Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension

[Derek Wyss]

Translation rule on CUCM.  MGCP or H323 should not matter.  You would match
on the called number which is the voicemail pilot number then manipulate
the calling number and send it on its way.  It would not affect standard
calls into Site A as it would not match the rule.

Derek



On Wed, Mar 20, 2013 at 4:51 PM, Steve Keller  wrote:

> Thanks Derek
> On the surface it seems like that would chop down my ANI to 4 digits for
> any call into site A not just calls to vm. Also in my case site A is MGCP
> controlled so I that is not an option for me...
> On Mar 20, 2013 5:16 PM, "Derek Wyss"  wrote:
>
>> Alternatively, you could also create a translation rule in a partition
>> accessible only by the inbound gateway that translates the calling number
>> to 4 digits before sending it to voicemail.  The hunt pilot calling
>> transform mask will work, but you could have issues if you have any caller
>> input requirements to route back out to the PSTN from UCON.
>>
>> Derek Wyss
>> CCIE#38238(Voice)
>>
>> On Wed, Mar 20, 2013 at 3:44 PM, Steve Keller wrote:
>>
>>> In SRST mode, when the vm button is pressed, i have a dial-peer to route
>>> this call to the vm hunt pilot on the UCM.
>>>
>>> dial-peer voice 2600 pots
>>> description voicemail-pilot
>>> destination-pattern 2600
>>> no digit-strip
>>> port 0/0/0:23
>>> prefix 1408202
>>> If i have to adhere to the requirement that LD calls should be 10 digit
>>> ANI, then i am sending the full 10 digit ANI for this call as well ( even
>>> though it more of a hidden number rather than an implicit user dialed
>>> number)
>>> Thus the call arrives at site A GW with 10 digits , say 9723033002.
>>> In order to route this call to the correct mailbox i would have to use
>>> Alternate Extension of 9723033002 and then i will be prompted to login.
>>> However, if i am not allowed to use alternate extension then i must have
>>> another strategy.
>>>
>>> here are the choices i can think of, please chime in if you too have
>>> experienced this dilemma and what is the best way to solve it.
>>>
>>> 1) do not send the full 10 digit ANI for this call and it will arrive at
>>> site a GW as 4 digit ANI and then land in the mailbox, but not adhere to LD
>>> calls should be 10 digit ANI requirement.
>>>
>>> 2) put  as calling party transform mask on the Hunt Pilot, thus
>>> stripping the caller ANI to 4 digits and i can be prompted to log in.
>>> However i think with this method, anytime the caller ANI is read to before
>>> the message is played the caller id would incorrectly state from "3002"
>>> instead of from "9723033002"
>>>
>>> essentially, what is the best way for SRST users to access voicemail
>>> when you are not permitted to use Alternate Extension.
>>>
>>> thanks in advance all!!
>>>
>>> steve
>>>
>>>
>>> ___
>>> 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] SRST to voicemail without Alternate Extension

[Derek Wyss]

Alternatively, you could also create a translation rule in a partition
accessible only by the inbound gateway that translates the calling number
to 4 digits before sending it to voicemail.  The hunt pilot calling
transform mask will work, but you could have issues if you have any caller
input requirements to route back out to the PSTN from UCON.

Derek Wyss
CCIE#38238(Voice)

On Wed, Mar 20, 2013 at 3:44 PM, Steve Keller  wrote:

> In SRST mode, when the vm button is pressed, i have a dial-peer to route
> this call to the vm hunt pilot on the UCM.
>
> dial-peer voice 2600 pots
> description voicemail-pilot
> destination-pattern 2600
> no digit-strip
> port 0/0/0:23
> prefix 1408202
> If i have to adhere to the requirement that LD calls should be 10 digit
> ANI, then i am sending the full 10 digit ANI for this call as well ( even
> though it more of a hidden number rather than an implicit user dialed
> number)
> Thus the call arrives at site A GW with 10 digits , say 9723033002.
> In order to route this call to the correct mailbox i would have to use
> Alternate Extension of 9723033002 and then i will be prompted to login.
> However, if i am not allowed to use alternate extension then i must have
> another strategy.
>
> here are the choices i can think of, please chime in if you too have
> experienced this dilemma and what is the best way to solve it.
>
> 1) do not send the full 10 digit ANI for this call and it will arrive at
> site a GW as 4 digit ANI and then land in the mailbox, but not adhere to LD
> calls should be 10 digit ANI requirement.
>
> 2) put  as calling party transform mask on the Hunt Pilot, thus
> stripping the caller ANI to 4 digits and i can be prompted to log in.
> However i think with this method, anytime the caller ANI is read to before
> the message is played the caller id would incorrectly state from "3002"
> instead of from "9723033002"
>
> essentially, what is the best way for SRST users to access voicemail when
> you are not permitted to use Alternate Extension.
>
> thanks in advance all!!
>
> steve
>
>
> ___
> 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] Dail Plan Consideration in SRST Mode

[Derek Wyss]

If you don't want forward by uncheck outbound redirecting ie on hq gw

Sent from my iPhone

On Feb 24, 2013, at 5:30 AM, CISCO CCIE VOICE  wrote:

> Hi 
> 
> Can any one help me with Dial Plan consideration when calling from HQ Site to 
> Branch 1 Site,following what i have configure.But the problem is that on 
> B1PH1 screen  its showing  as below 
> 
> From +14082021001
> Forward by:2001
> 
> 
> HQ SITE:
> 
> Extension Range:1XXX
> 
> Partition:Branch_1_SRST_PT
> CSS :Branch_1_SRST_CSS--contains-Branch_1_SRST_PT
> 
> B1PH1:i have assign CFUR as 2001 with CSS:Branch_1_SRST_CSS
> 
> 
> Route Pattern: 2XXX--Branch_1_SRST_PT
> Route List  : Standard Local Route Group
> Prefix :91972303
> 
> 
> 
> BRANCH 1 SITE:
> 
> Extension Range :2XXX
> 
> dial-peer voice 10 pots
> destination-pattern 1...
> perfix 14082021
> port 0/0/0:23
> 
> thanks
> 
> ___
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> visit www.ipexpert.com
> 
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> www.PlatinumPlacement.com
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Re: [OSL | CCIE_Voice] Rumours or True

[Derek Wyss]

My suggestion is to config all the way through maybe doing spot checks as
you go.  Make sure PRI's come up, GW's/phones register etc and then do full
detailed testing after you finish configuring.  Your time available for
testing will depend on your speed of configuring.

HTH

Derek

On Mon, Jan 14, 2013 at 11:52 AM, Chrysostomos Christofi <
ch.christ...@logicom.net> wrote:

>  Guys 
>
> ** **
>
> they say that you have to be very fast in the lab
>
> what actual that means?
>
> ** **
>
> do you have time for check every task?
>
> ** **
>
> you can check every task when you finish it ,or if you check every task
> then you will not have a time to finish the remaining lab
>
> ** **
>
> can you pls send your feedback?
>
> ** **
>
> ** **
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
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> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] CUE in SRST

[Derek Wyss]

I would always put mwi sip unsolicited under the sip subsystem as well.

Depends how you intend on setting up your MWI though, outcalling vs
unsolicited sip notifications.

Derek

On Mon, Jan 14, 2013 at 6:52 AM, Chrysostomos Christofi <
ch.christ...@logicom.net> wrote:

>  Hi Guys
>
> ** **
>
> We have cue registered with cucm
>
> if the GW become in srst mode then the below commands will take care about
> voicemail?
>
> ** **
>
> ccn subsystem sip
>
> gateway address "192.168.3.254"
>
> ** **
>
> !
>
> ** **
>
> ccn trigger sip phonenumber 
>
> application "voicemail"
>
> enabled
>
> maxsessions 6
>
> end trigger
>
> ** **
>
> and also a sip dial peer pointed to cue number in VG
>
> ** **
>
> ___
> 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] VOL2 lab 1 - HQ Phone cannot register ? WTH

[Derek Wyss]

I have also seen weird issues surrounded DHCP in my home lab in practice.
What I would suggest and has seemed to HELP alleviate issues is to issue
clear ip dhcp binding *

On Wed, Jan 9, 2013 at 5:43 AM, Bill  wrote:

> Does the phone actually get the ip from your config or is it stuck?
>
> I have had a phone stuck before at proctor labs and it would not take
> anything I did to resolve it.  That is because some of the phones are not
> actually attached to the rack device but layer two tunneled to the rack.
>  That means they do not recycle when you bounce them.
>
> Try this change your ip range and see if the phone gets a new ip, if so
> then it is something ou are missing.
>
> If not try pinging the phone, does it reply?
>
> If so then you might recover it if not then open a ticket and see if you
> can get another pod or use a softphone for now.  You can also hardware VPN
> into proctor labs and I find that to be the best solution for them to give
> me more of a true lab experience.
>
> Bill
>
>
> On Jan 8, 2013, at 11:32 PM, "Piotr Puchalski"  wrote:
>
> Nic,
>
> ** **
>
> I’ve been battling a somewhat similar problem in my own lab (I have all
> the hardware on my rack with full access) and the problem was that somehow
> one of the phones was not taking the default gateway set in PUB’s DHCP. I
> configured a pool locally on the router and it worked then. Next day (and
> next lab) it worked fine again with CUCM. However now it is still not
> working. Always make sure you have the default router configured correctly
> on the phone, that is the first thing. The rest should be a cake.
>
> ** **
>
> Peter
>
> ** **
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Nicolas MICHEL
> *Sent:* Tuesday, January 08, 2013 4:54 PM
> *To:* William Bell; Steffen Bruening
> *Cc:* OSL Voice
> *Subject:* Re: [OSL | CCIE_Voice] VOL2 lab 1 - HQ Phone cannot register ?
> WTH
>
> ** **
>
> Hi Guys !
>
> Yeah I have checked fa0/0.10 and it is ok !
>
> I cannot access to the IP phone since it is PL facilities :) DHCP scope
> has been created/deleted/checked etc etc
> I had no problem with BR1 or BR2 ... only with the 7962 in HQ .. Weird !
>
> I have the same rack tomorrow for the same lab (VOL2 LAB1) that I need to
> finish so I will try again :)
>
> Many thanks for the help guys :)
>
>
> More to come later and hope to gain experience to finally be able to
> answer questions here :)
>
>
> Nic
>
>
> 
>
> Le 08/01/13 18:03, William Bell a écrit :
>
> Nic, 
>
> ** **
>
> On your HQ router, can you check Fa0/0.10? Make sure you have the
> following config:
>
> ** **
>
> *encapsulation dot1q 10 native*
>
> ** **
>
> I don't think it is DHCP snooping or you would not receive an IP address.
> However, you could (and should)  Erase the config on the IP phone to ensure
> it isn't just using an "old" DHCP config. 
>
> ** **
>
> I would also check the DHCP scope. Check the server level (where you are
> likely specifying global Option 150 params) and check the scope itself.
> Make sure you haven't fat fingered a subnet mask, subnet, begin/end
> address, or default gateway. When I have had a problem where the phone was
> getting an address but was not progressing through TFTP and registration it
> was usually related to a misconfiguration of the DHCP scope in CUCM. 
>
> ** **
>
> If you have to change the DHCP scope params then remember to restart the
> DHCP monitor service. It is a finicky beast.
>
> ** **
>
> HTH.
>
> ** **
>
> -Bill
>
> ** **
>
> On Jan 8, 2013, at 9:57 AM, Nicolas MICHEL wrote:
>
>
>
> 
>
> Hey guys.
>
> I m in trouble with the phone connected to Fa1/0/23 of the 3750 in HQ.
>
> Vlan assignement are OK:
> HQ-3750#sh run int fa1/0/23
> Building configuration...
>
> Current configuration : 113 bytes
> !
> interface FastEthernet1/0/23
> switchport access vlan 10
> switchport voice vlan 20
> spanning-tree portfast
> end
>
> HQ-3750#sh run int fa1/0/1
> Building configuration...
>
> Current configuration : 153 bytes
> !
> interface FastEthernet1/0/1
> switchport trunk encapsulation dot1q
> switchport trunk native vlan 10
> switchport mode trunk
> speed 100
> duplex full
> end
>
> HQ-3750#sh int trunk
>
> PortMode Encapsulation  StatusNative vlan
> Fa1/0/1 on   802.1q trunking  10
>
> PortVlans allowed on trunk
> Fa1/0/1 1-4094
>
> PortVlans allowed and active in management domain
> Fa1/0/1 1,10,20,30
>
> PortVlans in spanning tree forwarding state and not pruned
> Fa1/0/1 1,10,20,30
>
>
>
>
> IP HELPER on the L3 Device is OK
>
> interface FastEthernet0/0.20
> encapsulation dot1Q 20
> ip address 10.10.200.3 255.255.255.0
> ip helper-address 10.10.210.10
> end
>
>
> IP Phone can get an IP from the PUB (10.10.210.10)
>
> HQ-3750#sh cdp neigh fa1/0/23 det
> -
> Device ID: SEP0021A086825D

Re: [OSL | CCIE_Voice] Unity connection recording for UCCX prompts

[Derek Wyss]

Bill,

I haven't personally seen a scenario with the recording script not
working.  Unless they specifically ask for 1 way or the other.

Derek

On Fri, Jan 4, 2013 at 9:06 AM, William Bell  wrote:

> Derek,
>
> Is it possible to expand on your statement without violating NDA? I ask
> because I struggle trying to imagine a scenario where I could get to UCCX
> to run the script that plays the prompts but I would be unable to create a
> script that records the prompts (thus forcing me to use CUC or some other
> method).
>
> -Bill
> --
> William Bell
> blog: http://ucguerrilla.com
> twitter: @ucguerrilla
>
>
>
> On Jan 4, 2013, at 7:58 AM, Derek Wyss wrote:
>
> I would recommend knowing how to do it both ways as certain circumstances
> might require it.
>
> Derek
>
> On Thu, Jan 3, 2013 at 11:21 PM, William Bell wrote:
>
>> I assume everyone has their own approach here. I do the following:
>>
>> 1. For Unity Connection recordings (call handlers) I use CUGA
>>
>> 2. For UCCX prompts, I write a script in UCCX and record/upload the
>> prompts from the UCCX server
>>
>> 3. For BACD prompts, I use the UCCX to record the prompt / upload to
>> UCM-TFTP / TFTP copy the file to flash
>>
>> 4. For CUE prompts, I use the CUE prompt management app
>>
>> -Bill
>>  --
>> William Bell
>> blog: http://ucguerrilla.com
>> twitter: @ucguerrilla
>>
>>
>>
>> On Jan 4, 2013, at 12:02 AM, singh wrote:
>>
>>
>> HI Guys,
>>
>> I am planning to use Unity connection to record and download prompts for
>> the UCCX scripts . I am just wondering if this is the best approach or a
>> recording script needs to be written on UCCX.
>>
>>
>> Also from machine on which UCCX is installed can the Unity connection web
>> interface be accessed directly ?
>>
>>
>> -singh
>>
>>
>> Get Yourself a cool, short *@in.com* Email ID 
>> now!<http://www3.in.com/sso/commonregister.php?ref=IN&utm_source=invite&utm_medium=outgoing>
>>  ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com
>>
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com
>>
>
>
>
___
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Re: [OSL | CCIE_Voice] Add voicemail service

[Derek Wyss]

If you mean the service you can use the following SQL command on the CUCM
CLI:

run sql insert into telecasterservice
(pkid,Name,NameASCII,Description,URLTemplate,tkPhoneService,EnterpriseSubscription,Priority)
values('ca69f2e4-d088-47f8-acb2-ceea6722272e','Voicemail','Voicemail','Voicemail','Application:Cisco/Voicemail',2,'t',1)


This is documented at Products -> Voice -> IP Telephony -> Platform -> CUCM
-> Release notes -> 7.01 -> Find telecaster.

Thanks,

Derek

On Fri, Jan 4, 2013 at 4:18 AM, Chrysostomos Christofi <
ch.christ...@logicom.net> wrote:

>  Hi
>
> ** **
>
> Its one configuration for cli if the voicemail its not exist in cucm
>
> Can you provide pls the appropriate link?
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> *Regards*
>
> Chrysostomos
>
> ** **
>
> ___
> 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] PHONES REGISTRATION ISSUE WITH SITES

[Derek Wyss]

Your screenshots don't really help much.  Show us your dhcp config for site
B and HQ.  Are you using dhcp helper or local dhcp servers?  Please
elaborate...

Thanks,

Derek

On Fri, Jan 4, 2013 at 1:48 AM, SAIKAT SEN  wrote:

> Hello Friends !!
> I am practicing my own lab. I had created three
> sites, HQ , SiteB, SiteC and I was trying to register local phones with
> those sites. SiteC local phone register with CUCM-SUB easily, but having
> trouble to register SiteB and HQ local phones  with CUCM-SUB. I am using
> PhoenViewer to access phones remotely. I was getting phone display and it
> was showing DN number which one I configure in SiteC, but when I was trying
> to refresh the phone. IT showing error and could not able to dail any
> number. Please !!! help friends !!! your help will be much appreciated !!
>
> ___
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> visit www.ipexpert.com
>
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> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] Unity connection recording for UCCX prompts

[Derek Wyss]

I would recommend knowing how to do it both ways as certain circumstances
might require it.

Derek

On Thu, Jan 3, 2013 at 11:21 PM, William Bell  wrote:

> I assume everyone has their own approach here. I do the following:
>
> 1. For Unity Connection recordings (call handlers) I use CUGA
>
> 2. For UCCX prompts, I write a script in UCCX and record/upload the
> prompts from the UCCX server
>
> 3. For BACD prompts, I use the UCCX to record the prompt / upload to
> UCM-TFTP / TFTP copy the file to flash
>
> 4. For CUE prompts, I use the CUE prompt management app
>
> -Bill
> --
> William Bell
> blog: http://ucguerrilla.com
> twitter: @ucguerrilla
>
>
>
> On Jan 4, 2013, at 12:02 AM, singh wrote:
>
>
> HI Guys,
>
> I am planning to use Unity connection to record and download prompts for
> the UCCX scripts . I am just wondering if this is the best approach or a
> recording script needs to be written on UCCX.
>
>
> Also from machine on which UCCX is installed can the Unity connection web
> interface be accessed directly ?
>
>
> -singh
>
>
> Get Yourself a cool, short *@in.com* Email ID 
> now!
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
___
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Re: [OSL | CCIE_Voice] AAR Requirements / Implications

[Derek Wyss]

Bill,

You are absolutely right it depends and while we all hope the test would
clearly identify the "success criteria" my experience has told me that
certain things are also implied as opposed to being spelled out.  I thought
I would poll the audience.  Thank you for the feedback.

Derek

On Thu, Jan 3, 2013 at 7:40 PM, William Bell  wrote:

> Derek,
>
> That is an excellent question. The answer is the ubiquitous: "it depends".
> It is possible the question may state something like "you are done with
> this task when Phone A at Site 2 can call Phone A at Site 1" or something
> similar that basically should be read as "our script is going to do this
> call, make sure it works.". It is also possible that the question won't
> specify any testing parameters. In this case, I assume any call test is
> possible. I'd actually even expect the test to ask the candidate to ensure
> that integration to voicemail is still functional.
>
> -Bill
>
>
> --
> William Bell
> blog: http://ucguerrilla.com
> twitter: @ucguerrilla
>
>
>
> On Jan 3, 2013, at 5:30 PM, Derek Wyss wrote:
>
> List,
>
> I have myself wondering about the implied requirements of AAR involvement
> in the lab.  Let's say that AAR is a requirement between Site 1 and Site 2
> and system devices such as UCCX and UCON live in Site 2.   Is it implied
> that you need to provide for AAR services to those UCCX / UCON system
> devices (Ports, Pilot Pts)?  Or is it safe to provision AAR routing for
> only the phones at Site 1 and Site 2?
>
> Just curious on others thoughts.  Feedback is much appreciated.
>
> Thanks,
>
> Derek
> ___
> 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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[OSL | CCIE_Voice] AAR Requirements / Implications

[Derek Wyss]

List,

I have myself wondering about the implied requirements of AAR involvement
in the lab.  Let's say that AAR is a requirement between Site 1 and Site 2
and system devices such as UCCX and UCON live in Site 2.   Is it implied
that you need to provide for AAR services to those UCCX / UCON system
devices (Ports, Pilot Pts)?  Or is it safe to provision AAR routing for
only the phones at Site 1 and Site 2?

Just curious on others thoughts.  Feedback is much appreciated.

Thanks,

Derek
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Re: [OSL | CCIE_Voice] Procedure to check if Unity Connection , UCCX and Presence server ( CUPS) are working with CUCM or not?

[Derek Wyss]

I would suggest ensuring the integration point is set up properly.  Things
could work fine, but you would still lose points.  Example - UCON ports
registered to PUB instead of SUB.  I would suggest the same approach as
UCCX.  I would always go through the entire config as if I were doing it
instead of assuming everything is correct.  It won't be.

Derek

On Thu, Dec 20, 2012 at 10:18 PM, Cory Gray wrote:

> IF CUC and CUCCX were already integrated for you, then dial the hunt pilot
> on CUC and CTI route point for CUCCX.  Presence is not dial able so not
> sure there.  You would have to have a phone registered first though.  I
> just cross my fingers that they will work for me when I get there because
> in my opinion it would be too much out of the way to test up front.  Even
> if you found something, good luck convincing the proctor that there is
> really something wrong
>
> Sent from my iPhone
>
> On Dec 20, 2012, at 11:07 PM, "Symon Phares" 
> wrote:
>
> you will need to do the tests from cup n unity connection side. On unity
> connection, run check configuration from phone system menu and on cups on
> system troubleshooter dasshboard.
>
> Regards
> Symon Phares
>
> --
> From: sanity insanity 
> Sent: Friday, December 21, 2012 06:45
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] Procedure to check if Unity Connection , UCCX
> and Presence server ( CUPS) are working with CUCM or not?
>
> hi Guys,
>
> I was just wondering if there is a short procedure or check we can use
> before starting any configurations  just to see if the integration of CUCM
> with the following servers is working fine or not...
>
> 1) Cisco Unity Connection
> 2) UCCX
> 3) CUPS
>
>
> -I know if can to ping tests.  Is there any other way to confirm if the
> integration is working fine?
>
> Thanks,
> MJ
>
> ___
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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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> visit www.ipexpert.com
>
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> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] Meet me conference - Unity connection.

[Derek Wyss]

You must create two meet me partitions.  One directly dialable to open the
bridge and one that routes off to CUCM then comes back from UCON to the
previously mentioned.  You use a call handler to supervise transfer the
call back to CUCM after "asking for callers name."  It is important to
understand UCON's order of operations for this.

Derek

On Fri, Dec 21, 2012 at 5:04 AM, virajith  wrote:

>
> hi All,
>
> Do anyone know to configure unity connection such that when a phone user
> presses the Meet me
> conf button ..it announces the participant name and when somebody dials
> the meet me number
> it asks " who may I say is calling"
>
> Please elaborate the configuration steps used if possible?
>
> - Vir
>
>
> 
> Catch India as it happens with the *Rediff News App*. To download it for
> FREE, click 
> here
>
> ___
> 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] Hyphothetical Voice Infrastructure Issue Question

[Derek Wyss]

CDP advertise enabled on the switch?
Verified DHCP scopes are correct?  IE Contain the right default
gateway,network, mask etc...

On Thu, Dec 13, 2012 at 6:38 AM, Michael Davis
wrote:

> Scenario for campus intrasture problem:
>
> Phones are not registering. Phones are obtaining a valid IP address and
> have been erased to ensure they are getting a fresh IP address.
>
> Solutions investigated
>
> Verified VLAN's are correct and active
> Verified option 150 is correct in the DHCP options
> Verified Helper IP address is on was  the voice vlan's when in different
> subnet
> TFTP and DHCP on CM was restarted.
> NTP was verified and CM DB replication is normal.
>
> Idea that not checkd: WAN Qos may have been enabled thereby causing phone
> registration issues.
>
> Am I on the right track?
>
> Michael Davis
>
>
>
> ___
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> visit www.ipexpert.com
>
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> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] What access we will have in the lab

[Derek Wyss]

Yes

Sent from my iPhone

On Dec 5, 2012, at 8:52 AM, Chrysostomos Christofi  
wrote:

> Hi Derek
>  
> To the below link for example I will have access?
>  
> Its configuration guide and not configuration example
>  
> http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/cme40tcl.html
>  
> From: Derek Wyss [mailto:wys...@gmail.com] 
> Sent: Τετάρτη, 5 Δεκεμβρίου 2012 4:51 μμ
> To: Chrysostomos Christofi
> Cc: Online Study (ccie_voice@onlinestudylist.com)
> Subject: Re: [OSL | CCIE_Voice] What access we will have in the lab
>  
> You have access to the products page and all associated docs. 
> 
> Sent from my iPhone
> 
> On Dec 5, 2012, at 7:39 AM, Chrysostomos Christofi  
> wrote:
> 
> Hi folks
>  
> I am wondering if we will have access in the configuration guides
>  
> For example this one:
>  
> http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/cme40tcl.html
>  
>  
> Regards
> cc
> ___
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> visit www.ipexpert.com
> 
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> www.PlatinumPlacement.com
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Re: [OSL | CCIE_Voice] What access we will have in the lab

[Derek Wyss]

You have access to the products page and all associated docs. 

Sent from my iPhone

On Dec 5, 2012, at 7:39 AM, Chrysostomos Christofi  
wrote:

> Hi folks
>  
> I am wondering if we will have access in the configuration guides
>  
> For example this one:
>  
> http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/cme40tcl.html
>  
>  
> Regards
> cc
> ___
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> visit www.ipexpert.com
> 
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> www.PlatinumPlacement.com
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Re: [OSL | CCIE_Voice] Big UCCX worry

[Derek Wyss]

Also script is basically the simple queuing script just remove the prompt 
steps. 

Sent from my iPhone

On Dec 5, 2012, at 8:22 AM, Tanner Ezell  wrote:

> Setup after call work timer (configured on the CSQ configuration page) to 
> give your user time after the call has ended. This will apply to everyone who 
> is taking calls from that queue.
> 
> On Wed, Dec 5, 2012 at 2:21 AM, sanity insanity 
>  wrote:
> HI Guys,
> 
> I am trying to write a UCCX script for the following...
> 
> Trigger 24044000 called from PSTN or from internally to 4000, it should be
> directed to SC Phone 1 4101 or 4102 SC Phone 2 depending on longest idle
> time.
> Configure ip phone service for one button login for these agents
> Write another script which will transfer calls to available agent based on
> longest idle time. Phone 2 user has requested some time after ending a
> call. During this time, Phone 2 user should be marked NOT READY.
> User has to change status to READY manually before call can be directed
> 
> 
> Can someone help me with how to proceed? I know very little on scripting.
> 
> 
> -MJ
> 
> ___
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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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> visit www.ipexpert.com
> 
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Re: [OSL | CCIE_Voice] clear and remove qos and mls commands from switch

[Derek Wyss]

If you are trying to get most of your LAN qos from auto qos such as the police 
map and dscp map then you would run auto qos then copy out the relevant config 
and reload the sw without writing memory. The only other way would be to no out 
the cmds individually as far as I know. 

Sent from my iPhone

On Dec 4, 2012, at 2:45 AM, "virajith "  wrote:

> hi guys,
> 
> I am unable to clear / remove the mls and qos commands from my switch...
> 
> 
> Switch#sh running-config | b mls
> mls qos map cos-dscp 0 8 16 24 32 46 48 56
> mls qos srr-queue input bandwidth 90 10
> mls qos srr-queue input threshold 1 8 16
> mls qos srr-queue input threshold 2 34 66
> mls qos srr-queue input buffers 67 33
> mls qos srr-queue input cos-map queue 1 threshold 2  1
> mls qos srr-queue input cos-map queue 1 threshold 3  0
> mls qos srr-queue input cos-map queue 2 threshold 1  2
> mls qos srr-queue input cos-map queue 2 threshold 2  4 6 7
> mls qos srr-queue input cos-map queue 2 threshold 3  3 5
> mls qos srr-queue input dscp-map queue 1 threshold 2  9 10 11 12 13 14 15
> mls qos srr-queue input dscp-map queue 1 threshold 3  0 1 2 3 4 5 6 7
> mls qos srr-queue input dscp-map queue 1 threshold 3  32
> mls qos srr-queue input dscp-map queue 2 threshold 1  16 17 18 19 20 21 22 23
> mls qos srr-queue input dscp-map queue 2 threshold 2  33 34 35 36 37 38 39 48
> mls qos srr-queue input dscp-map queue 2 threshold 2  49 50 51 52 53 54 55 56
> mls qos srr-queue input dscp-map queue 2 threshold 2  57 58 59 60 61 62 63
> mls qos srr-queue input dscp-map queue 2 threshold 3  24 25 26 27 28 29 30 31
> mls qos srr-queue input dscp-map queue 2 threshold 3  40 41 42 43 44 45 46 47
> mls qos srr-queue output cos-map queue 1 threshold 3  5
> mls qos srr-queue output cos-map queue 2 threshold 3  3 6 7
> mls qos srr-queue output cos-map queue 3 threshold 3  2 4
> mls qos srr-queue output cos-map queue 4 threshold 2  1
> mls qos srr-queue output cos-map queue 4 threshold 3  0
> mls qos srr-queue output dscp-map queue 1 threshold 3  40 41 42 43 44 45 46 47
> mls qos srr-queue output dscp-map queue 2 threshold 3  24 25 26 27 28 29 30 31
> mls qos srr-queue output dscp-map queue 2 threshold 3  48 49 50 51 52 53 54 55
> mls qos srr-queue output dscp-map queue 2 threshold 3  56 57 58 59 60 61 62 63
> mls qos srr-queue output dscp-map queue 3 threshold 3  16 17 18 19 20 21 22 23
> mls qos srr-queue output dscp-map queue 3 threshold 3  32 33 34 35 36 37 38 39
> mls qos srr-queue output dscp-map queue 4 threshold 1  8
> mls qos srr-queue output dscp-map queue 4 threshold 2  9 10 11 12 13 14 15
> mls qos srr-queue output dscp-map queue 4 threshold 3  0 1 2 3 4 5 6 7
> mls qos queue-set output 1 threshold 1 138 138 92 138
> mls qos queue-set output 1 threshold 2 138 138 92 400
> mls qos queue-set output 1 threshold 3 36 77 100 318
> mls qos queue-set output 1 threshold 4 20 50 67 400
> mls qos queue-set output 2 threshold 1 149 149 100 149
> mls qos queue-set output 2 threshold 2 118 118 100 235
> mls qos queue-set output 2 threshold 3 41 68 100 272
> mls qos queue-set output 2 threshold 4 42 72 100 242
> mls qos queue-set output 1 buffers 10 10 26 54
> mls qos queue-set output 2 buffers 16 6 17 61
> !
> 
> class-map match-all AutoQoS-VoIP-RTP-Trust
>  match ip dscp ef
> !
> !
> policy-map AutoQoS-Police-CiscoPhone
>  class AutoQoS-VoIP-RTP-Trust
>   set dscp ef
>   police 32 8000 exceed-action policed-dscp-transmit
> !
> 
> 
> 
> -- How do I clear the above in one or 2 commands?
> 
> 
> 
> I have tried do no auto qos voip and no mls qos still no good.
> 
> 
> Thanks,
> Vir
> 
> 
> 
> 
> 
> 
> Catch India as it happens with the Rediff News App. To download it for FREE, 
> click here
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> visit www.ipexpert.com
> 
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> www.PlatinumPlacement.com
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[OSL | CCIE_Voice] Time Tracking Excel Template

[Derek Wyss]

Hi all,

New member here.  I am looking for others' opinions on the best way to track 
time/progress as they prepare for the CCIE Voice Lab exam?  I have looked into 
a few excel templates, but am looking to see if anyone has any recommendations 
on what is working for them.

Thanks,

Derek Wyss
Unified Communications Engineer


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