[OSL | CCIE_Voice] OFF TOPIC: Avaya ACS integration.

[ccielabrat]

To All,

I apologize for distracting you from your studying.

I'm looking for some input from anyone with Avaya ACS experience.

I'm trying to provide CME to Avaya Partner ACS integration while we migrate
to CME.

I currently have VIC-4FXO and VIC2-FXO connected to "extension" ports on
the avaya.
I'm able to call to the PSTN without issue, but can not dial internal two
digit extensions.
As I understand it , the user must press the "intercom" button prior to
dialing an internal extension.
A quick look at the avaya documentation indicates the system supports
"single-line" pots phones and should be able to dial internal extensions by
using a switch hook signal before dialing.

I'm trying to figure out if this switch hook support needs to be enabled
and also if enabling the "flash" button in CME would be the same as a
"switch hook" to the avaya.

Any ideas or opinions are greatly appreciated.
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[OSL | CCIE_Voice] CUE voiceview authentication headache :)

[ccielabrat]

Hi group,

I have my CUE configured with CUCM.
I can call into CUE voicemail and check messages, so I know my PIN is good.

I've subscribed my phone to the voiceview service and can get the
authentication url to show on the phone.
I login in with username scphX (thats the username) and the pin, but I get
"login failed" everytime.

Can anyone help me understand where the voiceview app is referencing for
authentication?
Does it change if the CUE module it integrated to CME or CUCM ?

Thanks
Lab Rat
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[OSL | CCIE_Voice] Question regarding dialing NATIONAL calls outside the U.S.

[CCIELabRat]

Hi All,

It's probably a silly question, but it's some I struggle with more often
than I like.

Can some clarify for me how a "LOCAL" and "NATIONAL" call generally need to
be dialed / presented outside the U.S.

In all the practice labs I've done, dialing international is usually
straight forward - something like 90014085551212.

But my head spins when trying to understand what is a national call verus a
local call.

Lets say I have a International e164 phone number at +743 2202 4xxx at my
Site C location.

Is a "local" number anything in the   format?
What would a National number in this example ?

It seems to me anything other than    , would be a International
call.
So I don't see where "National" fits in.

Also , when I see a number like XX   on the PSTN phone, is that
automatically a International call ?

- Lab Rat
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Re: [OSL | CCIE_Voice] MVA -"Your call can not be completed as dialed"

[CCIELabRat]

Ramy,

Can you clarify how the re-routing CSS is used?
Is the "regular" CSS ignored when a remote destination phone is dialing
through MVA?

-Scott


On Tue, Aug 7, 2012 at 4:34 PM, Ramy Abdelrahim wrote:

>  Guys,
>
> Re-routing CSS is used for SNR but CSS is used for MVA.
>
> Therefore, you've to check if the number you're dialing is in a route
> pattern that the configured CSS in the remote destination profile can
> reach. I recommend to separate the dial plan for MVA or SNR as follows.
>
> - Create a new partition.
> - Create a new CSS containing the above new partition.
> - Create a new route list
> - Create a new route pattern.
>
> hope this will help.
>
> Regards,
> Ramy
>
> --
> Date: Tue, 7 Aug 2012 16:21:17 -0400
> From: ccielab...@gmail.com
> To: vipji...@cisco.com
> CC: ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] MVA -"Your call can not be completed as
> dialed" b
>
>
> Thank you Vipul.
> I thought re-routing CSS was used only for the SNR configuration.
> I will take a look and test again.
> :)
>
> On Tue, Aug 7, 2012 at 3:29 PM, Vipul Jindal (vipjinda) <
> vipji...@cisco.com> wrote:
>
>  It uses the re routing CSS on the remote destination number.
>
>  If you check the call manager traces, you can easily check it.
>
>
>
>   From: ccielabrat 
> Date: Tuesday, August 7, 2012 2:09 PM
> To: "ccie_voice@onlinestudylist.com" 
> Subject: [OSL | CCIE_Voice] MVA -"Your call can not be completed as
> dialed"
>
>  To All,
>
> I'm hoping the group can help me understand the call flow for an MVA call.
> I'm able to call into the MVA pilot number , have my remote destination
> number recognized and be prompted for my PIN and to dial .
>
> But I get the message "Your call can not be completed as dialed" for
> anything I try to call.
>
> I understand that the number configured under the mobile voice access page
> is used as an "anchor" , as per Vik's vlecture, but I'm unclear what device
> is referenced regarding CSS and what should and should be reachable.
>
> Can anyone please help get closure on this last piece of the puzzle.
>
> -Lab Rat
>
>
> ___
> 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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> 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] MVA -"Your call can not be completed as dialed"

[CCIELabRat]

Actually, the MVA Prompt was working.

I believe the issue was a owner/device mismatch actually.
I did try to adjust  the re-route CSS but it didn't change the behavior.

I actually deleted the user (which surprising wipes all the remote
destination info for the user also) and recreated it.

It's all working as expected now.

Thanks for the clarification regarding Re-routing CSS.
I didn't think it would apply here.



On Tue, Aug 7, 2012 at 4:18 PM, Krishna  wrote:

> Rat,
>
> make sure mva is enabled on the service parameters and number as well in
> the cucm service parameters, and also check with the dial-peer and
> application url on the router with the right number...
>
> Vipul.
>
> it uses rerouting css when it makes outbound calls, but in this case he
> can't even get to the prompt of mva...
>
> thank you
> krishna.
>   --
> *From:* Vipul Jindal (vipjinda) 
> *To:* ccielabrat ; "ccie_voice@onlinestudylist.com"
> 
> *Sent:* Tuesday, August 7, 2012 2:29 PM
> *Subject:* Re: [OSL | CCIE_Voice] MVA -"Your call can not be completed as
> dialed"
>
>  It uses the re routing CSS on the remote destination number.
>
>  If you check the call manager traces, you can easily check it.
>
>
>
>   From: ccielabrat 
> Date: Tuesday, August 7, 2012 2:09 PM
> To: "ccie_voice@onlinestudylist.com" 
> Subject: [OSL | CCIE_Voice] MVA -"Your call can not be completed as
> dialed"
>
>  To All,
>
> I'm hoping the group can help me understand the call flow for an MVA call.
> I'm able to call into the MVA pilot number , have my remote destination
> number recognized and be prompted for my PIN and to dial .
>
> But I get the message "Your call can not be completed as dialed" for
> anything I try to call.
>
> I understand that the number configured under the mobile voice access page
> is used as an "anchor" , as per Vik's vlecture, but I'm unclear what device
> is referenced regarding CSS and what should and should be reachable.
>
> Can anyone please help get closure on this last piece of the puzzle.
>
> -Lab Rat
>
>
> ___
> 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] MVA -"Your call can not be completed as dialed"

[CCIELabRat]

Thank you Vipul.
I thought re-routing CSS was used only for the SNR configuration.
I will take a look and test again.
:)

On Tue, Aug 7, 2012 at 3:29 PM, Vipul Jindal (vipjinda)
wrote:

>  It uses the re routing CSS on the remote destination number.
>
>  If you check the call manager traces, you can easily check it.
>
>
>
>   From: ccielabrat 
> Date: Tuesday, August 7, 2012 2:09 PM
> To: "ccie_voice@onlinestudylist.com" 
> Subject: [OSL | CCIE_Voice] MVA -"Your call can not be completed as
> dialed"
>
>  To All,
>
> I'm hoping the group can help me understand the call flow for an MVA call.
> I'm able to call into the MVA pilot number , have my remote destination
> number recognized and be prompted for my PIN and to dial .
>
> But I get the message "Your call can not be completed as dialed" for
> anything I try to call.
>
> I understand that the number configured under the mobile voice access page
> is used as an "anchor" , as per Vik's vlecture, but I'm unclear what device
> is referenced regarding CSS and what should and should be reachable.
>
> Can anyone please help get closure on this last piece of the puzzle.
>
> -Lab Rat
>
>
> ___
> 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] MVA -"Your call can not be completed as dialed"

[ccielabrat]

To All,

I'm hoping the group can help me understand the call flow for an MVA call.
I'm able to call into the MVA pilot number , have my remote destination
number recognized and be prompted for my PIN and to dial .

But I get the message "Your call can not be completed as dialed" for
anything I try to call.

I understand that the number configured under the mobile voice access page
is used as an "anchor" , as per Vik's vlecture, but I'm unclear what device
is referenced regarding CSS and what should and should be reachable.

Can anyone please help get closure on this last piece of the puzzle.

-Lab Rat
___
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[OSL | CCIE_Voice] CUE and SIP-Notify

[CCIELabRat]

Just working on some CUE stuff today and stumbled across the option to
configure "dtmf-relay rtp-nte" under the sip subsystem of the CUE.

In general , I've always followed the consensus that you shoud use
RTP-NTE everywhere BUT when communicating with CUE which uses
"SIP-NOTIFY".

I wanted to reach out the group to see if anyone has input on any of
the implications of changing the SIP subsystem to  use "RTP-NTE"
instead of "SIP-NOTIFY"
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Re: [OSL | CCIE_Voice] BACD using builtin: services.

[ccielabrat]

Nevermind

I messed so much on the previous attempt , it's not worth looking at. :)

Here is a config that seems to work.
application
 service app-b-acd
  param queue-len 10
  param aa-hunt1 4010
  param number-of-hunt-grps 1
  param queue-manager-debugs 1
 !
 service app-b-acd-aa
  paramspace english index 0
  param number-of-hunt-grps 1
  param drop-through-option 1
  param handoff-string app-b-acd-aa
  paramspace english language en
  param max-time-vm-retry 2
  param aa-pilot 4000
  paramspace english location flash:/
  param second-greeting-time 60
  param drop-through-prompt _dt_prompt.au
  param call-retry-timer 15
  param voice-mail 4001
  param max-time-call-retry 700
  param service-name app-b-acd
 !

dial-peer voice 4000 pots
 service app-b-acd-aa
 incoming called-number 4000
 direct-inward-dial




On Sat, Dec 10, 2011 at 3:11 PM, ccielabrat  wrote:

> I'm trying to use the newer builtin b-acd services in my lab setup.
>
> Can anyone confirm my config and see if there is anything I'm missing?
> I'm using the usual B-acd documentation as a template.
>
> I'm getting the a problem where the call drops as soon as I call the pilot
> number
>
> application
>  service app-b-acd-aa
>   paramspace english index 0
>   param max-time-call-retry 700
>   param voice-mail 4001
>   param service-name app-b-acd
>   param number-of-hunt-grps 1
>   param drop-through-option 1
>   paramspace english language en
>   param handoff-string app-b-acd-aa
>   param max-time-vm-retry 2
>   paramspace english location flash:/
>   param app-b-acd-aa-pilot 4000
>   param drop-through-prompt _dt_prompt.au
>   param second-greeting-time 60
>   param call-retry-timer 15
>  !
>  service app-b-acd
>   param queue-len 10
>   param number-of-hunt-grps 1
>   param queue-manager-debugs 1
>  !
>
> dial-peer voice 4000 pots
>  service app-b-acd-aa
>  incoming called-number 4000
>  direct-inward-dial
>
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[OSL | CCIE_Voice] BACD using builtin: services.

[ccielabrat]

I'm trying to use the newer builtin b-acd services in my lab setup.

Can anyone confirm my config and see if there is anything I'm missing?
I'm using the usual B-acd documentation as a template.

I'm getting the a problem where the call drops as soon as I call the pilot
number

application
 service app-b-acd-aa
  paramspace english index 0
  param max-time-call-retry 700
  param voice-mail 4001
  param service-name app-b-acd
  param number-of-hunt-grps 1
  param drop-through-option 1
  paramspace english language en
  param handoff-string app-b-acd-aa
  param max-time-vm-retry 2
  paramspace english location flash:/
  param app-b-acd-aa-pilot 4000
  param drop-through-prompt _dt_prompt.au
  param second-greeting-time 60
  param call-retry-timer 15
 !
 service app-b-acd
  param queue-len 10
  param number-of-hunt-grps 1
  param queue-manager-debugs 1
 !

dial-peer voice 4000 pots
 service app-b-acd-aa
 incoming called-number 4000
 direct-inward-dial
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Re: [OSL | CCIE_Voice] UCCX and transcoding

[ccielabrat]

As I spend more time thinking about this, I think I'm complicating the
whole scenario.
As you mention, the RP is simply forwarding the call to a CTI Port.

The only time a transcoder would have to be invoked is for the aef script
when playing prompts.
Otherwise the handoff to an agent would follow the normal regions/codec
rules.



On Fri, Dec 9, 2011 at 1:55 PM, datucha123 datucha123
wrote:

> I cannot understand why do you have CTI RP and CTI ports in a different
> Device Pool?
>
> Also please ensure that the Transcoder is assigned the correct Device Pool
> - HQ Device Pool.
> Let me test it in my LAB tommorow as well, and will infrom you.
>
>
> On Fri, Dec 9, 2011 at 4:50 PM, ccielabrat  wrote:
>
>> So if that is the case, I guess it makes sense that the RP would need to
>> be in the HQ device pool , to force the call to g.729 based on the GW being
>> in the SiteB region and the RP being in the HQ region.
>>
>> I will test that today.
>>
>> Strange thing is if I call the RP from a phone in SiteB , the codec gets
>> set properly.
>> It's only when I call from PSTN via SiteB's Gw that the codec doesn't get
>> forced to g.729 as I would expect.
>>
>>
>>
>> On Fri, Dec 9, 2011 at 3:01 AM, datucha123 datucha123 <
>> datucha...@gmail.com> wrote:
>>
>>> Well I think you need to use the transcoding based on the CTI ports, not
>>> the Route Point, as the RP just forwards calls, there are not RTP stream
>>> ever to Route Point.
>>>
>>> On Fri, Dec 9, 2011 at 8:23 AM, ccielabrat  wrote:
>>>
>>>> I need some help clarifying where I need transcoding for a UCCX where
>>>> the trigger DN is located across the WAN (Site B)
>>>>
>>>> I have the MGCP GW in Site B in a Device Pool (SiteB)
>>>> UCCX CTI-RP  is in Device Pool (SiteB)
>>>> and CTI-Ports are all in Device Pool (HQ)
>>>>
>>>> Per my region configuration, g.729 is required between HQ and SiteB.
>>>>
>>>> My expectation is the call will arrive in SiteB and based on region
>>>> requirements, will be setup as a g.729 call.
>>>>
>>>> A transcoder would be needed at the UCCX side to get to g.711.
>>>>
>>>> Would I be correct in thinking the CTI-Port should invoke trancoding
>>>> out of the MRGL assigned to the HQ dev pool ?
>>>>
>>>> Currently the call comes in and I see the MGCP gateway with a g.711
>>>> call, so it appears it's going across the WAN as g.711.
>>>>
>>>> I know I'm missing some understanding here.
>>>> Any input is appreciated.
>>>>
>>>>
>>>>
>>>> ___
>>>> 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 <http://www.platinumplacement.com/>
>>>>
>>>
>>>
>>
>
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Re: [OSL | CCIE_Voice] UCCX and transcoding

[ccielabrat]

So if that is the case, I guess it makes sense that the RP would need to be
in the HQ device pool , to force the call to g.729 based on the GW being in
the SiteB region and the RP being in the HQ region.

I will test that today.

Strange thing is if I call the RP from a phone in SiteB , the codec gets
set properly.
It's only when I call from PSTN via SiteB's Gw that the codec doesn't get
forced to g.729 as I would expect.


On Fri, Dec 9, 2011 at 3:01 AM, datucha123 datucha123
wrote:

> Well I think you need to use the transcoding based on the CTI ports, not
> the Route Point, as the RP just forwards calls, there are not RTP stream
> ever to Route Point.
>
> On Fri, Dec 9, 2011 at 8:23 AM, ccielabrat  wrote:
>
>> I need some help clarifying where I need transcoding for a UCCX where the
>> trigger DN is located across the WAN (Site B)
>>
>> I have the MGCP GW in Site B in a Device Pool (SiteB)
>> UCCX CTI-RP  is in Device Pool (SiteB)
>> and CTI-Ports are all in Device Pool (HQ)
>>
>> Per my region configuration, g.729 is required between HQ and SiteB.
>>
>> My expectation is the call will arrive in SiteB and based on region
>> requirements, will be setup as a g.729 call.
>>
>> A transcoder would be needed at the UCCX side to get to g.711.
>>
>> Would I be correct in thinking the CTI-Port should invoke trancoding out
>> of the MRGL assigned to the HQ dev pool ?
>>
>> Currently the call comes in and I see the MGCP gateway with a g.711 call,
>> so it appears it's going across the WAN as g.711.
>>
>> I know I'm missing some understanding here.
>> Any input is appreciated.
>>
>>
>>
>> ___
>> 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 <http://www.platinumplacement.com/>
>>
>
>
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[OSL | CCIE_Voice] UCCX and transcoding

[ccielabrat]

I need some help clarifying where I need transcoding for a UCCX where the
trigger DN is located across the WAN (Site B)

I have the MGCP GW in Site B in a Device Pool (SiteB)
UCCX CTI-RP  is in Device Pool (SiteB)
and CTI-Ports are all in Device Pool (HQ)

Per my region configuration, g.729 is required between HQ and SiteB.

My expectation is the call will arrive in SiteB and based on region
requirements, will be setup as a g.729 call.

A transcoder would be needed at the UCCX side to get to g.711.

Would I be correct in thinking the CTI-Port should invoke trancoding out of
the MRGL assigned to the HQ dev pool ?

Currently the call comes in and I see the MGCP gateway with a g.711 call,
so it appears it's going across the WAN as g.711.

I know I'm missing some understanding here.
Any input is appreciated.
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Re: [OSL | CCIE_Voice] UCCX on VMWARE problems.

[CCIELabRat]

Hi Ash,

I stand corrected, I had uploaded the license file.
I was at the point right after the initial setup completes and prompts
you to close your browser.

At that point, it presented the jtapi error.

I got it working though.
I ran setup /x from the UCCX media image and uninstalled UCCX.
I then re-ran setup and have a clean UCCX server to mess with.

Thank you as always for jumping in to help.


On Thu, Dec 8, 2011 at 5:02 PM, Ashraf Ayyash  wrote:
> What do you mean ? Where are you at exactly in the CCX installation ?
> Screenshot?
> Ash
>
> On Thursday, December 8, 2011, ccielabrat  wrote:
>> I can't get to the point to upload the license.
>>
>>
>> On Thu, Dec 8, 2011 at 2:07 PM, Ashraf Ayyash 
>> wrote:
>>>
>>> do you have the License uploaded to the CCX , this can happen when you
>>> have no license
>>>
>>> Ash
>>>
>>> On Thu, Dec 8, 2011 at 12:20 PM, ccielabrat  wrote:
>>> > I just wanted to do some testing on UCCX so I booted a vmware image of
>>> > UCCX
>>> > that I've used before.
>>> > It's a fresh install with no integration.
>>> >
>>> > When I log in, it says the JTAPI is out of sync.
>>> > I've fixed the JTAPI problem related to moving C:\windows\java files to
>>> > c:\winnt\java
>>> >
>>> > It wants me to rerun jtapi sync, but the menus on the UCCX page page
>>> > will
>>> > not display correctly.
>>> > Do dropdown menus appear.
>>> >
>>> > Can anyone help?
>>> >
>>> >
>>> > ___
>>> > 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] UCCX on VMWARE problems.

[ccielabrat]

I can't get to the point to upload the license.


On Thu, Dec 8, 2011 at 2:07 PM, Ashraf Ayyash  wrote:

> do you have the License uploaded to the CCX , this can happen when you
> have no license
>
> Ash
>
> On Thu, Dec 8, 2011 at 12:20 PM, ccielabrat  wrote:
> > I just wanted to do some testing on UCCX so I booted a vmware image of
> UCCX
> > that I've used before.
> > It's a fresh install with no integration.
> >
> > When I log in, it says the JTAPI is out of sync.
> > I've fixed the JTAPI problem related to moving C:\windows\java files to
> > c:\winnt\java
> >
> > It wants me to rerun jtapi sync, but the menus on the UCCX page page will
> > not display correctly.
> > Do dropdown menus appear.
> >
> > Can anyone help?
> >
> >
> > ___
> > 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] UCCX on VMWARE problems.

[ccielabrat]

I just wanted to do some testing on UCCX so I booted a vmware image of UCCX
that I've used before.
It's a fresh install with no integration.

When I log in, it says the JTAPI is out of sync.
I've fixed the JTAPI problem related to moving C:\windows\java files to
c:\winnt\java

It wants me to rerun jtapi sync, but the menus on the UCCX page page will
not display correctly.
Do dropdown menus appear.

Can anyone help?
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Re: [OSL | CCIE_Voice] Calling party transformations on a gateway

[ccielabrat]

Hey Chris,

Great catch.
You were spot on with your advice.

Thanks for correcting my thinking.

I guess it doesn't make sense as I think about it now, but I thought by the
time the gateway got the call had "left" CUCM and any reference to external
# mask, etc. wouldn't be relevant.
So I was trying to get the send the + globalized # to the gw and have it
trim off the +.


On Thu, Dec 8, 2011 at 8:50 AM, ccielabrat  wrote:

> Thanks Chris,
>
> I'll look into that now.
> So , with this in mind, should I be able to "check" the use external #
> mask within the calling party transformation and then apply a "XX"
> mask? or do I have to match against the 4 digit DN and then prefix the DN
> in the mask?
>
>
>
>
> On Thu, Dec 8, 2011 at 8:42 AM, Chris Martin  wrote:
>
>> What pattern do you have setup for Calling Party Transformations?  You
>> mention you are seeing + Globalized numbers in your isdn q931 debug, this
>> leads me to believe you have a RP/RL with External Number Mask applied.  My
>> guess is you are trying to match on the Globalized Calling numbers in your
>> calling transformation patterns.
>>
>> The calling party transformation patterns need to be matched on the
>> pre-RP/RL digits.  IE: 5XXX not +12127775XXX.
>>
>> HTH,
>> Chris
>>
>> On Wed, Dec 7, 2011 at 10:13 PM, ccielabrat  wrote:
>>
>>> To All,
>>>
>>> I just needed to check to see if anyone knows about a problem using
>>> calling party transformations at the gateway level.
>>>
>>> I have a setup where I am send a fully globalized called and calling #
>>> to my gateways.
>>> I wanted to make all needed adjustments just before it goes to the PSTN.
>>>
>>> My transformation CSS's are setup properly and I can adjust the called
>>> party number
>>> but the calling party number will not adjust according to the calling
>>> party xform pattern.
>>>
>>> I can see the calling number is getting to the gw in "+" format based on
>>> what I see in Q.931 on the router.
>>>
>>> Anybody experience this behavior?
>>>
>>>
>>>
>>> ___
>>> 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] Calling party transformations on a gateway

[ccielabrat]

Thanks Chris,

I'll look into that now.
So , with this in mind, should I be able to "check" the use external # mask
within the calling party transformation and then apply a "XX" mask?
or do I have to match against the 4 digit DN and then prefix the DN in the
mask?



On Thu, Dec 8, 2011 at 8:42 AM, Chris Martin  wrote:

> What pattern do you have setup for Calling Party Transformations?  You
> mention you are seeing + Globalized numbers in your isdn q931 debug, this
> leads me to believe you have a RP/RL with External Number Mask applied.  My
> guess is you are trying to match on the Globalized Calling numbers in your
> calling transformation patterns.
>
> The calling party transformation patterns need to be matched on the
> pre-RP/RL digits.  IE: 5XXX not +12127775XXX.
>
> HTH,
> Chris
>
> On Wed, Dec 7, 2011 at 10:13 PM, ccielabrat  wrote:
>
>> To All,
>>
>> I just needed to check to see if anyone knows about a problem using
>> calling party transformations at the gateway level.
>>
>> I have a setup where I am send a fully globalized called and calling # to
>> my gateways.
>> I wanted to make all needed adjustments just before it goes to the PSTN.
>>
>> My transformation CSS's are setup properly and I can adjust the called
>> party number
>> but the calling party number will not adjust according to the calling
>> party xform pattern.
>>
>> I can see the calling number is getting to the gw in "+" format based on
>> what I see in Q.931 on the router.
>>
>> Anybody experience this behavior?
>>
>>
>>
>> ___
>> 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] Calling party transformations on a gateway

[ccielabrat]

It shows the full globalized + format of the calling number, which matches
the external number mask on the phone.

On Thu, Dec 8, 2011 at 12:37 AM, Abel ...  wrote:

> What your Q.931 debug tell you on the gateway and the PSTN?
>
> On Thu, Dec 8, 2011 at 12:13 AM, ccielabrat  wrote:
>
>> To All,
>>
>> I just needed to check to see if anyone knows about a problem using
>> calling party transformations at the gateway level.
>>
>> I have a setup where I am send a fully globalized called and calling # to
>> my gateways.
>> I wanted to make all needed adjustments just before it goes to the PSTN.
>>
>> My transformation CSS's are setup properly and I can adjust the called
>> party number
>> but the calling party number will not adjust according to the calling
>> party xform pattern.
>>
>> I can see the calling number is getting to the gw in "+" format based on
>> what I see in Q.931 on the router.
>>
>> Anybody experience this behavior?
>>
>>
>>
>> ___
>> 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] Calling party transformations on a gateway

[ccielabrat]

To All,

I just needed to check to see if anyone knows about a problem using calling
party transformations at the gateway level.

I have a setup where I am send a fully globalized called and calling # to
my gateways.
I wanted to make all needed adjustments just before it goes to the PSTN.

My transformation CSS's are setup properly and I can adjust the called
party number
but the calling party number will not adjust according to the calling party
xform pattern.

I can see the calling number is getting to the gw in "+" format based on
what I see in Q.931 on the router.

Anybody experience this behavior?
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[OSL | CCIE_Voice] Dtmf problem with MVA

[ccielabrat]

I have MVA configured on my h323 router, with the appropriate dial peers as
per the Cucm help pages.
I am able to call into the piloting dn and I can hear the MVA application
prompt me for my pin.
When I press any digits, they are not recognized.
I've tried to force a g711 codec and ensured Dtmf-relay is configured, but
it doesn't change the problem.

Has anyone run into this issue?
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[OSL | CCIE_Voice] WAN qos question

[CCIELabRat]

I ran into a problem the other day that has me confused.
I ran auto qos on the hq side, changed values as needed and pasted the
modified policy in the hq router.
I then took the same policy and pasted it into the SB router config and
bound it to the dlci.
All seemed to be ok until I tried to get phones registered.
I could get a dhcp address but never register.

I knew something with the WAN qos was screwed up.
I've done it to myself in practice and in the real lab.

It turns out that I didn't have frts on the physical interface.
Once I put it on, everything started working.

My questions are:

1.) I thought IOS would refuse attaching a class to a dlci without frts on
the physical interface.
2.) Without frts configured on the physical interface, wouldn't the class
assignment on the dlci (384k) effectively be ignored?

I know what the config problem was, but I understand what it was actually
causing to happen at the Pvc level.
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Re: [OSL | CCIE_Voice] WAN QOS Strategy Question.

[ccielabrat]

Thanks for clarifying this Vik.

I wasn't aware of the crtp one way audio trap. But I'll keep it in mind.

One question : I thought we could avoid having the frame relay traffic
shaping on the physical interface if we did generic traffic shaping via
class-map within the frame-relay map-class.

Is this not the case?
I vaguely remember what you are talking about where I wasn't able to apply
a class to a DLCI and it prompted a message asking for traffic shaping to
be enabled.

Just wondering if there is work around or is it a hard and fast rule that
Frame traffic shaping MUST be on the physical interface.



On Sat, Dec 3, 2011 at 4:15 PM, Vik Malhi  wrote:

> This is correct. It's not AutoQoS that causes the problem- its because you
> MUST have FRTS enabled in order for the map-class to be attached to the
> DLCI. And this must happen since the service policy is inside the map-class.
>
> I recommend you run AutoQoS on all routers when doing WAN QoS or at the
> very least attach a map-class to all DLCI's.
>
> Also be careful if you are using cRTP. You should ensure that if you are
> using cRTP that both ends of the pipe are configured with cRTP otherwise
> you will experience one way audio.
>
>
> Vik Malhi – CCIE #13890
> Managing Partner - IPexpert, Inc.
>
> Telephone: +1.810.326.1444 ext 420
> Fax: +1.810.454.0130
> Mailto: vma...@ipexpert.com
>
>
>
>
> On Dec 3, 2011, at 10:59 AM, ccielabrat wrote:
>
> Hey everyone,
> I can confirm after A LOT of testing, if you are given a QOS requirement
> for only one of the two Frame PVCs, and you use Auto QOS, you will have a
> problem.
>
> Auto QOS will automatically config "Frame-relay Traffic shaping" on the
> physical WAN interface and then configure the PVC you are QOS'ing to the
> bandwidth that is noted under the sub interface.
> The other sub interface gets left with the default "frame-relay traffic
> shaping" behavior which is to drop CIR to 56k on the PVC.
>
> Do a "Show Frame PVC " on both PVC's after running auto qos.
>
> I think this could be an intended Rat hole on the exam.
> If you only have to configure Hq-SB QOS and you don't know much more the
> to run autoqos and tweak a couple parameters, your SC communications will
> start to fail with a PVC CIR of 56k.
>
>
> On Sat, Dec 3, 2011 at 10:00 AM, Ken Wyan  wrote:
>
>> Hi,
>>
>> I think you got 56k value from this document which was published in 2005
>> with IOS version 11. (somehow same age as QoS SRND)
>>
>>
>> http://www.cisco.com/en/US/tech/tk713/tk237/technologies_configuration_example09186a00800942f8.shtml
>>
>> I think it's better to put auto qos voip or auto qos voip fr-atm in the
>> remaining interface as well (without any bandwidth as it's not mentioned in
>> exam). Then it"ll take 1.5M by default.
>>
>> Is there a command to verify that FRTS use 56k bandwidth because above
>> documents are very old.
>>
>> Ken
>>
>>
>>
>>
>>
>>
>> On Sat, Dec 3, 2011 at 8:54 AM, ccielabrat  wrote:
>>
>>> To All,
>>>
>>> I've been trying to figure the best/fastest way to get a WAN QOS
>>> requirement completed on exam day.
>>>
>>> I've become very comfortable with Auto-QOS and making the needed tweaks,
>>> so Auto-QOS is the way I'm going to use.
>>>
>>> The one piece of the strategy that I'm stilll wondering about is if WAN
>>> QOS is specified for only one of the PVC's.
>>> Auto-QOS will automatically put Frame-relay traffic shaping on the
>>> physical interface which has the side effect of leaving the "other" pvc
>>> with a 56k PVC speed.
>>>
>>> My solution here is to create a frame-relay map-class with the following
>>> parameters.
>>>
>>> map-class frame-relay Not56k
>>>  frame-relay traffice-rate 1536
>>>
>>> I apply this map-class to the "other" sub-interface/PVC which negates
>>> the 56k problem.
>>>
>>> I'm curious if anyone has an opinion on the Pros/Cons of this approach
>>> and if it might negate requirements somehow.
>>>
>>>
>>>
>>>
>>> ___
>>> 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 <http://www.platinumplacement.com/>
>>>
>>
>>
> ___
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> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
>
>
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Re: [OSL | CCIE_Voice] WAN QOS Strategy Question.

[ccielabrat]

Hey everyone,
I can confirm after A LOT of testing, if you are given a QOS requirement
for only one of the two Frame PVCs, and you use Auto QOS, you will have a
problem.

Auto QOS will automatically config "Frame-relay Traffic shaping" on the
physical WAN interface and then configure the PVC you are QOS'ing to the
bandwidth that is noted under the sub interface.
The other sub interface gets left with the default "frame-relay traffic
shaping" behavior which is to drop CIR to 56k on the PVC.

Do a "Show Frame PVC " on both PVC's after running auto qos.

I think this could be an intended Rat hole on the exam.
If you only have to configure Hq-SB QOS and you don't know much more the to
run autoqos and tweak a couple parameters, your SC communications will
start to fail with a PVC CIR of 56k.


On Sat, Dec 3, 2011 at 10:00 AM, Ken Wyan  wrote:

> Hi,
>
> I think you got 56k value from this document which was published in 2005
> with IOS version 11. (somehow same age as QoS SRND)
>
>
> http://www.cisco.com/en/US/tech/tk713/tk237/technologies_configuration_example09186a00800942f8.shtml
>
> I think it's better to put auto qos voip or auto qos voip fr-atm in the
> remaining interface as well (without any bandwidth as it's not mentioned in
> exam). Then it"ll take 1.5M by default.
>
> Is there a command to verify that FRTS use 56k bandwidth because above
> documents are very old.
>
> Ken
>
>
>
>
>
>
> On Sat, Dec 3, 2011 at 8:54 AM, ccielabrat  wrote:
>
>> To All,
>>
>> I've been trying to figure the best/fastest way to get a WAN QOS
>> requirement completed on exam day.
>>
>> I've become very comfortable with Auto-QOS and making the needed tweaks,
>> so Auto-QOS is the way I'm going to use.
>>
>> The one piece of the strategy that I'm stilll wondering about is if WAN
>> QOS is specified for only one of the PVC's.
>> Auto-QOS will automatically put Frame-relay traffic shaping on the
>> physical interface which has the side effect of leaving the "other" pvc
>> with a 56k PVC speed.
>>
>> My solution here is to create a frame-relay map-class with the following
>> parameters.
>>
>> map-class frame-relay Not56k
>>  frame-relay traffice-rate 1536
>>
>> I apply this map-class to the "other" sub-interface/PVC which negates the
>> 56k problem.
>>
>> I'm curious if anyone has an opinion on the Pros/Cons of this approach
>> and if it might negate requirements somehow.
>>
>>
>>
>>
>> ___
>> 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 <http://www.platinumplacement.com/>
>>
>
>
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[OSL | CCIE_Voice] WAN QOS Strategy Question.

[ccielabrat]

To All,

I've been trying to figure the best/fastest way to get a WAN QOS
requirement completed on exam day.

I've become very comfortable with Auto-QOS and making the needed tweaks, so
Auto-QOS is the way I'm going to use.

The one piece of the strategy that I'm stilll wondering about is if WAN QOS
is specified for only one of the PVC's.
Auto-QOS will automatically put Frame-relay traffic shaping on the physical
interface which has the side effect of leaving the "other" pvc with a 56k
PVC speed.

My solution here is to create a frame-relay map-class with the following
parameters.

map-class frame-relay Not56k
 frame-relay traffice-rate 1536

I apply this map-class to the "other" sub-interface/PVC which negates the
56k problem.

I'm curious if anyone has an opinion on the Pros/Cons of this approach and
if it might negate requirements somehow.
___
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Re: [OSL | CCIE_Voice] Need some help with GK/CUBE Remote Zone call debugging.

[ccielabrat]

Hi Chris,
So is CUBE waiting for CUCM to send TCS ?
And holding up the whole process?

Also, I guess this means the the BBGK is waiting for TCS from CUBE.
Is CUBE sending TCS independent of the CUCM call leg? or does it wait for
CUCM TCS before sending TCS to BBGK?



On Fri, Dec 2, 2011 at 3:08 PM, Chris Martin  wrote:

> Because the IOS gateway trunks follow the rfc that the initiator will send
> TCS, while by default CUCM waits for the TCS from the end point.  Per the
> SRND this was done for potential video issues, so either you may have video
> problems with certain endpoints or voice issues with gateways trunks.  You
> can review this on CUCM SRND, search for TCS.
>
> HTH,
> Chris
>
>
> On Fri, Dec 2, 2011 at 12:45 PM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>> By the way, why the BBGK endpoint does not send the TCS?
>>
>>
>>
>>
>> On Fri, Dec 2, 2011 at 10:02 PM, ccielabrat  wrote:
>>
>>> Hi Ash,
>>>
>>> Thanks very much for taking the time to reply.
>>> I would really like to understand all the pieces to this scenario.
>>>
>>> The debug I posted was from the HQ router/GK/CUBE, I'm not sure how to
>>> read it yet, so I can't say what call leg it represents.
>>>
>>> We are talking about the same scenario you mention in your reply.
>>>
>>> CUCM  via GK-0Trunk (No MTP configured) (Wait for H245 unchecked)
>>> GK configured with a Remote Zone using a outvia to local CUBE.
>>> CUBE is configured with One inbound dial-peer 011! with a fixed codec of
>>> g.711 and an outbound dial-peer targeting RAS (default g.729)
>>> The call setup works and I can answer the call but the rtp never works.
>>>
>>> Please confirm my understanding of the problem.
>>> 1.) CUCM does ARQ/setup via GK
>>> 2.) GK sends LRQ to BBGK and gets LCF that it's a routable DN
>>> 3.) GK tells CUCM to target CUBE for H.323 call setup because of Outvia
>>> config for BBGK Zone.
>>> 4.) CUCM sends H.225 to CUBE which triggers CUBE to do H.225 to endpoint
>>> in BBGK Zone.
>>> 5.) CUBE waits for h.245 TCS and doesn't send H.225 connect back to CUCM
>>> 6.) BBGK Endpoint doesn't send any TCS and causes CUBE to wait /timeout
>>> for H.245
>>> 7.) Call fails with CUBE disconnecting both BBGK call leg and CUCM call
>>> leg.
>>>
>>> I think this is due to the fact that CUCM (without MTP) is forced to do
>>> slow start , while CUBE will automatically do fast start.
>>> As I understand , CUBE can't compensate for the difference between
>>> slow/fast start call legs.
>>>
>>> So is the only option to have an MTP configured at CUCM side?
>>> Can the CUBE be forced to do slow start? Would that fix the issue?
>>>
>>>
>>>
>>>
>>> On Fri, Dec 2, 2011 at 1:20 AM, Ashraf Ayyash wrote:
>>>
>>>> What are you looking at in the debugs and which leg is this ?
>>>>
>>>> you said you have CUBE in between so you will see 2 seprated H245
>>>> negotiation for each leg ,
>>>>
>>>> can you post the H225 and h245 debugs for me please ?
>>>>
>>>> just to make sure that we both talking about the same thing , this
>>>> call is Slow start and you have CUBE with Transcoder in it and the
>>>> issue you trying to trace is that once you connected the call it got
>>>> dropped by the remote GK ?
>>>>
>>>> On the CUBE you have inbound dial-peer with codec G711 and outbound
>>>> dial-peer with G729 and then you have transcoder to fix this in the
>>>> cube , but on the remote GK you have dial-peer with G711u call only,
>>>>
>>>> if any of the above is not what you have please correct me ,
>>>>
>>>>
>>>>
>>>> Ash
>>>>
>>>> On Thu, Dec 1, 2011 at 2:53 PM, ccielabrat 
>>>> wrote:
>>>> > ok, I'm getting to understand this better.
>>>> >
>>>> > I don't see any mention of a tcs failure though
>>>> > See the output of debug h245 asn1 below. Where is the indication of a
>>>> > failure?
>>>> >
>>>> > Also, I have CUBE running with a Hw transcoder registered locally on
>>>> HQ
>>>> > telephony-service.
>>>> > I would think the CUBE should allocate the xcoder to get around the
>>>> codec
>>>> > mism

Re: [OSL | CCIE_Voice] Need some help with GK/CUBE Remote Zone call debugging.

[ccielabrat]

Hi Ash,

Thanks very much for taking the time to reply.
I would really like to understand all the pieces to this scenario.

The debug I posted was from the HQ router/GK/CUBE, I'm not sure how to read
it yet, so I can't say what call leg it represents.

We are talking about the same scenario you mention in your reply.

CUCM  via GK-0Trunk (No MTP configured) (Wait for H245 unchecked)
GK configured with a Remote Zone using a outvia to local CUBE.
CUBE is configured with One inbound dial-peer 011! with a fixed codec of
g.711 and an outbound dial-peer targeting RAS (default g.729)
The call setup works and I can answer the call but the rtp never works.

Please confirm my understanding of the problem.
1.) CUCM does ARQ/setup via GK
2.) GK sends LRQ to BBGK and gets LCF that it's a routable DN
3.) GK tells CUCM to target CUBE for H.323 call setup because of Outvia
config for BBGK Zone.
4.) CUCM sends H.225 to CUBE which triggers CUBE to do H.225 to endpoint in
BBGK Zone.
5.) CUBE waits for h.245 TCS and doesn't send H.225 connect back to CUCM
6.) BBGK Endpoint doesn't send any TCS and causes CUBE to wait /timeout for
H.245
7.) Call fails with CUBE disconnecting both BBGK call leg and CUCM call leg.

I think this is due to the fact that CUCM (without MTP) is forced to do
slow start , while CUBE will automatically do fast start.
As I understand , CUBE can't compensate for the difference between
slow/fast start call legs.

So is the only option to have an MTP configured at CUCM side?
Can the CUBE be forced to do slow start? Would that fix the issue?



On Fri, Dec 2, 2011 at 1:20 AM, Ashraf Ayyash  wrote:

> What are you looking at in the debugs and which leg is this ?
>
> you said you have CUBE in between so you will see 2 seprated H245
> negotiation for each leg ,
>
> can you post the H225 and h245 debugs for me please ?
>
> just to make sure that we both talking about the same thing , this
> call is Slow start and you have CUBE with Transcoder in it and the
> issue you trying to trace is that once you connected the call it got
> dropped by the remote GK ?
>
> On the CUBE you have inbound dial-peer with codec G711 and outbound
> dial-peer with G729 and then you have transcoder to fix this in the
> cube , but on the remote GK you have dial-peer with G711u call only,
>
> if any of the above is not what you have please correct me ,
>
>
>
> Ash
>
> On Thu, Dec 1, 2011 at 2:53 PM, ccielabrat  wrote:
> > ok, I'm getting to understand this better.
> >
> > I don't see any mention of a tcs failure though
> > See the output of debug h245 asn1 below. Where is the indication of a
> > failure?
> >
> > Also, I have CUBE running with a Hw transcoder registered locally on HQ
> > telephony-service.
> > I would think the CUBE should allocate the xcoder to get around the codec
> > mismatch.
> >
> > Output from Debug of H245 ASN1 on HQ/GK/CUBE
> >
> > Dec  1 20:45:36.806: h245_decode_one_pdu: more_pdus = 0,
> bytesLeftToDecode =
> > 97
> > Dec  1 20:45:36.806: H245 MSC INCOMING ENCODE BUFFER::=
> >
> 0270010600088175000A801380003C000101010CC0010001000680240001058124080105822280058322C005848501408585011080002B85015000820300010002000301000400052B
> > Dec  1 20:45:36.806:
> > Dec  1 20:45:36.806: H245 MSC INCOMING PDU ::=
> >
> > value MultimediaSystemControlMessage ::= request : terminalCapabilitySet
> :
> > {
> >   sequenceNumber 1
> >   protocolIdentifier { 0 0 8 245 0 10 }
> >   multiplexCapability h2250Capability :
> >   {
> > maximumAudioDelayJitter 60
> > receiveMultipointCapability
> > {
> >   multicastCapability FALSE
> >   multiUniCastConference FALSE
> >   mediaDistributionCapability
> >   {
> >
> > {
> >   centralizedControl FALSE
> >   distributedControl FALSE
> >   centralizedAudio FALSE
> >   distributedAudio FA
> > HQ#LSE
> >   centralizedVideo FALSE
> >   distributedVideo FALSE
> > }
> >   }
> > }
> > transmitMultipointCapability
> > {
> >   multicastCapability FALSE
> >   multiUniCastConference FALSE
> >   mediaDistributionCapability
> >   {
> >
> > {
> >   centralizedControl FALSE
> >   distributedControl FALSE
> >   centralizedAudio FALSE
> >   distributedAudio FALSE
> >   centralizedVideo FALSE
> >   

Re: [OSL | CCIE_Voice] cisco IOU

[ccielabrat]

They are real routers.
Cisco IOU can't fake voice interfaces or DSPs.
As per conversation at CiscoLive.

The phones are connected over L2 VPN to be able to keep all lab equipment
remote from test centers.
Rumor is that RTP and SJ are now homed in Texas.


On Thu, Dec 1, 2011 at 10:21 PM, Ken Wyan  wrote:

> Hi,
>
> Does CCIE Voice lab routers are real or run on IOU?
>
> Guys say not to restart IOU during lab
>
>
> ___
> 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] Strange behavior on MGCP gateways

[ccielabrat]

I noticed a weird thing while testing MGCP.

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

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

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

Weird.
___
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Re: [OSL | CCIE_Voice] CME as SRST won't forward to voicemail using ephone-dn-template

[ccielabrat]

A what ?


On Thu, Dec 1, 2011 at 9:10 PM, Ken Wyan  wrote:

> Did you try adding a ephone-dn (octo line) with conference adhoc for
> cBarge?
>
> On Fri, Dec 2, 2011 at 5:24 AM, ccielabrat  wrote:
>
>> Hey all,
>>
>> I'm finishing up testing my CME as SRST testing and ran in to a problem
>> where it looks like the phones are getting the ephone-dn-template
>> configured under telephony-service.
>> The phone just rings and rings 
>> I'm also wondering if the DND button should automatically push calls to
>> voicemail in CME as SRST?
>>
>> One last question : Is Cbarge possible in CME as SRST? I have a hw conf
>> resource registered successfully in SRST mode but it ALWAYS fails to setup
>> the cbarge.
>>
>>
>> Does this look correct?
>>
>> telephony-service
>>  sdspfarm units 1
>>  sdspfarm tag 1 BR1CONF
>>  no privacy
>>  conference hardware
>>  srst mode auto-provision none
>>  srst ephone template 1
>>  srst ephone description "FallBack" : Dec 01 2011 16:04:15
>>  srst dn template 1
>>  srst dn line-mode octo
>>  em logout 0:0 0:0 0:0
>>  max-ephones 4
>>  max-dn 10 preference 9
>>  ip source-address 10.10.2.1 port 2000
>>  system message Your phone are in fallback mode
>>  date-format dd-mm-yy
>>  keepalive 10 auxiliary 1
>>  voicemail 2230
>>  max-conferences 8 gain -6
>>  call-forward pattern .T
>>  transfer-system full-consult
>>  transfer-pattern .T
>>  create cnf-files version-stamp 7960 Dec 01 2011 17:24:55
>>
>> ephone-dn-template  1
>>  call-forward busy 2230
>>  call-forward noan 2230 timeout 20
>>  huntstop channel 4
>>
>> ephone-template  1
>>  privacy off
>>  privacy-button
>>  transfer max-length 4
>>  softkeys remote-in-use  CBarge Newcall
>>  softkeys idle  Redial Newcall Cfwdall
>>
>> ___
>> 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 <http://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] Adding second language to CUE

[ccielabrat]

Hi Ashraf,

I removed everything but the one set of files from the directory and
ran "Software install add .pkg
but it didn't change the behavior.


On Thu, Dec 1, 2011 at 3:16 PM,  wrote:

> oh, ok.
> I'll give it a try.
>
>
> On Thu, Dec 1, 2011 at 1:44 PM, Ashraf Ayyash 
> wrote:
> > Hello ,
> >
> > why you have tow packages in the root directory ?
> >
> > you have to have the full package of 7.0.6 and the lang pack of GB
> > 7.0.6 ONLY on the root directory , run the installation again and see
> > how it will go
> >
> > Ash
> >
> > On Thu, Dec 1, 2011 at 8:02 AM, ccielabrat  wrote:
> >> Hi Ashraf,
> >>
> >> See below. Thank you!
> >>
> >> ftp> ls
> >> 200 Port command successful
> >> 150 Opening data channel for directory list.
> >> cue-installer.nm-aim.7.0.1
> >> cue-installer.nm-aim.7.0.6
> >> cue-vm-en_GB-langpack.nm-aim.7.0.6.prt1
> >> cue-vm-full-k9.nm-aim.7.0.1.prt1
> >> cue-vm-full-k9.nm-aim.7.0.6.prt1
> >> cue-vm-installer-k9.nm-aim.7.0.1.prt1
> >> cue-vm-installer-k9.nm-aim.7.0.6.prt1
> >> cue-vm-k9.nm-aim.7.0.1.pkg
> >> cue-vm-k9.nm-aim.7.0.1.zip
> >> cue-vm-k9.nm-aim.7.0.6.pkg
> >> cue-vm-k9.nm-aim.7.0.6.zip
> >> cue-vm-k9.nmx.7.1.2.zip
> >> cue-vm-langpack.nm-aim.7.0.1.pkg
> >> cue-vm-langpack.nm-aim.7.0.6.pkg
> >> cue-vm-license_12mbx_ccm_7.0.1.pkg
> >> cue-vm-license_12mbx_ccm_7.0.6.pkg
> >> cue-vm-license_12mbx_cme_7.0.1.pkg
> >> cue-vm-license_12mbx_cme_7.0.6.pkg
> >>
> >>
> >> On Thu, Dec 1, 2011 at 2:57 AM, Ashraf Ayyash 
> wrote:
> >>>
> >>> This mean you are installing the Wrong Language files , or you missing
> >>> on critical file ,
> >>>
> >>> can you please paste what you have in the FTP directory root ?
> >>>
> >>> Ash
> >>>
> >>> On Wed, Nov 30, 2011 at 1:03 PM, ccielabrat 
> wrote:
> >>> > I'm trying to add a second language to an AIM-CUE.
> >>> >
> >>> > I use the command "software install add url  ftp://x.x.x.x/xyz.pkg
> >>> > and it seems to run without a problem but when it finishes processing
> >>> > the
> >>> > file,
> >>> >
> >>> > I get the follow message :
> >>> >
> >>> > Language add-ons found on the system (1):
> >>> >
> >>> >   Installed   SKUName (version)
> >>> > --
> >>> >   *  ENU   CUE Voicemail US English (7.0.6)
> >>> >
> >>> > Maximum allowed language add-ons (=1) already installed.
> >>> > You can use "software uninstall" to remove add-ons.
> >>> >
> >>> > ui_install scripts executed successfully.
> >>> >
> >>> > The issue is if I run "Show software licenses" , it indicates a max
> of 2
> >>> > languages are allowed.
> >>> >
> >>> > CUE# sho software licenses
> >>> > Installed license files:
> >>> >  - voicemail_lic.sig : 12 MAILBOX LICENSE
> >>> >
> >>> > Core:
> >>> >  - Application mode: CCME
> >>> >  - Total usable system ports: 6
> >>> >
> >>> > Voicemail/Auto Attendant:
> >>> >  - Max system mailbox capacity time: 840
> >>> >  - Default # of general delivery mailboxes: 5
> >>> >  - Default # of personal mailboxes: 12
> >>> >
> >>> >  - Max # of configurable mailboxes: 17
> >>> >
> >>> > Interactive Voice Response:
> >>> >  - Max # of IVR sessions: Not Available
> >>> >
> >>> > Languages:
> >>> >  - Max installed languages: 2
> >>> >  - Max enabled languages: 2
> >>> >
> >>> > ___
> >>> > 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
>
___
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] CME as SRST won't forward to voicemail using ephone-dn-template

[ccielabrat]

Hey all,

I'm finishing up testing my CME as SRST testing and ran in to a problem
where it looks like the phones are getting the ephone-dn-template
configured under telephony-service.
The phone just rings and rings 
I'm also wondering if the DND button should automatically push calls to
voicemail in CME as SRST?

One last question : Is Cbarge possible in CME as SRST? I have a hw conf
resource registered successfully in SRST mode but it ALWAYS fails to setup
the cbarge.


Does this look correct?

telephony-service
 sdspfarm units 1
 sdspfarm tag 1 BR1CONF
 no privacy
 conference hardware
 srst mode auto-provision none
 srst ephone template 1
 srst ephone description "FallBack" : Dec 01 2011 16:04:15
 srst dn template 1
 srst dn line-mode octo
 em logout 0:0 0:0 0:0
 max-ephones 4
 max-dn 10 preference 9
 ip source-address 10.10.2.1 port 2000
 system message Your phone are in fallback mode
 date-format dd-mm-yy
 keepalive 10 auxiliary 1
 voicemail 2230
 max-conferences 8 gain -6
 call-forward pattern .T
 transfer-system full-consult
 transfer-pattern .T
 create cnf-files version-stamp 7960 Dec 01 2011 17:24:55

ephone-dn-template  1
 call-forward busy 2230
 call-forward noan 2230 timeout 20
 huntstop channel 4

ephone-template  1
 privacy off
 privacy-button
 transfer max-length 4
 softkeys remote-in-use  CBarge Newcall
 softkeys idle  Redial Newcall Cfwdall
___
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Re: [OSL | CCIE_Voice] Need some help with GK/CUBE Remote Zone call debugging.

[ccielabrat]

ok, I'm getting to understand this better.

I don't see any mention of a tcs failure though
See the output of debug h245 asn1 below. Where is the indication of a
failure?

Also, I have CUBE running with a Hw transcoder registered locally on HQ
telephony-service.
I would think the CUBE should allocate the xcoder to get around the codec
mismatch.

Output from Debug of H245 ASN1 on HQ/GK/CUBE

Dec  1 20:45:36.806: h245_decode_one_pdu: more_pdus = 0, bytesLeftToDecode
= 97
Dec  1 20:45:36.806: H245 MSC INCOMING ENCODE BUFFER::=
0270010600088175000A801380003C000101010CC0010001000680240001058124080105822280058322C005848501408585011080002B85015000820300010002000301000400052B
Dec  1 20:45:36.806:
Dec  1 20:45:36.806: H245 MSC INCOMING PDU ::=

value MultimediaSystemControlMessage ::= request : terminalCapabilitySet :
{
  sequenceNumber 1
  protocolIdentifier { 0 0 8 245 0 10 }
  multiplexCapability h2250Capability :
  {
maximumAudioDelayJitter 60
receiveMultipointCapability
{
  multicastCapability FALSE
  multiUniCastConference FALSE
  mediaDistributionCapability
  {

{
  centralizedControl FALSE
  distributedControl FALSE
  centralizedAudio FALSE
  distributedAudio FA
HQ#LSE
  centralizedVideo FALSE
  distributedVideo FALSE
}
  }
}
transmitMultipointCapability
{
  multicastCapability FALSE
  multiUniCastConference FALSE
  mediaDistributionCapability
  {

{
  centralizedControl FALSE
  distributedControl FALSE
  centralizedAudio FALSE
  distributedAudio FALSE
  centralizedVideo FALSE
  distributedVideo FALSE
}
  }
}
receiveAndTransmitMultipointCapability
{
  multicastCapability FALSE
  multiUniCastConference FALSE
  mediaDistributionCapability
  {

{
  centralizedControl FALSE
  distributedControl FALSE
  centralizedAudio FALSE
  distributedAudio FALSE
  centralizedVideo FALSE
  distributedVideo FALSE
}
  }
}
mcCapability
{
  centralizedConferenceMC FALSE
  decentralizedConferenceMC FALSE
}
rtcpVideoControlCapability FALSE
mediaPacketizationCapability
{
  h261aVideoPacketization FALSE
}
logicalChannelSwitchingCapability FALSE
t120DynamicPortCapability FALSE
  }
  capabilityTable
  {

{
  capabilityTableEntryNumber 1
  capability receiveAudioCapability : g729wAnnexB : 6
},
{
  capabilityTableEntryNumber 2
  capability receiveAudioCapability : g729AnnexAwAnnexB : 6
},
{
  capabilityTableEntryNumber 3
  capability receiveAudioCapability : g729 : 6
},
{
  capabilityTableEntryNumber 4
  capability receiveAudioCapability : g729AnnexA : 6
},
{
  capabilityTableEntryNumber 5
  capability receiveAndTransmitUserInputCapability : dtmf : NULL
},
{
  capabilityTableEntryNumber 6
  capability receiveAndTransmitUserInputCapability : basicString :
NULL
},
{
  capabilityTableEntryNumber 44
  capability receiveAndTransmitUserInputCapability : hookflash :
NULL
}
  }
  capabilityDescriptors
  {

{
  capabilityDescriptorNumber 0
  simultaneousCapabilities
  {

{
  1,
  2,
  3,
  4
},

{
  5,
  6
},

{
  44
}
  }
}
  }
}



Dec  1 20:45:36.810: h245_decode_one_pdu: H245ASNDecodePdu rc = 0,
bytesLeftToDecode = 0
Dec  1 20:45:36.810: h245_decode_one_pdu: Read Pkt body: more_pdus:0 rc:0
asn_rc:0
HQ#
HQ#
HQ#
HQ#sho deb

H.245:
  H.245 ASN1 Messages debugging is on


On Thu, Dec 1, 2011 at 2:00 PM, Ashraf Ayyash  wrote:

> The ccapi debug will show you the cause code which doesn't explain why
> the call failed ,
>
> you have to debug the h245 asn1 and check the TCS and see the codecs
> advertised and received and then you will get the TCS negotiation
> failure so you can explain that there is codec mismatch
>
> Ash
>
> On Thu, Dec 1, 2011 at 11:55 AM, Mohd Baqari 
> wrote:
> > Use the command debug voice ccapi inout. H323 debugs won't show in this
> case.
> >
> > Regards,
> > Mohammed Al Baqari
> >
> > Sent from my iPhone
> >
> > On Dec 1, 2011, 

Re: [OSL | CCIE_Voice] Adding second language to CUE

[CCIELabRat]

oh, ok.
I'll give it a try.


On Thu, Dec 1, 2011 at 1:44 PM, Ashraf Ayyash  wrote:
> Hello ,
>
> why you have tow packages in the root directory ?
>
> you have to have the full package of 7.0.6 and the lang pack of GB
> 7.0.6 ONLY on the root directory , run the installation again and see
> how it will go
>
> Ash
>
> On Thu, Dec 1, 2011 at 8:02 AM, ccielabrat  wrote:
>> Hi Ashraf,
>>
>> See below. Thank you!
>>
>> ftp> ls
>> 200 Port command successful
>> 150 Opening data channel for directory list.
>> cue-installer.nm-aim.7.0.1
>> cue-installer.nm-aim.7.0.6
>> cue-vm-en_GB-langpack.nm-aim.7.0.6.prt1
>> cue-vm-full-k9.nm-aim.7.0.1.prt1
>> cue-vm-full-k9.nm-aim.7.0.6.prt1
>> cue-vm-installer-k9.nm-aim.7.0.1.prt1
>> cue-vm-installer-k9.nm-aim.7.0.6.prt1
>> cue-vm-k9.nm-aim.7.0.1.pkg
>> cue-vm-k9.nm-aim.7.0.1.zip
>> cue-vm-k9.nm-aim.7.0.6.pkg
>> cue-vm-k9.nm-aim.7.0.6.zip
>> cue-vm-k9.nmx.7.1.2.zip
>> cue-vm-langpack.nm-aim.7.0.1.pkg
>> cue-vm-langpack.nm-aim.7.0.6.pkg
>> cue-vm-license_12mbx_ccm_7.0.1.pkg
>> cue-vm-license_12mbx_ccm_7.0.6.pkg
>> cue-vm-license_12mbx_cme_7.0.1.pkg
>> cue-vm-license_12mbx_cme_7.0.6.pkg
>>
>>
>> On Thu, Dec 1, 2011 at 2:57 AM, Ashraf Ayyash  wrote:
>>>
>>> This mean you are installing the Wrong Language files , or you missing
>>> on critical file ,
>>>
>>> can you please paste what you have in the FTP directory root ?
>>>
>>> Ash
>>>
>>> On Wed, Nov 30, 2011 at 1:03 PM, ccielabrat  wrote:
>>> > I'm trying to add a second language to an AIM-CUE.
>>> >
>>> > I use the command "software install add url  ftp://x.x.x.x/xyz.pkg
>>> > and it seems to run without a problem but when it finishes processing
>>> > the
>>> > file,
>>> >
>>> > I get the follow message :
>>> >
>>> > Language add-ons found on the system (1):
>>> >
>>> >   Installed   SKU    Name (version)
>>> > --
>>> >   *  ENU   CUE Voicemail US English (7.0.6)
>>> >
>>> > Maximum allowed language add-ons (=1) already installed.
>>> > You can use "software uninstall" to remove add-ons.
>>> >
>>> > ui_install scripts executed successfully.
>>> >
>>> > The issue is if I run "Show software licenses" , it indicates a max of 2
>>> > languages are allowed.
>>> >
>>> > CUE# sho software licenses
>>> > Installed license files:
>>> >  - voicemail_lic.sig : 12 MAILBOX LICENSE
>>> >
>>> > Core:
>>> >  - Application mode: CCME
>>> >  - Total usable system ports: 6
>>> >
>>> > Voicemail/Auto Attendant:
>>> >  - Max system mailbox capacity time: 840
>>> >  - Default # of general delivery mailboxes: 5
>>> >  - Default # of personal mailboxes: 12
>>> >
>>> >  - Max # of configurable mailboxes: 17
>>> >
>>> > Interactive Voice Response:
>>> >  - Max # of IVR sessions: Not Available
>>> >
>>> > Languages:
>>> >  - Max installed languages: 2
>>> >  - Max enabled languages: 2
>>> >
>>> > ___
>>> > 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

[OSL | CCIE_Voice] Need some help with GK/CUBE Remote Zone call debugging.

[ccielabrat]

I'm trying to setup a call from HQ CUCM via GK-Trunk to a Remote Gk Zone.

I have the Gatekeeper configured with OutVia for the remote zone
referencing a CUBE on the HQ router.

I didn't realize (but it makes sense now) that with "Wait for H.245"
unchecked on on the CUCM trunk, the call setup goes to the GK/CUBE as g.711.

This obviously causes a problem when the CUBE (by default) tries to create
the outgoing call leg to the remote zone using G.729.

I don't have an XCoder available to CUBE at this point.

My problem is that I can't see the codec mismatch failure in "debug cch323
h225 " or "debug cch323 h245".
(If it's in there , I'm not seeing :) )

Can someone help me understand if the failure is noted in either of these
debugs
Or
Point me towards a debug that would show the codec mismatch problem?
___
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Re: [OSL | CCIE_Voice] Adding second language to CUE

[ccielabrat]

Hi Ashraf,

See below. Thank you!

ftp> ls
200 Port command successful
150 Opening data channel for directory list.
cue-installer.nm-aim.7.0.1
cue-installer.nm-aim.7.0.6
cue-vm-en_GB-langpack.nm-aim.7.0.6.prt1
cue-vm-full-k9.nm-aim.7.0.1.prt1
cue-vm-full-k9.nm-aim.7.0.6.prt1
cue-vm-installer-k9.nm-aim.7.0.1.prt1
cue-vm-installer-k9.nm-aim.7.0.6.prt1
cue-vm-k9.nm-aim.7.0.1.pkg
cue-vm-k9.nm-aim.7.0.1.zip
cue-vm-k9.nm-aim.7.0.6.pkg
cue-vm-k9.nm-aim.7.0.6.zip
cue-vm-k9.nmx.7.1.2.zip
cue-vm-langpack.nm-aim.7.0.1.pkg
cue-vm-langpack.nm-aim.7.0.6.pkg
cue-vm-license_12mbx_ccm_7.0.1.pkg
cue-vm-license_12mbx_ccm_7.0.6.pkg
cue-vm-license_12mbx_cme_7.0.1.pkg
cue-vm-license_12mbx_cme_7.0.6.pkg


On Thu, Dec 1, 2011 at 2:57 AM, Ashraf Ayyash  wrote:

> This mean you are installing the Wrong Language files , or you missing
> on critical file ,
>
> can you please paste what you have in the FTP directory root ?
>
> Ash
>
> On Wed, Nov 30, 2011 at 1:03 PM, ccielabrat  wrote:
> > I'm trying to add a second language to an AIM-CUE.
> >
> > I use the command "software install add url  ftp://x.x.x.x/xyz.pkg
> > and it seems to run without a problem but when it finishes processing the
> > file,
> >
> > I get the follow message :
> >
> > Language add-ons found on the system (1):
> >
> >   Installed   SKUName (version)
> > --
> >   *  ENU   CUE Voicemail US English (7.0.6)
> >
> > Maximum allowed language add-ons (=1) already installed.
> > You can use "software uninstall" to remove add-ons.
> >
> > ui_install scripts executed successfully.
> >
> > The issue is if I run "Show software licenses" , it indicates a max of 2
> > languages are allowed.
> >
> > CUE# sho software licenses
> > Installed license files:
> >  - voicemail_lic.sig : 12 MAILBOX LICENSE
> >
> > Core:
> >  - Application mode: CCME
> >  - Total usable system ports: 6
> >
> > Voicemail/Auto Attendant:
> >  - Max system mailbox capacity time: 840
> >  - Default # of general delivery mailboxes: 5
> >  - Default # of personal mailboxes: 12
> >
> >  - Max # of configurable mailboxes: 17
> >
> > Interactive Voice Response:
> >  - Max # of IVR sessions: Not Available
> >
> > Languages:
> >  - Max installed languages: 2
> >  - Max enabled languages: 2
> >
> > ___
> > 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 <http://www.platinumplacement.com/>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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[OSL | CCIE_Voice] Adding second language to CUE

[ccielabrat]

I'm trying to add a second language to an AIM-CUE.

I use the command "software install add url  ftp://x.x.x.x/xyz.pkg
and it seems to run without a problem but when it finishes processing the
file,

I get the follow message :

Language add-ons found on the system (1):

  Installed   SKUName (version)
--
  *  ENU   CUE Voicemail US English (7.0.6)

Maximum allowed language add-ons (=1) already installed.
You can use "software uninstall" to remove add-ons.

ui_install scripts executed successfully.

The issue is if I run "Show software licenses" , it indicates a max of 2
languages are allowed.

CUE# sho software licenses
Installed license files:
 - voicemail_lic.sig : 12 MAILBOX LICENSE

Core:
 - Application mode: CCME
 - Total usable system ports: 6

Voicemail/Auto Attendant:
 - Max system mailbox capacity time: 840
 - Default # of general delivery mailboxes: 5
 - Default # of personal mailboxes: 12

 - Max # of configurable mailboxes: 17

Interactive Voice Response:
 - Max # of IVR sessions: Not Available

Languages:
 - Max installed languages: 2
 - Max enabled languages: 2
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[OSL | CCIE_Voice] Creating log files in the lab.

[ccielabrat]

I don't think this is an NDA topic.
If you think it is, please disregard.

Is there any restriction to create log files on the Lab pc you work off in
the lab?
I like to grab the default CUE config into a text file and modify then
paste back.

Just wondering if the PC is locked down that I can't save files along the
way.
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Re: [OSL | CCIE_Voice] Calling Name with H323 GW

[ccielabrat]

The problem was the check box on the gateway configuration.
After looking at every available option on the gateway, I found out CUCM
wasn't send the name in the first place :)

I do have a follow up question though.

I'm really confused about the logic around "no supplementary-service
h225-notify cid-update"
If the command is by default "supplementary-service h225-notify
cid-update", it seems to me this would ENABLE
the CID to be updated.
It doesn't make sense to me why DISABLING this with "NO
supplementary-service h225-notify cid-update" actually allows the gateway
to trigger an updated display on the phone.


On Tue, Nov 29, 2011 at 5:06 AM, datucha123 datucha123  wrote:

> Also you have to check the "Display IE" checkbox in CUCM H323 gateway
> configuration.
>
>
> On Tue, Nov 29, 2011 at 7:05 AM, Rrcrumm  wrote:
>
>> Do you have "isdn outgoing display-ie" under serial 0/0/0:23 or 15
>> interface?
>>
>> Randall
>>
>>
>>
>> Sent from my iPhone
>>
>> On Nov 28, 2011, at 5:52 PM, ccielabrat  wrote:
>>
>> > I must be missing something easy.
>> > I'm trying to get "Calling Name" to display on my PSTN phone when
>> receiving a call from a IP phone going through a H323 gateway.
>> >
>> > I've found many links online suggesting it's not supported but then
>> others suggesting it's possible.
>> >
>> > Can someone point me to a good link?
>> >
>> >
>>  > ___
>> > 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 <http://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 <http://www.platinumplacement.com/>
>>
>
>
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[OSL | CCIE_Voice] Question about Called Number displayed on phone.

[ccielabrat]

Can someone help me understand what determines what gets displayed on the
phone display when calling outbound.

I have a setup where I have a h323 Gw and MGCP Gw in a single RL.
I create a route pattern of 9.2345678 and assign it to the RL.
If it goes to the H323 GW , I don't drop the 9 prefix in the RL and it
displays 92345678 on the phone.
If it goes to the MGCP GW, the 9 prefix is dropped in the RL and it
displays 2345678 on the phone.

So I figured the display value must be based on what gets sent to the GW,
but this doesn't seem to be true either.

If I adjust my dial-peers on H323 to match on 2345678 (no 9 prefix) , and
drop the 9 in the RL ,  I still see the 9 prefix as dialed on the phone.
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[OSL | CCIE_Voice] Calling Name with H323 GW

[ccielabrat]

I must be missing something easy.
I'm trying to get "Calling Name" to display on my PSTN phone when receiving
a call from a IP phone going through a H323 gateway.

I've found many links online suggesting it's not supported but then others
suggesting it's possible.

Can someone point me to a good link?
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[OSL | CCIE_Voice] TFTP Question

[ccielabrat]

Group,

Is there a service parameter to set for TFTP on CUCM to allow the TFTP
server to "see" new files uploaded without restarting the TFTP Service ?
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[OSL | CCIE_Voice] Question about CUBE & Gatekeepers

[ccielabrat]

I need clarification about Gatekeepers using outvia to a  zone.

I've always thought a CUBE config needed the underlying "Telephony-service"
config to be operational.
Is that the case?

I suppose if the call setup is using g.729 in/out of the CUBE , there is no
need to have anything but a matching dial-peer
and the "allow h323 to h323" in the voice service voip.

Can someone confirm or correct my understanding.

Thanks
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Re: [OSL | CCIE_Voice] cue-cme mwi unsolicited question

[ccielabrat]

Hey Ashraf,

You got me thinking the right way.
I had a mismatch between my sip interface and the gateway configured on CUE.

Thanks!


On Sat, Oct 22, 2011 at 4:43 PM, Ashraf Ayyash  wrote:

> did you binded the SIP to the correct interface from the CME config
> Voice service Voip ?
>
> Any chance to reload the Funky CUE ?
>
> Ash
>
> On Sat, Oct 22, 2011 at 12:12 PM, ccielabrat  wrote:
> > I can't get CUE MWI working either.
> >
> > This is my cue config for SIP
> >
> > ccn subsystem sip
> >  gateway address "10.1.131.1"
> >  mwi envelope-info
> >  mwi sip unsolicited
> >  end subsystem
> >
> > I've tried all kinds of config on the CME router without success.
> >
> > When running "debug ccsip messages" on the CME router , I don't see
> anything
> > if I issue "mwi refresh all" on CUE, even though I can dial into CUE and
> > check to hear a voicemail on dn 4001
> >
> >
> >
> > On Fri, Oct 21, 2011 at 6:47 PM, Brian  wrote:
> >>
> >> hi - this is an excellent summary of mwi for  cue that is worth a read
> >>
> >> http://blog.ipexpert.com/2010/07/19/sip-mwi-mechansims-on-cue-notify/
> >>
> >> Sent from my iPad
> >>
> >> On 21 Oct 2011, at 21:23, Ashraf Ayyash  wrote:
> >>
> >> > Hello Zamuel ,
> >> >
> >> > the mwi relay command is only needed in case of the subscribe notify
> >> > MWI and its not needed in case of using Unsolicited because it does
> >> > send the event to the phone using NOTIFY message no matter it
> >> > subscribed to the MWI server Or not .
> >> >
> >> > Ash
> >> >
> >> > On Fri, Oct 21, 2011 at 11:21 AM, zamuel del Toro
> >> >  wrote:
> >> >> Hi Vic, how is it going?,
> >> >>
> >> >> about mwi unsolicited.
> >> >>
> >> >> sip-ua
> >> >> mwi.. unsolicited
> >> >>
> >> >> telephony-ser
> >> >> mwi relay
> >> >>
> >> >> ephone-dn
> >> >> nothing
> >> >>
> >> >>
> >> >> works mwi
> >> >>
> >> >> if subscribe notify
> >> >> sip-ua
> >> >> mwi...
> >> >> telephony-ser
> >> >> nothing
> >> >>
> >> >> ephone-dn
> >> >> mwi sip
> >> >>
> >> >>
> >> >> both works fine
> >> >> what if make mistake if on unsolicited include on ephone-dn
> >> >> mwi sip,
> >> >> that work too.is wrong do this?
> >> >>
> >> >>
> >> >> thanks
> >> >>
> >> >>
> >> >>
> >> >>
> >> >>
> >> >> ___
> >> >> 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
> >
> >
>
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Re: [OSL | CCIE_Voice] cue-cme mwi unsolicited question

[ccielabrat]

I can't get CUE MWI working either.

This is my cue config for SIP

ccn subsystem sip
 gateway address "10.1.131.1"
 mwi envelope-info
 mwi sip unsolicited
 end subsystem

I've tried all kinds of config on the CME router without success.

When running "debug ccsip messages" on the CME router , I don't see anything
if I issue "mwi refresh all" on CUE, even though I can dial into CUE and
check to hear a voicemail on dn 4001



On Fri, Oct 21, 2011 at 6:47 PM, Brian  wrote:

> hi - this is an excellent summary of mwi for  cue that is worth a read
>
> http://blog.ipexpert.com/2010/07/19/sip-mwi-mechansims-on-cue-notify/
>
> Sent from my iPad
>
> On 21 Oct 2011, at 21:23, Ashraf Ayyash  wrote:
>
> > Hello Zamuel ,
> >
> > the mwi relay command is only needed in case of the subscribe notify
> > MWI and its not needed in case of using Unsolicited because it does
> > send the event to the phone using NOTIFY message no matter it
> > subscribed to the MWI server Or not .
> >
> > Ash
> >
> > On Fri, Oct 21, 2011 at 11:21 AM, zamuel del Toro 
> wrote:
> >> Hi Vic, how is it going?,
> >>
> >> about mwi unsolicited.
> >>
> >> sip-ua
> >> mwi.. unsolicited
> >>
> >> telephony-ser
> >> mwi relay
> >>
> >> ephone-dn
> >> nothing
> >>
> >>
> >> works mwi
> >>
> >> if subscribe notify
> >> sip-ua
> >> mwi...
> >> telephony-ser
> >> nothing
> >>
> >> ephone-dn
> >> mwi sip
> >>
> >>
> >> both works fine
> >> what if make mistake if on unsolicited include on ephone-dn
> >> mwi sip,
> >> that work too.is wrong do this?
> >>
> >>
> >> thanks
> >>
> >>
> >>
> >>
> >>
> >> ___
> >> 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 
>
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[OSL | CCIE_Voice] Using "Builtin:" B-ACD files ?

[ccielabrat]

Can anyone let me know if they have been able to use the "Built-In" b-acd
tcl scripts ?

You can see both app-b-acd and app-b-acd-aa are available internal to the
IOS by running
"Show call application voice summary"

I'm not able to get them to work properly and wanted to find out if it
should be possible in case they expect to use the "built-in" versions on the
lab.


Thanks
LabRat
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[OSL | CCIE_Voice] Resend: Task Order in the lab

[ccielabrat]

No takers?

-- Forwarded message --
From: ccielabrat 
Date: Sat, Sep 17, 2011 at 12:11 PM
Subject: Task Order in the lab
To: ccie_voice@onlinestudylist.com


To All,

I know this question has been asked 1000 times.
Now that I am ready to schedule my lab exam, I need to ask it again.

How are successful exam takers breaking down and grouping the tasks that
make up the lab
to allow for time savings ?

My natural approach so far is to take a sheet of paper and divide it into 4
boxes (HQ, BR1, BR2, CUCM)
Then I read through the practice exam and note the task # (i.e. 1.1 ) and a
1-3 word description in the device box that it relates to.
If a task includes or impacts multiple devices, it goes in multiple boxes.

I also note DN's in each location box and call restrictions for each dn in
shorthand.

In the CUCM box, I keep an area that notes "service parameters" that need
attention , in addition to the more general stuff.

In trying this a couple times, it seems I can get a straw man topology
working in short order.

But I'm concerned that it leaves me in a position where I will end up
jumping around to fine tune devices/locations multiple times.

I know the common strategy that was used in the prior version of the lab
("Touch each device once") is generally not the way with the new format.
Maybe with the troubleshooting in the exam now, my approach is flawed.

Any opinion or insight is appreciated.

LabRat
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[OSL | CCIE_Voice] Task Order in the lab

[ccielabrat]

To All,

I know this question has been asked 1000 times.
Now that I am ready to schedule my lab exam, I need to ask it again.

How are successful exam takers breaking down and grouping the tasks that
make up the lab
to allow for time savings ?

My natural approach so far is to take a sheet of paper and divide it into 4
boxes (HQ, BR1, BR2, CUCM)
Then I read through the practice exam and note the task # (i.e. 1.1 ) and a
1-3 word description in the device box that it relates to.
If a task includes or impacts multiple devices, it goes in multiple boxes.

I also note DN's in each location box and call restrictions for each dn in
shorthand.

In the CUCM box, I keep an area that notes "service parameters" that need
attention , in addition to the more general stuff.

In trying this a couple times, it seems I can get a straw man topology
working in short order.

But I'm concerned that it leaves me in a position where I will end up
jumping around to fine tune devices/locations multiple times.

I know the common strategy that was used in the prior version of the lab
("Touch each device once") is generally not the way with the new format.
Maybe with the troubleshooting in the exam now, my approach is flawed.

Any opinion or insight is appreciated.

LabRat
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[OSL | CCIE_Voice] UnifiedFX phoneview question

[CCIELabRat]

Anyone successfully get Phoneview working with CUCME?
I see "express" as an option in the group configuration with telnet://
as the protocol to use but I don't have any idea what would be needed
in the CUCME config.

Thanks
LabRat
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[OSL | CCIE_Voice] Couple questions about MOH on CUCME

[ccielabrat]

Hi All,

I'm working on a better understanding of the options with MOH from CME.
I have one endpoint registered as SCCP and one as SIP.

Questions:
1.) Does the SIP phone configuration for MOH get defined in the
telephony-service area?
2.) And does multicast MOH to sip endpoints work the same as SCCP ?
3.) If multicast moh is configured , is there any option for the phone to
fallback to unicast if it can't join the multicast?

What I have now is a call from SIP to SCCP works and if the SIP phone
"holds" the call, I do get moh on the SCCP.

If the SCCP phone "holds" the call, I get beeps on the SIP phone.

Any insight is appreciated.
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Re: [OSL | CCIE_Voice] OT : Missing normal softkeys from 9971.

[ccielabrat]

Randall & Roger,

Are you kidding me?

I admit it was a dumb question and I tried to be polite by marking it OT:
(Off Topic)
I was stuck in a lab with no direct internet access or a manual for a phone
I never used before.
I leveraged the mailer group on my phone looking for a hand.
A group, which for the most part, is helpful whether or not it has anything
specifically to do with the voice lab.

I got several quick replies with useful information.
The thread could have died 5 minutes after I posted it.
In the time it took to reply, both of you could have deleted it and moved on
with your day.
I don't see either of you replying back to morons who blatantly ask for NDA
information.

*Your arrogance is laughable*, to think either of you have a place to
suggest what does or doesn't belong on this mailing list.

If you have a problem with it , take it up with the mailer admin.
Or email me a picture of your mailer police badge , and then I'll consider
your email.

I honestly hope if you are ever in the same position, you both get a helpful
reply instead of the nonsense you both provided.



On Wed, Mar 16, 2011 at 9:21 PM, Randall Saborío Cubero
wrote:

> It's difficult to ignore it if it is addressed to a specialized mailing
> list. It also takes extra time to sort out the crap that does not belong
> from the important emails.
>
> I'm sure you will get some attention at the cisco support forums and 0
> rejection.
>
> El mié, 16-03-2011 a las 17:35 -0400, ccielabrat escribió:
> > Right, thats why I included OT : (Off topic) in the subject line.
> >
> > That way, for people who don't want to be bothered can simply ignore
> > it.
> >
> >
> > On Wed, Mar 16, 2011 at 4:14 PM, Roger Carpio 
> > wrote:
> > 9971 are not evaluated in CCIE Lab as far as I know therefore,
> > this is not the right place for this question. Try doing some
> > google or post this question at Cisco Support forums.
> >
> >
> http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/9971_9951_8961/8_5/english/user_guide/book_911/callingfeatures_rt.html#wp1027556
> >
> > Regards,
> > Roger Carpio.
> >
> >
> >
> > On Wed, Mar 16, 2011 at 1:41 PM,  wrote:
> > I'm hoping i missed something simple.
> > I just got a 9971 registered on a CUCM 7.x server.
> >
> > it works great but I noticed there are no available
> > softkeys for hold,
> > park , etc during a call.
> >
> > It's my first SIP phone I'm using, so is there
> > something I'm missing
> > regarding supplemental services?
> > ___
> > For more information regarding industry leading CCIE
> > Lab training, please visit www.ipexpert.com
> >
> >
> >
> > ___
> > For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
>
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Re: [OSL | CCIE_Voice] OT : Missing normal softkeys from 9971.

[ccielabrat]

Right, thats why I included OT : (Off topic) in the subject line.

That way, for people who don't want to be bothered can simply ignore it.


On Wed, Mar 16, 2011 at 4:14 PM, Roger Carpio wrote:

> 9971 are not evaluated in CCIE Lab as far as I know therefore, this is not
> the right place for this question. Try doing some google or post this
> question at Cisco Support forums.
>
>
> http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/9971_9951_8961/8_5/english/user_guide/book_911/callingfeatures_rt.html#wp1027556
>
> Regards,
> Roger Carpio.
>
>
> On Wed, Mar 16, 2011 at 1:41 PM,  wrote:
>
>> I'm hoping i missed something simple.
>> I just got a 9971 registered on a CUCM 7.x server.
>>
>> it works great but I noticed there are no available softkeys for hold,
>> park , etc during a call.
>>
>> It's my first SIP phone I'm using, so is there something I'm missing
>> regarding supplemental services?
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>
>
___
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[OSL | CCIE_Voice] OT : Missing normal softkeys from 9971.

[CCIELabRat]

I'm hoping i missed something simple.
I just got a 9971 registered on a CUCM 7.x server.

it works great but I noticed there are no available softkeys for hold,
park , etc during a call.

It's my first SIP phone I'm using, so is there something I'm missing
regarding supplemental services?
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[OSL | CCIE_Voice] Inbound SIP call to CUCM from CUBE goes unanswered

[CCIELabRat]

I'm testing a cube configuration in my lab setup.
I have H.323 coming from CME to CUBE running on R1 and then SIP to the CUCM
via a SIP trunk.
I see the proper dialpeers being triggered in CUBE, but the CUCM doesn't
seem to respond to the SIP call setup inbound.

Calls from CUCM to CME via CUBE work , so I'm pretty confident the SIP trunk
is functional.

Short of trying to look through CUCM traces, is there a good debug on
R1/Cube that would provide some insight into whats going on?
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Re: [OSL | CCIE_Voice] Converting SCCP phones to SIP

[ccielabrat]

Looks like I'm right back in the same mess again.
Now I can't seem to get the CIPC currently running SIP to run as a SCCP
phone.
I even changed the TFTP away from CME and pointed it to CUCM thinking that
would force the phone to register as an SCCP phone.
No good.

Is there a "factory reset'" type action available on CIPC ?

Looking at the logs , phone is simply requesting the SIP config file from
CUCM and failing with an "unprovisioned" message on the phone.
I've tried to configure the CIPC endpoint as SCCP and SIP without getting
either to work.

I'm in the weeds without a plan.
If anyone can point me in any direction, I'd appreciate it .



On Wed, Aug 11, 2010 at 7:55 PM, Cisco CCIE  wrote:

> Great feeling when you figure it out on your own
>
>
> I figured out the problem.
> And as usual, I'm the architect of my own demise :)
>
> I had redundant tftp-server statements for the term62.default.loads file.
>
> (along with others as well)
> One for SCCP and one for SIP.
> tftp-server flash:/SCCP.8-3-3s/term62.default.loads alias
> term62.default.loads
> tftp-server flash:/SIP.8-3-3/term62.default.loads alias term62.default.loads
>
> Apparently, if there are redundant tftp-server commands, the first one (in
> this case SCCP) takes precedence.
> As soon as I removed the first statement pointing towards the SCCP
> term62.default.loads, the phone took the SIP image.
>
> One more problem solved by endless hours of frustration ...
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
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[OSL | CCIE_Voice] RTMT for Dummies

[ccielabrat]

Can anyone point me towards a good RTMT link that will give the basics of
how to collect a trace from CUCM 7.x ?
I'm looking into a GK related issue and what to see what messaging is
happening on the CUCM side.

I'm stumbling badly with RTMT.

Any help  or links are greatly appreciated.
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Re: [OSL | CCIE_Voice] Converting SCCP phones to SIP

[ccielabrat]

I figured out the problem.
And as usual, I'm the architect of my own demise :)

I had redundant tftp-server statements for the term62.default.loads file.
(along with others as well)
One for SCCP and one for SIP.
tftp-server flash:/SCCP.8-3-3s/term62.default.loads alias
term62.default.loads
tftp-server flash:/SIP.8-3-3/term62.default.loads alias term62.default.loads

Apparently, if there are redundant tftp-server commands, the first one (in
this case SCCP) takes precedence.
As soon as I removed the first statement pointing towards the SCCP
term62.default.loads, the phone took the SIP image.

One more problem solved by endless hours of frustration ...




On Sat, Jul 31, 2010 at 7:09 PM, ccielabrat  wrote:

> Maybe I'm missing a part of this process.
> Should a factory reset be necessary?
>
> On 7/31/10, Miron Kobelski  wrote:
> > You have your load command in voice register global? Restore the phones
> to
> > factory defaults and provide on the TFTP all the files it will ask for.
> It
> > always worked for me. But I know longer change firmware on CUCME itself -
> > it's so much easier and quicker by temporary registering the phone on
> CUCM.
> >
> >
> > regards
> > kobel
> >
> > On Sun, Aug 1, 2010 at 12:13 AM, ccielabrat 
> wrote:
> >
> >> ete the ephone config (sccp)
> >> - issue no create cnf (to delete the sccp cnf info)
> >> - Created voice register pool with the mac address of the phone.
> >> - Loaded the sip firmware on the router and made files available via
> >> tftp-server command.
> >>
> >> No matter what I do, the thing will NOT ask for the SIP firmware.
> >>
> >
>
___
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Re: [OSL | CCIE_Voice] Converting SCCP phones to SIP

[ccielabrat]

Maybe I'm missing a part of this process.
Should a factory reset be necessary?

On 7/31/10, Miron Kobelski  wrote:
> You have your load command in voice register global? Restore the phones to
> factory defaults and provide on the TFTP all the files it will ask for. It
> always worked for me. But I know longer change firmware on CUCME itself -
> it's so much easier and quicker by temporary registering the phone on CUCM.
>
>
> regards
> kobel
>
> On Sun, Aug 1, 2010 at 12:13 AM, ccielabrat  wrote:
>
>> ete the ephone config (sccp)
>> - issue no create cnf (to delete the sccp cnf info)
>> - Created voice register pool with the mac address of the phone.
>> - Loaded the sip firmware on the router and made files available via
>> tftp-server command.
>>
>> No matter what I do, the thing will NOT ask for the SIP firmware.
>>
>
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[OSL | CCIE_Voice] Converting SCCP phones to SIP

[ccielabrat]

I'm running into a strange problem where an existing phone running sccp
firmware will not register using SIP firmware.
I've done the following

- Delete the ephone config (sccp)
- issue no create cnf (to delete the sccp cnf info)
- Created voice register pool with the mac address of the phone.
- Loaded the sip firmware on the router and made files available via
tftp-server command.

No matter what I do, the thing will NOT ask for the SIP firmware.

I'm using 8.3.3 firmware for SCCP and SIP.

Thanks in advance.
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[OSL | CCIE_Voice] Question on "Alias Static" configuration.

[CCIELabRat]

Assuming the following requirements for BR2 to HQ calling via Gatekeeper:
- Do not use tech prefix or default tech prefix
- Route calls to Sub and then to Pub if Sub unavailable.

So in this case a "Static Alias" configuration would be needed to allow the
call to route to CUCM without having the usual 1# tech prefix.

The problem I'm seeing is you can only target a single IP for any given e164
number.

Is it possible to use "static alias" and still allow for redundancy ?
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Re: [OSL | CCIE_Voice] H323 gateway inbound calls issue.

[CCIELabRat]

Maybe a codec selection issue.
Do you have any restrictions on CUCM (Regions, Locations) that would
disallow the call setup?


On Sat, Jun 12, 2010 at 11:03 AM, jammer jones wrote:

> What would cause the following issues with a gw configured as h323 to cucm
>
> outbound calls work just fine
>
> inbound calls from a pstn phone does not work.  PSTN phone displays unknown
> number and then a fast busy - this is the problem.
>
> when in srst mode outbound and inbound calls work just fine. - this makes
> me believe it is something in cucm.  I verified the config on the gateway in
> cucm 6 times.
>
>
> My config on the gateway was this.
> voice translation-rule 1
> rule 1 /.*\(\)/ /\1/
>
> dial-peer voice 1 pots
> translation-profile incoming frompstn
> incoming called-number .
> direct-inward-dial
> port 0/0/0:23
>
> card type t1 0 0
> network-clock participate wic 0
> isdn switch-type primary-ni
>
> controller t1 0/0/0
> pri-group timeslots 1-24
>
> inter ser 0/0/0:23
> isdn outgoing display-ie
> isdn outgoing ie redirecting-number
>
> interface vlan 302
> h323-gateway voip bind srcaddr 1.1.1.1
>
> dial-peer voice 100 voip
> destination-pattern 3...$
> session target ipv4:2.2.2.2
> dtmf-relay h245-alphanumeric
> no vad
>
>
> dial-peer voice 101 voip
> preference 1
> destination-pattern 7...$
> session target ipv4:2.2.2.1
> dtmf-relay h245-alphanumeric
> no vad
>
>
> config on cucm was all of the usual.  I verified i was pointing to 1.1.1.1
> several times, verified i had a correct inbound css on gateway, etc..
>
>
> the debug isdn q931 showed invalid information element.
> the debug cch323 h225 showed what appeared to be a proper call flow.
>
>
> Unfortunately i am unable to reproduce the same behavior so i do not have
> better debug messages.
>
>
> could something me set coming in from the telco that would break inbound
> calls going to phones registered to cucm, but not phones registered to srst?
>
>
>
> Thanks
>
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
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Re: [OSL | CCIE_Voice] Expensive Lunch

[CCIELabRat]

Just my 2 cents.

I had a very similar experience as you in terms of scoring. Some items I
know where working got no points.
It's my impression (and others I've spoken with) is that once the proctor
gets to -21 on your score, they simply stop any further testing in most
cases.
I could be wrong and I would suspect they are instructed by Cisco to score
the whole exam but I don't believe they do.



On Fri, May 21, 2010 at 2:25 PM, Ashar Siddiqui  wrote:

>  Don't worry Jeff, I am also in the same boat as you. Few questions where I
> was sure to get 100%, I didn't get! You will have to sit and carefully look
> at what was asked and what you configured. Sometimes getting desired results
> is not the solution, getting desired results as per Cisco way is what they
> are looking for.
>
> Ash>
>
>
> Kevin Damisch wrote:
>
> I feel your pain.  I remember not getting points for a section that I triple 
> checked from each phone at the end of the day, ran the debugs to make sure 
> number/type were set to what is asked, checked the display on both the 
> calling/called phones, all matched up to what was asked as it was slightly 
> different than what you would expect due to the curveballs they like to throw 
> at you.  Very tough to know what was missed in these situations.
>
> Make sure you document everything, go home, and mock it up again.  I've 
> caught myself on a few things by doing that.
>
> Best of luck next time.
>
> 
> From: ccie_voice-boun...@onlinestudylist.com 
> [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Cotter 
> [jcot...@voxns.com]
> Sent: Friday, May 21, 2010 11:59 AM
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] Expensive Lunch
>
> Sat the lab yesterday….failed.
>
> So I need to vent a little bit.  There is no doubt I failed the exam and it 
> was painfully obvious I am not ready to be a CCIE.  However the most 
> unsettling  piece is not getting the points in areas that I “thought” were 
> working and verified.  I do not know how to address this the next time.  My 
> fear is, I will probably program these items exactly the same way next time… 
> because that is the way I know how….they coincide with the training materials 
> available and most importantly they seem to work…. and I will not get the 
> points AGAIN!!  I just do not understand why I did not get these points or 
> how to fix it the next time….frustrated.
>
> Lunch was good though!
>
>
>
>
>
>
> 
> This communication (including any attachments) is intended only for the use 
> of the individual or entity to which it is addressed, and may contain 
> information that is privileged, confidential and exempt from disclosure under 
> applicable law. If you are not the intended recipient, any dissemination, 
> distribution or copying of this communication is strictly prohibited. If you 
> have received this communication in error, please notify Vital Support 
> Systems at 515 334 5700 and delete or destroy all copies and the original 
> document.
> ___
> For more information regarding industry leading CCIE Lab training, please 
> visit www.ipexpert.com
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
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[OSL | CCIE_Voice] Console cable for a 7970

[CCIELabRat]

I've found a couple links about being able to get a console session to an IP
phone via the AUX port.Completely unsupported but apparently do-able.

Has anyone messed with this?
I have a 7970 that refuses to boot, it flashes but then goes slient.
Before I trash it, I figured I would try to find out if it posts an error
through this interface.
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[OSL | CCIE_Voice] DSCP for SCCP

[CCIELabRat]

In terms of modifying the signaling DSCP value for  phones and what is
generated by CallManager,
Are the entries noted below the only ones I need to check?


Enterprise Parameters

- DSCP for Cisco CallManager to Device Interface*
- DSCP for SCCP Phone Configuration*

Service Parameters:
- Voice Streaming APP:
  IP Type of Service to Cisco CallManager*

- CTI Manager Services "Advanced:
  DSCP for ICCP Protocol Links*
  DSCP IP CTIManager to Application*
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[OSL | CCIE_Voice] Clarifying VPM Broadcast setup

[CCIELabRat]

I think I have the VPIM Broadcast config straight now.But I want to confirm
I'm doing it the "normal" way.

CUE is configured with it's location VPIM-Broadcast ID as 852.

I'm creating a VPIM subscriber with an remote mailbox of "852" and assigning
it to the cue vpim location.
I'm creating a PDL "" and including all HQ and SB phones and VPIM user
"852" in it.

>From Unity Broadcast admin , i'm selecting "" as a PDL and leaving the
message.

What I'm noticing is that the message is delivered as a "normal" VM to the
HQ and SB sites but as a true broadcast to the CUE users.

It seems to work (MWI, etc.) , just curious if there is a variant on the
config .

Also, I'm assuming the same basic scenario is used for CUE to Unity
Broadcast ?
But Unity doesn't light an MWI for broadcast delivery right ?

[OSL | CCIE_Voice] Follow up on an old question (Unity MWI)

[CCIELabRat]

I just discovered a behavior that surprised me.
If I register only 3 voicemail ports in CM but have 8 port configured in
Unity, the mwi's behavior is
unpredictable.
I can only guess that Unity is attempting to use port 4-8 for MWI ,
regardless of there status within the system.
As soon as I uncheck the enabled button in the UTIM for ports 4-8, MWI
starts working great.

Just something to watch out for.

- Scott

Re: [OSL | CCIE_Voice] Voice translation question.

[CCIELabRat]

Thanks for the quick reply.
Actually it is working , I fat fingered it in the config.


On Wed, Jun 24, 2009 at 3:30 PM, Cyrus  wrote:

> Hi,
> It's not work with "translation-rule 1"
>
> use
>
> voice translation-rule 1
> rule 1 /^\*\(400.\)/ /\1/
>
>
> Cheers,
>
> Cyrus
>
>
> On Thu, Jun 25, 2009 at 5:18 AM,  wrote:
> > I'd like to be able to match a dial string with the format *400.
> > I'd like to match the beginning * and then strip it.
> > I can't seem to get a valid voice translation rule though.
> > I assumed I would have to "escape" the * by using \*
> > but I get an error when I try to use the following.
> > rule 1 /^\*\(400.\)/ /\1/
> > I'm I thinking about this incorrectly?
> > I also tried just having * as a single character (no escape).
> > The documentation suggests that I should be able to do this.
> > But I get an error with this one also.
> > rule 1 /*\(400.\)/ /\1/
> > 
> >
>
>
>
> --
> Sirus Moghadasian
> CCIE #21862 (R&S)
>

[OSL | CCIE_Voice] Voice translation question.

[CCIELabRat]

I'd like to be able to match a dial string with the format *400.I'd like to
match the beginning * and then strip it.

I can't seem to get a valid voice translation rule though.
I assumed I would have to "escape" the * by using \*
but I get an error when I try to use the following.

rule 1 /^\*\(400.\)/ /\1/

I'm I thinking about this incorrectly?

I also tried just having * as a single character (no escape).
The documentation suggests that I should be able to do this.
But I get an error with this one also.

rule 1 /*\(400.\)/ /\1/

[OSL | CCIE_Voice] What is the difference between VPIM delivery ?

[CCIELabRat]

I'm testing VPIM delivery from Unity to CUE using a PDL with an extension of
666.
If I send a vm directly to PDL ext 666 , the message goes through and the
MWI light goes on for CUE extensions.
If I send a vm via "Broadcast Manager" to PDL 666 , the message doesn't go
through and the MWI light doesn't go on (obviously).

The only difference I can see is the subject line in the VPIM trace on CUE
reads "Broadcast Message" instead of "From x...@xxx.xxx"

?

Re: [OSL | CCIE_Voice] CME Fast Transfer Problem

[CCIELabRat]

I haven't tried that.I thought if you have full-consult configured, you
would have to press transfer a second time to complete the transfer.
I'll test it on my setup which is a 3825 running 12.4.5b


On Wed, Apr 22, 2009 at 7:00 PM, Sergio Polizer wrote:

>  Have you entered  "transfer-system full-consult" ?
>
> I got make this work with same config as you posted (plus the above line)
> for internal and external calls with IOS 12.4(3g).
>
> Sergio.
>
> --
> Date: Wed, 22 Apr 2009 09:10:04 -0400
> From: ccielab...@gmail.com
> To: lmord...@cisco.com
> CC: ccie_voice@onlinestudylist.com; prabaha...@gmail.com
>
> Subject: Re: [OSL | CCIE_Voice] CME Fast Transfer Problem
>
> The monitored line is a ephone-dn of 54xxx.
> It's not assigned as a standard line to any phone, so it's ALWAYS idle.
>
> So , by pressing the transfer softkey and then the button assigned as
> monitor like it acts as a transfer + speed dial.
>
> Strange thing is I wasn't able to get this to work with an actual
> "speed-dial" configured on the transferring phone.
> I would think that it should work the same way but it doesn't.
>
>
>
>
> On Wed, Apr 22, 2009 at 8:32 AM, Linda Mordosky (lmordosk) <
> lmord...@cisco.com> wrote:
>
> Out of curiosity, how are you getting the transfer to go to VM when the
> monitored line is not busy?
>
> The System Guide states:
>
> When a monitored line is idle, pressing the monitor button will speed-dial
> the monitored line. This functionality is sometimes known as fast transfer
> or direct station select.
>
>
>
>
> -Original Message-
> From: ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] On Behalf Of Prabahar M
> Sent: Wednesday, April 22, 2009 12:55 AM
> To: ccie_voice@onlinestudylist.com
>  Subject: Re: [OSL | CCIE_Voice] CME Fast Transfer Problem
>
>  transfer-system full-blind will work for this problem. Is there
> anyway just to press transfer softkey without answering the call and
> the monitor button to send call to greeting?  There is no transfer
> sofkey available on the softkey template for alert. Callers hears
> music before call connects to the greeting after pressing transfer
> softkey.
>
> Thanks,
> Prabahar
>
> On Tue, Apr 21, 2009 at 10:02 PM,
>  wrote:
> > Date: Tue, 21 Apr 2009 22:07:44 -0600
> > From: J Delgado 
> > Subject: [OSL | CCIE_Voice] CME Fast Transfer Problem
> > To: 
> > Cc: ccie_voice@onlinestudylist.com
> >
> >
> > I had the same issue. I recommend you to configure under
> telephony-service transfer-system full-blind so when you transfer the call
> to Unity, you only need to press the Transfer button once.
> >
> > Cheers,
> >
> > Juan
> >
> >
> >> From: ccie_voice-requ...@onlinestudylist.com
> >> Subject: CCIE_Voice Digest, Vol 38, Issue 127
> >> To: ccie_voice@onlinestudylist.com
> >> Date: Wed, 22 Apr 2009 00:35:28 -0400
> >>
> >> Send CCIE_Voice mailing list submissions to
> >>   ccie_voice@onlinestudylist.com
> >>
> >> To subscribe or unsubscribe via the World Wide Web, visit
> >>   http://onlinestudylist.com/mailman/listinfo/ccie_voice
> >> or, via email, send a message with subject or body 'help' to
> >>   ccie_voice-requ...@onlinestudylist.com
> >>
> >> You can reach the person managing the list at
> >>   ccie_voice-ow...@onlinestudylist.com
> >>
> >> When replying, please edit your Subject line so it is more specific
> >> than "Re: Contents of CCIE_Voice digest..."
> >>
> >>
> >> Today's Topics:
> >>
> >>1. Re: AAR and Offnet Transfers (James Key)
> >>2. Re: Version 3 Lab equipment... (Michael Ciarfello)
> >>3. CME Fast Transfer Problem. (ccielab...@gmail.com)
> >>4. Re: CME Fast Transfer Problem. (Cliff McGlamry)
> >>5. Re: CME Fast Transfer Problem. (Cliff McGlamry)
> >>
> >>
> >> --
> >>
> >> Message: 1
> >> Date: Tue, 21 Apr 2009 19:50:42 -0500
> >> From: James Key 
> >> Subject: Re: [OSL | CCIE_Voice] AAR and Offnet Transfers
> >> To: Prabahar M ,
> >>   "ccie_voice@onlinestudylist.com"<
> ccie_voice@onlinestudylist.com>
> >> Message-ID:
> >>   <
> 7105bfd589769844b43b67bbfa844a9b015957c...@mmoexchmbs01.jhacorp.com>
> >> Content-Type: text/plain; charset="us-ascii"
> >>
> >> Hi Prabahar.  It is still not working for me and I can't seem to find a
> working solution.
> >>
> >> Has anyone been able to get this to work?
> >>
> >> James Key
> >>
> >> 
> >> From: Prabahar M [prabaha...@gmail.com]
> >> Sent: Tuesday, April 21, 2009 5:50 PM
> >> To: James Key
> >> Subject: Re: AAR and Offnet Transfers
> >>
> >> Hi James,
> >>  Does it work for you? I tried  and it is not working for me. If it
> >> works for you, could you send me the config steps please.
> >>
> >> Thansk,
> >> Prabahar
> >>
> >> Thanks for the reply Cliff.  That is what I was thinking and did
> >> configure it this way, but when site B tries to transfer i get the
> >> me

Re: [OSL | CCIE_Voice] CME Fast Transfer Problem

[CCIELabRat]

The monitored line is a ephone-dn of 54xxx.
It's not assigned as a standard line to any phone, so it's ALWAYS idle.

So , by pressing the transfer softkey and then the button assigned as
monitor like it acts as a transfer + speed dial.

Strange thing is I wasn't able to get this to work with an actual
"speed-dial" configured on the transferring phone.
I would think that it should work the same way but it doesn't.




On Wed, Apr 22, 2009 at 8:32 AM, Linda Mordosky (lmordosk) <
lmord...@cisco.com> wrote:

> Out of curiosity, how are you getting the transfer to go to VM when the
> monitored line is not busy?
>
> The System Guide states:
>
> When a monitored line is idle, pressing the monitor button will speed-dial
> the monitored line. This functionality is sometimes known as fast transfer
> or direct station select.
>
>
>
>
> -Original Message-
> From: ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] On Behalf Of Prabahar M
> Sent: Wednesday, April 22, 2009 12:55 AM
> To: ccie_voice@onlinestudylist.com
>  Subject: Re: [OSL | CCIE_Voice] CME Fast Transfer Problem
>
>  transfer-system full-blind will work for this problem. Is there
> anyway just to press transfer softkey without answering the call and
> the monitor button to send call to greeting?  There is no transfer
> sofkey available on the softkey template for alert. Callers hears
> music before call connects to the greeting after pressing transfer
> softkey.
>
> Thanks,
> Prabahar
>
> On Tue, Apr 21, 2009 at 10:02 PM,
>  wrote:
> > Date: Tue, 21 Apr 2009 22:07:44 -0600
> > From: J Delgado 
> > Subject: [OSL | CCIE_Voice] CME Fast Transfer Problem
> > To: 
> > Cc: ccie_voice@onlinestudylist.com
> >
> >
> > I had the same issue. I recommend you to configure under
> telephony-service transfer-system full-blind so when you transfer the call
> to Unity, you only need to press the Transfer button once.
> >
> > Cheers,
> >
> > Juan
> >
> >
> >> From: ccie_voice-requ...@onlinestudylist.com
> >> Subject: CCIE_Voice Digest, Vol 38, Issue 127
> >> To: ccie_voice@onlinestudylist.com
> >> Date: Wed, 22 Apr 2009 00:35:28 -0400
> >>
> >> Send CCIE_Voice mailing list submissions to
> >>   ccie_voice@onlinestudylist.com
> >>
> >> To subscribe or unsubscribe via the World Wide Web, visit
> >>   http://onlinestudylist.com/mailman/listinfo/ccie_voice
> >> or, via email, send a message with subject or body 'help' to
> >>   ccie_voice-requ...@onlinestudylist.com
> >>
> >> You can reach the person managing the list at
> >>   ccie_voice-ow...@onlinestudylist.com
> >>
> >> When replying, please edit your Subject line so it is more specific
> >> than "Re: Contents of CCIE_Voice digest..."
> >>
> >>
> >> Today's Topics:
> >>
> >>1. Re: AAR and Offnet Transfers (James Key)
> >>2. Re: Version 3 Lab equipment... (Michael Ciarfello)
> >>3. CME Fast Transfer Problem. (ccielab...@gmail.com)
> >>4. Re: CME Fast Transfer Problem. (Cliff McGlamry)
> >>5. Re: CME Fast Transfer Problem. (Cliff McGlamry)
> >>
> >>
> >> --
> >>
> >> Message: 1
> >> Date: Tue, 21 Apr 2009 19:50:42 -0500
> >> From: James Key 
> >> Subject: Re: [OSL | CCIE_Voice] AAR and Offnet Transfers
> >> To: Prabahar M ,
> >>   "ccie_voice@onlinestudylist.com"<
> ccie_voice@onlinestudylist.com>
> >> Message-ID:
> >>   <
> 7105bfd589769844b43b67bbfa844a9b015957c...@mmoexchmbs01.jhacorp.com>
> >> Content-Type: text/plain; charset="us-ascii"
> >>
> >> Hi Prabahar.  It is still not working for me and I can't seem to find a
> working solution.
> >>
> >> Has anyone been able to get this to work?
> >>
> >> James Key
> >>
> >> 
> >> From: Prabahar M [prabaha...@gmail.com]
> >> Sent: Tuesday, April 21, 2009 5:50 PM
> >> To: James Key
> >> Subject: Re: AAR and Offnet Transfers
> >>
> >> Hi James,
> >>  Does it work for you? I tried  and it is not working for me. If it
> >> works for you, could you send me the config steps please.
> >>
> >> Thansk,
> >> Prabahar
> >>
> >> Thanks for the reply Cliff.  That is what I was thinking and did
> >> configure it this way, but when site B tries to transfer i get the
> >> message that transfer is not allowed.  I will take a closer look.
> >>
> >> James Key
> >> 
> >> From: Cliff McGlamry [cliff at mcglamry.net]
> >> Sent: Monday, April 20, 2009 7:08 PM
> >> To: James Key; ccie_voice at onlinestudylist.com
> >> Subject: Re: [OSL | CCIE_Voice] AAR and Offnet Transfers
> >>
> >> Usually when you get something like this it's because you have a
> >> situation where Site A calls site B via AAR.  Then the user at site B
> >> is transferring the call outside to somewhere else.
> >>
> >> What you do is set offnet to offnet transfers as disabled.  Then you
> >> create an AAR route pattern to support the call from A to B, but
> >> classify the call as on net.  Then the 

Re: [OSL | CCIE_Voice] CME Fast Transfer Problem.

[CCIELabRat]

I also found I can go with the original voice translation setup and just add
a more specific pots dial-peer like below


dial-peer voice 5 pots
 incoming called-number 54...
 direct-inward-dial


Any idea why the CME would have to reprocess the call like this?



On Tue, Apr 21, 2009 at 11:47 PM,  wrote:

> Link :
> http://ftp.ntu-kpi.kiev.ua/www.cisco.com/univercd/cc/td/doc/product/voice/its/cme33/cme33sa/cme33atd.htm#wp2198094
>
> *I found the problem but I don't understand it.*
> **
> If I turn on debug voice ccapi inout , I can see the initial is released
> upon transfer and another call is being set up as if the external number is
> calling the 54xxx number.
>
> The dial-peer voice 1 is being matched again and the translation pattern is
> invoked, thereby causing the destination # to be trimmed from 54xxx to 4xxx.
>
> I adjusted the translation rules as follows and now it works.
> I'm guessing this has something to do with the transfer method being
> invoked by CME.
> (h450.3 ?)
>
> voice translation-rule 1
>  rule 1 /^54...$/ /\0/
>  rule 2 /.*\(\)/ /\1/
>
> I'd REALLY like to understand what is going on under the hook on this one.
>
>
>
>
> On Tue, Apr 21, 2009 at 11:36 PM, Cliff McGlamry wrote:
>
>>  Not sure, but you might add either transfer-system full-consult or
>> transfer-system full-blind.
>>
>> How about providing a link to the doc you're saying you're looking at?
>>
>>
>>
>> - Original Message -
>> *From:* ccielab...@gmail.com
>> *To:* OSL Group 
>> *Sent:* Tuesday, April 21, 2009 11:17 PM
>> *Subject:* [OSL | CCIE_Voice] CME Fast Transfer Problem.
>>
>> I'm seeing an unexpected behavior while setting up a fast transfer to
>> voicemail scenario.
>>
>> The intent is as follows:
>>
>> Call comes into extension 4001
>> Phone is configured with a primary ext of 4001
>> and a second button labeled "transfer to 4002 VM"
>>
>> The call needs to be transferred directly to VM greeting for 4002.
>> The user needs to be able to do this by pressing softkey "transfer"
>> and then the second button.
>> The user SHOULD NOT have to press transfer a second time.
>>
>> In researching this , I came across a document on "Fast Transfers" using
>> the monitor line configuration.
>>
>> The issue I'm seeing is that the scenario works for calls initiated
>> internally.
>> 4003->4001 ->(fast transfer) -> VM for 4002
>>
>> But if the call comes in from the PSTN and I attempt a fast transfer, the
>> call
>> gets forwarded to the actual ext 4002.
>>
>> I can't see what I'm  missing.
>> I've pasted in most of the config from my SiteC router.
>>
>> Any ideas are very appreciated.
>>
>>
>>
>> voice translation-rule 1
>>  rule 1 /.*\(\)/ /\1/
>> !
>> voice translation-rule 2
>>  rule 1 /^4.../ /8522405\0/
>> !
>> voice translation-rule 54000
>>  rule 1 /.*\(\)/ /\1/
>> !
>> voice translation-profile pstn-in
>>  translate called 1
>> !
>> voice translation-profile pstn-out-11
>>  translate calling 2
>> !
>> voice translation-profile vmdirect
>>  translate redirect-called 54000
>> !
>> dial-peer voice 1 pots
>>  translation-profile incoming pstn-in
>>  incoming called-number .
>>  direct-inward-dial
>> !
>> dial-peer voice 4000 voip
>>  translation-profile outgoing vmdirect
>>  destination-pattern 4[15]..
>>  session protocol sipv2
>>  session target ipv4:142.105.66.253
>>  dtmf-relay sip-notify
>>  codec g711ulaw
>> !
>> telephony-service
>>  ip source-address 142.105.66.254 port 2000
>>  sdspfarm units 1
>>  sdspfarm transcode sessions 3
>>  sdspfarm tag 1 MTP00141cd53fc1
>>  voicemail 4111
>>  call-forward pattern .T
>>  transfer-pattern .T
>> !
>>
>> ephone-dn  1  dual-line
>>  number 4001
>>  call-forward busy 4111
>>  call-forward noan 4111 timeout 6
>> !
>> ephone-dn  2  dual-line
>>  number 4002
>>  call-forward busy 4111
>>  call-forward noan 4111 timeout 6
>> !
>> ephone-dn  3  dual-line
>>  number 4003
>>  call-forward busy 4111
>>  call-forward noan 4111 timeout 6
>> !
>> ephone-dn  24
>>  number 54002
>>  label call.vm.4002
>>  call-forward all 4111
>>
>> ephone  1
>>  mac-address 0004.27E6.921A
>>  type 7940
>>  button  1:1 2m24
>>
>>
>

[OSL | CCIE_Voice] CME Fast Transfer Problem.

[CCIELabRat]

I'm seeing an unexpected behavior while setting up a fast transfer to
voicemail scenario.

The intent is as follows:

Call comes into extension 4001
Phone is configured with a primary ext of 4001
and a second button labeled "transfer to 4002 VM"

The call needs to be transferred directly to VM greeting for 4002.
The user needs to be able to do this by pressing softkey "transfer"
and then the second button.
The user SHOULD NOT have to press transfer a second time.

In researching this , I came across a document on "Fast Transfers" using
the monitor line configuration.

The issue I'm seeing is that the scenario works for calls initiated
internally.
4003->4001 ->(fast transfer) -> VM for 4002

But if the call comes in from the PSTN and I attempt a fast transfer, the
call
gets forwarded to the actual ext 4002.

I can't see what I'm  missing.
I've pasted in most of the config from my SiteC router.

Any ideas are very appreciated.



voice translation-rule 1
 rule 1 /.*\(\)/ /\1/
!
voice translation-rule 2
 rule 1 /^4.../ /8522405\0/
!
voice translation-rule 54000
 rule 1 /.*\(\)/ /\1/
!
voice translation-profile pstn-in
 translate called 1
!
voice translation-profile pstn-out-11
 translate calling 2
!
voice translation-profile vmdirect
 translate redirect-called 54000
!
dial-peer voice 1 pots
 translation-profile incoming pstn-in
 incoming called-number .
 direct-inward-dial
!
dial-peer voice 4000 voip
 translation-profile outgoing vmdirect
 destination-pattern 4[15]..
 session protocol sipv2
 session target ipv4:142.105.66.253
 dtmf-relay sip-notify
 codec g711ulaw
!
telephony-service
 ip source-address 142.105.66.254 port 2000
 sdspfarm units 1
 sdspfarm transcode sessions 3
 sdspfarm tag 1 MTP00141cd53fc1
 voicemail 4111
 call-forward pattern .T
 transfer-pattern .T
!

ephone-dn  1  dual-line
 number 4001
 call-forward busy 4111
 call-forward noan 4111 timeout 6
!
ephone-dn  2  dual-line
 number 4002
 call-forward busy 4111
 call-forward noan 4111 timeout 6
!
ephone-dn  3  dual-line
 number 4003
 call-forward busy 4111
 call-forward noan 4111 timeout 6
!
ephone-dn  24
 number 54002
 label call.vm.4002
 call-forward all 4111

ephone  1
 mac-address 0004.27E6.921A
 type 7940
 button  1:1 2m24

[OSL | CCIE_Voice] CME Transfer

[CCIELabRat]

I'm trying to setup a quick transfer to voicemail option.
I'd like to have a scenario as follows:

Extension 5001 receives a call.
A speed dial is configured on the second button of this ephone (*5002)
The user on 5001 presses transfer and the speed dial button for *5002.
Pressing Transfer again is NOT needed.

The call should go directly to voicemail greeting for 5002.

I have the whole vmdirect thing setup with translation patterns so if I
simply press the speed-dial on 5001, the call goes directly to 5002's
greeting.
I can't get a call from PSTN to be directly transferred to 5002's Voicemail
greeting though.
The PSTN call simply gets disconnected when I press the speed-dial button.

I've tried a couple of options with call-forward and transfer pattern, but
nothing works.

Any ideas?

[OSL | CCIE_Voice] Digging for VATS

[CCIELabRat]

Can someone tell me where to look for the VATS configuration info ?I know I
can do a search and find it but in case I have to go digging for it on lab
day, I need to know how to navigate to it.

[OSL | CCIE_Voice] Call-forward vs transfer-pattern in UCME

[CCIELabRat]

Can someone help me understand what I'm missing regarding call-forward
pattern and transfer-pattern in CME.
I'd like to control where a call can be transferred to.
Either as a consult transfer or using CfwdAll .

[OSL | CCIE_Voice] IPCC Prompt Variable Format.

[CCIELabRat]

Probably a very elementary question but it's never been clear to me.
Regarding how prompt files are referenced within an IPCC script:
How do it related to WHERE the prompt files are expected to be?

Example

QueuePrompt = SP[ICD\ICDQUEUE.wav]
vs
QueuePrompt = P[myprompt.wav]

Re: [OSL | CCIE_Voice] Extension Mobility Question

[CCIELabRat]

Alex,
Thanks for the reply.
I got it working.



On Thu, Mar 26, 2009 at 2:46 PM, Alex  wrote:

>  Have 911/9911 route patterns only in device CSS? Line CSS should not have
> any such pattern.
> Rgds
> Alex
>
> - Original Message -
> *From:* ccielab...@gmail.com
> *To:* ccie_voice@onlinestudylist.com
> *Sent:* Thursday, March 26, 2009 6:01 PM
> *Subject:* [OSL | CCIE_Voice] Extension Mobility Question
>
> Can someone point me towards info on how to approach EM to allow a user to
> move between HQ and BR1 but make sure 911 calls go out only the local GW.
>
>

[OSL | CCIE_Voice] Extension Mobility Question

[CCIELabRat]

Can someone point me towards info on how to approach EM to allow a user to
move between HQ and BR1 but make sure 911 calls go out only the local GW.

[OSL | CCIE_Voice] Understanding Transcoding

[CCIELabRat]

I'm a little confused about how transcoding is invoked.
Assuming I have an endpoint that makes a call (g.729) to another endpoint
requiring g.711u.
Is it ALWAYS the g.711 side that must allocate the transcoder resource?

[OSL | CCIE_Voice] Call forward to CUE failing.

[CCIELabRat]

I think I'm running into a bug but wanted to check if I might be missing
something.
I have a call coming into CME via the HQ-rtr GK.
The call goes through fine if answered on the CME phone. (g729).

If the call go noan to the CUE, I get a fast busy on the HQ phone.
I also get a traceback msg on BR2.

I have the voice services voip configured to allow  H323 to SIP connections
(all four options).

Is there something else I'm missing?

[OSL | CCIE_Voice] Trunk Type.

[CCIELabRat]

Outside of being told specifically to configure a h.225 trunk vs. an IC
Trunk, is there any reason to choose one over another in terms of connecting
to the BR2 CME ?

[OSL | CCIE_Voice] Local Directory on CME ?

[CCIELabRat]

Any trick to getting local directory to work on CME?I have Service
local-directory configured and a handful of entries but can't get it to come
up on the phone.

I also have http server and http path configured.

I've re-created the cnf files for good mesaure but nothing works.

Re: [OSL | CCIE_Voice] Gatekeeper blues :|

[CCIELabRat]

 Media Termination Point Required

I had to have a check in the checkbox to get the call to work.



On Mon, Mar 23, 2009 at 12:54 PM, Joel Jose  wrote:

> ohh.. i take that back, so CCIELabRat... what was it??.. u had the mtp
> "on" ? or "off"??.. very curious now to see how it worked..
>
> On Mon, Mar 23, 2009 at 10:14 PM, Joel Jose  wrote:
>
>> no way... he must have meant that.. he "off" ed it.. and then it worked...
>> the first case..the signaling went through..but the media must have
>> terminated at the gateways...
>>
>>
>> On Mon, Mar 23, 2009 at 10:09 PM, Chris Parker wrote:
>>
>>> Does the call complete as 729 withthe mtp checked?
>>>
>>> --
>>> From: ccielab...@gmail.com
>>> Sent: Monday, March 23, 2009 12:28 PM
>>> To: Chris Parker 
>>> Cc: ccie_voice@onlinestudylist.com 
>>> Subject: Re: [OSL | CCIE_Voice] Gatekeeper blues :|
>>>
>>> Yep, everything started working once I checked off requires MTP on the
>>> trunk. .
>>>
>>> But why? The call is coming in as g.729 and the device pool the trunk is
>>> in is set to g.729
>>>
>>>
>>>
>>>
>>>
>>> On Mon, Mar 23, 2009 at 11:54 AM, Chris Parker wrote:
>>>
>>>> If you are getting ring through when you place the call, then your GK
>>>> config is probably OK. If its ringing that tells me the GK sent the call to
>>>> the UCM.
>>>>
>>>> The fact that it goes fast busy when you pick up sounds like a
>>>> codec/media problem to me. Check the media resources config and your
>>>> regions.
>>>>
>>>> Chris
>>>>
>>>> ccielab...@gmail.com wrote:
>>>>
>>>>> I'm working through scenario , trying to get 4 digit dial from BR2 to
>>>>> HQ working via the HQ GK.
>>>>> The GK config looks like this.
>>>>>
>>>>> gatekeeper
>>>>>  zone local UCM ipexpert.com <http://ipexpert.com> 142.1.1.1
>>>>>  zone prefix UCM 1... gw-priority 10 Trunk_2
>>>>>  zone prefix UCM 1... gw-priority 9 Trunk_1
>>>>>  zone prefix UCM 1... gw-priority 0 cmegw
>>>>>  zone prefix UCM 3... gw-priority 10 cmegw
>>>>>  zone prefix UCM 3... gw-priority 0 Trunk_2 Trunk_1
>>>>>  gw-type-prefix 1#* gw ipaddr 142.2.64.12 1720 gw ipaddr 142.2.64.11
>>>>> 1720
>>>>>  no shutdown
>>>>>
>>>>> I have the UCM registering without a tech-prefix defined and I've
>>>>> adjusted the "service parameter" to insure the UCM listens on port 1720.
>>>>>
>>>>> The problem is the call rings through to the HQ DN but comes up fast
>>>>> busy when I answer the call.
>>>>>
>>>>> I've attempted to configure the "Trunk" from UCM as both an H.225 Trunk
>>>>> as well as an ICT trunk.
>>>>> I've unchecked "Wait for far end h.245 terminal capability" as well.
>>>>>
>>>>> Any help is greatly appreciated.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>
>>>
>>
>>
>> --
>> "it's not the years in your life that count. It's the life in your years."
>> Abraham Lincoln
>>
>
>
>
> --
> "it's not the years in your life that count. It's the life in your years."
> Abraham Lincoln
>

Re: [OSL | CCIE_Voice] Gatekeeper blues :|

[CCIELabRat]

Yep, everything started working once I checked off requires MTP on the
trunk. .

But why? The call is coming in as g.729 and the device pool the trunk is in
is set to g.729





On Mon, Mar 23, 2009 at 11:54 AM, Chris Parker  wrote:

> If you are getting ring through when you place the call, then your GK
> config is probably OK. If its ringing that tells me the GK sent the call to
> the UCM.
>
> The fact that it goes fast busy when you pick up sounds like a codec/media
> problem to me. Check the media resources config and your regions.
>
> Chris
>
> ccielab...@gmail.com wrote:
>
>> I'm working through scenario , trying to get 4 digit dial from BR2 to HQ
>> working via the HQ GK.
>> The GK config looks like this.
>>
>> gatekeeper
>>  zone local UCM ipexpert.com  142.1.1.1
>>  zone prefix UCM 1... gw-priority 10 Trunk_2
>>  zone prefix UCM 1... gw-priority 9 Trunk_1
>>  zone prefix UCM 1... gw-priority 0 cmegw
>>  zone prefix UCM 3... gw-priority 10 cmegw
>>  zone prefix UCM 3... gw-priority 0 Trunk_2 Trunk_1
>>  gw-type-prefix 1#* gw ipaddr 142.2.64.12 1720 gw ipaddr 142.2.64.11 1720
>>  no shutdown
>>
>> I have the UCM registering without a tech-prefix defined and I've adjusted
>> the "service parameter" to insure the UCM listens on port 1720.
>>
>> The problem is the call rings through to the HQ DN but comes up fast busy
>> when I answer the call.
>>
>> I've attempted to configure the "Trunk" from UCM as both an H.225 Trunk as
>> well as an ICT trunk.
>> I've unchecked "Wait for far end h.245 terminal capability" as well.
>>
>> Any help is greatly appreciated.
>>
>>
>>
>>
>

[OSL | CCIE_Voice] Gatekeeper blues :|

[CCIELabRat]

I'm working through scenario , trying to get 4 digit dial from BR2 to HQ
working via the HQ GK.The GK config looks like this.

gatekeeper
 zone local UCM ipexpert.com 142.1.1.1
 zone prefix UCM 1... gw-priority 10 Trunk_2
 zone prefix UCM 1... gw-priority 9 Trunk_1
 zone prefix UCM 1... gw-priority 0 cmegw
 zone prefix UCM 3... gw-priority 10 cmegw
 zone prefix UCM 3... gw-priority 0 Trunk_2 Trunk_1
 gw-type-prefix 1#* gw ipaddr 142.2.64.12 1720 gw ipaddr 142.2.64.11 1720
 no shutdown

I have the UCM registering without a tech-prefix defined and I've adjusted
the "service parameter" to insure the UCM listens on port 1720.

The problem is the call rings through to the HQ DN but comes up fast busy
when I answer the call.

I've attempted to configure the "Trunk" from UCM as both an H.225 Trunk as
well as an ICT trunk.
I've unchecked "Wait for far end h.245 terminal capability" as well.

Any help is greatly appreciated.

Re: [OSL | CCIE_Voice] VMware ESXi for CCM servers

[CCIELabRat]

Any tricks to getting CUCM 7 working on ESXi ?My install keeps failing .
I haven't invested too much time yet, as I'm trying hard to pass with the
current blueprint :)



On Sat, Mar 21, 2009 at 1:15 PM, Arun Kumar  wrote:

> Hi
>
> I'm running CUCM 7 and Unity Connection 7 on ESXi with 6GB of RAM and 500GB
> of HDD and it's working fine. Not tested on Linux.
>
> Thanks
>
>
> On Sat, Mar 21, 2009 at 6:44 PM, WorkerBee  wrote:
>
>> Anyone has tried using ESXi with Quad core/8G ram instead of using
>> VMware server on a Linux?
>>
>> Does ESXi gives a better performance?
>>
>> Thanks.
>>
>
>

[OSL | CCIE_Voice] CME Gatekeeper Registration

[CCIELabRat]

Can someone confirm for me what the requirements are to have ephone-dn's in
CME register to a gatekeeper.
Is it dependent on using a dialplan command ?

I keep getting inconsistent results. (I think)

I also have run into the problem where I use No-Reg on the number
configuration and it still ends up listed as an e.614 on the GK.
Multiple reboots don't seem to clear up the problem either.

This is driving me nuts.

Re: [OSL | CCIE_Voice] Lab attack order.

[CCIELabRat]

Chris ,
Great layout and exactly what I was looking for.

Thanks


On Tue, Mar 3, 2009 at 10:27 PM, Chris Parker  wrote:

> I think everyone has their own way of doing it, but here's how I go about
> it:
>
> STEP ZERO - read the whole lab carefully. It's hard to do when you are
> nervous. Try and pay attention the the details and look for tricks. Then
> work things out like what partitions and css you'll need, how the gatekeeper
> will work. Do you need multicast. Try and get you head around the lab as
> much as possible.
>
> 1. Gather info
>
> This is where I log into everything and look at CDP to get the MAC
> addresses of the phones and of the 6608 devices. I put all of this info into
> notepad for easy cut and paste. I hardly use any paper in the lab everything
> goes into notepad for easy access and no retyping.
>
> 2. 6500
>
> Next I set up the 6500. I do all the vlans, aux vlans, voice ports,
> muulticast, and any QoS
>
> 3. HQ
>
> Here I set up interfaces, NTP, timezone, DHCP if need be, multicast if
> needed, QoS and the basic Gatekeeper config. I like doing all the QoS at the
> beginning of the lab, and by doing it on HQ first it helps with time because
> you can just cut and paste what you do on HQ to BR1 and BR2. I set up the GK
> enough so that I know the CME and UCM will register in the correct zones.
> Also if I see I need VIA-GK I go ahead and set up IPIPGW, dial peers and
> transcoder (as needed) on HQ. I go back and tweak the GK config later after
> I have set up UCM and CME.
>
> 4. BR1
>
> I paste in QoS config copied from HQ, create the vlans, create interfaces,
> set helper address and multicast if needed, set up switch ports, set up dhcp
> if needed, set up ntp and timezone, set isdn switchtype, configure PRI for
> MGCP or H323, tweak isdn settings on serial, set up mgcp if needed, set up
> translation rules, set up CoR if needed, configure dial peers if needed for
> H323 and/or SRST, config transcoder and conf bridge if needed, set up SRST.
>
> 5. BR2
>
> I paste in QoS config copied from HQ, create interfaces, set helper address
> and multicast if needed, set up dhcp if needed, set up ntp and timezone, set
> isdn switchtype, configure PRI, tweak isdn settings on serial, set voice
> service to allow h323 to sip, set up translation rules, set up CoR if
> needed, configure dial peers (these can often be copied with minor tweaks
> from BR1), config transcoder and conf bridge if needed (another section
> copied fro BR1), run telephony service setup, paste mac address from step 1
> to appropriate ephone, set up H323 gateway and confirm registered to HQ, set
> voip dial peers for HQ GK and CUE, add ephone-dns for MWI, set up BACD, add
> any hunt groups as needed.
>
> 6. CUE
>
> I normally do this step while doing something else at the same time like
> setting up the 3550. There's alot of waiting in this step while CUE just
> "does its thing". Once CUE is up and ready to be configured, I normally do
> the basic setup with the GUI (this will set up mwi, and your sip triggers
> and applications) and then do user creation from the CUE CLI.
>
> 7. 3550
>
> Here I do vlans and set up ports, and do any QoS.
>
> 8. CME Testing
>
> So at this point 90% of the CLI work is done and all the CME/CUE stuff is
> complete. So here I do some quick testing on CME dialing in / out to PSTN,
> CUE MWI, and BACD.
>
> 9. Callmanager Basic
>
> So here Im doing all the CM stuff I can think of except call routing and
> phones. I go into serviceability first and turn on whatever services I need.
> Then I go back to admin. I use the Top Left to Right method. So System menu
> first and usually touch everything except Device Defaults. Then in Route
> Plan I do AAR, Partitions, CSS, and I try to do any  translation patterns
> that I think I'll need. Then I jump over to the Device menu. I do any custom
> soft keys or button templates. I set up EM profiles if I need them. I set up
> all my gateways and make sure they register, I set up the GK and trunk I set
> up any CTI route points I need. I DO NOT set up the phones yet however. Then
> I move to the Feature Menu. I setup my phone services, voice mail, and park
> or pickup if need be. Then I go to the services menu. I setup all the media
> resources, lists and groups and make sure everything is registered and in
> the right device pool. After the media stuff is done I go back to the device
> pools and assign the MRGL. I always do IPMA or AC later after the phones are
> set up.
>
> 10. Eat Lunch
>
> Yep all the stuff listed above needs to be done before lunch.
>
> 11. Callmanager routing and Phones
>
> So this part is all about the Route Groups, Route Lists, and Route
> Patterns. I plan most of this out in step zero or at lunch. Once thats all
> done I start setting up the phones and users. If I did everything right in
> step 9 and I can do everything I need to on every phones without going back
> and add partitions or what not. When this is done I do some quick basic
> t

[OSL | CCIE_Voice] Lab attack order.

[CCIELabRat]

I'm curious how people have approached the order of completing a lab.
What order do you use to ensure gathering the most points and build the lab
correctly without doubling back to adjust things as you get further into the
lab.

I know Vik and Mark have their own opinions on this, but I wanted to throw
it out to the more general audience for feedback.

[OSL | CCIE_Voice] Parked Call problem

[CCIELabRat]

I'm working with call park.
If I place an internal call (phone to phone) , I can park the call and
pickup it up on another phone by dialing the call park DN.
If I place a call from PSTN into HQ via the 6608 , I can park the call and
pickup it up on another phone by dialing the call park DN.
If I place a call from PSTN into HQ via the br1-rtr (MGCP) , I can park the
call but get a message that "your call can not be completed" when I dial the
call park DN.

Any ideas?

Re: [OSL | CCIE_Voice] Antw: Re: Working on Unity Hunt group scenario.

[CCIELabRat]

That would make sense but it doesn't seem to work that way.
I changed the CSS assigned to the VM PILOT to a CSS without any access to
the internal partitions.
All the VM ports were in an "internal" partition.
When I dial the VM pilot, it shouldn't work based on my changes, but it
does.

- Scott



On Wed, Feb 18, 2009 at 1:36 PM, Robert Schuknecht wrote:

> Scott,
>
> as far as i know, the CSS of the VM-Pilot is used to reach the Hunt-Pilot
> of the Hunt-List/Line-Group of the VM-Ports.
> See the Voicemail-Pilot as a Speeddial to the Hunt-Pilot.
>
>
> /Robert
>
> >>>  schrieb am Mittwoch, 18. Februar 2009 um 17:59
> in
> Nachricht 884fa9c92a4d8b0e56bd24f2851093dc:
> > Agreed, the CSS that is assigned to the VM port is used to determine what
> > dial patterns are available.
> > My question is the CSS that is assigned to the actual VM PILOT DN.
> > As I see it, the PILOT DN wouldn't be used for outbound calling.
> >
> > - Scott
> >
> >
> > On Wed, Feb 18, 2009 at 11:55 AM, Cliff McGlamry 
> wrote:
> >
> >>  You're making an outbound call from the port when you transfer to that
> >> last number.  That uses the CSS to figure out how to route the call.
> >>
> >> - Original Message -
> >> *From:* ccielab...@gmail.com
> >> *To:* Jose Gregorio Linero (jlinero) 
> >> *Cc:* OSL Group 
> >> *Sent:* Wednesday, February 18, 2009 10:54 AM
> >> *Subject:* Re: [OSL | CCIE_Voice] Working on Unity Hunt group scenario.
> >>
> >> Jose,
> >> The CSS was set correctly, but I never "reset" the ports :)
> >> It's all working now.
> >>
> >> BTW:  On a semi-related note, What is the significance of the CSS
> assigned
> >> to the VM Pilot number?
> >>   I can't see how that DN would ever be leveraged for outbound
> >> service.
> >>
> >> - Scott
> >>
> >>
> >>  On Wed, Feb 18, 2009 at 10:46 AM, Jose Gregorio Linero (jlinero) <
> >> jlin...@cisco.com> wrote:
> >>
> >>>  Hi:
> >>>
> >>> Take a look at the CSS configuration in Unity ports.
> >>>
> >>> Regards,
> >>>
> >>> Jose
> >>>
> >>>  --
> >>> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> >>> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *
> >>> ccielab...@gmail.com
> >>> *Sent:* Miércoles, Febrero 18, 2009 10:45 AM
> >>> *To:* OSL Group
> >>> *Subject:* [OSL | CCIE_Voice] Working on Unity Hunt group scenario.
> >>>
> >>>   I'm trying to get a "Poor mans" hunt group working by using Unity
> Call
> >>> Handlers to ring a couple of extensions and then ultimately forward the
> call
> >>> out to a PSTN phone. I've gotten everything working but the last
> forward
> >>> out to a PSTN number (91408xxx)
> >>>
> >>> I've checked and changed the default restriction table in Unity , so I
> >>> think the call should be allowed.
> >>> Where can I look to troubleshoot this? The call viewer application
> doesn't
> >>> show any good information.
> >>>
> >>> Thanks
> >>> Scott
> >>>
> >>>
> >>
>

Re: [OSL | CCIE_Voice] Working on Unity Hunt group scenario.

[CCIELabRat]

Agreed, the CSS that is assigned to the VM port is used to determine what
dial patterns are available.
My question is the CSS that is assigned to the actual VM PILOT DN.
As I see it, the PILOT DN wouldn't be used for outbound calling.

- Scott


On Wed, Feb 18, 2009 at 11:55 AM, Cliff McGlamry  wrote:

>  You're making an outbound call from the port when you transfer to that
> last number.  That uses the CSS to figure out how to route the call.
>
> - Original Message -
> *From:* ccielab...@gmail.com
> *To:* Jose Gregorio Linero (jlinero) 
> *Cc:* OSL Group 
> *Sent:* Wednesday, February 18, 2009 10:54 AM
> *Subject:* Re: [OSL | CCIE_Voice] Working on Unity Hunt group scenario.
>
> Jose,
> The CSS was set correctly, but I never "reset" the ports :)
> It's all working now.
>
> BTW:  On a semi-related note, What is the significance of the CSS assigned
> to the VM Pilot number?
>   I can't see how that DN would ever be leveraged for outbound
> service.
>
> - Scott
>
>
>  On Wed, Feb 18, 2009 at 10:46 AM, Jose Gregorio Linero (jlinero) <
> jlin...@cisco.com> wrote:
>
>>  Hi:
>>
>> Take a look at the CSS configuration in Unity ports.
>>
>> Regards,
>>
>> Jose
>>
>>  --
>> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
>> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *
>> ccielab...@gmail.com
>> *Sent:* Miércoles, Febrero 18, 2009 10:45 AM
>> *To:* OSL Group
>> *Subject:* [OSL | CCIE_Voice] Working on Unity Hunt group scenario.
>>
>>   I'm trying to get a "Poor mans" hunt group working by using Unity Call
>> Handlers to ring a couple of extensions and then ultimately forward the call
>> out to a PSTN phone. I've gotten everything working but the last forward
>> out to a PSTN number (91408xxx)
>>
>> I've checked and changed the default restriction table in Unity , so I
>> think the call should be allowed.
>> Where can I look to troubleshoot this? The call viewer application doesn't
>> show any good information.
>>
>> Thanks
>> Scott
>>
>>
>

Re: [OSL | CCIE_Voice] Working on Unity Hunt group scenario.

[CCIELabRat]

Jose,
The CSS was set correctly, but I never "reset" the ports :)
It's all working now.

BTW:  On a semi-related note, What is the significance of the CSS assigned
to the VM Pilot number?
  I can't see how that DN would ever be leveraged for outbound
service.

- Scott


On Wed, Feb 18, 2009 at 10:46 AM, Jose Gregorio Linero (jlinero) <
jlin...@cisco.com> wrote:

>  Hi:
>
> Take a look at the CSS configuration in Unity ports.
>
> Regards,
>
> Jose
>
>  --
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *
> ccielab...@gmail.com
> *Sent:* Miércoles, Febrero 18, 2009 10:45 AM
> *To:* OSL Group
> *Subject:* [OSL | CCIE_Voice] Working on Unity Hunt group scenario.
>
> I'm trying to get a "Poor mans" hunt group working by using Unity Call
> Handlers to ring a couple of extensions and then ultimately forward the call
> out to a PSTN phone. I've gotten everything working but the last forward
> out to a PSTN number (91408xxx)
>
> I've checked and changed the default restriction table in Unity , so I
> think the call should be allowed.
> Where can I look to troubleshoot this? The call viewer application doesn't
> show any good information.
>
> Thanks
> Scott
>
>

[OSL | CCIE_Voice] Working on Unity Hunt group scenario.

[CCIELabRat]

I'm trying to get a "Poor mans" hunt group working by using Unity Call
Handlers to ring a couple of extensions and then ultimately forward the call
out to a PSTN phone.I've gotten everything working but the last forward out
to a PSTN number (91408xxx)

I've checked and changed the default restriction table in Unity , so I think
the call should be allowed.
Where can I look to troubleshoot this? The call viewer application doesn't
show any good information.

Thanks
Scott

Re: [OSL | CCIE_Voice] Device Pool Question

[CCIELabRat]

Thanks for the reply Mark.I'll give it a try.


On Fri, Jan 9, 2009 at 1:16 PM, Mark Snow  wrote:

> Well with all as you said, it 'should' have worked fine. That being said,
> maybe at one time there was another DP assiged or a different R within one
> of the DPs. Sometimes the UCM 4 DB had to be 'bumped' to get it to reconize
> some changes and possibly your change did just that. If all is as you say
> then I would imagine that changing it back would also result in a completed
> call.
>
> HTH,
>
> Mark Snow
> Sr Technical Instructor
> IPexpert, Inc.
>
> Sent from my iPhone
>
>
> On Jan 9, 2009, at 11:52 AM, ccielab...@gmail.com wrote:
>
>  I had been struggling with getting the IPIPGW scenario working (Task 4.9)
>> in the IPEXPERT workbook.
>> The specific problem was making an H.323 call from CME and have it
>> delivered via SIP Trunk to CM.
>>
>> I found , although I had an MRGL assigned to the trunk, the MTP within the
>> MRGL wasn't in the same Device Pool as the trunk. Once I assigned the MTP to
>> the same Device Pool as the Trunk (DP_711only) , the call completed.
>>
>> This doesn't make sense to me.
>> The MTP was in a device pool called "Default" which contained a region
>> called "default"
>> The Trunk was in a device pool called "DP_711only" which contained a
>> region called "g711"
>>
>> The regions were configured to use G.711 between each other.
>>
>> What am I not understanding here?  What would require the MTP to be in the
>> same Device pool?
>>
>> - Scott
>>
>>
>>
>>

[OSL | CCIE_Voice] Device Pool Question

[CCIELabRat]

I had been struggling with getting the IPIPGW scenario working (Task 4.9) in
the IPEXPERT workbook.
The specific problem was making an H.323 call from CME and have it delivered
via SIP Trunk to CM.

I found , although I had an MRGL assigned to the trunk, the MTP within the
MRGL wasn't in the same Device Pool as the trunk. Once I assigned the MTP to
the same Device Pool as the Trunk (DP_711only) , the call completed.

This doesn't make sense to me.
The MTP was in a device pool called "Default" which contained a region
called "default"
The Trunk was in a device pool called "DP_711only" which contained a region
called "g711"

The regions were configured to use G.711 between each other.

What am I not understanding here?  What would require the MTP to be in the
same Device pool?

- Scott

[OSL | CCIE_Voice] Device Pool Question

[CCIELabRat]

I had been struggling with getting the IPIPGW scenario working (Task 4.9) in
the IPEXPERT workbook.
The specific problem was making an H.323 call from CME and have it delivered
via SIP Trunk to CM.

I found , although I had an MRGL assigned to the trunk, the MTP within the
MRGL wasn't in the same Device Pool as the trunk. Once I assigned the MTP to
the same Device Pool as the Trunk (DP_711only) , the call completed.

This doesn't make sense to me.
The MTP was in a device pool called "Default" which contained a region
called "default"
The Trunk was in a device pool called "DP_711only" which contained a region
called "g711"

The regions were configured to use G.711 between each other.

What am I not understanding here?  What would require the MTP to be in the
same Device pool?

- Scott

[OSL | CCIE_Voice] Correction: Transcoding Question.

[CCIELabRat]

After some review, I see that the MTP being configured is a software MTP .
It is being assigned to the device pool that will be used for the ICT trunk
going to the IPIPGw.

-- Forwarded message --
From: 
Date: Sat, Jan 3, 2009 at 11:17 AM
Subject: Transcoding Question.
To: OSL Group 


I need to get a better understanding of how transcoding is invoked.
I've setup an IPIPGW on the HQ Router.
I'm trying to setup H.323(g711) -> SIP (g729) calls to/from Hq/SiteC
I've configured a transcoder and registered it to the telephony service on
the HQ RT.

I've noticed in the IPExpert proctor guide that the HQ router is registered
as a  HW transcoder in the UCM.
Is that needed?

I assumed the a g711 from UCM would come into the IPIPGW and the IPIPGW
would invoke a transcoder.
I didn't think the UCM needed anything for this.

Can someone help me out?

- Scott

[OSL | CCIE_Voice] Transcoding Question.

[CCIELabRat]

I need to get a better understanding of how transcoding is invoked.
I've setup an IPIPGW on the HQ Router.
I'm trying to setup H.323(g711) -> SIP (g729) calls to/from Hq/SiteC
I've configured a transcoder and registered it to the telephony service on
the HQ RT.

I've noticed in the IPExpert proctor guide that the HQ router is registered
as a  HW transcoder in the UCM.
Is that needed?

I assumed the a g711 from UCM would come into the IPIPGW and the IPIPGW
would invoke a transcoder.
I didn't think the UCM needed anything for this.

Can someone help me out?

- Scott