Re: [OSL | CCIE_Voice] Cme background image

[Ki Wi]

I always use alias to avoid some path related issue

Use debug tftp events to see what the phone is looking for when you are 
pressing around. 

Sent from my iPhone
Pls pardon my fat fingers.

On Oct 18, 2010, at 2:58 AM, Goran Selthofer  wrote:

> what do you get on the phone when you try to click on background images 
> selection?
> 
> did you enable http server on your router?
> 
> also, folder path is very important as for the phone types you are using.
> i recommend to use this document for that as it lists formats/folders for 
> different phone types (and that is still valid for CME as well)
> http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configuration_example09186a0080b3690c.shtml
> 
> 
> 
> 
> On Sun, Oct 17, 2010 at 8:04 PM, Prashant Patel  
> wrote:
> Make sure the filename is exactly as requested in the tftp request from phone.
>  
> HTH
> Prashant
> On Sun, Oct 17, 2010 at 2:01 PM, fatai_adeku...@yahoo.com 
>  wrote:
> I worked on putting a background image on a cucme router. I uploaded the 
> background image successfully n configure ''tftp server flash ..'' on the 
> cme. I  created cnf files in telephony service, reloaded d router and checked 
> if d image is available for the phone but to know avail. Anybody with an idea 
> of what i am doing wrong?
> Tks.
> 
> Sent from my Nokia phone
> ___
> 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
> 
> 
> ___
> 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

Re: [OSL | CCIE_Voice] Call Forward Unregistered

[Mark Holloway]

I think the main thing to understand is that it should work using E164 in 
For/By under normal circumstances and everything else we are suggesting is a 
work around to a known bug with CUCM 7.0 and VMWare. 


On Oct 17, 2010, at 3:56 PM, Daniel Berlinski wrote:

> Hello guys
> 
> If you want to manipulate this with CUCM the place to change the redirected 
> number is the VM profile as indicated by Mark.  Alternatively you could 
> attach an additional rule to the translation-profile plugged inbound to the 
> POTS call leg in the branch router in SRST mode and configure it to change 
> the redirect-called number from  to the e164 that you are after.
> 
> Cheers
> 
> On Mon, Oct 18, 2010 at 11:36 AM, Mark Holloway  wrote:
> I ran into this same issue. Supposedly it's a problem with CUCM 7.0 and 
> VMWare.  If you go to the Device > Phone and click on the Site B phones > 
> Line and specifically assign the Voicemail Profile to the Line it might work. 
>  I had success a couple of times doing this, but then after resetting my rack 
> the last time and assigning the VM profile to the Line I still had this 
> issue. 
> 
> On Oct 17, 2010, at 3:28 PM, Afzal Bhutta wrote:
> 
>> Scenario:
>> 
>> In SRST mode: HQ MGCP gateway and Site-B H322 gateway Site-C H322 gateway cme
>> 
>> HQ and Site C phones are being able to call SiteB Phone 1 using 4 digits 
>> dialing in SRST.(Wan failure)
>> 
>> I use call forward unregistered feature.
>> 
>> When I call from HQ Phone-1 call routed through HQ Gateway.
>> When I call from Site-C Phone-1 call routed through the GK first and then HQ 
>> Gateway.
>> Below is the display I am getting on my Site-B phone display.
>>  
>> Forward HQ Phone 1
>> (2001)
>> For   3001
>> By3001
>>  
>> Forward Site-C Phone 1
>> (4001)
>> For   3001
>> By3001
>>  
>> My question how can I achieve below display in FOR and BY field it should be 
>> E.164 number format and than 4 digits internal ID
>>  
>>  
>> Forward
>> (2001)
>> For   +19723033001 (3...)
>> By+19723033001 (3...)
>> Forward
>> (4001)
>> For   +19723033001 (3...)
>> By+19723033001 (3...)
>>  
>> Thanking you in anticipation folks.
>> 
>> ___
>> 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
> 
> 

___
For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] Cme background image

[Francisco .]

Check...
1. Device Pool  -> Any Local route group selected?
2. CSS -> Any patition in selected partitions?
 

 
> Date: Sun, 17 Oct 2010 18:01:19 +
> From: fatai_adeku...@yahoo.com
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] Cme background image
> 
> I worked on putting a background image on a cucme router. I uploaded the 
> background image successfully n configure ''tftp server flash ..'' on the 
> cme. I created cnf files in telephony service, reloaded d router and checked 
> if d image is available for the phone but to know avail. Anybody with an idea 
> of what i am doing wrong?
> Tks.
> 
> Sent from my Nokia phone
> ___
> 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

Re: [OSL | CCIE_Voice] Call Forward Unregistered

[Daniel Berlinski]

Hello guys

If you want to manipulate this with CUCM the place to change the redirected
number is the VM profile as indicated by Mark.  Alternatively you could
attach an additional rule to the translation-profile plugged inbound to the
POTS call leg in the branch router in SRST mode and configure it to change
the redirect-called number from  to the e164 that you are after.

Cheers

On Mon, Oct 18, 2010 at 11:36 AM, Mark Holloway  wrote:

> I ran into this same issue. Supposedly it's a problem with CUCM 7.0 and
> VMWare.  If you go to the Device > Phone and click on the Site B phones >
> Line and specifically assign the Voicemail Profile to the Line it might
> work.  I had success a couple of times doing this, but then after resetting
> my rack the last time and assigning the VM profile to the Line I still had
> this issue.
>
> On Oct 17, 2010, at 3:28 PM, Afzal Bhutta wrote:
>
> Scenario:
>
> In SRST mode: HQ MGCP gateway and Site-B H322 gateway Site-C H322 gateway
> cme
>
> HQ and Site C phones are being able to call SiteB Phone 1 using 4 digits
> dialing in SRST.(Wan failure)
>
> I use call forward unregistered feature.
> When I call from HQ Phone-1 call routed through HQ Gateway.
> When I call from Site-C Phone-1 call routed through the GK first and then
> HQ Gateway.
> Below is the display I am getting on my Site-B phone display.
>
>
> Forward HQ Phone 1
> (2001)
> For   3001
> By3001
>
>
> Forward Site-C Phone 1
> (4001)
> For   3001
> By3001
>
>
> My question how can I achieve below display in FOR and BY field it should
> be E.164 number format and than 4 digits internal ID
>
>
>
>
> Forward
> (2001)
> For   +19723033001 (3...)
> By+19723033001 (3...)
> Forward
> (4001)
> For   +19723033001 (3...)
> By+19723033001 (3...)
>
>
> Thanking you in anticipation folks.
> ___
> 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
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

[OSL | CCIE_Voice] CUC WMI issue

[Francisco .]

Fellows,
 
MWI issue
 
CUC integrated with CUCM.
Users imported from CUCM
MWI extension same (1998/1999) on both CUCM & CUC.
Null partition & CSS
Able to call both extensions for light on/off from all phones.
One phone system in CUC
Rebooted both CUC & CUCM
 
After all the tasks above, No MWI when message is left on any of the phones, 
though by pressing the message button you can retrieve these messages.
 
Anyone with similar issue or an idea?
 
Thanks in advance.___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Call Forward Unregistered

[Mark Holloway]

I ran into this same issue. Supposedly it's a problem with CUCM 7.0 and VMWare. 
 If you go to the Device > Phone and click on the Site B phones > Line and 
specifically assign the Voicemail Profile to the Line it might work.  I had 
success a couple of times doing this, but then after resetting my rack the last 
time and assigning the VM profile to the Line I still had this issue. 

On Oct 17, 2010, at 3:28 PM, Afzal Bhutta wrote:

> Scenario:
> 
> In SRST mode: HQ MGCP gateway and Site-B H322 gateway Site-C H322 gateway cme
> 
> HQ and Site C phones are being able to call SiteB Phone 1 using 4 digits 
> dialing in SRST.(Wan failure)
> 
> I use call forward unregistered feature.
> 
> When I call from HQ Phone-1 call routed through HQ Gateway.
> When I call from Site-C Phone-1 call routed through the GK first and then HQ 
> Gateway.
> Below is the display I am getting on my Site-B phone display.
>  
> Forward HQ Phone 1
> (2001)
> For   3001
> By3001
>  
> Forward Site-C Phone 1
> (4001)
> For   3001
> By3001
>  
> My question how can I achieve below display in FOR and BY field it should be 
> E.164 number format and than 4 digits internal ID
>  
>  
> Forward
> (2001)
> For   +19723033001 (3...)
> By+19723033001 (3...)
> Forward
> (4001)
> For   +19723033001 (3...)
> By+19723033001 (3...)
>  
> Thanking you in anticipation folks.
> 
> ___
> 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

[OSL | CCIE_Voice] Call Forward Unregistered

[Afzal Bhutta]

Scenario:

In SRST mode: HQ MGCP gateway and Site-B H322 gateway Site-C H322 gateway
cme

HQ and Site C phones are being able to call SiteB Phone 1 using 4 digits
dialing in SRST.(Wan failure)

I use call forward unregistered feature.

When I call from HQ Phone-1 call routed through HQ Gateway.

When I call from Site-C Phone-1 call routed through the GK first and then HQ
Gateway.

Below is the display I am getting on my Site-B phone display.



Forward HQ Phone 1

(2001)

For   3001

By3001



Forward Site-C Phone 1

(4001)

For   3001

By3001



My question how can I achieve below display in FOR and BY field it should be
E.164 number format and than 4 digits internal ID





Forward

(2001)

For   +19723033001 (3...)

By+19723033001 (3...)

Forward

(4001)

For   +19723033001 (3...)

By+19723033001 (3...)



Thanking you in anticipation folks.
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

[OSL | CCIE_Voice] maximum sessions 0

[Bill Lake]

Hello,



I am going to assume that you have a FXO or FXS voice card, given that, you
will need a second PVDM to be able to use both conference bridges and the
voice FXO/FXS in your Cisco 2811.  You can also migrate this voice card to a
network module with DSP resources to free up more DSP resources for your
router.



I know PVDM’s are not inexpensive but if you need hardware conference
ability and FXO/FXS cards you will need multiple PVDM’s



http://www.cisco.com/cgi-bin/Support/DSP/cisco_prodsel.pl







Sincerely,

Bill
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Digit Manipulation on H323

[Cristobal Priego]

thank you all for your comments

I've been debugging the gateway

i have digit manipulation on the rp, rl. I also have a called pattern
transformation css applied to the h323 gateway
i've reset the GW multiple times

however when i make the call

this is what i see

 Exclusive, Channel 1
Display i = 'HQ Phone 2'
Calling Party Number
HQ-RTR(config-gk)# i = 0x0081, '12123945002'
Plan:Unknown, Type:Unknown
Called Party Number i = 0x80, '16178632683'
Plan:Unknown, Type:Unknown


My calling party transformation CSS is being hit by the GW

10/17/2010 16:01:47.979 CCM|SPROC  DATransformMatch - matchNumber
[5002]transformCSSPkid [c85a253f-f88d-7fbb-50b0-ff8aaacde5c3]
transformationCss
[gw-hq-pt] patternUsage [15] paternNodeID
[b071ad82-eaf4-c3c5-09df-6acba1ad0832] OutpulsedNum.nd [+12123945002] tn
 [2] pi [1] npi
[0]|


however i didn't see the called party transformation pattern being hit

and i can't get the call type to work properly nor the + dialing

without Dialing Rules

with dialing rules works great
however i was wondering if it could be done from CUCM


2010/10/17 Prashant Patel 

> To set Called and Calling type on H323 GW it is preferred to use
> voice-translation rules on the gateway so that if you go into srst you dont
> have to do it again. However if no srst you can set it in CUCM.
>
> HTH,
> Prashant
>
> On Sun, Oct 17, 2010 at 3:02 PM, Cristobal Priego <
> cristobalpri...@gmail.com> wrote:
>
>> thank you very much it really make sense
>>
>> and what would happen to the call type ?
>>
>> if i use a called party transformation pattern
>>
>> to set the called type to: national, subscriber, intl,
>>
>> on the MGCP gateway it's no problem the call is sent out on the PRI with
>> the proper digit manipulation and the proper call type
>>
>> on the other hand on H323. stripping digits from ccm are passed to the
>> h323 gw, but that's it, no call type at all
>>
>> this is when the translation rules comes in place, right ?
>>
>>
>> 2010/10/17 Goran Selthofer 
>>
>> Hi C.P,
>>>
>>> Digit manipulation will be done on CUCM and will be sent to H323 as well,
>>> and the preference would be on the manipulations done within RL rather than
>>> on RP.
>>> So, i.e.
>>> RP is 91608.[2-9]XX and
>>> - if you put pre-dot and prefix 608 under RP,
>>> - and then you also do pre-dot and prefix 9 for specific RG (for your
>>> h323 gw) under RL,
>>>
>>>  then your h323 gw will receive 9[2-9]XX
>>>
>>> hence, dial-peer pots on your h323 gw needed to terminate this call
>>> should have the same/similar destination-pattern configured, i.e:
>>>
>>> dial-peer voice 9 pots
>>> destination-pattern 9[2-9]..$
>>> port 0/1/0:23
>>>
>>>
>>> Now, the real trick comes if you want to actually influence your calling
>>> phone LCD digit presentations of DNIS (so, not ANI on the receiving end, but
>>> the actual dialed number on the calling end being presented on your phone
>>> from which you are dialing those digits - this is where the difference
>>> between mgcp and h323 gw can be seen).
>>>
>>> mgcp will present whatever manipulations you've done using RP (will not
>>> present back to calling phone LCD what you have done withing RG/RL
>>> manipulations though it will use those manipulations to send to the GW).
>>>
>>> however, in case of h323 gw, manipulations on DNIS done withing RG/RL
>>> will be also presented back to calling phone LCD.
>>> Now, since that is H323, you can still have one more chance to do your
>>> digits manipulations and influence back presenting of dialed digits to
>>> calling phone - voice transformation rules/profiles attached to pots
>>> dial-peer (or forward-digits under dial-peer but that one will not influence
>>> LCD DNIS presentation on the calling phone)
>>>
>>> i.e. if for above example we want to actually show 9 in front of local
>>> number, we can just put 'forward-digits 7' under above pots and that's it.
>>>  dial-peer voice 9 pots
>>> destination-pattern 9[2-9]..$
>>> port 0/1/0:23
>>> forward-digits 7
>>>
>>> But, if we would like to show ONLY local number, without leading 9 back
>>> to the caller on his ip phone LCD, then we would have to strip that 9 inside
>>> voice translation-rule, i.e:
>>>
>>> voice translation-rule 9
>>>  rule 1 /^9\([2-9]..$\)/ /\1/ p any sub t any sub
>>>
>>> voice translation-profile 9
>>>  translate called 9
>>>
>>> and then add that to above dp:
>>>
>>>  dial-peer voice 9 pots
>>>  translation-profile out 9
>>> destination-pattern 9[2-9]..$
>>> port 0/1/0:23
>>>
>>> so this will result in showing only 7 digits back to LCD of the calling
>>> phone. (if dialed number was 91234567, it will show back only 1234567).
>>>
>>> here, you can also include forward-digits as well, but
>>> translation-profile will still have precedence
>>>
>>>  dial-peer voice 9 pots
>>>  translation-profile out 9
>>> destination-pattern 9[2-9]..$
>>> port 0/1/0:23
>>> forward 7
>>>
>>>
>>> in both cases 

Re: [OSL | CCIE_Voice] Digit Manipulation on H323

[Prashant Patel]

To set Called and Calling type on H323 GW it is preferred to use
voice-translation rules on the gateway so that if you go into srst you dont
have to do it again. However if no srst you can set it in CUCM.

HTH,
Prashant

On Sun, Oct 17, 2010 at 3:02 PM, Cristobal Priego  wrote:

> thank you very much it really make sense
>
> and what would happen to the call type ?
>
> if i use a called party transformation pattern
>
> to set the called type to: national, subscriber, intl,
>
> on the MGCP gateway it's no problem the call is sent out on the PRI with
> the proper digit manipulation and the proper call type
>
> on the other hand on H323. stripping digits from ccm are passed to the h323
> gw, but that's it, no call type at all
>
> this is when the translation rules comes in place, right ?
>
>
> 2010/10/17 Goran Selthofer 
>
> Hi C.P,
>>
>> Digit manipulation will be done on CUCM and will be sent to H323 as well,
>> and the preference would be on the manipulations done within RL rather than
>> on RP.
>> So, i.e.
>> RP is 91608.[2-9]XX and
>> - if you put pre-dot and prefix 608 under RP,
>> - and then you also do pre-dot and prefix 9 for specific RG (for your h323
>> gw) under RL,
>>
>>  then your h323 gw will receive 9[2-9]XX
>>
>> hence, dial-peer pots on your h323 gw needed to terminate this call should
>> have the same/similar destination-pattern configured, i.e:
>>
>> dial-peer voice 9 pots
>> destination-pattern 9[2-9]..$
>> port 0/1/0:23
>>
>>
>> Now, the real trick comes if you want to actually influence your calling
>> phone LCD digit presentations of DNIS (so, not ANI on the receiving end, but
>> the actual dialed number on the calling end being presented on your phone
>> from which you are dialing those digits - this is where the difference
>> between mgcp and h323 gw can be seen).
>>
>> mgcp will present whatever manipulations you've done using RP (will not
>> present back to calling phone LCD what you have done withing RG/RL
>> manipulations though it will use those manipulations to send to the GW).
>>
>> however, in case of h323 gw, manipulations on DNIS done withing RG/RL will
>> be also presented back to calling phone LCD.
>> Now, since that is H323, you can still have one more chance to do your
>> digits manipulations and influence back presenting of dialed digits to
>> calling phone - voice transformation rules/profiles attached to pots
>> dial-peer (or forward-digits under dial-peer but that one will not influence
>> LCD DNIS presentation on the calling phone)
>>
>> i.e. if for above example we want to actually show 9 in front of local
>> number, we can just put 'forward-digits 7' under above pots and that's it.
>>  dial-peer voice 9 pots
>> destination-pattern 9[2-9]..$
>> port 0/1/0:23
>> forward-digits 7
>>
>> But, if we would like to show ONLY local number, without leading 9 back to
>> the caller on his ip phone LCD, then we would have to strip that 9 inside
>> voice translation-rule, i.e:
>>
>> voice translation-rule 9
>>  rule 1 /^9\([2-9]..$\)/ /\1/ p any sub t any sub
>>
>> voice translation-profile 9
>>  translate called 9
>>
>> and then add that to above dp:
>>
>>  dial-peer voice 9 pots
>>  translation-profile out 9
>> destination-pattern 9[2-9]..$
>> port 0/1/0:23
>>
>> so this will result in showing only 7 digits back to LCD of the calling
>> phone. (if dialed number was 91234567, it will show back only 1234567).
>>
>> here, you can also include forward-digits as well, but translation-profile
>> will still have precedence
>>
>>  dial-peer voice 9 pots
>>  translation-profile out 9
>> destination-pattern 9[2-9]..$
>> port 0/1/0:23
>> forward 7
>>
>>
>> in both cases you are sending 7 digits to PSTN, just the difference is
>> what you will present back to the caller who actually dialed this number.
>>
>> and that is the difference with mgcp, as you don't have that extra step to
>> manipulate DNIS - all needs to be done on the CUCM withing RP, RG of RL or
>> CalledPartTransformationPattern attached to outgoing mgcp gw.
>>
>>
>> hope this will give you some clues how it works...
>>
>> cheers,
>> G.
>>
>>
>>
>>   On Sun, Oct 17, 2010 at 8:17 PM, Cristobal Priego <
>> cristobalpri...@gmail.com> wrote:
>>
>>>  hello all,
>>>
>>> I'm working on the workbook 1 lab 5
>>>
>>> and i noticed what when i do digit manipulation either on the RP, RL or
>>> by using transformation patterns, the changes aren't sent to the GW, if my
>>> protocol is H.323 usually I need to create some dial rules on the Voice
>>> Gateway
>>>
>>> when I'm using MGCP i have no problem
>>>
>>> i was wondering if there is a setting on the ccm that will allow ccm to
>>> send the digit manipulation to the GW or does it has to be manually done at
>>> the GW level ?
>>>
>>> could you please explain a bit for me
>>>
>>> thank you
>>>
>>> ___
>>> For more information regarding industry leading CCIE Lab training, please
>>> visit www.ipexpert.com
>>>
>>>
>>
>

Re: [OSL | CCIE_Voice] Digit Manipulation on H323

[Cristobal Priego]

thank you very much it really make sense

and what would happen to the call type ?

if i use a called party transformation pattern

to set the called type to: national, subscriber, intl,

on the MGCP gateway it's no problem the call is sent out on the PRI with the
proper digit manipulation and the proper call type

on the other hand on H323. stripping digits from ccm are passed to the h323
gw, but that's it, no call type at all

this is when the translation rules comes in place, right ?


2010/10/17 Goran Selthofer 

> Hi C.P,
>
> Digit manipulation will be done on CUCM and will be sent to H323 as well,
> and the preference would be on the manipulations done within RL rather than
> on RP.
> So, i.e.
> RP is 91608.[2-9]XX and
> - if you put pre-dot and prefix 608 under RP,
> - and then you also do pre-dot and prefix 9 for specific RG (for your h323
> gw) under RL,
>
>  then your h323 gw will receive 9[2-9]XX
>
> hence, dial-peer pots on your h323 gw needed to terminate this call should
> have the same/similar destination-pattern configured, i.e:
>
> dial-peer voice 9 pots
> destination-pattern 9[2-9]..$
> port 0/1/0:23
>
>
> Now, the real trick comes if you want to actually influence your calling
> phone LCD digit presentations of DNIS (so, not ANI on the receiving end, but
> the actual dialed number on the calling end being presented on your phone
> from which you are dialing those digits - this is where the difference
> between mgcp and h323 gw can be seen).
>
> mgcp will present whatever manipulations you've done using RP (will not
> present back to calling phone LCD what you have done withing RG/RL
> manipulations though it will use those manipulations to send to the GW).
>
> however, in case of h323 gw, manipulations on DNIS done withing RG/RL will
> be also presented back to calling phone LCD.
> Now, since that is H323, you can still have one more chance to do your
> digits manipulations and influence back presenting of dialed digits to
> calling phone - voice transformation rules/profiles attached to pots
> dial-peer (or forward-digits under dial-peer but that one will not influence
> LCD DNIS presentation on the calling phone)
>
> i.e. if for above example we want to actually show 9 in front of local
> number, we can just put 'forward-digits 7' under above pots and that's it.
> dial-peer voice 9 pots
> destination-pattern 9[2-9]..$
> port 0/1/0:23
> forward-digits 7
>
> But, if we would like to show ONLY local number, without leading 9 back to
> the caller on his ip phone LCD, then we would have to strip that 9 inside
> voice translation-rule, i.e:
>
> voice translation-rule 9
>  rule 1 /^9\([2-9]..$\)/ /\1/ p any sub t any sub
>
> voice translation-profile 9
>  translate called 9
>
> and then add that to above dp:
>
> dial-peer voice 9 pots
>  translation-profile out 9
> destination-pattern 9[2-9]..$
> port 0/1/0:23
>
> so this will result in showing only 7 digits back to LCD of the calling
> phone. (if dialed number was 91234567, it will show back only 1234567).
>
> here, you can also include forward-digits as well, but translation-profile
> will still have precedence
>
> dial-peer voice 9 pots
>  translation-profile out 9
> destination-pattern 9[2-9]..$
> port 0/1/0:23
> forward 7
>
>
> in both cases you are sending 7 digits to PSTN, just the difference is what
> you will present back to the caller who actually dialed this number.
>
> and that is the difference with mgcp, as you don't have that extra step to
> manipulate DNIS - all needs to be done on the CUCM withing RP, RG of RL or
> CalledPartTransformationPattern attached to outgoing mgcp gw.
>
>
> hope this will give you some clues how it works...
>
> cheers,
> G.
>
>
>
> On Sun, Oct 17, 2010 at 8:17 PM, Cristobal Priego <
> cristobalpri...@gmail.com> wrote:
>
>> hello all,
>>
>> I'm working on the workbook 1 lab 5
>>
>> and i noticed what when i do digit manipulation either on the RP, RL or by
>> using transformation patterns, the changes aren't sent to the GW, if my
>> protocol is H.323 usually I need to create some dial rules on the Voice
>> Gateway
>>
>> when I'm using MGCP i have no problem
>>
>> i was wondering if there is a setting on the ccm that will allow ccm to
>> send the digit manipulation to the GW or does it has to be manually done at
>> the GW level ?
>>
>> could you please explain a bit for me
>>
>> thank you
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>
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[OSL | CCIE_Voice] OSL | CCIE_Voice] Mgcp outbound call fails.

[Pithog Oil]

 
Check your route group, route list and device pool ,properly, also ensure 
correct CSS on the gateway, check if that CSS has the PT assigned to phone.
 
Check the few i have mentioned Critically.
 
you could also recreate your PRI-group
 
Pithog Oil


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Re: [OSL | CCIE_Voice] Cme background image

[Goran Selthofer]

what do you get on the phone when you try to click on background images
selection?

did you enable http server on your router?

also, folder path is very important as for the phone types you are using.
i recommend to use this document for that as it lists formats/folders for
different phone types (and that is still valid for CME as well)
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configuration_example09186a0080b3690c.shtml




On Sun, Oct 17, 2010 at 8:04 PM, Prashant Patel
wrote:

> Make sure the filename is exactly as requested in the tftp request from
> phone.
>
> HTH
> Prashant
> On Sun, Oct 17, 2010 at 2:01 PM, fatai_adeku...@yahoo.com <
> fatai_adeku...@yahoo.com> wrote:
>
>> I worked on putting a background image on a cucme router. I uploaded the
>> background image successfully n configure ''tftp server flash ..'' on
>> the cme. I  created cnf files in telephony service, reloaded d router and
>> checked if d image is available for the phone but to know avail. Anybody
>> with an idea of what i am doing wrong?
>> Tks.
>>
>> Sent from my Nokia phone
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>
>
> ___
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> visit www.ipexpert.com
>
>
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Re: [OSL | CCIE_Voice] Mgcp outbound call fails.

[Goran Selthofer]

do debug isdn q931, place the call and let us know what do you get



On Sun, Oct 17, 2010 at 8:35 PM, Cristobal Priego  wrote:

> do you see your call hitting the gateway ?
> do you have a pri ?
>
> check your digit manipulation
>
> 2010/10/17 fatai_adeku...@yahoo.com 
>
> Hello guys,
>> I experienced a problem where i cannot make any outbound call from an cucm
>> through an mgcp gateway. My mgcp gateway was registered to d cucm, right
>> pt/css were applied to d calling phone and correct partition to the route
>> pattern. When 911 is called, a disconnect tone is heard, when i call any
>> other number it says ''number you have dialed cannot be ...'' note that
>> inbound calls are working fine.
>>
>> Anybody seen dis issue before?
>>
>> Tjs.
>>
>> Sent from my Nokia phone
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>
>
> ___
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> visit www.ipexpert.com
>
>
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Re: [OSL | CCIE_Voice] Digit Manipulation on H323

[Goran Selthofer]

Hi C.P,

Digit manipulation will be done on CUCM and will be sent to H323 as well,
and the preference would be on the manipulations done within RL rather than
on RP.
So, i.e.
RP is 91608.[2-9]XX and
- if you put pre-dot and prefix 608 under RP,
- and then you also do pre-dot and prefix 9 for specific RG (for your h323
gw) under RL,

 then your h323 gw will receive 9[2-9]XX

hence, dial-peer pots on your h323 gw needed to terminate this call should
have the same/similar destination-pattern configured, i.e:

dial-peer voice 9 pots
destination-pattern 9[2-9]..$
port 0/1/0:23


Now, the real trick comes if you want to actually influence your calling
phone LCD digit presentations of DNIS (so, not ANI on the receiving end, but
the actual dialed number on the calling end being presented on your phone
from which you are dialing those digits - this is where the difference
between mgcp and h323 gw can be seen).

mgcp will present whatever manipulations you've done using RP (will not
present back to calling phone LCD what you have done withing RG/RL
manipulations though it will use those manipulations to send to the GW).

however, in case of h323 gw, manipulations on DNIS done withing RG/RL will
be also presented back to calling phone LCD.
Now, since that is H323, you can still have one more chance to do your
digits manipulations and influence back presenting of dialed digits to
calling phone - voice transformation rules/profiles attached to pots
dial-peer (or forward-digits under dial-peer but that one will not influence
LCD DNIS presentation on the calling phone)

i.e. if for above example we want to actually show 9 in front of local
number, we can just put 'forward-digits 7' under above pots and that's it.
dial-peer voice 9 pots
destination-pattern 9[2-9]..$
port 0/1/0:23
forward-digits 7

But, if we would like to show ONLY local number, without leading 9 back to
the caller on his ip phone LCD, then we would have to strip that 9 inside
voice translation-rule, i.e:

voice translation-rule 9
 rule 1 /^9\([2-9]..$\)/ /\1/ p any sub t any sub

voice translation-profile 9
 translate called 9

and then add that to above dp:

dial-peer voice 9 pots
 translation-profile out 9
destination-pattern 9[2-9]..$
port 0/1/0:23

so this will result in showing only 7 digits back to LCD of the calling
phone. (if dialed number was 91234567, it will show back only 1234567).

here, you can also include forward-digits as well, but translation-profile
will still have precedence

dial-peer voice 9 pots
 translation-profile out 9
destination-pattern 9[2-9]..$
port 0/1/0:23
forward 7


in both cases you are sending 7 digits to PSTN, just the difference is what
you will present back to the caller who actually dialed this number.

and that is the difference with mgcp, as you don't have that extra step to
manipulate DNIS - all needs to be done on the CUCM withing RP, RG of RL or
CalledPartTransformationPattern attached to outgoing mgcp gw.


hope this will give you some clues how it works...

cheers,
G.



On Sun, Oct 17, 2010 at 8:17 PM, Cristobal Priego  wrote:

> hello all,
>
> I'm working on the workbook 1 lab 5
>
> and i noticed what when i do digit manipulation either on the RP, RL or by
> using transformation patterns, the changes aren't sent to the GW, if my
> protocol is H.323 usually I need to create some dial rules on the Voice
> Gateway
>
> when I'm using MGCP i have no problem
>
> i was wondering if there is a setting on the ccm that will allow ccm to
> send the digit manipulation to the GW or does it has to be manually done at
> the GW level ?
>
> could you please explain a bit for me
>
> thank you
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
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Re: [OSL | CCIE_Voice] Mgcp outbound call fails.

[Cristobal Priego]

do you see your call hitting the gateway ?
do you have a pri ?

check your digit manipulation

2010/10/17 fatai_adeku...@yahoo.com 

> Hello guys,
> I experienced a problem where i cannot make any outbound call from an cucm
> through an mgcp gateway. My mgcp gateway was registered to d cucm, right
> pt/css were applied to d calling phone and correct partition to the route
> pattern. When 911 is called, a disconnect tone is heard, when i call any
> other number it says ''number you have dialed cannot be ...'' note that
> inbound calls are working fine.
>
> Anybody seen dis issue before?
>
> Tjs.
>
> Sent from my Nokia phone
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
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[OSL | CCIE_Voice] Digit Manipulation on H323

[Cristobal Priego]

hello all,

I'm working on the workbook 1 lab 5

and i noticed what when i do digit manipulation either on the RP, RL or by
using transformation patterns, the changes aren't sent to the GW, if my
protocol is H.323 usually I need to create some dial rules on the Voice
Gateway

when I'm using MGCP i have no problem

i was wondering if there is a setting on the ccm that will allow ccm to send
the digit manipulation to the GW or does it has to be manually done at the
GW level ?

could you please explain a bit for me

thank you
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Re: [OSL | CCIE_Voice] Cme background image

[Prashant Patel]

Make sure the filename is exactly as requested in the tftp request from
phone.

HTH
Prashant
On Sun, Oct 17, 2010 at 2:01 PM, fatai_adeku...@yahoo.com <
fatai_adeku...@yahoo.com> wrote:

> I worked on putting a background image on a cucme router. I uploaded the
> background image successfully n configure ''tftp server flash ..'' on
> the cme. I  created cnf files in telephony service, reloaded d router and
> checked if d image is available for the phone but to know avail. Anybody
> with an idea of what i am doing wrong?
> Tks.
>
> Sent from my Nokia phone
> ___
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> visit www.ipexpert.com
>
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[OSL | CCIE_Voice] Cme background image

[fatai_adeku...@yahoo.com]

I worked on putting a background image on a cucme router. I uploaded the 
background image successfully n configure ''tftp server flash ..'' on the 
cme. I  created cnf files in telephony service, reloaded d router and checked 
if d image is available for the phone but to know avail. Anybody with an idea 
of what i am doing wrong?
Tks.

Sent from my Nokia phone
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[OSL | CCIE_Voice] (no subject)

[Bhushan Paranjape]

Hi,
 
In PL WB 1 Lab 6 and onwards in the config tasks -prerequisites I see a NOTE:
If you are using your own hw cisco 7961 phones instead of 7962 phones, please 
perform the following: first delete all 7962 phones and then run the BAT tool 
for phone install. We have laready preprovisioned a file that you simply need 
to import (and change MAC add) containing the 7961 phones types.
 
My question is where do I find this file to import? I am not getting any 
response from ipexpert support team on this. Currently I just delete the 7962s 
and add my 7961s.
 
Pls advise
 
Thanks,
 
Bhushan.


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[OSL | CCIE_Voice] Mgcp outbound call fails.

[fatai_adeku...@yahoo.com]

Hello guys,
I experienced a problem where i cannot make any outbound call from an cucm 
through an mgcp gateway. My mgcp gateway was registered to d cucm, right pt/css 
were applied to d calling phone and correct partition to the route pattern. 
When 911 is called, a disconnect tone is heard, when i call any other number it 
says ''number you have dialed cannot be ...'' note that inbound calls are 
working fine.

Anybody seen dis issue before?

Tjs.  

Sent from my Nokia phone
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Re: [OSL | CCIE_Voice] Vol2 Lab7 cue aar

[Roger Källberg]

Hi David,
Sounds like your call never gets to the VGW. Have you verified that your RP 
used for AAR will match to send the call to the gateway? You might also want to 
verify that you have reset the RL used, pretty common thing to forget.

What kind of VGW do you use, H.323 or MGCP. With the first you might want to 
run "debug voip dialpeer", to see what ingress dp that is used. That is if the 
call even gets sent to the VGW. If not the problem is within the UCM. Might be 
pretty obvious, but have you activated AAR in the SP?

About AAR group, you should only set that on the line, setting it on the device 
might cause some unpredicted behaviour.

Sincerely
Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB

Från: David A [david.a...@gmail.com]
Skickat: den 17 oktober 2010 16:10
Till: Roger Källberg
Kopia: ccie_voice@onlinestudylist.com
Ämne: Re: [OSL | CCIE_Voice] Vol2 Lab7 cue aar

Hi Roger,

I have the EPNM on the CTI ports set to the CTI-RP/VM pilot ie 3600 and EPMN is 
02077353600. I have no idea why the call doesnt pass thru to the gateway.

Thanks,
DA




2010/10/17 Roger Källberg 
mailto:roger.kallb...@cygate.se>>
You need to set the EPNM on the CTI ports to point to the number of the CTI RP 
for CUE. This is since the call can not go directly to the CTI ports, it has to 
first be sent to the CTI RP, then on to the CTI port.

Sincerely

Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB

Från: David A [david.a...@gmail.com]
Skickat: den 16 oktober 2010 19:18
Till: ccie_voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] Vol2 Lab7 cue aar

Hi all,


I always get a busy when I configure AAR for cue and dial from HQ or SiteB.

I have aar group on all the phones and lines. cue external mask is same as the 
sietc phones. cti ports and cti rp have aar css and aar group. I do not see the 
call go out of any of the gateways.

Anyone face similar issues.


Thanks,
DA
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Re: [OSL | CCIE_Voice] maximum sessions 0

[Ayman_labib]

Xcoder an conf require 1 dsp to configure.  You need more dsp's in order to 
configure your t1 up to 16 ports and the second dsp will be used for your conf 
and Xcoder 

Sent from my iPhone

On Oct 17, 2010, at 8:44 AM, "Tamer Ismail"  wrote:

> Hello,
> Knowing that installing any voice card on router, consuming the DSPs.
> Try to plug out the Te card and try again (just to be sure).
> 
> Tamer,
> 
> -Original Message-
> From: ccie_voice-boun...@onlinestudylist.com
> [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mohammad Dewan
> Sent: Sunday, October 17, 2010 1:50 PM
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] maximum sessions 0
> 
> hi,
> this is my first mail to the group
> 
> i have a problem with conference bridge my H/w is cisco2811 12.4.22.T
> with a PVDM2-16
> 
> nothing configured yet and i got this
> 
> Router(config-dspfarm-profile)#maximum sessions ?
>  <0-0>  Number of sessions assigned to this profile
> 
> my config is:
> Router(config-dspfarm-profile)#do sh run
> 
> hostname Router
> !
> boot-start-marker
> boot system flash:c2800nm-adventerprisek9_ivs_li-mz.124-22.T.bin
> 
> memory-size iomem 5
> no network-clock-participate wic 3
> !
> ip cef
> !
> !
> no ip domain lookup
> !
> voice-card 0
> dsp services dspfarm
> !
> !
> !
> !
> !
> controller T1 0/3/0
> !
> controller T1 0/3/1
> !
> !
> !
> !
> !
> interface FastEthernet0/0
> ip address 192.168.1.10 255.255.255.0
> duplex auto
> speed auto
> !
> interface FastEthernet0/1
> no ip address
> shutdown
> duplex auto
> speed auto
> !
> interface Serial0/0/0
> no ip address
> shutdown
> clock rate 200
> !
> !
> !
> !
> voice-port 0/2/0
> !
> voice-port 0/2/1
> !
> ccm-manager fax protocol cisco
> !
> mgcp fax t38 ecm
> mgcp behavior g729-variants static-pt
> !
> sccp local FastEthernet0/0
> sccp ccm 192.168.1.20 identifier 1 priority 1 version 7.0
> sccp
> !
> sccp ccm group 1
> associate ccm 1 priority 1
> associate profile 1 register GW_CFB
> !
> dspfarm profile 1 conference
> codec g711ulaw
> codec g711alaw
> codec g729ar8
> codec g729abr8
> codec g729r8
> codec g729br8
> shutdown
> !
> !
> !
> !
> !
> gatekeeper
> shutdown
> !
> !
> line con 0
> line aux 0
> line vty 0 4
> login
> !
> scheduler allocate 2 1000
> end
> 
> can anyone tell me what is happening with my router
> ___
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> visit www.ipexpert.com
> 
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Re: [OSL | CCIE_Voice] E1 doesn't accept more than one incoming call

[George Goglidze]

What session limit did you put on the Application trigger? It has to be
increased maybe?
CTI Group that 888 App trigger is using, should have enough CTI Ports too.

I'd go through traces in the following order:
1) Voice Gateway -> deb h225 q931
if it's Call Manager that sends "release" then step 2...
2) You might want to leave CUCM traces for the last, as probably it has
something to do with resource allocation on UCCX.
Therefore check UCCX Traces, try to look for that application trigger,
search by name/number.
It will give you a clue about what's happening.
3) On CCM make sure locations are not limiting bandwidth to only one call
between h323 gateway and UCCX App Route Point.

Hope this helps,


2010/10/17 khaled Saholy 

>
> Thanks Goerge for the reply.
>
> I haven't tried this debug of h225. I tried debug voip dialpeer all .I'll
> try it with tracing of the call manager .
>
> That number is for the Contact Center CTI port. Any incoming call will be
> answered by CUCCX  by the Application triggered by that number (888).
>
> The setup on the cm7 side , router is configured as H323 gateway.  Is there
> any other information you need to know about?
>
>
> Thanks and regards.
>
> Khaled
>
> --
> Date: Sun, 17 Oct 2010 10:41:41 +0100
> Subject: Re: [OSL | CCIE_Voice] E1 doesn't accept more than one incoming
> call
> From: gogli...@gmail.com
> To: khaled_sah...@hotmail.com
> CC: ccie_voice@onlinestudylist.com
>
>
> Hi,
>
> Have you done any debugs? The router sends the h225 setup to the CUCM or it
> just denies the call itself?
> deb h225 q931
>
> Then you might need to see the traces of the Call Manager.
> What is the 888 on the call manager? is it a hunt group? particular
> extension?
> Tell us more about setup on CallManager side...
>
> Cheers,
>
> 2010/10/17 khaled Saholy 
>
>
>
> Hi All,
>
> I have a weird problem. I configured the router voice controller E1 as one
> pri-group with H.323 the signaling protocol. The router accepts the incoming
> calls and forward them to CCM7.
> The problem is the router only accept one incoming call and it can process
> more than one outgoing call. Any other incoming call , the PSTN answers the
> call saying the number is busy at the moment.
>
> Any ideas??
>
> I'm thinking of dividing the pri-group to two groups.
>
> Thanks in advance for any help.
>
>
> Here is my config:
>
> interface Serial0/2/0:15
>  no ip address
>  encapsulation hdlc
>  isdn switch-type primary-net5
>  isdn incoming-voice voice
>  isdn send-alerting
>  isdn bchan-number-order ascending
>  isdn sending-complete
> !
> voice-port 0/2/0:15
>  cptone GB
>  connection plar 0
>  bearer-cap Speech
>
>
> num-exp 0 888  > to the contact center
> !
> dial-peer voice 1 pots
>  destination-pattern .T
>  incoming called-number .
>  direct-inward-dial ---> the customer dosn't have this service at
> the moment
>  port 0/2/0:15
> !
> dial-peer voice 12 voip
>  destination-pattern ...
>  voice-class h323 1
>  session target ipv4:192.168.77.105
>  dtmf-relay h245-alphanumeric
>  codec g711alaw
>
>
>
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> visit www.ipexpert.com
>
>
>
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[OSL | CCIE_Voice] CUE Msg Notification via CLI configurations - Lab 8 4.6

[Tam Nhu]

Hi Experts,

I am working on Vol 2 Lab 8 with has CME integrated with CUE.  I can
configure almost the requirements on this lab via CUE CLI, but cannot
configure CLI commands to turn on the Msg Notification for user to ring out
the PSTN phone when he got a new message.  I only got a portion of it, but
not complete the task without access the GUI to configure the 'cell phone
number' and the 'schedule'.  I did an output of the CLI configurations
before and after the completion via GUI, but could not find any addition CLI
commands that configured for the 'cell phone number' and 'the schedule' for
msg notifications.

Have anyone try to configure this in CLI before and have it work?  If so,
please share the configurations or shed the light.  I know that access GUI
is very quick way to complete this task via VM > Msg Notification tab, but
WHAT IF WE CANNOT ACCESS GUI FOR SOME REASON OR NO GUI ALLOW IN THE REAL
LAB.

Here is my portion of the CLI command to turn on the msg notification, but
not the whole thing to turn on the cell phone and the schedule.

voicemail notification owner scphn4 enable

voicemail notification enable

voicemail notification preference all

voicemail notification allow-login

username scphn4 phonenumberE164 "032141891"

Thanks,
TN.
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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 56, Issue 141

[Pithog Oil]

 
Hi Dewan,
 
>From your configuration i can see that your Dsp farm profile is shutdown, do a 
>no shutdown on your conference dspfarm profile.
 
Pithog oil
 
 
Message: 1
Date: Sun, 17 Oct 2010 14:44:21 +0200
From: "Tamer Ismail" 
To: "'Mohammad Dewan'" 
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice]


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Re: [OSL | CCIE_Voice] maximum sessions 0

[Prashant Patel]

I am assuming you see the PVDM in sh diag. A conference bridge uses 240 MIPS
ie a single PVDM-16. In the config you need to use

voice-card 0
no dspfarm
dsp serviced dspfarm

Also make sure none of the dsp's are being used ie in a pri or else you
would not see any resources if you have a single pvdm-16.

HTH
Prashant

On Sun, Oct 17, 2010 at 8:44 AM, Tamer Ismail  wrote:

> Hello,
> Knowing that installing any voice card on router, consuming the DSPs.
> Try to plug out the Te card and try again (just to be sure).
>
> Tamer,
>
> -Original Message-
> From: ccie_voice-boun...@onlinestudylist.com
> [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mohammad
> Dewan
> Sent: Sunday, October 17, 2010 1:50 PM
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] maximum sessions 0
>
> hi,
> this is my first mail to the group
>
> i have a problem with conference bridge my H/w is cisco2811 12.4.22.T
> with a PVDM2-16
>
> nothing configured yet and i got this
>
> Router(config-dspfarm-profile)#maximum sessions ?
>  <0-0>  Number of sessions assigned to this profile
>
> my config is:
> Router(config-dspfarm-profile)#do sh run
>
> hostname Router
> !
> boot-start-marker
> boot system flash:c2800nm-adventerprisek9_ivs_li-mz.124-22.T.bin
>
> memory-size iomem 5
> no network-clock-participate wic 3
> !
> ip cef
> !
> !
> no ip domain lookup
> !
> voice-card 0
>  dsp services dspfarm
> !
> !
> !
> !
> !
> controller T1 0/3/0
> !
> controller T1 0/3/1
> !
> !
> !
> !
> !
> interface FastEthernet0/0
>  ip address 192.168.1.10 255.255.255.0
>  duplex auto
>  speed auto
> !
> interface FastEthernet0/1
>  no ip address
>  shutdown
>  duplex auto
>  speed auto
> !
> interface Serial0/0/0
>  no ip address
>  shutdown
>  clock rate 200
> !
> !
> !
> !
> voice-port 0/2/0
> !
> voice-port 0/2/1
> !
> ccm-manager fax protocol cisco
> !
> mgcp fax t38 ecm
> mgcp behavior g729-variants static-pt
> !
> sccp local FastEthernet0/0
> sccp ccm 192.168.1.20 identifier 1 priority 1 version 7.0
> sccp
> !
> sccp ccm group 1
>  associate ccm 1 priority 1
>  associate profile 1 register GW_CFB
> !
> dspfarm profile 1 conference
>  codec g711ulaw
>  codec g711alaw
>  codec g729ar8
>  codec g729abr8
>  codec g729r8
>  codec g729br8
>  shutdown
> !
> !
> !
> !
> !
> gatekeeper
>  shutdown
> !
> !
> line con 0
> line aux 0
> line vty 0 4
>  login
> !
> scheduler allocate 2 1000
> end
>
> can anyone tell me what is happening with my router
> ___
> 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
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Vol2 Lab7 cue aar

[David A]

Hi Roger,

I have the EPNM on the CTI ports set to the CTI-RP/VM pilot ie 3600 and EPMN
is 02077353600. I have no idea why the call doesnt pass thru to the gateway.

Thanks,
DA




2010/10/17 Roger Källberg 

>  You need to set the EPNM on the CTI ports to point to the number of the
> CTI RP for CUE. This is since the call can not go directly to the CTI ports,
> it has to first be sent to the CTI RP, then on to the CTI port.
>
> Sincerely
>
>  *Roger Källberg*
> CCIE #26199 (Voice)
> Consultant
> Cygate AB
>  --
> *Från:* David A [david.a...@gmail.com]
> *Skickat:* den 16 oktober 2010 19:18
> *Till:* ccie_voice@onlinestudylist.com
> *Ämne:* [OSL | CCIE_Voice] Vol2 Lab7 cue aar
>
>Hi all,
>
>
> I always get a busy when I configure AAR for cue and dial from HQ or SiteB.
>
>
> I have aar group on all the phones and lines. cue external mask is same as
> the sietc phones. cti ports and cti rp have aar css and aar group. I do not
> see the call go out of any of the gateways.
>
> Anyone face similar issues.
>
>
> Thanks,
> DA
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

[OSL | CCIE_Voice] Lab 8, Q2.3

[Paul Dardinski]

Kind of stuck here and really cannot find the issue. Calls out to BR2
work fine, 323 to the cube, sip to br2. (Task 2.2). 

 

However, calls to UCM are getting reorder. I don't see the incoming
calls on CDRs. Running ccapi and 245 debugs seems to point to a q.850,
cause 57. The dialp inout shows correct ingress and egress on the peers.
The ICT on UCM has inbound faststart enabled.

 

Any ideas where to look? I've been all over the PG and I'm sure I'm
compliant with their solution, but getting frustrated : )

 

Paul Dardinski (RS/Sec #16842)

___
For more information regarding industry leading CCIE Lab training, please visit 
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[OSL | CCIE_Voice] Website saying LAB not Scheduled

[Stern, Larry]

Could someone from Proctor Labs please get back to me, I gave up so much to LAB 
today including tickets for a New York Giants Football Game. If someone out 
there has the Tech Support Hotline Number that would be great, it is notg 
posted on the Web Site, live Chat is only available apparently only when you 
are in a LAB.

My receipt info for Today is Below.





Help

I have a Proctor LABS Voice Lab scheduled today 10/17 at 8:00AM and it is 
telling me no LAB scheduled.
Here is my receipt

56434521101017800E89D9 Voice - Oct 17, 2010 at 08:00EST 
Order#: pli110983572

What is the Technical support number for Proctor Labs, unless you are on a LAB 
the website does not show this.


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com on behalf of 
ccie_voice-requ...@onlinestudylist.com
Sent: Sun 10/17/2010 8:36 AM
To: ccie_voice@onlinestudylist.com
Subject: CCIE_Voice Digest, Vol 56, Issue 140
 
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: E1 doesn't accept more than one incoming call
  (George Goglidze)
   2. Re: E1 doesn't accept more than one incoming call (khaled Saholy)
   3. Re: Vol2 Lab7 cue aar (Roger K?llberg)
   4. maximum sessions 0 (Mohammad Dewan)
   5. Re: CCIE_Voice Digest, Vol 56, Issue 139 (Stern, Larry)


--

Message: 1
Date: Sun, 17 Oct 2010 10:41:41 +0100
From: George Goglidze 
To: khaled Saholy 
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] E1 doesn't accept more than one
incoming call
Message-ID:

Content-Type: text/plain; charset="iso-8859-1"

Hi,

Have you done any debugs? The router sends the h225 setup to the CUCM or it
just denies the call itself?
deb h225 q931

Then you might need to see the traces of the Call Manager.
What is the 888 on the call manager? is it a hunt group? particular
extension?
Tell us more about setup on CallManager side...

Cheers,

2010/10/17 khaled Saholy 

>
>
> Hi All,
>
> I have a weird problem. I configured the router voice controller E1 as one
> pri-group with H.323 the signaling protocol. The router accepts the incoming
> calls and forward them to CCM7.
> The problem is the router only accept one incoming call and it can process
> more than one outgoing call. Any other incoming call , the PSTN answers the
> call saying the number is busy at the moment.
>
> Any ideas??
>
> I'm thinking of dividing the pri-group to two groups.
>
> Thanks in advance for any help.
>
>
> Here is my config:
>
> interface Serial0/2/0:15
>  no ip address
>  encapsulation hdlc
>  isdn switch-type primary-net5
>  isdn incoming-voice voice
>  isdn send-alerting
>  isdn bchan-number-order ascending
>  isdn sending-complete
> !
> voice-port 0/2/0:15
>  cptone GB
>  connection plar 0
>  bearer-cap Speech
>
>
> num-exp 0 888  > to the contact center
> !
> dial-peer voice 1 pots
>  destination-pattern .T
>  incoming called-number .
>  direct-inward-dial ---> the customer dosn't have this service at
> the moment
>  port 0/2/0:15
> !
> dial-peer voice 12 voip
>  destination-pattern ...
>  voice-class h323 1
>  session target ipv4:192.168.77.105
>  dtmf-relay h245-alphanumeric
>  codec g711alaw
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
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--

Message: 2
Date: Sun, 17 Oct 2010 13:27:27 +0300
From: khaled Saholy 
To: 
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] E1 doesn't accept more than one
incoming call
Message-ID: 
Content-Type: text/plain; charset="windows-1256"


 
Thanks Goerge for the reply.
 
I haven't tried this debug of h225. I tried debug voip dialpeer all .I'll try 
it with tracing of the call manager .
 
That number is for the Contact Center CTI port. Any incoming call will be 
answered by CUCCX  by the Application triggered by that number (888).
 
The setup on the cm7 side , router is configured as H323 gateway.  Is there any 
other information you need to know about?
 
 
Thanks and regards.
 
Khaled
 


Date: Sun, 17 Oct 2010 10:41:41 +0100
Subject: Re: [OSL | CCIE_Voice] E1 doesn't accept more than one incoming call
From: gogli...@gmail.com
To: khaled_sah...@hotmail.com
CC: ccie_voice@onlinestudylist.com

Hi,


Have you done any debugs? The router sends the h225 setup to the CUCM or it 
jus

Re: [OSL | CCIE_Voice] maximum sessions 0

[Tamer Ismail]

Hello,
Knowing that installing any voice card on router, consuming the DSPs.
Try to plug out the Te card and try again (just to be sure).

Tamer,

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mohammad Dewan
Sent: Sunday, October 17, 2010 1:50 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] maximum sessions 0

hi,
this is my first mail to the group

i have a problem with conference bridge my H/w is cisco2811 12.4.22.T
with a PVDM2-16

nothing configured yet and i got this

Router(config-dspfarm-profile)#maximum sessions ?
  <0-0>  Number of sessions assigned to this profile

my config is:
Router(config-dspfarm-profile)#do sh run

hostname Router
!
boot-start-marker
boot system flash:c2800nm-adventerprisek9_ivs_li-mz.124-22.T.bin

memory-size iomem 5
no network-clock-participate wic 3
!
ip cef
!
!
no ip domain lookup
!
voice-card 0
 dsp services dspfarm
!
!
!
!
!
controller T1 0/3/0
!
controller T1 0/3/1
!
!
!
!
!
interface FastEthernet0/0
 ip address 192.168.1.10 255.255.255.0
 duplex auto
 speed auto
!
interface FastEthernet0/1
 no ip address
 shutdown
 duplex auto
 speed auto
!
interface Serial0/0/0
 no ip address
 shutdown
 clock rate 200
!
!
!
!
voice-port 0/2/0
!
voice-port 0/2/1
!
ccm-manager fax protocol cisco
!
mgcp fax t38 ecm
mgcp behavior g729-variants static-pt
!
sccp local FastEthernet0/0
sccp ccm 192.168.1.20 identifier 1 priority 1 version 7.0
sccp
!
sccp ccm group 1
 associate ccm 1 priority 1
 associate profile 1 register GW_CFB
!
dspfarm profile 1 conference
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 codec g729r8
 codec g729br8
 shutdown
!
!
!
!
!
gatekeeper
 shutdown
!
!
line con 0
line aux 0
line vty 0 4
 login
!
scheduler allocate 2 1000
end

can anyone tell me what is happening with my router
___
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

Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 56, Issue 139

[Stern, Larry]

Help

I have a Proctor LABS Voice Lab scheduled today 10/17 at 8:00AM and it is 
telling me no LAB scheduled.
Here is my receipt

56434521101017800E89D9 Voice - Oct 17, 2010 at 08:00EST  
Order#: pli110983572

What is the Technical support number for Proctor Labs, unless you are on a LAB 
the website does not show this.



-Original Message-
From: ccie_voice-boun...@onlinestudylist.com on behalf of 
ccie_voice-requ...@onlinestudylist.com
Sent: Sun 10/17/2010 4:25 AM
To: ccie_voice@onlinestudylist.com
Subject: CCIE_Voice Digest, Vol 56, Issue 139
 
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. Proctorlabs website down ? (Cristobal Priego)
   2. (no subject) (Pithog Oil)
   3. Re: Vol1, Lab 4A GK (Paul Kruger)
   4. Time Synchronization All over (Pithog Oil)
   5. Assistance with IPBlue Multilab Version (Mann Chaddha)
   6. Re: Assistance with IPBlue Multilab Version (Mann Chaddha)
   7. E1 doesn't accept more than one incoming call (khaled Saholy)


--

Message: 1
Date: Sat, 16 Oct 2010 12:46:12 -0700
From: Cristobal Priego 
To: ccie_voice@onlinestudylist.com, support 
Subject: [OSL | CCIE_Voice] Proctorlabs website down ?
Message-ID:

Content-Type: text/plain; charset="iso-8859-1"

I was in the middle of my lab and can't do anything on the proctorlab
website

i wanted to load a final config and couldn't do so

can you guys access proctorlabs ?
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--

Message: 2
Date: Sat, 16 Oct 2010 13:42:08 -0700 (PDT)
From: Pithog Oil 
To: schwab...@shaw.ca
Cc: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] (no subject)
Message-ID: <254944.39603...@web120420.mail.ne1.yahoo.com>
Content-Type: text/plain; charset="iso-8859-1"

?
?
What type of phones are you using for this scenario? the anormally you have 
just mentioned is what i get when using softphones for IPMA.
?
Pithog oil


  
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--

Message: 3
Date: Sat, 16 Oct 2010 23:06:51 +0200
From: Paul Kruger 
To: Cristobal Priego 
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Vol1, Lab 4A GK
Message-ID:

Content-Type: text/plain; charset="iso-8859-1"

Does this happen for all calls inbound on CUCM, or only 1002? It might be
that you have ext 1002 created as an auto-reg ext in no partition, and that
this is getting matched and causing the conflict?

On Sat, Oct 16, 2010 at 8:52 PM, Cristobal Priego  wrote:

> hello all
>
> i have an issue with my GK implementation
>
> i have the CCM cluster and br2 (CME) registered to a GK.
>
> when i make calls from CMM to CCME using the GK, i have no issues
>
> however when i try to call from CME to CCM. i always get "your call cannot
> be completed as dialed"
>
> I already checked the ptts assigned to the extensions
> the CSS for incoming calls on the Trunk that indeed matches the ptss of the
> extensions
> the significant digits set to 4
>
> on the GK this is what i have
>
> gatekeeper
>  zone local PL ipexpert.com 10.10.110.1
>  zone prefix PL 1... gw-priority 10 gk-trunk_2
>  zone prefix PL 1... gw-priority 9 gk-trunk_1
>  zone prefix PL 1... gw-priority 0 BR2-RTR
>  zone prefix PL 5... gw-priority 10 gk-trunk_2
>  zone prefix PL 5... gw-priority 9 gk-trunk_1
>  zone prefix PL 5... gw-priority 0 BR2-RTR
>  no shutdown
> !
>
>
> HQ-RTR(config-gk)#do sh gateke end
> GATEKEEPER ENDPOINT REGISTRATION
> 
> CallSignalAddr  Port  RASSignalAddr   Port  Zone Name Type
>  Flags
> --- - --- - - 
>  -
> 10.10.110.3 1820  10.10.110.3 61408 PLH323-GW
> H323-ID: BR2-RTR
> Voice Capacity Max.=  Avail.=  Current.= 0
> 10.10.210.101720  10.10.210.1032785 PLVOIP-GW
> H323-ID: gk-trunk_1
> Voice Capacity Max.=  Avail.=  Current.= 0
> 10.10.210.111720  10.10.210.1132780 PLVOIP-GW
> H323-ID: gk-trunk_2
> Voice Capacity Max.=  Avail.=  Current.= 0
> Total number of active registrations = 3
>
> HQ-RTR(config-gk)#do sh gateke gw
> GATEWAY TYPE PREFIX TABLE
> =
> Prefix: 3*
>   Zone PL master gateway list:
> 10.10.110.3:1820 BR2-RTR
>
> Prefix: 1#*
>   Zone PL master gateway list:

[OSL | CCIE_Voice] maximum sessions 0

[Mohammad Dewan]

hi,
this is my first mail to the group

i have a problem with conference bridge my H/w is cisco2811 12.4.22.T
with a PVDM2-16

nothing configured yet and i got this

Router(config-dspfarm-profile)#maximum sessions ?
  <0-0>  Number of sessions assigned to this profile

my config is:
Router(config-dspfarm-profile)#do sh run

hostname Router
!
boot-start-marker
boot system flash:c2800nm-adventerprisek9_ivs_li-mz.124-22.T.bin

memory-size iomem 5
no network-clock-participate wic 3
!
ip cef
!
!
no ip domain lookup
!
voice-card 0
 dsp services dspfarm
!
!
!
!
!
controller T1 0/3/0
!
controller T1 0/3/1
!
!
!
!
!
interface FastEthernet0/0
 ip address 192.168.1.10 255.255.255.0
 duplex auto
 speed auto
!
interface FastEthernet0/1
 no ip address
 shutdown
 duplex auto
 speed auto
!
interface Serial0/0/0
 no ip address
 shutdown
 clock rate 200
!
!
!
!
voice-port 0/2/0
!
voice-port 0/2/1
!
ccm-manager fax protocol cisco
!
mgcp fax t38 ecm
mgcp behavior g729-variants static-pt
!
sccp local FastEthernet0/0
sccp ccm 192.168.1.20 identifier 1 priority 1 version 7.0
sccp
!
sccp ccm group 1
 associate ccm 1 priority 1
 associate profile 1 register GW_CFB
!
dspfarm profile 1 conference
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 codec g729r8
 codec g729br8
 shutdown
!
!
!
!
!
gatekeeper
 shutdown
!
!
line con 0
line aux 0
line vty 0 4
 login
!
scheduler allocate 2 1000
end

can anyone tell me what is happening with my router
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Vol2 Lab7 cue aar

[Roger Källberg]

You need to set the EPNM on the CTI ports to point to the number of the CTI RP 
for CUE. This is since the call can not go directly to the CTI ports, it has to 
first be sent to the CTI RP, then on to the CTI port.

Sincerely

Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB

Från: David A [david.a...@gmail.com]
Skickat: den 16 oktober 2010 19:18
Till: ccie_voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] Vol2 Lab7 cue aar

Hi all,


I always get a busy when I configure AAR for cue and dial from HQ or SiteB.

I have aar group on all the phones and lines. cue external mask is same as the 
sietc phones. cti ports and cti rp have aar css and aar group. I do not see the 
call go out of any of the gateways.

Anyone face similar issues.


Thanks,
DA___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] E1 doesn't accept more than one incoming call

[khaled Saholy]

 
Thanks Goerge for the reply.
 
I haven't tried this debug of h225. I tried debug voip dialpeer all .I'll try 
it with tracing of the call manager .
 
That number is for the Contact Center CTI port. Any incoming call will be 
answered by CUCCX  by the Application triggered by that number (888).
 
The setup on the cm7 side , router is configured as H323 gateway.  Is there any 
other information you need to know about?
 
 
Thanks and regards.
 
Khaled
 


Date: Sun, 17 Oct 2010 10:41:41 +0100
Subject: Re: [OSL | CCIE_Voice] E1 doesn't accept more than one incoming call
From: gogli...@gmail.com
To: khaled_sah...@hotmail.com
CC: ccie_voice@onlinestudylist.com

Hi,


Have you done any debugs? The router sends the h225 setup to the CUCM or it 
just denies the call itself? 
deb h225 q931


Then you might need to see the traces of the Call Manager.
What is the 888 on the call manager? is it a hunt group? particular extension? 
Tell us more about setup on CallManager side...


Cheers,


2010/10/17 khaled Saholy 


 
 
Hi All,
 
I have a weird problem. I configured the router voice controller E1 as one 
pri-group with H.323 the signaling protocol. The router accepts the incoming 
calls and forward them to CCM7. 
The problem is the router only accept one incoming call and it can process more 
than one outgoing call. Any other incoming call , the PSTN answers the call 
saying the number is busy at the moment.
 
Any ideas??   
 
I'm thinking of dividing the pri-group to two groups. 
 
Thanks in advance for any help.
 
 
Here is my config:
 
interface Serial0/2/0:15
 no ip address
 encapsulation hdlc
 isdn switch-type primary-net5
 isdn incoming-voice voice
 isdn send-alerting
 isdn bchan-number-order ascending
 isdn sending-complete
!
voice-port 0/2/0:15
 cptone GB
 connection plar 0
 bearer-cap Speech
 
 
num-exp 0 888  > to the contact center
!
dial-peer voice 1 pots
 destination-pattern .T
 incoming called-number .
 direct-inward-dial ---> the customer dosn't have this service at the 
moment
 port 0/2/0:15
!
dial-peer voice 12 voip
 destination-pattern ...
 voice-class h323 1
 session target ipv4:192.168.77.105
 dtmf-relay h245-alphanumeric
 codec g711alaw
 
 

___
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

Re: [OSL | CCIE_Voice] E1 doesn't accept more than one incoming call

[George Goglidze]

Hi,

Have you done any debugs? The router sends the h225 setup to the CUCM or it
just denies the call itself?
deb h225 q931

Then you might need to see the traces of the Call Manager.
What is the 888 on the call manager? is it a hunt group? particular
extension?
Tell us more about setup on CallManager side...

Cheers,

2010/10/17 khaled Saholy 

>
>
> Hi All,
>
> I have a weird problem. I configured the router voice controller E1 as one
> pri-group with H.323 the signaling protocol. The router accepts the incoming
> calls and forward them to CCM7.
> The problem is the router only accept one incoming call and it can process
> more than one outgoing call. Any other incoming call , the PSTN answers the
> call saying the number is busy at the moment.
>
> Any ideas??
>
> I'm thinking of dividing the pri-group to two groups.
>
> Thanks in advance for any help.
>
>
> Here is my config:
>
> interface Serial0/2/0:15
>  no ip address
>  encapsulation hdlc
>  isdn switch-type primary-net5
>  isdn incoming-voice voice
>  isdn send-alerting
>  isdn bchan-number-order ascending
>  isdn sending-complete
> !
> voice-port 0/2/0:15
>  cptone GB
>  connection plar 0
>  bearer-cap Speech
>
>
> num-exp 0 888  > to the contact center
> !
> dial-peer voice 1 pots
>  destination-pattern .T
>  incoming called-number .
>  direct-inward-dial ---> the customer dosn't have this service at
> the moment
>  port 0/2/0:15
> !
> dial-peer voice 12 voip
>  destination-pattern ...
>  voice-class h323 1
>  session target ipv4:192.168.77.105
>  dtmf-relay h245-alphanumeric
>  codec g711alaw
>
>
>
> ___
> 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

[OSL | CCIE_Voice] E1 doesn't accept more than one incoming call

[khaled Saholy]

 
 
Hi All,
 
I have a weird problem. I configured the router voice controller E1 as one 
pri-group with H.323 the signaling protocol. The router accepts the incoming 
calls and forward them to CCM7. 
The problem is the router only accept one incoming call and it can process more 
than one outgoing call. Any other incoming call , the PSTN answers the call 
saying the number is busy at the moment.
 
Any ideas??   
 
I'm thinking of dividing the pri-group to two groups. 
 
Thanks in advance for any help.
 
 
Here is my config:
 
interface Serial0/2/0:15
 no ip address
 encapsulation hdlc
 isdn switch-type primary-net5
 isdn incoming-voice voice
 isdn send-alerting
 isdn bchan-number-order ascending
 isdn sending-complete
!
voice-port 0/2/0:15
 cptone GB
 connection plar 0
 bearer-cap Speech
 
 
num-exp 0 888  > to the contact center
!
dial-peer voice 1 pots
 destination-pattern .T
 incoming called-number .
 direct-inward-dial ---> the customer dosn't have this service at the 
moment
 port 0/2/0:15
!
dial-peer voice 12 voip
 destination-pattern ...
 voice-class h323 1
 session target ipv4:192.168.77.105
 dtmf-relay h245-alphanumeric
 codec g711alaw
 
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