Re: [OSL | CCIE_Voice] SRST with Unity Connections

2012-01-06 Thread Matthew Saskin
Check out the use of DTMF patterns in the following document:
http://www.cisco.com/en/US/docs/voice_ip_comm/cusrst/admin/sccp_sip_srst/configuration/guide/srst_voicemail.html

-matthew

On Fri, Jan 6, 2012 at 6:20 AM, datucha123 datucha123
datucha...@gmail.comwrote:


 Hello,

 As I gues there are following ways to integrate SRST with Unity Connection
 through PRI:

 1) when the IP phone are in SRST mode, configure the *isdn outgoing ie
 redirecting-number*, so that the RDNIS will be send over PSTN Route to
 HQ Unity Connection, and it can select the correct Mailbox.

 But I have an issue with the following  -  Let the Users press the
 Voicemail Button on their IP Phones in SRST mode and hear their messages.

 Well, for that that I have to configure the VM Pilot number in
 call-manager-fallback (or CME SRST) as the PSTN Global number. But when
 that call will come to Unity, (where the Users are present with 4 digit
 extensions), how to strip unnecessary digits from ANI, so that the Unity
 Connection can select the correct users mailbox?

 VM Profile does not work in that case any more. And the VM Hunt Pilot,
 setting for Calling Number Translation to  is not a good idea.

 As I know, there are several ways:

 1) Configure the Secondary Extension for those users as global PSTN Number
 2)  Or use the Initial Unity Connection with Global Numbers for users, and
 have the VM Profile which will expand the 4 digits extension to respective
 Global PSTN Numbers.

 Is there another/better way?

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Re: [OSL | CCIE_Voice] CUE License

2012-01-06 Thread Matthew Saskin
Agree with the earlier messages.  What you're describing is not a real
world scenario in any way.  In the off-chance that something this obscure
is asked as part of the CCIE lab, I'd suggest that you would be OK to use
either CUE license unless you are explicitly told otherwise, or ask the
proctor.

-matthew

On Fri, Jan 6, 2012 at 2:11 PM, Bill Lake whl...@gmail.com wrote:

 I do not think that using you want to try to use CUE as backup as I
 believe that would only allow you access to VM stored on CUE when in SRST.

 I would think that you would prefer to manipulate the dial plan under SRST
 to use the PSTN to get calls to your UCEx VM.

 On Fri, Jan 6, 2012 at 12:10 PM, datucha123 datucha123 
 datucha...@gmail.com wrote:


 Ok, thank you very much for your answer.

 But now look, If I have CUCM license in CUE, and the Router goes to SRST
 mode, that license will work for SRST IP Phones through SIP Notify or
 Subscribe MWI option.

 If we are using CUCME License in CUE, it also works for SRST IP Phones
 either through SIP notify or Subscribe MWI.

 Now imagine, that we have CUCM with Unity Connection at HQ, and the
 Remote Branch Router is having CUE module. *But the CUCM does not use
 this CUE module for Voice Mail.* CUE Voice Mail system is going to be
 used only when the Remote Branch goes into SRST.
  So if we upload the CUCM license, there will be no problem, as this
 license can work for SRST IP Phones.
 Also if we upload the CUCME License, it will also work, as there is no
 need to register CUE with CUCM.

 So which License is better to use in such case? I am not using either
 CUCM or CUCME, I am using only SRST for CUE.
 On Fri, Jan 6, 2012 at 7:48 PM, Rrcrumm rrcr...@yahoo.com wrote:

 If you have CME then CME, if you have CUCM then CUCM license.

 SRST does not dictate the license.

 Rc

 Sent from my iPhone

 On Jan 6, 2012, at 4:02 AM, datucha123 datucha123 datucha...@gmail.com
 wrote:

  When configuring CUE for SRST (Traditional or CME), and assuming that
 CUE will not operate for CUCM -  it will operate only for SRST, which
 license is better to upload to CUE? CUCM and CCME?
 
  Because, for SRST both license works fine.
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 please visit www.ipexpert.com
 
  Are you a CCNP or CCIE and looking for a job? Check out
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Re: [OSL | CCIE_Voice] New Lab #2 - CUE ntp

2012-01-04 Thread Matthew Saskin
Also, make sure that 10.10.110.3 is indeed acting as an NTP server.  If
it's a router, ntp master and ntp source to the interface that has the
IP 10.10.110.3, etc.

I know that I typically forget to put the static route back to the CUE
module since you can get session/console access with the service-module
ip/default gateway set, but no external traffic without the static route.

-matthew

On Wed, Jan 4, 2012 at 3:14 PM, datucha123 datucha123
datucha...@gmail.comwrote:

 ping 10.10.110.3   -  and see if it pings.

 then take a look at the Integrated Service Engine Interface, to see the
 default gateway set.

 On Thu, Jan 5, 2012 at 12:03 AM, Rrcrumm rrcr...@yahoo.com wrote:

  What commands?

 Sent from my iPhone

 On Jan 4, 2012, at 11:53 AM, datucha123 datucha123 datucha...@gmail.com
 wrote:

  Verify that CUE can reach the NTP IP Address. and the CUE module has
 the default gateway set

 On Wed, Jan 4, 2012 at 11:27 PM, Randall Crumm rrcr...@yahoo.com wrote:

  HI,
 I am working on the new lab #2.

 When configuring CUE the question asks for the primary NTP server to be
 10.10.110.3. The default is lo1, 10.10.115.1

 When I enter that IP address, I get an error :

  I could not reach 10.10.110.3 using NTP.
  If 10.10.110.3 really is an NTP server, then
  there may be a firewall blocking NTP packets.
  NTP uses UDP port 123.


 Any thoughts?


 Thanks,
 Randall



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Re: [OSL | CCIE_Voice] New Lab #2 - CUE ntp

2012-01-04 Thread Matthew Saskin
Is ntp source set on the router to the interface with the IP you're trying
to pull NTP from?

-matthew

On Wed, Jan 4, 2012 at 4:53 PM, Rrcrumm rrcr...@yahoo.com wrote:

 Here are my NTP commands on the router
 Ntp server 10.10.110.1
 NTP master 9


 I did confirm the default route and static route

 Rc

 Sent from my iPhone

 On Jan 4, 2012, at 1:41 PM, Matthew Saskin m...@saskin.net wrote:

 Also, make sure that 10.10.110.3 is indeed acting as an NTP server.  If
 it's a router, ntp master and ntp source to the interface that has the
 IP 10.10.110.3, etc.

 I know that I typically forget to put the static route back to the CUE
 module since you can get session/console access with the service-module
 ip/default gateway set, but no external traffic without the static route.

 -matthew

 On Wed, Jan 4, 2012 at 3:14 PM, datucha123 datucha123 
 datucha...@gmail.com wrote:

 ping 10.10.110.3   -  and see if it pings.

 then take a look at the Integrated Service Engine Interface, to see the
 default gateway set.

 On Thu, Jan 5, 2012 at 12:03 AM, Rrcrumm rrcr...@yahoo.com wrote:

  What commands?

 Sent from my iPhone

 On Jan 4, 2012, at 11:53 AM, datucha123 datucha123 datucha...@gmail.com
 wrote:

  Verify that CUE can reach the NTP IP Address. and the CUE module has
 the default gateway set

 On Wed, Jan 4, 2012 at 11:27 PM, Randall Crumm rrcr...@yahoo.comwrote:

  HI,
 I am working on the new lab #2.

 When configuring CUE the question asks for the primary NTP server to
 be 10.10.110.3. The default is lo1, 10.10.115.1

 When I enter that IP address, I get an error :

  I could not reach 10.10.110.3 using NTP.
  If 10.10.110.3 really is an NTP server, then
  there may be a firewall blocking NTP packets.
  NTP uses UDP port 123.


 Any thoughts?


 Thanks,
 Randall



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 please visit www.ipexpert.com

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




 ___
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 visit www.ipexpert.com

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 www.PlatinumPlacement.com



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

2011-12-28 Thread Matthew Saskin
The initial SIP offer specifies G.729 for media, which the (original)
dial-peer can't negotiate since it is fixed at g.711.  After setting an
appropriate inbound dial-peer, even though G711 is the primary choice in
the voice-class, G.729 is what is negotiated since the initial INVITE
specifies G729.

-matthew

On Wed, Dec 28, 2011 at 1:26 PM, John McGaughey (jomcgaug) 
jomcg...@cisco.com wrote:

 Yep it is, but it still works.  I think because the inbound dial-peer
 still has g729 in it even though it's not the 1st choice.

 John

 -Original Message-
 From: brajesh kumaR [mailto:brjku...@gmail.com]
 Sent: Wednesday, December 28, 2011 11:58 AM
 To: datucha123 datucha123
 Cc: John McGaughey (jomcgaug); ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] Calls to CUE fail over WAN

 But as per voice class codec still preference one is g711ulaw codec
 which is on incoming dial-peer same codec as outbound peer.



 On Wed, Dec 28, 2011 at 2:00 PM, datucha123 datucha123
 datucha...@gmail.com wrote:
  Nice to hear that you have solved the issue.
 
  The solution is correct. When you a have a Voice-Class Codec that has the
  G711 in it, then the Xcoder is not invoked as the G711 is supported by
 the
  incoming dial-peer.
 
  In such cases, you have to force the incoming dial-peer from CUCM (or any
  other sources) to use the G729, so that xcoder will get invoked.
 
  On Wed, Dec 28, 2011 at 4:44 AM, John McGaughey (jomcgaug)
  jomcg...@cisco.com wrote:
 
  Nevermind.  I figured it out.  I created a separate inbound dial-peer
 with
  g729 and that caused the xcoder to get invoked.
 
 
 
  voice class codec 1
 
  codec preference 1 g711ulaw
 
  codec preference 2 g729r8
 
 
 
  dial-peer voice 36002 voip
 
  voice-class codec 1
 
  incoming called-number 3600
 
  dtmf-relay rtp-nte
 
  no vad
 
 
 
  dial-peer voice 3600 voip
 
  destination-pattern 3600
 
  session protocol sipv2
 
  session target ipv4:10.10.202.2
 
  dtmf-relay sip-notify
 
  codec g711ulaw
 
  no vad
 
 
 
  Before I had it with only this dial-peer.
 
 
 
  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
 
 
 
  I guess if the inbound dial-peer can't satisfy the codec it fails.  But
 if
  it does, and the outbound dial-peer does not, then it invokes the
 xcoder.
 
 
 
  John
 
 
 
 
 
  From: bkvalent...@gmail.com [mailto:bkvalent...@gmail.com]
  Sent: Tuesday, December 27, 2011 6:42 PM
  To: John McGaughey (jomcgaug); ccie_voice@onlinestudylist.com
  Subject: Re: [OSL | CCIE_Voice] Calls to CUE fail over WAN
 
 
 
  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 

Re: [OSL | CCIE_Voice] Emergency Calls (911 or 999)

2011-12-08 Thread Matthew Saskin
In practice, the PS-ALI database contains everything the PSAP needs (in
theory), hence no need for calling name to go with the call.

In the lab, do only what is asked, nothing more, nothing less.  If there is
no explicit requirement to ensure calling party name is displayed, don't
worry about it.  Remember, it's not a test of best practices or real-world
situations.

-matthew

On Wed, Dec 7, 2011 at 11:52 PM, Ken Wyan kew...@gmail.com wrote:

 Experts,

 I want to know best practice of sending caller-name for emergency calls.

 Usually Telco doesn't forward calling party name to PSAP  it sends
 calling number only. PSAP uses ALI database (which is seperate  secured
 from PSTN) to find the location of caller.

  Is there a specific reason ( regulatory or practical reason ) why telco
 strip caller-name for 911/999 calls?

 In our CUCM or CCME in exam , should we strip caller-name of Emergency
 calls ?  ( if so , additional configuration step needed)

 K.W.










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www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] CCIE Voice Lab Guidance Req !

2011-12-07 Thread Matthew Saskin
I'm using a combination of VM's, physical phones, and dynamips to run
virtual routers for my home lab - about the only things that can't really
be done with dynamips are some QOS, CUE, T1's, and media resources, but
that's about it.  My total investment in my home lab is well under $1500
and I can probably do 75-80% of the blueprint material.

5 x phones - 7960s/7961's
VMware server - 2 x quad core athlon processors, 16gb ram, 2x750gb hdd,
ESXi 4.0

On that VMware server I'm running a full CCIE lab (pub/sub/cups/uccx/ucxn),
as well as 4 x XP desktops for IP communicator, CAD installed on a few as
well as a virtualized linux server running 4 virtual routers under
dynamips.

-matthew


On Wed, Dec 7, 2011 at 11:37 AM, Bill Lake whl...@gmail.com wrote:

 I think the least expensive and less time consuming method is to use
 remote rack rental. They have all the hardware needed to practice.  That
 said, plan ahead and have different vendors so that if say Remote lab A is
 down, you can schedule with remote lab b or c and not waste your study
 time.  Also make sure they support remote phone connections so that you can
 use your local resources (cisco router and phones) to connect them in to
 have hands on phone use.

 That all said a Pod is nice as you can check things out and test anytime.
 Here are my recommendation to save money and build a pod.

 I am running all my servers on a VMWare EXSi 5  server w/i7 870, 8 GB
 RAM, 4 HDD, 2 NIC's and using resource pooling.

 From other post with edits:

 - PSTN phone can be 7960 so you can save big there!  3 or more 7961 or
 7962 Phones should be fine, use soft phones or move phones from pod to pod
 (ie move CUCM hardware phone to CME when needed for testing)
 - 1 2801 w/256MB RAM,w/128MB Flash, 1 PVDM2-16, 1 VWIC-2MFT-T1, 1
 HWIC-4ESW (this will be BR1)
 - 1 2801 w/256MB RAM,w/128MB Flash, 1 PVDM2-16, 1 VWIC-2MFT-T1  (this will
 be HQ)
 - 1 2801 w/256MB RAM,w/128MB Flash, 1 PVDM2-16, 2 VWIC-2MFT-T1, 1
 VWIC-2MFT-E1(this will be PSTN)
 - 1 3560 or 3750 switch
 - NM-CUE or AIM-CUE, NM is more powerful but AIM is cheaper. (this will go
 into your 2821)
 - 1 VWIC-2MFT-T1/E1 (this will go into your 2821)
 - 1 HWIC-4ESW (this will go into your 2821)

 You save money using the VWIC-2MFT-T1 and VWIC-2MFT-E1 over the
 programable ones, you just have to remember that you need to know the
 commands for programing the card just in case.

 You will want more PVDM module(s) for testing but you can get by at first
 with this. Also you can shut down one router and move PVDM over to do the
 more demanding testing.

 You can also get by with 64MB of Compact Flash for most of the routers,
 just look at the requirements of the IOS you want to install.  You will
 need more for the CME router.

 Some will say you can't use 2801 but I am currently running two in my pod
 (including my PSTN) and it is working well.



 Hope this helps some


 On Wed, Dec 7, 2011 at 7:50 AM, Michael Miller kf4...@gmail.com wrote:

 The latest list of hardware and software that you'll see in the lab can
 be found on the Cisco Learning Network.

 https://learningnetwork.cisco.com/docs/DOC-5292

 Thanks,

 Michael

 On Wed, Dec 7, 2011 at 11:46 AM, Muhammad Zubair 
 muhammad.zub...@gmail.com wrote:

 My Friend
 the link you send to me ? is that updated one... in software section it
 is showing CCM version 4.1(3) with latest SR ? Can you believe that
 CCIE Logo is new but . interesting

 the things which you wrote is like a sea. ok any idea how much it cost
 total. where to start and how to use that efficiently ?
 confused totally

 On Wed, Dec 7, 2011 at 12:22 PM, AJ BG ciscoie2...@gmail.com wrote:

  Muhammad,

 · Stay away from GNS3, you will end up troubleshooting GNS3
 instead of troubleshooting real voice issues.

 · You will need a server with at least 16GB ram; don’t forget
 that you have to have 5 virtual servers.  You don’t need FXO for the
 current version. I agree with Emanuel,  I would get four 2800 series
 routers, you would need to have PVDMS, T1 and E1 cards, CUE module, switch
 module, a layer 3 switch, and at least 6 type B phones.  You can get
 it a little bit cheaper from ebay, or closely follow this group study
 treads and time to time people sell their equipment all at once maybe you
 can bargain and get it really cheap.

 · If you haven’t seen the CCIE blue print yet. I suggest you
 check the following link. You will have a good idea what you will need
 after you checked the blueprint.
 cisco.com/web/learning/le3/ccie/voice/lab_equipment.html
 · This is one investment that can really pay off .
 Thanks, Joe


 On Tue, Dec 6, 2011 at 9:39 PM, Muhammad Zubair 
 muhammad.zub...@gmail.com wrote:

 Emanuel

 i m scared buddy, i think 3rd world war started for me.
 right now i have 2821 with FXO card n another 2821 without. also i can
 arrange one or 2 2600 series.
 before that i had setup my virtual machines on my laptop which has 8Gb
 ram and simulation on 

Re: [OSL | CCIE_Voice] CCIE Voice Lab Guidance Req !

2011-12-07 Thread Matthew Saskin
Of course.  Speaking personally, I know enough about CUE and PSTN
connectivity that I don't need to spend time focusing on my home lab.  I
basically use my home lab for day-to-day studying, testing out scenarios,
etc.  I also use remote labs (via proctorlabs) for full lab simulations.


On Wed, Dec 7, 2011 at 1:38 PM, datucha123 datucha123
datucha...@gmail.comwrote:

 QoS, CUE T1 and Media Resources are very very important on the exam.
 T1 and Media Resources are one of the Major requirements on the exam, I
 think.

 On Wed, Dec 7, 2011 at 9:41 PM, Matthew Saskin m...@saskin.net wrote:

 I'm using a combination of VM's, physical phones, and dynamips to run
 virtual routers for my home lab - about the only things that can't really
 be done with dynamips are some QOS, CUE, T1's, and media resources, but
 that's about it.  My total investment in my home lab is well under $1500
 and I can probably do 75-80% of the blueprint material.

 5 x phones - 7960s/7961's
 VMware server - 2 x quad core athlon processors, 16gb ram, 2x750gb hdd,
 ESXi 4.0

 On that VMware server I'm running a full CCIE lab
 (pub/sub/cups/uccx/ucxn), as well as 4 x XP desktops for IP communicator,
 CAD installed on a few as well as a virtualized linux server running 4
 virtual routers under dynamips.

 -matthew



 On Wed, Dec 7, 2011 at 11:37 AM, Bill Lake whl...@gmail.com wrote:

 I think the least expensive and less time consuming method is to use
 remote rack rental. They have all the hardware needed to practice.  That
 said, plan ahead and have different vendors so that if say Remote lab A is
 down, you can schedule with remote lab b or c and not waste your study
 time.  Also make sure they support remote phone connections so that you can
 use your local resources (cisco router and phones) to connect them in to
 have hands on phone use.

 That all said a Pod is nice as you can check things out and test
 anytime.  Here are my recommendation to save money and build a pod.

 I am running all my servers on a VMWare EXSi 5  server w/i7 870, 8 GB
 RAM, 4 HDD, 2 NIC's and using resource pooling.

 From other post with edits:

 - PSTN phone can be 7960 so you can save big there!  3 or more 7961 or
 7962 Phones should be fine, use soft phones or move phones from pod to pod
 (ie move CUCM hardware phone to CME when needed for testing)
 - 1 2801 w/256MB RAM,w/128MB Flash, 1 PVDM2-16, 1 VWIC-2MFT-T1, 1
 HWIC-4ESW (this will be BR1)
 - 1 2801 w/256MB RAM,w/128MB Flash, 1 PVDM2-16, 1 VWIC-2MFT-T1  (this
 will be HQ)
 - 1 2801 w/256MB RAM,w/128MB Flash, 1 PVDM2-16, 2 VWIC-2MFT-T1, 1
 VWIC-2MFT-E1(this will be PSTN)
 - 1 3560 or 3750 switch
 - NM-CUE or AIM-CUE, NM is more powerful but AIM is cheaper. (this will
 go into your 2821)
 - 1 VWIC-2MFT-T1/E1 (this will go into your 2821)
 - 1 HWIC-4ESW (this will go into your 2821)

 You save money using the VWIC-2MFT-T1 and VWIC-2MFT-E1 over the
 programable ones, you just have to remember that you need to know the
 commands for programing the card just in case.

 You will want more PVDM module(s) for testing but you can get by at
 first with this. Also you can shut down one router and move PVDM over to do
 the more demanding testing.

 You can also get by with 64MB of Compact Flash for most of the routers,
 just look at the requirements of the IOS you want to install.  You will
 need more for the CME router.

 Some will say you can't use 2801 but I am currently running two in my
 pod (including my PSTN) and it is working well.



 Hope this helps some


 On Wed, Dec 7, 2011 at 7:50 AM, Michael Miller kf4...@gmail.com wrote:

 The latest list of hardware and software that you'll see in the lab can
 be found on the Cisco Learning Network.

 https://learningnetwork.cisco.com/docs/DOC-5292

 Thanks,

 Michael

 On Wed, Dec 7, 2011 at 11:46 AM, Muhammad Zubair 
 muhammad.zub...@gmail.com wrote:

 My Friend
 the link you send to me ? is that updated one... in software section
 it is showing CCM version 4.1(3) with latest SR ? Can you believe that
 CCIE Logo is new but . interesting

 the things which you wrote is like a sea. ok any idea how much it cost
 total. where to start and how to use that efficiently ?
 confused totally

 On Wed, Dec 7, 2011 at 12:22 PM, AJ BG ciscoie2...@gmail.com wrote:

  Muhammad,

 · Stay away from GNS3, you will end up troubleshooting GNS3
 instead of troubleshooting real voice issues.

 · You will need a server with at least 16GB ram; don’t
 forget that you have to have 5 virtual servers.  You don’t need FXO
 for the current version. I agree with Emanuel,  I would get four
 2800 series routers, you would need to have PVDMS, T1 and E1 cards, CUE
 module, switch module, a layer 3 switch, and at least 6 type B phones.
  You can get it a little bit cheaper from ebay, or closely follow
 this group study treads and time to time people sell their equipment all 
 at
 once maybe you can bargain and get it really cheap.

 · If you haven’t seen the CCIE blue

Re: [OSL | CCIE_Voice] CCIE Voice Lab Guidance Req !

2011-12-07 Thread Matthew Saskin
I use virtual audio cable (http://software.muzychenko.net/eng/vac.htm) to
create a dummy audio driver within windows XP.  This will let you run
applications that require audio (eg; IP communicator) under VMware.

On Wed, Dec 7, 2011 at 1:49 PM, Matthew Saskin m...@saskin.net wrote:

 Of course.  Speaking personally, I know enough about CUE and PSTN
 connectivity that I don't need to spend time focusing on my home lab.  I
 basically use my home lab for day-to-day studying, testing out scenarios,
 etc.  I also use remote labs (via proctorlabs) for full lab simulations.



 On Wed, Dec 7, 2011 at 1:38 PM, datucha123 datucha123 
 datucha...@gmail.com wrote:

 QoS, CUE T1 and Media Resources are very very important on the exam.
 T1 and Media Resources are one of the Major requirements on the exam, I
 think.

 On Wed, Dec 7, 2011 at 9:41 PM, Matthew Saskin m...@saskin.net wrote:

 I'm using a combination of VM's, physical phones, and dynamips to run
 virtual routers for my home lab - about the only things that can't really
 be done with dynamips are some QOS, CUE, T1's, and media resources, but
 that's about it.  My total investment in my home lab is well under $1500
 and I can probably do 75-80% of the blueprint material.

 5 x phones - 7960s/7961's
 VMware server - 2 x quad core athlon processors, 16gb ram, 2x750gb hdd,
 ESXi 4.0

 On that VMware server I'm running a full CCIE lab
 (pub/sub/cups/uccx/ucxn), as well as 4 x XP desktops for IP communicator,
 CAD installed on a few as well as a virtualized linux server running 4
 virtual routers under dynamips.

 -matthew



 On Wed, Dec 7, 2011 at 11:37 AM, Bill Lake whl...@gmail.com wrote:

 I think the least expensive and less time consuming method is to use
 remote rack rental. They have all the hardware needed to practice.  That
 said, plan ahead and have different vendors so that if say Remote lab A is
 down, you can schedule with remote lab b or c and not waste your study
 time.  Also make sure they support remote phone connections so that you can
 use your local resources (cisco router and phones) to connect them in to
 have hands on phone use.

 That all said a Pod is nice as you can check things out and test
 anytime.  Here are my recommendation to save money and build a pod.

 I am running all my servers on a VMWare EXSi 5  server w/i7 870, 8 GB
 RAM, 4 HDD, 2 NIC's and using resource pooling.

 From other post with edits:

 - PSTN phone can be 7960 so you can save big there!  3 or more 7961 or
 7962 Phones should be fine, use soft phones or move phones from pod to pod
 (ie move CUCM hardware phone to CME when needed for testing)
 - 1 2801 w/256MB RAM,w/128MB Flash, 1 PVDM2-16, 1 VWIC-2MFT-T1, 1
 HWIC-4ESW (this will be BR1)
 - 1 2801 w/256MB RAM,w/128MB Flash, 1 PVDM2-16, 1 VWIC-2MFT-T1  (this
 will be HQ)
 - 1 2801 w/256MB RAM,w/128MB Flash, 1 PVDM2-16, 2 VWIC-2MFT-T1, 1
 VWIC-2MFT-E1(this will be PSTN)
 - 1 3560 or 3750 switch
 - NM-CUE or AIM-CUE, NM is more powerful but AIM is cheaper. (this will
 go into your 2821)
 - 1 VWIC-2MFT-T1/E1 (this will go into your 2821)
 - 1 HWIC-4ESW (this will go into your 2821)

 You save money using the VWIC-2MFT-T1 and VWIC-2MFT-E1 over the
 programable ones, you just have to remember that you need to know the
 commands for programing the card just in case.

 You will want more PVDM module(s) for testing but you can get by at
 first with this. Also you can shut down one router and move PVDM over to do
 the more demanding testing.

 You can also get by with 64MB of Compact Flash for most of the routers,
 just look at the requirements of the IOS you want to install.  You will
 need more for the CME router.

 Some will say you can't use 2801 but I am currently running two in my
 pod (including my PSTN) and it is working well.



 Hope this helps some


 On Wed, Dec 7, 2011 at 7:50 AM, Michael Miller kf4...@gmail.comwrote:

 The latest list of hardware and software that you'll see in the
 lab can be found on the Cisco Learning Network.

 https://learningnetwork.cisco.com/docs/DOC-5292

 Thanks,

 Michael

 On Wed, Dec 7, 2011 at 11:46 AM, Muhammad Zubair 
 muhammad.zub...@gmail.com wrote:

 My Friend
 the link you send to me ? is that updated one... in software section
 it is showing CCM version 4.1(3) with latest SR ? Can you believe
 that
 CCIE Logo is new but . interesting

 the things which you wrote is like a sea. ok any idea how much it
 cost total. where to start and how to use that efficiently ?
 confused totally

 On Wed, Dec 7, 2011 at 12:22 PM, AJ BG ciscoie2...@gmail.com wrote:

  Muhammad,

 · Stay away from GNS3, you will end up troubleshooting GNS3
 instead of troubleshooting real voice issues.

 · You will need a server with at least 16GB ram; don’t
 forget that you have to have 5 virtual servers.  You don’t need FXO
 for the current version. I agree with Emanuel,  I would get four
 2800 series routers, you would need to have PVDMS, T1 and E1 cards, CUE
 module, switch module

Re: [OSL | CCIE_Voice] DHCP bindings

2011-12-05 Thread Matthew Saskin
Look up using an external file (either via TFTP or on the router flash) for
the easiest (in my opinion) way to accomplish manual DHCP address binding
within a single pool.  In fact, I'm fairly certain that a number of us
responded with various was to accomplish this in an earlier email thread.

-matthew

On Mon, Dec 5, 2011 at 8:49 AM, Emanuel Damasceno aedamasc...@gmail.comwrote:

 Hello experts,

 We can't just ignore all the messages we see here. And the last guy who
 mentioned about DHCP manual bind, that really drew my attention.

 My knowledge goes as far as the documents I read. In all those documents,
 I really didn't see a way to make a manual bind without using more than 1
 pool. I think those guys come and say they saw lab 1, or 2, or 3, or
 whatever, they just wanna scare us.

 If somebody really thinks this is authentic, please respond to this
 e-mail. I am NOT talking about Lab X, nor I wanna screw with Cisco's NDA.

 I wanna know about this DHCP manual binding with one pool.

 *Emanuel Damasceno*



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Re: [OSL | CCIE_Voice] Strange behavior on MGCP gateways

2011-12-02 Thread Matthew Saskin
I'd look a bit further, the silence is not causing the call to terminate
- what about calls being muted, etc.  What do the q931 debugs show when the
call gets disconnected?


On Thu, Dec 1, 2011 at 11:02 PM, ccielabrat ccielab...@gmail.com wrote:

 I noticed a weird thing while testing MGCP.

 If I call out to my pstn phone and answer the call by pressing the answer
 soft key , the call will disconnect after about two minutes.
 If I answer the call with the speaker button , it stays up forever.

 I'm guessing the problem is I don't have handsets on any of my phones , so
 when I answer with the answer softkey , the phone is off hook and sending
 dead air packets.
 Somehow , mgcp see this as an error condition after two minutes or so and
 kills the call.

 If the call is on speaker, I guess it picks up enough noise to keep the
 call from being considered inactive.

 Weird.


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 visit www.ipexpert.com

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Re: [OSL | CCIE_Voice] Route Group digit manipulation / TEHO

2011-11-28 Thread Matthew Saskin
Route group manipulations that occur within route list membership are for
that route list only.

As an example, if I have the following RL/RG's

RL-HQ: RG-HQ
RL-BR1: RG-BR1
RL-BR2: RG-BR2
RL-HQ-BR1-TEHO: RG-BR1 (**), RG-SLRG

I can make calling party modifications in the RG-BR1 denoted with (**)
without impacting the RG-BR1 reference in the RL-BR1 route list.

-matthew

On Mon, Nov 28, 2011 at 9:59 AM, Emanuel Damasceno aedamasc...@gmail.comwrote:

 Hello Experts,

 I was labbing this weekend (Lab 5A IPExpert), and I started trying out
 things to clarify any doubts I would have. I didn't have enough time on my
 sessions to try this out, and I am wondering if you guys can give me any
 input on this.

 I create route groups for HQ (RG-HQ), and BR1 (RG-BR1) sites with its
 respective Route Lists. I also create a route list for HQ calling BR1 PSTN
 (RG-BR1  Standard Local Route Group).

 If I choose to manipulate CALLING numbers on the route groups inside the
 route list for TEHO, would I be altering the Route Group manipulation first
 created? I was sticking with Calling/ed party transformation patterns...

 Thanks in advance for any input
 *Emanuel Damasceno*



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 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

2011-11-21 Thread Matthew Saskin
Option 60  (VCI / Vendor Class Information), at least for Cisco phones, is
the same for the same model, so if indeed all IP phones are the same type,
they'll send the same VCI, meaning you can't use dhcp classes to manually
bind separate IP addresses.

Given that, the only option I'm aware of is to use a dhcp binding file
hosted (for example) on a TFTP server.  See the following for reference:
http://www.cisco.com/en/US/docs/ios/12_3t/12_3t11/feature/guide/gtdhcpsm.html

-matthew

On Mon, Nov 21, 2011 at 8:23 AM, Ken Wyan kew...@gmail.com wrote:

 Hi,

 Excellent method  thanks for publishing.

 option 60 VCI values are practically used for Lightweight Wireless Access
 Points to find the WLC. Cisco has published option 60 ascii strings for all
 access points. (it's something like  Cisco AP c1240). Nowhere I could
 find ascii string for 7965 phones. Decoding your hex value gives
 ascii string Cisco Systems, Inc. IP Phone CP-7960?

 BTW , CCIE lab has all 7965 phones as mentioned in cisco website.

 Thanks






 On Mon, Nov 21, 2011 at 4:44 PM, Mohammed Al Baqari 
 baqari.voic...@gmail.com wrote:

  Regarding DHCP it can be done. The configuration should be as follow.
 This will work for different phone models which is the case in CCIE LAB as
 Phone 1 model is different from Phone 2 model in SC.

 ** **

 *STEP 1* Configure DHCP class with random value for option 60.

 ** **

 ip dhcp class SCPH1

option 60 hex 123456

 ** **

 ip dhcp class SCPH2

option 60 hex 123456

 ** **

 *STEP 2* Configure DHCP Pool as below

 ** **

 ip dhcp pool SC-POOL

network 142.102.66.0 255.255.255.0

default-router 142.102.66.1

option 150 ip CUCM IP* 

class SCPH1

   address range 142.102.66.11 142.102.66.11

class SCPH2

   address range 142.102.66.12 142.102.66.12

 ** **

 *STEP 3* Use the commands “debug ip dhcp server packet” and “debug ip
 dhcp server class”. The output will be as follow:

 ** **

 Nov 21 14:43:46.981: DHCPD: Sending notification of DISCOVER:

 Nov 21 14:43:46.981:   DHCPD: htype 1 chaddr
 0030.94c3.e5ab *!!!... This is my SC Phone 1 MAC
 Address*

 Nov 21 14:43:46.981:   DHCPD: remote id 020ac0a8641f001f

 Nov 21 14:43:46.981:   DHCPD: circuit id 

 Nov 21 14:43:46.981: DHCPD: Seeing if there is an internally specified
 pool class:

 Nov 21 14:43:46.981:   DHCPD: htype 1 chaddr 0030.94c3.e5ab
 *!!!... This is my SC Phone 1 MAC Address*

 Nov 21 14:43:46.981:   DHCPD: remote id 020ac0a8641f001f

 Nov 21 14:43:46.981:   DHCPD: circuit id 

 Nov 21 14:43:46.981: DHCPD: input does not contain option 82

 Nov 21 14:43:46.981: DHCPD: Searching for a match to '   option 60 *
 436973636f2053797374656d732c20496e632e2049502050686f6e652043502d3739363000
 *' in class SCPH1   *!!!... The underlined value is the correct option
 60 value for SC Phone 1*

 ** **

 *STEP 4* Reconfigure the correct option 60 value for phone 1.

 ** **

 ip dhcp class SCPH1

 no   option 60 hex 123456

option 60 hex *
 436973636f2053797374656d732c20496e632e2049502050686f6e652043502d3739363000
 *

 ** **

 Steps 3  4 can be reused to find out and correct option 60 value for
 phone 2.

 ** **

 Regards,

 Mohammed Al Baqari

 ** **

 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Robert Schuknecht
 *Sent:* Monday, November 21, 2011 12:02 PM
 *To:* 'Abel ...'; 'datucha123 datucha123'
 *Cc:* ccie_voice@onlinestudylist.com
 *Subject:* Re: [OSL | CCIE_Voice] lab - 6

 ** **

 Hi,

 ** **

 SIP early offer is similar to H323 Fast-Start. With SIP EO the INVITE
 includes the SDP with the possible Media Options, Codecs for example.
 Without an NTP/Transcoder CUCM is not able to do EO on SIP Trunks.

 ** **

 But, regarding the DHCP Part, there I am totally lost. I know how to
 configure DHCP on IOS, so that each host will become his “own” IP Address,
 every time. Its done with the host and client-id commands under the DHCP
 Pools. But, I have no glue hwo to do for a couple of devices with only ONE
 DHCP Pool. The way I do it I need 2 Pools, minimum.

 ** **

 Could you please provide a complete Link to the documentation which
 explains how to do it? That would be great!

 ** **

 /Robert

 ** **

 *Von:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *Im Auftrag von *Abel ...
 *Gesendet:* Montag, 21. November 2011 00:48
 *An:* datucha123 datucha123
 *Cc:* ccie_voice@onlinestudylist.com
 *Betreff:* Re: [OSL | CCIE_Voice] lab - 6

 ** **

 Well, the DHCP issue can be done easy if you look 5 minutes on 
 cisco.comdocumentation, by Early offer what you mean? no interdigit timeout?
 

 On Sun, Nov 20, 2011 at 2:22 PM, datucha123 datucha123 
 datucha...@gmail.com wrote:

 Do 

Re: [OSL | CCIE_Voice] lab - 6

2011-11-21 Thread Matthew Saskin
But to the original point, if the requirement is manual reservations *
without* creating multiple dhcp  pools, this won't work.

-matthew

On Mon, Nov 21, 2011 at 9:19 AM, Bill Lake whl...@gmail.com wrote:

 Or maybe you can do a DHCP client reservation, something like this

 ip dhcp pool Server_static
 host 10.0.0.10 255.255.255.0
 client-identifier 0102.0010.0012.34

 http://blog.ioshints.info/2007/09/assigning-server-ip-addresses-with-dhcp.html

 On Mon, Nov 21, 2011 at 7:23 AM, Ken Wyan kew...@gmail.com wrote:

 Hi,

 Excellent method  thanks for publishing.

 option 60 VCI values are practically used for Lightweight Wireless Access
 Points to find the WLC. Cisco has published option 60 ascii strings for all
 access points. (it's something like  Cisco AP c1240). Nowhere I could
 find ascii string for 7965 phones. Decoding your hex value gives
 ascii string Cisco Systems, Inc. IP Phone CP-7960?

 BTW , CCIE lab has all 7965 phones as mentioned in cisco website.

 Thanks






 On Mon, Nov 21, 2011 at 4:44 PM, Mohammed Al Baqari 
 baqari.voic...@gmail.com wrote:

  Regarding DHCP it can be done. The configuration should be as follow.
 This will work for different phone models which is the case in CCIE LAB as
 Phone 1 model is different from Phone 2 model in SC.

 ** **

 *STEP 1* Configure DHCP class with random value for option 60.

 ** **

 ip dhcp class SCPH1

option 60 hex 123456

 ** **

 ip dhcp class SCPH2

option 60 hex 123456

 ** **

 *STEP 2* Configure DHCP Pool as below

 ** **

 ip dhcp pool SC-POOL

network 142.102.66.0 255.255.255.0

default-router 142.102.66.1

option 150 ip CUCM IP* 

class SCPH1

   address range 142.102.66.11 142.102.66.11

class SCPH2

   address range 142.102.66.12 142.102.66.12

 ** **

 *STEP 3* Use the commands “debug ip dhcp server packet” and “debug ip
 dhcp server class”. The output will be as follow:

 ** **

 Nov 21 14:43:46.981: DHCPD: Sending notification of DISCOVER:

 Nov 21 14:43:46.981:   DHCPD: htype 1 chaddr
 0030.94c3.e5ab *!!!... This is my SC Phone 1 MAC
 Address*

 Nov 21 14:43:46.981:   DHCPD: remote id 020ac0a8641f001f

 Nov 21 14:43:46.981:   DHCPD: circuit id 

 Nov 21 14:43:46.981: DHCPD: Seeing if there is an internally specified
 pool class:

 Nov 21 14:43:46.981:   DHCPD: htype 1 chaddr 0030.94c3.e5ab
 *!!!... This is my SC Phone 1 MAC Address*

 Nov 21 14:43:46.981:   DHCPD: remote id 020ac0a8641f001f

 Nov 21 14:43:46.981:   DHCPD: circuit id 

 Nov 21 14:43:46.981: DHCPD: input does not contain option 82

 Nov 21 14:43:46.981: DHCPD: Searching for a match to '   option 60 *
 436973636f2053797374656d732c20496e632e2049502050686f6e652043502d3739363000
 *' in class SCPH1   *!!!... The underlined value is the correct option
 60 value for SC Phone 1*

 ** **

 *STEP 4* Reconfigure the correct option 60 value for phone 1.

 ** **

 ip dhcp class SCPH1

 no   option 60 hex 123456

option 60 hex *
 436973636f2053797374656d732c20496e632e2049502050686f6e652043502d3739363000
 *

 ** **

 Steps 3  4 can be reused to find out and correct option 60 value for
 phone 2.

 ** **

 Regards,

 Mohammed Al Baqari

 ** **

 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Robert Schuknecht
 *Sent:* Monday, November 21, 2011 12:02 PM
 *To:* 'Abel ...'; 'datucha123 datucha123'
 *Cc:* ccie_voice@onlinestudylist.com
 *Subject:* Re: [OSL | CCIE_Voice] lab - 6

 ** **

 Hi,

 ** **

 SIP early offer is similar to H323 Fast-Start. With SIP EO the INVITE
 includes the SDP with the possible Media Options, Codecs for example.
 Without an NTP/Transcoder CUCM is not able to do EO on SIP Trunks.

 ** **

 But, regarding the DHCP Part, there I am totally lost. I know how to
 configure DHCP on IOS, so that each host will become his “own” IP Address,
 every time. Its done with the host and client-id commands under the DHCP
 Pools. But, I have no glue hwo to do for a couple of devices with only ONE
 DHCP Pool. The way I do it I need 2 Pools, minimum.

 ** **

 Could you please provide a complete Link to the documentation which
 explains how to do it? That would be great!

 ** **

 /Robert

 ** **

 *Von:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *Im Auftrag von *Abel ...
 *Gesendet:* Montag, 21. November 2011 00:48
 *An:* datucha123 datucha123
 *Cc:* ccie_voice@onlinestudylist.com
 *Betreff:* Re: [OSL | CCIE_Voice] lab - 6

 ** **

 Well, the DHCP issue can be done easy if you look 5 minutes on 
 cisco.comdocumentation, by Early offer what you mean? no interdigit timeout?
 

 On Sun, Nov 20, 2011 at 2:22 PM, datucha123 datucha123 
 datucha...@gmail.com wrote:

 Do you mean under the DHCP 

Re: [OSL | CCIE_Voice] How to bypass VMWare found pop up after powering up CUCM ?

2011-11-21 Thread Matthew Saskin
Use the vsphere remote console...

On Mon, Nov 21, 2011 at 6:19 PM, Voiper datapack...@gmail.com wrote:


  Hi Friends :)

 I moved my VMware CUCM image to a different server and when I power up I
 get the pop up (attached) mentioning This system has been identified as  a
 VMWare server and I have to click on Continue and again click on
 Agree on the next pop up every time I start my CUCMs.
 As I don't have a monitor connected to the ESXi box, and I use the vsphere
 client to connect to ESXi, I will need to either get a monitor and keyboard
 for the ESXi or find a way to bypass the pop up.
 I googled but came up empty. Most topics on Operarting system not found.I
 am using Demo version of Call Manager 7.1.0. on ESXi 5 (same issue with
 esxi 4 too)
 Any help will be thanked.

 Voiper

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Re: [OSL | CCIE_Voice] How to bypass VMWare found pop up after powering up CUCM ?

2011-11-21 Thread Matthew Saskin
use the vsphere remote console, hit enter.

What version of CUCM are you running?  7.1(x) shouldn't give this error on
boot, earlier versions may.


On Mon, Nov 21, 2011 at 8:21 PM, Voiper datapack...@gmail.com wrote:

 Forgot to attach file

 Thanks,
 Voiper

 On Mon, Nov 21, 2011 at 6:19 PM, Voiper datapack...@gmail.com wrote:


  Hi Friends :)

 I moved my VMware CUCM image to a different server and when I power up I
 get the pop up (attached) mentioning This system has been identified as  a
 VMWare server and I have to click on Continue and again click on
 Agree on the next pop up every time I start my CUCMs.
 As I don't have a monitor connected to the ESXi box, and I use the
 vsphere client to connect to ESXi, I will need to either get a monitor and
 keyboard for the ESXi or find a way to bypass the pop up.
 I googled but came up empty. Most topics on Operarting system not found.I
 am using Demo version of Call Manager 7.1.0. on ESXi 5 (same issue with
 esxi 4 too)
 Any help will be thanked.

 Voiper



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 For more information regarding industry leading CCIE Lab training, please
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Re: [OSL | CCIE_Voice] SRST Gatekeeper

2011-11-14 Thread Matthew Saskin
What does the rest of your configuration on the SRST router look like?  Do
you have the requisite dial peers in place to direct calls from SRST
registered phones to the gatekeeper?  Without that you're not going to get
very far...the SRST router has no knowledge of the callmanager gatekeeper
configuration.

-matthew

On Mon, Nov 14, 2011 at 1:18 PM, datucha123 datucha123 datucha...@gmail.com
 wrote:


 Hello everyone,

 I have a problem with SRST to Gatekeeper calls.

 When the IP Phone are registered with CUCM calls to CUCME through GK works
 just fine.
 But as soon as I will shutdown the CUCM Service on all servers, so that
 the IP Phones will register with SRST Router, the outgoing calls to CUCME
 through the GK fail.

 But the inbound call to SRST IP Phone through the GK from CUCME works
 fine.

 SRST Config:

 *interface Loopback0
  ip address X.X.X.X 255.255.255.255
  h323-gateway voip interface
  h323-gateway voip id CUCM ipaddr Y.Y.Y.Y 1719
  h323-gateway voip h323-id srst
  h323-gateway voip tech-prefix 1#
  h323-gateway voip bind srcaddr Z.Z.Z.Z*
 * *
 *call-manager-fallback
  max-conferences 4 gain -6
  transfer-system full-consult
  ip source-address X.X.X.X port 2000
  max-ephones 10
  max-dn 10 no-reg
  system message primary SRST
  moh music-on-hold.au
  multicast moh 239.5.5.6 port 16384
  time-zone 32
  time-format 24
  date-format dd-mm-yy*

 GK Config:

 *gatekeeper
  zone local CUCME test.com A.A.A.A
  zone local CUCM test.com
  zone prefix CUCM 1... gw-priority 10 ccm_2
  zone prefix CUCM 1... gw-priority 9 ccm_1
  zone prefix CUCM 1... gw-priority 8 srst
  zone prefix CUCM 2... gw-priority 10 ccm_2
  zone prefix CUCM 2... gw-priority 9 ccm_1*
 * zone prefix CUCM 2... gw-priority 8 srst
  zone prefix CUCME 3...
  zone prefix CUCME 5...
  gw-type-prefix 1#* default-technology
  gw-type-prefix 3#* gw ipaddr B.B.B.B 1720
  gw-type-prefix 2#*
   arq reject-unknown-prefix
  no shutdown

 *
  Maybe there is a known issue with SRST and Gatekeeper finctionality. Or
 maybe SRST does not work with GK at all?




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Re: [OSL | CCIE_Voice] CUCM 8.6 and CUPS 8.6 NTP issue

2011-11-05 Thread Matthew Saskin
A CCM publisher always runs an NTP server to provide time services to the
rest of the CCM cluster - this also extends to adjuncts like CUPS, which by
default (and non configurable as far as I know) pulls time from the cluster
publisher that it is integrated to.

-matthew

On Sat, Nov 5, 2011 at 9:20 AM, Ravindra Lakpriya lakpr...@gmail.comwrote:

 Hi Guys,

 Im setting up a testbed on vmware workstation with CUCM 8.6 and CUPS 8.6.
 Im using the windows domain controller as the NTP server. i have notice
 once i went through the initialization wizard of CUPS automatically it
 changes its NTP to CUCM publisher. But CUCM publisher is sync to external
 NTP and CUCM is not running any NTP service on publisher it self.

 Once i went through the troubleshooter from CUPS i notice its keeps on
 warning me about the NTP and from the platform management there is no way
 to add or remove NTP servers. Can we remove the external NTP sync and
 configure CUCM 8.6 publisher as the NTP  server ?

 Any help on this is really appreciated.

 --
 Ravindra Lakpriya


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Re: [OSL | CCIE_Voice] IP Expert mock labs

2011-10-21 Thread Matthew Saskin
There are no base configs for CUCM (and UCXN, UCCX, etc.) per se.  It's
mostly just server hostnames  IP addresses, plus some basic simple
integration in the case of UCXN and UCCX.  Your best bet for a home lab is
to get the various application servers to a usable starting point - OS/App
installed and IP addresses assigned, etc and then take a VMware snapshot to
use as a starting point for all labs.

-matthew


On Fri, Oct 21, 2011 at 9:53 AM, Mike mik...@msn.com wrote:

 Hi all,

 ** **

 I have my own lab with 2811’s, HWIC4s, 7961 phones and VMWARE for all UC
 servers. In my RS days I had my own gear and could load up labs with no
 issues.

 ** **

 My question is will I be able to load up mock labs on my routers and UC
 servers? If so how? I’m sure the router configs are text files, but are the
 CM configs tar files? Bulk imports? How do I can the base configs into CM,
 Unity Cx , Presence and UCCX?

 ** **

 ** **

 Thanks,

 Mike

 CCIE #17982m (RS)

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Re: [OSL | CCIE_Voice] thoughts on plan / type / name requirements forani / dnis

2011-10-17 Thread Matthew Saskin
It's less telco specific as it is geography-specific.  At this point there
are still a large number of NPA's where 7D dialing is permitted.  From
NANPA, ~129 NPA's require 10D dialing with some exceptions:
http://www.nanpa.com/nas/public/npasRequiring10DigitReport.do?method=displayNpasRequiring10DigitReport

-matthew


On Sun, Oct 16, 2011 at 9:16 PM, Anthony Alba ascanio.al...@gmail.comwrote:

 Slightly OT: is the US shifting to allowing 10D or 1+10D for all calls or
 is it still highly telco specific? Is 7D still used? I thought that many
 cities require 10D even for local calls.


 Obviously-not-living-in-the-US Anthony






 On 16 Oct 2011, at 16:38, Brian btmulg...@gmail.com wrote:

 thanks a lot Kshitij, i am not too familiar with US telco requirement so
 this  is good to know.

 Sent from my iPad

 On 15 Oct 2011, at 20:39, Kshitij Singhi  martinian.ksin...@gmail.com
 martinian.ksin...@gmail.com wrote:

 Hi Brian,

 That's quite a common requirement for Telcos. (not so much for 11 digit
 calls but definitely for the international calls)

 There are quite a few Service Providers who expect one of the following
 (pertaining only to international calls)

 If 011 is prefixed, then they will pass the call only if it is flagged as
 unknown.
 If 011 is not being sent across, then they will pass the call only if it is
 flagged as international.

 From a Gateway perspective, what we usually do is make use of the following
 command to set the plan/type on the Serial Interface:

 interface serial x/y/z:23
 isdn map address ^011 plan isdn type unknown (or plan unknown type unknown
 depending on what the Telco expects)

 (this obviously pertains to calls where the 011 is being sent across - be
 careful with the command since it changes the calling/called number
 plan/type). For calls where the 011 is not sent across, it's safer to use
 translation profiles on the outgoing dial peers. For MGCP, it's all done on
 the CUCM and translations are eventually leveraged for sending the correct
 plan/type/number of digits in SRST.

 On Sun, Oct 16, 2011 at 12:59 AM, Brian Mulgrew  
 btmulg...@gmail.combtmulg...@gmail.com
 btmulg...@gmail.com wrote:

 Thanks for the response Kshitij

 just curious - but what is the thinking behind sending type as unknown
 when not presenting leading digits in the US?

 Brian




 On Sat, Oct 15, 2011 at 8:27 AM, Kshitij Singhi 
 martinian.ksin...@gmail.commartinian.ksin...@gmail.com
 martinian.ksin...@gmail.com wrote:

 Here is what I had followed:

 ALWAYS send a plan of ISDN.

 If it has been stated that the Telco expects a specific plan/type, then
 ensure that the plan/type is sent in the ani/dnis. Also, for an outgoing
 call ensure that you take care of the digits seen on the calling phone ONLY
 if specified in the paper.

 If it has not been specifically stated, then for called number:

 1. Send a plan of ISDN and type of subscriber for 7 digit calls (US)
 2. Send a plan of ISDN a type of national for 11 digit calls (US)
 (assuming the Telco does NOT expect the leading 1)
 3. Send a plan of ISDN and type of international for International calls
 (US) (assuming the Telco does not expect the leading 011)
 4. Send a plan of ISDN and type of unknown for 11 digit calls (US)
 (assuming the Telco expects the leading 1)
 5. Send a plan of ISDN and type of unknown for international calls (US)
 (assuming the Telco expects the leading 011)
 6. Send a plan of ISDN and type of unknown for 911 calls.

 For non-US sites:

 1. Send a plan of ISDN and type of subscriber for local calls
 2. Send a plan of ISDN and type of international for international calls
 3. For emergency numbers, send a plan of ISDN and type of unknown.

 For the calling number:

 1. Send a plan of ISDN and type of subscriber for 7 digit calls. (US)
 2. Send a plan of ISDN and type of national for 11 digit calls. (US)
 3. Send a plan of ISDN and type of international for international calls.
 (US)
 4. Send a plan of ISDN and type of national for emergency calls

 For non US sites:

 1. Send a plan of ISDN and type of subscriber for emergency calls.
 2. Send a plan of ISDN and type of subscriber for local calls
 3. Send a plan of ISDN and type of international for international calls.


 On Sat, Oct 15, 2011 at 7:21 AM, 
 ccie_voice-requ...@onlinestudylist.comccie_voice-requ...@onlinestudylist.com
 ccie_voice-requ...@onlinestudylist.com wrote:

 Send CCIE_Voice mailing list submissions to
 ccie_voice@onlinestudylist.comccie_voice@onlinestudylist.com
 ccie_voice@onlinestudylist.com

 To subscribe or unsubscribe via the World Wide Web, visit
 
 http://onlinestudylist.com/mailman/listinfo/ccie_voicehttp://onlinestudylist.com/mailman/listinfo/ccie_voice
 http://onlinestudylist.com/mailman/listinfo/ccie_voice
 or, via email, send a message with subject or body 'help' to
 
 ccie_voice-requ...@onlinestudylist.comccie_voice-requ...@onlinestudylist.com
 ccie_voice-requ...@onlinestudylist.com

 You can reach the 

Re: [OSL | CCIE_Voice] MRGMRGL

2011-09-30 Thread Matthew Saskin
It shouldn't matter provided, in the example provided, each MRG contains
only resources of the specified type.  Precedence for a MRG within an MRGL
should only come into play if your MRG's contain the same types of
resources.

On Fri, Sep 30, 2011 at 8:14 AM, eng_firasoq...@yahoo.com 
eng_firasoq...@yahoo.com wrote:

 Dear All,

  If we have to configure MTP,Xcoder,hd_Conference in the HQ site then we
 will configure 3 MRG MTP_MRG,Xcoder_MRG,conf_MRG after that we will create
 MRGL for hd_hq_MRGL but now what is correct order for MRG inside the MRGL

 Regards



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Re: [OSL | CCIE_Voice] CUCM - UCCS upgrading options

2010-10-08 Thread Matthew Saskin
If UCSS wasn't purchased initially or within 90 days (if I recall correctly)
of the initial purchase, they can't purchase UCSS to enable the upgrade.
They need to re-purchase 8.0 software + licenses.

Matthew Saskin
msas...@gmail.com
203-253-9571

July 18, 2010 - 1500m swim (in the hudson), 40k bike, 10k run
Please support the Leukemia  Lyphoma Society
http://pages.teamintraining.org/nyc/nyctri10/msaskin


On Wed, Oct 6, 2010 at 4:48 PM, Saeed IDris saee...@gmail.com wrote:

 *Hi everyone,*

 *I need to ask regarding Cisco UCCS, is it enough to include the below
 part number according to number of users in case this is new implementation:
 *

 *Other question, one of my clients already is hosted CUCMBE version 6.5.1
 but now they plan to upgrade their CUCM to version 8.0 …..They didn’t have
 UCCS ….what is the possible Scenario to achieve such upgrade?*

 **

 *Regards,*

 *Saeed *

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Re: [OSL | CCIE_Voice] SQL replication check\fix Q.

2008-02-25 Thread Matthew Saskin
No dblhelper in the lab as it's not installed by default on the 
callmanager.  You typically have to grab it from another CM that has it, 
or get it from TAC.


-matt

Jonathan Charles wrote:

Jumping in late, just subscribed... do they not have DBLHelper on the
Publisher? Is it there on the CCIE lab? If not, do we have to rebuild
the publication manually?



Jonathan

On Sun, Feb 24, 2008 at 3:11 PM, Mark Snow [EMAIL PROTECTED] wrote:

At the time of publishing those last WBs we did not have a Subscriber - and
perhaps the wording wasn't the best.

Now (and for sometime now) we do have both Pub (10.x.200.21) and a Sub
(10.x.200.20) live in every pod.


Hope that clears some things up :)


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

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


On Feb 23, 2008, at 2:27 PM, Mark Cardwell wrote:


Question and concern. On the ebooks and printed WB, It states that there is
no publisher (.20) and just to assume a publisher. So given how can we fix
replication on a server that does not exist. I am very confused and am
wonder what you are referring to when you state this is an inherent
troubleshooting task. And another thing I am confused on is if there is a
Pub and a Sub then why am I logging into the Sub (according to the WB
documentation).

Cheers!

Mark Cardwell | System Engineer | Presidio Networked Solutions |
[EMAIL PROTECTED]| Cell: 571.225.0132  | Office: 301.623.2000| FAX:
301.313.2400



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Snow
Sent: Friday, February 22, 2008 1:19 PM
To: Moore, Neil
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] SQL replication check\fix Q.

SA user (even though mentioned in that article) is not ever to be used.

Use local authentication instead - will work every time - even between
boxes.

And yes - SQL replication is broken on all of our Subscribers in every pod -
and that is no accident! :)
It is there as an inherent troubleshooting task - which is a testable
'topic' in the lab - so beware - things may not always be as they seem on
the surface!


HTH,


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

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




On Feb 22, 2008, at 12:38 PM, Moore, Neil wrote:




Hi there folks!

I'm after any thoughts\tips on checking the sql replication's all good
before you start you lab configs (virtual and real).

My last couple of vrack sessions I've had lots of probs with this.  The
replication is never setup and running at the start of the vrack sessions so
I update lmhosts (checked the cache is good), check connectivity etc, and
then tried the following procedure:

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a00801d11a6.shtml

for the sa account pwrd I'm trying cisco and blank, I've reset these on both
boxes aswell, same.  Seems to me like some sort of authentication issue.

I've even put dblhelper on the box  that can't sort it  this always works
a treat in the field!!

So I guess my questions are:

What check do you folks do to check the replication (add a dev  check the
device table?)
What procedure do you follow to get the replication up and running at the
start of your vrack sessions (proctorlabs.com)?
Is this a 'task' for all proctorlab vrack sessions only?  (is this just a
vmare 'feature'?)
In the real lab would we be expected to do this kind of sql troubleshooting,
reseting sa pwrds, assigning roles etc in SQL?

What am I missing!!?

Any help'd be appreciated.

Cheers,

Neil (UK)




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Partner and HP Business Partner Select / Authorised Services Delivery
Partner.
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Re: [OSL | CCIE_Voice] Video Codec Rate Table

2008-02-14 Thread Matthew Saskin
Don't quote me, but I believe audio is fixed as 24k for G.729 and 80K 
for G.711

not sure on video...

-matt

Earnieball78 wrote:

Hello all,
I'm am trying to find out where one would fine the table that lists 
the audio and video codec rates information required to configure 
locations based CAC in CCM?
 
Thanks!



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Re: [OSL | CCIE_Voice] Video Codec Rate Table

2008-02-14 Thread Matthew Saskin
It was my understanding that CallManager uses the fixed numbers 
(24k/80k) for location based CAC regardless of the sampling rate.


Can anyone confirm/deny this?

-matt

Earnieball78 wrote:

Matt,
Thanks for your response. Well, that would depend on the sampling rate.
I don't have any problems with the voice codecs, however I do want to 
use the table for the video part.
 
Anybody knows where they got the table that is at the begining of the 
workbook?
 
Thanks!


*/Matthew Saskin [EMAIL PROTECTED]/* wrote:

Don't quote me, but I believe audio is fixed as 24k for G.729 and 80K
for G.711
not sure on video...

-matt

Earnieball78 wrote:
 Hello all,
 I'm am trying to find out where one would fine the table that lists
 the audio and video codec rates information required to configure
 locations based CAC in CCM?

 Thanks!



 Never miss a thing. Make Yahoo your homepage.




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Re: [OSL | CCIE_Voice] (no subject)

2008-02-13 Thread Matthew Saskin

What do you mean scripting?

To the first point, you'll have to deal with your telco to have them 
point 800 numbers inbound.  Will they just be routing an 800 number to 
your existing DID's (like remote call-forward) or will the 800 numbers 
be assigned directly to your circuits and they'll route DNIS digits inbound.


Regardless, handling these within CME is just going to be a matter of 
setting up proper dial-peers (remember incoming called-number and 
direct-inward-dial statements) to handle the calls and send them where 
they need to go (phones registered to CCME, CCM via trunks, other H.323 
or SIP peers).


-matt

Jonathan Preston wrote:
I need to provision some trunks going into CME and using scripting 
done in CME.  Can someone please give me some guidance on this request?
 


* Negotiate with telco to point existing 800# lines into the CME
  trunks.
* Conduct appropriate scripting activiteis to make 800#s
  functional withing CME

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Re: [OSL | CCIE_Voice] MoH Question

2008-02-11 Thread Matthew Saskin
I don't believe so - the held device will attempt to either subscribe to 
the multicast group (or request a unicast stream) without any care as to 
whether or not it is actually receiving any audio.


-matt

Earnieball78 wrote:

Hello all,
 
I have question regarding MoH. In a scenario where you provide both 
multicast and unicast streams, is it possible for the MRG/MRGL to 
automatically switch between multicast and unicast depending on the 
capabilities of the network? Let's say that I have two MRGs, one with 
multicast enabled, and one without. In the MRGL, I have the multicast 
MRG on top, and the unicast MRG below. So the end result would be, with 
a multicast enabled network, to always use multicast MoH. But let's say 
that multicast is not enabled on a particular segment of a network, and 
the phones are still using the same MRGL. Since multicast will not work, 
will it switch to unicast?
I have tried this in my lab but I keep getting tone on hold. Hence, my 
question is: Is it even possible for MoH resources to behave in this 
fashion?
 
Thanks!

/Christian


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Re: [OSL | CCIE_Voice] BR1 and Cat6K showed registered but CCM is NOT

2007-12-24 Thread Matthew Saskin

Cisco lists a specific method on their site:

-Add a recognizable phone via the web interface (set description = TEST 
PHONE, etc...)
-Run a query on all nodes in the cluster to see if the database entry is 
there


-matt

Chad Stachowicz wrote:
Is that really a testable topic?  What is the best way to manually 
figure out replication is out of sync?



Chad

Sent from my iPhone

On Dec 23, 2007, at 10:26 PM, Mark Snow [EMAIL PROTECTED] wrote:


No DBHelper available in the real lab.
Delete if there the Subscription to the CCM0301 DB on the Subscriber 
server, and manually set it back up again.
I will have a video up on our webserver soon that you can view on 
specifically how to manually setup the Pub/Sub pull subscription.


Cheers,

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

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



On Dec 23, 2007, at 1:57 PM, Chad Stachowicz wrote:


Mark how are we expected to fix replication ?   Dbhelper?


Thanks.

Chad

Sent from my iPhone

On Dec 22, 2007, at 1:00 PM, Mark Snow [EMAIL PROTECTED] wrote:

When you do a sh port on the 6608 port, what ccm server does it show 
reg to? Are you sure that SQL replication is fixed between pub and sub?
(by default all of our pods are broken on purpose as a task for the 
student to fix)


Mark Snow
Sr Technical Instructor
IPexpert, Inc.

Sent from my iPhone

On Dec 22, 2007, at 11:47 AM, Toyin Bakare 
[EMAIL PROTECTED] wrote:



Hello All,

I have registered BR1 and Cat6K in the CCM, when i do sh port on 
cat6k or sh ccm on BR1, it shows status as registered but CCM shows 
registration as unknown.


I have restarted CCM several times, i deleted and re-add also 
several time.


thnks






Re: [OSL | CCIE_Voice] Switch Module For 2800

2007-12-10 Thread Matthew Saskin
If I recall correctly and you want to run PoE, you need both.  One is 
just the switch card, the other is a daughter-card that provides PoE 
capabilities.


-matt

Daniel Galati wrote:

Hello,
  I would like to know which module to use for the 2800 acting as the 
remote site.

 Is the module a 16 port power ESW or 16 port ES power module.
 Thank you.
   Dan