: Sending notification of DISCOVER:
 
 Jul 30 19:05:53.922:   DHCPD: htype 1 chaddr 001b.d4c6.16a0
 
 Jul 30 19:05:53.922:   DHCPD: remote id 020a0a0aca0103000190
 
 Jul 30 19:05:53.922:   DHCPD: circuit id 
 
 Jul 30 19:05:53.922:   DHCPD: Seeing if there is an internally specified pool 
 class:
 
 Jul 30 19:05:53.922:   DHCPD: htype 1 chaddr 001b.d4c6.16a0
 
 Jul 30 19:05:53.922:   DHCPD: remote id 020a0a0aca0103000190
 
 Jul 30 19:05:53.922:   DHCPD: circuit id 
 
 Jul 30 19:05:53.922:   DHCPD: subnet [10.10.202.1,10.10.202.254 in address 
 pool VOICE is empty.
 
 Jul 30 19:05:53.922:   DHCPD: Sending notification of ASSIGNMENT FAILURE:
 
  
 
 When I change the configuration to the following everything works but it 
 doesn’t meet the requirements:
 
  
 
 Configuration:
 
  
 
 ip dhcp excluded-address 10.10.202.1 10.10.202.10
 
  
 
 ip dhcp pool VOICE
 
network 10.10.202.0 255.255.255.0
 
default-router 10.10.202.1
 
option 150 ip 10.10.201.1
 
  
 
 Any input would be appreciated, I think it’s a bug, but haven’t done a scrub 
 yet because I’m in the middle of doing a lab.
 
  
 
 Michael Sears
 
 CompuCom
 
 US West, Professional Services
 
 Senior IPT Engineer
 
 Office:+1 720.344.6833
 
 Mobile:   +1 303.328.5590
 
 Fax:+1 978.863.0740
 
 Conf:  +1 866-389-9494 #6159225  Host Code: 879 7896
 
 E-Mail:   michael.se...@compucom.com
 
 Web:   http://www.compucom.com
 
  
 
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 www.PlatinumPlacement.com
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Re: [OSL | CCIE_Voice] Fwd: MGCP Gateway reg issue (not l2 TEI or l3 bind)

2011-07-17 Thread bkvalent...@gmail.com
Or if the port is physically down... like not plugged in.

- Reply message -
From: Shabeeb Mohammed shabeebc...@gmail.com
Date: Sun, Jul 17, 2011 9:19 am
Subject: [OSL | CCIE_Voice] Fwd: MGCP Gateway reg issue (not l2 TEI or l3 bind)
To: Ashraf Ayyash ash.ayy...@gmail.com
Cc: ccie_voice@onlinestudylist.com, steven moran smoran...@gmail.com


The only reasons for MGCP to not register with the CUCM are -

a. Hostname mismatch
b. Domain Mismatch
c. cucm in not binded to the serial interface..

regards
shabeeb


On Sun, Jul 17, 2011 at 2:46 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote:

 have you got it registers with the CCM before on any of your VM spanshots ?

 i faced this issue before and it only fixed when i have reverted the
 ccm snapshot to older one


 Thanks
 Ash

 On Sun, Jul 17, 2011 at 11:35 AM, steven moran smoran...@gmail.com
 wrote:
 
 
  No matter what I do I cannot get the MGCP gateway to show as registered
 in
  CUCM 7.02.  I think I have configured up the gateway in CUCM correctly.
 I
  cut and paste the router name from the sh ccm-m and the card is
 configured
  in slot 0 (aim-voice-30 (2 subunits) subunit 1 vwic-2mft-E1 0/0.  After
 all
  the config is done I apply the no mgcp/mgcp command.
 
  I can see from the DNA that the dialled number is correctly routed to
  the MGCP gateway but a sh isdn q931 shows no output - ie the call never
 gets
  there.  I aslo tried restarting the ris data collector and CM service on
  both pub and Sub to see if this was just mis represented in the GUI.
 When I
  reload the router the L2 isdn status goes back to TEI assigned until I
  remove the isdn bind-l3 from the loopback restart mgcp and then add it
 back
  in again.
  Here is the sh run
 
  Current configuration : 2717 bytes
  !
  ! No configuration change since last restart
  !
  version 12.4
  service timestamps debug datetime msec
  service timestamps log datetime msec
  no service password-encryption
  !
  hostname BR1-RTR
  !
  boot-start-marker
  boot system flash:c2600-adventerprisek9_ivs-mz.124-15.T14.bin
  boot-end-marker
  !
  enable secret 5 $1$LaQO$FR59LroSkjvjxYkzDsZ1N/
  !
  no aaa new-model
  clock timezone AEST 9
  clock summer-time AEST recurring 1 Sun Apr 1:00 last Sun Oct 1:00
  network-clock-participate wic 1
  network-clock-select 1 E1 0/1
  voice-card 0
   dspfarm
   no local-bypass
  !
  ip cef
  !
  !
  !
  !
  ip domain name smlab.com
  !
  multilink bundle-name authenticated
  !
  isdn switch-type primary-net5
  !
  !
  !
  !
  !
  !
  !
  !
  !
  !
  !
  !
  !
  !
  !
  !
  !
  !
  !
  !
  archive
   log config
hidekeys
  !
  !
  !
  !
  controller E1 0/1
   pri-group timeslots 1-6,16 service mgcp
  !
  controller E1 0/2
  !
  !
  !
  !
  !
  interface Loopback0
   description management ip address
   ip address 10.10.110.2 255.255.255.255
   ip ospf network point-to-point
  !
  interface FastEthernet0/0
   no ip address
   duplex auto
   speed auto
  !
  interface FastEthernet0/0.130
   description data vlan to switch
   encapsulation dot1Q 130 native
   ip address 10.10.101.1 255.255.255.0
   ip helper-address 10.10.210.10
  !
  interface FastEthernet0/0.240
   description br1 voice vlan
   encapsulation dot1Q 240
   ip address 10.10.201.1 255.255.255.0
   ip helper-address 10.10.210.10
  !
  interface Serial0/0
   no ip address
   encapsulation frame-relay
   frame-relay lmi-type ansi
  !
  interface Serial0/0.1 point-to-point
   ip address 10.10.111.2 255.255.255.0
   snmp trap link-status
   frame-relay interface-dlci 101
  !
  interface FastEthernet0/1
   no ip address
   shutdown
   duplex auto
   speed auto
  !
  interface Serial0/1:15
   no ip address
   encapsulation hdlc
   isdn switch-type primary-net5
   isdn incoming-voice voice
   isdn bind-l3 ccm-manager
   no cdp enable
  !
  router eigrp 1
   network 10.0.0.0
   no auto-summary
  !
  ip forward-protocol nd
  !
  !
  no ip http server
  no ip http secure-server
  !
  !
  !
  !
  !
  !
  control-plane
  !
  !
  !
  voice-port 0/1:15
  !
  ccm-manager switchback immediate
  ccm-manager redundant-host 10.10.210.10
  ccm-manager mgcp
  !
  mgcp
  mgcp call-agent 10.10.210.11 service-type mgcp version 0.1
  mgcp dtmf-relay voip codec all mode out-of-band
  mgcp bind control source-interface Loopback0
  mgcp bind media source-interface Loopback0
  !
  mgcp profile default
  !
  !
  !
  !
  !
  !
  gatekeeper
   shutdown
  !
  !
  line con 0
   exec-timeout 0 0
   logging synchronous
  line aux 0
  line vty 0 4
   exec-timeout 0 0
   privilege level 15
   logging synchronous
   no login
   transport preferred none
   transport input telnet
  line vty 5 15
   exec-timeout 0 0
   privilege level 15
   logging synchronous
   no login
   transport preferred none
   transport input telnet
  !
  ntp clock-period 17207966
  ntp server 10.10.___
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Re: [OSL | CCIE_Voice] MGCP Gateway reg issue (not l2 TEI or l3 bind)

2011-07-17 Thread bkvalent...@gmail.com
CUCM does not support fractional PRIs in MGCP officially.  

- Reply message -
From: steven moran smoran...@gmail.com
Date: Sun, Jul 17, 2011 8:45 pm
Subject: [OSL | CCIE_Voice] MGCP Gateway reg issue (not l2 TEI or l3 bind)
To: ccie_voice@onlinestudylist.com

Dear all,

thanks for all of the suggestions.

I have removed the gateway from CM and restarted the Pub abd Sub servers
from the CLI (utils system restart)

I removed the mgcp bind media and control from the loopback interface and
reloaded the BR1_RTR (after saving).  This results in the gateway binding to
the 1st BR1 serial int 10.10.111.2.

I am curious as to why the logs show that the mgcp tries to bin to the full
PRI even though the pri-group specifies that it should only bind to channels
1-6 fro mgcp.

Does the 2651xm router support fractional pri bind for mgcp?

Here is the config

BR1-RTR(config)#do sh run
Building configuration...
Current configuration : 2653 bytes
!
! Last configuration change at 07:53:31 AEST Mon Jul 18 2011
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname BR1-RTR
!
boot-start-marker
boot system flash:c2600-adventerprisek9_ivs-mz.124-15.T14.bin
boot-end-marker
!
enable secret 5 $1$LaQO$FR59LroSkjvjxYkzDsZ1N/
!
no aaa new-model
clock timezone AEST 9
clock summer-time AEST recurring 1 Sun Apr 1:00 last Sun Oct 1:00
network-clock-participate wic 1
network-clock-select 1 E1 0/1
voice-card 0
 dspfarm
 no local-bypass
!
ip cef
!
!
!
!
ip domain name lab.com
!
multilink bundle-name authenticated
!
isdn switch-type primary-net5
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
archive
 log config
  hidekeys
!
!
!
!
controller E1 0/1
 pri-group timeslots 1-6,16 service mgcp
!
controller E1 0/2
 shutdown
!
!
!
!
!
interface Loopback0
 description management ip address
 ip address 10.10.110.2 255.255.255.255
 ip ospf network point-to-point
!
interface FastEthernet0/0
 no ip address
 duplex auto
 speed auto
!
interface FastEthernet0/0.130
 description data vlan to switch
 encapsulation dot1Q 130 native
 ip address 10.10.101.1 255.255.255.0
 ip helper-address 10.10.210.10
!
interface FastEthernet0/0.240
 description br1 voice vlan
 encapsulation dot1Q 240
 ip address 10.10.201.1 255.255.255.0
 ip helper-address 10.10.210.10
!
interface Serial0/0
 no ip address
 encapsulation frame-relay
 frame-relay lmi-type ansi
!
interface Serial0/0.1 point-to-point
 ip address 10.10.111.2 255.255.255.0
 snmp trap link-status
 frame-relay interface-dlci 101
!
interface FastEthernet0/1
 no ip address
 shutdown
 duplex auto
 speed auto
!
interface Serial0/1:15
 no ip address
 encapsulation hdlc
 isdn switch-type primary-net5
 isdn incoming-voice voice
 isdn bind-l3 ccm-manager
 no cdp enable
!
router eigrp 1
 network 10.0.0.0
 no auto-summary
!
ip forward-protocol nd
!
!
no ip http server
no ip http secure-server
!
!
!
!
!
!
control-plane
!
!
!
voice-port 0/1:15
!
ccm-manager switchback immediate
ccm-manager redundant-host 10.10.210.10
ccm-manager mgcp
!
mgcp
mgcp call-agent 10.10.210.11 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
!
mgcp profile default
!
!
!
!
!
!
gatekeeper
 shutdown
!
!
line con 0
 exec-timeout 0 0
 logging synchronous
line aux 0
line vty 0 4
 exec-timeout 0 0
 privilege level 15
 logging synchronous
 no login
 transport preferred none
 transport input telnet
line vty 5 15
 exec-timeout 0 0
 privilege level 15
 logging synchronous
 no login
 transport preferred none
 transport input telnet
!
ntp clock-period 17207971
ntp server 10.10.100.2
!
end
Here is the sh isdn stat


BR1-RTR(config)#do sh isdn stat
Global ISDN Switchtype = primary-net5
%Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output may not
apply
ISDN Serial0/1:15 interface
dsl 0, interface ISDN Switchtype = primary-net5
L2 Protocol = Q.921 0x  L3 Protocol(s) = CCM MANAGER 0x0003
Layer 1 Status:
ACTIVE
Layer 2 Status:
TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
Layer 3 Status:
0 Active Layer 3 Call(s)
Active dsl 0 CCBs = 0
The Free Channel Mask:  0x803F
Number of L2 Discards = 0, L2 Session ID = 1
Total Allocated ISDN CCBs = 0
BR1-RTR(config)#
Here is the sh ccm

MGCP Domain Name: BR1-RTR.lab.com
PriorityStatus   Host

Primary Registered   10.10.210.11
First BackupBackup Ready 10.10.210.10
Second Backup   None
Current active Call Manager:10.10.210.11
Backhaul/Redundant link port:   2428
Failover Interval:  30 seconds
Keepalive Interval: 15 seconds
Last keepalive sent:10:24:58 AEST Jul 18 2011 (elapsed time:
00:00:08)
Last MGCP traffic time: 10:24:58 AEST Jul 18 2011 (elapsed time:
00:00:08)
Last failover time: 10:11:26 AEST Jul 18 2011 from
(10.10.210.11)
Last switchback 

Re: [OSL | CCIE_Voice] Unicode support in CUCM

2011-07-12 Thread bkvalent...@gmail.com
You are looking for locales.


- Reply message -
From: Ravindra Lakpriya lakpr...@gmail.com
Date: Tue, Jul 12, 2011 7:11 am
Subject: [OSL | CCIE_Voice] Unicode support in CUCM
To: ccie_voice@onlinestudylist.com

Guy first of all my apologies... even though my CCIE lab date is 20 days
away from today...But this question is something related to work :D

I wonder any of you came across a situation where you have to configure
display name and line text lable in other languages. I believe for that we
use uni code. I can use unicode in the admin pages. But still at the phone
level it will display as junk character. If you have any experience on that
please share. really appreciated.

-- 
Ravindra Lakpriya
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Re: [OSL | CCIE_Voice] PSTN-WAN Enable Password

2011-07-04 Thread bkvalent...@gmail.com
Disabling password recovery does NOT mean you cannot recover the router.  It 
means if you recover the router, you will lose the config.  Recover the router. 
 Then put  your config back on it.  Then change the password to something you 
know.

Sent from my Verizon Wireless Phone

- Reply message -
From: Bryan Byrne ccie.25...@gmail.com
Date: Mon, Jul 4, 2011 11:31 am
Subject: [OSL | CCIE_Voice] PSTN-WAN Enable Password
To: Bill Lake whl...@gmail.com
Cc: ccie voice ccie_voice@onlinestudylist.com


It's in the default config.  I didn't catch it until after I put it in the 
router.  


On Jul 4, 2011, at 11:29 AM, Bill Lake wrote:

 Did you disable password recovery?
 
 On Mon, Jul 4, 2011 at 8:14 AM, Bryan Byrne ccie.25...@gmail.com wrote:
 I'm in the process of finishing out a home lab build and I ran into a small 
 problem with the PSTN-WAN router.  I forgot to remove the enable password 
 when I pasted in the config and I am not running the right version of ROMMON 
 that will let me do a factory reset of the router.  Does anyone happen to 
 know the enable secret for the PSTN-WAN router?
 
 -Bryan
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Re: [OSL | CCIE_Voice] Lab result status

2011-06-23 Thread bkvalent...@gmail.com
Maybe your proctor is on summer vacation.

- Reply message -
From: Adil Shaikh adil.sha...@gmail.com
Date: Thu, Jun 23, 2011 6:44 pm
Subject: [OSL | CCIE_Voice] Lab result status
To: Pablo Meneses pmenese...@gmail.com
Cc: ccie voice ccie_voice@onlinestudylist.com


One week no result. This is too long.

Has anyone read any fine print which specifies within what timeframe they
will be providing result?

-adil

On Fri, Jun 24, 2011 at 3:50 AM, Pablo Meneses pmenese...@gmail.com wrote:

 I just sent her an email. Let's see who it goes.

 -Pablo Meneses.

 On Thu, Jun 23, 2011 at 10:38 AM, William Affeldt 
 william.affe...@yahoo.com wrote:

 Her name is Kathe Saccenti. I would just call Support and ask to talk to
 her. She is in charge of the proctors and the testing environment.

 --- On *Thu, 6/23/11, Pablo Meneses pmenese...@gmail.com* wrote:


 From: Pablo Meneses pmenese...@gmail.com
 Subject: Re: [OSL | CCIE_Voice] Lab result status
 To: William Affeldt william.affe...@yahoo.com
 Cc: George Goglidze gogli...@gmail.com, ccie voice 
 ccie_voice@onlinestudylist.com
 Date: Thursday, June 23, 2011, 4:23 PM


 Hello Bill,

 Thanks for that information. Do you know Kathe's last name?

 Regards.

 -Pablo Meneses.

 On Wed, Jun 22, 2011 at 9:09 PM, William Affeldt 
 william.affe...@yahoo.comhttp://us.mc1146.mail.yahoo.com/mc/compose?to=william.affe...@yahoo.com
  wrote:

  You should look at this as a positive. If they deleted your test they
 may just give you a passing grade. You will never know though. The person
 you should contact is the lab ops manager. Her name is Kathe. She is a nice
 person and will take your call at any time.

 Sent from my iPhone

 On Jun 22, 2011, at 4:56 PM, George Goglidze 
 gogli...@gmail.comhttp://us.mc1146.mail.yahoo.com/mc/compose?to=gogli...@gmail.com
 wrote:

  WTF man, I can't believe this is happening to you.
 I wouldn't be able to sleep until I got the results.


 On Wed, Jun 22, 2011 at 10:46 PM, Pablo Meneses 
 http://us.mc1146.mail.yahoo.com/mc/compose?to=pmenese...@gmail.com
 pmenese...@gmail.comhttp://us.mc1146.mail.yahoo.com/mc/compose?to=pmenese...@gmail.com
  wrote:

 Can you believe that I haven't received the answer yet!
 This has been the longest wait ever.
 I even opened a ticket with Cert Support and so far I have just received
 templates.
 Does someone else have gone through this?
 I'm starting to think that they messed up on my exam.

 -Pablo Meneses.


 On Wed, Jun 22, 2011 at 2:47 AM, Julien Krieger 
 http://us.mc1146.mail.yahoo.com/mc/compose?to=krieger.jul...@gmail.com
 krieger.jul...@gmail.comhttp://us.mc1146.mail.yahoo.com/mc/compose?to=krieger.jul...@gmail.com
  wrote:

 Hi Pablo,
 How did it go? Have you passed?


 2011/6/21 Pablo Meneses 
 http://us.mc1146.mail.yahoo.com/mc/compose?to=pmenese...@gmail.com
 pmenese...@gmail.comhttp://us.mc1146.mail.yahoo.com/mc/compose?to=pmenese...@gmail.com
 

 Thanks Cris.

 Looking forward to getting my result.


 On Tue, Jun 21, 2011 at 1:23 PM, Cristobal Priego 
 http://us.mc1146.mail.yahoo.com/mc/compose?to=cristobalpri...@gmail.com
 cristobalpri...@gmail.comhttp://us.mc1146.mail.yahoo.com/mc/compose?to=cristobalpri...@gmail.com
  wrote:

 i know man
 i took my last attempt on a Friday as well In San Jose

 i didn't get my score until next Monday at 14:45 PST


 2011/6/21 Pablo Meneses 
 http://us.mc1146.mail.yahoo.com/mc/compose?to=pmenese...@gmail.com
 pmenese...@gmail.comhttp://us.mc1146.mail.yahoo.com/mc/compose?to=pmenese...@gmail.com
 

 Hopefully that's true. Longest wait EVER!!!

 -Pablo Meneses.


 On Tue, Jun 21, 2011 at 1:19 PM, Cristobal Priego 
 http://us.mc1146.mail.yahoo.com/mc/compose?to=cristobalpri...@gmail.com
 cristobalpri...@gmail.comhttp://us.mc1146.mail.yahoo.com/mc/compose?to=cristobalpri...@gmail.com
  wrote:

 you will get today, wait 2.5 more hours
 that's what happens when you take it on Friday

   2011/6/21 Pablo Meneses 
 http://us.mc1146.mail.yahoo.com/mc/compose?to=pmenese...@gmail.com
 pmenese...@gmail.comhttp://us.mc1146.mail.yahoo.com/mc/compose?to=pmenese...@gmail.com
 

  Hi experts,

 I am wondering how long it takes for you to received your score report
 this days. I've taken my lab last Friday and I have NOT received the answer
 yet. Yesterday I logged to the CCIE Online Tool and it said HOLD on the
 PASS/FAIL field, does someone know what that means? Has someone else gone
 through this?

 I will look forward to your reply as I figure most of you understand the
 feeling of not having the answer.

 Regards.

 -Pablo Meneses.

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Re: [OSL | CCIE_Voice] Need vouchers

2011-06-20 Thread bkvalent...@gmail.com
Looking for $20 each.  

- Reply message -
From: adam compton com...@gmail.com
Date: Mon, Jun 20, 2011 3:06 pm
Subject: [OSL | CCIE_Voice] Need vouchers
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com

If you have any extra vouchers you need to get rid of let me know.  I'm
looking for 10.

Adam Compton
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Re: [OSL | CCIE_Voice] OT: MeetingPlace Express

2011-06-09 Thread bkvalent...@gmail.com
www.cisco.com/tac


- Reply message -
From: George Goglidze gogli...@gmail.com
Date: Thu, Jun 9, 2011 12:39 pm
Subject: [OSL | CCIE_Voice]  OT: MeetingPlace Express
To: CCIE Voice online groupstudy ccie_voice@onlinestudylist.com

Hi all,

Cisco has announced MeetingPlace Express will be in the labs, starting
NOW!


Don't panick guys, MeetingPlace Express is not in a lab :)

This is an OT, so feel free to ignore. Unless of course you think you can
help.

I had to change the domain name, on MeetingPlace Express 7.0 server,
did the following:

1) sudo mpx_sys stop
2) net
option 4 - change to new domain name
3) save
4) /sbin/shutdown -r now

The server is back up, and I can access the web interface, and schedule a
meeting. But I can't join the web conference.
I ran the Browser Test and it gives me the following error:

--

Player:
Version: 10 Revision:  102
Platform: WIN
Cisco Unified MeetingPlace Express Add-In: Installed
App Server Connection: Successful
Flash Com Connection: Timed Out
server: rtmp://
CLN-MPX02.myserver.com:1935/?rtmp://localhost:1935,rtmp://CLN-MPX02.
myserver.com:80/?rtmp://localhost:1935/,rtmpt://CLN-MPX02.
myserver.com:80/?rtmp://localhost:1935/
--

The problem is, the myserver.com domain, is OLD domain, that I changed.
and I do not understand why is it still trying to connect to old domain
name?
When I do net - list current configuration... the new domain is listed
correctly.

Many thanks to all for helping,
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Re: [OSL | CCIE_Voice] I just got my Fail result

2011-06-08 Thread bkvalent...@gmail.com
Your score reports are protected by the NDA agreement. Obviously so are the 
test questions.  Stop breaking NDA.

- Reply message -
From: Michael Luo hout...@gmail.com
Date: Wed, Jun 8, 2011 12:11 pm
Subject: [OSL | CCIE_Voice] I just got my Fail result
To: Ki Wi kiwi.vo...@gmail.com
Cc: ccie_voice@onlinestudylist.com, Chris Green voice5...@yahoo.com


Customized prompt was required.  Thus I recorded my prompt.

Set agent to not ready as required.

For CSQ, there's no specific requirement.  Thus I created my own CSQ.

I'm pretty familiar with UCCX.  I can do whatever they ask me to do.  I just
need to understand their grading philosophy so I don't miss the trivial
things and lose the whole portion.

Michael

On Wed, Jun 8, 2011 at 8:54 AM, Ki Wi kiwi.vo...@gmail.com wrote:

 Maybe you missed out some requirement?

 Such as they want you to use system recording but you recorded your own?

 They want you to set agent to not ready after the call ended? Set the agent
 to work state after the call ended?

 Queue the call to special CSQ, etc etc.

 UCCX is a nightmare.!

 On Wed, Jun 8, 2011 at 10:37 PM, Michael Luo hout...@gmail.com wrote:

 I meant - I've done everything right in UCCX.


 On Wed, Jun 8, 2011 at 7:37 AM, Michael Luo hout...@gmail.com wrote:

 I also got my report.  That's very confusing to me as well.

 For example, I'm pretty sure I made a mistake on QoS.  But I got 100%.

 Then I'm pretty sure I've don't everything right in UCCX but got 38%.


 On Wed, Jun 8, 2011 at 4:02 AM, adam compton com...@gmail.com wrote:

 I know what you mean.  I thought I nailed QOS 100% and got a zero.

 On Wed, Jun 8, 2011 at 3:04 AM, Chris Green voice5...@yahoo.comwrote:

  Hi All,

 I just got my fail result, and what it shocked me is the Call Routing
 mark for 40% while I am sure I done what was asked for and I tested as 
 well.

 I also got 100% in 4 different section but what it killed me is Call
 Routing which I was quite confident about that section :(

 Chris

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 ___
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 please visit www.ipexpert.com

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 www.PlatinumPlacement.com




 ___
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Re: [OSL | CCIE_Voice] UCME/CUE

2011-05-29 Thread bkvalent...@gmail.com
Make sure vlan 400 is up/up.

If not, that would do it.

Hth

- Reply message -
From: Randall Crumm rrcr...@yahoo.com
Date: Sun, May 29, 2011 5:07 pm
Subject: [OSL | CCIE_Voice] UCME/CUE
To: Online Study ccie_voice@onlinestudylist.com

HI,
I'm having a weird issue. (see snipits below)

I configured mt CME for CUE(not telephony-service yet) and I cannot ser ser 0/0 
ses to it
I can ping 10.10.202.2 from hq rtr
The cli just has a blank like

Anyone have an ideas or run into this?

Thanks,
Randall


interface Service-Engine0/0
 ip unnumbered Vlan400
 service-module ip address 10.10.202.2 255.255.255.0
 service-module ip default-gateway 10.10.202.1

ip route 10.10.202.2 255.255.255.255 Service-Engine0/0


interface Vlan400
 ip address 10.10.202.1 255.255.255.0___
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Re: [OSL | CCIE_Voice] Lab strategy

2011-05-20 Thread bkvalent...@gmail.com
Emin,

You will not have to guess at what they want you to do.  The instructions will 
be pretty clear as to what will earn you points.  Doing things that aren't 
asked for will not help you, unless they are required for you to accomplish the 
task that is asked of you.  

For example, if a question asks you to build a call handler in CUC, you 
obviously will have to build a port group.  Don't think they care about time of 
day routing or time zones if the question doesn't ask for it.  Does that make 
sense?

If you are unclear about what a question is asking you to do, you should ask 
the proctor.  That is why they are there.

Best of luck on your attempt.

- Reply message -
From: Emin Guliyev eguli...@fidelus.com
Date: Fri, May 20, 2011 1:23 pm
Subject: [OSL | CCIE_Voice] Lab strategy
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com

Hey guys,

I will be going for my 1st attempt this month. Wanted to verify 
some minor details. Are we supposed to configure correct cadence/locale on 
endpoints and gateways? Also, do you lose any points for mismatching timezones 
on UC mailboxes? For afterhours mailbox, are we supposed to use schedules on UC 
and use Closed greeting , or just add a new call handler or subscriber?

Regards,

Emin
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Re: [OSL | CCIE_Voice] Problem with Time on CME

2011-05-09 Thread bkvalent...@gmail.com
7940/60 phones use the router's time.  The newer phones obey the timezone 
command under telephony service.

HTH

Brian

- Reply message -
From: George Goglidze gogli...@gmail.com
Date: Mon, May 9, 2011 7:36 pm
Subject: [OSL | CCIE_Voice] Problem with Time on CME
To: Abel ... midga...@gmail.com
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com


Put the configs here...

I don't know about 7965 but CIPC gets it's time from computer I think, 
therefore you can't provide time for CIPC from a router.

For 7965 you need to have correct timezone set in telephony-service... timezone 
number
And then create cnf is mandatory, otherwise it doesn't do it straight away.

And if it's SIP try setting time-zone and ntp server too under voice register 
global
SIP phones get time from ntp server directly, so you should provide one for it.

Regards,

Sent from my iPad

On 10 May 2011, at 00:03, Abel ... midga...@gmail.com wrote:

 Hello everyone, I'm having this stange issue with CME 7.0.1, the router is 
 getting correct time from NTP and some phones are getting the right time from 
 the CME (7940) and other don't (7965 and CIPC), if someone know how solve 
 this issue, please let me know.
 
 Thanks
 
 Abel...
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Re: [OSL | CCIE_Voice] Problem with Time on CME

2011-05-09 Thread bkvalent...@gmail.com
Is ny -4 the same as timezone 14?

Can't remember.

Is the ntp in synch on the router?

Show ntp status
Show clock

- Reply message -
From: Bill Lake whl...@gmail.com
Date: Mon, May 9, 2011 7:50 pm
Subject: [OSL | CCIE_Voice] Problem with Time on CME
To: Abel ... midga...@gmail.com
Cc: ccie_voice@onlinestudylist.com


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Re: [OSL | CCIE_Voice] UCCX issues in Proctor labs with hardware vpn

2011-05-08 Thread bkvalent...@gmail.com
You might need to review logs from the rmcm subsystem.   I discovered a bug in 
5.0 where a database change of skill wouldn't impact rmcm after a nic failure.  
You could be running into the same bug.  Not sure if it was docs in 7.0. A 
reboot or reset of the node was required to reset the rmcm.   

Do you have a minimum skill level set in the queue?   Agents are more skilled 
than minimum?

Brian

- Reply message -
From: CCIE for Me cciefo...@hotmail.com
Date: Sun, May 8, 2011 1:58 pm
Subject: [OSL | CCIE_Voice] UCCX issues in Proctor labs with hardware vpn
To: ccie_voice@onlinestudylist.com

Hi all,

just trying to eliminate some possibilities.  Recently I have been having 
issues with uccx recognizing the ready status of agents.  I do this all the 
time for my job and in the boot camp with no issues and am confident everything 
is set up correctly.  The only difference I see is that I am using hardware vpn 
to join Proctor Labs and thinking everything in between (VPN tunnell, etc) may 
be preventing calls from routing properly to agents, or the VM images are 
screwy.  I can log in to the agents fine, change my status, etc.  If I do a CSQ 
with a resource group, calls route fine.  If I change the same CSQ to 
resource/skill based, calls won't route.

Any thoughts or suggestions?

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Re: [OSL | CCIE_Voice] Cisco has identified multiple vulnerabilities in Cisco Unified Communications Manager.

2011-05-05 Thread bkvalent...@gmail.com
This is not an uncommon occurance.

- Reply message -
From: George Goglidze gogli...@gmail.com
Date: Thu, May 5, 2011 6:09 am
Subject: [OSL | CCIE_Voice] Cisco has identified multiple vulnerabilities in 
Cisco Unified Communications Manager.
To: mihal caro mihalc...@gmail.com
Cc: ccie_voice@onlinestudylist.com


Thanks for sharing Mihal...

I wonder who exposed these vulnerabilities... If it were Cisco itself, I bet
they wouldn't have announced it... :)

Regards,

On Thu, May 5, 2011 at 10:02 AM, mihal caro mihalc...@gmail.com wrote:

 *Hi All *
 **
 *Just to let you know that **Cisco has identified multiple vulnerabilities
 in Cisco Unified Communications Manager.  The vulnerabilities include a SIP
 DoS vulnerability, Directory Transversal Vulnerability, and a SQL Injection
 vulnerability.*

 * *

 The vulnerability is documented at the following address:
 *
 http://www.cisco.com/en/US/products/products_security_advisory09186a0080b79904.shtml
 *
 **
 *regards.*

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Re: [OSL | CCIE_Voice] sh run for router

2011-04-19 Thread bkvalent...@gmail.com
show run all 

- Reply message -
From: Erwan Erwan e_er...@yahoo.com
Date: Tue, Apr 19, 2011 12:46 pm
Subject: [OSL | CCIE_Voice] sh run for router
To: ccie_voice@onlinestudylist.com

hi all,
 
does anyone know what is the sh run ' command to see any hidden  config that 
not show up in usual sh run , like : sh run all or whatever /
 
tks

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Re: [OSL | CCIE_Voice] CUE to use loopback IP

2011-04-18 Thread bkvalent...@gmail.com

Your loopback should have a 32 bit mask.  Your static route isn't active 
because the connected route has a lower administrative distance. 

- Reply message -
From: Rogers Ochieng rogersochi...@gmail.com
Date: Mon, Apr 18, 2011 12:31 pm
Subject: [OSL | CCIE_Voice] CUE to use loopback IP
To: Wilson Bolanos wbola...@pvt.com
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com


You are right Wilson, it's only needed on the Loopback interface.

On 18 April 2011 19:25, Wilson Bolanos wbola...@pvt.com wrote:

  Just for giggles wr the config and then reboot.



 The Design guide for CUE does not show putting the ip ospf network
 point-to-point on the interface Service-Engine0/0 from what I saw.  Is the
 Design guide incorrect?



 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *romain mullier
 *Sent:* Monday, April 18, 2011 9:58 AM
 *To:* Rogers Ochieng
 *Cc:* ccie_voice@onlinestudylist.com
 *Subject:* Re: [OSL | CCIE_Voice] CUE to use loopback IP



 Tried that too. that still does not do it.

 interface Loopback1
  ip address 10.10.222.1 255.255.255.252
  ip ospf network point-to-point
 !
 interface Service-Engine0/0
  ip unnumbered Loopback1
  ip ospf network point-to-point
  service-module ip address 10.10.222.2 255.255.255.252
  service-module ip default-gateway 10.10.222.1
 !
 ip route 10.10.222.2 255.255.255.255 Service-Engine0/0

 Still can't ping for the router iself.
 BR2-RTR#ping 10.10.222.2
 Type escape sequence to abort.
 Sending 5, 100-byte ICMP Echos to 10.10.222.2, timeout is 2 seconds:
 .

 It has got to be something specific to loopbacks be cause when I use the
 vlan interface it works right away.

  On Mon, Apr 18, 2011 at 11:49 AM, Rogers Ochieng rogersochi...@gmail.com
 wrote:

 You need ip ospf network point-to-point on the int service-engine 0/0 too

 On 18 April 2011 18:11, romain mullier romain.mull...@gmail.com wrote:


 Hi guys,
 So instead of using a vlan interface to bind the CUE module on BR2, I tried
 to use a loopback and for some reason I cannot ping the module. config
 below. (it works fine with a vlan or  ethernet interface)

 interface Loopback1
  ip address 10.10.222.1 255.255.255.252
  ip ospf network point-to-point
 !
  interface Service-Engine0/0
  ip unnumbered Loopback1
  service-module ip address 10.10.222.2 255.255.255.252
  service-module ip default-gateway 10.10.222.1
 !
 ip route 10.10.222.2 255.255.255.255 Service-Engine0/0
 !

 BR2-RTR#ping 10.10.222.2
 Type escape sequence to abort.
 Sending 5, 100-byte ICMP Echos to 10.10.222.2, timeout is 2 seconds:
 ...

 Any thoughts?

 Thx

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Re: [OSL | CCIE_Voice] UCCX Script (Redirect call based on the calling number)

2011-04-10 Thread bkvalent...@gmail.com
I believe uccx does not understand the + symbol.  

- Reply message -
From: Naoufal Kerboute naou...@mhdinfotech.com
Date: Sun, Apr 10, 2011 7:21 am
Subject: [OSL | CCIE_Voice] UCCX Script (Redirect call based on the calling 
number)
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com

Dear gents,

I'm working on UCCX section and I'm trying to reroute some calls coming from 
Spain (+34) to a specific extension. I've setup the script and it's working 
only if I set the calling number variable to full Spain PSTN number, but let 
take the case for many number from Spain.
How can I reroute calls coming from spain to a specific extension (I don't want 
to much the full muber, I want to much only calling number start with +34)

Any ideas?

Thanks a lot
Naoufal



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Re: [OSL | CCIE_Voice] Hardware conference is stuck in state TCP_CONN_ERROR

2011-03-29 Thread bkvalent...@gmail.com
Possibly a replication issue?... is ccm3 the Pub?

Sent from my Verizon Wireless Phone

- Reply message -
From: Shrini linuxbos...@gmail.com
Date: Tue, Mar 29, 2011 12:44 pm
Subject: [OSL | CCIE_Voice] Hardware conference is stuck in state   
TCP_CONN_ERROR
To: Rahul Kapor rahul.kapo...@gmail.com
Cc: ccie_voice@onlinestudylist.com


Try this.

voice card x
no dspfarm
dsp services dspfarm


copy the sccp config

delete all and paste again.

It will fix your issue

Thanks
Shrini

On 3/29/2011 8:32 AM, Rahul Kapor wrote:
 Hi All,

 I did reset , reload , no sccp , sccp but did not help. DIf f conf 
 bridge from diff router can be registered with CUCM so looks like , 
 PVDM has gone into hung state.
 Is there any way to reset the PVDM ?

 th,
 Rahul

 On Tue, Mar 29, 2011 at 8:30 PM, Rogers Ochieng 
 rogersochi...@gmail.com mailto:rogersochi...@gmail.com wrote:

 Since i think this is CUCM with SRST scenario. Try this
 sccp local GigabitEthernet0/0
 sccp ccm 14.160.115.200 identifier 1 version 7.0
 sccp ccm 14.160.116.40 identifier 2 version 7.0
 sccp ccm 14.160.110.22 identifier 3 version 7.0
 sccp
 sccp ccm group 1
 associate ccm 1 priority 1
 associate ccm 2 priority 2
 associate ccm 3 priority 3
 associate profile 1 register BR1-CONF
 dspfarm profile 1 conference
  codec g711ulaw
  codec g729r8

  maximum sessions 1
  conference-join custom-cptone join
  conference-leave custom-cptone leave
  associate application SCCP
 !
 telephony-service
 sdspfarm units 1
 sdspfarm tag 1 BR1-CONF
 conference hardware
 ephone-dn 10 octo-line
 number B
 conference meet-me
 no huntstop

 On 29 March 2011 14:31, Rahul Kapor rahul.kapo...@gmail.com
 mailto:rahul.kapo...@gmail.com wrote:

 Hi All ,

 thanks for replying.
 here is my sccp config

 sccp local GigabitEthernet0/0
 sccp ccm 14.160.115.200 identifier 1 version 7.0
 sccp ccm 14.160.116.40 identifier 2 version 7.0
 sccp ccm 14.160.110.22 identifier 3 version 7.0
 sccp
 sccp ccm group 1
  associate ccm 3 priority 1
  associate profile 1 register BR1-CONF

 dspfarm profile 1 conference
  codec g711ulaw
  codec g711alaw
  codec g729ar8
  codec g729abr8
  codec g729r8
  codec g729br8
  maximum sessions 1
  conference-join custom-cptone join
  conference-leave custom-cptone leave
  associate application SCCP
 !

 I am adding Hardware conf in CUCM as BR1-CONF but status is
 unknown-unknown.
 sh sccp
 SCCP Admin State: UP
 Gateway Local Interface: GigabitEthernet0/0
 IPv4 Address: 14.160.110.35
 Port Number: 2000
 IP Precedence: 5
 User Masked Codec list: None
 Call Manager: 14.160.115.200, Port Number: 2000
 Priority: N/A, Version: 7.0, Identifier: 1
 Trustpoint: N/A
 Call Manager: 14.160.116.40, Port Number: 2000
 Priority: N/A, Version: 7.0, Identifier: 2
 Trustpoint: N/A
 Call Manager: 14.160.110.22, Port Number: 2000
 Priority: N/A, Version: 7.0, Identifier: 3
 Trustpoint: N/A


 Conferencing Oper State: ACTIVE_IN_PROGRESS - Cause Code:
 TCP_CONN_ERROR
 Active Call Manager: NONE
 TCP Link Status: CONNECT_PENDING, Profile Identifier: 1
 Reported Max Streams: 8, Reported Max OOS Streams: 0
 Supported Codec: g711ulaw, Maximum Packetization Period: 30
 Supported Codec: g711alaw, Maximum Packetization Period: 30
 Supported Codec: g729ar8, Maximum Packetization Period: 60
 Supported Codec: g729abr8, Maximum Packetization Period: 60
 Supported Codec: g729r8, Maximum Packetization Period: 60
 Supported Codec: g729br8, Maximum Packetization Period: 60
 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
 Supported Codec: rfc2833 pass-thru, Maximum Packetization
 Period: 30
 Supported Codec: inband-dtmf to rfc2833 conversion, Maximum
 Packetization Period: 30

 reloaded the router again but no help.

 thx,
 Rahul





 On Tue, Mar 29, 2011 at 12:28 PM, Rogers Ochieng
 rogersochi...@gmail.com mailto:rogersochi...@gmail.com wrote:

 Your sccp and telephony service config will help

 On 28 March 2011 18:57, Rahul Kapor
 rahul.kapo...@gmail.com mailto:rahul.kapo...@gmail.com
 wrote:


 Hi all ,

 I am trying to register my hardware conference to
 cucm/cme   but it is stuck in TCP_CONN_ERROR error
 state.

 sh sccp
 

Re: [OSL | CCIE_Voice] encapsulation frame-relay cisco?

2011-03-26 Thread bkvalent...@gmail.com
Try just encapsulation frame-relay... the default is Cisco.

Sent from my Verizon Wireless Phone

- Reply message -
From: Michael Luo hout...@gmail.com
Date: Sat, Mar 26, 2011 1:27 pm
Subject: [OSL | CCIE_Voice] encapsulation frame-relay cisco?
To: ccie_voice@onlinestudylist.com

I'm running ISO 12.4(24)T4.

When I tried to configure Auto QoS under the frame-relay DLCI, I got the
complain that the sub-interface encapsulation type was not Cisco.

But I don't see cisco as an option while configuring encapsulation.  Only
ietf and MFR are available.  Can anyone help?

R1(config-if)#encapsulation frame-relay ?
  MFR   Multilink Frame Relay bundle interface
  ietf  Use RFC1490/RFC2427 encapsulation
  cr

Thanks!
Michael
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Re: [OSL | CCIE_Voice] RE : BUG ????? QoS with MLP LFI

2011-02-24 Thread bkvalent...@gmail.com
Are you using frts or mqc based shaping?   I believe I saw this behaviour when 
trying to use mqc based shaping. 

- Reply message -
From: Ilya Rubinchik ilya.rubinc...@followmars.com
Date: Thu, Feb 24, 2011 12:16 am
Subject: [OSL | CCIE_Voice] RE : BUG ? QoS with MLP LFI
To: Friderich Claude cfrider...@netcore.lu, ShinGei Yong 
shingei.y...@gmail.com, mihal caro mihalc...@gmail.com, 
ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com

Thanks for clarification, will try it today.

But you mean, that MPL, which is very common technology in the RL, just cannot 
come-up automagically after router reboots?
It sounds strange. ☺


--
Best regards,
Ilya Rubinchik
Chief UC Engineer
Mars Solutions Ltd.

Skype: im_citius
Mob: +998 (97) 712-8456
Office: +998 (71) 290-7364

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Friderich Claude
Sent: Thursday, February 24, 2011 1:28 AM
To: ShinGei Yong; mihal caro; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] RE : BUG ? QoS with MLP LFI

Guys,

My Lab is 100 percent Cisco  guarantee


After making more tests, I can confirm that :

If I only reboot  a couple of times BR2 router, virtual access becomes up after 
each reboot

But rebooting a couple of times HQ router,  the virtual access is going to be 
down after each reboot

Conclusion:
On HQ, Just have to put the command line no frame-relay interface-dlci 202 ppp 
Virtual-Template100
and shut the Serial interface  0/1/0 and put the command  frame-relay 
interface-dlci 202 ppp Virtual-Template100 and no shut  to make it work again


Hope this can help you during the real lab …  Think it’s the easiest solution 
to make it work …. Have to keep this in mind ☺
And again, thanks to all for your remarks and suggestions

Regards

Claude.

Claude Friderich
PreSales Support
[ccvp_voice_sm]
NETCORE PSF S.A.
49 rue du Baerendall
B.P.65 L-8201 Mamer
Téléphone: 31 33 80-407
Fax: 31 33 80 8-407
GSM: 621 303 616
E-mail: cfrider...@netcore.lumailto:cfrider...@netcore.lu

From: ShinGei Yong [mailto:shingei.y...@gmail.com]
Sent: mercredi 23 février 2011 06:40
To: mihal caro; Friderich Claude; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] RE : BUG ? QoS with MLP LFI

Hi Caro,

So what you're trying to said is that,this MLP LFI issue could be potentially 
caused by the non-genuine WIC instead
of an IOS bugs?

shingei
On Wed, Feb 23, 2011 at 3:22 AM, mihal caro 
mihalc...@gmail.commailto:mihalc...@gmail.com wrote:

The interfaces

90% of the WIC-1T interfaces that are on ebay are  fake they don't come from 
Cisco but from china cloned.So maybe you are working on a fake ones. I have 
Cisco WIC-1T original and they all work after reboot.

regards

On 22 February 2011 17:19, Friderich Claude 
cfrider...@netcore.lumailto:cfrider...@netcore.lu wrote:
Hi Caro,

What do you mean by Cisco Original ?

Regards
Claude

Claude Friderich
PreSales Support
[ccvp_voice_sm]
NETCORE PSF S.A.
49 rue du Baerendall
B.P.65 L-8201 Mamer
Téléphone: 31 33 80-407
Fax: 31 33 80 8-407
GSM: 621 303 616
E-mail: cfrider...@netcore.lumailto:cfrider...@netcore.lu

From: mihal caro [mailto:mihalc...@gmail.commailto:mihalc...@gmail.com]
Sent: mardi 22 février 2011 12:00
To: Stutz, Bernhard
Cc: Friderich Claude; Ilya Rubinchik; Miron Kobelski; Roger Källberg; 
ccieid...@gmail.commailto:ccieid...@gmail.com; 
ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com; 
ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] RE : BUG ? QoS with MLP LFI

Hi there

Yes they can reboot the router at anytime you have to explain the issue to 
proctor, however he is not going to help you. I had similar issues here and I 
found out that the Serial Interface was fake. I replaced the serial module with 
Cisco orginal and it worked.



 regards.



On 22 February 2011 09:52, Stutz, Bernhard 
st...@pandacom.demailto:st...@pandacom.de wrote:
Why should your proctor reboot the router?
I woG5___
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Re: [OSL | CCIE_Voice] Slips on E1

2011-02-24 Thread bkvalent...@gmail.com
Do show network on both and paste output.

- Reply message -
From: Vinay Kumar6 vinayjaisw...@in.ibm.com
Date: Thu, Feb 24, 2011 11:34 am
Subject: [OSL | CCIE_Voice] Slips on  E1
To: ccie_voice-boun...@onlinestudylist.com, ccie_voice@onlinestudylist.com

Hi,

I have connected two routers Back-to-Back using the E1 Cross over cable, 
but I am unable to get the Multiple_Frame_Established.

I see on one of the side as slips are increasing, I tried to change the 
Clock source to Internal.


E1 0/1/0 is up.
Applique type is Channelized E1 - balanced
  No alarms detected.
  alarm-trigger is not set
  Version info Firmware: 20090113, FPGA: 20, spm_count = 0
  Framing is NO-CRC4, Line Code is HDB3, Clock Source is Internal.
  CRC Threshold is 320. Reported from firmware  is 320.
  Data in current interval (287 seconds elapsed):
 4 Line Code Violations, 1 Path Code Violations
 1 Slip Secs, 0 Fr Loss Secs, 2 Line Err Secs, 0 Degraded Mins
 3 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail 
Secs


E1 0/2/0 is up.
  Applique type is Channelized E1 - balanced
  No alarms detected.
  alarm-trigger is not set
  Version info Firmware: 20090113, FPGA: 20, spm_count = 0
  Framing is NO-CRC4, Line Code is HDB3, Clock Source is Line Primary.
  CRC Threshold is 320. Reported from firmware  is 320.
  Data in current interval (137 seconds elapsed):
 0 Line Code Violations, 0 Path Code Violations
 29 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
 29 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail 
Secs


Warm Regards, 
Vinay Kumar
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Re: [OSL | CCIE_Voice] Gatekeeper to Route Group

2011-02-20 Thread bkvalent...@gmail.com
If you have a route pattern pointed directly to a trunk, you are not able to 
add it to an RG.  Vice Versa is also true.

Brian

- Reply message -
From: Jay Woods cisco...@live.com
Date: Sun, Feb 20, 2011 5:54 pm
Subject: [OSL | CCIE_Voice] Gatekeeper to Route Group
To: ccie_voice@onlinestudylist.com


Does anyone know if I should be able to add an H.225 Gatekeeper controlled 
trunk to a Route Group? It's not showing up in my RG list in CUCM.



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Re: [OSL | CCIE_Voice] Cross Connecting PRIs issue

2011-02-19 Thread bkvalent...@gmail.com
Mark, check your cable.  It looks like you are taking physical errors... t1s 
are slipping. 

- Reply message -
From: Mark Reed marklr...@gmail.com
Date: Sat, Feb 19, 2011 8:56 am
Subject: [OSL | CCIE_Voice] Cross Connecting PRIs issue
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com

Yes network clock select is set. Show network clock at all the sites
shows clocking coming from the PSTN.



On Saturday, February 19, 2011, Nelu Cirstea nelu_cirs...@yahoo.com wrote:

 Hi Mark,

 It looks like you have sleeps on your E1s.
 You have to do network-clock-select with priority on HQ router.

 Regards,
 Nelu




-- 
Thanks,

Mark L Reed
Home: 260-637-1585
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Re: [OSL | CCIE_Voice] Cross Connecting PRIs issue

2011-02-18 Thread bkvalent...@gmail.com
You have these connected with a t1 cross over cable, correct?

I believe pinout should be 1to4, 2to5.

Brian

- Reply message -
From: Mark Reed marklr...@gmail.com
Date: Fri, Feb 18, 2011 5:27 pm
Subject: [OSL | CCIE_Voice] Cross Connecting PRIs issue
To: ccie_voice@onlinestudylist.com

Some additional information as I'm finding new things as I dig deeper.

FYI, PSTN is a 3725 with two NM-HDVs running Cisco IOS Software, 3700
Software (C3725-ADVENTERPRISEK9_IVS-M), Version 12.4(15
)T14, RELEASE SOFTWARE (fc2)

Debug isdn q921 on either side shows all TX's and no RX's.

Sh serial on the PSTN side shows my Total Output Drops incrementing rapidly
with 0 packets output.  On the other side there are no drops and output
packets is incrementing.


On Fri, Feb 18, 2011 at 4:03 PM, Mark Reed marklr...@gmail.com wrote:

 I'm not able to bring up any of my PRIs and I can't come up with a
 solution.  I've gone though every guide Cisco has and brought up many PRIs
 in production but I'm thinking maybe there is something special for setting
 up the PSTN router I'm missing?  No alarms, serial interface is up/up show
 isdn is Awaiting Establishment and TEI Assigned.  It never goes to Multiple
 Frames.  I tried multiple cables just in case.  As soon as I get back I'm
 going to try connecting two branches together to eliminate the PSTN router
 but while I had a minute I wanted to go ahead and throw this out there for
 assistance.

 This is my relevant config.  Show controller for my E1, my two T1s look
 exactly the same.

 network-clock-participate slot 1
 network-clock-participate slot 2

 controller E1 1/0
  clock source internal
  pri-group timeslots 1-3,16
  description ** E1 VOICE CONNECTION TO BR2-RTR **
 !
 controller T1 2/0
  framing esf
  clock source internal
  linecode b8zs
  pri-group timeslots 1-3,23-24
  description ** T1 VOICE CONNECTION TO HQ-RTR **
 !
 controller T1 2/1
  framing esf
  clock source internal
  linecode b8zs
  pri-group timeslots 1-3,23-24
  description ** T1 VOICE CONNECTION TO BR1-RTR **

 interface Serial1/0:15
  no ip address
  encapsulation hdlc
  isdn switch-type primary-net5
  isdn protocol-emulate network
  isdn incoming-voice voice
 !
 interface Serial2/0:23
  no ip address
  encapsulation hdlc
  isdn switch-type primary-ni
  isdn protocol-emulate network
  isdn incoming-voice voice
  no cdp enable
 !
 interface Serial2/1:23
  no ip address
  encapsulation hdlc
  isdn switch-type primary-ni
  isdn protocol-emulate network
  isdn incoming-voice voice
  no cdp enable

 voice-port 1/0:15
  translation-profile incoming block-call-into-BR2
  translation-profile outgoing display-proper-ani-into-BR2
  description ** PRI VOICE TRUNK TO BR2-RTR **
 !
 voice-port 2/0:23
  translation-profile incoming block-call-into-HQ
  translation-profile outgoing display-proper-ani-into-HQ
  description ** PRI VOICE TRUNK TO HQ-RTR **
 !
 voice-port 2/1:23
  translation-profile incoming block-call-into-BR1
  translation-profile outgoing display-proper-ani-into-BR1
  description ** PRI VOICE TRUNK TO BR1-RTR **

 All three of my controllers are doing this:

 PSTN-WAN#   sh controller e1 1/0
 E1 1/0 is up.
   Applique type is Channelized E1 - balanced
   Cablelength is Unknown
   Description: ** E1 VOICE CONNECTION TO BR2-RTR **
   No alarms detected.
   alarm-trigger is not set
   Version info Firmware: 20100222, FPGA: 13, spm_count = 0
   Framing is CRC4, Line Code is HDB3, Clock Source is Internal.
   CRC Threshold is 320. Reported from firmware  is 320.
   Data in current interval (661 seconds elapsed):
  0 Line Code Violations, 0 Path Code Violations
  661 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
  661 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail
 Secs


 Here is BR2, they all look pretty much the same

 network-clock-participate wic 2
 network-clock-select 1 E1 0/2/0

 controller E1 0/2/0
  pri-group timeslots 1-3,16

 interface Serial0/2/0:15
  no ip address
  encapsulation hdlc
  isdn switch-type primary-net5
  isdn incoming-voice voice
  isdn bchan-number-order ascending
  no isdn outgoing display-ie

 voice-port 0/2/0:15
  translation-profile incoming 4digitDNIS

 E1 0/2/0 is up.
   Applique type is Channelized E1 - balanced
   No alarms detected.
   alarm-trigger is not set
   Version info Firmware: 20090408, FPGA: 13, spm_count = 0
   Framing is CRC4, Line Code is HDB3, Clock Source is Line.
   Data in current interval (59 seconds elapsed):
  0 Line Code Violations, 0 Path Code Violations
  59 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
  59 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail
 Secs

 --
 Thanks,

 Mark L Reed




-- 
Thanks,

Mark L Reed
Home: 260-637-1585
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Re: [OSL | CCIE_Voice] High Traffic Try Again Later

2011-02-14 Thread bkvalent...@gmail.com
Search the Cisco support forms for code yellow.  It usually means the calls 
are being throttled due to high cpu/memory utilization.

Time for a reboot?

Sent from my Verizon Wireless Phone

- Reply message -
From: Tatiana Oliveira tati_olivei...@yahoo.com
Date: Mon, Feb 14, 2011 11:08 am
Subject: [OSL | CCIE_Voice] High Traffic Try Again Later
To: ccie_voice@onlinestudylist.com

I´m receiving High Traffic Try Again Later on Cisco 7965 Phones and I´m using 
the 7.1.5.1-12. Do you have any idea about how to solve that?
 


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Re: [OSL | CCIE_Voice] Gatekeeper trunk registration (SUB)

2011-01-09 Thread bkvalent...@gmail.com
Could be a replication problem.  Make sue the ccm cluster is in sync.

Sent from my Verizon Wireless Phone

- Reply message -
From: Joli-coeur Wouter jwou...@gmail.com
Date: Sun, Jan 9, 2011 5:52 pm
Subject: [OSL | CCIE_Voice] Gatekeeper trunk registration (SUB)
To: ccie_voice@onlinestudylist.com

Thanks for the info
I have a sub configured for the correct CM group and the device pool has the
correct CM group and the trunk is in the correct device pool.

I think it might be a vmware  problem or database problem?
i just want to be sure that normally both trunks should show up if they are
in the correct device group and the device group has the correct CM group.

Regards,
Joli-coeur Wouter

On Sun, Jan 9, 2011 at 11:30 PM, Miron Kobelski findko...@gmail.com wrote:

 Hi,

 It seems you don't have Sub in CM Group configured for your trunk. Remember
 to restart it later.

 regards
 kobel

   On Sun, Jan 9, 2011 at 22:39, Joli-coeur Wouter jwou...@gmail.comwrote:

   Hello,

 I am running into some problems with cucm/gatekeeper configuration.
 For some reason only the PUB trunk is being registered.

 Is the SUB trunk to register automatically or do i need to configure
 something for that to happen  ?

 interface Loopback0
  ip address 177.1.254.1 255.255.255.255
  ip pim dense-mode
  h323-gateway voip interface
  h323-gateway voip bind srcaddr 177.1.254.1
 gatekeeper
  zone local GK cisco.com 177.1.254.1
  zone prefix GK 2*
  zone prefix GK 4*
  no shutdown
 !
 I created a gatekeeper in CUCM (pub) andi created a  H.225 Trunk
 (Gatekeeper Controlled trunk)
 I added the trunk name GK-Trunk in the Device Name of GK-controlled Trunk
 That Will Use Port 1720 service

 Am i missing something ? PUB/SUB are vmware.

 Thanks for your help


 GATEKEEPER ENDPOINT REGISTRATION
 
 CallSignalAddr  Port  RASSignalAddr   Port  Zone Name Type
 Flags
 --- - --- - - 
 -
 *177.1.10.10 1720  177.1.10.10 32816 GKVOIP-GW
 *H323-ID: GK-Trunk_1
 Voice Capacity Max.=  Avail.=  Current.= 0
 177.1.254.2 1720  177.1.254.2 61105 GKH323-GW
 Voice Capacity Max.=  Avail.=  Current.= 0
 177.1.254.3 1720  177.1.254.3 61431 GKVOIP-GW
 H323-ID: CUCME
 E164-ID: 4001
 E164-ID: 4002
 E164-ID: 4003
 E164-ID: 4300
 E164-ID: 4301
 E164-ID: 4302
 Voice Capacity Max.=  Avail.=  Current.= 0
 Total number of active registrations = 3


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Re: [OSL | CCIE_Voice] CUCM Huntgroup (No Fwd)

2011-01-05 Thread bkvalent...@gmail.com
You must also set a call forward no coverage (and the associated CSS) on the 
dn that forwarded the call to the hunt pilot. 

HTH

Sent from my Verizon Wireless Phone

- Reply message -
From: Naoufal Kerboute naou...@mhdinfotech.com
Date: Sun, Jan 2, 2011 4:06 am
Subject: [OSL | CCIE_Voice] CUCM Huntgroup (No Fwd)
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com

Hi Gents,

I'm working on Hunt group configuration on CUCM. I wanted the CallFwdNoAn and 
CallFwdBusy  settings shoud be honored for the user who forwarded the call 
instead of the Hunt pilot/
Below My config:

One Hunt Group
One Hunt List
One Hunt Pilot (5000) and I've checked Use Personal preferences for CallFwdNoAn 
and CallFwdBusy

The problem is whenever I receive a call on the hunt Pilot the same ring in 
circular fashion but the CallFwdNoAn and CallFwdBusy  doesn't happened even the 
forward settings has been set on the members of the hunt group. Simply I heard 
busy tone without forward.

Any Ideas?

Regards,

Naoufal


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Re: [OSL | CCIE_Voice] mobility connect - retrieve call at desk phone

2010-12-30 Thread bkvalent...@gmail.com
I would believe at least software MTP would be required.  The call has to land 
on the cucm somehow.

- Reply message -
From: vcciev vcc...@gmail.com
Date: Thu, Dec 30, 2010 9:31 pm
Subject: [OSL | CCIE_Voice] mobility connect - retrieve call at desk phone
To: ccie_voice@onlinestudylist.com

I know the Max Wait Time for Desk Pickup under End User parameter. It is
not the cause.

If PSTN is sending inband tone to indicate call release then there is
nothing I can do, right?

I think I am looking for the setting on the CUCM/VG (suppose the PSTN suits
the requirement)

Thanks.

On Fri, Dec 31, 2010 at 12:31 AM, Miron Kobelski findko...@gmail.comwrote:

 Hi,

 On the PSTN gw check if the disconnect msg received from PSTN is properly
 relayed to CUCM (e.g. BYE is sent in case of SIP).
 I've seen this happening, when the PSTN sent inband tones to indicate call
 release.

 regards
 kobel



   On Thu, Dec 30, 2010 at 05:11, vcciev vcc...@gmail.com wrote:

   Hi,

   When the mobile phone hangs up, one is supposed to be able to press
 resume softkey at the desk phone to retrieve the call. I couldn't get it
 work, instead the call disconnects immediately. Where I should check first?
 Is MOH required for this?

   Thanks.

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Re: [OSL | CCIE_Voice] IPCC trouble

2010-12-26 Thread bkvalent...@gmail.com
Sounds like it could be a replication issue.  Try disabling the subscriber to 
see if it works.

Sent from my Verizon Wireless Phone

- Reply message -
From: Mritunjay Kumar mjs...@gmail.com
Date: Sun, Dec 26, 2010 2:49 am
Subject: [OSL | CCIE_Voice] IPCC trouble
To: bkvalent...@gmail.com bkvalent...@gmail.com, Shrini 
linuxbos...@gmail.com, ccie_voice@onlinestudylist.com

Hi All,
thanks for responding.
Did DNA , its hitting right CTI Route Point
CCX is configured with g711.

-Mritunjay

On Sun, Dec 26, 2010 at 1:46 AM, bkvalent...@gmail.com 
bkvalent...@gmail.com wrote:

 Is uccx configured for g711?  In the uccx config area.

 Sent from my Verizon Wireless Phone

 - Reply message -
 From: Shrini linuxbos...@gmail.com
 Date: Sat, Dec 25, 2010 1:46 pm

 Subject: [OSL | CCIE_Voice] IPCC trouble
 To: 'Mritunjay Kumar' mjs...@gmail.com, 
 ccie_voice@onlinestudylist.com


 Kumar,

 Use DNA to see whets going on. Try calling directly the CTI port numbers.

 -S

  _

 From: ccie_voice-boun...@onlinestudylist.com
 [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mritunjay
 Kumar
 Sent: Saturday, December 25, 2010 8:46 AM
 To: ccie_voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice] IPCC trouble


 hi All ,

 Wish U ALL  MARRY CHRISTMAS !

 Getting issue in invoking IPCC's  AA

 Created CTI group , application and JTAPI trigger.  all phone , CTI PORT ,
 Trigger DN  are in null partition.
 when trigger number is called , BUSY message is seen on phone display and
 AA
 is not being invoked.
 All are in same Region ie g711.
 same server  was workig earlier.
 i deleted  respective config and tried to confg . i rebooted the cucm and
 IPCC server but no luck :(
 all service are up and running. CTI PORT and ROUTE point are registered.

 m missing something ?

 Regards,
 MJ





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Re: [OSL | CCIE_Voice] IPCC trouble

2010-12-25 Thread bkvalent...@gmail.com
Is uccx configured for g711?  In the uccx config area.

Sent from my Verizon Wireless Phone

- Reply message -
From: Shrini linuxbos...@gmail.com
Date: Sat, Dec 25, 2010 1:46 pm
Subject: [OSL | CCIE_Voice] IPCC trouble
To: 'Mritunjay Kumar' mjs...@gmail.com, ccie_voice@onlinestudylist.com

Kumar,
 
Use DNA to see whets going on. Try calling directly the CTI port numbers.
 
-S

  _  

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mritunjay Kumar
Sent: Saturday, December 25, 2010 8:46 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] IPCC trouble


hi All ,

Wish U ALL  MARRY CHRISTMAS !

Getting issue in invoking IPCC's  AA

Created CTI group , application and JTAPI trigger.  all phone , CTI PORT ,
Trigger DN  are in null partition.
when trigger number is called , BUSY message is seen on phone display and AA
is not being invoked.
All are in same Region ie g711.
same server  was workig earlier.
i deleted  respective config and tried to confg . i rebooted the cucm and
IPCC server but no luck :(
all service are up and running. CTI PORT and ROUTE point are registered.

m missing something ?

Regards,
MJ


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Re: [OSL | CCIE_Voice] Introduction

2010-12-23 Thread bkvalent...@gmail.com
I knew this guy was lying... John Smith from UK with broken English. 

Sent from my Verizon Wireless Phone

- Reply message -
From: John Smith john.smith.cc...@gmail.com
Date: Wed, Dec 22, 2010 10:46 am
Subject: [OSL | CCIE_Voice] Introduction
To: ccie_voice@onlinestudylist.com

HI Team,

I am john smith located in UK.

I am new to this forum. SO hello to all you guys.

I have started preparation for ccie voice since last sixth month.

Guys i would need your help and as far as i know this group is very helpful.

So many of my friends are member of this group.

I hope my query will be answered and i look to learn from this forum as
well.

Thanks once again guys.

Rgds,
John Smith
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Re: [OSL | CCIE_Voice] Workbook 2 LAB 1

2010-12-11 Thread bkvalent...@gmail.com
B8ZS?  ESF? All 24 channels? Top down vs. Bottom up?  The PSTN gateway at PL is 
set to channels 1-3...  how about sending config?

Sent from my Verizon Wireless Phone

- Reply message -
From: Cristobal Priego cristobalpri...@gmail.com
Date: Sat, Dec 11, 2010 12:36 pm
Subject: [OSL | CCIE_Voice] Workbook 2 LAB 1
To: ccie_voice@onlinestudylist.com

hello All,

I'm working on Lab 1 from Woorkbook 2

I've been having problems with the HQ PRI it doesn't seems to be working
properly and i think the problem is on the PSTN router

i don't receive incoming calls at all, i tested it with debug isdn q931,
when i call from my pstn phone to a HQ phone i get a fast busy right away
and the call never hits the pri on the HQ router


i have the HQ PRI configured as MGCP. i have multiple frame established on
the show isdn status
however when i want to make a call from a HQ phone, i get a fast busy after
4 secs, if i debug the PRI i see that i'm sending the proper digits and it
tries 3 times before it fails


what should i do to fix this problem ?

if i reload the HQ router the PRI starts flapping

if i enable the debug isdn q921 i only send info and never receive anything
from the pstn


any help is greatly appreciated

thanks
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Re: [OSL | CCIE_Voice] Deleting and Creating VM Box on Unity 5.0

2010-12-10 Thread bkvalent...@gmail.com
COBRAS can help.  www.ciscounitytools.com.

- Reply message -
From: Mark Padilla clane...@hotmail.com
Date: Fri, Dec 10, 2010 1:20 pm
Subject: [OSL | CCIE_Voice] Deleting and Creating VM Box on Unity 5.0
To: ccie_voice@onlinestudylist.com


I was wondering if anyone knew of a way to delete and create and or move a 
users VM Box with saving the users VM and Greeting. Please let me know if 
anyone has a better way of moving this information other than deleting and 
having the user recreate evertyhing. 
 
 From: ccie_voice-requ...@onlinestudylist.com
 Subject: CCIE_Voice Digest, Vol 58, Issue 32
 To: ccie_voice@onlinestudylist.com
 Date: Fri, 10 Dec 2010 12:00:03 -0500
 
 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. CFUR and SRST - is there a bug? (Tam Nhu)
 2. Re: CFUR display For and By (Tam Nhu)
 3. UCCX and CCM Integration Issue (shafqut hamid)
 4. Re: UCCX and CCM Integration Issue (Prashant Patel)
 
 
 --
 
 Message: 1
 Date: Fri, 10 Dec 2010 09:11:47 -0600
 From: Tam Nhu tamnhu...@gmail.com
 To: ccie_voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice] CFUR and SRST - is there a bug?
 Message-ID:
 aanlktikxyosqbxlbbg+s0s-tb=ftfecaqcjq8clyw...@mail.gmail.com
 Content-Type: text/plain; charset=iso-8859-1
 
 Hi Experts,
 
 Not sure if this subject has been posted before, but I could not find it in
 this forum. Here is the problem I just discovered a couple days back and
 can be duplicated in many labs.
 
 I have configured CFUR to display the correct full E164 digits in FOR and BY
 field; everything configured on CUCM, not the router. But when I finished
 my testing in SRST, and things are back to normal CUCM mode; I tested it
 again by shutdown the serial interface again to force it go back to SRST
 mode. Now, the FOR and BY field only display 4-digit, not full E164 as on
 the first test.
 
 I rebooted the router and went to SRST mode again, then everything is
 working with FOR and BY as expected. I then performed the same step above,
 no shut to back to CUCM mode, then shut to go in SRST mode, now I duplicated
 the same issue with FOR and BY display only 4-digit.
 
 I duplicated this many times on my own lab and IPX vRack lab as well. It
 does not matter the remote gateway is MGCP or H323 (I did test both with the
 same result). So looks like it is a bug to me and required to reboot the
 router every time after it went to SRST. I didn't know about this issue
 since we always have a fresh vRack session, and I only tested SRST once
 time, but not 2nd time on the same vrack session.
 
 Has anyone run into this before, and what is the bug ID if you have it? Is
 there another work-round to fix this?
 
 Thanks,
 Tam.
 -- next part --
 An HTML attachment was scrubbed...
 URL: /archives/ccie_voice/attachments/20101210/d191a755/attachment-0001.html
 
 --
 
 Message: 2
 Date: Fri, 10 Dec 2010 09:17:44 -0600
 From: Tam Nhu tamnhu...@gmail.com
 To: ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] CFUR display For and By
 Message-ID:
 aanlktinwzkx671mg8j+spd6j1rrjo-ezbxml+=ll6...@mail.gmail.com
 Content-Type: text/plain; charset=iso-8859-1
 
 Hi study2b,
 
 You can look at this thread and make it works.
 
 http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg19418.html
 
 
 I've just posted a thread related to this issue, so if you make it works and
 run into the same problem, please let me know.
 
 Thanks,
 TN.
 -- next part --
 An HTML attachment was scrubbed...
 URL: /archives/ccie_voice/attachments/20101210/b529c717/attachment-0001.html
 
 --
 
 Message: 3
 Date: Fri, 10 Dec 2010 21:27:39 +0500
 From: shafqut hamid shafqut.ha...@gmail.com
 To: ccie_voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice] UCCX and CCM Integration Issue
 Message-ID:
 aanlktik_c=nmwpmmdfd0moryvf9mva7p1vkajh6wm...@mail.gmail.com
 Content-Type: text/plain; charset=iso-8859-1
 
 Hi,
 
 I have installed CCM 7.0 and UCCX 7.0 with all necessary configuration in
 VMWare but i am unable to test very first auto attendant script. Whenever i
 dial number for CTI route point i receive busy tone, although i can see both
 CTI Route Point and CTI ports registered on CCM. Also on UCCX Control Center
 operational state is in-service. I already checked the Partitions and
 Calling Search spa___
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Re: [OSL | CCIE_Voice] CIPC hangs at Registering when connecting to lab CUCM

2010-11-30 Thread bkvalent...@gmail.com
Did you build the cipc as the cipc device type?  You didn't build in cucm as a 
7960/1 phone, as you did with ip blue, did you?

Just a thought...

Brian

Sent from my Verizon Wireless Phone

- Reply message -
From: ccieid1ot ccieid...@gmail.com
Date: Tue, Nov 30, 2010 1:24 pm
Subject: [OSL | CCIE_Voice] CIPC hangs at Registering when connecting to lab 
CUCM
To: Ginther, Scott [NTK] scott.gint...@sprint.com
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com


Go into your settings unlock, then Device and erased your CCM stuff.

On Tue, Nov 30, 2010 at 11:58 AM, Ginther, Scott [NTK] 
scott.gint...@sprint.com wrote:

  Cristobal,



 That’s not a silly question at all.



 I have CIPC pointed at the CUCM Pub (10.10.210.10). The same TFTP server
 that I have my IP Blue client pointed at. The IP Blue client is registered
 and working. The CIPC client just says “registering”.



 Both the IP Blue and CIPC phones are configured in CUCM as SCCP with the
 correct MAC addresses. Nothing fancy.





 *From:* Cristobal Priego [mailto:cristobalpri...@gmail.com]
 *Sent:* Tuesday, November 30, 2010 8:54 AM
 *To:* Ginther, Scott [NTK]
 *Cc:* ccie_voice@onlinestudylist.com
 *Subject:* Re: [OSL | CCIE_Voice] CIPC hangs at Registering when
 connecting to lab CUCM



 A silly question



 Dd you change the tftp settings of your Ipc to point to the proper server?

 Right click on the ipc, preferences, tftp

 Sent from my iPhone


 On Nov 30, 2010, at 6:45, Ginther, Scott [NTK] scott.gint...@sprint.com
 wrote:

  I’m having trouble connecting CIPC 7.0.5 to the virtual lab CUCM. CIPC
 hangs at “Registering”, and does not give an error.



 If I disconnect my virtual lab VPN connection and connect to my corporate
 network, CIPC will register to our corporate CUCM cluster. Am I missing
 something?



 I can't register my CIPC phone to the lab CUCM via SCCP or SIP. I could use
 a little help getting it configured correctly for the virtual lab.



 Thanks,



 -Scott-


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Re: [OSL | CCIE_Voice] IP RSVP

2010-11-14 Thread bkvalent...@gmail.com
Heres a hint.  Read the qos srnd.  Some WAN qos mechanisms are incompatible 
with ip rsvp.  If I recall, the point of this lab is to teach you this lesson 
at the cost of your time and frustration.

- Reply message -
From: Stern, Larry larry.st...@nuvt.com
Date: Sun, Nov 14, 2010 3:54 pm
Subject: [OSL | CCIE_Voice] IP RSVP
To: ccie_voice@onlinestudylist.com


Hi all

I going crazy trying to figure out this issue. I use a Hardware VPN to connect 
to PL
with 7960 and 7961 phones. I am doing LAB 10A part 10.1 RSVP call agent.

I have BR1 to HQ set to Manadtory under the Locations in CUCM and have all the 
DP settings and MRG and MRGL's as per the PG guide. My HQ and BR1 routers have 
ip rsvp bandwidth set to 80K on interface serial 0/0/1.0.1 on both Routers and 
my software MTP's are registered. But when I make one call between HQ and BR1 
or vice versa, I get not enough bandwidth. My max sessions software is set for 
4 on my router dspfarm MTP profile on both sides. If I remove the mandatory 
RSVP setting on the locations in CUCM, I can make the call no problem. Has 
anyone run into a similar issue?

After getting frustrated I dumped the final config's into the routers and have 
the same issue.

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Re: [OSL | CCIE_Voice] UCCX get digit string

2010-09-24 Thread bkvalent...@gmail.com
Exactly... use the music files built into uccx.

- Reply message -
From: Stutz, Bernhard st...@pandacom.de
Date: Fri, Sep 24, 2010 6:39 am
Subject: AW: [OSL | CCIE_Voice] UCCX get digit string
To: Tanner Ezell tanner.ez...@gmail.com, bkvalent...@gmail.com
Cc: ccie_voice@onlinestudylist.com


Hi,

This solution works as desired. Thanks :-)
You don't need to waste your time to copy across the music-on-hold.au file and 
convert it to wav.
You can use this wave file for example: ICM\ICMMusic_30.wav
There is nothing mentioned to use the music on hold music ;-)

Btw, converting a .au file to .wav is not so easy and it doesn't work with the 
windows audio recorder.
Just renaming .au to .wav doesn't work either and callmanager can't convert the 
.au file to .wav like with mp3 files.
You will need a converter and this will probably not be available for you at 
the lab.

Cheers,
Bernhard

-Ursprüngliche Nachricht-
Von: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] Im Auftrag von Stutz, Bernhard
Gesendet: Freitag, 24. September 2010 10:27
An: Tanner Ezell; bkvalent...@gmail.com
Cc: ccie_voice@onlinestudylist.com
Betreff: Re: [OSL | CCIE_Voice] UCCX get digit string

Hi,

Thanks for your replies. 
This solution sounds reasonable. You probably need to copy across the 
music-on-hold.au file on the routers and convert it to wav with the windows 
audio recorder if this works. I will try this.

Cheers,
Bernhard

-Ursprüngliche Nachricht-
Von: Tanner Ezell [mailto:tanner.ez...@gmail.com] 
Gesendet: Donnerstag, 23. September 2010 21:56
An: bkvalent...@gmail.com
Cc: Stutz, Bernhard; ccie_voice@onlinestudylist.com
Betreff: Re: [OSL | CCIE_Voice] UCCX get digit string

They give you a prompt to play music with?

Weird.

On Thu, Sep 23, 2010 at 12:44 PM, bkvalent...@gmail.com
bkvalent...@gmail.com wrote:
 No.  This is easier than you think.

 Use 2 menu steps.  The first is your prompt with the 0 out to AA.  Make the
 timeout 0 and the retries 0.

 The next step is a second menu.  Make it the same, but for the audio prompt,
 use the music file.  Still has 0 out to AA.  Still has 0 timeout and 0
 retries.

 HTH

 Brian

 - Reply message -
 From: Tanner Ezell tanner.ez...@gmail.com
 Date: Thu, Sep 23, 2010 3:24 pm
 Subject: [OSL | CCIE_Voice] UCCX get digit string
 To: Stutz, Bernhard st...@pandacom.de
 Cc: ccie_voice@onlinestudylist.com


 Actually,

 I take that back. Putting the call on hold should prevent Get Digit
 String from working as desired. Ah well.

 On Thu, Sep 23, 2010 at 12:22 PM, Tanner Ezell tanner.ez...@gmail.com
 wrote:
 Unless there is some context I am missing this is *not possible* with
 UCCX.

 I suspect they expect you to play a menu step that is interruptible.

 Having a user enter digits during MoH is not possible.

 But..
 now that I think about it, you could do the following:

 Queue:
 Call Hold
 Get Digit String( P[], timeout=30, digits=1 )
  - Success
   Call Unhold
   Redirect
 Call Unhold
 Prompt (Please continue to wait)
 Go to Queue

 Hmm. Hot damn, sometimes I amaze myself..

 On Thu, Sep 23, 2010 at 12:13 PM, Stutz, Bernhard st...@pandacom.de
 wrote:
 Hi,

 I am stucked with a question on UCCX:

 snip
 ... At any time during the prompt or whilst music is being played the
 caller
 can press 0 to be routed to the UC AutoAttendant.
 /snip

 When i try to achieve this with GetDigitString the routine will wait
 for
 an input and not going further the script unless its timed out. when
 timed
 out i am not in that routine GetDigitString anymore to catch the caller
 input.

 How can this be solved? I see nothing what i can use at the select
 resource Routine.

 Anyone solved this? Any help on this is much appreciated.

 cheers,
 Bernhard
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 visit www.ipexpert.com



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



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Re: [OSL | CCIE_Voice] Unity Connection Greeting

2010-09-24 Thread bkvalent...@gmail.com
Remove the display name.


- Reply message -
From: Mark Holloway m...@markholloway.com
Date: Fri, Sep 24, 2010 5:53 pm
Subject: [OSL | CCIE_Voice] Unity Connection Greeting
To: osl osl ccie_voice@onlinestudylist.com

Is there an option so instead of the VM greeting being Sorry, Jon Doe is not 
available it says Sorry, extension 5001 is not available.  Basically, I want 
it to say the extension instead of the name.

Thanks,
Mark

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Re: [OSL | CCIE_Voice] One button login - Phone service

2010-09-15 Thread bkvalent...@gmail.com
Wondering if replication is down between pub and sub.  Ccx could be hitting 
sub, which maybe doesn't have the phone associated to the rmcm user... just an 
idea.

- Reply message -
From: Tanner Ezell tanner.ez...@gmail.com
Date: Wed, Sep 15, 2010 6:02 pm
Subject: [OSL | CCIE_Voice] One button login - Phone service
To: KatGuru gkr2...@yahoo.com, ccie_voice@onlinestudylist.com

Give that a shot and let us know. Beyond that you may have a broken system.

On Wed, Sep 15, 2010 at 2:47 PM, KatGuru gkr2...@yahoo.com wrote:

 No i didn't.

 --- On *Thu, 9/16/10, Tanner Ezell tanner.ez...@gmail.com* wrote:


 From: Tanner Ezell tanner.ez...@gmail.com
 Subject: Re: [OSL | CCIE_Voice] One button login - Phone service
 To: KatGuru gkr2...@yahoo.com
 Date: Thursday, September 16, 2010, 5:39 AM


 Curious.

 Have you tried logging in via CAD to verify at least that works?

 On Wed, Sep 15, 2010 at 2:18 PM, KatGuru 
 gkr2...@yahoo.comhttp://mc/compose?to=gkr2...@yahoo.com
  wrote:

 Please find the attached. I'm running out of ideas and no clue.


 Thanks.

 --- On *Thu, 9/16/10, Tanner Ezell 
 tanner.ez...@gmail.comhttp://mc/compose?to=tanner.ez...@gmail.com
 * wrote:


 From: Tanner Ezell 
 tanner.ez...@gmail.comhttp://mc/compose?to=tanner.ez...@gmail.com
 
 Subject: Re: [OSL | CCIE_Voice] One button login - Phone service
 To: KatGuru gkr2...@yahoo.com http://mc/compose?to=gkr2...@yahoo.com
 Cc: Amy Ryan ar...@ipexpert.comhttp://mc/compose?to=ar...@ipexpert.com,
 ccie_voice@onlinestudylist.comhttp://mc/compose?to=ccie_vo...@onlinestudylist.com
 Date: Thursday, September 16, 2010, 4:47 AM


 Just to verify,

 Have you set the Ext parameter to match the IPCC Extension field off the
 end user profile?

 On Wed, Sep 15, 2010 at 1:39 PM, KatGuru 
 gkr2...@yahoo.comhttp://mc/compose?to=gkr2...@yahoo.com
  wrote:

 Yes I did. I reset the end user password few times. After that I reset
 phone, reset tftp but still the same. This is weird.


 --- On *Thu, 9/16/10, Amy Ryan 
 ar...@ipexpert.comhttp://mc/compose?to=ar...@ipexpert.com
 * wrote:


 From: Amy Ryan ar...@ipexpert.comhttp://mc/compose?to=ar...@ipexpert.com
 
 Subject: Re: [OSL | CCIE_Voice] One button login - Phone service
 To: KatGuru gkr2...@yahoo.com http://mc/compose?to=gkr2...@yahoo.com,
 Tanner Ezell 
 tanner.ez...@gmail.comhttp://mc/compose?to=tanner.ez...@gmail.com
 

 Cc: 
 ccie_voice@onlinestudylist.comhttp://mc/compose?to=ccie_vo...@onlinestudylist.com
 Date: Thursday, September 16, 2010, 4:25 AM


 When you assigned the service to the phone, did you populate the correct
 credentials to match what was set in the End User?



 ---
 Amy Ryan – CCIE #24677 (Voice)
 Technical Instructor - IPexpert, Inc.
 Mailto: *ar...@ipexpert.com
 *Telephone: +1.810.326.1444
 Live Assistance, Please visit: www.ipexpert.com/chat *
 http://www.ipexpert.com/chat*
 eFax: +1.810.454.0130

 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand,
 Audio Tools, Online Hardware Rental and Classroom Training for the Cisco
 CCIE (RS, Voice, Wireless, Security  Service Provider) certification(s)
 with training locations throughout the United States, Europe, South Asia and
 Australia. Be sure to visit our online communities at
 www.ipexpert.com/communities *http://www.ipexpert.com/communities*  and
 our public website at www.ipexpert.com *http://www.ipexpert.com/*



 --
 *From: *KatGuru gkr2...@yahoo.com
 *Date: *Wed, 15 Sep 2010 13:18:27 -0700 (PDT)
 *To: *Tanner Ezell tanner.ez...@gmail.com
 *Cc: *ccie_voice@onlinestudylist.com, Amy Ryan ar...@ipexpert.com
 *Subject: *Re: [OSL | CCIE_Voice] One button login - Phone service

 No luck.  I changed all the parameters with proper case. Actually i deleted
 the one  i had and recreate the new one. But this time it didn't even gave
 me an option to input the agent credentials after pressing the service
 button -- One button login. It gives me the below response

 Either the agent ID or the password you entered is invalid. But i'm
 pretty sure the ID and password is correct.

 Do you guys think is it something to do with my IPC?  I don't have the
 access to physical phone today otherwise i should have been tested.

 Thanks.

 --- On *Thu, 9/16/10, Tanner Ezell tanner.ez...@gmail.com* wrote:


 From: Tanner Ezell tanner.ez...@gmail.com
 Subject: Re: [OSL | CCIE_Voice] One button login - Phone service
 To: KatGuru gkr2...@yahoo.com
 Cc: ccie_voice@onlinestudylist.com, Amy Ryan ar...@ipexpert.com
 Date: Thursday, September 16, 2010, 3:39 AM

 Right off the bat I see you have you have the parameters as Extn, ID, pwd.
 Extn should be Ext, pwd should be Pwd (case matters).
 Change those and let me know.

 On Wed, Sep 15, 2010 at 12:09 PM, KatGuru gkr2...@yahoo.com 
 /mc/compose?to=gkr2...@yahoo.com  wrote:

 Please find the attached.

 Thanks.

 --- On *Thu, 9/16/10, Tanner Ezell tanner.ez...@gmail.com 
 /mc/compose?to=tanner.ez...@gmail.com * wrote:


 From: Tanner Ezell tanner.ez...@gmail.com 
 

Re: [OSL | CCIE_Voice] Hide a particular user in theCorpora teDirectory

2010-09-13 Thread bkvalent...@gmail.com
Disable the empty search.  It's a call manager service parameter. 

Sent from my Verizon Wireless Phone

- Reply message -
From: Tam Nhu tamnhu...@gmail.com
Date: Mon, Sep 13, 2010 10:26 am
Subject: [OSL | CCIE_Voice] Hide a particular user in theCorporateDirectory
To: bkvalent...@gmail.com bkvalent...@gmail.com
Cc: cciefo...@hotmail.com, Hough, Earl earl.ho...@pcmall.com, 
ccie_voice-boun...@onlinestudylist.com, Randall Saborio 
ill2...@gmail.com, ccie_voice@onlinestudylist.com


I have tried all the workaround and suggestions, but none of them appear to
be worked or fulfill the requirement of 'no display in the corporate
directory'.  Yes, you can put ~ or some special char in the front of
First/Last (in both AD or DC directory) to hide them from the search by
name, but it still appears in the search at the bottom as ~admin, ~crs.
Again, the requirement is NO DISPLAY.

Thanks for all the input and suggestion...I give up this one and except no
credit from customer then:)

TN.
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Re: [OSL | CCIE_Voice] ccie voice

2010-09-02 Thread bkvalent...@gmail.com
There is no way anyone can answer this question without violating the Non 
Disclosure Agreement.  



- Reply message -
From: Afzal Bhutta azhar.bhu...@gmail.com
Date: Thu, Sep 2, 2010 9:17 am
Subject: [OSL | CCIE_Voice] ccie voice
To: ccie_voice@onlinestudylist.com

Hi all,
Does any body know the voice lab also testing VPIM and extension mobility.I
heard there is new scenario.Also there is some changes in the previous lab
they are not testing C-barge in SRST any more.*Is that true?
So if there is some changes in the lab what new features they are testing.
Hopefully those who pass will shade something on the topic.
Thanks,

*
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Re: [OSL | CCIE_Voice] [OSL | CCIE_RS] Proctorlabs not resolving

2010-08-29 Thread bkvalent...@gmail.com
Tyson, 

Sorry, bud. That doesn't cut it.  It's completely absurd to think that people 
can work on labs that they can't load.  Even if I wanted to work a lab 
completely from scratch without any base config, I can't configure the PSTN-wan 
router.  So, its a complete waste of time.  You need to address an ongoing 
availability problem with the PL website.  Otherwise, you need to come up with 
a new way to allow us to load any given lab we wish to load (for instance via a 
menu once we telnet to the termserv).  I know that I found this issue maddening 
when it impacted me.  It's painful to set aside a day to study and then not 
have access for a few hours to even begin.   As CCIE candidates and PL 
customers, we already sacrifice a lot of personal time to dedicate ourselves to 
this journey.  Please don't waste our time. Telling me that you expect me to 
work on a lab that I cannot load is like slashing my tires and telling me to 
just drive anyway.  Could I?  I suppose I could, but not very far and it will 
likely be quicker to wait until the tires are fixed before I begin.  Speaking 
from personal experience, I know that's how I felt. 

Brian


- Reply message -
From: Tyson Scott tsc...@ipexpert.com
Date: Sun, Aug 29, 2010 6:03 pm
Subject: [OSL | CCIE_Voice] [OSL | CCIE_RS] Proctorlabs not resolving
To: 'marc abel' marca...@gmail.com, 'Jason Maynard' 
jason.mayn...@rogers.com
Cc: ccie...@onlinestudylist.com, ccie_voice@onlinestudylist.com, 'OSL CCIE 
Security Lab  Exam' ccie_secur...@onlinestudylist.com


Now you guys know there really there is no reason you have to login to the
web interface except to load/revert/save configurations?  For access to the
routers/switches/servers simply connecting to them using the instructions
Marko has given on our blog will work well enough and can be adapted for
voice customers.  In the future if PL is down there is no reason that  it
should stop you from beginning and working on your session.  Just connect in
and start your work on what you can and we will always eventually fix any
problems on our end.

 

Regards,

 

Tyson Scott - CCIE #13513 RS, Security, and SP

Managing Partner / Sr. Instructor - IPexpert, Inc.

Mailto:  mailto:tsc...@ipexpert.com tsc...@ipexpert.com

Telephone: +1.810.326.1444, ext. 208

Live Assistance, Please visit:  http://www.ipexpert.com/chat
www.ipexpert.com/chat

eFax: +1.810.454.0130

 

IPexpert is a premier provider of Self-Study Workbooks, Video on Demand,
Audio Tools, Online Hardware Rental and Classroom Training for the Cisco
CCIE (RS, Voice, Security  Service Provider) certification(s) with
training locations throughout the United States, Europe, South Asia and
Australia. Be sure to visit our online communities at
http://www.ipexpert.com/communities www.ipexpert.com/communities and our
public website at  http://www.ipexpert.com/ www.ipexpert.com

 

From: ccie_rs-boun...@onlinestudylist.com
[mailto:ccie_rs-boun...@onlinestudylist.com] On Behalf Of marc abel
Sent: Sunday, August 29, 2010 5:28 PM
To: Jason Maynard
Cc: ccie...@onlinestudylist.com
Subject: Re: [OSL | CCIE_RS] Proctorlabs not resolving

 

I was finally able to get in just now.

On Sun, Aug 29, 2010 at 3:39 PM, Jason Maynard jason.mayn...@rogers.com
wrote:

That is correct --- I was also doing an nslookup but I had a  typo
protolabs.com --- here nor there it is down

 

And I am missing my session again

 

 

 

From: marc abel [mailto:marca...@gmail.com] 
Sent: August-29-10 4:38 PM
To: Jason Maynard
Cc: ccie...@onlinestudylist.com
Subject: Re: [OSL | CCIE_RS] Proctorlabs not resolving

 

Actually, I believe it is resolving, I can do an NSlookup and then ping the
returned IP address. It is just the webserver throws an error after a long
period of attempting to connect.

The error is:


Server Error


The server encountered an internal error and was unable to complete your
request.

JRun closed connection.

 

On Sun, Aug 29, 2010 at 3:37 PM, marc abel marca...@gmail.com wrote:

Same here.

On Sun, Aug 29, 2010 at 3:14 PM, Jason Maynard jason.mayn...@rogers.com
wrote:

Having problems getting to protorlabs.com

 

This happened yesterday during my lab and again today.

 

Looks like protorlabs cannot be resolved. All other sites including ipexpert
are working.

 

Anyone else??

 

I cannot contact support because I cannot get to the site.

 

This is frustrating as heck as I will be taking the lab in a few weeks and
cannot afford missing prep time.

 

Regards

 

 

___
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visit www.ipexpert.com

 

 

 

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Re: [OSL | CCIE_Voice] [OSL | CCIE_RS] Proctorlabs not resolving

2010-08-29 Thread bkvalent...@gmail.com
Maybe I am in the whiny minority... who feels that the thousands of dollars 
I've spent with IPExpert/PL gives me a sense of entitlement to a certain 
quality.  It's a tough sell that your work around solution is to copy and paste 
when the voice labs mostly contain 5+ VMWare servers that can't be copied and 
pasted.  There are more servers in the voice pods than all the other devices. 

Tyson, i completely understand that there will be snags in the technology 
occasionally.  It happens. I was perfectly silent on this issue given that when 
I got bit on a weekend, I was issued a replacement voucher.  It didn't replace 
my time, but it was the right thing to do.  

I just wanted to make sure this recurring issue is being properly addressed.  I 
may have misunderstood your initial email.  I thought you had resigned yourself 
to the fact that this is just the way it is going to be.  Please say it isn't 
so.

Brian

- Reply message -
From: Tyson Scott tsc...@ipexpert.com
Date: Sun, Aug 29, 2010 8:23 pm
Subject: [OSL | CCIE_Voice] [OSL | CCIE_RS] Proctorlabs not resolving
To: 'marc abel' marca...@gmail.com, bkvalent...@gmail.com
Cc: 'Jason Maynard' jason.mayn...@rogers.com, 
ccie...@onlinestudylist.com, ccie_voice@onlinestudylist.com, 'OSL CCIE 
Security Lab Exam' ccie_secur...@onlinestudylist.com


Marc,

 

You guys do realize this is a lab environment right?  I read this and feel
that you want enterprise class service in an ever changing environment.
This is somewhat disheartening to me as well.  Do I want problems to occur,
of course not.  But I read these posts and feel a bunch of ingratitude to
the way we have changed the training environment.   I remember studying for
my RS exam, paying 50+ dollars for a session with no remote power, no save
load function, no scripts to wipe out previous students configuration.  I
would be lucky to get a rack that actually supported the topology of the
workbook I was using without having to make changes to the startup
configuration to account for differences in interfaces, routers, etc.  And I
would usually have to book a session three weeks in advance to even have a
chance to get on a rack.

 

It surprises me that I reach out to give you guys ideas to better utilize
your time when a problem occurs; giving simple suggestions for things you
can do to utilize your time in the event there is a problems.  And get
complaints back to my emails.  Being a CCIE is having the ability to think
outside the box.  

 

Getting responses back like this for providing constructive ideas simply
reminds me again why I don't want to waste my time writing new grading
scripts and other new advanced features.  They simply become something else
for a minority  of our students to complain about.

 

Regards,

 

Tyson Scott - CCIE #13513 RS, Security, and SP

Managing Partner / Sr. Instructor - IPexpert, Inc.

Mailto:  mailto:tsc...@ipexpert.com tsc...@ipexpert.com

Telephone: +1.810.326.1444, ext. 208

Live Assistance, Please visit:  http://www.ipexpert.com/chat
www.ipexpert.com/chat

eFax: +1.810.454.0130

 

IPexpert is a premier provider of Self-Study Workbooks, Video on Demand,
Audio Tools, Online Hardware Rental and Classroom Training for the Cisco
CCIE (RS, Voice, Security  Service Provider) certification(s) with
training locations throughout the United States, Europe, South Asia and
Australia. Be sure to visit our online communities at
http://www.ipexpert.com/communities www.ipexpert.com/communities and our
public website at  http://www.ipexpert.com/ www.ipexpert.com

 

From: marc abel [mailto:marca...@gmail.com] 
Sent: Sunday, August 29, 2010 7:56 PM
To: bkvalent...@gmail.com
Cc: Tyson Scott; Jason Maynard; ccie...@onlinestudylist.com;
ccie_voice@onlinestudylist.com; OSL CCIE Security Lab Exam
Subject: Re: [OSL | CCIE_Voice] [OSL | CCIE_RS] Proctorlabs not resolving

 

Brian, I would totally agree. I was trying to continue a lab from last
night, so I didn't even have a copy that I could paste in. Then when I
finally did get logged in an hour and a half after my session was to start I
find that the script didn't save my configs from last night, despite
starting the script 30 minutes before the session ended. This follows Friday
night where The script processor wouldn't load other saved configs (my
session was refunded). 
Tyson your comment makes it sound like you believe we should be ok with this
kind of performance. This is disheartening because I know that you are
managing a complex system and there are going to be glitches. I am willing
to cut you some slack on that front if you acknowledge that this is not
acceptable performance. I will say that the desire not to waste my study
time troubleshooting things not on the blueprint is why I use proctor labs
over dynamips. When this kind of stuff happens it can totally take the wind
out of my study motivation.


I would say that my desire not to battle technical issues outside of the
study related ones is why I use Proctor labs

Re: [OSL | CCIE_Voice] PHONE REGISTplRATION

2010-08-21 Thread bkvalent...@gmail.com
That's the dhcp pool for the remote phones.  Are you doing a hardware vpn?   If 
so, create a similar scope for your local phones.  Otherwise, manually assign 
ips and tftp.

Brian

- Reply message -
From: Pithog Oil pithog...@yahoo.com
Date: Sat, Aug 21, 2010 7:40 pm
Subject: [OSL | CCIE_Voice] PHONE REGISTplRATION
To: ccie_voice@onlinestudylist.com

I am new on proctor labs 

Please  i need some  one that has worked on this lab to help me out 

i have done all that is needed as per config to make my phones register but to 
no avail.

please how do i get the phones local to me registered.

HERE IS MY CONFIG 

ip dhcp excluded-address 10.10.200.1 10.10.200.49
ip dhcp excluded-address 10.10.200.70 10.10.200.254
ip dhcp excluded-address 10.10.201.1 10.10.201.49
ip dhcp excluded-address 10.10.201.70 10.10.201.254
ip dhcp excluded-address 10.10.202.1 10.10.202.49
ip dhcp excluded-address 10.10.202.70 10.10.202.254
!
ip dhcp pool HQPHONES
   network 10.10.200.0 255.255.255.0
   option 150 ip 10.10.210.10
   default-router 10.10.200.3
!
ip dhcp pool br1phones
   network 10.10.201.0 255.255.255.0
   option 150 ip 10.10.210.10
   default-router 10.10.201.1
!
ip dhcp pool br2phones
   network 10.10.202.0 255.255.255.0
   option 150 ip 10.10.210.10
   default-router 10.10.202.1



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Re: [OSL | CCIE_Voice] gatekeeper max calls per endpoint

2010-08-19 Thread bkvalent...@gmail.com
Bingo.  Endpoint resource-threshold was missing.  Thanks.

- Reply message -
From: Miron Kobelski findko...@gmail.com
Date: Thu, Aug 19, 2010 1:29 pm
Subject: [OSL | CCIE_Voice] gatekeeper max calls per endpoint
To: Brian Valentine bkvalent...@gmail.com
Cc: OSL Group ccie_voice@onlinestudylist.com


Hi,

add  endpoint resource-threshold under gatekeeper and maybe arq
reject-resource-low

regards
kobel

On Thu, Aug 19, 2010 at 19:26, Brian Valentine bkvalent...@gmail.comwrote:

  endpoint max-calls h323id HQ-GW 1
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Re: [OSL | CCIE_Voice] SIP Phone Not Cancelling Session Upo n Request

2010-08-19 Thread bkvalent...@gmail.com
Change the dtmf relay on the voice register pool to rtp-nte.

Brian

- Reply message -
From: DeShon Crayton dcrayto...@comcast.net
Date: Thu, Aug 19, 2010 8:33 pm
Subject: [OSL | CCIE_Voice] SIP Phone Not Cancelling Session Upon Request
To: ccie_voice@onlinestudylist.com

I am running IOS 12.4.24. T3 on a 3845 

 

I have a 7941 IP Phone with a SIP load of :

SIP41.8-5-3SR1S

 

My issue is a follows:

1. Place a call to the SIP phone

2. End call remotely prior to SIP phone answering

3. The SIP phone still shows the line button blinking with caller id showing

 

I am not sure if I have a timer issue or a bug related problem on the 7941.

 

I see 3003 send the CANCEL message:

Aug 20 00:13:17.326: //-1//SIP/Msg/ccsipDisplayMsg:

Sent: 

CANCEL sip:3...@172.16.32.51:5060 SIP/2.0

Via: SIP/2.0/UDP 10.10.3.1:5060;branch=z9hG4bK7207D

From: SiteC Ph3 sip:3...@10.10.3.1;tag=52EFA4-114E

To: sip:3...@172.16.32.51

Date: Fri, 20 Aug 2010 00:13:09 GMT

Call-ID: 8c3c31b1-ab2611df-8053e949-17e37...@10.10.3.1

CSeq: 101 CANCEL

Max-Forwards: 70

Timestamp: 1282263197

Reason: Q.850;cause=16

Content-Length: 0

 

 

 

My config is as follows:

 

voice service voip 

 allow-connections h323 to h323

 allow-connections h323 to sip

 allow-connections sip to h323

 allow-connections sip to sip

 fax protocol cisco 

 sip

  bind control source-interface Loopback0

  bind media source-interface Loopback0

  registrar server expires max 600 min 60

 

voice register global

 mode cme

 source-address 10.10.3.1 port 5060

 max-dn 5

 max-pool 3

 load 7941 SIP41.8-5-3SR1S

 timezone 23

 time-format 24

 date-format D/M/Y

 tftp-path flash:

 create profile sync 0022063813214158

 ntp-server 172.16.30.250 mode unicast

 

 

voice register dn  2

 number 3002

 name SiteC Ph2

 no-reg

 label SiteC PH2 3002

!

voice register template  1

 dialplan 1

 softkeys connected  Endcall Hold Park Trnsfer

 no conference enable

!

voice register dialplan  1

 type 7940-7960-others

 pattern 1 3...

 pattern 2 999

!

voice register pool  2

 id mac 0022.9004.4147

 type 7941

 number 1 dn 2

 template 1

 dtmf-relay sip-notify

 username 3002 password cisco

 description 33132143002

 codec g711ulaw

 

 

 

Aug 20 00:13:09.430: //-1//SIP/Msg/ccsipDisplayMsg:

Sent: 

INVITE sip:3...@172.16.32.51:5060 SIP/2.0

Via: SIP/2.0/UDP 10.10.3.1:5060;branch=z9hG4bK7207D

Remote-Party-ID: SiteC Ph3
sip:3...@10.10.3.1;party=calling;screen=no;privacy=off

From: SiteC Ph3 sip:3...@10.10.3.1;tag=52EFA4-114E

To: sip:3...@172.16.32.51

Date: Fri, 20 Aug 2010 00:13:09 GMT

Call-ID: 8c3c31b1-ab2611df-8053e949-17e37...@10.10.3.1

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 2345154899-2871398879-2152655177-400783259

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Max-Forwards: 70

Timestamp: 1282263189

Contact: sip:3...@10.10.3.1:5060

Call-Info:
sip:10.10.3.1:5060;method=NOTIFY;Event=telephone-event;Duration=2000

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 206

 

v=0

o=CiscoSystemsSIP-GW-UserAgent 7078 7937 IN IP4 10.10.3.1

s=SIP Call

c=IN IP4 10.10.3.1

t=0 0

m=audio 17734 RTP/AVP 0 19

c=IN IP4 10.10.3.1

a=rtpmap:0 PCMU/8000

a=rtpmap:19 CN/8000

a=ptime:20

 

Aug 20 00:13:09.446: //-1//SIP/Msg/ccsipDisplayMsg:

Received: 

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 10.10.3.1:5060;branch=z9hG4bK7207D

From: SiteC Ph3 sip:3...@10.10.3.1;tag=52EFA4-114E

To: sip:3...@172.16.32.51

Call-ID: 8c3c31b1-ab2611df-8053e949-17e37...@10.10.3.1

Date: Fri, 20 Aug 2010 00:13:09 GMT

CSeq: 101 INVITE

Server: Cisco-CP7941G/8.5.3

Contact: sip:3...@172.16.32.51:5060;transport=udp

Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE

Allow-Events: kpml,dialog

Content-Length: 0

 

 

Aug 20 00:13:09.630: //-1//SIP/Msg/ccsipDisplayMsg:

Received: 

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 10.10.3.1:5060;branch=z9hG4bK7207D

From: SiteC Ph3 sip:3...@10.10.3.1;tag=52EFA4-114E

To: sip:3...@172.16.32.51;tag=00229004414700077048b4c3-81528c56

Call-ID: 8c3c31b1-ab2611df-8053e949-17e37...@10.10.3.1

Date: Fri, 20 Aug 2010 00:13:09 GMT

CSeq: 101 INVITE

Server: Cisco-CP7941G/8.5.3

Contact: sip:3...@172.16.32.51:5060;transport=udp

Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE

Remote-Party-ID: SiteC Ph2
sip:3...@10.10.3.1;party=called;id-type=subscriber;privacy=off;screen=yes

Allow-Events: kpml,dialog

Content-Length: 0

 

 

Aug 20 00:13:17.326: //-1//SIP/Msg/ccsipDisplayMsg:

Sent: 

CANCEL sip:3...@172.16.32.51:5060 SIP/2.0

Via: SIP/2.0/UDP 10.10.3.1:5060;branch=z9hG4bK7207D

From: SiteC Ph3 sip:3...@10.10.3.1;tag=52EFA4-114E

To: sip:3...@172.16.32.51

Date: Fri, 20 Aug 2010 00:13:09 GMT


Re: [OSL | CCIE_Voice] SIP Phone Not Cancelling Session Upo n Request

2010-08-19 Thread bkvalent...@gmail.com
Make sure you have a dial-peer.  show voice register dial-peer.  I suspect 
you may need to do authenticate register under voice register global as 
well.  Is your SIP bind command a different interface than the source ip under 
voice reg global?  Also did you do create profile and reset the pool?

Brian

- Reply message -
From: DeShon Crayton dcrayto...@gmail.com
Date: Thu, Aug 19, 2010 9:18 pm
Subject: [OSL | CCIE_Voice] SIP Phone Not Cancelling Session Upon Request
To: bkvalent...@gmail.com bkvalent...@gmail.com
Cc: DeShon Crayton dcrayto...@comcast.net, ccie_voice@onlinestudylist.com 
ccie_voice@onlinestudylist.com


hello Brian,

the dtmf-relay change did not work





Sent from my iPad

On Aug 19, 2010, at 9:02 PM, bkvalent...@gmail.com bkvalent...@gmail.com 
wrote:

 Change the dtmf relay on the voice register pool to rtp-nte.
 
 Brian
 
 - Reply message -
 From: DeShon Crayton dcrayto...@comcast.net
 Date: Thu, Aug 19, 2010 8:33 pm
 Subject: [OSL | CCIE_Voice] SIP Phone Not Cancelling Session Upon Request
 To: ccie_voice@onlinestudylist.com
 
 I am running IOS 12.4.24. T3 on a 3845 
 
 
 
 I have a 7941 IP Phone with a SIP load of :
 
 SIP41.8-5-3SR1S
 
 
 
 My issue is a follows:
 
 1. Place a call to the SIP phone
 
 2. End call remotely prior to SIP phone answering
 
 3. The SIP phone still shows the line button blinking with caller id showing
 
 
 
 I am not sure if I have a timer issue or a bug related problem on the 7941.
 
 
 
 I see 3003 send the CANCEL message:
 
 Aug 20 00:13:17.326: //-1//SIP/Msg/ccsipDisplayMsg:
 
 Sent: 
 
 CANCEL sip:3...@172.16.32.51:5060 SIP/2.0
 
 Via: SIP/2.0/UDP 10.10.3.1:5060;branch=z9hG4bK7207D
 
 From: SiteC Ph3 sip:3...@10.10.3.1;tag=52EFA4-114E
 
 To: sip:3...@172.16.32.51
 
 Date: Fri, 20 Aug 2010 00:13:09 GMT
 
 Call-ID: 8c3c31b1-ab2611df-8053e949-17e37...@10.10.3.1
 
 CSeq: 101 CANCEL
 
 Max-Forwards: 70
 
 Timestamp: 1282263197
 
 Reason: Q.850;cause=16
 
 Content-Length: 0
 
 
 
 
 
 
 
 My config is as follows:
 
 
 
 voice service voip 
 
 allow-connections h323 to h323
 
 allow-connections h323 to sip
 
 allow-connections sip to h323
 
 allow-connections sip to sip
 
 fax protocol cisco 
 
 sip
 
  bind control source-interface Loopback0
 
  bind media source-interface Loopback0
 
  registrar server expires max 600 min 60
 
 
 
 voice register global
 
 mode cme
 
 source-address 10.10.3.1 port 5060
 
 max-dn 5
 
 max-pool 3
 
 load 7941 SIP41.8-5-3SR1S
 
 timezone 23
 
 time-format 24
 
 date-format D/M/Y
 
 tftp-path flash:
 
 create profile sync 0022063813214158
 
 ntp-server 172.16.30.250 mode unicast
 
 
 
 
 
 voice register dn  2
 
 number 3002
 
 name SiteC Ph2
 
 no-reg
 
 label SiteC PH2 3002
 
 !
 
 voice register template  1
 
 dialplan 1
 
 softkeys connected  Endcall Hold Park Trnsfer
 
 no conference enable
 
 !
 
 voice register dialplan  1
 
 type 7940-7960-others
 
 pattern 1 3...
 
 pattern 2 999
 
 !
 
 voice register pool  2
 
 id mac 0022.9004.4147
 
 type 7941
 
 number 1 dn 2
 
 template 1
 
 dtmf-relay sip-notify
 
 username 3002 password cisco
 
 description 33132143002
 
 codec g711ulaw
 
 
 
 
 
 
 
 Aug 20 00:13:09.430: //-1//SIP/Msg/ccsipDisplayMsg:
 
 Sent: 
 
 INVITE sip:3...@172.16.32.51:5060 SIP/2.0
 
 Via: SIP/2.0/UDP 10.10.3.1:5060;branch=z9hG4bK7207D
 
 Remote-Party-ID: SiteC Ph3
 sip:3...@10.10.3.1;party=calling;screen=no;privacy=off
 
 From: SiteC Ph3 sip:3...@10.10.3.1;tag=52EFA4-114E
 
 To: sip:3...@172.16.32.51
 
 Date: Fri, 20 Aug 2010 00:13:09 GMT
 
 Call-ID: 8c3c31b1-ab2611df-8053e949-17e37...@10.10.3.1
 
 Supported: 100rel,timer,resource-priority,replaces,sdp-anat
 
 Min-SE:  1800
 
 Cisco-Guid: 2345154899-2871398879-2152655177-400783259
 
 User-Agent: Cisco-SIPGateway/IOS-12.x
 
 Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
 NOTIFY, INFO, REGISTER
 
 CSeq: 101 INVITE
 
 Max-Forwards: 70
 
 Timestamp: 1282263189
 
 Contact: sip:3...@10.10.3.1:5060
 
 Call-Info:
 sip:10.10.3.1:5060;method=NOTIFY;Event=telephone-event;Duration=2000
 
 Expires: 180
 
 Allow-Events: telephone-event
 
 Content-Type: application/sdp
 
 Content-Disposition: session;handling=required
 
 Content-Length: 206
 
 
 
 v=0
 
 o=CiscoSystemsSIP-GW-UserAgent 7078 7937 IN IP4 10.10.3.1
 
 s=SIP Call
 
 c=IN IP4 10.10.3.1
 
 t=0 0
 
 m=audio 17734 RTP/AVP 0 19
 
 c=IN IP4 10.10.3.1
 
 a=rtpmap:0 PCMU/8000
 
 a=rtpmap:19 CN/8000
 
 a=ptime:20
 
 
 
 Aug 20 00:13:09.446: //-1//SIP/Msg/ccsipDisplayMsg:
 
 Received: 
 
 SIP/2.0 100 Trying
 
 Via: SIP/2.0/UDP 10.10.3.1:5060;branch=z9hG4bK7207D
 
 From: SiteC Ph3 sip:3...@10.10.3.1;tag=52EFA4-114E
 
 To: sip:3...@172.16.32.51
 
 Call-ID: 8c3c31b1-ab2611df-8053e949-17e37...@10.10.3.1
 
 Date: Fri, 20 Aug 2010 00:13:09 GM___
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Re: [OSL | CCIE_Voice] I have a Question?

2010-08-18 Thread bkvalent...@gmail.com
CONGRATS! 

Sent from my Verizon Wireless Phone

- Reply message -
From: Beck, Ken kb...@vectorusa.com
Date: Wed, Aug 18, 2010 8:45 pm
Subject: [OSL | CCIE_Voice] I have a Question?
To: ccie_voice@onlinestudylist.com

What do I do with my time now that I
PASSED!!
!

 

 

Ken Beck CCIE Voice #26726

 

 

To Vik  Amy,

 

It was absolutely invaluable that first OWLE class I took and both of
you were there.  It truly changed my life.

 

Amy, I have listened to the audio on demand like 10X.  it's all I
listened to in the car and taking walks.

 

Vik, I can't express how much of an inspiration you are to us all.

 

 

Coffee is on me GUYS!

 


Information contained in this e-mail message is intended only for the 
individual to whom it is addressed and is private and confidential.  If you are 
not the intended recipient, or the employee or agent responsible for delivering 
this message to the intended recipient, any dissemination, distribution or 
copying of this communication is strictly prohibited.  If you are not the 
intended recipient of this e-mail, please kindly destroy it and notify the 
sender immediately by reply e-mail.  Thank you for your cooperation.


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Re: [OSL | CCIE_Voice] CCIE #26721 - I PASSED!

2010-08-17 Thread bkvalent...@gmail.com
Congrats!  Well done.  It's an amazing accomplishment. 

- Reply message -
From: Matthew Hall 1.matt.h...@gmail.com
Date: Tue, Aug 17, 2010 6:19 pm
Subject: [OSL | CCIE_Voice] CCIE #26721 - I PASSED!
To: Matthew Berry ciscovoiceg...@gmail.com
Cc: CCIE Voice OSL ccie_voice@onlinestudylist.com


Congrats I got my number on Monday, so the two Matts are in!!

On Aug 17, 2010, at 5:05 PM, Matthew Berry wrote:

 I just got my score report. I passed guys.
 
 More follow-up to come later.  Right now I'm now on cloud nine. :)
 
 CCIE #26271
 
 Thanks,
 
 Matthew Berry
 ciscovoiceg...@gmail.com
 http://ciscovoiceguru.com
 
 ___
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Re: [OSL | CCIE_Voice] Infrastructure Brainstorm request

2010-08-14 Thread bkvalent...@gmail.com
I once forgot to put a default gateway in my dhcp scope.  Makes it hard for the 
dhcp session to transition to unicast.  

Also, if  the ip helper-address command is absent or misconfigured.

The voice vlan on the branch router does not exist or has been disabled.  

Brian

- Reply message -
From: Daniel Berlinski dberlin...@gmail.com
Date: Sat, Aug 14, 2010 1:14 am
Subject: [OSL | CCIE_Voice] Infrastructure Brainstorm request
To: osl osl ccie_voice@onlinestudylist.com

Hello List

I would like to initiate a thread which I invite all of you to collaborate
sharing ideas on the following:

What would you think could break DHCP relay traffiic to succeed with the
scenario below:

(CUCM DHCP) 3750WAN-RTR WAN   WAN-RTR  Etherswitch   IP Phone

So far I can think of the following:
1- CSA enabled in CUCMs
2- Vlan access-maps in Switch
3- DHCP snooping configured in switch
4- service dhcp disabled in the branch router
5- L2/L3 problem within the network
6- ACLs in L3 interfaces along the way
7-DHCP service disabled in CUCM
8-DHCP service server/subnet page misconfiguration


I'm trying to go through all sections of the blueprint I believe I completed
100% at this stage and only practice troubleshooting scenarios.  Would be
thankful for any input

Cheers
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Re: [OSL | CCIE_Voice] CUPC won't TFTP

2010-08-08 Thread bkvalent...@gmail.com
You built a cupc phone is cucm with device name UPCUSERNAME?  

You can collect detailed cupc logs from within the cupc client to see why it is 
failing.  

Brian

- Reply message -
From: Warren Heaviside (wheavisi) wheav...@cisco.com
Date: Sun, Aug 8, 2010 7:56 pm
Subject: [OSL | CCIE_Voice] CUPC won't TFTP
To: ccie_voice@onlinestudylist.com

All,

 

I'm working through Vol. 1 lab 13A (Presence).  Integration is complete.
In CUPS DiagnosticsSystem Dashboard shows all systems good.  Trouble is
the CUPC won't make it through the TFTP process.  After launching CUPC
I'm looking at HelpShow Server Health on the client itself.  The
connection/authentication to the CUPS itself is fine but it fails TFTP.
I've disabled IPS on my laptop and verified TFTP Server (CUCM) settings
in CUPS.  I'm discounting my home based 871 due to the physical phones
being able to TFTP.  Thanks,

 

Warren

 

Warren Heavisidewheav...@cisco.com

ENGINEER.CUSTOMER SUPPORT

High Touch Technical Support

Phone: +1 408 853 7995

Office Hour 9 am - 5 pm Pacific Monday - Friday

 

For corporate legal information go to:

http://www.cisco.com/web/about/doing_business/legal/cri/index.html

 

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Re: [OSL | CCIE_Voice] SIP Integration between CUCM and CUC

2010-08-05 Thread bkvalent...@gmail.com
Yes.  On the SIP trunk in cucm, make sure to enable outbound redirect.

Enjoy.

Brian

- Reply message -
From: Ryan Schwab schwab...@shaw.ca
Date: Thu, Aug 5, 2010 7:48 pm
Subject: [OSL | CCIE_Voice] SIP Integration between CUCM and CUC
To: OSL Group ccie_voice@onlinestudylist.com

I've configured a SIP trunk integration from CUCM to CUC, everything is
working properly but with one major problem.

 

When calling the phone configured to forward-no answer to voicemail (via the
RP-SIP Trunk) the CALLING phone gets sent to CUC as a DIRECT call, not a
forwarded call. Therefore, I cannot reach the called party's voicemail.

 

Anyone have ideas?

 

 

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Re: [OSL | CCIE_Voice] class-based shaping

2010-08-03 Thread bkvalent...@gmail.com
I believe the qos srnd is provided on the computer's desktop in off format  
when you take the lab exam.  No need to find it through the web interface.  

- Reply message -
From: CCIE Voice GMAIL givemeccievoice2...@gmail.com
Date: Tue, Aug 3, 2010 6:02 pm
Subject: [OSL | CCIE_Voice] class-based shaping
To: ccie_voice@onlinestudylist.com

Has anyone figured out how to access this through the documentation site?
I’ve been using the search function and looking through all of the menus
without any luck.  

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Daniel
Berlinski
Sent: Tuesday, August 03, 2010 2:33 PM
To: Stutz, Bernhard
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] class-based shaping

 

Hello there

Check this path in the SRND
WAN Aggregator QoS Design - WAN Edge Link-Specific QoS Design - Frame Relay
- Slow-Speed (£ 768 kbps) Frame Relay Links, then turn the page and you will
find it.

HTH



On Wed, Aug 4, 2010 at 5:12 AM, Stutz, Bernhard st...@pandacom.de wrote:

Hi all,

 

In Proctorlabs there is always mentioned that the QOS SRND provides a
adequate example for class-based shaping but i can't find them. 

 

Also i heard rumors that the QOS SRND document provided at the labs is not
at Cisco Website available anymore. However i can find a version here: 

http://www.cisco.com/en/US/docs/solutions/Enterprise/WAN_and_MAN/QoS_SRND/Qo
S-SRND-Book.html

 

but i can't find there an example for class-based shaping. Does anyone know
where i can find such a adequate example?

 

good luck and have fun in your studies,

Bernhard


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visit www.ipexpert.com

 

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Re: [OSL | CCIE_Voice] First attempt

2010-07-30 Thread bkvalent...@gmail.com
Guys be careful. Score reports are subject to NDA.

- Reply message -
From: Daniel Berlinski dberlin...@gmail.com
Date: Thu, Jul 29, 2010 6:14 pm
Subject: [OSL | CCIE_Voice] First attempt
To: CCIE Voice GMAIL givemeccievoice2...@gmail.com
Cc: OSL Group ccie_voice@onlinestudylist.com


I've heard this from Cisco employees as well that when the proctors reach
the conclusion that the candidate cannot make it up the 80% they just stop
the correction. This is something I will definetely ask whenever there is
another ask the expert forum.  By the way has anyone ever looked for this
info in the ask the expert archives?

I know this forum is packed with Cisco staff.  Can any of you clarify this
for us?

On Fri, Jul 30, 2010 at 7:29 AM, CCIE Voice GMAIL 
givemeccievoice2...@gmail.com wrote:

  I have heard this from a couple of people and even on this mailer.  That
 is why I am bringing it up.  I am not 100% sure if it is accurate or not.



 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Graham Hopkins
 *Sent:* Thursday, July 29, 2010 12:09 PM
 *To:* OSL Group

 *Subject:* Re: [OSL | CCIE_Voice] First attempt







 I don't see how this can be correct, if it is it makes the report
 meaningless. You could screw up a few early sections,  fail on 79% and still
 have most of the report as 0.  Of course as the score report is subject to
 NDA we'll never know.



 Still Ohamien keep working on it and you will get there.





 Graham





 On 29 Jul 2010, at 19:59, CCIE Voice GMAIL wrote:



   It’s also important to note, and correct me if I’m wrong, that the 0’s
 don’t necessarily mean you configured that section incorrectly.



 To my knowledge, once you lose more than 20 points, they simply stop
 grading your exam.  So the later section may have 0’s but you configured
 them correctly.



 I feel like this is a big problem with the already vague score reports.  I
 wish they would change this.  If you are paying $1400, you deserve a full
 report in my opinion.



 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Ashar Siddiqui
 *Sent:* Thursday, July 29, 2010 11:25 AM
 *To:* Ohamien Uhakheme
 *Cc:* OSL Group
 *Subject:* Re: [OSL | CCIE_Voice] First attempt



 I am sure you will figure out what mistakes you made which resulted in 0%.
 I know its very hard to find out when you are sure your solution is 100%
 but believe me I have been through this and you will come to know how a tiny
 mistake in that particular section or may be in some other section resulted
 in 0% for this section :)

 I hope you pass in 2nd attempt. Don't forget to break down your scores and
 analyze exactly which question you lost points. That will help you to work
 out on specific areas.

 Ash

 Ohamien Uhakheme wrote:

 Hey guys --

 I've been lurking for a while, so I figured that I'd chime in.  I sat for
 my first attempt yesterday with less than passing results.  Like other
 people have mentioned, it is heart breaking to see 0% in areas that you are
 sure that you nailed completely.  It's cool though, I needed to get the
 psychological first attempt out of the way, and I will probably schedule
 again for early September.

 IPExpert is spot on with their training material, and I definitely
 appreciate the effort that has gone into it.

 Thanks guys,

 Ohamien





 --







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Re: [OSL | CCIE_Voice] Meetme: cannot use join select sof tkeys to join 3rd call to meetme conference

2010-07-22 Thread bkvalent...@gmail.com
Just transfer the caller to the meet me extension. 

Select and join are for ad-hoc.

- Reply message -
From: Pavan pav.c...@gmail.com
Date: Thu, Jul 22, 2010 7:16 am
Subject: [OSL | CCIE_Voice] Meetme: cannot use join select softkeys to join 3rd 
call to meetme conference
To: jeremy co jeremy.coo...@gmail.com
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com, 
cisco-v...@puck.nether.net cisco-v...@puck.nether.net


Join is used to trigger ad hoc conference. It is not used for meetme conf.

I am not sure if cme allows you to chain a add a ad hoc participant into meetme 
conf using join

Sent from my phone

On Jul 22, 2010, at 3:49 AM, jeremy co jeremy.coo...@gmail.com wrote:

 From admin guideL
 
 
 Soft Keys for Conference Functions
 In Cisco Unified CME 4.1 and later versions, the following soft keys provide 
 conferencing functions for
 multi-party conferencing enhancements on your phone:
 • ConfList—Conference list. Lists all parties in a conference. For ad hoc 
 conferences, this soft key is
 available for all parties in a conference. For meet-me conferences, this soft 
 key is available for the
 creator only. Press Update to update the list of parties in the conference, 
 for instance, to verify that
 a party has been removed from the conference.
 • Join—Joins an established call to a conference. You must first press Select 
 to choose each
 connected call that you want to join in a conference, then press Join to join 
 the selected calls to the
 conference.
 • RmLstC—Remove last caller. Removes the last party added to the conference. 
 This soft key works
 for the creator only.
 • Select—Selects a call or conference to join to a conference and selects a 
 call to remove from a
 conference. The creator can remove other parties by pressing the ConfList 
 soft key, then use the
 Select and Remove soft keys to remove the appropriate parties.
 • MeetMe—Initiates a meet-me conference. The creator presses this soft key 
 before dialing the
 conference number. Other meet-me conference parties only dial the conference 
 number to join the
 conference. This soft key must be configured before you can initiate meet-me 
 conferences.
 
 
 SCCP: Configuring Conferencing Options for a Phone
 
 
 
 cheers,
 
 
 Jeremy
 
 also look at section called:  
 On Thu, Jul 22, 2010 at 6:33 PM, Ashar Siddiqui siddas...@gmail.com wrote:
 From the admin guide:
 
  
 
 Meet-Me Conferencing in Cisco Unified CME 4.1 and Later versions
 
 In Cisco Unified CME 4.1 and later versions, meet-me conferences consist of 
 at least three parties dialing a meet-me conference number predetermined by a 
 system administrator. For example, the conference shown in Figure 24 is 
 created when the conference creator at extension 1215 presses the MeetMe soft 
 key and hears a confirmation tone, then dials the meet-me conference number 
 1500. Extension 1225 and extension 1235 join the meet-me conference by 
 dialing 1500. Extensions 1215, 1225, and 1235 are now parties in a meet-me 
 conference on extension 1500.
 
 MeetMe—Initiates a meet-me conference. The creator presses this soft key 
 before dialing the conference number. Other meet-me conference parties only 
 dial the conference number to join the conference. This soft key must be 
 configured before you can initiate meet-me conferences.
 
 Join key can be used in normal conferencing.
 
  
 
 Ash
 
  
 
 From: jeremy co [mailto:jeremy.coo...@gmail.com] 
 Sent: 22 July 2010 09:19
 
 
 To: Ashar Siddiqui
 Cc: ccie_voice@onlinestudylist.com; cisco-v...@puck.nether.net
 Subject: Re: [OSL | CCIE_Voice] Meetme: cannot use join select softkeys to 
 join 3rd call to meetme conference
  
 
 I'm not disagreeing with you.   So what is the usage of select and join key 
 that mentioned in CME admin guide , conference section ?
 
  
 
  
 
 Cheers,
 
  
 
 Jeremy
 
 On Thu, Jul 22, 2010 at 6:00 PM, Ashar Siddiqui siddas...@gmail.com wrote:
 
 I don’t think it would work that way.
 
  
 
 All participants must come through by dialling meetme number to join a meetme 
 conf (this is what I think) otherwise it won’t work.
 
  
 
 Ash
 
  
 
 From: jeremy co [mailto:jeremy.coo...@gmail.com] 
 Sent: 22 July 2010 08:51
 To: Ashar Siddiqui
 Cc: ccie_voice@onlinestudylist.com; cisco-v...@puck.nether.net
 Subject: Re: [OSL | CCIE_Voice] Meetme: cannot use join select softkeys to 
 join 3rd call to meetme conference
 
  
 
 Ph1 go to seized state and then press meetme softkey then dial 4200 (meetme 
 number) and ph2 and ph3 call to 4200 to join the conference. 
 
  
 
 As I want to test join softkey feature , I made a call from PSTN to 4013.   
 PH1 answer the call and then I try to join 4013 and 4200 togther using 
 select and join key which is not working.
 
  
 
  
 
  
 
 Cheers,
 
  
 
  
 
 Jeremy
 
 On Thu, Jul 22, 2010 at 5:42 PM, Ashar Siddiqui siddas...@gmail.com wrote:
 
 Sorry didn’t understand this bit “Ph1 create meetme conference and ph2 and 
 ph3 joins to conference. PSTN 

Re: [OSL | CCIE_Voice] CUE CLI UserPrompt

2010-07-19 Thread bkvalent...@gmail.com
Bump

- Reply message -
From: Brian Valentine bkvalent...@gmail.com
Date: Sat, Jul 17, 2010 9:04 pm
Subject: CUE CLI UserPrompt
To: OSL Group ccie_voice@onlinestudylist.com

I'm experimenting with the CLI on CUE.  I'm trying to figure out how
much I can do from the Command Line.  So, If I'm asked to create a
custom promptand let's say there was a requirement not to record
over any existing greetings...

Is it possible to figure out the filename of the new custom prompt
from the CUE CLI?

Thanks folks for any help you can give.

Brian Valentine
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Re: [OSL | CCIE_Voice] proctor rack 15, CUE issue

2010-07-15 Thread bkvalent...@gmail.com
I had to bootload it during my session today.  It was stuck in and endless 
reboot loop. Thought about reporting it.  Figured it was worth the effort to 
learn the process.  I thought I had fixed it.  It worked after I reimaged it.  

Brian

- Reply message -
From: Erwan Erwan e_er...@yahoo.com
Date: Thu, Jul 15, 2010 7:53 pm
Subject: [OSL | CCIE_Voice] proctor rack 15, CUE  issue
To: ccie_voice@onlinestudylist.com

hi all,
 
Is anybody experience  CUE issue in rack 15 ?
 
Has been 2 days stuck in   27% boot and no ipaddress in Service Engine 0/0
 
I tried to reboot many times, here is my config, pls help
 

 
interface Service-Engine0/0
 ip unnumbered Vlan400
 service-module ip address 10.10.202.2 255.255.255.0
 service-module ip default-gateway 10.10.202.1

BR2-RTR#sh run | s ip route
ip route 10.10.202.2 255.255.255.255 Service-Engine0/0

Vlan400 is up, line protocol is up
 
interface Vlan400
 ip address 10.10.202.1 255.255.255.0
end

 
tks

 
 
 


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Re: [OSL | CCIE_Voice] CME SRST Mode

2010-07-14 Thread bkvalent...@gmail.com
Is your ip source address on the phones config as the srst? Is that interface 
up/up?  Ntp status?

- Reply message -
From: ccieid1ot ccieid...@gmail.com
Date: Wed, Jul 14, 2010 6:20 pm
Subject: [OSL | CCIE_Voice] CME SRST Mode
To: ccie_voice@onlinestudylist.com CCIE_Voice@onlinestudylist.com

Hi gangs,

I can not seem to get the ephones to register with their assigned
DN's.  Tried all types of srst mode auto-pro commands.  Someone please
give me a sanity check.


!
telephony-service
 sdspfarm units 5
 sdspfarm tag 1 SC-CFB
 no privacy
 conference hardware
 srst mode auto-provision none
 srst ephone template 1
 srst dn template 1
 srst dn line-mode octo
 max-ephones 10
 max-dn 10
 ip source-address 192.102.66.254 port 2000
 system message Your phones are in fallback
 max-conferences 8 gain -6
 call-forward pattern .T
 transfer-system full-consult
 create cnf-files version-stamp 7960 Apr 05 2009 12:47:04
!
!
ephone-dn-template  1
 call-forward busy 91658200
 call-forward noan 91658200 timeout 20
!
!
ephone-template  1
 privacy off
 privacy-button
 softkeys remote-in-use  Newcall CBarge
 softkeys idle  Newcall Redial
 softkeys seized  Redial Endcall Cfwdall Pickup Gpickup Meetme
 softkeys connected  Hold Endcall Trnsfer Park Confrn ConfList
!
!
ephone-dn  4  octo-line
 number B16A55 no-reg primary
 conference ad-hoc
!
!
ephone-dn  10  octo-line
 number A100
 conference ad-hoc
!
!
ephone  1
 privacy-button
 device-security-mode none
 max-calls-per-button 4
 busy-trigger-per-button 1
!
!
!
ephone  2
 privacy-button
 device-security-mode none
 max-calls-per-button 4
 busy-trigger-per-button 1
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Re: [OSL | CCIE_Voice] CME SRST Mode

2010-07-14 Thread bkvalent...@gmail.com
Phones are sccp in cucm? 

Sent from my Verizon Wireless Phone

- Reply message -
From: ccieid1ot ccieid...@gmail.com
Date: Wed, Jul 14, 2010 7:53 pm
Subject: [OSL | CCIE_Voice] CME SRST Mode
To: CCIE Voice ccievoiced...@gmail.com
Cc: ccie_voice@onlinestudylist.com CCIE_Voice@onlinestudylist.com, 
ccie_voice-boun...@onlinestudylist.com


Tried again, removing telephony-s reloaded, phones registered to ccm
after router boot up, failback to SRST, still just the system message
and no extension.

On Wed, Jul 14, 2010 at 6:32 PM, CCIE Voice ccievoiced...@gmail.com wrote:
 Jeff is correct...whenever I have to make a change to CME SRST config,
 especially the ephones, I copy all of my related config to notepad and issue
 the no telephony-service command.  I then paste everything back in and it
 works fine.  Don't know if this is a bug or what but it happens every single
 time.

 -Original Message-
 From: ccie_voice-boun...@onlinestudylist.com
 [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Price
 (jeffpric)
 Sent: Wednesday, July 14, 2010 7:12 PM
 To: ccieid1ot
 Cc: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] CME SRST Mode

 I have definitely had this problem before, but never really figure out why
 it happens.

 Try this:
 Copy your telephony-service config to Notepad
 do a no telephony-service
 Register the phones with CUCM again
 Paste in your telephony-service config
 Bring down the WAN connection.

 Eventually the phones will register with the proper DNs.

 It's not ideal, but it works most of the time.

 Hope this helps,
 Jeff

 -Original Message-
 From: ccieid1ot [mailto:ccieid...@gmail.com]
 Sent: Wednesday, July 14, 2010 4:10 PM
 To: Jeff Price (jeffpric)
 Cc: Edwin Dotson; ccie_voice@onlinestudylist.com;
 ccie_voice-boun...@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] CME SRST Mode

 Jeff,

 same thing.  I had it registered back to CCM then failback to SRST.


 ccievoice-sitec#show telephony-service ephone-dn

 ephone-dn 4 octo-line
 number B16A55 no-reg primary
 preference 0 secondary 9
 huntstop
 huntstop channel 8
 call-waiting beep
 conference ad-hoc

 ephone-dn 10 octo-line
 number A100
 preference 0 secondary 9
 huntstop
 huntstop channel 8
 call-waiting beep
 conference ad-hoc

 ccievoice-sitec#show run | s telephony
 telephony-service
  sdspfarm units 5
  sdspfarm tag 1 SC-CFB
  no privacy
  conference hardware
  srst mode auto-provision all
  srst ephone template 1
  srst dn template 1
  srst dn line-mode octo
  max-ephones 10
  max-dn 10
  ip source-address 142.102.66.254 port 2000
  system message Your phones are in fallback
  max-conferences 8 gain -6
  call-forward pattern .T
  transfer-system full-consult
  create cnf-files version-stamp 7960 Apr 05 2009 12:47:04
 ccievoice-sitec#


 On Wed, Jul 14, 2010 at 5:57 PM, Jeff Price (jeffpric)
 jeffp...@cisco.com wrote:
 You have the auto provision mode set to none.  You should set to all so
 that CME will download the phone's config completely.  This is most likely
 your problem.

 Try and change the command to srst mode auto all and let me know what
 happens.

 -Original Message-
 From: ccie_voice-boun...@onlinestudylist.com
 [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccieid1ot
 Sent: Wednesday, July 14, 2010 3:54 PM
 To: Edwin Dotson
 Cc: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] CME SRST Mode

 Done that, and phones get nothing.

 On Wed, Jul 14, 2010 at 5:51 PM, Edwin Dotson edot...@ams.net wrote:
 Have you tried removing the static ephone configuration? They may be
 picking that profile which don't have buttons specified.

 Thanks
 Edwin
 Sent from my Verizon Wireless BlackBerry

 -Original Message-
 From: ccieid1ot ccieid...@gmail.com
 Sender: ccie_voice-boun...@onlinestudylist.com
        ccie_voice-boun...@onlinestudylist.com
 Date: Wed, 14 Jul 2010 15:20:27
 To: ccie_voice@onlinestudylist.comCCIE_Voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice] CME SRST Mode

 Hi gangs,

 I can not seem to get the ephones to register with their assigned
 DN's.  Tried all types of srst mode auto-pro commands.  Someone please
 give me a sanity check.


 !
 telephony-service
  sdspfarm units 5
  sdspfarm tag 1 SC-CFB
  no privacy
  conference hardware
  srst mode auto-provision none
  srst ephone template 1
  srst dn template 1
  srst dn line-mode octo
  max-ephones 10
  max-dn 10
  ip source-address 192.102.66.254 port 2000
  system message Your phones are in fallback
  max-conferences 8 gain -6
  call-forward pattern .T
  transfer-system full-consult
  create cnf-files version-stamp 7960 Apr 05 2009 12:47:04
 !
 !
 ephone-dn-template  1
  call-forward busy 91658200
  call-forward noan 91658200 timeout 20
 !
 !
 ephone-template  1
  privacy off
  privacy___
For more information regarding 

Re: [OSL | CCIE_Voice] problems getting SIP phone registere d to BR2

2010-07-10 Thread bkvalent...@gmail.com
Do not use SIP-notify.  Use rtp-nte for sip endpoints on cucme.  

Sent from my Verizon Wireless Phone

- Reply message -
From: Miron Kobelski findko...@gmail.com
Date: Sat, Jul 10, 2010 3:50 pm
Subject: [OSL | CCIE_Voice] problems getting SIP phone registered to BR2
To: CCIE Voice cc...@corb.net
Cc: ccie_voice@onlinestudylist.com


Hello,

Apart from everything else suggested, I don't thing router reset is
necessary for this (you loose some time). I don't remember performing reload
when I had this issue. Although for stubborn IP phones I usually perform
factory reset and provide everything what the phone requests from TFTP.

regards
kobel

On Sat, Jul 10, 2010 at 9:27 PM, CCIE Voice cc...@corb.net wrote:

 Thanks Adam.  Your suggestion worked.  I did add the phone type to the
 config as well as the steps below.  I think it was the clearing/recreating
 the profiles and bouncing the router that did it.

 Thanks again...scd



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Re: [OSL | CCIE_Voice] NTP

2010-07-09 Thread bkvalent...@gmail.com
A router that has ntp server configured also acts as a server.  You don't 
need ntp master to act as a server.

Brian

- Reply message -
From: cisco voip voip.ccieci...@gmail.com
Date: Thu, Jul 8, 2010 10:42 pm
Subject: [OSL | CCIE_Voice] NTP
To: Mark Holloway m...@markholloway.com
Cc: OSL osl ccie_voice@onlinestudylist.com


But ntp master command is necessary for the router to be able to be ntp
server for anyone else. i agree about stratum chain, but without ntp master
command publisher should not be able to get its clock.

you can check show ntp associations on the router, after ntp master command
only it will show you the local interface.

now if you want to use the stratum to be 1 more from whatever the stratum is
on actual source, you do not need stratum command, for anything else you
need that commnd

On Fri, Jul 9, 2010 at 4:42 AM, Mark Holloway m...@markholloway.com wrote:

 Cool, thanks Graham and Randall.

 On Jul 8, 2010, at 4:11 PM, Graham Hopkins wrote:

  Default stratum is 8 so a simple ntp master will work
 
  Graham
 
  On 8 Jul 2010, at 23:59, Mark Holloway m...@markholloway.com wrote:
 
  Yikes, I meant ntp master stratum X not ntp server stratum X
 
  On Jul 8, 2010, at 3:57 PM, Mark Holloway wrote:
 
  If a router (for example, HQ) is configured with the ntp server
 x.x.x.x command to sync time from another source, but I want another device
 (such as PUB) to get its time from the HQ router, do I also need to
 configure the HQ router with ntp server stratum X or can UCM simply get
 the time sync from HQ without the stratum command?
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 please visit www.ipexpert.com
 
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Re: [OSL | CCIE_Voice] VATS in the VOD

2010-07-07 Thread bkvalent...@gmail.com
Thanks for clarifying!

Brian

- Reply message -
From: Vik Malhi vma...@ipexpert.com
Date: Wed, Jul 7, 2010 6:08 pm
Subject: VATS in the VOD
To: Brian Valentine bkvalent...@gmail.com, OSL Group 
ccie_voice@onlinestudylist.com

Fragmentation values should be based on sending rate when there are active
voice sessions- the target serialization delay of 10ms applies when there is
voice traffic over the FR WAN. Therefore the fragmentation should be based
on the adaptive rate. Sorry for any confusion.

-- 
Vik Malhi ­ CCIE #13890
Managing Partner / Instructor - IPexpert, Inc.
Mailto: vma...@ipexpert.com
Telephone: +1.810.326.1444 ext 420
Fax: +1.810.454.0130
Live Assistance, Please visit: www.ipexpert.com/chat
http://www.ipexpert.com/chat

IPexpert is a premier provider of Self-Study Workbooks, Video on Demand,
Audio Tools, Online Hardware Rental and Classroom Training for the Cisco
CCIE (RS, Voice, Security  Service Provider) certification(s) with
training locations throughout the United States, Europe, South Asia and
Australia. Be sure to visit our online communities at
www.ipexpert.com/communities http://www.ipexpert.com/communities  and our
public website at www.ipexpert.com http://www.ipexpert.com/



 From: Brian Valentine bkvalent...@gmail.com
 Date: Wed, 7 Jul 2010 09:33:05 -0400
 To: Vik Malhi vma...@ipexpert.com, OSL Group
 ccie_voice@onlinestudylist.com
 Subject: Re: VATS in the VOD
 
 Anyone else care to shed some light?
 
 Thanks,
 
 Brian
 
 On Mon, Jul 5, 2010 at 3:28 PM, Brian Valentine bkvalent...@gmail.com wrote:
 Vik,
 
 Well done on the VoD product.  It's really very helpful.  I was going
 through the WAN QoS video today.  Question for you on VATS - should
 the fragment size be based on the adaptive rate or the cir?
 
 In the VoD you mention that the fragment size in the class should be
 960, not 80.  I understand that 80 was nonsense, but I was thinking it
 should be 480.   I would appreciate it if you could clarify for me.
 See attached screenshot.
 
 Thanks in advance!
 
 Brian Valentine
 


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Re: [OSL | CCIE_Voice] Offering to ProctorLabs Clients

2010-07-02 Thread bkvalent...@gmail.com
I think its easy to test with a simple acl on hq-rtr.

If br1phn2 is 192.168.5.13, my acl is 

ip access-list extended SRST
 deny ip 10.10.201.0 0.0.0.255 10.10.210.0 0.0.0.255
 deny ip host 10.10.110.2 10.10.210.0 0.0.0.255
 deny ip host 192.168.5.13 10.10.210.0 0.0.0.255
 permit ip any any

Applied outbound on fas0/0.30.

Seems easy enough to me.  Don't see much need to automate it. Plus in the real 
lab, how will people verify that their config is good?  I wouldn't want to 
cripple people from learning how to verify.  It's good practice for the big 
day.  My .02.  

Brian

- Reply message -
From: Tyson Scott tsc...@ipexpert.com
Date: Fri, Jul 2, 2010 5:21 pm
Subject: [OSL | CCIE_Voice] Offering to ProctorLabs Clients
To: ccie_voice@onlinestudylist.com

Hey Voice Team,

 

We recently made some design changes to our Voice bootcamp offerings to
extend our PL infrastructure directly to the bootcamp phones.  Meaning The
Voice candidates in our bootcamps are able to test QoS, SRST, and other
services offered at each branch site using the phones in the classes as if
they were connected to the rack equipment.

 

We can extend this same service to you as voice customers but I first wanted
to check into the interest of you all to see if there is interest in the
ability to do this. 

 

What it would provide is the phones that you have at your remote location,
physical phones, would be able to appear as though they are directly
connected to the PL devices.  Now obviously this is going to add a lot of
complexity that I will need to resolve so I want to first find if the
interest is there.

 

Please respond if interested.  I will test this with a few students once we
have the infrastructure in place to support this capability.  It would be
later next week.

 

Regards,

 

Tyson Scott - CCIE #13513 RS, Security, and SP

Managing Partner / Sr. Instructor - IPexpert, Inc.

Mailto: tsc...@ipexpert.com

Telephone: +1.810.326.1444, ext. 208

Live Assistance, Please visit: www.ipexpert.com/chat

eFax: +1.810.454.0130

 

IPexpert is a premier provider of Self-Study Workbooks, Video on Demand,
Audio Tools, Online Hardware Rental and Classroom Training for the Cisco
CCIE (RS, Voice, Security  Service Provider) certification(s) with
training locations throughout the United States, Europe, South Asia and
Australia. Be sure to visit our online communities at
www.ipexpert.com/communities and our public website at www.ipexpert.com
http://www.ipexpert.com/ 

 

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Re: [OSL | CCIE_Voice] Fwd: VOIP issue between CUCM 7 to UC500

2010-07-01 Thread bkvalent...@gmail.com
Open a TAC case with Cisco.

- Reply message -
From: Saeed IDris saee...@gmail.com
Date: Thu, Jul 1, 2010 1:10 pm
Subject: [OSL | CCIE_Voice] Fwd: VOIP issue between CUCM 7 to UC500
To: ccie_voice@onlinestudylist.com

 Hi expert,

I have strange issue with VOIP calls between two sites, the issue is when I
dial from site B to site A the call is going fine, but when I call from site
A to site B also is going fine except  with Cisco IP Phone 7945 once they
answer the call immediately goes down but If I try to reach another phone
type all calls going fine ……any suggestion why 7945 drops calls,  here is my
setup details:

Site A info:

-  Call Processing : CUCM 7.0

-  Cisco IP Phones 7941 , 7911  7970

-  Cisco IP Phones is registered as SIP with CUCM 7.0

-  I add UC 500 “Site B” call processing as H.323 device

-  Configure route pattern , route list and route Group “UC 500”

Site B info:

-  Call Processing : UC 500

-  Cisco IP Phones 7945 , 7985  Soft phone

-  Cisco IP Phones is registered as SCCP with UC500

-  Dial-peer already configured pointing to CUCM 7.0 “Site A” call
processing



Please note that I attached the debug on UC 500 while I make calls:

Called Number: 406 (7945)

Calling number : 106  (7941)



Thanks in advance.

SAEED
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Re: [OSL | CCIE_Voice] Vol 1 Lab 5C Q12 Mobile Voice Access

2010-06-17 Thread bkvalent...@gmail.com
Yes. On your remote destination, don't put any leading digits.  Don't put the 9 
trunk access code.  Leave it the same as your mobile ani.  

Use application dial rules to prefix the trunk access code. Mobile connect uses 
application dial rules to xform the redirecting number.

MVA doesn't like it when the ANI of the caller is shorter than the configured 
remote destination.


Brian

- Reply message -
From: Jones, Brett brett.jo...@redstone.co.uk
Date: Thu, Jun 17, 2010 7:22 pm
Subject: [OSL | CCIE_Voice] Vol 1 Lab 5C Q12 Mobile Voice Access
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com

Hi,

I have configure mobile voice access as describe in the video walk through, 
however when I dial in from my mobile or any other number the IVR asks for me 
to enter my remote destination number followed by the pound key and not my 
pin number. When I enter  12345# (which is the pin number) it tell me that it's 
not a recognised remote destination number.

I have set the service parameter to partial match and even changed the matched 
digits to 7.

Anyone see this before?

Thanks



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Re: [OSL | CCIE_Voice] VMware Connectivity and VLAN Interfa ce Issues

2010-06-13 Thread bkvalent...@gmail.com
Used pod 12 last night without incident. 

Brian 

- Reply message -
From: Matthew Berry ciscovoiceg...@gmail.com
Date: Sun, Jun 13, 2010 5:09 pm
Subject: [OSL | CCIE_Voice] VMware Connectivity and VLAN Interface  Issues
To: OSL Group ccie_voice@onlinestudylist.com

Has anyone else experienced issues with Pod 12 on PL?

I am using that pod right now and have two issues that prevent me from 
moving forward.

Issue 1:  I can ping the virtual servers from my own hardware and from 
the remote routers.  However, I cannot load the web page admin  nor can 
I SSH into the CUCM/CUC devices.

Issue 2: BR1 gateway has a line protocol down for VLAN interfaces 130 
and 240.  I mentioned this on the OSL a few weeks prior.  The basic 
issue is that those interfaces will never show an UP/UP state.  I have 
reloaded the router, removed configs, compared with previous working 
configs, etc.  Still no dice.  I made sure that the interfaces were 
configured properly.  In fact, the same configuration works on BR2 
(different VLAN ID and IPs, off course).

Below are some outputs to defend issue two:

BR1-RTR#show int vlan 130
Jun 14 00:44:05.575: %SYS-5-CONFIG_I: Configured from console by console
BR1-RTR#show int vlan 130
Vlan130 is up, line protocol is down
   Hardware is EtherSVI, address is 0017.9460.8d40 (bia 0017.9460.8d40)
   Internet address is 10.10.101.1/24
   MTU 1500 bytes, BW 10 Kbit/sec, DLY 100 usec,
  reliability 255/255, txload 1/255, rxload 1/255
   Encapsulation ARPA, loopback not set
   ARP type: ARPA, ARP Timeout 04:00:00
   Last input never, output never, output hang never
   Last clearing of show interface counters never
   Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops: 0
   Queueing strategy: fifo
   Output queue: 0/40 (size/max)
   5 minute input rate 0 bits/sec, 0 packets/sec
   5 minute output rate 0 bits/sec, 0 packets/sec
  0 packets input, 0 bytes, 0 no buffer
  Received 0 broadcasts, 0 runts, 0 giants, 0 throttles
  0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored
  0 packets output, 0 bytes, 0 underruns
  0 output errors, 0 interface resets
  0 unknown protocol drops
  0 output buffer failures, 0 output buffers swapped out

BR1-RTR#show ip int vlan 130
Vlan130 is up, line protocol is down
   Internet address is 10.10.101.1/24
   Broadcast address is 255.255.255.255
   Address determined by setup command
   MTU is 1500 bytes
   Helper address is not set
   Directed broadcast forwarding is disabled
   Multicast reserved groups joined: 224.0.0.5
   Outgoing access list is not set
   Inbound  access list is not set
   Proxy ARP is enabled
   Local Proxy ARP is disabled
   Security level is default
   Split horizon is enabled
   ICMP redirects are always sent
   ICMP unreachables are always sent
   ICMP mask replies are never sent
   IP fast switching is enabled
   IP fast switching on the same interface is disabled
   IP Flow switching is disabled
   IP CEF switching is enabled
   IP CEF switching turbo vector
   IP Null turbo vector
   IP multicast fast switching is enabled
   IP multicast distributed fast switching is disabled
   IP route-cache flags are Fast, CEF
   Router Discovery is disabled
   IP output packet accounting is disabled
   IP access violation accounting is disabled
   TCP/IP header compression is disabled
   RTP/IP header compression is disabled
   Policy routing is disabled
   Network address translation is disabled
   BGP Policy Mapping is disabled
   WCCP Redirect outbound is disabled
   WCCP Redirect inbound is disabled
   WCCP Redirect exclude is disabled


BR1-RTR#show vlan-switch brief

VLAN Name StatusPorts
  - 
---
1default  activeFa1/0, Fa1/1, Fa1/2, Fa1/3
 Fa1/4, Fa1/5, Fa1/6, Fa1/7
 Fa1/8, Fa1/9, Fa1/10, 
Fa1/11
 Fa1/12, Fa1/13, Fa1/14, 
Fa1/15
130  DATA active
240  PHONES   activeFa1/1, Fa1/2, Fa1/3, Fa1/4
 Fa1/5, Fa1/6, Fa1/7, Fa1/8
 Fa1/9, Fa1/10, Fa1/11, 
Fa1/12
 Fa1/13, Fa1/14, Fa1/15
1002 fddi-default act/unsup
1003 token-ring-default   act/unsup
1004 fddinet-default  act/unsup
1005 trnet-defaultact/unsup

interface FastEthernet1/1
  switchport trunk native vlan 130
  switchport mode trunk
  switchport voice vlan 240
  spanning-tree portfast
!
interface FastEthernet1/2
  switchport trunk native vlan 130
  switchport mode trunk
  switchport voice vlan 240
  spanning-tree portfast
!
interface Vlan130
  ip address 10.10.101.1 255.255.255.0
!
interface Vlan240
 

Re: [OSL | CCIE_Voice] CUC Voicmeail diversion

2010-06-09 Thread bkvalent...@gmail.com
Just make 2999 an alternate ext of 2602.

HTH,
 
Brian

- Reply message -
From: Ashar Siddiqui siddas...@gmail.com
Date: Wed, Jun 9, 2010 8:37 am
Subject: [OSL | CCIE_Voice] CUC Voicmeail diversion
To: 'wolfsrudel' wolfsru...@gmail.com, ccie_voice@onlinestudylist.com

Thanks Wolf,
If I disable VM on 2999, will an incoming call go to VM of2602?


-Original Message-
From: wolfsrudel [mailto:wolfsru...@gmail.com] 
Sent: 09 June 2010 13:33
To: Ashar Siddiqui; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CUC Voicmeail diversion

resuming to the original called number is the expected behaviour.
remove call-coverage settings from 2999 or disable it's diversion to vm.

hth

On 6/9/10, Ashar Siddiqui siddas...@gmail.com wrote:
 Hello,



 I have an issue with one of our customer where the call diversion is not
 working properly.

 The incoming number (352999) to extension 2602 after 1800hrs.



 Ext. 2602 has voicemail setup on it, which works internally and also when
 dialled as DDI - but this does not work when calls are diverted to it from
 352999.



 I think it's a Unity issue that it gets back to the original called party
 for voicemail. Any workaround that it should go to voicemail of 2602?



 Thanks









-- 
Sent from my mobile device

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Re: [OSL | CCIE_Voice] CUE integration with CCM problems

2010-06-09 Thread bkvalent...@gmail.com
Do your phones and gateways have the pt-vm partition in their css?  

Brian

Sent from my Verizon Wireless Phone

- Reply message -
From: Pavan K pav.c...@gmail.com
Date: Wed, Jun 9, 2010 9:14 pm
Subject: [OSL | CCIE_Voice] CUE integration with CCM problems
To: osl osl ccie_voice@onlinestudylist.com

If i place all my CTI ports in the NULL partition everything works, If i put
them in PT-VM i get BUSY (CTI rejecting call).

Any additional CSS settings needed ?

=
Placed all my CTI Ports in PT-VM
Placed all my CTI Routepoints in Null partition.

CSS for Routepoints and Ports is CSS-VM { PT-VM + PT-Phones }
Doesn't work
==










-- 
- Pavan
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Re: [OSL | CCIE_Voice] CCIE voice workbook 1

2010-06-08 Thread bkvalent...@gmail.com
On the hwic ports on the router, you should use the trunk method.  

On the 3750, you should use the access port method.  

Brian

Sent from my Verizon Wireless Phone

- Reply message -
From: Dew Swen dew.s...@gmail.com
Date: Tue, Jun 8, 2010 9:19 am
Subject: [OSL | CCIE_Voice] CCIE voice workbook 1
To: ccie_voice@onlinestudylist.com

I used the trunk method on the same lab a couple of days ago. It was 
just ok.

-- 
Dew Swen
CCVP, CCNP


08.06.2010 16:15, rsmail...@solcon.nl yazmış:
 Hello,
   i am preparing for the CCIE voice Lab. in the Lab1A solution the
 switchports are configured like: int fa0/3/0 switchport trunk native vlan
 200 switchport mode trunk switchport voice vlan 400 should it be:
 switchport access vlan 200 switchport voice vlan 400 switchport mode
 access i know the trunk config is the old way. or are the 2 correct on
 the Lab exam. in the Lab3A the sip phone should use RFC2833 dtmf in the
 solution it is: dtmf-relay sip-notify should be: dtmf-relay rtp-nte or
 not ? regards Ron Schutte

 regards Ron

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Re: [OSL | CCIE_Voice] Passed CCIE Voice ( # 26129 )

2010-06-05 Thread bkvalent...@gmail.com
Congrats!

Sent from my Verizon Wireless Phone

- Reply message -
From: wolfsrudel wolfsru...@gmail.com
Date: Sat, Jun 5, 2010 11:32 am
Subject: [OSL | CCIE_Voice] Passed CCIE Voice ( # 26129 )
To: Moataz Mamdouh moataz_m...@yahoo.com
Cc: ccie_voice@onlinestudylist.com


congratulations moataz!!!
time to relax now ;-)

On Sat, Jun 5, 2010 at 12:02 PM, Moataz Mamdouh moataz_m...@yahoo.comwrote:

 ALL
 I passed the exam from the first attempt , thank you all for you help ,
 really i learned a lot from
 you.

 Regards
 Moataz Mamdouh Tolba
 SEEGYPT
 Website:www.seegypt.com
 Technical Support Engineer
 CCIE # 26129





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Re: [OSL | CCIE_Voice] VLAN interfaces down

2010-06-04 Thread bkvalent...@gmail.com
Also if the vlans are disabled.  Make sure the vlans are active.

Sent from my Verizon Wireless Phone

- Reply message -
From: Amy Ryan ar...@ipexpert.com
Date: Fri, Jun 4, 2010 7:33 am
Subject: [OSL | CCIE_Voice] VLAN interfaces down
To: Angel Perez gorr...@hotmail.com, amccar...@cciequest.com, 
ciscovoiceg...@gmail.com
Cc: osl osl ccie_voice@onlinestudylist.com


Matthew, 

Did you do a ³sh cdp neighber² and verify what port the phone was plugged
into?  I did notice below that you are showing the configuration for int
fa1/1 and the Proctor Lab Racks usually have the phone at BR1 plugged into
interface fa1/0 which is also not shown in your output as being set up.  If
the phone was plugged into interface fa1/0 and was not configured properly
or shut, you would have this behavior.

If the vlan.dat file was not present in flash and the VLAN¹s were configured
as you have shown below, then they would not show up when you do a ³sh
vlan-s bri², but they would show up in a ³sh ip int bri².  See ScreenShots:

BR1-RTR#sh vlan-s bri (with vlan.dat deleted from flash)

VLAN Name StatusPorts
  -
---
1default  activeFa1/1, Fa1/2, Fa1/3, Fa1/4,
Fa1/5
Fa1/6, Fa1/7, Fa1/8, Fa1/9,
Fa1/10
Fa1/11, Fa1/12, Fa1/13,
Fa1/14, Fa1/15
1002 fddi-default act/unsup
1003 token-ring-default   act/unsup
1004 fddinet-default  act/unsup
1005 trnet-defaultact/unsup
!
!
BR1-RTR#sh ip int bri
Interface  IP-Address  OK? Method Status
Protocol
...
Vlan13010.10.101.1 YES NVRAM  up
down
Vlan24010.10.201.1 YES NVRAM  up
down 


I will be around on Sunday.  If you are able to recreate this then, contact
me and I can hop into your session and look around.

Thank you,
Amy

---
Amy Ryan ­ CCIE #24677 (Voice)
Technical Instructor - IPexpert, Inc.
Mailto: ar...@ipexpert.com
Telephone: +1.810.326.1444
Live Assistance, Please visit: www.ipexpert.com/chat
http://www.ipexpert.com/chat
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IPexpert is a premier provider of Self-Study Workbooks, Video on Demand,
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public website at www.ipexpert.com http://www.ipexpert.com/




From: Angel Perez gorr...@hotmail.com
Date: Fri, 4 Jun 2010 08:03:41 +
To: amccar...@cciequest.com, ciscovoiceg...@gmail.com
Cc: osl osl ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] VLAN interfaces down

Hi:
 
If the vlan.dat file is deleted you will get this result
 
Make sure that the vlan exists and also that it is active:
 
vlan 130
create 
name data
status active
 
vlan 240
create
name voice
status active
 
hth
 
 Date: Thu, 3 Jun 2010 19:08:22 -0400
 From: amccar...@cciequest.com
 To: ciscovoiceg...@gmail.com
 CC: ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] VLAN interfaces down
 
 Hey Bro,
 I ran into an issue similar to that before but mine was because there
 was no phone connected to the port.
 
 Antonio
 
 Quoting Matthew Berry ciscovoiceg...@gmail.com:
 
  I see this issue from time to time. The VLAN interfaces on my BR1-RTR
  show a state of up, but line protocol is down.
 
  I made sure that there are ports with the vlans configured. I reload
  the router. I also made sure the vlans were in existence.
 
  At this point, my BR1-RTR is useless until I get this working. Any ideas?
 
  interface Vlan130
  ip address 10.10.101.1 255.255.255.0
  !
  interface Vlan240
  ip address 10.10.201.1 255.255.255.0
 
  ...
 
  interface FastEthernet1/1
  switchport trunk native vlan 130
  switchport mode trunk
  switchport voice vlan 240
 
  
 
  BR1-RTR#show vlan-switch br
 
  VLAN Name Status Ports
    -
  ---
  ...
  130 DATA active Fa1/1, Fa1/15
  240 PHONES active Fa1/1, Fa1/2, Fa1/3, Fa1/4
  Fa1/5, Fa1/6, Fa1/7, Fa1/8
  Fa1/9, Fa1/10, Fa1/11, Fa1/12
  Fa1/13, Fa1/14, Fa1/15
 
  
 
  BR1-RTR#show ip int bri
  Interface IP-Address OK? Method Status
  Protocol
  ...
  Vlan130 10.10.101.1 YES manual up
  down
  Vlan240 10.10.201.1 YES NVRAM up
  down
 
  -- 
 
  *Matthew Berry*
 
  /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/
 
  *_Vitals:_*
 
  *GVoice: *+1.612.424.5044
 
  *Gmail*: ciscovoiceg...@gmail.com
 
  *Skype*: ciscovoiceguru
 
  *Twitter*: ciscovoiceguru
 
  *_Cert Stats:_*
 
  Cisco Cert Journey Began: Jan 1, 2009
 
  1st Lab Attempt: Aug 16, 2010
 
 
 
 

Re: [OSL | CCIE_Voice] Direct transfer to Original called p arty Voicemail

2010-05-26 Thread bkvalent...@gmail.com
Just give the phones the trnsfvm softkey.  

Brian

- Reply message -
From: Ashar Siddiqui siddas...@gmail.com
Date: Wed, May 26, 2010 7:03 am
Subject: [OSL | CCIE_Voice] Direct transfer to Original called party
Voicemail
To: r.ochi...@mfient.com
Cc: 'osl osl' ccie_voice@onlinestudylist.com


So you mean in CUE, I just have to assign alternate extension for every user
starting with 6 like 62904, 62905, 62906 ...

If x2905 transfer the call to 62904, would it go straight to VM for 2904 or
will it first ring for 10s and then go to voicemail?

Do I have to create ephone-dn for all of these? (remember customer has 100+
users and dn)

 

Thanks for your help

 

Ash

 

From: Rogers Ochieng [mailto:r.ochi...@mfient.com] 
Sent: 26 May 2010 10:57
To: 'Ashar Siddiqui'
Subject: RE: [OSL | CCIE_Voice] Direct transfer to Original called party
Voicemail

 

I'm thinking secondary number in CUE for the user say 62904 and you route
that to CUE so 6 can be your assumed prefix for diverting calls to CUE for
other subscriber

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ashar Siddiqui
Sent: Wednesday, May 26, 2010 12:20 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Direct transfer to Original called party
Voicemail

 

Hello all,

 

One of my customer is interested in direct transfer of an incoming call to
voicemail after it has been picked up by someone else in the pickup group.

 

For e.g. If a call comes in and ring x2904 but he is not available, person
at x2905 picks up the call but the calling party wants to leave a VM for
x2904. How the person at x2905 can direct transfer the call to x2904
voicemail.

 

One way is to transfer the call back to x2904 which will ring and ring for
10s and then go to voicemail. This is not what they want. They want the
ability to transfer the call directly to voicemail of Original called party.

 

 

ephone-dn  1  octo-line

 number 2904

 pickup-group 1

 label Tim Flynn (2904)

name Tim Flynn

 call-forward busy 8005

 call-forward noan 8005 timeout 10

 corlist incoming User-international

!

!

ephone-dn  2  octo-line

 number 2905

 pickup-group 1

 label Steve Zander (2905)

  name Steve Zander

 call-forward busy 8005

 call-forward noan 8005 timeout 10

 corlist incoming User-international

!

!

 

 

 

Ash

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Re: [OSL | CCIE_Voice] Unable to Bind L3 to CCM

2010-05-21 Thread bkvalent...@gmail.com
You forgot service mgcp at the end of your pri-group command in controller 
config. 

Sent from my Verizon Wireless Phone

- Reply message -
From: Jeff Price (jeffpric) jeffp...@cisco.com
Date: Fri, May 21, 2010 6:55 pm
Subject: [OSL | CCIE_Voice] Unable to Bind L3 to CCM
To: CCIE Voice Maillist ccie_voice@onlinestudylist.com

Hey everyone,

 

Have you ever seen a situation where you can register a MGCP GW to CUCM
but you are unable to bind L3 to CCM in IOS?

 

Here's what I see:

 

R1(config-if)#isdn bind-l3 ?

  q931  Select IOS Q.931

 

R1(config-if)#   

 

Here is my config:

 

R1#show run

Building configuration...

 

 

Current configuration : 3127 bytes

!

version 12.4

service timestamps debug datetime msec

service timestamps log datetime msec

service password-encryption

!

hostname R1

!

boot-start-marker

boot system flash:c2800nm-adventerprisek9_ivs_li-mz.124-24.T.bin

boot-end-marker

!

logging message-counter syslog

enable password 7 110A1A0C12

!

no aaa new-model

network-clock-participate wic 0 

!

dot11 syslog

ip source-route

!

! 

ip cef

ip dhcp excluded-address 10.5.200.1

!

ip dhcp pool HQ_PHONES

   network 10.5.200.0 255.255.255.0

   option 150 ip 172.21.51.204 

   default-router 10.5.200.1 

!

!

no ipv6 cef

!

multilink bundle-name authenticated

!

!

!

!

isdn switch-type primary-ni

!

!

!

!

!

! 

!

!

!

!

!

!

!

!

!

!

!

!

!

voice-card 0

!

!

!

!

!

archive

 log config

  hidekeys

! 

!

!

!

!

controller T1 0/0/0

 pri-group timeslots 1-3,24

!

controller T1 0/0/1

!

!

!

!

!

interface GigabitEthernet0/0

 description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$

 no ip address

 duplex auto

 speed auto

!

interface GigabitEthernet0/0.102

 encapsulation dot1Q 102 native

 ip address 172.21.51.196 255.255.255.224

! 

interface GigabitEthernet0/0.150

 encapsulation dot1Q 150

 ip address 10.5.100.1 255.255.255.0

!

interface GigabitEthernet0/0.250

 encapsulation dot1Q 250

 ip address 10.5.200.1 255.255.255.0

!

interface GigabitEthernet0/1

 no ip address

 shutdown

 duplex auto

 speed auto

!

interface Serial0/0/0:23

 no ip address

 encapsulation hdlc

 isdn switch-type primary-ni

 isdn incoming-voice voice

 isdn outgoing display-ie

 no cdp enable

!

interface Serial0/1/0

 no ip address

 encapsulation frame-relay IETF

 tx-ring-limit 128

 tx-queue-limit 128

 serial restart-delay 0

 frame-relay lmi-type ansi

!

interface Serial0/1/0.1 point-to-point

 ip address 162.5.101.1 255.255.255.0

 ip ospf mtu-ignore

 frame-relay interface-dlci 201   

!

interface Serial0/1/0.2 point-to-point

 ip address 162.5.102.1 255.255.255.0

 ip ospf mtu-ignore

 frame-relay interface-dlci 202   

!

router ospf 1

 log-adjacency-changes

 network 10.5.100.0 0.0.0.255 area 0

 network 10.5.200.0 0.0.0.255 area 0

 network 162.5.101.0 0.0.0.255 area 0

 network 162.5.102.0 0.0.0.255 area 0

 network 172.5.100.0 0.0.0.255 area 0

 network 172.21.51.0 0.0.0.255 area 0

!

ip forward-protocol nd

ip route 0.0.0.0 0.0.0.0 172.21.51.193

ip http server

no ip http secure-server

!

!

!

nls resp-timeout 1

cpd cr-id 1

!

!

!

!

!

!

control-plane

!

!

!

voice-port 0/0/0:23

!

ccm-manager redundant-host 172.21.51.204

ccm-manager mgcp

!

mgcp

mgcp call-agent 172.21.51.205 service-type mgcp version 0.1

mgcp fax t38 ecm

mgcp bind control source-interface GigabitEthernet0/0.250

mgcp bind media source-interface GigabitEthernet0/0.250

!

mgcp profile default

!

!

!

!

!

!

gatekeeper

 shutdown

!

line con 0

 exec-timeout 0 0

 logging synchronous

 terminal-type mon

line aux 0

line vty 0 4

 exec-timeout 0 0

 password 7 110A1A0C12

 logging synchronous

 login

 terminal-type mon

!

scheduler allocate 2 1000

end 

 

 

Here is the status of CCM registration:

R1(config)#do show ccm

MGCP Domain Name: R1

PriorityStatus   Host



Primary Registered   172.21.51.205

First BackupBackup Ready 172.21.51.204

Second Backup   None 

 

Current active Call Manager:172.21.51.205

Backhaul/Redundant link port:   2428

Failover Interval:  30 seconds

Keepalive Interval: 15 seconds

Last keepalive sent:00:03:46 UTC May 22 2010 (elapsed time:
00:00:06)

Last MGCP traffic time: 00:03:46 UTC May 22 2010 (elapsed time:
00:00:06)

Last failover time: 23:59:51 UTC May 21 2010 from
(172.21.51.205)

Last switchback time:   00:00:21 UTC May 22 2010 from
(172.21.51.204)

Switchback mode:Graceful

MGCP Fallback mode: Not Selected

Last MGCP Fallback start time:  None

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