Re: [OSL | CCIE_Voice] CUC VM Mask

[datucha123 datucha123]

I also guess so,

But I have created the Dummy DN in CUCM with the same Extension number with
VM profile. But it did not work still.

I will try the SRST and see if it works

On Thu, Nov 24, 2011 at 2:09 AM, Chris Martin  wrote:

> Since these are registered with CME and not associated phones with CUCM in
> any way, I am not sure the voice mail profile is invoked since that is tied
> to a line.  I don't think I have tested CME phones locally registered
> redirecting into CUCM/CUC without SRST involved.  One option would be a
> voice-translation rule that strips the RDNIS to 4 digits, then associate
> this to a dial-peer going to voicemail.
>
> Chris
>
>
> On Wed, Nov 23, 2011 at 3:14 PM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>> Yes, sure, it is enabled on the Incoming Gateway.
>>
>> I have also noticed that the Voice Mail Box Mask is working only for the
>> CUCM Registered IP Phones. It transforms the Redirecting and Calling Number
>> as necessary. But when the IP Phone, that is not registered with CUCM
>> (CUCME IP Phones) is redirecting the call to CUC, that Transformation does
>> not work :(
>>
>>   On Thu, Nov 24, 2011 at 1:06 AM, Chris Martin wrote:
>>
>>> At a glance your config seems fine and it should work as you intend.  I
>>> assume you also allowed incoming redirecting number for the CUCM gateway?
>>>
>>> Chris
>>>
>>>   On Wed, Nov 23, 2011 at 11:38 AM, datucha123 datucha123 <
>>> datucha...@gmail.com> wrote:
>>>
   Hello,

 I have configured the Voice Mail Box Mask for  (As I know, this
 will transform the Redricting Number to last 4 digits).

 but somehow it does not work :(

 So here what is happening:

 I have configure the User and VM Box in CUC for CUCME IP Phone, (ext
 3012). I have configure the Call Forward All on CUCME IP Phone to CUC VM
 Pilot through the PSTN (0001911444888)

 Also configured the Voice Mail Box Mask for VM Porfile to .

 CUCME IP Phone has also the DialPlan pattern assigned, so that the
 Caller ID is 2553012.

 CUCME IP Phone is calling CUC through the PRI PSTN. (I enabled the
 Redirecting IE on PR interfaces).

 Also I have enabled the "call-forward system redirect" in
 Telephony-service, so that the redirecting number will be expanded based on
 the DialPlan pattern.

 So now when the some Phone calls this CUCME IP Phones (That has CFWDALL
 to CUC PILOT NUMBER Through PSTN) the caller hears the AA Greeting instead
 of Voice Mail leaving greeting.

 So as I guess the Voice Mail Box Mask transformation is not working.

 Maybe I am missing something?





 ___
 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] Unity VM Box Creation

[datucha123 datucha123]

I have tryed that but still not success :(

There is still the same error



On Thu, Nov 24, 2011 at 2:03 AM, Chris Martin  wrote:

> Try removing "CCMSysUser" and only have your axl admin user in there.  If
> that doesn't work restart your axl web servers on both devices.  I just had
> time to run through it and worked fine for me.  When I had both users I got
> the same error you are.
>
> Chris
>
>
> On Wed, Nov 23, 2011 at 3:15 PM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>> Yes, I have that user "admin" (Just in my case it is called
>> "CCMAdministrator"), and this user is also configured in CUC Phone System.
>> But still no success :(
>>
>>
>> On Thu, Nov 24, 2011 at 1:08 AM, Chris Martin  wrote:
>>
>>> Been a while since I tested pushing unity users through CUCM, but I
>>> thought you had to set the application user to an axl enabled user that is
>>> also set in unity connection phone system.  IE: admin.
>>>
>>> I may be wrong.. Has been a while since I tried that and I don't have
>>> access to my lab right now.
>>>
>>> Chris
>>>
>>>   On Wed, Nov 23, 2011 at 11:20 AM, datucha123 datucha123 <
>>> datucha...@gmail.com> wrote:
>>>
   I am trying to create the Voice Mail Box for the User from the CUCM
 directly, meaning that I am going into Line configuration and then
 selecting the "Create Voice Mail Box" from the upper right corner.

 But when I choose the User Template (it is visible in CUCM when adding
 the VM Box) and click SAVE, it gaves me an error:"   *Unmapped
 Exception (401)Unauthorized*"

 Thus I have changed the Name in Application Server to IP address, and
 assigned the CUCM Administrator and CCMSysUser as the Selected Application
 Users for CUC Server.

 Also has unset the "Voice Mail Box Mask" in the VM Profile (as the CUCM
 7.0 has BUG related to that).

 But still I got that error when tryin to add a VM Box from CUCM Device
 Line configuration.



 ___
 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] Dialed Display Number.

[datucha123 datucha123]

What do you mean under the No Supplementary service?

So you mean *no supplementary-service h225-notify cid-update*?

On Thu, Nov 24, 2011 at 4:00 AM, Gurpreet Singh Kukreja <
tycoononway1...@gmail.com> wrote:

> Hi,
>
> This depends on what is asked in the Lab. If the lab tells you to display
> 10 digit number then you do it otherwise you don't.
>
> Logically, a user when dials a 7 digit number has no idea where the call
> is going from. For him, he is dialing a 7 digit #. So ideally, it should
> always display 7 digit called # in this situation.
>
> If nothing is mentioned in lab, i would leave it to 7 digit called #.
>
> HTH
>
>
> Regards
> Gurpreet
>
>   On Mon, Nov 21, 2011 at 2:00 PM, Pradeep Kumar Sharma <
> sonu.netwo...@gmail.com> wrote:
>
>>   Hello Guys,
>>
>> Do we really have to take care of dialed display number on the phone
>> while call is rerouting to the secondary path in CCIE Lab.
>>
>> For Example:
>>
>> "HQ phone should be able make Local Calls. If HQ gateway is not
>> available, the call should reroute via BR1 Gateway."
>>
>> HQ local number is : 394-2XXX(+1-212-394-2XXX).
>>
>> While rerouting this call via BR1, we have to prefix this number with
>> 1212 in RL/RG . (91212 in case BR1 is H.323)
>>
>> At this point the dialed display number on the phone will automatically
>> change from 393-2XXX to 12123942XXX or something like that.
>>
>> I know we can take care of it using no supplementary command and route
>> pattern.
>>
>> But my question is , do we really have to worry about this ?
>>
>>
>> -SONU-
>>
>> ___
>> 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] Plus Dialing

[datucha123 datucha123]

I have tryed to add te mentioned translation rules, and everything is just
the same way as Ken mentioned.

On Thu, Nov 24, 2011 at 10:20 AM, Ken Wyan  wrote:

> This is a limitation with CME- and the same applies for 79XX phones
> registered to CME too.
>
>
> Vik Malhi – CCIE #13890
> Managing Partner - IPexpert, Inc.
>
>
> Telephone: +1.810.326.1444 ext 420
> Fax: +1.810.454.0130
> Mailto: vma...@ipexpert.com
>
>
>
>
>
>
>
>
> - Hide quoted text -
> On Oct 24, 2011, at 6:23 AM, Ken Wyan wrote:
>
>
> - Hide quoted text -
> I have CIPC registered to CUCME. PSTN gateway is sending full e164 number
> with + sign ( caller-id for incoming calls).
>
> But in my CIPC main display from field doesn't show + sign (but all digits
> in number are shown). In same CIPC bottom of the display (in small size) it
> shows full number with + sign.
>
> If I register CIPC to CUCM , then both numbers show correctly with leading
> + sign.
>
> How can I correct this?
>
>   On Thu, Nov 24, 2011 at 4:14 AM, Emanuel Damasceno <
> aedamasc...@gmail.com> wrote:
>
>> Hmm
>>
>> Have you tried:
>> translation rule XX
>> rule 1 /^1/ /+1/
>>
>> translation-profile 1
>> translate calling 1
>>
>> voice-port x/x/x:x
>> translation-profile incoming 1
>>
>> If you did, and didn't work, I am gonna start trying this out here...
>> Cheers
>> *Emanuel Damasceno*
>>
>>
>>
>>
>>   On Wed, Nov 23, 2011 at 7:12 PM, datucha123 datucha123 <
>> datucha...@gmail.com> wrote:
>>
>>>
>>> Hello,
>>>
>>> How can I force CUCME IP Phones to display the Numbers in
>>> Missed/Received calls directory with + sign?
>>>
>>> When the call comes to CUCME with Caller ID with +, the CUCME IP Phone
>>> dislpays that + sign in lower place of the screen. But in the Directories
>>> (Missed/Received) calls, that number is listed without +.
>>>
>>> So how can I configure the CUCME not to strip the + sign from the
>>> Calling Number, and also display that sign in Directories (Missed/Received)
>>> calls?
>>>
>>> On CUCM, I do not have a problem, the + sign is listed in the
>>> Missed/Received calls, but as for CUCME I cannot make it working.
>>>
>>> ___
>>> 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] User Import to Unity Connection from CallManager

[datucha123 datucha123]

I think you can do both as nothing will be ruined after the synch.

On Thu, Nov 24, 2011 at 10:01 AM, Ken Wyan  wrote:

> Hi Guys,
>
> I have a very basic question with manually created users in CUCM. ( *No*AD or 
> LDAP integration)
>
> When integrating CUCM with CUC in the exam , is it sufficient to import
> users from CUCM to CUC  or  need to Sync users after importing ?
>
> I saw in a CCIE video ,  user import & sync both to be done in order.
>
> As mentioned in cisco docs , sync is required if any later change is done
> to CUCM users.
>
> Do you normally do both user import & user sync both or  do import only?
>
> Thanks
>
> Ken
>
>
>
> ___
> 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] CUC VM Mask

[datucha123 datucha123]

I have tryed the same with SRST, but that Mask in VM Profile did not take
effect :(

Can anybody tell me, when does that Mask is activated?

For CUCM registered IP Phones it is working file, but for Phones in SRST
mode, it does not work any more.

On Thu, Nov 24, 2011 at 2:09 AM, Chris Martin  wrote:

> Since these are registered with CME and not associated phones with CUCM in
> any way, I am not sure the voice mail profile is invoked since that is tied
> to a line.  I don't think I have tested CME phones locally registered
> redirecting into CUCM/CUC without SRST involved.  One option would be a
> voice-translation rule that strips the RDNIS to 4 digits, then associate
> this to a dial-peer going to voicemail.
>
> Chris
>
>
> On Wed, Nov 23, 2011 at 3:14 PM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>> Yes, sure, it is enabled on the Incoming Gateway.
>>
>> I have also noticed that the Voice Mail Box Mask is working only for the
>> CUCM Registered IP Phones. It transforms the Redirecting and Calling Number
>> as necessary. But when the IP Phone, that is not registered with CUCM
>> (CUCME IP Phones) is redirecting the call to CUC, that Transformation does
>> not work :(
>>
>>   On Thu, Nov 24, 2011 at 1:06 AM, Chris Martin wrote:
>>
>>> At a glance your config seems fine and it should work as you intend.  I
>>> assume you also allowed incoming redirecting number for the CUCM gateway?
>>>
>>> Chris
>>>
>>>   On Wed, Nov 23, 2011 at 11:38 AM, datucha123 datucha123 <
>>> datucha...@gmail.com> wrote:
>>>
   Hello,

 I have configured the Voice Mail Box Mask for  (As I know, this
 will transform the Redricting Number to last 4 digits).

 but somehow it does not work :(

 So here what is happening:

 I have configure the User and VM Box in CUC for CUCME IP Phone, (ext
 3012). I have configure the Call Forward All on CUCME IP Phone to CUC VM
 Pilot through the PSTN (0001911444888)

 Also configured the Voice Mail Box Mask for VM Porfile to .

 CUCME IP Phone has also the DialPlan pattern assigned, so that the
 Caller ID is 2553012.

 CUCME IP Phone is calling CUC through the PRI PSTN. (I enabled the
 Redirecting IE on PR interfaces).

 Also I have enabled the "call-forward system redirect" in
 Telephony-service, so that the redirecting number will be expanded based on
 the DialPlan pattern.

 So now when the some Phone calls this CUCME IP Phones (That has CFWDALL
 to CUC PILOT NUMBER Through PSTN) the caller hears the AA Greeting instead
 of Voice Mail leaving greeting.

 So as I guess the Voice Mail Box Mask transformation is not working.

 Maybe I am missing something?





 ___
 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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www.ipexpert.com

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[OSL | CCIE_Voice] AAR with Route Patterns

[Rynard Coetzee]

Hi guys
I`m trying to get the following scenario to work ,but can`t seem to get the AAR 
to invoke. I have 2 GW`s out to the PSTN ,they are located at HQ site and BR1 
site ,what I have a RPattern that sends all calls from HQ/BR1 phones out of the 
HQ GW. I then have the same RP`s created in a AAR partition ,pointing out the 
BR1 GW ,so when I limit my Location bandwidth to 23k so as to not allow the 
call across the WAN to the remote GW ,it fails the call with "Not Enough 
Bandwidth" message displaying on the phone but it does not invoke AAR ,so I 
never see the "Network Congestion ,Rerouting" message and the call just fails. 
AAR is enabled on the cluster and I can get AAR working between the HQ and BR1 
phones so AAR definitely works. Any ideas ?


Rynard
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[OSL | CCIE_Voice] E1

[muhammad nouman]

Hi 
 
Quick Quesion to connect E1 back to back do I need any specail cable or simple 
crossover cable with RJ 45 connector  is OK.
 
Thanks
 
Nomi___
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Re: [OSL | CCIE_Voice] E1

[datucha123 datucha123]

simple crossover Cable is OK

On Thu, Nov 24, 2011 at 3:07 PM, muhammad nouman wrote:

>  Hi
>
> Quick Quesion to connect E1 back to back do I need any specail cable or
> simple crossover cable with RJ 45 connector  is OK.
>
> Thanks
>
> Nomi
>
> ___
> 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] Lab 6

[Jonathan Bourne]

HI Guys,

I attempted last week @ dubai and to my surprised i got lab 6.

I was not prepared for it.

I was only prepared for lab 3 4 and 5.

I have noted the lab and ready to share.

Please ping me at mail id, if anyone interested.

Rgds,
Jon
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[OSL | CCIE_Voice] Lab 6

[Jonathan Bourne]

HI Guys,

I attempted last week @ dubai and to my surprised i got lab 6.

I was not prepared for it.

I was only prepared for lab 3 4 and 5.

Few of the questions asked

1) I divert
2) 6 Ringtones
3) SIP Early offer

I have noted the lab and ready to share.

Please ping me at mail id, if anyone interested.

Rgds,
Jon
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Re: [OSL | CCIE_Voice] E1

[Emanuel Damasceno]

You need a Crossover cable with the pins 1245 crossed over. Here is a chart
to make it easier :) (I have a BUNCH of those cables =D)

END 1
PIN 1 ORANGE/WHITE
PIN 2 WHITE/ORANGE
PIN 4 BLUE/WHITE
PIN 5 WHITE/BLUE

END 2
PIN 1 BLUE/WHITE
PIN 2 WHITE/BLUE
PIN 4 ORANGE/WHITE
PIN 5 WHITE/ORANGE
*
Emanuel Damasceno*




On Thu, Nov 24, 2011 at 9:07 AM, muhammad nouman wrote:

> Hi
>
> Quick Quesion to connect E1 back to back do I need any specail cable or
> simple crossover cable with RJ 45 connector  is OK.
>
> Thanks
>
> Nomi
>
> ___
> 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] Dialed Display Number.

[datucha123 datucha123]

Can we also use the CUCM Service Parameter  - "Always display dialed
digits" or not?

On Thu, Nov 24, 2011 at 4:39 PM, Gurpreet Singh Kukreja <
tycoononway1...@gmail.com> wrote:

> Yes,
>
> I think he means to enter these commands on the gateway to disable
> sending of H.225 messages with caller-ID updates. You can enter the
> command on both, under the dial-peer & under the global config mode. I do
> this under the global config mode;
>
> *voice service voip*
>   *no supplementary-service h225-notify cid-update*
>
>
> Regards
> Gurpreet
>
>
> On Thu, Nov 24, 2011 at 3:53 AM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>> What do you mean under the No Supplementary service?
>>
>> So you mean *no supplementary-service h225-notify cid-update*?
>>
>>  On Thu, Nov 24, 2011 at 4:00 AM, Gurpreet Singh Kukreja <
>> tycoononway1...@gmail.com> wrote:
>>
>>> Hi,
>>>
>>> This depends on what is asked in the Lab. If the lab tells you to
>>> display 10 digit number then you do it otherwise you don't.
>>>
>>> Logically, a user when dials a 7 digit number has no idea where the call
>>> is going from. For him, he is dialing a 7 digit #. So ideally, it should
>>> always display 7 digit called # in this situation.
>>>
>>> If nothing is mentioned in lab, i would leave it to 7 digit called #.
>>>
>>> HTH
>>>
>>>
>>> Regards
>>> Gurpreet
>>>
>>>   On Mon, Nov 21, 2011 at 2:00 PM, Pradeep Kumar Sharma <
>>> sonu.netwo...@gmail.com> wrote:
>>>
   Hello Guys,

 Do we really have to take care of dialed display number on the phone
 while call is rerouting to the secondary path in CCIE Lab.

 For Example:

 "HQ phone should be able make Local Calls. If HQ gateway is not
 available, the call should reroute via BR1 Gateway."

 HQ local number is : 394-2XXX(+1-212-394-2XXX).


 While rerouting this call via BR1, we have to prefix this number with
 1212 in RL/RG . (91212 in case BR1 is H.323)

 At this point the dialed display number on the phone will automatically
 change from 393-2XXX to 12123942XXX or something like that.

 I know we can take care of it using no supplementary command and route
 pattern.

 But my question is , do we really have to worry about this ?


 -SONU-

 ___
 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] Dialed Display Number.

[Pradeep Kumar Sharma]

Correct, i also apply this command globally.

Well the point is, Its very easy to configure TEHO with Called Party
Transformation Pattern.
But If proctor is expecting to maintain the Called Display number in case
of TEHO,i am not suppose to use Called Party Transformation Pattern.
because it'll take preference over "*no supplementary-service h225-notify
cid-update*" command or Route Pattern Digit manipulation and change the
display number number on the calling phone.

So if nothing is mentioned in the lab about display number , we have to
take our own decision whether we want to maintain the display number or not.

What you guys will recommend, should it go for CPTP Digit manipulation dial
plan or RL/RG Digit manipulation Dial plan ?



On Thu, Nov 24, 2011 at 6:16 PM, datucha123 datucha123  wrote:

>
> Can we also use the CUCM Service Parameter  - "Always display dialed
> digits" or not?
>
> On Thu, Nov 24, 2011 at 4:39 PM, Gurpreet Singh Kukreja <
> tycoononway1...@gmail.com> wrote:
>
>> Yes,
>>
>> I think he means to enter these commands on the gateway to disable
>> sending of H.225 messages with caller-ID updates. You can enter the
>> command on both, under the dial-peer & under the global config mode. I do
>> this under the global config mode;
>>
>> *voice service voip*
>>   *no supplementary-service h225-notify cid-update*
>>
>>
>> Regards
>> Gurpreet
>>
>>
>> On Thu, Nov 24, 2011 at 3:53 AM, datucha123 datucha123 <
>> datucha...@gmail.com> wrote:
>>
>>> What do you mean under the No Supplementary service?
>>>
>>> So you mean *no supplementary-service h225-notify cid-update*?
>>>
>>>  On Thu, Nov 24, 2011 at 4:00 AM, Gurpreet Singh Kukreja <
>>> tycoononway1...@gmail.com> wrote:
>>>
 Hi,

 This depends on what is asked in the Lab. If the lab tells you to
 display 10 digit number then you do it otherwise you don't.

 Logically, a user when dials a 7 digit number has no idea where the
 call is going from. For him, he is dialing a 7 digit #. So ideally, it
 should always display 7 digit called # in this situation.

 If nothing is mentioned in lab, i would leave it to 7 digit called #.

 HTH


 Regards
 Gurpreet

  On Mon, Nov 21, 2011 at 2:00 PM, Pradeep Kumar Sharma <
 sonu.netwo...@gmail.com> wrote:

>  Hello Guys,
>
> Do we really have to take care of dialed display number on the phone
> while call is rerouting to the secondary path in CCIE Lab.
>
> For Example:
>
> "HQ phone should be able make Local Calls. If HQ gateway is not
> available, the call should reroute via BR1 Gateway."
>
> HQ local number is : 394-2XXX
> (+1-212-394-2XXX).
>
> While rerouting this call via BR1, we have to prefix this number with
> 1212 in RL/RG . (91212 in case BR1 is H.323)
>
> At this point the dialed display number on the phone will
> automatically change from 393-2XXX to 12123942XXX or something like that.
>
> I know we can take care of it using no supplementary command and route
> pattern.
>
> But my question is , do we really have to worry about this ?
>
>
> -SONU-
>
> ___
> 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

Re: [OSL | CCIE_Voice] AAR with Route Patterns

[Rrcrumm]

Hi
Check if the phone devices  and have an aar CSS, aar group is on the line . Add 
both to your gateways and route list can use local route group, make sure aar 
rp is sending the correct number of digits to PSTN 

HTH
Randall


Sent from my iPhone

On Nov 24, 2011, at 2:36 AM, Rynard Coetzee  wrote:

> Hi guys
> I`m trying to get the following scenario to work ,but can`t seem to get the 
> AAR to invoke. I have 2 GW`s out to the PSTN ,they are located at HQ site and 
> BR1 site ,what I have a RPattern that sends all calls from HQ/BR1 phones out 
> of the HQ GW. I then have the same RP`s created in a AAR partition ,pointing 
> out the BR1 GW ,so when I limit my Location bandwidth to 23k so as to not 
> allow the call across the WAN to the remote GW ,it fails the call with “Not 
> Enough Bandwidth” message displaying on the phone but it does not invoke AAR 
> ,so I never see the “Network Congestion ,Rerouting” message and the call just 
> fails. AAR is enabled on the cluster and I can get AAR working between the HQ 
> and BR1 phones so AAR definitely works. Any ideas ?
>  
>  
> Rynard
> ___
> 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 
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Re: [OSL | CCIE_Voice] Dialed Display Number.

[Gurpreet Singh Kukreja]

Hey Pradeep,

I've never used CPTP and have scored well in Dial Plan Section. I always
use RP/RL but i also enter "*no supplementary-service h225-notify cid-update
*" on BR1 gateway too. This way, the display on my HQ phones is correct and
the calling party number on the Destination phone too.

Let me know if you have more questions

HTH

Regards
Gurpreet

On Thu, Nov 24, 2011 at 10:32 AM, Pradeep Kumar Sharma <
sonu.netwo...@gmail.com> wrote:

> Correct, i also apply this command globally.
>
> Well the point is, Its very easy to configure TEHO with Called Party
> Transformation Pattern.
> But If proctor is expecting to maintain the Called Display number in case
> of TEHO,i am not suppose to use Called Party Transformation Pattern.
> because it'll take preference over "*no supplementary-service h225-notify
> cid-update*" command or Route Pattern Digit manipulation and change the
> display number number on the calling phone.
>
> So if nothing is mentioned in the lab about display number , we have to
> take our own decision whether we want to maintain the display number or not.
>
> What you guys will recommend, should it go for CPTP Digit manipulation
> dial plan or RL/RG Digit manipulation Dial plan ?
>
>
>
> On Thu, Nov 24, 2011 at 6:16 PM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>>
>> Can we also use the CUCM Service Parameter  - "Always display dialed
>> digits" or not?
>>
>> On Thu, Nov 24, 2011 at 4:39 PM, Gurpreet Singh Kukreja <
>> tycoononway1...@gmail.com> wrote:
>>
>>> Yes,
>>>
>>> I think he means to enter these commands on the gateway to disable
>>> sending of H.225 messages with caller-ID updates. You can enter the
>>> command on both, under the dial-peer & under the global config mode. I do
>>> this under the global config mode;
>>>
>>> *voice service voip*
>>>   *no supplementary-service h225-notify cid-update*
>>>
>>>
>>> Regards
>>> Gurpreet
>>>
>>>
>>> On Thu, Nov 24, 2011 at 3:53 AM, datucha123 datucha123 <
>>> datucha...@gmail.com> wrote:
>>>
 What do you mean under the No Supplementary service?

 So you mean *no supplementary-service h225-notify cid-update*?

  On Thu, Nov 24, 2011 at 4:00 AM, Gurpreet Singh Kukreja <
 tycoononway1...@gmail.com> wrote:

> Hi,
>
> This depends on what is asked in the Lab. If the lab tells you to
> display 10 digit number then you do it otherwise you don't.
>
> Logically, a user when dials a 7 digit number has no idea where the
> call is going from. For him, he is dialing a 7 digit #. So ideally, it
> should always display 7 digit called # in this situation.
>
> If nothing is mentioned in lab, i would leave it to 7 digit called #.
>
> HTH
>
>
> Regards
> Gurpreet
>
>  On Mon, Nov 21, 2011 at 2:00 PM, Pradeep Kumar Sharma <
> sonu.netwo...@gmail.com> wrote:
>
>>  Hello Guys,
>>
>> Do we really have to take care of dialed display number on the phone
>> while call is rerouting to the secondary path in CCIE Lab.
>>
>> For Example:
>>
>> "HQ phone should be able make Local Calls. If HQ gateway is not
>> available, the call should reroute via BR1 Gateway."
>>
>> HQ local number is : 394-2XXX
>> (+1-212-394-2XXX).
>>
>> While rerouting this call via BR1, we have to prefix this number with
>> 1212 in RL/RG . (91212 in case BR1 is H.323)
>>
>> At this point the dialed display number on the phone will
>> automatically change from 393-2XXX to 12123942XXX or something like that.
>>
>> I know we can take care of it using no supplementary command and
>> route pattern.
>>
>> But my question is , do we really have to worry about this ?
>>
>>
>> -SONU-
>>
>> ___
>> 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

Re: [OSL | CCIE_Voice] User Import to Unity Connection from CallManager

[Gurpreet Singh Kukreja]

Ken,

Usually I make all the necessary changes to Voicemail User Template before
importing the users into Unity CX unless there are any specific changes
asked for a particular user which would need to be carried out separately
for that user after the import.

I never felt the need to run a sync to update anything after the user
import but doing so should not harm anything. Everything works fine for me.
I think it's more related to sync any changes if made to users on CUCM.

HTH

Regards
Gurpreet

On Thu, Nov 24, 2011 at 1:01 AM, Ken Wyan  wrote:

> Hi Guys,
>
> I have a very basic question with manually created users in CUCM. ( *No*AD or 
> LDAP integration)
>
> When integrating CUCM with CUC in the exam , is it sufficient to import
> users from CUCM to CUC  or  need to Sync users after importing ?
>
> I saw in a CCIE video ,  user import & sync both to be done in order.
>
> As mentioned in cisco docs , sync is required if any later change is done
> to CUCM users.
>
> Do you normally do both user import & user sync both or  do import only?
>
> Thanks
>
> Ken
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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

Re: [OSL | CCIE_Voice] Unity VM Box Creation

[Gurpreet Singh Kukreja]

Hi,

Try this:

In Unity:
1) Go to Phone System > Edit > CUCM AXL Servers
2) Enter the Call Manager IP Addresses, then enter the CCM Administrator
(AXL USER) username and password.
3) Re-type the correct password on Edit AXL servers. Click on Test.
4) Now try to create Unity user from CUCM End user page.

AXL user should have Super User rights. Try restarting the Cisco AXL Web
Service in both CUC and CUCM. If the CUCM is integrated with LDAP, try
removing LDAP config from the CUCM and try re-adding it into CUCM again.


HTH


Regards
Gurpreet



On Thu, Nov 24, 2011 at 3:47 AM, datucha123 datucha123  wrote:

> I have tryed that but still not success :(
>
> There is still the same error
>
>
>
> On Thu, Nov 24, 2011 at 2:03 AM, Chris Martin  wrote:
>
>> Try removing "CCMSysUser" and only have your axl admin user in there.  If
>> that doesn't work restart your axl web servers on both devices.  I just had
>> time to run through it and worked fine for me.  When I had both users I got
>> the same error you are.
>>
>> Chris
>>
>>
>> On Wed, Nov 23, 2011 at 3:15 PM, datucha123 datucha123 <
>> datucha...@gmail.com> wrote:
>>
>>> Yes, I have that user "admin" (Just in my case it is called
>>> "CCMAdministrator"), and this user is also configured in CUC Phone System.
>>> But still no success :(
>>>
>>>
>>> On Thu, Nov 24, 2011 at 1:08 AM, Chris Martin wrote:
>>>
 Been a while since I tested pushing unity users through CUCM, but I
 thought you had to set the application user to an axl enabled user that is
 also set in unity connection phone system.  IE: admin.

 I may be wrong.. Has been a while since I tried that and I don't have
 access to my lab right now.

 Chris

   On Wed, Nov 23, 2011 at 11:20 AM, datucha123 datucha123 <
 datucha...@gmail.com> wrote:

>   I am trying to create the Voice Mail Box for the User from the CUCM
> directly, meaning that I am going into Line configuration and then
> selecting the "Create Voice Mail Box" from the upper right corner.
>
> But when I choose the User Template (it is visible in CUCM when adding
> the VM Box) and click SAVE, it gaves me an error:"   *Unmapped
> Exception (401)Unauthorized*"
>
> Thus I have changed the Name in Application Server to IP address, and
> assigned the CUCM Administrator and CCMSysUser as the Selected Application
> Users for CUC Server.
>
> Also has unset the "Voice Mail Box Mask" in the VM Profile (as the
> CUCM 7.0 has BUG related to that).
>
> But still I got that error when tryin to add a VM Box from CUCM Device
> Line configuration.
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training,
> please visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>


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

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

Re: [OSL | CCIE_Voice] Unity VM Box Creation

[Gurpreet Singh Kukreja]

This should also help:

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a0080a98d27.shtml


Regards
Gurpreet


On Thu, Nov 24, 2011 at 11:35 AM, Gurpreet Singh Kukreja <
tycoononway1...@gmail.com> wrote:

> Hi,
>
> Try this:
>
> In Unity:
> 1) Go to Phone System > Edit > CUCM AXL Servers
> 2) Enter the Call Manager IP Addresses, then enter the CCM Administrator
> (AXL USER) username and password.
> 3) Re-type the correct password on Edit AXL servers. Click on Test.
> 4) Now try to create Unity user from CUCM End user page.
>
> AXL user should have Super User rights. Try restarting the Cisco AXL Web
> Service in both CUC and CUCM. If the CUCM is integrated with LDAP, try
> removing LDAP config from the CUCM and try re-adding it into CUCM again.
>
>
> HTH
>
>
> Regards
> Gurpreet
>
>
>
>
> On Thu, Nov 24, 2011 at 3:47 AM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>> I have tryed that but still not success :(
>>
>> There is still the same error
>>
>>
>>
>> On Thu, Nov 24, 2011 at 2:03 AM, Chris Martin  wrote:
>>
>>> Try removing "CCMSysUser" and only have your axl admin user in there.
>>> If that doesn't work restart your axl web servers on both devices.  I just
>>> had time to run through it and worked fine for me.  When I had both users I
>>> got the same error you are.
>>>
>>> Chris
>>>
>>>
>>> On Wed, Nov 23, 2011 at 3:15 PM, datucha123 datucha123 <
>>> datucha...@gmail.com> wrote:
>>>
 Yes, I have that user "admin" (Just in my case it is called
 "CCMAdministrator"), and this user is also configured in CUC Phone System.
 But still no success :(


 On Thu, Nov 24, 2011 at 1:08 AM, Chris Martin wrote:

> Been a while since I tested pushing unity users through CUCM, but I
> thought you had to set the application user to an axl enabled user that is
> also set in unity connection phone system.  IE: admin.
>
> I may be wrong.. Has been a while since I tried that and I don't have
> access to my lab right now.
>
> Chris
>
>   On Wed, Nov 23, 2011 at 11:20 AM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>>   I am trying to create the Voice Mail Box for the User from the
>> CUCM directly, meaning that I am going into Line configuration and then
>> selecting the "Create Voice Mail Box" from the upper right corner.
>>
>> But when I choose the User Template (it is visible in CUCM when
>> adding the VM Box) and click SAVE, it gaves me an error:"   *Unmapped
>> Exception (401)Unauthorized*"
>>
>> Thus I have changed the Name in Application Server to IP address, and
>> assigned the CUCM Administrator and CCMSysUser as the Selected 
>> Application
>> Users for CUC Server.
>>
>> Also has unset the "Voice Mail Box Mask" in the VM Profile (as the
>> CUCM 7.0 has BUG related to that).
>>
>> But still I got that error when tryin to add a VM Box from CUCM
>> Device Line configuration.
>>
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training,
>> please visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com 
>>
>
>

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

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

Re: [OSL | CCIE_Voice] Unity VM Box Creation

[datucha123 datucha123]

Still no success,

I think I have problems with my CUC installation/configuration, so I will
try to revert it back to fresh one. And then will try to do it.

On Thu, Nov 24, 2011 at 8:35 PM, Gurpreet Singh Kukreja <
tycoononway1...@gmail.com> wrote:

> Hi,
>
> Try this:
>
> In Unity:
> 1) Go to Phone System > Edit > CUCM AXL Servers
> 2) Enter the Call Manager IP Addresses, then enter the CCM Administrator
> (AXL USER) username and password.
> 3) Re-type the correct password on Edit AXL servers. Click on Test.
> 4) Now try to create Unity user from CUCM End user page.
>
> AXL user should have Super User rights. Try restarting the Cisco AXL Web
> Service in both CUC and CUCM. If the CUCM is integrated with LDAP, try
> removing LDAP config from the CUCM and try re-adding it into CUCM again.
>
>
> HTH
>
>
> Regards
> Gurpreet
>
>
>
>
> On Thu, Nov 24, 2011 at 3:47 AM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>> I have tryed that but still not success :(
>>
>> There is still the same error
>>
>>
>>
>> On Thu, Nov 24, 2011 at 2:03 AM, Chris Martin  wrote:
>>
>>> Try removing "CCMSysUser" and only have your axl admin user in there.
>>> If that doesn't work restart your axl web servers on both devices.  I just
>>> had time to run through it and worked fine for me.  When I had both users I
>>> got the same error you are.
>>>
>>> Chris
>>>
>>>
>>> On Wed, Nov 23, 2011 at 3:15 PM, datucha123 datucha123 <
>>> datucha...@gmail.com> wrote:
>>>
 Yes, I have that user "admin" (Just in my case it is called
 "CCMAdministrator"), and this user is also configured in CUC Phone System.
 But still no success :(


 On Thu, Nov 24, 2011 at 1:08 AM, Chris Martin wrote:

> Been a while since I tested pushing unity users through CUCM, but I
> thought you had to set the application user to an axl enabled user that is
> also set in unity connection phone system.  IE: admin.
>
> I may be wrong.. Has been a while since I tried that and I don't have
> access to my lab right now.
>
> Chris
>
>   On Wed, Nov 23, 2011 at 11:20 AM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>>   I am trying to create the Voice Mail Box for the User from the
>> CUCM directly, meaning that I am going into Line configuration and then
>> selecting the "Create Voice Mail Box" from the upper right corner.
>>
>> But when I choose the User Template (it is visible in CUCM when
>> adding the VM Box) and click SAVE, it gaves me an error:"   *Unmapped
>> Exception (401)Unauthorized*"
>>
>> Thus I have changed the Name in Application Server to IP address, and
>> assigned the CUCM Administrator and CCMSysUser as the Selected 
>> Application
>> Users for CUC Server.
>>
>> Also has unset the "Voice Mail Box Mask" in the VM Profile (as the
>> CUCM 7.0 has BUG related to that).
>>
>> But still I got that error when tryin to add a VM Box from CUCM
>> Device Line configuration.
>>
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training,
>> please visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com 
>>
>
>

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

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

Re: [OSL | CCIE_Voice] Unity VM Box Creation

[datucha123 datucha123]

Great document,

That doc helped me, and now I am able to add users to CUC from CUCM :)

Thanks a lot for you help

On Thu, Nov 24, 2011 at 8:45 PM, Gurpreet Singh Kukreja <
tycoononway1...@gmail.com> wrote:

> This should also help:
>
>
> http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a0080a98d27.shtml
>
>
> Regards
> Gurpreet
>
>
>
> On Thu, Nov 24, 2011 at 11:35 AM, Gurpreet Singh Kukreja <
> tycoononway1...@gmail.com> wrote:
>
>> Hi,
>>
>> Try this:
>>
>> In Unity:
>> 1) Go to Phone System > Edit > CUCM AXL Servers
>> 2) Enter the Call Manager IP Addresses, then enter the CCM Administrator
>> (AXL USER) username and password.
>> 3) Re-type the correct password on Edit AXL servers. Click on Test.
>> 4) Now try to create Unity user from CUCM End user page.
>>
>> AXL user should have Super User rights. Try restarting the Cisco AXL Web
>> Service in both CUC and CUCM. If the CUCM is integrated with LDAP, try
>> removing LDAP config from the CUCM and try re-adding it into CUCM again.
>>
>>
>> HTH
>>
>>
>> Regards
>> Gurpreet
>>
>>
>>
>>
>> On Thu, Nov 24, 2011 at 3:47 AM, datucha123 datucha123 <
>> datucha...@gmail.com> wrote:
>>
>>> I have tryed that but still not success :(
>>>
>>> There is still the same error
>>>
>>>
>>>
>>> On Thu, Nov 24, 2011 at 2:03 AM, Chris Martin wrote:
>>>
 Try removing "CCMSysUser" and only have your axl admin user in there.
 If that doesn't work restart your axl web servers on both devices.  I just
 had time to run through it and worked fine for me.  When I had both users I
 got the same error you are.

 Chris


 On Wed, Nov 23, 2011 at 3:15 PM, datucha123 datucha123 <
 datucha...@gmail.com> wrote:

> Yes, I have that user "admin" (Just in my case it is called
> "CCMAdministrator"), and this user is also configured in CUC Phone System.
> But still no success :(
>
>
> On Thu, Nov 24, 2011 at 1:08 AM, Chris Martin wrote:
>
>> Been a while since I tested pushing unity users through CUCM, but I
>> thought you had to set the application user to an axl enabled user that 
>> is
>> also set in unity connection phone system.  IE: admin.
>>
>> I may be wrong.. Has been a while since I tried that and I don't have
>> access to my lab right now.
>>
>> Chris
>>
>>   On Wed, Nov 23, 2011 at 11:20 AM, datucha123 datucha123 <
>> datucha...@gmail.com> wrote:
>>
>>>   I am trying to create the Voice Mail Box for the User from the
>>> CUCM directly, meaning that I am going into Line configuration and then
>>> selecting the "Create Voice Mail Box" from the upper right corner.
>>>
>>> But when I choose the User Template (it is visible in CUCM when
>>> adding the VM Box) and click SAVE, it gaves me an error:"   *Unmapped
>>> Exception (401)Unauthorized*"
>>>
>>> Thus I have changed the Name in Application Server to IP address,
>>> and assigned the CUCM Administrator and CCMSysUser as the Selected
>>> Application Users for CUC Server.
>>>
>>> Also has unset the "Voice Mail Box Mask" in the VM Profile (as the
>>> CUCM 7.0 has BUG related to that).
>>>
>>> But still I got that error when tryin to add a VM Box from CUCM
>>> Device Line configuration.
>>>
>>>
>>>
>>> ___
>>> For more information regarding industry leading CCIE Lab training,
>>> please visit www.ipexpert.com
>>>
>>> Are you a CCNP or CCIE and looking for a job? Check out
>>> www.PlatinumPlacement.com 
>>>
>>
>>
>

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

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

Re: [OSL | CCIE_Voice] Unity VM Box Creation

[Gurpreet Singh Kukreja]

Great :)

Glad you did not have to re-image your CUC.


Cheers
Gurpreet

On Thu, Nov 24, 2011 at 11:56 AM, datucha123 datucha123 <
datucha...@gmail.com> wrote:

> Great document,
>
> That doc helped me, and now I am able to add users to CUC from CUCM :)
>
> Thanks a lot for you help
>
> On Thu, Nov 24, 2011 at 8:45 PM, Gurpreet Singh Kukreja <
> tycoononway1...@gmail.com> wrote:
>
>> This should also help:
>>
>>
>> http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a0080a98d27.shtml
>>
>>
>> Regards
>> Gurpreet
>>
>>
>>
>> On Thu, Nov 24, 2011 at 11:35 AM, Gurpreet Singh Kukreja <
>> tycoononway1...@gmail.com> wrote:
>>
>>> Hi,
>>>
>>> Try this:
>>>
>>> In Unity:
>>> 1) Go to Phone System > Edit > CUCM AXL Servers
>>> 2) Enter the Call Manager IP Addresses, then enter the CCM Administrator
>>> (AXL USER) username and password.
>>> 3) Re-type the correct password on Edit AXL servers. Click on Test.
>>> 4) Now try to create Unity user from CUCM End user page.
>>>
>>> AXL user should have Super User rights. Try restarting the Cisco AXL Web
>>> Service in both CUC and CUCM. If the CUCM is integrated with LDAP, try
>>> removing LDAP config from the CUCM and try re-adding it into CUCM again.
>>>
>>>
>>> HTH
>>>
>>>
>>> Regards
>>> Gurpreet
>>>
>>>
>>>
>>>
>>> On Thu, Nov 24, 2011 at 3:47 AM, datucha123 datucha123 <
>>> datucha...@gmail.com> wrote:
>>>
 I have tryed that but still not success :(

 There is still the same error



 On Thu, Nov 24, 2011 at 2:03 AM, Chris Martin wrote:

> Try removing "CCMSysUser" and only have your axl admin user in there.
> If that doesn't work restart your axl web servers on both devices.  I just
> had time to run through it and worked fine for me.  When I had both users 
> I
> got the same error you are.
>
> Chris
>
>
> On Wed, Nov 23, 2011 at 3:15 PM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>> Yes, I have that user "admin" (Just in my case it is called
>> "CCMAdministrator"), and this user is also configured in CUC Phone 
>> System.
>> But still no success :(
>>
>>
>> On Thu, Nov 24, 2011 at 1:08 AM, Chris Martin wrote:
>>
>>> Been a while since I tested pushing unity users through CUCM, but I
>>> thought you had to set the application user to an axl enabled user that 
>>> is
>>> also set in unity connection phone system.  IE: admin.
>>>
>>> I may be wrong.. Has been a while since I tried that and I don't
>>> have access to my lab right now.
>>>
>>> Chris
>>>
>>>   On Wed, Nov 23, 2011 at 11:20 AM, datucha123 datucha123 <
>>> datucha...@gmail.com> wrote:
>>>
   I am trying to create the Voice Mail Box for the User from the
 CUCM directly, meaning that I am going into Line configuration and then
 selecting the "Create Voice Mail Box" from the upper right corner.

 But when I choose the User Template (it is visible in CUCM when
 adding the VM Box) and click SAVE, it gaves me an error:"   *Unmapped
 Exception (401)Unauthorized*"

 Thus I have changed the Name in Application Server to IP address,
 and assigned the CUCM Administrator and CCMSysUser as the Selected
 Application Users for CUC Server.

 Also has unset the "Voice Mail Box Mask" in the VM Profile (as the
 CUCM 7.0 has BUG related to that).

 But still I got that error when tryin to add a VM Box from CUCM
 Device Line configuration.



 ___
 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] User Import to Unity Connection from CallManager

[Edgar Feliz]

I've done 6-7 CUCM/CUC intergration in the practice labs and had only to do
a Sync when I had a DB issue on the CUCMs.

E

On Thu, Nov 24, 2011 at 3:59 AM, datucha123 datucha123  wrote:

> I think you can do both as nothing will be ruined after the synch.
>
> On Thu, Nov 24, 2011 at 10:01 AM, Ken Wyan  wrote:
>
>> Hi Guys,
>>
>> I have a very basic question with manually created users in CUCM. ( *No*AD 
>> or LDAP integration)
>>
>> When integrating CUCM with CUC in the exam , is it sufficient to import
>> users from CUCM to CUC  or  need to Sync users after importing ?
>>
>> I saw in a CCIE video ,  user import & sync both to be done in order.
>>
>> As mentioned in cisco docs , sync is required if any later change is done
>> to CUCM users.
>>
>> Do you normally do both user import & user sync both or  do import only?
>>
>> Thanks
>>
>> Ken
>>
>>
>>
>> ___
>> 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] Need help understanding this behavior

[Priyank Kiran]

Experts,

Need help understanding the following behavior conceptually -

Have the subscriber as dedicated MOH multicast server incrementing on port
with default address 239.1.1.1 port 16384
Remote H323 gateway, with local music-on-hold wav file spoofing the above
source address.
This works as expected when put on HOLD and I see all the right output via
"show ccm-manager music-on-hold" and "debug ccm-manager music events" and
"show perf query class"

However, when I check the "*Media Termination Point Required*" box on the
gateway page in CUCM - I no longer see it sourcing off of the local router
flash and it now becomes a unicast stream sourcing off of the Subscriber
which I can see from the "show perf query class" command.

Couple questions I have is
1) What forces it to go unicast when you check the MTP required box?
2) Can you still have multicast music-on-hold stream off the local router
flash with MTP required check ON?


Thanks,
Priyank
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[OSL | CCIE_Voice] Type/Plan not as should be.

[Edgar Feliz]

I am working a a lab that requires TEHO >SiteA> calls to siteB/PSTN = calls
are routed SB-RTR and if not available SG-SLRG.


   1. SB=H323
   2. Calls out of SB should have plan/type as ISDN/National
   3. calling number = 10
   4. called number = 7

My problem is that as seen on the output below my calling # plan/type are
not showing up as ISDN/National but rather as U/U. Called Plan/Type are
correct.

 Display i = 'SiteA phone 2'
Calling Party Number
SiteB-RTR# i = 0x0081, '2025552002'
Plan:Unknown, Type:Unknown *** Should
be ISDN/National.
Called Party Number i = 0x80, '8397263'
Plan:Unknown, Type:Unknown


Any Ideas??? I am including the SB translation part of the config.

voice translation-rule 10
 rule 1 /^3...$/ /408387\0/ type any subscriber plan any isdn
 rule 2 /^2...$/ /202555\0/ type any national plan any isdn
!
!
voice translation-profile 10
 translate calling 10
!
!
dial-peer voice 7 pots
 translation-profile outgoing 10 calls on translation profile
10 above
 destination-pattern 9[2-9]..
 port 0/0/0:23
 forward-digits 7
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Re: [OSL | CCIE_Voice] Lab 6

[Peter Jeff]

Here is something new which i was stucked with
 2.2 IP Phone customization (Part I)  Directory Services
 
SB Phone 1 user is alleging unauthorized access of his Corporate directory 
services from his phone and has asked to disable access to his IP Phone’s 
Corporate Directory. You have management approval to disable the corporate 
directory for this phone only. When the directory button is depressed it should 
display no Corporate Directory.
 
(3 points)
 
thanks



From: Jonathan Bourne 
To: ccie_voice@onlinestudylist.com 
Sent: Thursday, November 24, 2011 6:08 PM
Subject: [OSL | CCIE_Voice] Lab 6


HI Guys, 

I attempted last week @ dubai and to my surprised i got lab 6.

I was not prepared for it. 

I was only prepared for lab 3 4 and 5.

Few of the questions asked 

1) I divert
2) 6 Ringtones
3) SIP Early offer

I have noted the lab and ready to share.

Please ping me at mail id, if anyone interested.

Rgds,
Jon
___
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Re: [OSL | CCIE_Voice] Lab 6

[Amit Singh]

Mate. 

Ipexperts labs have covered all these topics. 

If u have prepared for some other labs. Can't say anything. 

If u had used ipexperts. Shouldnt be a problem with these topics at least. 

Regards
Amit

Sent from my iPad

On 25/11/2011, at 1:38 AM, Jonathan Bourne  wrote:

> HI Guys,
> 
> I attempted last week @ dubai and to my surprised i got lab 6.
> 
> I was not prepared for it. 
> 
> I was only prepared for lab 3 4 and 5.
> 
> Few of the questions asked 
> 
> 1) I divert
> 2) 6 Ringtones
> 3) SIP Early offer
> 
> I have noted the lab and ready to share.
> 
> Please ping me at mail id, if anyone interested.
> 
> Rgds,
> Jon
> ___
> 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] Type/Plan not as should be.

[Edgar Feliz]

OK I found the problem...Operator Error!!!

I had this under the s0/0/0.0.1 int,

*isdn map address ^[2-9] plan unknown type unknown *than i tried this isdn
map address *^[2-9].. plan unknown type unknown* but if you don't put
the the $ at the end it will not put the correct plat/type for the calling
number. so at last i put this which i normally had been doing but had a
brain fart i guess. So fixing the ISDN map fixed the calling plan/type.

This works!!

isdn map address ^[2-9]..$ plan unknown type unknown

On Thu, Nov 24, 2011 at 1:28 PM, Edgar Feliz  wrote:

> I am working a a lab that requires TEHO >SiteA> calls to siteB/PSTN =
> calls are routed SB-RTR and if not available SG-SLRG.
>
>
>1. SB=H323
>2. Calls out of SB should have plan/type as ISDN/National
>3. calling number = 10
>4. called number = 7
>
> My problem is that as seen on the output below my calling # plan/type are
> not showing up as ISDN/National but rather as U/U. Called Plan/Type are
> correct.
>
>  Display i = 'SiteA phone 2'
> Calling Party Number
> SiteB-RTR# i = 0x0081, '2025552002'
> Plan:Unknown, Type:Unknown *** Should
> be ISDN/National.
> Called Party Number i = 0x80, '8397263'
> Plan:Unknown, Type:Unknown
>
>
> Any Ideas??? I am including the SB translation part of the config.
>
> voice translation-rule 10
>  rule 1 /^3...$/ /408387\0/ type any subscriber plan any isdn
>  rule 2 /^2...$/ /202555\0/ type any national plan any isdn
> !
> !
> voice translation-profile 10
>  translate calling 10
> !
> !
> dial-peer voice 7 pots
>  translation-profile outgoing 10 calls on translation profile
> 10 above
>  destination-pattern 9[2-9]..
>  port 0/0/0:23
>  forward-digits 7
>
___
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Re: [OSL | CCIE_Voice] Need help understanding this behavior

[datucha123 datucha123]

Very interesting thing.

I am not at my Lab side now, but tommorow will get there and test it also.

On Thu, Nov 24, 2011 at 9:50 PM, Priyank Kiran wrote:

> Experts,
>
> Need help understanding the following behavior conceptually -
>
> Have the subscriber as dedicated MOH multicast server incrementing on port
> with default address 239.1.1.1 port 16384
> Remote H323 gateway, with local music-on-hold wav file spoofing the above
> source address.
> This works as expected when put on HOLD and I see all the right output via
> "show ccm-manager music-on-hold" and "debug ccm-manager music events" and
> "show perf query class"
>
> However, when I check the "*Media Termination Point Required*" box on the
> gateway page in CUCM - I no longer see it sourcing off of the local router
> flash and it now becomes a unicast stream sourcing off of the Subscriber
> which I can see from the "show perf query class" command.
>
> Couple questions I have is
> 1) What forces it to go unicast when you check the MTP required box?
> 2) Can you still have multicast music-on-hold stream off the local router
> flash with MTP required check ON?
>
>
> Thanks,
> Priyank
>
>
>
>
>
>
>
> ___
> 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] Need help understanding this behavior

[datucha123 datucha123]

What codec are you using for normal Voice calls? is it G729?

If yes, then I have the following on my mind:

Look, when we use G729 for Voice Calls, and then try to hold the PSTN
phone, then the gateway need to connect to MoH with G711, as the Routers
Flash MoH file is G711 only.

And in your case, I think that the following happens:

When the G729 MTP is used, and the PSTN Phone is placed on hold, the MTP
needs to connect to MoH through G711, but is connects through G729 becuase
the Router software MTP cannot use the G711 at the same time.
So as the G729 MoH is choosed, the gateway does not receive it, because I
think you deny the Multicast reaching to gateway, as it uses the local
Multicast Source.
As the gateway fails to play the G729 Multicast, it falls back to Unicast.

I do not whether it will help or not, but try to add the G711 MTP to the
Gateways MRGL.

But there is also another restriction, that the Multicast cannot be
transcoded. Based on that, it comes out that the Multicast cannot traverse
to some proxies, it must be always from the reported source to destination.
And in your case the MTP comes out as some kind of proxy, and that's why
the Multicast MoH does not work. I do not know, whether I am 100% correct
or not.
On Thu, Nov 24, 2011 at 9:50 PM, Priyank Kiran wrote:

> Experts,
>
> Need help understanding the following behavior conceptually -
>
> Have the subscriber as dedicated MOH multicast server incrementing on port
> with default address 239.1.1.1 port 16384
> Remote H323 gateway, with local music-on-hold wav file spoofing the above
> source address.
> This works as expected when put on HOLD and I see all the right output via
> "show ccm-manager music-on-hold" and "debug ccm-manager music events" and
> "show perf query class"
>
> However, when I check the "*Media Termination Point Required*" box on the
> gateway page in CUCM - I no longer see it sourcing off of the local router
> flash and it now becomes a unicast stream sourcing off of the Subscriber
> which I can see from the "show perf query class" command.
>
> Couple questions I have is
> 1) What forces it to go unicast when you check the MTP required box?
> 2) Can you still have multicast music-on-hold stream off the local router
> flash with MTP required check ON?
>
>
> Thanks,
> Priyank
>
>
>
>
>
>
>
> ___
> 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] Lab 6

[datucha123 datucha123]

Can you please post the Solution for that task Amit?

Or point to the IP Expert LAB number, where the solution is?

On Thu, Nov 24, 2011 at 11:19 PM, Amit Singh  wrote:

>  Mate.
>
> Ipexperts labs have covered all these topics.
>
> If u have prepared for some other labs. Can't say anything.
>
> If u had used ipexperts. Shouldnt be a problem with these topics at least.
>
> Regards
> Amit
>
> Sent from my iPad
>
> On 25/11/2011, at 1:38 AM, Jonathan Bourne 
> wrote:
>
>  HI Guys,
>
> I attempted last week @ dubai and to my surprised i got lab 6.
>
> I was not prepared for it.
>
> I was only prepared for lab 3 4 and 5.
>
> Few of the questions asked
>
> 1) I divert
> 2) 6 Ringtones
> 3) SIP Early offer
>
>  I have noted the lab and ready to share.
>
> Please ping me at mail id, if anyone interested.
>
> Rgds,
> Jon
>
>  ___
>
> 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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www.ipexpert.com

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www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] Need help understanding this behavior

[Priyank Kiran]

Using G711. The region setting b/w the MOH server DP and phone DP is G711.

Priyank

On Thu, Nov 24, 2011 at 3:06 PM, datucha123 datucha123  wrote:

> What codec are you using for normal Voice calls? is it G729?
>
> If yes, then I have the following on my mind:
>
> Look, when we use G729 for Voice Calls, and then try to hold the PSTN
> phone, then the gateway need to connect to MoH with G711, as the Routers
> Flash MoH file is G711 only.
>
> And in your case, I think that the following happens:
>
> When the G729 MTP is used, and the PSTN Phone is placed on hold, the MTP
> needs to connect to MoH through G711, but is connects through G729 becuase
> the Router software MTP cannot use the G711 at the same time.
> So as the G729 MoH is choosed, the gateway does not receive it, because I
> think you deny the Multicast reaching to gateway, as it uses the local
> Multicast Source.
> As the gateway fails to play the G729 Multicast, it falls back to Unicast.
>
> I do not whether it will help or not, but try to add the G711 MTP to the
> Gateways MRGL.
>
> But there is also another restriction, that the Multicast cannot be
> transcoded. Based on that, it comes out that the Multicast cannot traverse
> to some proxies, it must be always from the reported source to destination.
> And in your case the MTP comes out as some kind of proxy, and that's why
> the Multicast MoH does not work. I do not know, whether I am 100% correct
> or not.
>  On Thu, Nov 24, 2011 at 9:50 PM, Priyank Kiran 
> wrote:
>
>>  Experts,
>>
>> Need help understanding the following behavior conceptually -
>>
>> Have the subscriber as dedicated MOH multicast server incrementing on
>> port with default address 239.1.1.1 port 16384
>> Remote H323 gateway, with local music-on-hold wav file spoofing the above
>> source address.
>> This works as expected when put on HOLD and I see all the right output
>> via "show ccm-manager music-on-hold" and "debug ccm-manager music events"
>> and "show perf query class"
>>
>> However, when I check the "*Media Termination Point Required*" box on
>> the gateway page in CUCM - I no longer see it sourcing off of the local
>> router flash and it now becomes a unicast stream sourcing off of the
>> Subscriber which I can see from the "show perf query class" command.
>>
>> Couple questions I have is
>> 1) What forces it to go unicast when you check the MTP required box?
>> 2) Can you still have multicast music-on-hold stream off the local router
>> flash with MTP required check ON?
>>
>>
>> Thanks,
>> Priyank
>>
>>
>>
>>
>>
>>
>>
>> ___
>> 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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www.ipexpert.com

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

Re: [OSL | CCIE_Voice] Lab 6

[Ashraf Ayyash]

In addition ,

This is a Clear NDA violation on a Public Alias , may i ask you to
stop Publish the Question you Got in the real Exam ?

Ash

On Thu, Nov 24, 2011 at 1:19 PM, Amit Singh  wrote:
> Mate.
> Ipexperts labs have covered all these topics.
> If u have prepared for some other labs. Can't say anything.
> If u had used ipexperts. Shouldnt be a problem with these topics at least.
>
> Regards
> Amit
> Sent from my iPad
> On 25/11/2011, at 1:38 AM, Jonathan Bourne 
> wrote:
>
> HI Guys,
> I attempted last week @ dubai and to my surprised i got lab 6.
> I was not prepared for it.
> I was only prepared for lab 3 4 and 5.
> Few of the questions asked
> 1) I divert
> 2) 6 Ringtones
> 3) SIP Early offer
> I have noted the lab and ready to share.
> Please ping me at mail id, if anyone interested.
> Rgds,
> Jon
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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

Re: [OSL | CCIE_Voice] Need help understanding this behavior

[datucha123 datucha123]

And what's the Regiong settings between the GW to MoH and and GW to IP
Phones?

If everything is G711, then maybe the Multicast cannot traverse the MTP?

On Fri, Nov 25, 2011 at 12:36 AM, Priyank Kiran wrote:

> Using G711. The region setting b/w the MOH server DP and phone DP is G711.
>
> Priyank
>
>   On Thu, Nov 24, 2011 at 3:06 PM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>> What codec are you using for normal Voice calls? is it G729?
>>
>> If yes, then I have the following on my mind:
>>
>> Look, when we use G729 for Voice Calls, and then try to hold the PSTN
>> phone, then the gateway need to connect to MoH with G711, as the Routers
>> Flash MoH file is G711 only.
>>
>> And in your case, I think that the following happens:
>>
>> When the G729 MTP is used, and the PSTN Phone is placed on hold, the MTP
>> needs to connect to MoH through G711, but is connects through G729 becuase
>> the Router software MTP cannot use the G711 at the same time.
>> So as the G729 MoH is choosed, the gateway does not receive it, because I
>> think you deny the Multicast reaching to gateway, as it uses the local
>> Multicast Source.
>> As the gateway fails to play the G729 Multicast, it falls back to
>> Unicast.
>>
>> I do not whether it will help or not, but try to add the G711 MTP to the
>> Gateways MRGL.
>>
>> But there is also another restriction, that the Multicast cannot be
>> transcoded. Based on that, it comes out that the Multicast cannot traverse
>> to some proxies, it must be always from the reported source to destination.
>> And in your case the MTP comes out as some kind of proxy, and that's why
>> the Multicast MoH does not work. I do not know, whether I am 100% correct
>> or not.
>>  On Thu, Nov 24, 2011 at 9:50 PM, Priyank Kiran 
>> wrote:
>>
>>>  Experts,
>>>
>>> Need help understanding the following behavior conceptually -
>>>
>>> Have the subscriber as dedicated MOH multicast server incrementing on
>>> port with default address 239.1.1.1 port 16384
>>> Remote H323 gateway, with local music-on-hold wav file spoofing the
>>> above source address.
>>> This works as expected when put on HOLD and I see all the right output
>>> via "show ccm-manager music-on-hold" and "debug ccm-manager music events"
>>> and "show perf query class"
>>>
>>> However, when I check the "*Media Termination Point Required*" box on
>>> the gateway page in CUCM - I no longer see it sourcing off of the local
>>> router flash and it now becomes a unicast stream sourcing off of the
>>> Subscriber which I can see from the "show perf query class" command.
>>>
>>> Couple questions I have is
>>> 1) What forces it to go unicast when you check the MTP required box?
>>> 2) Can you still have multicast music-on-hold stream off the local
>>> router flash with MTP required check ON?
>>>
>>>
>>> Thanks,
>>> Priyank
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> ___
>>> 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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www.ipexpert.com

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

Re: [OSL | CCIE_Voice] Lab 6

[Amit Singh]

Ipexperts volume 2 lab 1-10. 

Do them 2 times. U will have the answers. 


Regards
Amit

Sent from my iPad

On 25/11/2011, at 9:31 AM, datucha123 datucha123  wrote:

> Can you please post the Solution for that task Amit?
>  
> Or point to the IP Expert LAB number, where the solution is?
> 
> On Thu, Nov 24, 2011 at 11:19 PM, Amit Singh  wrote:
> Mate. 
> 
> Ipexperts labs have covered all these topics. 
> 
> If u have prepared for some other labs. Can't say anything. 
> 
> If u had used ipexperts. Shouldnt be a problem with these topics at least. 
> 
> Regards
> Amit
> 
> Sent from my iPad
> 
> On 25/11/2011, at 1:38 AM, Jonathan Bourne  
> wrote:
> 
>> HI Guys,
>> 
>> I attempted last week @ dubai and to my surprised i got lab 6.
>> 
>> I was not prepared for it. 
>> 
>> I was only prepared for lab 3 4 and 5.
>> 
>> Few of the questions asked 
>> 
>> 1) I divert
>> 2) 6 Ringtones
>> 3) SIP Early offer
>> 
>> I have noted the lab and ready to share.
>> 
>> Please ping me at mail id, if anyone interested.
>> 
>> Rgds,
>> Jon
>> ___
>> 
>> 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] Need help understanding this behavior

[Priyank Kiran]

I think that is exactly what I am getting at after all the observations -
perhaps multicast cannot traverse the MTP and that is where I would like to
understand the reasoning behind it.

Everything is set at G711 with my setup.



On Thu, Nov 24, 2011 at 4:08 PM, datucha123 datucha123  wrote:

> And what's the Regiong settings between the GW to MoH and and GW to IP
> Phones?
>
> If everything is G711, then maybe the Multicast cannot traverse the MTP?
>
>  On Fri, Nov 25, 2011 at 12:36 AM, Priyank Kiran 
> wrote:
>
>> Using G711. The region setting b/w the MOH server DP and phone DP is G711.
>>
>> Priyank
>>
>>   On Thu, Nov 24, 2011 at 3:06 PM, datucha123 datucha123 <
>> datucha...@gmail.com> wrote:
>>
>>> What codec are you using for normal Voice calls? is it G729?
>>>
>>> If yes, then I have the following on my mind:
>>>
>>> Look, when we use G729 for Voice Calls, and then try to hold the PSTN
>>> phone, then the gateway need to connect to MoH with G711, as the Routers
>>> Flash MoH file is G711 only.
>>>
>>> And in your case, I think that the following happens:
>>>
>>> When the G729 MTP is used, and the PSTN Phone is placed on hold, the MTP
>>> needs to connect to MoH through G711, but is connects through G729 becuase
>>> the Router software MTP cannot use the G711 at the same time.
>>> So as the G729 MoH is choosed, the gateway does not receive it,
>>> because I think you deny the Multicast reaching to gateway, as it uses the
>>> local Multicast Source.
>>> As the gateway fails to play the G729 Multicast, it falls back to
>>> Unicast.
>>>
>>> I do not whether it will help or not, but try to add the G711 MTP to the
>>> Gateways MRGL.
>>>
>>> But there is also another restriction, that the Multicast cannot be
>>> transcoded. Based on that, it comes out that the Multicast cannot traverse
>>> to some proxies, it must be always from the reported source to destination.
>>> And in your case the MTP comes out as some kind of proxy, and that's why
>>> the Multicast MoH does not work. I do not know, whether I am 100% correct
>>> or not.
>>>  On Thu, Nov 24, 2011 at 9:50 PM, Priyank Kiran >> > wrote:
>>>
  Experts,

 Need help understanding the following behavior conceptually -

 Have the subscriber as dedicated MOH multicast server incrementing on
 port with default address 239.1.1.1 port 16384
 Remote H323 gateway, with local music-on-hold wav file spoofing the
 above source address.
 This works as expected when put on HOLD and I see all the right output
 via "show ccm-manager music-on-hold" and "debug ccm-manager music events"
 and "show perf query class"

 However, when I check the "*Media Termination Point Required*" box on
 the gateway page in CUCM - I no longer see it sourcing off of the local
 router flash and it now becomes a unicast stream sourcing off of the
 Subscriber which I can see from the "show perf query class" command.

 Couple questions I have is
 1) What forces it to go unicast when you check the MTP required box?
 2) Can you still have multicast music-on-hold stream off the local
 router flash with MTP required check ON?


 Thanks,
 Priyank







 ___
 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] Need help understanding this behavior

[Mohd Baqari]

What is the MRGL assigned to the device pool of your MTP. Is it the same MoH 
multicast MRGL.

Regards,
Mohammed Al Baqari

Sent from my iPhone

On Nov 24, 2011, at 9:50 PM, Priyank Kiran  wrote:

> Experts,
>  
> Need help understanding the following behavior conceptually -
>  
> Have the subscriber as dedicated MOH multicast server incrementing on port 
> with default address 239.1.1.1 port 16384
> Remote H323 gateway, with local music-on-hold wav file spoofing the above 
> source address.
> This works as expected when put on HOLD and I see all the right output via 
> "show ccm-manager music-on-hold" and "debug ccm-manager music events" and 
> "show perf query class"
>  
> However, when I check the "Media Termination Point Required" box on the 
> gateway page in CUCM - I no longer see it sourcing off of the local router 
> flash and it now becomes a unicast stream sourcing off of the Subscriber 
> which I can see from the "show perf query class" command.
>  
> Couple questions I have is
> 1) What forces it to go unicast when you check the MTP required box?
> 2) Can you still have multicast music-on-hold stream off the local router 
> flash with MTP required check ON?
>  
>  
> Thanks,
> Priyank
>  
>  
>  
>  
>  
>  
> ___
> 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

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Re: [OSL | CCIE_Voice] Need help understanding this behavior

[Priyank Kiran]

No it's not, have 2 MRGLs

1) MRGL attached to DP of gateway and MTP are same > mrgl-siteXX > mrg-moh
+ mrg-mtp-siteXX
2) MRGL attached to DP of MoH server >  mrgl-moh  > mrg-moh

Would like to point out that I only have 1 muticast source and server in my
cluster which has been bound to mrg-moh.



On Thu, Nov 24, 2011 at 4:44 PM, Mohd Baqari wrote:

>  What is the MRGL assigned to the device pool of your MTP. Is it the same
> MoH multicast MRGL.
>
> Regards,
> Mohammed Al Baqari
>
> Sent from my iPhone
>
> On Nov 24, 2011, at 9:50 PM, Priyank Kiran 
> wrote:
>
>   Experts,
>
> Need help understanding the following behavior conceptually -
>
> Have the subscriber as dedicated MOH multicast server incrementing on port
> with default address 239.1.1.1 port 16384
> Remote H323 gateway, with local music-on-hold wav file spoofing the above
> source address.
> This works as expected when put on HOLD and I see all the right output via
> "show ccm-manager music-on-hold" and "debug ccm-manager music events" and
> "show perf query class"
>
> However, when I check the "*Media Termination Point Required*" box on the
> gateway page in CUCM - I no longer see it sourcing off of the local router
> flash and it now becomes a unicast stream sourcing off of the Subscriber
> which I can see from the "show perf query class" command.
>
> Couple questions I have is
> 1) What forces it to go unicast when you check the MTP required box?
> 2) Can you still have multicast music-on-hold stream off the local router
> flash with MTP required check ON?
>
>
> Thanks,
> Priyank
>
>
>
>
>
>
>
>  ___
>
> 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

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www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] Need help understanding this behavior

[Mohd Baqari]

The MRGL of MTP should have MoH multicast.

Regards,
Mohammed Al Baqari

Sent from my iPhone

On Nov 25, 2011, at 1:57 AM, Priyank Kiran  wrote:

> No it's not, have 2 MRGLs
>  
> 1) MRGL attached to DP of gateway and MTP are same > mrgl-siteXX > mrg-moh + 
> mrg-mtp-siteXX
> 2) MRGL attached to DP of MoH server >  mrgl-moh  > mrg-moh
>  
> Would like to point out that I only have 1 muticast source and server in my 
> cluster which has been bound to mrg-moh.
>  
> 
>  
> On Thu, Nov 24, 2011 at 4:44 PM, Mohd Baqari  wrote:
> What is the MRGL assigned to the device pool of your MTP. Is it the same MoH 
> multicast MRGL.
> 
> Regards,
> Mohammed Al Baqari
> 
> Sent from my iPhone
> 
> On Nov 24, 2011, at 9:50 PM, Priyank Kiran  wrote:
> 
>> Experts,
>>  
>> Need help understanding the following behavior conceptually -
>>  
>> Have the subscriber as dedicated MOH multicast server incrementing on port 
>> with default address 239.1.1.1 port 16384
>> Remote H323 gateway, with local music-on-hold wav file spoofing the above 
>> source address.
>> This works as expected when put on HOLD and I see all the right output via 
>> "show ccm-manager music-on-hold" and "debug ccm-manager music events" and 
>> "show perf query class"
>>  
>> However, when I check the "Media Termination Point Required" box on the 
>> gateway page in CUCM - I no longer see it sourcing off of the local router 
>> flash and it now becomes a unicast stream sourcing off of the Subscriber 
>> which I can see from the "show perf query class" command.
>>  
>> Couple questions I have is
>> 1) What forces it to go unicast when you check the MTP required box?
>> 2) Can you still have multicast music-on-hold stream off the local router 
>> flash with MTP required check ON?
>>  
>>  
>> Thanks,
>> Priyank
>>  
>>  
>>  
>>  
>>  
>>  
> 
>> ___
>> 
>> For more information regarding industry leading CCIE Lab training, please 
>> visit www.ipexpert.com
>> 
>> Are you a CCNP or CCIE and looking for a job? Check out 
>> www.PlatinumPlacement.com
> 
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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

Re: [OSL | CCIE_Voice] Need help understanding this behavior

[Priyank Kiran]

It does

On Thursday, November 24, 2011, Mohd Baqari 
wrote:
> The MRGL of MTP should have MoH multicast.
>
> Regards,
> Mohammed Al Baqari
> Sent from my iPhone
> On Nov 25, 2011, at 1:57 AM, Priyank Kiran 
wrote:
>
> No it's not, have 2 MRGLs
>
> 1) MRGL attached to DP of gateway and MTP are same > mrgl-siteXX >
mrg-moh + mrg-mtp-siteXX
> 2) MRGL attached to DP of MoH server >  mrgl-moh  > mrg-moh
>
> Would like to point out that I only have 1 muticast source and server in
my cluster which has been bound to mrg-moh.
>
>
> On Thu, Nov 24, 2011 at 4:44 PM, Mohd Baqari 
wrote:
>>
>> What is the MRGL assigned to the device pool of your MTP. Is it the same
MoH multicast MRGL.
>> Regards,
>> Mohammed Al Baqari
>> Sent from my iPhone
>> On Nov 24, 2011, at 9:50 PM, Priyank Kiran 
wrote:
>>
>> Experts,
>>
>> Need help understanding the following behavior conceptually -
>>
>> Have the subscriber as dedicated MOH multicast server incrementing on
port with default address 239.1.1.1 port 16384
>> Remote H323 gateway, with local music-on-hold wav file spoofing the
above source address.
>> This works as expected when put on HOLD and I see all the right output
via "show ccm-manager music-on-hold" and "debug ccm-manager music events"
and "show perf query class"
>>
>> However, when I check the "Media Termination Point Required" box on the
gateway page in CUCM - I no longer see it sourcing off of the local router
flash and it now becomes a unicast stream sourcing off of the Subscriber
which I can see from the "show perf query class" command.
>>
>> Couple questions I have is
>> 1) What forces it to go unicast when you check the MTP required box?
>> 2) Can you still have multicast music-on-hold stream off the local
router flash with MTP required check ON?
>>
>>
>> Thanks,
>> Priyank
>>
>>
>>
>>
>>
>>
>>
>> ___
>> 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 
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[OSL | CCIE_Voice] CCIE Payment

[Ccie Voice]

Hi all,

I added my credit card to pay for Cisco but they did not proceed the payment. I 
did not take care about it, because someone told me that Cisco now proceeding 
the payment after the lab, I went for CCIE lab and I did my lab but till now I 
did not receive my result and  the payment still pending.

Any body attempt soon can tell me if he paid before or after the lab?

Regards,
___
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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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