[OSL | CCIE_Voice] voice translation-rule 1

[Anthony Nwachukwu]

Hi,I need someone to help me explain the translation-rule below.This having issues with thisvoice translation-rule 1 rule 1 /0803902\(3\)$/ /\1/ rule 2 /^3/ /0803902/ rule 3 /4/ /0803902/ rule 4 /5/ /0803902/!voice translation-rule 2 rule 1 /0803902\(3\)$/ /\1/ rule 2 /^3/ /+234803902/ rule 3 /4/ /+234803902/ rule 4 /5/ /+234803902/!voice translation-rule 5034 rule 1 /5034/ /35034/!voice translation-rule 803902 rule 1 /0803902\(3\)$/ /\1/ rule 2 /\(\)$/ /3\1/
___
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Re: [OSL | CCIE_Voice] Gatekeeper

[Justin Carney]

In your HQ GK config you have not specified the IP address to which the GK
will be bound, commented below in blue:

gatekeeper
 zone local HQ cisco.com <<< NO IP specified here, router will pick
one (see below)
 no zone subnet HQ default enable
 zone subnet HQ 10.1.5.3/32 enable
 zone subnet HQ 10.1.5.2/32 enable
 zone subnet HQ 10.1.130.1/32 enable
 no shutdown
!

If you do not specify an IP on the "zone local" command the router will
pick a specific IP - I don't recall the exact rule offhand, but it may be
the highest numbered loopback, and if none the highest physical interface.
 That default method doesn't matter since you can (ie, should) manually
assign the specific IP you want to use and not worry about how the router
will pick if you don't.


*FIX -> Assign an IP for the GK to listen on:*

gatekeeper
 zone local HQ cisco.com *10.1.110.1  *  <<< specify the desired IP
here after the domain - you can optionally specify a port after the IP, but
if you don't it will default to1719 (and will be listed in show run with
port 1719), which is fine unless the question tells you otherwise

You should probably shut/no shut your gatekeeper after this change.  It may
even require you to remove the existing zone local HQ first.  To speed up
your BR2 gateway re-registering do a "no gateway" then gateway on that
side.  When I lab I always specific the GK IP even if not called out by the
question.  I typically use the loopback0 for GK and if using a CUBE I
typically will use the voice vlan SVI - but it doesn't matter unless the
question states what to use.



For reference, the other related commands commented below:

*! BR2 side*
interface Vlan130
 ip address 10.1.130.1 255.255.255.0
 h323-gateway voip interface  <<< specifies to use this interface to
source RAS
 h323-gateway voip id HQ ipaddr 10.1.5.1 1719   <<< register to zone "HQ"
at gatekeeper with IP 10.1.5.1 on port 1719
 h323-gateway voip h323-id BR2   <<< the local gateway's h323 identifier is
"BR2"
 h323-gateway voip tech-prefix 56   <<< tell the GK that my tech-prefix (to
get to me, BR2) is 56 (on the GK you will see tech prefix 56*)

^make sure the above BR2 commands are applied on the interface you want BR2
to source, and the IP called out on the HQ side is listed on the "zone
local"


*! HQ side*
interface GigabitEthernet0/0.110
 encapsulation dot1Q 110
 ip address 10.1.110.1 255.255.255.0
 ip helper-address 10.1.5.2
 h323-gateway voip interface   <<< use this interface to source RAS
messages (when talking TO another GK, not when you ARE the GK...when you
ARE the GK that is the zone local IP indicated above).  When setting up a
CUBE on the same router as the GK then this command will determine which IP
the "CUBE" registers to the "GK" with even though they are the same router.
 Both addresses can even be the same, i believe.
 h323-gateway voip bind srcaddr 10.1.110.1   <<< use this interface to
source h.225/h.245/rtp traffic.  If your question states something like
"the phone's IP address should not be know by the "cloud gatekeeper and
should only see media from IP w.x.y.z" then this command is used to pin
media (along with media flow through on the voip dial peer) to the desired
IP.  Otherwise, this IP needs to match on the CUCM side if the gateway is
using h323 to CUCM instead of mgcp.

^make sure the above HQ commands are applied on the interface you want the
HQ router to source for CUBE, they are unnecessary (and don't do anything)
for the GK process and are needed for cube (or if the gateway is h.323 then
the voip bind src is needed to match the configured IP in CUCM)


Hope this helps...

-Justin



On Mon, Oct 7, 2013 at 8:56 PM, Josh Petro  wrote:

> Hi All,
> I have a strange issue I ran into on a lab recently. The BR2 gateway would
> not register to the HQ gatekeeper unless I changed the IP address from the
> 'voice' subnet IP to the 'data' subnet IP.
>
> The question said I could not configure the gatekeeper with Zone Prefixes,
> Aliases nor could I register any e.164 addresses with it. It also said I
> could only allow the CUCM and BR2 endpoints to register to it. That
> basically left me to use the Zone Subnet commands.
>
> Why would the BR2 gateway not register until I changed the command on the
> VLAN interface from this:
> interface Vlan130
>  ip address 10.1.130.1 255.255.255.0
>  h323-gateway voip interface
> * h323-gateway voip id HQ ipaddr 10.1.110.1 1719 G0/0.110 interface*
>  h323-gateway voip h323-id BR2
>  h323-gateway voip tech-prefix 56
>
> to this
>
> interface Vlan130
>  ip address 10.1.130.1 255.255.255.0
>  h323-gateway voip interface
> * h323-gateway voip id HQ ipaddr 10.1.5.1 1719 !gig0/0.5 interface*
>  h323-gateway voip h323-id BR2
>  h323-gateway voip tech-prefix 56
>
>
>
>
>
> Here's the config
>
> HQ
> interface GigabitEthernet0/0
>  no ip address
>  duplex auto
>  speed auto
>  media-type rj45
> !
> interface GigabitEthernet0/0.5
>  encapsulation dot1Q 5
>  ip address 10.1.5.1 255.255.255.0
> !
> i

Re: [OSL | CCIE_Voice] Gatekeeper

[Hesham Abdelkereem]

Josh,

I think the reason why you have this because you are missing the binding
under the Voice vlan interface
make h323-gateway voip bind srcaddr 10.1.130.1
*h323-gateway voip id HQ ipaddr 10.1.110.1 1719*
*
*
*
*
*also there could be routing issue so you might need to do this in all your
routers*
*
*
*router ospf 2 or 1*
*network 0.0.0.0 0.0.0.0 area 0*
*
*
*
*
*Try this and let me know and if it didnt work plz share your HQ and BR2
show run and will take it from there*
*
*
*
*
*Thanks,*
*Hesham*


On 7 October 2013 17:56, Josh Petro  wrote:

> Hi All,
> I have a strange issue I ran into on a lab recently. The BR2 gateway would
> not register to the HQ gatekeeper unless I changed the IP address from the
> 'voice' subnet IP to the 'data' subnet IP.
>
> The question said I could not configure the gatekeeper with Zone Prefixes,
> Aliases nor could I register any e.164 addresses with it. It also said I
> could only allow the CUCM and BR2 endpoints to register to it. That
> basically left me to use the Zone Subnet commands.
>
> Why would the BR2 gateway not register until I changed the command on the
> VLAN interface from this:
> interface Vlan130
>  ip address 10.1.130.1 255.255.255.0
>  h323-gateway voip interface
> * h323-gateway voip id HQ ipaddr 10.1.110.1 1719 G0/0.110 interface*
>  h323-gateway voip h323-id BR2
>  h323-gateway voip tech-prefix 56
>
> to this
>
> interface Vlan130
>  ip address 10.1.130.1 255.255.255.0
>  h323-gateway voip interface
> * h323-gateway voip id HQ ipaddr 10.1.5.1 1719 !gig0/0.5 interface*
>  h323-gateway voip h323-id BR2
>  h323-gateway voip tech-prefix 56
>
>
>
>
>
> Here's the config
>
> HQ
> interface GigabitEthernet0/0
>  no ip address
>  duplex auto
>  speed auto
>  media-type rj45
> !
> interface GigabitEthernet0/0.5
>  encapsulation dot1Q 5
>  ip address 10.1.5.1 255.255.255.0
> !
> interface GigabitEthernet0/0.10
>  encapsulation dot1Q 10
>  ip address 10.1.10.1 255.255.255.0
>  ip helper-address 10.1.5.2
> !
> interface GigabitEthernet0/0.110
>  encapsulation dot1Q 110
>  ip address 10.1.110.1 255.255.255.0
>  ip helper-address 10.1.5.2
>  h323-gateway voip interface
>  h323-gateway voip bind srcaddr 10.1.110.1
> !
> gatekeeper
>  zone local HQ cisco.com
>  no zone subnet HQ default enable
>  zone subnet HQ 10.1.5.3/32 enable
>  zone subnet HQ 10.1.5.2/32 enable
>  zone subnet HQ 10.1.130.1/32 enable
>  no shutdown
> !
> !
>
>
> BR2
> interface Vlan130
>  ip address 10.1.130.1 255.255.255.0
>  h323-gateway voip interface
>  h323-gateway voip id HQ ipaddr 10.1.5.1 1719
>  h323-gateway voip h323-id BR2
>  h323-gateway voip tech-prefix 56
> !
> dial-peer voice 855 voip
>  translation-profile outgoing SiteCode
>  destination-pattern 855
>  session target ras
>  tech-prefix 55
>  dtmf-relay h245-alphanumeric
> !
> dial-peer voice 887 voip
>  translation-profile outgoing SiteCode
>  destination-pattern 887
>  session target ras
>  tech-prefix 87
>  dtmf-relay h245-alphanumeric
> !
>
>
> ___
> 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 connection - transfer option for users.

[ramesh]

Hi Guys ,

Thanks for your inputs here.


Hi Ramcharan,

I am not fully understanding your suggestion here of using translation pattern. 
Would you be able to illustrate this with an example?


Also my VM ports are set with the external mask of  +1408202 . Do you 
guys recommend doing so?


-Ramesh





From: Ramcharan Arya 
Sent: Thu, 03 Oct 2013 06:41:54 
To: Martin Sloan 
Cc: Justin Carney , 
"ccie_voice@onlinestudylist.com" , ramesh 

Subject: Re: [OSL | CCIE_Voice] unity connection - transfer option for users.
Hi Marty,

In order to preserve Original calling party TON you have to consider existing 
route pattern should not override  so two possible ways to achieve this 
you can try to use a translation patter which evaluation prior to route 
pattern. Let us assume you prefix some additional character and create clng 
party x-formation pattern and DDI -predot what was prefix in TP and set 
appropriate plan and type. Use separate pt/css for clng party x-formation 
pattern.


Another option is using application dial-rule can also use for this.

Regards,
Ramcharan Arya CCIE # 28926 (Voice/Routing & Switching)


On Wed, Oct 2, 2013 at 12:53 PM, Martin Sloan  
wrote:

Yeah, I wasn't sure on that one either and had to test it out.  I can't 
recall what the exact requirement, if any, for calling party TON was on the 
'practice test' that I had with a similar task but I'm thinking the only way to 
properly set the calling TON would be with Xforms on the port level since it 
could be any number on the PSTN phone, even the number you're trying to dial 
out to from VM.  They'd have to be very specific Xforms though since it 
could potentially override the current dial-plan manipulations in RL and RP if 
general masks are used like 10 X's.




On Wed, Oct 2, 2013 at 1:21 PM, Justin Carney  
wrote:


I wasn't sure RDNIS would matter here but figured I would throw it out there 
anyway (as it applies when redirecting TO CUC).  It seems the unity 
service parameter mentioned earlier obviates the need to use RDNIS.



With the option you proposed on creating a new RP/RL just for this requirement 
I would just set the digit manipulation/TON on the RL to whatever you see 
inbound from that specific PSTN ANI to HQ - unless the question told you what 
the expected outbound ANI/TON should be.  Another option would be to 
compare the original PSTN number with the destination PSTN and set to local if 
same NPA, LD if different NPA, or international different country codes. 
 If it comes in unknown/unknown then send it back out that way.




On Wed, Oct 2, 2013 at 12:48 PM, Martin Sloan  
wrote:



The RDNIS shouldn't be a factor here.  I just labbed this up and there is 
no Redirecting Number IE in the ISDN messages for this scenario.  It's 
more of a straight dial from Unity.





I think the places to be checked are:

CUCM service parameter
Call Routing Path

Whatever Route Pattern -> Route List is being used needs to have the "Use 
Calling Party's External Phone Number Mask" checked and no masking being done 
below, like truncating the calling party number to 7 digits if that was part of 
the requirement for the sites local PSTN dialing.  I recommend 
partitioning out a new pattern that matches the number you're trying to dial 
and handling the digit manipulation separately from the rest of the dial plan 
to keep it conceptually simple, but not necessarily 'cleaner'.  Kind of 
along the lines of keeping AAR, CFUR, SNR separate.





As for the calling party TON on this, your guess is as good as mine.  If 
the task doesn't specifically ask to set the calling party TON and it says to 
use any line from the PSTN phone, what do you do?





Marty


On Wed, Oct 2, 2013 at 12:14 PM, Justin Carney 
 wrote:




What do you see on the voice gateway for ANI/DNIS of the two separate calls - 
inbound (PSTN to Phone (which is then CFNA to CUC)) and outbound (CUC to PSTN 
alt dest 9515)?





Take a look your gateways settings for "Redirecting Number IE Delivery" (RDNIS) 
for both inbound/outbound.  These checkboxes are adjacent to the "Display 
IE Delivery" (which is usually turned on).





To test and understand the behavior of these settings I would recommend ticking 
these boxes on/off and retrying your inbound/outbound calls in this (and other) 
scenario.  As a test try setting up a call such as PSTN > SA phone > 
CFA to a different PSTN number and look at the q931 debugs for ANI/DNIS/RDNIS.





I haven't tested this recently and not sure if it applies in your stated 
scenario but try checking the box on SA gateway for the outbound RDNIS. 
 This "should" allow CUC to send 3 IE out to the PSTN - the original ANI 
(PSTN caller), the redirecting number/RDNIS (would expect this to be ei

[OSL | CCIE_Voice] Gatekeeper

[Josh Petro]

Hi All,
I have a strange issue I ran into on a lab recently. The BR2 gateway would
not register to the HQ gatekeeper unless I changed the IP address from the
'voice' subnet IP to the 'data' subnet IP.

The question said I could not configure the gatekeeper with Zone Prefixes,
Aliases nor could I register any e.164 addresses with it. It also said I
could only allow the CUCM and BR2 endpoints to register to it. That
basically left me to use the Zone Subnet commands.

Why would the BR2 gateway not register until I changed the command on the
VLAN interface from this:
interface Vlan130
 ip address 10.1.130.1 255.255.255.0
 h323-gateway voip interface
* h323-gateway voip id HQ ipaddr 10.1.110.1 1719 G0/0.110 interface*
 h323-gateway voip h323-id BR2
 h323-gateway voip tech-prefix 56

to this

interface Vlan130
 ip address 10.1.130.1 255.255.255.0
 h323-gateway voip interface
* h323-gateway voip id HQ ipaddr 10.1.5.1 1719 !gig0/0.5 interface*
 h323-gateway voip h323-id BR2
 h323-gateway voip tech-prefix 56





Here's the config

HQ
interface GigabitEthernet0/0
 no ip address
 duplex auto
 speed auto
 media-type rj45
!
interface GigabitEthernet0/0.5
 encapsulation dot1Q 5
 ip address 10.1.5.1 255.255.255.0
!
interface GigabitEthernet0/0.10
 encapsulation dot1Q 10
 ip address 10.1.10.1 255.255.255.0
 ip helper-address 10.1.5.2
!
interface GigabitEthernet0/0.110
 encapsulation dot1Q 110
 ip address 10.1.110.1 255.255.255.0
 ip helper-address 10.1.5.2
 h323-gateway voip interface
 h323-gateway voip bind srcaddr 10.1.110.1
!
gatekeeper
 zone local HQ cisco.com
 no zone subnet HQ default enable
 zone subnet HQ 10.1.5.3/32 enable
 zone subnet HQ 10.1.5.2/32 enable
 zone subnet HQ 10.1.130.1/32 enable
 no shutdown
!
!


BR2
interface Vlan130
 ip address 10.1.130.1 255.255.255.0
 h323-gateway voip interface
 h323-gateway voip id HQ ipaddr 10.1.5.1 1719
 h323-gateway voip h323-id BR2
 h323-gateway voip tech-prefix 56
!
dial-peer voice 855 voip
 translation-profile outgoing SiteCode
 destination-pattern 855
 session target ras
 tech-prefix 55
 dtmf-relay h245-alphanumeric
!
dial-peer voice 887 voip
 translation-profile outgoing SiteCode
 destination-pattern 887
 session target ras
 tech-prefix 87
 dtmf-relay h245-alphanumeric
!
___
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] ccie written

[Karen Johnson]

cool


From: "wilson.sam...@bt.com" 
To: karen.johnson...@yahoo.ca; ramcharan.a...@gmail.com 
Cc: ccie_voice@onlinestudylist.com; networksanitytoinsan...@gmail.com 
Sent: Monday, October 7, 2013 1:24:35 PM
Subject: RE: [OSL | CCIE_Voice] ccie written



Karen,
 
Cisco believes if you pass any CCIE Written Exam, and hold a complete CCIE for 
any track, its automatically recertified for next  2 years as well as any other 
Certs one is already holding.
 
Regards
 
From:ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Karen Johnson
Sent: Monday, October 07, 2013 1:47 PM
To: Ramcharan Arya
Cc: ccie_voice@onlinestudylist.com; sanity insanity
Subject: Re: [OSL | CCIE_Voice] ccie written
 
but how CISCO verified if our skills is up to date if only re-certified one ?


 
From:Ramcharan Arya 
To: Karen Johnson  
Cc: Josh Petro ; "ccie_voice@onlinestudylist.com" 
; sanity insanity 
 
Sent: Monday, October 7, 2013 11:32:50 AM
Subject: Re: ccie written
 
Hi Karen,
When I passed my CCIE voice written exam after that my CCIE R&S automatically 
got renewed.
 
Regards,
Ramcharan Arya CCIE # 28926 ( Voice/Routing & Switching)
 
 
On Mon, Oct 7, 2013 at 12:19 PM, Karen Johnson  
wrote:
hi Arya and all,
 
when you have 2 ccie specialization, do you need to write WRITTEN exam for both 
or just one ?
 
K
 
From:Ramcharan Arya 
To: Josh Petro  
Cc: "ccie_voice@onlinestudylist.com" ; sanity 
insanity  
Sent: Wednesday, October 2, 2013 6:59:08 PM
Subject: Re: [OSL | CCIE_Voice] Presence - on hook and off hook status
 
Hi Josh,
I do not believe it is related to vmware environment. I am assuming your CUPS 
is integrated with CUCM using SIP trunk.

Can you enable SIP debug level to detail and run collect SIP debug logs ( on 
primary call processing engine i.e. Sub) and check SIP logs  why there is delay 
in status.?
Im my home lab I never had this issue it works almost instantly my CUPS client 
is installed on UCCX server.
Regards,
Ramcharan Arya CCIE # 28926 ( Voice/R&S)
 
On Wed, Oct 2, 2013 at 7:47 PM, Josh Petro  wrote:
I have a huge delay in my presence updates on my system. Im assuming 
that's from CUPS being installed in my lab vmware environment though. Try 
to go off hook or call a pstn number and let it sit for 1-2 minutes. If anyone 
knows how to fix the lag, please let me know. Im assuming its related to 
vmware. 
Josh
On Oct 1, 2013 11:20 AM, "Ovidiu Popa"  wrote:
Hi
 
You could try a reboot of the CUPS server. Worked for me a couple of times... 
 
Cheers, 
Ovidiu
 
On Tue, Oct 1, 2013 at 4:43 PM, Martin Sloan  wrote:
Hi MJ,

Is the end user assigned on the line level of the hard phone?  That assignment 
is unique per line appearance so if you make the association on the CUPC device 
it does not automatically populate to the hard phone/any other line appearance. 
 When the phone goes off-hook CUCM checks the end user assignment for that 
appearance and if there is an end user assigned it check whether that end user 
is assigned CUP licensing to decide if the publish message is sent over the 
CUPS SIP trunk.
BR,
Marty
 
On Tue, Oct 1, 2013 at 10:29 AM, sanity insanity 
 wrote:
Hello all,
>I have configured presence and both softphone and deskphone modes , IM  and 
>voicemail is working fine on the clients
>However I have a question when I lift the handset of the phone ( hard phone ) 
>that is assoicated
>with the CUPC clients . I see that the presence status does not show  "  On 
>the phone "  and does not turn yellow.
>I have tried reseting my sip trunk pointing to the presence server yet I see 
>the same issue.
>Please let me know what can be done to fix this ?  Also is this  a major issue 
>?
>-MJ
> 
>___
>For more information regarding industry leading CCIE Lab training, please 
>visit http://www.ipexpert.com/
>
>Are you a CCNP or CCIE and looking for a job? Check out 
>http://www.platinumplacement.com/
 

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

Are you a CCNP or CCIE and looking for a job? Check out 
http://www.platinumplacement.com/
 

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

Are you a CCNP or CCIE and looking for a job? Check out 
http://www.platinumplacement.com/

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

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

Are you a CCNP or CCIE and looking for a job? Check out 
http://www.platinumplacement.com/__

Re: [OSL | CCIE_Voice] ccie written

[wilson.samuel]

Karen,

Cisco believes if you pass any CCIE Written Exam, and hold a complete CCIE for 
any track, its automatically recertified for next  2 years as well as any other 
Certs one is already holding.

Regards

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Karen Johnson
Sent: Monday, October 07, 2013 1:47 PM
To: Ramcharan Arya
Cc: ccie_voice@onlinestudylist.com; sanity insanity
Subject: Re: [OSL | CCIE_Voice] ccie written

but how CISCO verified if our skills is up to date if only re-certified one ?


From: Ramcharan Arya mailto:ramcharan.a...@gmail.com>>
To: Karen Johnson mailto:karen.johnson...@yahoo.ca>>
Cc: Josh Petro mailto:josh.pe...@gmail.com>>; 
"ccie_voice@onlinestudylist.com" 
mailto:ccie_voice@onlinestudylist.com>>; sanity 
insanity 
mailto:networksanitytoinsan...@gmail.com>>
Sent: Monday, October 7, 2013 11:32:50 AM
Subject: Re: ccie written

Hi Karen,
When I passed my CCIE voice written exam after that my CCIE R&S automatically 
got renewed.

Regards,
Ramcharan Arya CCIE # 28926 ( Voice/Routing & Switching)


On Mon, Oct 7, 2013 at 12:19 PM, Karen Johnson 
mailto:karen.johnson...@yahoo.ca>> wrote:
hi Arya and all,

when you have 2 ccie specialization, do you need to write WRITTEN exam for both 
or just one ?

K

From: Ramcharan Arya mailto:ramcharan.a...@gmail.com>>
To: Josh Petro mailto:josh.pe...@gmail.com>>
Cc: "ccie_voice@onlinestudylist.com" 
mailto:ccie_voice@onlinestudylist.com>>; sanity 
insanity 
mailto:networksanitytoinsan...@gmail.com>>
Sent: Wednesday, October 2, 2013 6:59:08 PM
Subject: Re: [OSL | CCIE_Voice] Presence - on hook and off hook status

Hi Josh,
I do not believe it is related to vmware environment. I am assuming your CUPS 
is integrated with CUCM using SIP trunk.

Can you enable SIP debug level to detail and run collect SIP debug logs ( on 
primary call processing engine i.e. Sub) and check SIP logs  why there is delay 
in status.?
Im my home lab I never had this issue it works almost instantly my CUPS client 
is installed on UCCX server.
Regards,
Ramcharan Arya CCIE # 28926 ( Voice/R&S)

On Wed, Oct 2, 2013 at 7:47 PM, Josh Petro 
mailto:josh.pe...@gmail.com>> wrote:
I have a huge delay in my presence updates on my system. Im assuming 
that's from CUPS being installed in my lab vmware environment though. Try 
to go off hook or call a pstn number and let it sit for 1-2 minutes. If anyone 
knows how to fix the lag, please let me know. Im assuming its related to vmware.
Josh
On Oct 1, 2013 11:20 AM, "Ovidiu Popa" 
mailto:ovi.p...@gmail.com>> wrote:
Hi

You could try a reboot of the CUPS server. Worked for me a couple of times...

Cheers,
Ovidiu

On Tue, Oct 1, 2013 at 4:43 PM, Martin Sloan 
mailto:martinsloa...@gmail.com>> wrote:
Hi MJ,

Is the end user assigned on the line level of the hard phone?  That assignment 
is unique per line appearance so if you make the association on the CUPC device 
it does not automatically populate to the hard phone/any other line appearance. 
 When the phone goes off-hook CUCM checks the end user assignment for that 
appearance and if there is an end user assigned it check whether that end user 
is assigned CUP licensing to decide if the publish message is sent over the 
CUPS SIP trunk.
BR,
Marty

On Tue, Oct 1, 2013 at 10:29 AM, sanity insanity 
mailto:networksanitytoinsan...@gmail.com>> 
wrote:
Hello all,
I have configured presence and both softphone and deskphone modes , IM  and 
voicemail is working fine on the clients
However I have a question when I lift the handset of the phone ( hard phone ) 
that is assoicated
with the CUPC clients . I see that the presence status does not show  "  On the 
phone "  and does not turn yellow.
I have tried reseting my sip trunk pointing to the presence server yet I see 
the same issue.
Please let me know what can be done to fix this ?  Also is this  a major issue ?
-MJ

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

Are you a CCNP or CCIE and looking for a job? Check out 
http://www.platinumplacement.com/


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

Are you a CCNP or CCIE and looking for a job? Check out 
http://www.platinumplacement.com/


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

Are you a CCNP or CCIE and looking for a job? Check out 
http://www.platinumplacement.com/

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

Are you a CCNP or CCIE and looking for a job? Check out 
http://www.platinumplacement.com/


___

Re: [OSL | CCIE_Voice] ccie written

[Karen Johnson]

but how CISCO verified if our skills is up to date if only re-certified one ?


From: Ramcharan Arya 
To: Karen Johnson  
Cc: Josh Petro ; "ccie_voice@onlinestudylist.com" 
; sanity insanity 
 
Sent: Monday, October 7, 2013 11:32:50 AM
Subject: Re: ccie written



Hi Karen,

When I passed my CCIE voice written exam after that my CCIE R&S automatically 
got renewed.


Regards,
Ramcharan Arya CCIE # 28926 ( Voice/Routing & Switching)







On Mon, Oct 7, 2013 at 12:19 PM, Karen Johnson  
wrote:

hi Arya and all,
> 
>when you have 2 ccie specialization, do you need to write WRITTEN exam for 
>both or just one ?
> 
>K
>
>
>From: Ramcharan Arya 
>To: Josh Petro  
>Cc: "ccie_voice@onlinestudylist.com" ; sanity 
>insanity  
>Sent: Wednesday, October 2, 2013 6:59:08 PM
>Subject: Re: [OSL | CCIE_Voice] Presence - on hook and off hook status
>
>
>
>Hi Josh,
>
>
>I do not believe it is related to vmware environment. I am assuming your CUPS 
>is integrated with CUCM using SIP trunk.
>
>
>Can you enable SIP debug level to detail and run collect SIP debug logs ( on 
>primary call processing engine i.e. Sub) and check SIP logs  why there is 
>delay in status.?
>
>Im my home lab I never had this issue it works almost instantly my CUPS client 
>is installed on UCCX server.
>
>Regards,
>Ramcharan Arya CCIE # 28926 ( Voice/R&S)
>
>
>
>
>
>On Wed, Oct 2, 2013 at 7:47 PM, Josh Petro  wrote:
>
>I have a huge delay in my presence updates on my system. Im assuming 
>that's from CUPS being installed in my lab vmware environment though. Try 
>to go off hook or call a pstn number and let it sit for 1-2 minutes. If anyone 
>knows how to fix the lag, please let me know. Im assuming its related to 
>vmware. 
>>Josh
>>On Oct 1, 2013 11:20 AM, "Ovidiu Popa"  wrote:
>>
>>Hi
>>>
>>>You could try a reboot of the CUPS server. Worked for me a couple of 
>>>times... 
>>>
>>>
>>>Cheers, 
>>>Ovidiu
>>>
>>>
>>>
>>>On Tue, Oct 1, 2013 at 4:43 PM, Martin Sloan  wrote:
>>>
>>>Hi MJ,

Is the end user assigned on the line level of the hard phone?  That 
assignment is unique per line appearance so if you make the association on 
the CUPC device it does not automatically populate to the hard phone/any 
other line appearance.  When the phone goes off-hook CUCM checks the end 
user assignment for that appearance and if there is an end user assigned it 
check whether that end user is assigned CUP licensing to decide if the 
publish message is sent over the CUPS SIP trunk.

BR,
Marty




On Tue, Oct 1, 2013 at 10:29 AM, sanity insanity 
 wrote:

Hello all,
>
>I have configured presence and both softphone and deskphone modes , IM  
>and voicemail is working fine on the clients
>
>However I have a question when I lift the handset of the phone ( hard 
>phone ) that is assoicated
>with the CUPC clients . I see that the presence status does not show  "  
>On the phone "  and does not turn yellow.
>
>I have tried reseting my sip trunk pointing to the presence server yet I 
>see the same issue.
>
>Please let me know what can be done to fix this ?  Also is this  a major 
>issue ?
>
>-MJ
>
>___
>For more information regarding industry leading CCIE Lab training, please 
>visit http://www.ipexpert.com/
>
>Are you a CCNP or CCIE and looking for a job? Check out 
>http://www.platinumplacement.com/
>

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

Are you a CCNP or CCIE and looking for a job? Check out 
http://www.platinumplacement.com/

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

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

Re: [OSL | CCIE_Voice] ccie written

[Karen Johnson]

So I assume if you hold full CCIE RS and VOICE, in the 2 years time, you only 
need to write either one.


From: Ramcharan Arya 
To: Karen Johnson  
Cc: Josh Petro ; "ccie_voice@onlinestudylist.com" 
; sanity insanity 
 
Sent: Monday, October 7, 2013 11:32:50 AM
Subject: Re: ccie written



Hi Karen,

When I passed my CCIE voice written exam after that my CCIE R&S automatically 
got renewed.


Regards,
Ramcharan Arya CCIE # 28926 ( Voice/Routing & Switching)







On Mon, Oct 7, 2013 at 12:19 PM, Karen Johnson  
wrote:

hi Arya and all,
> 
>when you have 2 ccie specialization, do you need to write WRITTEN exam for 
>both or just one ?
> 
>K
>
>
>From: Ramcharan Arya 
>To: Josh Petro  
>Cc: "ccie_voice@onlinestudylist.com" ; sanity 
>insanity  
>Sent: Wednesday, October 2, 2013 6:59:08 PM
>Subject: Re: [OSL | CCIE_Voice] Presence - on hook and off hook status
>
>
>
>Hi Josh,
>
>
>I do not believe it is related to vmware environment. I am assuming your CUPS 
>is integrated with CUCM using SIP trunk.
>
>
>Can you enable SIP debug level to detail and run collect SIP debug logs ( on 
>primary call processing engine i.e. Sub) and check SIP logs  why there is 
>delay in status.?
>
>Im my home lab I never had this issue it works almost instantly my CUPS client 
>is installed on UCCX server.
>
>Regards,
>Ramcharan Arya CCIE # 28926 ( Voice/R&S)
>
>
>
>
>
>On Wed, Oct 2, 2013 at 7:47 PM, Josh Petro  wrote:
>
>I have a huge delay in my presence updates on my system. Im assuming 
>that's from CUPS being installed in my lab vmware environment though. Try 
>to go off hook or call a pstn number and let it sit for 1-2 minutes. If anyone 
>knows how to fix the lag, please let me know. Im assuming its related to 
>vmware. 
>>Josh
>>On Oct 1, 2013 11:20 AM, "Ovidiu Popa"  wrote:
>>
>>Hi
>>>
>>>You could try a reboot of the CUPS server. Worked for me a couple of 
>>>times... 
>>>
>>>
>>>Cheers, 
>>>Ovidiu
>>>
>>>
>>>
>>>On Tue, Oct 1, 2013 at 4:43 PM, Martin Sloan  wrote:
>>>
>>>Hi MJ,

Is the end user assigned on the line level of the hard phone?  That 
assignment is unique per line appearance so if you make the association on 
the CUPC device it does not automatically populate to the hard phone/any 
other line appearance.  When the phone goes off-hook CUCM checks the end 
user assignment for that appearance and if there is an end user assigned it 
check whether that end user is assigned CUP licensing to decide if the 
publish message is sent over the CUPS SIP trunk.

BR,
Marty




On Tue, Oct 1, 2013 at 10:29 AM, sanity insanity 
 wrote:

Hello all,
>
>I have configured presence and both softphone and deskphone modes , IM  
>and voicemail is working fine on the clients
>
>However I have a question when I lift the handset of the phone ( hard 
>phone ) that is assoicated
>with the CUPC clients . I see that the presence status does not show  "  
>On the phone "  and does not turn yellow.
>
>I have tried reseting my sip trunk pointing to the presence server yet I 
>see the same issue.
>
>Please let me know what can be done to fix this ?  Also is this  a major 
>issue ?
>
>-MJ
>
>___
>For more information regarding industry leading CCIE Lab training, please 
>visit http://www.ipexpert.com/
>
>Are you a CCNP or CCIE and looking for a job? Check out 
>http://www.platinumplacement.com/
>

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

Are you a CCNP or CCIE and looking for a job? Check out 
http://www.platinumplacement.com/

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

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

Re: [OSL | CCIE_Voice] ccie written

[Ramcharan Arya]

Hi Karen,

When I passed my CCIE voice written exam after that my CCIE R&S
automatically got renewed.


Regards,
Ramcharan Arya CCIE # 28926 ( Voice/Routing & Switching)





On Mon, Oct 7, 2013 at 12:19 PM, Karen Johnson wrote:

> hi Arya and all,
>
> when you have 2 ccie specialization, do you need to write WRITTEN exam
> for both or just one ?
>
> K
>
>   *From:* Ramcharan Arya 
> *To:* Josh Petro 
> *Cc:* "ccie_voice@onlinestudylist.com" ;
> sanity insanity 
> *Sent:* Wednesday, October 2, 2013 6:59:08 PM
> *Subject:* Re: [OSL | CCIE_Voice] Presence - on hook and off hook status
>
>   Hi Josh,
>
> I do not believe it is related to vmware environment. I am assuming your
> CUPS is integrated with CUCM using SIP trunk.
>
> Can you enable SIP debug level to detail and run collect SIP debug logs (
> on primary call processing engine i.e. Sub) and check SIP logs  why there
> is delay in status.?
>
> Im my home lab I never had this issue it works almost instantly my CUPS
> client is installed on UCCX server.
>
> Regards,
> Ramcharan Arya CCIE # 28926 ( Voice/R&S)
>
>
>
>  On Wed, Oct 2, 2013 at 7:47 PM, Josh Petro  wrote:
>
> I have a huge delay in my presence updates on my system. Im assuming
> that's from CUPS being installed in my lab vmware environment though.
> Try to go off hook or call a pstn number and let it sit for 1-2 minutes. If
> anyone knows how to fix the lag, please let me know. Im assuming its
> related to vmware.
> Josh
>  On Oct 1, 2013 11:20 AM, "Ovidiu Popa"  wrote:
>
>  Hi
>
> You could try a reboot of the CUPS server. Worked for me a couple of
> times...
>
> Cheers,
> Ovidiu
>
>
> On Tue, Oct 1, 2013 at 4:43 PM, Martin Sloan wrote:
>
>  Hi MJ,
>
> Is the end user assigned on the line level of the hard phone?  That
> assignment is unique per line appearance so if you make the association on
> the CUPC device it does not automatically populate to the hard phone/any
> other line appearance.  When the phone goes off-hook CUCM checks the end
> user assignment for that appearance and if there is an end user assigned it
> check whether that end user is assigned CUP licensing to decide if the
> publish message is sent over the CUPS SIP trunk.
>
> BR,
> Marty
>
>
>  On Tue, Oct 1, 2013 at 10:29 AM, sanity insanity <
> networksanitytoinsan...@gmail.com> wrote:
>
> Hello all,
>
> I have configured presence and both softphone and deskphone modes , IM
> and voicemail is working fine on the clients
>
> However I have a question when I lift the handset of the phone ( hard
> phone ) that is assoicated
> with the CUPC clients . I see that the presence status does not show  "
> On the phone "  and does not turn yellow.
>
> I have tried reseting my sip trunk pointing to the presence server yet I
> see the same issue.
>
> Please let me know what can be done to fix this ?  Also is this  a major
> issue ?
>
> -MJ
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit http://www.ipexpert.com/
>
> Are you a CCNP or CCIE and looking for a job? Check out
> http://www.platinumplacement.com/
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit http://www.ipexpert.com/
>
> Are you a CCNP or CCIE and looking for a job? Check out
> http://www.platinumplacement.com/
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit http://www.ipexpert.com/
>
> Are you a CCNP or CCIE and looking for a job? Check out
> http://www.platinumplacement.com/
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit http://www.ipexpert.com/
>
> Are you a CCNP or CCIE and looking for a job? Check out
> http://www.platinumplacement.com/
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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

[OSL | CCIE_Voice] ccie written

[Karen Johnson]

hi Arya and all,
 
when you have 2 ccie specialization, do you need to write WRITTEN exam for both 
or just one ?
 
K

From: Ramcharan Arya 
To: Josh Petro  
Cc: "ccie_voice@onlinestudylist.com" ; sanity 
insanity  
Sent: Wednesday, October 2, 2013 6:59:08 PM
Subject: Re: [OSL | CCIE_Voice] Presence - on hook and off hook status



Hi Josh,


I do not believe it is related to vmware environment. I am assuming your CUPS 
is integrated with CUCM using SIP trunk.


Can you enable SIP debug level to detail and run collect SIP debug logs ( on 
primary call processing engine i.e. Sub) and check SIP logs  why there is delay 
in status.?

Im my home lab I never had this issue it works almost instantly my CUPS client 
is installed on UCCX server.

Regards,
Ramcharan Arya CCIE # 28926 ( Voice/R&S)





On Wed, Oct 2, 2013 at 7:47 PM, Josh Petro  wrote:

I have a huge delay in my presence updates on my system. Im assuming 
that's from CUPS being installed in my lab vmware environment though. Try 
to go off hook or call a pstn number and let it sit for 1-2 minutes. If anyone 
knows how to fix the lag, please let me know. Im assuming its related to 
vmware. 
>Josh
>On Oct 1, 2013 11:20 AM, "Ovidiu Popa"  wrote:
>
>Hi
>>
>>You could try a reboot of the CUPS server. Worked for me a couple of times... 
>>
>>
>>Cheers, 
>>Ovidiu
>>
>>
>>
>>On Tue, Oct 1, 2013 at 4:43 PM, Martin Sloan  wrote:
>>
>>Hi MJ,
>>>
>>>Is the end user assigned on the line level of the hard phone?  That 
>>>assignment is unique per line appearance so if you make the association on 
>>>the CUPC device it does not automatically populate to the hard phone/any 
>>>other line appearance.  When the phone goes off-hook CUCM checks the end 
>>>user assignment for that appearance and if there is an end user assigned it 
>>>check whether that end user is assigned CUP licensing to decide if the 
>>>publish message is sent over the CUPS SIP trunk.
>>>
>>>BR,
>>>Marty
>>>
>>>
>>>
>>>
>>>On Tue, Oct 1, 2013 at 10:29 AM, sanity insanity 
>>> wrote:
>>>
>>>Hello all,

I have configured presence and both softphone and deskphone modes , IM  and 
voicemail is working fine on the clients

However I have a question when I lift the handset of the phone ( hard phone 
) that is assoicated
with the CUPC clients . I see that the presence status does not show  "  On 
the phone "  and does not turn yellow.

I have tried reseting my sip trunk pointing to the presence server yet I 
see the same issue.

Please let me know what can be done to fix this ?  Also is this  a major 
issue ?

-MJ

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

Are you a CCNP or CCIE and looking for a job? Check out 
http://www.platinumplacement.com/

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


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

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

[Justin Carney]

If all servers are online you should be registered to you primary server,
which in the IPexpert practice labs will be your sub.  You show this server
as "down" and are registered to the backup, which I would presume is your
pub.

Your config (from an earlier email)
mgcp call-agent 177.1.10.20 service-type mgcp version 0.1   <<< you should
be registered here (Primary)
ccm-manager redundant-host 177.1.10.10   <<< you are registered here (first
backup)
->you don't have a third cucm, or "second backup" configured so that will
always be "none" on "sho ccm"

You need to troubleshoot why the gateway is not registering to the sub
(your primary ccm).
1. is the ccm service activated and running on the sub?  (may need to
restart the service)
2. is your db replication good? (use reporting page or cli commands to check
3. to fix db you can try "utils dbreplication repair all" which is not the
most intrusive option but often can clear up issues.  beyond this lookup
cisco docs on all options/methods to check/fix db replication issues
4. can you register phones to the sub?  if no, you likely have db
replication issues.

Also, since you were able to get your gateway registered now, can you share
what you did to fix it with the study group mailer so that others may
benefit from what you learned?  I provided a lot of recommendations and it
would be helpful to understand which were helpful and applied to the error
messages you were getting.

Thanks,
Justin



On Sat, Oct 5, 2013 at 11:59 PM, Anthony Nwachukwu <
anwachu...@accesspointafrica.com> wrote:

> Looks ok now the only problem now is the primary is down should be standby
> 
>
> ** **
>
> Why
>
> CorpHQ#show ccm-manager 
>
> MGCP Domain Name: CorpHQ.ccievoice.com
>
> PriorityStatus   Host
>
> 
>
> Primary Down 177.1.10.20
>
> First BackupRegistered   177.1.10.10
>
> Second Backup   None 
>
> ** **
>
> Current active Call Manager:177.1.10.10
>
> Backhaul/Redundant link port:   2428
>
> Failover Interval:  30 seconds
>
> Keepalive Interval: 15 seconds
>
> Last keepalive sent:13:57:12 PDT Oct 5 2013 (elapsed time:
> 00:00:01)
>
> Last MGCP traffic time: 13:57:12 PDT Oct 5 2013 (elapsed time:
> 00:00:01)
>
> Last failover time: 13:57:12 PDT Oct 5 2013 from (177.1.10.20)
> 
>
> Last switchback time:   13:49:47 PDT Oct 5 2013 from (177.1.10.10)
> 
>
> Switchback mode:Immediate
>
> MGCP Fallback mode: Not Selected
>
> Last MGCP Fallback start time:  None
>
> Last MGCP Fallback end time:None
>
> MGCP Download Tones:Disabled 
>
> TFTP retry count to shut Ports: 2 
>
> ** **
>
> Backhaul Link info:
>
> Link Protocol:  TCP 
>
> Remote Port Number: 2428
>
> Remote IP Address:  177.1.10.10
>
> Current Link State: OPEN
>
> Statistics:
>
> Packets recvd:   1
>
> Recv failures:   0
>
> Packets xmitted: 1
>
> Xmit failures:   0
>
> PRI Ports being backhauled:
>
> Slot 0, VIC 0, port 0
>
> Configuration Auto-Download Information
>
> ===
>
> No configurations downloaded
>
> Current state: Waiting for commands
>
> Configuration Download statistics:
>
> Download Attempted : 6
>
>   Download Successful  : 0
>
>   Download Failed  : 6
>
>   TFTP Download Failed : 33
>
> Configuration Attempted: 0
>
>   Configuration Successful : 0
>
>   Configuration Failed(Parsing): 0
>
>   Configuration Failed(config) : 0
>
> Last config download command:
>
> ** **
>
> *From:* Justin Carney [mailto:justin.s.car...@gmail.com]
> *Sent:* 05 October 2013 20:31
> *To:* Anthony Nwachukwu
> *Cc:* Anthony Nwachukwu; ccie_voice@onlinestudylist.com, (
> ccie_voice@onlinestudylist.com)
> *Subject:* RE: [OSL | CCIE_Voice] TEI_ASSIGNED
>
> ** **
>
> You will not be able to bring up the pri until your mgcp gw registers to
> cucm.  In my very first reply I assumed it was already registered but I
> guess I should have asked.
>
> The most common reason for not registering is a mismatch on hostname.  The
> gateway will register as either "hostname" (if ip domain-name not set) or "
> hostname.domain.com (if ip domain-name domain.com).
>
> I see your hostname is "CorpHQ" and I didn't see the ip domain-name
> command in yiur posted config.  In this case make sure that in cucm you
> list the mgcp gw as "CorpHQ" exactly (I always type in the correct/matching
> case, I'm not sure if it is actually case sensitive but why take that
> chance?)
>
> Use "sh

Re: [OSL | CCIE_Voice] MVA confusion and quesiton

[Bill Lake]

On Number 2 you should remember in 7.0.x there is a bug with partial match and 
you might not want to risk hitting it

Bill

Sent from my iPhone

On Oct 7, 2013, at 7:43 AM, Martin Sloan  wrote:

> Hi Ramesh,
> 
> Here's some answers based on my approach to configuring MVA.
> 
> 1)  I would use the 4 digit number for my dial-peer and CUCM MVA number 
> (3300).  Since you probably already have a translation-profile in place on 
> the voice port or inbound dial-peer to chop the called number down to 4 
> digits, it makes sense to use that.
> 
> 2) I don't change the calling party number and I use 'complete match' on the 
> service parameter.  I set my remote destination to that full number (either 7 
> or 10 digits).
> 
> 3) No manipulation required, just set the remote destination to the full 
> number.
> 
> Marty
> 
> 
> On Sun, Oct 6, 2013 at 9:52 AM, ramesh  wrote:
>> 
>> Hi San,
>> 
>> Thanks for your reply.
>> 
>> 1) So you're suggestion is to use 3300 or 3033300 ?
>> 
>> 2)At the dial-peer level are you using 3300 or 3033300?  
>> 
>> 
>> I way I use it is as given below : -
>> =
>> 
>> (a) If I use 3300 at the dial-peer level  and on the callmanger as MVA 
>> number  with 525 as the calling party number  then I  am  able  to have  
>> MVA functionality .  
>> 
>> (b)  I  normally call from the pstn using 3033300 from line 525( pstn 
>> phone)  then   on my h323 gateway  I strip the called number to last 4 
>> digits and send to the callmanger .  
>> 
>> (c) On the callmanger my MVA number is 3300.
>> 
>> 
>> Are  the above steps ( a to b )  correct?
>> 
>> 
>> Regards,
>> Ramesh
>> 
>> 
>> 
>> 
>> 
>> From: san r 
>> Sent: Sun, 06 Oct 2013 13:37:52 
>> To: ramesh 
>> Subject: Re: [OSL | CCIE_Voice] MVA confusion and quesiton
>> 
>> if you're stripping number for MVA , then mostly it wont work. Should use 
>> the exactly same number in Dial peer & and CCM MVA configurations.
>> 
>> I had the same issue in lab
>> 
>> 
>> On Sat, Oct 5, 2013 at 8:12 PM, ramesh  wrote:
>>> Hello Guys,
>>> 
>>> I have the following questions for MVA.
>>> 
>>> 
>>> 1) I am following  a 4 digit  internal dial-plan for  my site B phones  and 
>>> there is a requirement that I use  3033300  ( 7 digit number )  as my MVA 
>>> number   then  can strip this 7 digit number to  the last 4 digit  number ( 
>>>  3300 ) as my MVA number ?
>>> 
>>> 2) Also  my calling number is  a 7 digit number coming from pstn  as 
>>> 525  then do I change it to 9525?
>>> 
>>> 3) If incase calling number is  a 10 digit number  then It would come into 
>>> site B as 972525 ( which is 10 digits) is manipulation required for 
>>> this or can I just use the complete match with 10 digits on the service 
>>> parameter level?
>>> 
>>> 
>>> -Ramesh Dollar
>>> 
>>> 
>>> 
>>> 
>>> Get your own FREE website, FREE domain & FREE mobile app with Company 
>>> email.  
>>> Know More >
>>> 
>>> ___
>>> 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
>> 
>> 
>> 
>> Get your own FREE website, FREE domain & FREE mobile app with Company email. 
>>  
>> Know More >
>> 
>> ___
>> 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] MVA confusion and quesiton

[Martin Sloan]

Hi Ramesh,

Here's some answers based on my approach to configuring MVA.

1)  I would use the 4 digit number for my dial-peer and CUCM MVA number
(3300).  Since you probably already have a translation-profile in place on
the voice port or inbound dial-peer to chop the called number down to 4
digits, it makes sense to use that.

2) I don't change the calling party number and I use 'complete match' on
the service parameter.  I set my remote destination to that full number
(either 7 or 10 digits).

3) No manipulation required, just set the remote destination to the full
number.

Marty


On Sun, Oct 6, 2013 at 9:52 AM, ramesh  wrote:

>
> Hi San,
>
> Thanks for your reply.
>
> 1) So you're suggestion is to use 3300 or 3033300 ?
>
> 2)At the dial-peer level are you using 3300 or 3033300?
>
>
> I way I use it is as given below : -
> =
>
> (a) If I use 3300 at the dial-peer level  and on the callmanger as MVA
> number  with 525 as the calling party number  then I  am  able  to
> have  MVA functionality .
>
> (b)  I  normally call from the pstn using 3033300 from line 525( pstn
> phone)  then   on my h323 gateway  I strip the called number to last 4
> digits and send to the callmanger .
>
> (c) On the callmanger my MVA number is 3300.
>
>
> Are  the above steps ( a to b )  correct?
>
>
> Regards,
> Ramesh
>
>
>
>
>
> From: san r 
> Sent: Sun, 06 Oct 2013 13:37:52
> To: ramesh 
> Subject: Re: [OSL | CCIE_Voice] MVA confusion and quesiton
>
> if you're stripping number for MVA , then mostly it wont work. Should use
> the exactly same number in Dial peer & and CCM MVA configurations.
>
> I had the same issue in lab
>
>
> On Sat, Oct 5, 2013 at 8:12 PM, ramesh wrote:
>
>> Hello Guys,
>>
>> I have the following questions for MVA.
>>
>>
>> 1) I am following  a 4 digit  internal dial-plan for  my site B phones
>> and there is a requirement that I use  3033300  ( 7 digit number )  as my
>> MVA number   then  can strip this 7 digit number to  the last 4 digit
>> number (  3300 ) as my MVA number ?
>>
>> 2) Also  my calling number is  a 7 digit number coming from pstn  as
>> 525  then do I change it to 9525?
>>
>> 3) If incase calling number is  a 10 digit number  then It would come
>> into site B as 972525 ( which is 10 digits) is manipulation required
>> for this or can I just use the complete match with 10 digits on the service
>> parameter level?
>>
>>
>> -Ramesh Dollar
>>
>>
>>
>>
>> 
>> Get your own *FREE* website, *FREE* domain & *FREE* mobile app with
>> Company email.
>> *Know More 
>> >*
>> ___
>> 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
>>
>
>
>
> 
> Get your own *FREE* website, *FREE* domain & *FREE* mobile app with
> Company email.
> *Know More 
> >*
>
> ___
> 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