Re: [OSL | CCIE_Voice] UCCX agent routing and script

[Michael.Sears]

Can you call from HQ Phone 1 to 4101?   By chance do you have RSVP configured?  
I had same issue and in my case RSVP wasn't working.  If you have it setup the 
easy way to test is to turn off mandatory in locations and try to call from HQ 
Phone 1 to 4101 again.  If this works try UCCX again.  This may not be your 
problem, but I had exactly the same problem when RSVP wasn't working correctly 
and it sounds like UCCX is trying to forward the call so this may not be a 
problem with UCCX but another issue outside of UCCX call routing, i.e. being 
able to forward calls through HQ.

Hope this helps.

Michael Sears
CCIE Voice 38404



___
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

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

[Chadi H Hassoune (chassoun)]

Calling Party Xform and assign it to the CUC Device Pool works fine for me.

HTH

From: Pixar Perfect mailto:pixarperf...@live.com>>
Date: Wednesday, March 20, 2013 11:43 PM
To: "Mark Thrash (marthras)" mailto:marth...@cisco.com>>, 
Steve Keller mailto:skeller...@gmail.com>>
Cc: CCIE Voice OSL 
mailto:ccie_voice@onlinestudylist.com>>
Subject: Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension

the requirement is always for SiteB calling into SiteA voicemail by hitting 
Messages button. SiteA is always MGCP gateway. Calling Party Xform isnt any use 
on MGCP gateway for incoming calls.

here is another way of doing it ...

Voicemail Pilot for CUC is 2200

call-manager-fallback
voicemail 2777   ---> siteB specific

translation-pattern on CUCM to convert 2777 into 2200 and mask calling number 
. The CSS of the translation pattern should have access to 2200.



there is no definitive answer as to which solution is graded positively. there 
is a reason why many leading CCIE instructors say this is not a test of best 
practices but a test of how like able is your solution to the script. .. :)






From: marth...@cisco.com
To: skeller...@gmail.com
Date: Thu, 21 Mar 2013 03:59:48 +
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension

What about a calling party transform mask on the incoming gateway?

Sent from my iPhone

On Mar 20, 2013, at 10:43 PM, "Steve Keller" 
mailto:skeller...@gmail.com>> wrote:

Thanks Bill, I like this option pretty well as it seems to limit treatment of 
calls this way to CUC when site B is in SRST mode only.  I will try to lab this 
up tomorrow morning. Question for you, will this only solve my issue of 
pressing the VM button to access my mailbox to retrieve a message. Meaning when 
PSTN calls in to site B phone and then gets forward(redirected) to voicemail, I 
use a dial-peer that provides RDNIS capabilites to route the caller to the 
correct mailbox and not the opening greeting. So with this would i still want 
to use the following to get the caller into my mailbox?

dial-peer voice 2600 pots
destination-pattern 2600
port 0/0/0:23
no digit-strip
prefix 202555 ( assuming no LD code at this site )

this is the way i get callers into my mailbox - using RDNIS.

On Wed, Mar 20, 2013 at 10:49 PM, William Bell 
mailto:b...@ucguerrilla.com>> wrote:
If you are told National calls must present a 10D ANI  AND you are restricted 
from using an alternate extension in CUC then I do the following. I am not sure 
whether this would be graded right or wrong

On the SRST device (assume basic SRST)


call-manager-fallback
 max-ephone 10
 max-dn 20 oct
 huntst chan 1
 voicemail 912025552699   ! or some unused DID on Site A
 call-forward noan 912025552600 time 20   !assume VM pilot is 2600
 call-forward busy 912025552600
 time-z 
 time-f 
 date-f 
 call-forward pattern .T
!



On CUCM:

Create a PT:   hq_gw-in_pt
Create a CSS: hq_gw_css

Assign CSS to hq gateway

Either

a.) create a translation in hq_gw-in_pt
Pattern: 2699
xform ANI: 
xform DNIS: 2600! as in, redirect to regular VM pilot
CSS: your regular HQ phone CSS will do

OR

b.) create a new hunt pilot in hq_gw-in_pt
Pattern: 2699
HL: your VM HL
xform ANI:


Why would I go this path?

1. We had a requirement that National calls are presented with a 10D ANI in 
SRST mode. I assume that you would already have a translation-p that handles 
this bit

2. We can't modify the CUC subscriber.

3. This method doesn't interfere with RDNIS to VM

4. This method doesn't interfere with direct or redirect calls from HQ or SiteC


Anyway, that is my 2 cents.

-Bill

--
William Bell
blog: http://ucguerrilla.com
twitter: @ucguerrilla



On Mar 20, 2013, at 9:33 PM, Bill wrote:

Traditionally you would use the alternate extension or a  on the pilot.  So 
if you we're denied the ability to use alternate extension for this task but 
had to use it for another, say allowing easy voicemail access to a user at 
home, then I think you are looking at a very specific inbound translation on 
your gateway or nay sending 4 digits if the PSTN allows.  I would definitely 
test out the translation setup to ensure you can do it.

Sent from my iPad

On Mar 20, 2013, at 3:44 PM, Steve Keller 
mailto:skeller...@gmail.com>> 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 972303300

Re: [OSL | CCIE_Voice] UCCX agent routing and script verification!!

[William Bell]

> 
> 1) when I call 4000 I can hear the greeting  saying "Press 1 to be 
> transferred to priority agent or stay online for next available agent" . The 
> call does
> not ring on the ipcc agent phones of 4101 and 4102 but when I press 1 it is 
> transferred to 4001 as expected.  My question is what is preventing it from
> ringing 4101 and 4102 even though the agents are in a "Ready" state?



Given the way you presented your script logic this behavior is expected. You 
are asking the contact to press 1 and handling the transfer action prior to  
the Select Resource step. 

> 2) The resources set for 4101 and 4102 are in Resource group name "S"  and 
> the CSQ for this is named as "CSQ". The resource criteria is " Longest 
> Available".
> Is this correct?


"Longest Idle" == "Longest Available"

> 3) Any other parameter that needs to be checked under the Resource group or 
> the CSQ?


Can't say. Assuming you have configured your resources and CSQ correctly and 
you have properly employed either Resource Group or Skills based routing then I 
think you are OK. If you have failed to configure resources/CSQ/etc. correctly 
then you are not OK.

> 4)Is the configuration steps correct ? What steps are missing if any and how 
> do we correct it? Is the script correct?


Is something not behaving the way you want or expect it to? If yes, then 
something is provisioned incorrectly.

Your script has a logic flaw. 
> -option 1:-
>a) call redirect to 4001
>b) If successful goto queueLoop


Step (b) doesn't make sense to me. If you successfully redirect the contact 
then the script logic shouldn't go to the queueLoop. You should terminate.




--
William Bell
blog: http://ucguerrilla.com
twitter: @ucguerrilla



On Mar 21, 2013, at 1:03 PM, sanity insanity wrote:

> Hi All,
> 
> Need your help.
> 
> 
> I am configuring DNs 4101 & 4102   ( both DNs are uccx agent extensions). 
> Calls to 4000 should here a greeting "Press 1 to be transferred to priority 
> agent or stay online for next available agent" . If the caller presses 1 , 
> calls should be transferred to 4001.
> Otherwise it should be hunted as per "Longest idle time".
> 
> 
> These are the configuration steps I followed --
> 
> 1) recorded a prompt for the greeting called operator.wav
> 
> 2) Configured one button login for the phone dns ( agent DNs - 4101 & 4102)
> 
> 3) Setup the CSQ and resources in UCCX
> 
> 
> 4) Wrote the following script...
> 
> -start
> -accept
> -play prompt ( welcome prompt)
> -menu( triggering contact , "operator.wav")
> -option 1:-
>a) call redirect to 4001
>b) If successful goto queueLoop
> - Under Select Resource ( triggering contact - from CSQ) :-
>a)queueLoop:
> -End
> 
> 
> 5) Configured a trigger for 4000
> 
> 
> 
> Questions :
>  
> 
> 1) when I call 4000 I can hear the greeting  saying "Press 1 to be 
> transferred to priority agent or stay online for next available agent" . The 
> call does
> not ring on the ipcc agent phones of 4101 and 4102 but when I press 1 it is 
> transferred to 4001 as expected.  My question is what is preventing it from
> ringing 4101 and 4102 even though the agents are in a "Ready" state?
> 
> 
> 2) The resources set for 4101 and 4102 are in Resource group name "S"  and 
> the CSQ for this is named as "CSQ". The resource criteria is " Longest 
> Available".
> Is this correct?
> 
> 
> 3) Any other parameter that needs to be checked under the Resource group or 
> the CSQ?
> 
> 
> 4)Is the configuration steps correct ? What steps are missing if any and how 
> do we correct it? Is the script correct?
> 
> 
> - MJ
> 
> ___
> 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

[OSL | CCIE_Voice] UCCX agent routing and script verification!!

[sanity insanity]

Hi All,

Need your help.


I am configuring DNs 4101 & 4102   ( both DNs are uccx agent extensions).
Calls to 4000 should here a greeting "Press 1 to be transferred to priority
agent or stay online for next available agent" . If the caller presses 1 ,
calls should be transferred to 4001.
Otherwise it should be hunted as per "Longest idle time".


These are the configuration steps I followed --

1) recorded a prompt for the greeting called operator.wav

2) Configured one button login for the phone dns ( agent DNs - 4101 & 4102)

3) Setup the CSQ and resources in UCCX


4) Wrote the following script...

-start
-accept
-play prompt ( welcome prompt)
-menu( triggering contact , "operator.wav")
-option 1:-
   a) call redirect to 4001
   b) If successful goto queueLoop
- Under Select Resource ( triggering contact - from CSQ) :-
   a)queueLoop:
-End


5) Configured a trigger for 4000



Questions :


1) when I call 4000 I can hear the greeting  saying "Press 1 to be
transferred to priority agent or stay online for next available agent" .
The call does
not ring on the ipcc agent phones of 4101 and 4102 but when I press 1 it is
transferred to 4001 as expected.  My question is what is preventing it from
ringing 4101 and 4102 even though the agents are in a "Ready" state?


2) The resources set for 4101 and 4102 are in Resource group name "S"  and
the CSQ for this is named as "CSQ". The resource criteria is " Longest
Available".
Is this correct?


3) Any other parameter that needs to be checked under the Resource group or
the CSQ?


4)Is the configuration steps correct ? What steps are missing if any and
how do we correct it? Is the script correct?


- MJ
___
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

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

[Steve Keller]

Thanks to all for all of your input on this question. Based on my testing
of the suggestions provided we are essentially always stripping down the
ANI to 4 digits before routing to CUC, that is really our only option since
I have stated that Alternate Extension is not allowed in my question. With
this technique , if the ANI is read back to the message recipient it will
be chopped down to the last 4 digits. I suppose this is okay as long as
there is no requirement to play Senders ANI before the message ( which is
off by default ) . I dont see a way around this drawback. But the  on
the hunt pilot calling party transform mask seems to do the job. If I were
to see a question like this on the lab exam, i hope the grading gods will
be on my side.

I do not think you can have it both ways, either CUC gets 4 digit ANI ,
finds the mailbox and prompts to sign in - and the ANI read back to the
message recipient is 4 digits, or CUC gets 10 digits and you must use the
Alternate Extension field to find the mailbox.

On Thu, Mar 21, 2013 at 12:43 AM, Pixar Perfect wrote:

> the requirement is always for SiteB calling into SiteA voicemail by
> hitting Messages button. SiteA is always MGCP gateway. Calling Party Xform
> isnt any use on MGCP gateway for incoming calls.
>
> here is another way of doing it ...
>
> Voicemail Pilot for CUC is 2200
>
> call-manager-fallback
> voicemail 2777   ---> siteB specific
>
> translation-pattern on CUCM to convert 2777 into 2200 and mask calling
> number . The CSS of the translation pattern should have access to 2200.
>
>
>
> there is no definitive answer as to which solution is graded positively.
> there is a reason why many leading CCIE instructors say this is not a test
> of best practices but a test of how like able is your solution to the
> script. .. :)
>
>
>
>
>
> --
> From: marth...@cisco.com
> To: skeller...@gmail.com
> Date: Thu, 21 Mar 2013 03:59:48 +
>
> CC: ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate
> Extension
>
> What about a calling party transform mask on the incoming gateway?
>
> Sent from my iPhone
>
> On Mar 20, 2013, at 10:43 PM, "Steve Keller"  wrote:
>
>   Thanks Bill, I like this option pretty well as it seems to limit
> treatment of calls this way to CUC when site B is in SRST mode only.  I
> will try to lab this up tomorrow morning. Question for you, will this only
> solve my issue of pressing the VM button to access my mailbox to retrieve a
> message. Meaning when PSTN calls in to site B phone and then gets
> forward(redirected) to voicemail, I use a dial-peer that provides RDNIS
> capabilites to route the caller to the correct mailbox and not the opening
> greeting. So with this would i still want to use the following to get the
> caller into my mailbox?
>
> dial-peer voice 2600 pots
> destination-pattern 2600
> port 0/0/0:23
> no digit-strip
> prefix 202555 ( assuming no LD code at this site )
>
> this is the way i get callers into my mailbox - using RDNIS.
>
>  On Wed, Mar 20, 2013 at 10:49 PM, William Bell wrote:
>
> If you are told National calls must present a 10D ANI  AND you are
> restricted from using an alternate extension in CUC then I do the
> following. I am not sure whether this would be graded right or wrong
>
>  On the SRST device (assume basic SRST)
>
>
>  call-manager-fallback
>  max-ephone 10
>  max-dn 20 oct
>  huntst chan 1
>  voicemail 912025552699   ! or some unused DID on Site A
>  call-forward noan 912025552600 time 20   !assume VM pilot is 2600
>  call-forward busy 912025552600
>  time-z 
>  time-f 
>  date-f 
>  call-forward pattern .T
> !
> 
>
>
>  On CUCM:
>
>  Create a PT:   hq_gw-in_pt
> Create a CSS: hq_gw_css
>
>  Assign CSS to hq gateway
>
>  Either
>
>  a.) create a translation in hq_gw-in_pt
> Pattern: 2699
> xform ANI: 
> xform DNIS: 2600! as in, redirect to regular VM pilot
> CSS: your regular HQ phone CSS will do
>
>  OR
>
>  b.) create a new hunt pilot in hq_gw-in_pt
> Pattern: 2699
> HL: your VM HL
> xform ANI:
>
>
>  Why would I go this path?
>
>  1. We had a requirement that National calls are presented with a 10D ANI
> in SRST mode. I assume that you would already have a translation-p that
> handles this bit
>
>  2. We can't modify the CUC subscriber.
>
>  3. This method doesn't interfere with RDNIS to VM
>
>  4. This method doesn't interfere with direct or redirect calls from HQ
> or SiteC
>
>
>  Anyway, that is my 2 cents.
>
>  -Bill
>
>  --
> William Bell
> blog: http://ucguerrilla.com
> twitter: @ucguerrilla
>
>
>
>  On Mar 20, 2013, at 9:33 PM, Bill wrote:
>
>  Traditionally you would use the alternate extension or a  on the
> pilot.  So if you we're denied the ability to use alternate extension for
> this task but had to use it for another, say allowing easy voicemail access
> to a user at home, then I think you are looking at a very specific inbound
> translation on your gateway or nay sending 4

[OSL | CCIE_Voice] [Accelerate Learning]

[ie ravindra]

Hellow Experts,

I made my exam payments today for my coming in CCIE exam. until today I was
not went through lab practice much. But I am expecting to start it today.
Is there any way to accelerate learning practice since I am busy with
office tasks also. This is my first attempt.

Thanks,
Ravi.
___
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