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

[Somphol Boonjing]

Sorry for revisiting this old thread.   The Calling Party Transformation at
the Device Pool level would come in handy for this particular need.

In the document starting 7.1.2, this is stated explicitly,
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_1_2/ccmfeat/fscallpn.html#wp1325305
.

*Cisco Unity/Cisco Unity Connection*


Because no calling party transformation options exist in Cisco Unified
Communications Manager Administration for voice-messaging ports, make sure
that you configure the calling party number transformations in the device
pool that is associated with the voice-messaging ports.
...

Table 7-8 Configuring the Calling Party Transformation CSS to Localize the
Calling Party Number
Also mentioned "Use Device Pool Calling Party Transformation CSS" as a
method to Localize the Calling Party Number.
...
...

The same document for 7.0.1 contained the table 7-8, but somehow doesn't
have that explicit section on Cisco Unity/Cisco Unity Connection's calling
party localization.  (REF:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmfeat/fscallpn.html#wp1276877)
   So, I am not so sure whether this is possible in CUCM 7.0.1/CUC 7.0.1.

I don't have lab access to test this now, but would appreciate if anyone
can help testing this.

Note: I recall seeing some sort of Technotes outlining the strategy to
perform Calling Party transformation for Call Manager 4.x or something that
doesn't rely on Gateway's Calling Party Transformation.I can't locate
it now, but if anyone could point me to the URL that would be great.

Regards,
--Somphol.





On Sat, Mar 23, 2013 at 12:34 AM, Leslie Meade wrote:

>  Easy way of doing this is to copy the hunt pilot and give it another
> number.. set user caller ID and mask it to 
>
> Then in the call-manager-fallback change the voicemail to the new hunt
> pilot and your done
>
> ** **
>
> ** **
>
> *Leslie Meade* 
>
> .. *
> Mobile:778.228.4339* | *Main:* *604.676.5239*
> *Email:* leslie.me...@lvs1.com
>
> ** **
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Chadi H Hassoune
> (chassoun)
> *Sent:* Thursday, March 21, 2013 7:10 PM
> *To:* Pixar Perfect; Mark Thrash (marthras); Steve Keller
> *Cc:* CCIE Voice OSL
>
> *Subject:* Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate
> Extension
>
>  ** **
>
> Calling Party Xform and assign it to the CUC Device Pool works fine for
> me. 
>
> ** **
>
> HTH
>
> ** **
>
> *From: *Pixar Perfect 
> *Date: *Wednesday, March 20, 2013 11:43 PM
> *To: *"Mark Thrash (marthras)" , Steve Keller <
> skeller...@gmail.com>
> *Cc: *CCIE Voice OSL 
> *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"  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 d

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

[Leslie Meade]

Easy way of doing this is to copy the hunt pilot and give it another number.. 
set user caller ID and mask it to 
Then in the call-manager-fallback change the voicemail to the new hunt pilot 
and your done


Leslie Meade


..
Mobile:778.228.4339 | Main: 604.676.5239
Email: leslie.me...@lvs1.com<mailto:leslie.me...@lvs1.com>

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Chadi H Hassoune 
(chassoun)
Sent: Thursday, March 21, 2013 7:10 PM
To: Pixar Perfect; Mark Thrash (marthras); Steve Keller
Cc: CCIE Voice OSL
Subject: Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension

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<mailto:marth...@cisco.com>
To: skeller...@gmail.com<mailto:skeller...@gmail.com>
Date: Thu, 21 Mar 2013 03:59:48 +
CC: ccie_voice@onlinestudylist.com<mailto: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

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<mailto:marth...@cisco.com>
To: skeller...@gmail.com<mailto:skeller...@gmail.com>
Date: Thu, 21 Mar 2013 03:59:48 +
CC: ccie_voice@onlinestudylist.com<mailto: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 num

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: @uc

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

[Pixar Perfect]

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-fallbackvoicemail 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 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  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 

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

[Mark Thrash (marthras)]

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 
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

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

[Steve Keller]

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 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  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
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking

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

[William Bell]

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  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
> ___
> 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

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

[Bill]

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  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
___
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

[Pixar Perfect]

As far as i remember the requirement under call routing doesn't specify the 
called party type "National" for long distance calls out SiteB gateway AND the 
question clearly said that YOU ARE ALLOWED TO SEND 4 DIGITS in Calling Number 
field to the Telco for these calls ... so why not use this dialpeer ..it still 
meets the requirement ... let me know if anyone has seen any different . 

dial-peer voice 2600 potsdescription voicemail-pilotdestination-pattern 2600no 
digit-stripport 0/0/0:23prefix 1408202

Mar 20 22:55:57.807: ISDN Se0/1/0:23 Q931: TX -> SETUP pd = 8  callref = 0x0084 
Bearer Capability i = 0x8090A2 Standard = CCITT 
Transfer Capability = Speech  Transfer Mode = Circuit   
  Transfer Rate = 64 kbit/s Channel ID i = 0xA98397 
Exclusive, Channel 23 Progress Ind i = 0x8183 - Origination 
address is non-ISDN  Display i = '+19723033002' Calling Party 
Number i = 0x0080, '3002' Plan:Unknown, Type:Unknown 
Called Party Number i = 0x80, '14082022600' Plan:Unknown, 
Type:Unknown
Date: Wed, 20 Mar 2013 17:01:48 -0500
From: wys...@gmail.com
To: skeller...@gmail.com
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension

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 potsdescription voicemail-pilotdestination-pattern 2600
no digit-stripport 0/0/0:23prefix 1408202If 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



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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 ___
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

[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
>>>
>>
>>
___
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 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
>>
>
>
___
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

[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
>
___
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] SRST to voicemail without Alternate Extension

[Steve Keller]

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