from:"Hesham Abdelkereem"

Dear Experts,

I have 12 PRI configured as MGCP gateways and would like to replace them by
a CUBE.

Now, I would like to make Statistics/Feasability study about the number of
concurrent calls on each PRI for example today from 8am to 5PM.
Is there is anyway I can do that? That will help me in the calculation to
order the number of concurrent calls properly when I migrate into SIP.


Thanks,
Hesham
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Re: [OSL | CCIE_Voice] 7970 Phones are black dead how to recover them?

Hi Alex,

Thank you very much for your help , I'll do that and I'll let you know.

Many Thanks,
Hesham


On 3 September 2013 07:44, Alex Mendoza  wrote:

> Hi, Hesham.
>
> Connect the ip phone (black screen)  to a PoE switch, turn on debug ip
> DHCP server all to see if the ip phone are trying to get an IP address.
>
> If not, I don't know what to do, but If the phone are trying to get an IP
> address you can use a CME to put a correct firmware and bring back to live.
>
>
> regards!
>
>
> On Sep 3, 2013, at 2:28 AM, Hesham Abdelkereem 
> wrote:
>
> Dear Experts,
>
> I have couple of 7970's using them for my homelab for practicing. Some of
> the phones were frozen due to normal boot/upgrade process then went black
> and unable to recover them.
> I have used this URL
>
>
> http://greenwirecommunications.com/phone-systems/cisco-ip-phones/guide-faq-unbrick-reflash-cisco-7970g/
>
> as well as tried to reboot then press # till lights became amber then
> release # then put 123456789*0# as well as the other one 1673492850*#
>
> All that never worked at all.
>
> I have CUCM , CME , POE Switches , Laptop.
> Whats the best way to recover a phone from a black screen?
> When you connect the phone to a POE switch. I just see the headset ,
> speaker and mute button blinks in the beginning then nothing. No logo or
> anything is shown.
>
> Thank you very much in advance,
>
> Hesham
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
>
>
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Re: [OSL | CCIE_Voice] MGCP incoming calling party prefix

Glad that i was able to help ok for your info. In the future when u make
any minor changes into mgcp gateways always restart and no mgcp and mgcp
even if it a tickbox on the gateway but in h323 or sip trunk u can just
restart the h323 gateway or sip trunk are good enough

On Tuesday, September 3, 2013, aman sinha wrote:

> Hi Hesham,
>
> I had tried resetting the Gateway multiple times.
>
> Tried no mgcp and mgcp ; and ir worked.
>
> Thanks !!
>
>
> On Tue, Sep 3, 2013 at 1:00 PM, Hesham Abdelkereem <
> heshamcentr...@gmail.com  'heshamcentr...@gmail.com');>> wrote:
>
>> Hi Aman,
>>
>> After applying that prefix make sure you restart the gateway on the
>> gateway page then Telnet/SSH to the MGCP Gateway/Router itself then conf t
>> and then NO MGCP then MGCP
>>
>> See if that will work
>>
>> Thanks,
>> Hesham
>>
>>
>> On 2 September 2013 23:58, aman sinha > 'cvml', 'aman.i...@gmail.com');>
>> > wrote:
>>
>>> Hi All.
>>>
>>> Prefixing +44 in calling number is not working on MGCP gateways.
>>>  Any suggestions ?
>>>
>>> ___
>>> For more information regarding industry leading CCIE Lab training,
>>> please visit www.ipexpert.com
>>>
>>> Are you a CCNP or CCIE and looking for a job? Check out
>>> www.PlatinumPlacement.com
>>>
>>
>>
>
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Re: [OSL | CCIE_Voice] MGCP incoming calling party prefix

Hi Aman,

After applying that prefix make sure you restart the gateway on the gateway
page then Telnet/SSH to the MGCP Gateway/Router itself then conf t
and then NO MGCP then MGCP

See if that will work

Thanks,
Hesham


On 2 September 2013 23:58, aman sinha  wrote:

> Hi All.
>
> Prefixing +44 in calling number is not working on MGCP gateways.
> Any suggestions ?
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
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> www.PlatinumPlacement.com
>
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[OSL | CCIE_Voice] 7970 Phones are black dead how to recover them?

Dear Experts,

I have couple of 7970's using them for my homelab for practicing. Some of
the phones were frozen due to normal boot/upgrade process then went black
and unable to recover them.
I have used this URL

http://greenwirecommunications.com/phone-systems/cisco-ip-phones/guide-faq-unbrick-reflash-cisco-7970g/

as well as tried to reboot then press # till lights became amber then
release # then put 123456789*0# as well as the other one 1673492850*#

All that never worked at all.

I have CUCM , CME , POE Switches , Laptop.
Whats the best way to recover a phone from a black screen?
When you connect the phone to a POE switch. I just see the headset ,
speaker and mute button blinks in the beginning then nothing. No logo or
anything is shown.

Thank you very much in advance,

Hesham
___
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Re: [OSL | CCIE_Voice] ACS 2509 Issues

2013-08-26 Thread Hesham Abdelkereem

Thank you very much Sam for that and I will try it on the weekend and I'll
let you know.

Have a wonderful day,
Hesham


On 26 August 2013 13:26, Sam Wilson  wrote:

>
> Hi
>
> Try entering "no service config" in the config mode, save the config and
> reload the router
>
> Hope that help,
>
> Rwgards
> Sent from my Windows Phone
> --
> From: Hesham Abdelkereem
> Sent: 8/26/2013 1:05 AM
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] ACS 2509 Issues
>
> Dear Experts,
> I have Cisco ACS 2509 using it for labs. I have suffering couple of issues
> with it
> First when I turn on the device i get the following
> System Bootstrap, Version 11.0(10c), SOFTWARE
> Copyright (c) 1986-1996 by cisco Systems
> 2500 processor with 16384 Kbytes of main memory
> >
> >
> >
> >
> Then I hit B to boot
> then I get the following
> Restricted Rights Legend
> Use, duplication, or disclosure by the Government is
> subject to restrictions as set forth in subparagraph
> (c) of the Commercial Computer Software - Restricted
> Rights clause at FAR sec. 52.227-19 and subparagraph
> (c) (1) (ii) of the Rights in Technical Data and Computer
> Software clause at DFARS sec. 252.227-7013.
> cisco Systems, Inc.
> 170 West Tasman Drive
> San Jose, California 95134-1706
> Cisco Internetwork Operating System Software
> IOS (tm) 3000 Bootstrap Software (IGS-BOOT-R), Version 11.0(10c), RELEASE
> SOFTWARE (fc1)
> Copyright (c) 1986-1996 by cisco Systems, Inc.
> Compiled Fri 27-Dec-96 17:33 by loreilly
> Image text-base: 0x0101, data-base: 0x1000
> cisco 2509 (68030) processor (revision D) with 16384K/2048K bytes of
> memory.
> Processor board ID 01886520, with hardware revision 
> X.25 software, Version 2.0, NET2, BFE and GOSIP compliant.
> 1 Ethernet/IEEE 802.3 interface.
> 2 Serial network interfaces.
> 8 terminal lines.
> 32K bytes of non-volatile configuration memory.
> 16384K bytes of processor board System flash (Read/Write)
> Loading network-confg ... [timed out]
> Loading cisconet.cfg ... [timed out]
> Loading acserver-confg ... [timed out]
> Loading acserver.cfg ... [timed out]
>  Press RETURN to get started!
>  I always get these messages for infinity
> Loading network-confg ... [timed out]
> Loading cisconet.cfg ... [timed out]
> Loading acserver-confg ... [timed out]
> Loading acserver.cfg ... [timed out]
>  Then finally when I access any device connected to the ASYNC cable
> I am able to connect but not unable to exit the session
> I hit Ctrl + Shift + 6 + X many times by the Laptop keyboard and even by
> On-Screen keyboard and still nothing is working well.
>  I have tried to do the following workaround
> 1-I have another 2509 which has a defective ASYNC Socket , I removed all
> the Memories and stuff and put it in the other one and still get the same
> results.
>   Your help would be highly appreciated.
> Thank you very much in advance,
>
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[OSL | CCIE_Voice] ACS 2509 Issues

2013-08-25 Thread Hesham Abdelkereem

Dear Experts,
I have Cisco ACS 2509 using it for labs. I have suffering couple of issues
with it
First when I turn on the device i get the following
System Bootstrap, Version 11.0(10c), SOFTWARE
Copyright (c) 1986-1996 by cisco Systems
2500 processor with 16384 Kbytes of main memory
>
>
>
>
Then I hit B to boot
then I get the following
Restricted Rights Legend
Use, duplication, or disclosure by the Government is
subject to restrictions as set forth in subparagraph
(c) of the Commercial Computer Software - Restricted
Rights clause at FAR sec. 52.227-19 and subparagraph
(c) (1) (ii) of the Rights in Technical Data and Computer
Software clause at DFARS sec. 252.227-7013.
cisco Systems, Inc.
170 West Tasman Drive
San Jose, California 95134-1706
Cisco Internetwork Operating System Software
IOS (tm) 3000 Bootstrap Software (IGS-BOOT-R), Version 11.0(10c), RELEASE
SOFTWARE (fc1)
Copyright (c) 1986-1996 by cisco Systems, Inc.
Compiled Fri 27-Dec-96 17:33 by loreilly
Image text-base: 0x0101, data-base: 0x1000
cisco 2509 (68030) processor (revision D) with 16384K/2048K bytes of memory.
Processor board ID 01886520, with hardware revision 
X.25 software, Version 2.0, NET2, BFE and GOSIP compliant.
1 Ethernet/IEEE 802.3 interface.
2 Serial network interfaces.
8 terminal lines.
32K bytes of non-volatile configuration memory.
16384K bytes of processor board System flash (Read/Write)
Loading network-confg ... [timed out]
Loading cisconet.cfg ... [timed out]
Loading acserver-confg ... [timed out]
Loading acserver.cfg ... [timed out]
Press RETURN to get started!
 I always get these messages for infinity
Loading network-confg ... [timed out]
Loading cisconet.cfg ... [timed out]
Loading acserver-confg ... [timed out]
Loading acserver.cfg ... [timed out]
 Then finally when I access any device connected to the ASYNC cable
I am able to connect but not unable to exit the session
I hit Ctrl + Shift + 6 + X many times by the Laptop keyboard and even by
On-Screen keyboard and still nothing is working well.
 I have tried to do the following workaround
1-I have another 2509 which has a defective ASYNC Socket , I removed all
the Memories and stuff and put it in the other one and still get the same
results.
 Your help would be highly appreciated.
Thank you very much in advance,
___
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Re: [OSL | CCIE_Voice] BACD limit max 2 call

2013-08-09 Thread Hesham Abdelkereem

Hi,

Thats the way you do it to fulfil your requirement

ccm-manager music-on-hold
ephone-hunt 1 longest-idle
pilot 4500
list 4101,4102
timeout 10,10
auto logout 2 dynamic <

application
service app-b-acd
param number-of-hunt-grps 1
param second-greeting-time 40 <<
param aa-hunt1 4500
param queue-len 2
param queue-manager-debugs 1
!
service app-b-acd-aa
paramspace english index 1
paramspace english language en
paramspace english location flash:
param service-name app-b-acd
param handoff-string app-b-acd-aa
param aa-pilot 4000
param number-of-hunt-grps 1
param dial-by-extension-option 1
param second-greeting-time 32 <
param call-retry-timer 10
param max-time-call-retry 60
param max-time-vm-retry 2
param voice-mail *4001
param drop-through-option 1
param drop-through-prompt _bacd_welcome.au
!
dial-peer voice 4000 voip
service app-b-acd-aa
destination-pattern 4000
session target ipv4:142.102.66.254
incoming called-number 4000
dtmf-relay h245-alphanumeric
codec g711ulaw

On 9 August 2013 16:43, Karen Johnson  wrote:

> all,
>
>
> is there a way to limit so BACD can only accept 2 call ?
>
> i have used
> -max-conn under dial-peer
> -param queue-len under sript app-b-acd
>
> however it still play " Thanks for calling " then reject the call.
>
> Can we achieve rejecting call right away, without play "Thanks for
> calling" ?
>
> K
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
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> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] Off net to off net transfer and Conference using Verizon SIP Trunk Issues

2013-07-28 Thread Hesham Abdelkereem

Please don't talk about the LAB because the LAB is a matter of luck and not
a matter of knowledge or experience.



On 28 July 2013 20:31, Ashok Boinpally  wrote:

> Ha ha...
>
> What can we do if we hit these kind of bugs in the lab except pulling our
> hair :)
>
>
> On Monday, 29 July 2013, Hesham Abdelkereem wrote:
>
>> Hi Ashok,
>>
>> Thanks for checking out. I was running a know bug of my CUCM
>>
>>
>> http://tools.cisco.com/Support/BugToolKit/search/getBugDetails.do?method=fetchBugDetails&bugId=CSCuf24788
>>
>>
>> The fix was upgrading the CUCM to a different version.
>>
>>
>> Many Thanks,
>>
>> Hesham
>>
>>
>> On 28 July 2013 20:27, Ashok Boinpally  wrote:
>>
>>> Hi Hesham,
>>>
>>> Have you checked Geolocation configs if they are active? The mid-call
>>> features can be  controlled through Geolocation policies.
>>>
>>>
>>> On Friday, 19 July 2013, Hesham Abdelkereem wrote:
>>>
>>>> Dear Experts,
>>>>
>>>> I have an issue when trying to complete an off net to off net transfer
>>>> for
>>>> calls using Verizon SIP trunking,
>>>>
>>>> When a Deskphone calls a PSTN number call connected then transfer it to
>>>> another internal extension or Conference , I get the
>>>> message cannot complete transfer on the phone and the transfered part
>>>> of
>>>> the call fails but the original call stays on hold.
>>>>
>>>> I did the following:-
>>>>
>>>> I have got the Block offnet to offnet transfer set to False on the
>>>> CUCM, also I tried to do this
>>>>
>>>> voice service voip
>>>> sip
>>>> pass-thru content sdp
>>>>
>>>> Also let me tell you other sutff for help
>>>> All the phones are 6945 , CIPC and Jabber
>>>>
>>>> I have tested the same situation with a Polycom Conference station that
>>>> is registered to CUCM as 3rd Party SIP Endpoint everything is working
>>>> perfect transfer to another extension and make a multiple conference with
>>>> multiple PSTN numbers and internal extensions
>>>>
>>>> also other weird thing is
>>>>
>>>> If any Deskphone receive inbound PSTN call and transfer it to another
>>>> extension then it works also but not calling outbound to PSTN and transfer.
>>>>
>>>> 6945 , CIPC and Jabber clients are doing the same issues while with 3rd
>>>> Party Polycom conference everything works perfect
>>>>
>>>> I have traced in the phones and I don't get any acknowledgment it just
>>>> try invite 3 times.
>>>>
>>>>
>>>> What could be the issue?
>>>>
>>>>
>>>> Many Thanks,
>>>>
>>>> Hesham
>>>>
>>>
>>>
>>> --
>>> Ashok Kumar Boinpally.
>>>
>>
>>
>
> --
> Ashok Kumar Boinpally.
>
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Re: [OSL | CCIE_Voice] Off net to off net transfer and Conference using Verizon SIP Trunk Issues

2013-07-28 Thread Hesham Abdelkereem

Hi Ashok,

Thanks for checking out. I was running a know bug of my CUCM

http://tools.cisco.com/Support/BugToolKit/search/getBugDetails.do?method=fetchBugDetails&bugId=CSCuf24788


The fix was upgrading the CUCM to a different version.


Many Thanks,

Hesham


On 28 July 2013 20:27, Ashok Boinpally  wrote:

> Hi Hesham,
>
> Have you checked Geolocation configs if they are active? The mid-call
> features can be  controlled through Geolocation policies.
>
>
> On Friday, 19 July 2013, Hesham Abdelkereem wrote:
>
>> Dear Experts,
>>
>> I have an issue when trying to complete an off net to off net transfer
>> for
>> calls using Verizon SIP trunking,
>>
>> When a Deskphone calls a PSTN number call connected then transfer it to
>> another internal extension or Conference , I get the
>> message cannot complete transfer on the phone and the transfered part of
>> the call fails but the original call stays on hold.
>>
>> I did the following:-
>>
>> I have got the Block offnet to offnet transfer set to False on the CUCM,
>> also I tried to do this
>>
>> voice service voip
>> sip
>> pass-thru content sdp
>>
>> Also let me tell you other sutff for help
>> All the phones are 6945 , CIPC and Jabber
>>
>> I have tested the same situation with a Polycom Conference station that
>> is registered to CUCM as 3rd Party SIP Endpoint everything is working
>> perfect transfer to another extension and make a multiple conference with
>> multiple PSTN numbers and internal extensions
>>
>> also other weird thing is
>>
>> If any Deskphone receive inbound PSTN call and transfer it to another
>> extension then it works also but not calling outbound to PSTN and transfer.
>>
>> 6945 , CIPC and Jabber clients are doing the same issues while with 3rd
>> Party Polycom conference everything works perfect
>>
>> I have traced in the phones and I don't get any acknowledgment it just
>> try invite 3 times.
>>
>>
>> What could be the issue?
>>
>>
>> Many Thanks,
>>
>> Hesham
>>
>
>
> --
> Ashok Kumar Boinpally.
>
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Re: [OSL | CCIE_Voice] overlapping dialplan using voice messaging through cisco unity.

Hi Dhara,


   -

   create a new VM Profile and a new Pilot Number in CUCM. So each location
   has its own profile and pilot.



   In CUC I created a Direct Routing Rule for each Pilot (Dialed Number
   equals ). Inside the Routing Rule config I'm able to set the
   search scope.


   Refer to this link https://supportforums.cisco.com/thread/1003530



   Thanks,

   Hesham



On 25 July 2013 12:13, Dharambir kumar varma  wrote:

> Hi Team,
> i have two location uk and india.
> both are using overlapping dial plan 5xxx.they are working fine by
> using translation pattern on CUCM.
> Right now india cisco phone user 5xxx are using voice messging thorough
> unity.
> but uk 5xxx extensions users are not using.But now we need to give the
> voice messaging facilty at UK also.but problem is that boty are using
> overlapping extensions.how can we achieve them
> india to uk  using  81-5XXX
> Uk to India- using  80-5XXX
>
> please comment on this.
>  Regards,
>  Dharambir Kumar
> ___
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> visit www.ipexpert.com
>
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> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] HWIC-4ESW

Just get that model is good enough and cheap  3550-PWR-24


On 25 July 2013 11:28, Hesham Abdelkereem  wrote:

> Yes sure just make DOT1Q on port 24
> switchport mode trunk
> switchport trunk encapsulation dot1q
>
> in the Branch router like gig0/1 or whatever make a router on stick (Sub
> interfaces)
> Int gig0/0.302
> encapsulation dot1q 302
> ip address 142.102.65.254 255.255.255.0
> no shut
>
> int gig0/0.402
> encpasulation dot1q 402
> ip address 142.202.65.254 255.255.255.0
> no shut
>
>
>
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Re: [OSL | CCIE_Voice] HWIC-4ESW

Yes sure just make DOT1Q on port 24
switchport mode trunk
switchport trunk encapsulation dot1q

in the Branch router like gig0/1 or whatever make a router on stick (Sub
interfaces)
Int gig0/0.302
encapsulation dot1q 302
ip address 142.102.65.254 255.255.255.0
no shut

int gig0/0.402
encpasulation dot1q 402
ip address 142.202.65.254 255.255.255.0
no shut
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Re: [OSL | CCIE_Voice] HWIC-4ESW

Sir,
Just use 3550 24 Poe switch it cost $70 on ebay.
That HWIC will cost u at least $200 or more. I know its better for
practicing labs but its not cost effective.
You can find it on ebay.com

Thanks,
Hesham

On 25 July 2013 09:45, CCIE Voice Aspirant wrote:

> Hello list
>
> I am looking for 2 HWIC-4ESW cards for my lab, does anyone have spares I
> can buy?
>
> Thanks
> ___
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> visit www.ipexpert.com
>
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>
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Re: [OSL | CCIE_Voice] Troubleshooting Cisco IP phone 7912

2013-07-23 Thread Hesham Abdelkereem

That has many possibilities
1-Make sure you have the correct Calling Search Space configured on the
phone configuration page
2-Make sure that the phone has the correct Device Pool configured that is
pointing to the Standard Local Route Group
3-Make sure the the phone has CSS that has access to the gateway that you
will dial-out via it.

Thanks,
Hesham

On 23 July 2013 13:27, cisco 2006  wrote:

>
>
> Dear All ,
>
> I have a problem in my Cisco IP Phone 7912 . I can receive a call from the
> outside , but I cannot place a call in my phone . Can anyone help me to
> troubleshoot this problem as soon as possible , please .
>
> Best Regards,
> Israa
>
>
>
> ___
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> visit www.ipexpert.com
>
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> www.PlatinumPlacement.com
>
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[OSL | CCIE_Voice] Off net to off net transfer and Conference using Verizon SIP Trunk Issues

2013-07-19 Thread Hesham Abdelkereem

Dear Experts,

I have an issue when trying to complete an off net to off net transfer for
calls using Verizon SIP trunking,

When a Deskphone calls a PSTN number call connected then transfer it to
another internal extension or Conference , I get the
message cannot complete transfer on the phone and the transfered part of
the call fails but the original call stays on hold.

I did the following:-

I have got the Block offnet to offnet transfer set to False on the CUCM,
also I tried to do this

voice service voip
sip
pass-thru content sdp

Also let me tell you other sutff for help
All the phones are 6945 , CIPC and Jabber

I have tested the same situation with a Polycom Conference station that is
registered to CUCM as 3rd Party SIP Endpoint everything is working perfect
transfer to another extension and make a multiple conference with multiple
PSTN numbers and internal extensions

also other weird thing is

If any Deskphone receive inbound PSTN call and transfer it to another
extension then it works also but not calling outbound to PSTN and transfer.

6945 , CIPC and Jabber clients are doing the same issues while with 3rd
Party Polycom conference everything works perfect

I have traced in the phones and I don't get any acknowledgment it just try
invite 3 times.


What could be the issue?


Many Thanks,

Hesham
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Re: [OSL | CCIE_Voice] SIP Gateway with Unity Connection issues

2013-07-19 Thread Hesham Abdelkereem

Dear All,

I have solved this issue by going into the SIP TRUNK and make Calling Party
Selection :- Last Redirect Number (External). However , If CellPhone called
unity connection and then transfer by extension that has CFA to another
cell then the Caller ID shown to the last destination is the External Phone
number mask of the phone that did the CFA and not the originator. Any Idea
how to transfer the originator caller-id which is the PSTN number to the
last forwarded hop.


Many Thanks,
Hesham


On 16 July 2013 13:45, Justin McIntyre  wrote:

> So what does the diversion header get translated to when you try the call
> via UCxN?  Are you saying that the SIP profile is working when you directly
> call the forwarded phone but not when UCxN AA calls the forwarded phone.
>  Can we see the comparing  SIP traffic.  Can we see associated SIP traffic
> when you call forwarded phone and then the SIP traffic when UCxN AA makes
> the call?  I'd like to see the difference.  Based upon your diversion
> header info being set to .*@.* this should apply the change to the UCxN VM
> pilot as well depending on the length of your VM pilot etc... If your VM
> pilot is only 4 digits and not 7 then that may be the reason it works by
> calling forwarded phone directly but not UCxN AA.  Maybe Provider isn't
> seeing enough incoming digits?  Excluding all of these options you could
> also check and see if your provider allows additional authentication
> methods for calls.  Trunk groups, digest authentications etc...
>
> Thanks,
>
> Justin
>
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Re: [OSL | CCIE_Voice] SIP Gateway with Unity Connection issues

2013-07-15 Thread Hesham Abdelkereem

yes I did that to make a normal call forwarding by doing so

oice class sip-profiles 1
 request INVITE sip-header Diversion modify "" "<
sip:305...@sbcglobal.com>" where XXX is a real DID range that make
it work with me when I call phone A to Phone B while Phone B is forwarded
to cell phone but doesn't work when I call Unity AA to call Phone B while
Phone B is forwarded to a cell phone


Thanks,


On 15 July 2013 19:36, Ashok Boinpally  wrote:

> Hello,
>
> Have you tried to modify SIP header with SIP profiles on Cisco VG while
> going finally out?
>
>
> On Tuesday, 16 July 2013, Hesham Abdelkereem wrote:
>
>> Dear All,
>>
>> I have SIP Verizon and Unity Connection.
>> I setup the Unity Connection Automated Attendant to make
>> dial-by-extension feature.
>>
>> Now suppose I have extension  is forwarded to a cell 408202
>>
>> If I called from PSTN to AA number then called extension  which is
>> forwarded to cell is not working.
>>
>> I did debug ccsip messasges and the reason why is because the
>> remote-party or ANI becomes the voicemail pilot
>>
>> this exactly related to that problem
>> http://www.gossamer-threads.com/lists/cisco/voip/148095
>>
>>
>>
>> How can I fix that in the SIP header? knowing that I did a change to let
>> Phone 1 calls Phone 2 and Phone 2 is forwarded to PSTN number
>>
>> that worked with me but didn't work when I do it via unity connection.
>>
>>
>> Please give me some advice.
>>
>>
>
> --
> Ashok Kumar Boinpally.
>
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[OSL | CCIE_Voice] SIP Gateway with Unity Connection issues

2013-07-15 Thread Hesham Abdelkereem

Dear All,

I have SIP Verizon and Unity Connection.
I setup the Unity Connection Automated Attendant to make dial-by-extension
feature.

Now suppose I have extension  is forwarded to a cell 408202

If I called from PSTN to AA number then called extension  which is
forwarded to cell is not working.

I did debug ccsip messasges and the reason why is because the remote-party
or ANI becomes the voicemail pilot

this exactly related to that problem
http://www.gossamer-threads.com/lists/cisco/voip/148095



How can I fix that in the SIP header? knowing that I did a change to let
Phone 1 calls Phone 2 and Phone 2 is forwarded to PSTN number

that worked with me but didn't work when I do it via unity connection.


Please give me some advice.
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Re: [OSL | CCIE_Voice] Access list for cue traffic marking

2013-07-07 Thread Hesham Abdelkereem

Guys ,

Get the biggest relief in your life. If you see this CUE QOS question just
give it up.
No one has ever scored that CUE Switch QOS and many people tried different
things.
My only advice give it up completely and never waste ur time or energy
solving it.
That particular lab is very long and if you have 2 hours left then try to
play with it and enjoy.
Knowing that the guys who passed this lab still didn't score that question
in particular.

In order for that question to be solved that needs to be consulted to a
very knowledgable Routing and Switching . SP and Voice simultaneously even
though the Cisco grading would be different than the real realistic world.

To conclude , Never waste ur time or energy solve this stupid question
trust me.
Your passing score is 80% and this stupid question could be about 4% of the
whole test.
I know for fact that every minor mark counts in the total but its really up
to the destiny.

To me CCIE Test is no longer a test that you are real knowledgable or not.
I definitely believe 100% CCIE test is like a gambling game , Jackpot or a
roulette in LAS VEGAS.

Don't have the faith that this thing is graded fairly with a standard.

On 7 July 2013 02:25, LorenzLGRC  wrote:

> Hello,
> you can use something like this:
>
> access-list 101 permit tcp host a.b.c.d any eq 2748
> !
> class-map match-all cti-qbe
>  match access-group 101
> !
> policy-map cti-qbe
>  class cti-qbe
>  set dscp af31
>  bandwidth 20
> !
> interface Serial0/1
>  service-policy output cti-qbe
>
>
>
>
> On Sun, Jul 7, 2013 at 6:06 AM, Piyush Jain wrote:
>
>> Hi Guys,
>>
>> I am trying to understand how we can mark CUE traffic on HQ Switch to
>> implement LAN QOS.
>>
>> I have come up with the below solution.
>>
>> ip access-list extended name CUE
>>  permit tcp host 142.100.64.12 host 142.1.66.253 eq 2748
>>
>>
>> class-map match-any CUE-CLASS
>>  match access group name CUE
>>
>> policy-map CUE-POLICY
>>  class CUE-CLASS
>>   set ip dhcp CS3
>>
>> int fa 1/0/4
>>  description * CONNECTED TO SUB CUCM ***
>>  service policy input CUE-POLICY
>>
>> In above config, 142.100.64.12 is SUB CUCM, 142.1.66.253 is CUE on SC
>> router.
>> Explanation: Since we are applying service policy in incoming direction
>> on switch port connected to CUCM, so the source port number (of CUCM) can
>> be anything but destination port number (i.e for CUE) should be 2748 (JTAPI
>> port).
>>
>> Any advice or inputs are most welcome.
>>
>> Cheers !!
>> Piyush Jain
>>
>>
>> ___
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>> visit www.ipexpert.com
>>
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>> www.PlatinumPlacement.com
>>
>
>
> ___
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>
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Re: [OSL | CCIE_Voice] Can I use 1 single T1 line for voice and data at the same time?

Thank you very much Bill.
As always I witness and confess you are the man.

It's really difficult and tricky question and you given me great experience.

Many Thanks,


On 3 July 2013 14:15, Bill Lake  wrote:

> Only if the service provider slows it
>
> So an example would be in your case Verizon accepts the SIP trunk from
> their IP 10.10.10.1 and they could also accept it from 20.20.20.1 on an
> AT&T circuit
>
> What they most likely wont do is promise it will work as well
>
> Sent from my iPhone
>
> On Jul 3, 2013, at 12:55 PM, Hesham Abdelkereem 
> wrote:
>
> Bill thanks for your great participating.
> Let me ask you more challenging question
> If I have SIP from Verizon and I have Dynamic/Integrated T1 from ATT. Can
> I have the Dynamic T1 work as backup for Verizon's SIP?
> I know that if you have the same service provider you can make T1 Line can
> work as backup for SIP but I don't know if that will work out on different
> providers?
>
> Thanks,
>
>
> On 3 July 2013 10:51, Bill Lake  wrote:
>
>> You can run voice over a data t1 from most providers. This could be as a
>> sip or h323 trunk (perhaps other ways too)
>>
>> My recommendation is to get one with QoS that matches your needs
>>
>> Sent from my iPhone
>>
>> On Jul 3, 2013, at 12:09 PM, Hesham Abdelkereem 
>> wrote:
>>
>> > Dear Experts,
>> >
>> > Can I use single T1 Line from any carrier such as ATT or Verizon for
>> Voice and Data at the same time?
>> > Or it must be one dedicated for each?
>> >
>> > Thanks,
>> > Hesham
>> > ___
>> > For more information regarding industry leading CCIE Lab training,
>> please visit www.ipexpert.com
>> >
>> > Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com
>>
>
>
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Re: [OSL | CCIE_Voice] Can I use 1 single T1 line for voice and data at the same time?

Bill thanks for your great participating.
Let me ask you more challenging question
If I have SIP from Verizon and I have Dynamic/Integrated T1 from ATT. Can I
have the Dynamic T1 work as backup for Verizon's SIP?
I know that if you have the same service provider you can make T1 Line can
work as backup for SIP but I don't know if that will work out on different
providers?

Thanks,

On 3 July 2013 10:51, Bill Lake  wrote:

> You can run voice over a data t1 from most providers. This could be as a
> sip or h323 trunk (perhaps other ways too)
>
> My recommendation is to get one with QoS that matches your needs
>
> Sent from my iPhone
>
> On Jul 3, 2013, at 12:09 PM, Hesham Abdelkereem 
> wrote:
>
> > Dear Experts,
> >
> > Can I use single T1 Line from any carrier such as ATT or Verizon for
> Voice and Data at the same time?
> > Or it must be one dedicated for each?
> >
> > Thanks,
> > Hesham
> > ___
> > For more information regarding industry leading CCIE Lab training,
> please visit www.ipexpert.com
> >
> > Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
___
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Re: [OSL | CCIE_Voice] Can I use 1 single T1 line for voice and data at the same time?

Hi Experts,
that answered my questions
If the T1 is Dynamic T1 or Integrated then It can combine voice and DATA?
http://www.carrierschoice.com/what_is_a_t1.html

I have another questions.
If I have SIP Circuit from a telco (Verizon) can I have a voice T1 from ATT
to work as backup for the SIP?

Thanks,
Hesham

On 3 July 2013 10:09, Hesham Abdelkereem  wrote:

> Dear Experts,
>
> Can I use single T1 Line from any carrier such as ATT or Verizon for Voice
> and Data at the same time?
> Or it must be one dedicated for each?
>
> Thanks,
> Hesham
>
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[OSL | CCIE_Voice] Can I use 1 single T1 line for voice and data at the same time?

Dear Experts,

Can I use single T1 Line from any carrier such as ATT or Verizon for Voice
and Data at the same time?
Or it must be one dedicated for each?

Thanks,
Hesham
___
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Re: [OSL | CCIE_Voice] B-ACD Problem

Somphol that was such a great point your have raised.

Yes I agree it was playing en_bacd_invalidoption.au even when I didn't
configure it on B-ACD
I have deleted the lab but thanks for your points.

Hesham


On 2 July 2013 05:15, Somphol Boonjing  wrote:

> Hi Hesham / Khaled,
>
> Fully agreed with that Kaled on that typo.
>
> Just a few more thought, there is also possibility that this is the actual
> audio of the file flash:en_bacd_welcome.au.
>
> "You have entered an invalid option , for sales press 1 for customer
> service press 2 for dialing by extension please press 3"
>
> You can use "debug voip application script" to quickly see what audio
> files are played.  Is it only "en_bacd_welcome.au" that is played or
>  "en_bacd_invalidoption.au" is played first then followed by
> "en_bacd_welcome.au".?
>
> Another quick isolation point is at POTS dial-peer, I think a quick change
> to number other than "4000" would help isolating the issue even further.
>   My rational is to scope down the problematic area.
>
> dial-peer voice 4001 pots
>  service app-b-acd-aa
>  incoming called-number 4008
> !
>
> Then, make a test call from PSTN.   Not that there is anything obvious,
> but isolation will make it easier to focus.
>
> Regards,
> --Somphol.
>
>
>
> On Tue, Jul 2, 2013 at 9:32 PM, khaled Saholy 
> wrote:
>
>> Hi Hesham,
>>
>> here are my comments:
>>
>> -I see under the application , no service app-b-acd-a , is this typo
>> error? It shouldn't preceded with no.
>>
>> -If you're using drop through option , change the
>>   (1)  param welcome-prompt _bacd_welcome.au  >>>  param
>> drop-through-prompt _bacd_welcome.au
>>   (2)  paramspace english index 1   from 1 to 0
>>
>> -And under service app-b-acd   , change param number-of-hunt-grps 2
>> from 2 to 1
>>
>> Try these changes and let us know how it went with you.
>>
>> Regards.
>>
>> Khaled
>>
>> --
>> Date: Tue, 2 Jul 2013 04:21:09 -0700
>> Subject: Re: [OSL | CCIE_Voice] B-ACD Problem
>> From: heshamcentr...@gmail.com
>> To: khaled_sah...@hotmail.com
>> CC: ccie_voice@onlinestudylist.com
>>
>>
>> Hi Khaled ,
>>
>> Here you are below
>>
>> application
>>  no service app-b-acd-aa
>>   param voice-mail 4220
>>   paramspace english index 1
>>   param max-time-call-retry 700
>>   param service-name app-b-acd
>>   param number-of-hunt-grps 1
>>   param drop-through-option 1
>>   paramspace english language en
>>   param handoff-string app-b-acd-aa
>>   param max-time-vm-retry 2
>>   paramspace english location flash:
>>   param aa-pilot 4000
>>   param second-greeting-time 60
>>   param welcome-prompt _bacd_welcome.au
>>   param call-retry-timer 15
>>  !
>>  service app-b-acd
>>   param queue-len 15
>>   param aa-hunt1 4500
>>   param number-of-hunt-grps 2
>>   param queue-manager-debugs 1
>>  !
>>  global
>>   service alternate default
>>  !
>> !
>> dial-peer voice 4000 voip
>>  service app-b-acd-aa
>>  destination-pattern 4000
>>  session target ipv4:142.102.66.254
>>  incoming called-number 4000
>>  dtmf-relay h245-alphanumeric
>>  codec g711ulaw
>>  no vad
>> !
>> dial-peer voice 4001 pots
>>  service app-b-acd-aa
>>  incoming called-number 4000
>>
>>
>> no ephone-hunt 10 longest-idle
>> ephone-hunt 10 longest-idle
>>  pilot 4500
>>  list 4101, 4102
>>  timeout 10, 10
>> !
>>
>>
>>
>> On 2 July 2013 04:06, khaled Saholy  wrote:
>>
>> Hi Hesham,
>>
>> Can you post the config of B-ACD and ephone-hunt ? Also the output of
>> show flash: | in au
>>
>> Regards.
>>
>> Khaled
>>
>> --
>> Date: Tue, 2 Jul 2013 03:24:10 -0700
>> From: heshamcentr...@gmail.com
>> To: ccie_voice@onlinestudylist.com
>> Subject: [OSL | CCIE_Voice] B-ACD Problem
>>
>>
>> Dear Experts,
>>
>> I have configured B-ACD. I have been configuring that everyday for months.
>> Today is the first time. when i call the pilot number it says
>> "You have entered an invalid option , for sales press 1 for customer
>> service press 2 for dialing by extension please press 3"
>>
>> What could be the problem?
>>
>> Thanks,
>> Hesham
>>
>> ___ For more information
>> regarding industry leading CCIE Lab training, please visit
>> www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com
>>
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com
>>
>
>
___
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Re: [OSL | CCIE_Voice] B-ACD Problem

Hi Khaled,

Thanks a lot for your reply

yes regarding no service (I was just trying to delete it after when it
didn't work)

I got your points and I have erased the whole lab I'd like to thank you so
much for your great efforts.

Hesham


On 2 July 2013 04:32, khaled Saholy  wrote:

> Hi Hesham,
>
> here are my comments:
>
> -I see under the application , no service app-b-acd-a , is this typo
> error? It shouldn't preceded with no.
>
> -If you're using drop through option , change the
>   (1)  param welcome-prompt _bacd_welcome.au  >>>  param
> drop-through-prompt _bacd_welcome.au
>   (2)  paramspace english index 1   from 1 to 0
>
> -And under service app-b-acd   , change param number-of-hunt-grps 2   from
> 2 to 1
>
> Try these changes and let us know how it went with you.
>
> Regards.
>
> Khaled
>
> --
> Date: Tue, 2 Jul 2013 04:21:09 -0700
> Subject: Re: [OSL | CCIE_Voice] B-ACD Problem
> From: heshamcentr...@gmail.com
> To: khaled_sah...@hotmail.com
> CC: ccie_voice@onlinestudylist.com
>
>
> Hi Khaled ,
>
> Here you are below
>
> application
>  no service app-b-acd-aa
>   param voice-mail 4220
>   paramspace english index 1
>   param max-time-call-retry 700
>   param service-name app-b-acd
>   param number-of-hunt-grps 1
>   param drop-through-option 1
>   paramspace english language en
>   param handoff-string app-b-acd-aa
>   param max-time-vm-retry 2
>   paramspace english location flash:
>   param aa-pilot 4000
>   param second-greeting-time 60
>   param welcome-prompt _bacd_welcome.au
>   param call-retry-timer 15
>  !
>  service app-b-acd
>   param queue-len 15
>   param aa-hunt1 4500
>   param number-of-hunt-grps 2
>   param queue-manager-debugs 1
>  !
>  global
>   service alternate default
>  !
> !
> dial-peer voice 4000 voip
>  service app-b-acd-aa
>  destination-pattern 4000
>  session target ipv4:142.102.66.254
>  incoming called-number 4000
>  dtmf-relay h245-alphanumeric
>  codec g711ulaw
>  no vad
> !
> dial-peer voice 4001 pots
>  service app-b-acd-aa
>  incoming called-number 4000
>
>
> no ephone-hunt 10 longest-idle
> ephone-hunt 10 longest-idle
>  pilot 4500
>  list 4101, 4102
>  timeout 10, 10
> !
>
>
>
> On 2 July 2013 04:06, khaled Saholy  wrote:
>
> Hi Hesham,
>
> Can you post the config of B-ACD and ephone-hunt ? Also the output of show
> flash: | in au
>
> Regards.
>
> Khaled
>
> --
> Date: Tue, 2 Jul 2013 03:24:10 -0700
> From: heshamcentr...@gmail.com
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] B-ACD Problem
>
>
> Dear Experts,
>
> I have configured B-ACD. I have been configuring that everyday for months.
> Today is the first time. when i call the pilot number it says
> "You have entered an invalid option , for sales press 1 for customer
> service press 2 for dialing by extension please press 3"
>
> What could be the problem?
>
> Thanks,
> Hesham
>
> ___ For more information
> regarding industry leading CCIE Lab training, please visit
> www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
>
>
___
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Re: [OSL | CCIE_Voice] B-ACD Problem

Hi Khaled ,

Here you are below

application
 no service app-b-acd-aa
  param voice-mail 4220
  paramspace english index 1
  param max-time-call-retry 700
  param service-name app-b-acd
  param number-of-hunt-grps 1
  param drop-through-option 1
  paramspace english language en
  param handoff-string app-b-acd-aa
  param max-time-vm-retry 2
  paramspace english location flash:
  param aa-pilot 4000
  param second-greeting-time 60
  param welcome-prompt _bacd_welcome.au
  param call-retry-timer 15
 !
 service app-b-acd
  param queue-len 15
  param aa-hunt1 4500
  param number-of-hunt-grps 2
  param queue-manager-debugs 1
 !
 global
  service alternate default
 !
!
dial-peer voice 4000 voip
 service app-b-acd-aa
 destination-pattern 4000
 session target ipv4:142.102.66.254
 incoming called-number 4000
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad
!
dial-peer voice 4001 pots
 service app-b-acd-aa
 incoming called-number 4000

no ephone-hunt 10 longest-idle
ephone-hunt 10 longest-idle
 pilot 4500
 list 4101, 4102
 timeout 10, 10
!

On 2 July 2013 04:06, khaled Saholy  wrote:

> Hi Hesham,
>
> Can you post the config of B-ACD and ephone-hunt ? Also the output of show
> flash: | in au
>
> Regards.
>
> Khaled
>
> --
> Date: Tue, 2 Jul 2013 03:24:10 -0700
> From: heshamcentr...@gmail.com
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] B-ACD Problem
>
>
> Dear Experts,
>
> I have configured B-ACD. I have been configuring that everyday for months.
> Today is the first time. when i call the pilot number it says
> "You have entered an invalid option , for sales press 1 for customer
> service press 2 for dialing by extension please press 3"
>
> What could be the problem?
>
> Thanks,
> Hesham
>
> ___ For more information
> regarding industry leading CCIE Lab training, please visit
> www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
___
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[OSL | CCIE_Voice] B-ACD Problem

Dear Experts,

I have configured B-ACD. I have been configuring that everyday for months.
Today is the first time. when i call the pilot number it says
"You have entered an invalid option , for sales press 1 for customer
service press 2 for dialing by extension please press 3"

What could be the problem?

Thanks,
Hesham
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Re: [OSL | CCIE_Voice] Clocking for GW

2013-07-01 Thread Hesham Abdelkereem

Thanks for that great information.

I wonder should I do that for all routers R1 , R2 and R3?
Because as far as i remember it's just mentioned in the beginning of the Voice 
Gateway section not individually per each router?


Thanks,
Hesham
On Jun 30, 2013, at 11:16 PM, LorenzLGRC  wrote:

> Under your se0/0/0:15 interface add:
> Isdn layer1-protocol-emulate network
> 
> Hth
> Lorenz
> 
> Il giorno lunedì 1 luglio 2013, Karen Johnson ha scritto:
> hi folks,
>  
> when we were asked to do below :  what is the right command and verification?
>  
> Take clocking for Layer 1 from Network side.
> Your PRI clocking of layer 2 should be user side.
> tks
> K
> ___
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> visit www.ipexpert.com
> 
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> www.PlatinumPlacement.com

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Re: [OSL | CCIE_Voice] Clocking for GW

2013-07-01 Thread Hesham Abdelkereem

Hi Lorenz,

I have tried that at my home lab now and under s0/0/0:23 or 15
i don't have an option for that

also I have removed network-clock-participate 1 t1 0/0/0
not working???

any ideas??

Thanks a lot
On Jun 30, 2013, at 11:49 PM, Hesham Abdelkereem  
wrote:

> Thanks for that great information.
> 
> I wonder should I do that for all routers R1 , R2 and R3?
> Because as far as i remember it's just mentioned in the beginning of the 
> Voice Gateway section not individually per each router?
> 
> 
> Thanks,
> Hesham
> On Jun 30, 2013, at 11:16 PM, LorenzLGRC  wrote:
> 
>> Under your se0/0/0:15 interface add:
>> Isdn layer1-protocol-emulate network
>> 
>> Hth
>> Lorenz
>> 
>> Il giorno lunedì 1 luglio 2013, Karen Johnson ha scritto:
>> hi folks,
>>  
>> when we were asked to do below :  what is the right command and verification?
>>  
>> Take clocking for Layer 1 from Network side.
>> Your PRI clocking of layer 2 should be user side.
>> tks
>> K
>> ___
>> For more information regarding industry leading CCIE Lab training, please 
>> visit www.ipexpert.com
>> 
>> Are you a CCNP or CCIE and looking for a job? Check out 
>> www.PlatinumPlacement.com
> 

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[OSL | CCIE_Voice] Single Alert in RTMT for MGCP when PRI Channel is Down

2013-06-30 Thread Hesham Abdelkereem

Dear Experts,


I would like to configure a single alert when MGCP PRI Channel is down to
be sent to a specific email.

I know how to configure it by going to RTMT ---> Alert Central -->
MGCPDCHANNEL is down ---> Set Alert/Propertis

but I don't know how to make a single alert to avoid excessive e-mail?

Kindly , Please advise


Thanks,
Hesham
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Re: [OSL | CCIE_Voice] Translation-rule help

2013-06-28 Thread Hesham Abdelkereem

Regist what about if i need it for 9011T

I would like to strip 9 from 011T how can i do it?


On 28 June 2013 10:02, Regis Reis  wrote:

> Hi Hesham,
>
> You make this form:
>
> voice translation-rule 1
> rule 1 /^91\(..$\)/ /\1/
> rule 2 /^9\(..$\)/ /\1/
> rule 3 /^9\(...$\)/ /\1/
>
> Test it. I put the "$" after last digit, because I understand that you
> want match with the total digits diled.
>
> **
>
> *Regis Reis*
>
>
>   --
>  *De:* Hesham Abdelkereem 
> *Para:* "ccie_voice@onlinestudylist.com" 
> *Enviadas:* Sexta-feira, 28 de Junho de 2013 13:29
> *Assunto:* [OSL | CCIE_Voice] Translation-rule help
>
> Dear All,
>
> I would like to make a translation-rule to do the following
> remove 9 from 91[10 digits]
> remove 9 from  9[10 digits]
> remove 9 from 9[7 digits]
>
> i did it the following but was invalid
>
> voice translation-rule 1
> rule 1 /^91../ /../
> rule 2 /^9../ /../
> rule 3 /^9.../ /.../
>
> when i did it like that it didn't work
> I would like to make it strict match not like /^9/ // this will overlap
>
> Please help me whats the other way to do it.
>
>
> Thanks,
> Hesham
>
> ___
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> visit www.ipexpert.com
>
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> www.PlatinumPlacement.com
>
>
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[OSL | CCIE_Voice] Translation-rule help

2013-06-28 Thread Hesham Abdelkereem

Dear All,

I would like to make a translation-rule to do the following
remove 9 from 91[10 digits]
remove 9 from  9[10 digits]
remove 9 from 9[7 digits]

i did it the following but was invalid

voice translation-rule 1
rule 1 /^91../ /../
rule 2 /^9../ /../
rule 3 /^9.../ /.../

when i did it like that it didn't work
I would like to make it strict match not like /^9/ // this will overlap

Please help me whats the other way to do it.


Thanks,
Hesham
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Re: [OSL | CCIE_Voice] Gatekeeper is unable to reach SiteB under Call Forward UnRegister

2013-06-27 Thread Hesham Abdelkereem

Guys I got the fix,

The problem was a typo error due to my fast copy and paste

in SB router i type gateway command by default and that resulted the
following

R1#sh gatekee end
GATEKEEPER ENDPOINT REGISTRATION

CallSignalAddr  Port  RASSignalAddr   Port  Zone Name TypeFlags
--- - --- - - -
142.100.64.11   41758 142.100.64.11   32793 GKVOIP-GW
H323-ID: GK-Trunk_1
Voice Capacity Max.=  Avail.=  Current.= 0
142.100.64.12   37277 142.100.64.12   32790 GKVOIP-GW
H323-ID: GK-Trunk_2
Voice Capacity Max.=  Avail.=  Current.= 0
142.102.65.254  1720  142.102.65.254  57138 GKH323-GW
E164-ID: 3002
E164-ID: 3001
Voice Capacity Max.=  Avail.=  Current.= 0
142.102.66.254  1720  142.102.66.254  51323 GKH323-GW
H323-ID: CUCME
Voice Capacity Max.=  Avail.=  Current.= 0
Total number of active registrations = 4

R1#



so it was invalid

when i deleted the gateway from SiteB gateway it fixed the problem



Thank you very much guys
Special Thanks to Bill , Ramy and Somphol

Hesham


On 23 June 2013 04:00, Somphol Boonjing  wrote:

> Sorry, I assume wrongly that SBGW will ever take the call for "3...".
>
>  Your normal path is for both "2..." and "3..." to be pointing to
> CUCMTRUNK only.  Given that both SBGW and CUCMTRUNK are registered to the
> same zone, it would be necessary to exclude SBGW from ever getting the call
> destined to "2..." or "3...".
>
> gw-type-prefix 1#* default-technology
> zone prefix THEZONE 3... gw-priority 0 SBGW
> zone prefix THEZONE 3... gw-priority 10 CUCMTRUNK
> zone prefix THEZONE 2... gw-priority 0 SBGW
> zone prefix THEZONE 2... gw-priority 10 CUCMTRUNK
>
> Sorry for the confusion.
>
> Even if you don't have "gw-priority", when SBGW is unreachable, it should
> not cause the problem and call should be sent correctly to CUCMTRUNK.
>
> Then, it is less likely that the problem would be in the gatekeeper call
> leg, unless you use some sort of tech-prefix in addition to zone prefix.
>
> Regards,
> --Somphol
>
>
> On Sun, Jun 23, 2013 at 8:43 PM, Somphol Boonjing wrote:
>
>> Hi Hesham,
>>
>> Essentially, the gw-priority is to advise the gatekeeper to choose SBGW
>> over CUCMTRUNK.   The higher the number, the higher the priority.   Without
>> this it will distribute the call to "3XXX" to both CUCMTRUNK and SBGW in a
>> round robin fashion.
>>
>> If you give higher priority to SBGW, then call will be routed to SBGW
>> unless it is not available.
>>
>>
>> gw-type-prefix 1#* default-technology
>> zone prefix THEZONE 3... gw-priority 100 SBGW
>> zone prefix THEZONE 3... gw-priority 10 CUCMTRUNK
>>
>> I'm fairly new to gatekeeper myself, so it would be great if you can lab
>> it up and see if I am wildly off the mark.
>>
>> Regards,
>> --Somphol.
>>
>>
>>
>> On Sun, Jun 23, 2013 at 8:37 PM, Hesham Abdelkereem <
>> heshamcentr...@gmail.com> wrote:
>>
>>> Hi Somphol,
>>>
>>> HQ & SB are in the same zone
>>> and i don't understand
>>>
>>> zone prefix THEZONE 3... gw-priority 100 SBGW
>>>
>>> I think I should disregard it as they are int he same zone
>>> It's all just the CUCM Trunk and has both 2XXX and 3XXX
>>> I think that could make it work
>>>
>>> Thank you very much for ur great input
>>> I will test it and let u know
>>>
>>> Thank you very much for ur great efforts.
>>>
>>> On Jun 23, 2013, at 3:30 AM, Somphol Boonjing  wrote:
>>>
>>> Hi Hesham,
>>>
>>> If the problem is on the gatekeeper, it could be as simple as the zone
>>> prefix not configured to point to CUCM for the pattern "3..."
>>>
>>> Given that in normal situation, the zone prefix would be pointing "SBGW"
>>> either dynamically or statically.
>>>
>>> The configure with static zone prefix set would look similar to this.
>>>
>>> gatekeeeper
>>> ...
>>> ...
>>> gw-type-prefix 1#* default-technology
>>> zone prefix THEZONE 3... gw-priority 100 SBGW
>>> zone prefix THEZONE 3... gw-priority 10 CUCMTRUNK
>>> ...
>>> ...
>>>
>>> If your CUCM & SBGW happens to be in the different zones, that is a
>>> different matter.  Looking at a configuration guide for "zone prefix"
>>> command, I do

Re: [OSL | CCIE_Voice] Gatekeeper is unable to reach SiteB under Call Forward UnRegister

2013-06-23 Thread Hesham Abdelkereem

Hi Somphol,

HQ & SB are in the same zone 
and i don't understand
> zone prefix THEZONE 3... gw-priority 100 SBGW
I think I should disregard it as they are int he same zone
It's all just the CUCM Trunk and has both 2XXX and 3XXX
I think that could make it work

Thank you very much for ur great input
I will test it and let u know

Thank you very much for ur great efforts.

On Jun 23, 2013, at 3:30 AM, Somphol Boonjing  wrote:

> Hi Hesham,
> 
> If the problem is on the gatekeeper, it could be as simple as the zone prefix 
> not configured to point to CUCM for the pattern "3..."
> 
> Given that in normal situation, the zone prefix would be pointing "SBGW" 
> either dynamically or statically.
> 
> The configure with static zone prefix set would look similar to this.
> 
> gatekeeeper
> ...
> ...
> gw-type-prefix 1#* default-technology
> zone prefix THEZONE 3... gw-priority 100 SBGW
> zone prefix THEZONE 3... gw-priority 10 CUCMTRUNK
> ...
> ...
> 
> If your CUCM & SBGW happens to be in the different zones, that is a different 
> matter.  Looking at a configuration guide for "zone prefix" command, I don't 
> think it is possible for a zone prefix to point to two different local zones. 
> (See: 
> http://www.cisco.com/en/US/docs/ios/12_3/vvf_r/vrg_z1_ps1839_TSD_Products_Command_Reference_Chapter.html#wp1002271)
> 
> So, in essence, I doubt that this would work.
> 
> gatekeeeper
> ...
> ...
> gw-type-prefix 1#* default-technology
> zone prefix SBZONE 3... gw-priority 100 SBGW
> zone prefix CUCMZONE 3... gw-priority 10 CUCMTRUNK
> ...
> ...
> 
> Regards,
> --Somphol.
> 
> 
> On Sun, Jun 23, 2013 at 6:45 PM, Hesham Abdelkereem 
>  wrote:
> Hi Somphol,
> 
> Of course all your sequence of ideas definitely make sense.
> However, I did exactly all that
> I made the Route List for CFUR is very specific to HQ Gateway and not SLRG.
> and Tried to change the Inbound Calls in the trunk and changed the CSS to 
> INTERNAL and still didn't work,
> 
> yes I am looking into the debug command that will show me the gatekeeper call 
> flow.
> I have been a long time never worked with that.
> 
> Thanks for your ideas,
> 
> I will keep you and the forum posted if I got any updates,
> 
> Thanks,
> Hesham
> 
> 
> On 23 June 2013 01:40, Somphol Boonjing  wrote:
> Hi Hesham,
> 
> I have a few ideas.   I want to remove a few things out of the equation, 
> first try to set codec for all inter-region to G711.  Second, if you are 
> using Local Route Group (LRG), replace it with a more straightforward 
> settings -- i.e. point the RL directly to HQ gateway in your case for 
> relevant route pattern. We can deal with them later on once we understand 
> this case to the bone.
> 
> There are two call legs.   The first call leg is from SC PH1 to reach x3001 
> via a H323 Trunk on CUCM -- the Trunk with gatekeeper control.   The call 
> should be directed to the gatekeeper who in turn should be routing it to the 
> H323 Trunk on CUCM.   The H323 Trunk should have significant digits set to 4 
> and a CSS that can reach x3001.
> 
> Upon hitting x3001, CUCM will discover that the number is forwarded to 
> 9723033001.  Assuming that you have set the CSS for CFUR on x3001 correctly, 
> that will match a Router Pattern that route the call toward HQ Gateway.
> This is a second call leg.(If you use the LRG, at this point, the LRG for 
> the incoming H323 Trunk will cause the call to route to the wrong RG.)
> 
> Once the second call leg is established, then CUCM will tell the two parities 
> to open the RTP channel directly to each other (i.e. between the CME and the 
> HQ Gateway.)   (Well, sort of, if you have MTP required check on the H323 
> Trunk, then an MTP will be involved.)
> 
> You problem could be on either one of this.   While I believe that since you 
> can make a call from HQ PH1 to x3001 successfully, the problem may not be in 
> the 2nd leg, I don't entirely want to rule out the CSS, the Significant 
> digits as well as the fact that HQ PH1 and the incoming H323 Trunk will be 
> more than likely belong to a different Device Pool & Region.
> 
> I think "debug gatekeeper main 10" on the gatekeeper would help.
> 
> On the H323 CUCM Trunk, RTMT Real Time monitoring with "Detailed Debug" turn 
> on would help you see whether the H323 Trunk has the right CSS to reach x3001.
> 
> Hope this gives you some idea to work on this case.
> 
> Regards,
> --Somphol.  
> 
> 
> 
> 
> 
> On Sun, Jun 23, 2013 at 5:27 PM, Somphol Boonjing  wrote:
> Hi Hesham,
> 
> Thanks for the detail explanation and well thanks for sharing

[OSL | CCIE_Voice] Gatekeeper is unable to reach SiteB under Call Forward UnRegister

2013-06-22 Thread Hesham Abdelkereem

Dear Experts,


SiteC is CME and connected with HQ and SB via Gatekeeper
Gatekeeper is working excellent with HQ and SB
I am configuring Call Forward Unregister for SiteB.
SiteB has Call-Manager-Fallback mode working excellent

Now, I have configured Call Forward Unregister
in the service parameter I changed maximum hops to DN unregister is 1

I have Created a Partitions and CSS for CFUR
I forward SiteB1 and SiteB2 telephones in unregisted internal and external
to be 9723033001 with forward css CFUR-CSS

I created Route List to point to HQ Router
and create route pattern for CFUR

Now gatekeeper is reaching both HQ and SiteB in normal operaiton
when I put SiteB under call-manager-fallback mode
when I dial from HQ 3001 the CFUR works and shows the E164 number
when I dial from SiteC 3001 via gatekeeper it shows unknown number

knowing that Gatekeeper is working with SiteB under normal operation but
doesn't work with CFUR

Any Ideas,

Thanks,
Hesham
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[OSL | CCIE_Voice] Bug in Cisco Unity with Unity Xfer becareful

2013-06-17 Thread Hesham Abdelkereem

Guys,

I have tried to configure Unity Xfer at one of my customers by making
CTI Route point * on the internal-pt and I made the alerting name
voicemail and then I forward all to voicemail.

Then making the Voicemail profile with  mask.

Very tricky.

If you made the alerting name of * as voicemail

when you call *+extension that would prompt you please enter your id
followed by pound

make sure the alerting name is anything different than Voicemail for
example Unity Xfer or anything as this will overlap with your Pilot

Thanks,
Hesham
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Re: [OSL | CCIE_Voice] SIP Timers fine tuning

2013-06-17 Thread Hesham Abdelkereem

Hi Robert,

Thank you for your reply. In the CUBE Level there is an early offer forced
but in the CUCM Level in the Trunk config , I didn't check MTP Required?
Will that fix the issue if I checked MTP required and I will use the soft
MTP resource then?

Thanks,
Hesham


On 16 June 2013 20:52, Robert Thomas  wrote:

>
> You should look into Early offer and Early media. Perhaps you might need
> PRACK enabled, to cut throught the audio before the call connects. Usually
> your Telco can give the specific requirements you need.
>
>
> On Sun, Jun 16, 2013 at 8:53 AM, Hesham Abdelkereem <
> heshamcentr...@gmail.com> wrote:
>
>> Dear All,
>>
>> I have a SIP Circuit to Verizon and when I call out I hear 3 rings first
>> before the call is actually routed to the PSTN.
>>
>> Also , I have Automated Attendant and when I dial in to the AA the first
>> 3 seconds are cut from the prompt?
>>
>> Any Ideas what parameters should I change to fix that.
>>
>>
>> Thanks,
>> Hesham
>>
>> ___
>> 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
>>
>
>
>
> --
> Robert Thomas Zamora
> tho...@gmail.com +50689389544
> http://cr.linkedin.com/pub/robert-thomas-zamora/29/913/8a8
> CCNP, CCNP Voice
>
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Re: [OSL | CCIE_Voice] H323 Gateway for POTS with CUCM issue

of course

voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 h323
!
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8
!
voice class h323 1
  h225 timeout tcp establish 3
  h225 timeout setup 3

interface Vlan100
 description ***Voice Vlan***
 ip address VLAN100IP 255.255.255.0
 ip pim dense-mode
 h323-gateway voip interface
 h323-gateway voip bind srcaddr VLAN100IP

voice-port 0/1/0
 no battery-reversal
 no comfort-noise
 connection plar opx 
 caller-id enable
!
voice-port 0/1/1
 no battery-reversal
 no comfort-noise
 connection plar opx 
 caller-id enable
!
voice-port 0/1/2
 no battery-reversal
 no comfort-noise
 connection plar opx 
 caller-id enable
!
voice-port 0/1/3
 no battery-reversal
 no comfort-noise
 connection plar opx 
caller-id enable
!
dial-peer voice 1 pots
 description ** FXO pots dial-peer **
 incoming called-number .
 port 0/1/0
!
dial-peer voice 2 pots
 description ** FXO pots dial-peer **
 incoming called-number .
 port 0/1/1
!
dial-peer voice 3 pots
 description ** FXO pots dial-peer **
 incoming called-number .
 port 0/1/2
!
dial-peer voice 4 pots
 description ** FXO pots dial-peer **
 incoming called-number .
 port 0/1/3
!
dial-peer voice 100 voip
 preference 2
 destination-pattern .T
 session target ipv4:172.30.55.11
 voice-class codec 1
 voice-class h323 1
 dtmf-relay h245-signal h245-alphanumeric
 no vad
!
dial-peer voice 5 pots
 translation-profile outgoing STRIP9
 preference 1
 destination-pattern .T
 port 0/1/0
!
dial-peer voice 6 pots
 translation-profile outgoing STRIP9
 preference 1
 destination-pattern .T
 port 0/1/1
!
dial-peer voice 7 pots
 translation-profile outgoing STRIP9
 preference 1
 destination-pattern .T
 port 0/1/2
!
dial-peer voice 8 pots
 translation-profile outgoing STRIP9
 preference 1
 destination-pattern .T
 port 0/1/3
!
!
!




Thank you very much


On 16 June 2013 09:14, Bill Lake  wrote:

> Do you think sharing your config might help?
>
>
> On Sun, Jun 16, 2013 at 10:51 AM, Hesham Abdelkereem <
> heshamcentr...@gmail.com> wrote:
>
>> Dear All.
>>
>> I have configured H323 Gateway to use the 4 FXO ports on the router with
>> CUCM.
>> When I call in to the POTS line sometimes its working perfectly and
>> sometimes when I call in it give me like a faxtone and i hear no voice then
>> I drop the call and call again it works.
>> Also , There is a big delay to reach the PLAR number.
>>
>> Do you have any ideas how to fix that?
>>
>>
>> Many Thanks,
>> Hesham
>>
>> ___
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>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com
>>
>
>
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[OSL | CCIE_Voice] Best Practice to block certain pattern

Dear All,


I'd like to block 91900 pattern efficient the CUCM.
What's the best and most efficent practice to do that?

Many Thanks,

Hesham
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[OSL | CCIE_Voice] SIP Timers fine tuning

Dear All,

I have a SIP Circuit to Verizon and when I call out I hear 3 rings first
before the call is actually routed to the PSTN.

Also , I have Automated Attendant and when I dial in to the AA the first 3
seconds are cut from the prompt?

Any Ideas what parameters should I change to fix that.


Thanks,
Hesham
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[OSL | CCIE_Voice] H323 Gateway for POTS with CUCM issue

Dear All.

I have configured H323 Gateway to use the 4 FXO ports on the router with
CUCM.
When I call in to the POTS line sometimes its working perfectly and
sometimes when I call in it give me like a faxtone and i hear no voice then
I drop the call and call again it works.
Also , There is a big delay to reach the PLAR number.

Do you have any ideas how to fix that?


Many Thanks,
Hesham
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[OSL | CCIE_Voice] Dialing *Extension to reach the person voicemail is not working on v9.1

Dear Experts,

I'd like to configure on CUCM when I dial *Extension then I can reach the
voicemail of the person directly and says "Sorry Extension 1130 is not
available please record ur message"

In V7 I just make a CTI Route Point with extension * on the internal
partition then forward all to voicemail then its absoultely working.

I have CUCM and Unity connection v9.1 and when I did that it just telling
enter your pin followed by pound like I am calling the normal voicemail
pilot number
I tried to tweak the forwarding routing rule and direct routing rule but no
chance unfortunately.

Any Ideas

Thank you very much in advance
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[OSL | CCIE_Voice] Verizon SIP Trunking + SRST configs questions

2013-06-07 Thread Hesham Abdelkereem

Dear Experts,

I would like to configure 2901 Gateway as SIP Trunking with Verizon.
I have been working 8 hours with Verizon and they are barely can help or
support while when I have dealed with AT&T they have provided me SIP script
that made everything smooth.
Now the issue inbound calls hitting the gateway but nothing received on the
phones and when i make outbound the SIP message make 408 request timeout.
Kindly , If anyone has done SIP with Verizon can you provide me with your
configurations?
Also , I am using for testing now CME V9.1 with 6945 SIP Phone is there are
any concerns needs to be addressed in CME its just temporarily maybe I need
trasncoder or CFB or something or?
Also, After that I will configure remote site for SRST as SIP is peer to
peer protocol like H323 can I use CALL MANAGER FALLBACK same as H323 will
it will work?
I don't want to hassle myself for learned configuration of CME as
SRST knowing that the phones are SIP Phones Cisco 6945?

Please share all your concerns.

Thank you very much in advance.

Best Regards,
Hesham
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[OSL | CCIE_Voice] TEHO BEST PRACTICE

2013-05-30 Thread Hesham Abdelkereem

Dear Experts,

Guys have a very tricky question for you.
Suppose you are asked to call from HQ (408) to 972 TEHO
1ST you will use remote gateway SB (972) and Second you will use SLRG
Ok my question here
If I will use the Remote gateway siteb

What should I do my pattern , ANI AND DNIS manipulation?
If I call 972 numbers from HQ via SB 972 Gateway
Should I make my pattern 91972.XXX
make my ANI 408XXX NATIONAL
DNIS 7 Digits Subscriber or 10 DIGIT NATIONAL
Please let me know the best practice
Which one makes more sense
to make ANI 10 DIGIT 408XX NATIONAL
DNIS 7 DIGIT LOCAL
or ANI 10 DIGIT 408XXX NATIONAL
DNIS 10 DIGIT NATIONAL and prefix 1972

Thank you so much in advance.

Hesham
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Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced (Hesham Abdelkereem)

2013-05-29 Thread Hesham Abdelkereem

Kamran,

i totally understand that your IE# will still be valid and you just need to 
upgrade it by written test every 2 years as well as you can gain Emeritus 
status after 10 years.
All that will be fine same as CCIE Storage , old CCIE's such as CCIE Cabling 
and etc.

The thing is there  is no quite difference between Voice or Collaboration 
however its just some video stuff and SAF protocol and etc..
Why we don't have a better migration option.
Cisco always change name from CCVP to CCNP V 
CCIP to CCNP SP and bla bla bla bla
then we can have a better and cheaper migration option.
We can have special migration test from CCIE-V to Collaboration for example for 
$500.
That test will concentrate on the differences between both tracks and will 
include some simulation labs.
that would definitely make more sense rather than making the people who 
invested a lot of money to tell them your name will be old and grandfathered.
How do you look to a CCIE Cabling nowadays???
I doubt you heard of that track
that track is very old or its the first CCIE is ever released.
You maybe impressed of the person who got it because his number is very old 
could be no 200 or 700 while people nowadays has 39XXX.
However , Career and employment wise CCIE Cabling add nothing like exactly u 
didn't talk it and i bet you people who has these certification.
They changed their career and became normal network engineers in any of other 
fields such as wireless , voice , R/S and etc.

I hope you got my point

On May 28, 2013, at 11:12 PM, Kamran Ahsanullah  
wrote:

> Hesham,
> 
> if you have a Voice CCIE already or pass before Feb 2013,you will
> return your CCIE Voice. That will not be taken away from you.
> How can you expect to be awarded the CCIE Collaboration if you haven't
> passed or sat it.
> 
> 
> Kamran
> 
> On 29 May 2013 06:37,   wrote:
>> 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. keyboard model in SJ and RTP exam (Karen Johnson)
>>   2. CCIE Collaboration officially announced (Vik Malhi)
>>   3. Re: keyboard model in SJ and RTP exam (nielsenj)
>>   4. Re: CCIE Collaboration officially announced (Mark Holloway)
>>   5. Re: CCIE Collaboration officially announced (Hesham Abdelkereem)
>>   6. Re: CCIE Collaboration officially announced (Karen Johnson)
>> 
>> 
>> --
>> 
>> Message: 1
>> Date: Tue, 28 May 2013 14:36:56 -0700 (PDT)
>> From: Karen Johnson 
>> To: "ccie_voice@onlinestudylist.com" 
>> Subject: [OSL | CCIE_Voice] keyboard model in SJ and RTP exam
>> Message-ID:
>><1369777016.74606.yahoomail...@web163903.mail.gq1.yahoo.com>
>> Content-Type: text/plain; charset="iso-8859-1"
>> 
>> 
>> 
>> hi folks,
>> ?
>> anyone remember what is the keyboard model use in SJ and RTP, need to 
>> duplicate for speed.
>> ?
>> tks
>> -- next part --
>> An HTML attachment was scrubbed...
>> URL: 
>> 
>> 
>> --
>> 
>> Message: 2
>> Date: Tue, 28 May 2013 16:08:33 -0700
>> From: Vik Malhi 
>> To: OSL Group 
>> Subject: [OSL | CCIE_Voice] CCIE Collaboration officially announced
>> Message-ID: 
>> Content-Type: text/plain; charset=windows-1252
>> 
>> For my initial reaction read here:
>> 
>> http://bit.ly/12MNK5t
>> 
>> 
>> Vik Malhi ? CCIE #13890
>> Managing Partner - IPexpert, Inc.
>> 
>> Telephone: +1.810.326.1444 ext 420
>> Fax: +1.810.454.0130
>> Mailto: vma...@ipexpert.com
>> 
>> 
>> 
>> 
>> 
>> 
>> --
>> 
>> Message: 3
>> Date: Tue, 28 May 2013 20:45:49 -0500
>> From: nielsenj 
>> To: Karen Johnson 
>> Cc: "ccie_voice@onlinestudylist.com" 
>> Subject: Re: [OSL | CCIE_Voice] keyboard model in SJ and RTP exam
>> Message-ID:
>>
>> Content-Type: text/plain; charset="iso-8859-1"
>> 
>> Logitech K1

Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced

2013-05-28 Thread Hesham Abdelkereem

Yes its really frustrating what Cisco is doing to us.
Ok let me tell you this.
People now have invested a lot of money in pursuing their CCIE Voice that
includes (Verious Workbook fees , Rack Rentals , Home Lab building , travel
expenses and Lab fees attempts for whatever times)
So when people achieve CCIE Voice nowadays a year or two later it would be
considered old and grandfathered.
Also , Cisco has released a new lab for 2 months while they are planning to
abolish the whole syllabus.
Why they do that to us They already make money out of everything
especially lab multiple times of lab attempts per each person.

CCIE Voice achievers has to send cisco request for Migration without Lab
test.
CCVP it was automatically migrated to CCNP Voice without any additional
tests.
CCNA is migrated to CCNA R/S without any additional tests.
In case of Video part then I suggest whether they force CCIE Voice people
to make CCNA VIDEO or CCNP Video if they will release or they make just a
migration lab track that includes VIDEO stuff only for a cheaper fee
something like $500.

Thats same for MICROSOFT they abolished MCSE to change it to MCITP people
usually just add 2 tracks to become full MCITP same when they migrate to
new MCSE (Microsoft Certified Solutions Experts) there is only an upgrade
track rather than taking the whole 5 tracks again.

Cisco obviously has to do something like that.It's really unfair retiring
the whole cisco voice totally.
Guys to make the new Collaboration lab that would cost anyone over 50K to
buy telepresence , X9XX routers stuff , 9971 Video Phones , TV's and etc..
Even the rack rentals would be 5 times the old voice track as the equipment
would be way more expensive.

Seriously , We have to agree all of us from multiple different voice study
group to have a migration track to Collaboration please share your thoughts
guys

On 28 May 2013 18:56, Mark Holloway  wrote:

> Bummer, I was really hoping CCIE Voice candidates would transition to
> Collaboration without any additional lab exams.
>
> On May 28, 2013, at 7:08 PM, Vik Malhi  wrote:
>
> > For my initial reaction read here:
> >
> > http://bit.ly/12MNK5t
> >
> >
> > Vik Malhi – CCIE #13890
> > Managing Partner - IPexpert, Inc.
> >
> > Telephone: +1.810.326.1444 ext 420
> > Fax: +1.810.454.0130
> > Mailto: vma...@ipexpert.com
> >
> >
> >
> >
> > ___
> > For more information regarding industry leading CCIE Lab training,
> please visit www.ipexpert.com
> >
> > Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
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[OSL | CCIE_Voice] VDSL2 Centurylink config with Cisco 887VA Router

2013-05-27 Thread Hesham Abdelkereem

Dear Experts,

First of all , I'd like to say Happy memorial days for everyone.
I have a Centurylink VDSL2 Multimode and I don't like to use their modem as
it's junk and I would like to access my CCIE Voicelab at home from work or
anywhere else.
However, their modem is very poor and I'd like to replace it with a Cisco
Router .
I've bought Cisco 887VA router as it support VDSL2+ multimode which should
be compatible with my ISP.
However , I'd like to have the confi , I have tried to make the dialer
interface for PPPoE and didn't work.
Centurylink support even doesn't know whats their own parameters such as
their authentication type and etc.. so they are not helpful at all and they
will tell u contact Cisco.

If anyone has Century link and configured it on Cisco Router please share
your configs urgently.

Many Thanks in advance,
Hesham
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[OSL | CCIE_Voice] How to solve the LFI & LLQ for Router QOS?

2013-05-22 Thread Hesham Abdelkereem

Dear Experts,

I have solved the LLQ & LFI for Router QOS with exactly the following
solution and I have never scored any points in the section.
Kindy , Please could you tell me what I am missing ?

ON HQ & SB Routers
Set the Bandwidth to 384 for LFI
under the frame relay dlci enable the autoqos
auto qos voip trust

map-class frame-relay AutoQoS-FR-Se0/0/1:0-101
frame-relay cir 384000
frame-relay bc 3840
frame-relay be 0
frame-relay mincir 384000
frame-relay fragment 480
service-policy output AutoQoS-Policy-Trust

to
frame-relay cir 364800
frame-relay bc 3648
frame-relay be 0
frame-relay mincir 364800



policy-map AutoQoS-Policy-Trust
class AutoQoS-VoIP-RTP-Trust
priority percent 70
class AutoQoS-VoIP-Control-Trust
bandwidth percent 5
class class-default
fair-queue

TO
policy-map AutoQoS-Policy-Trust
class AutoQoS-VoIP-RTP-Trust
priority 47
class AutoQoS-VoIP-Control-Trust
bandwidth 16
class class-default
fair-queue

UNDER HQ-SC LINK

wr need to apply a special class map for that link
map-class frame-relay FR-Se0/0/1:0-201
frame-relay cir 1466800
frame-relay bc 14668
frame-relay be 0
frame-relay mincir 1466800
interface Serial0/0/1:0.2 point-to-point
 frame-relay interface-dlci 201
class FR-Se0/0/1:0-201


After finishing then last thing just do on both HQ and SB Router
HQ Router
interface Serial0/1/1:0.102 point-to-point
no frame-relay ip rtp header-compression
!
policy-map AutoQoS-Policy-Trust
class AutoQoS-VoIP-RTP-Trust
priority 47
compress header ip rtp

SB Router
interface Serial0/1/0:0.102 point-to-point
no frame-relay ip rtp header-compression
!
policy-map AutoQoS-Policy-Trust
class AutoQoS-VoIP-RTP-Trust
priority 47
compress header ip rtp


Please let me know whats missing?

Thanks,
Hesham
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[OSL | CCIE_Voice] CUE QOS Configuration

2013-05-22 Thread Hesham Abdelkereem

Dear Experts,

I'd like to know how to configure LAN QOS for CUE Traffic?

As far as i know it's the following

Under all Phone Ports apply qos
Int range fas1/0/14-16
auto qos voip cisco-phone

no mls qos map policed-dscp 24 26 46 to 0
mls qos
mls qos map policed-dscp 24 to 8
mls qos map cos-dscp 0 8 16 24 32 46 48 56
Under the Server Ports AND TRUNK such as CUCM/Unity Connection/UCCX EXCEPT
CUPC TEST/CUPS SERVER
MLS QOS TRUST DSCP

access-list 100 permit tcp host 142.1.66.253 any eq 2748
access-list 101 permit udp host 142.1.66.253 any range 16384 32767

class-map match-any CUE-SIG
match access-group 100
class-map match-any CUE-RTP
match access-group 101

policy-map CUE-POLICY
class CUE-SIG
set dscp cs3
police 32000 8000 exceed-action policed-dscp-transmit
class CUE-RTP
set dscp ef

class class-default
interface gi 1/0/4   UCCX
service-policy input cue
mls qos trust dscp

interface gi 1/0/3   CUCM
mls qos trust dscp

interface gi 1/0/13.15 <<< Phones
 mls qos trust cos
 mls qos trust device cisco-phone


Please let me know If I am missing anything in my configuration such as
port numbers .

Thanks,
Hesham
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Re: [OSL | CCIE_Voice] Cisco evils erase configurations needs magician to explain how?

2013-05-20 Thread Hesham Abdelkereem

Hi pIYUSH,

Yes , I was using both Manual first to define the Switch type , Controller
, Serial Interface then I was doing ccm-manager config server
(subscriberip) then ccm-manager config
OK , Don't you think in the service parameters when you go advanced and
Make MGCP B-Channel Maintenanance status and you can busy out the channels
by making 12 zeros and then 20 One's.

Anyway , What you are saying is definitely make sense although I did the
above and I still find issues and I'd like to thank you so much for your
great input.

Best Regards,
Hesham



On 20 May 2013 04:10, Piyush Jain  wrote:

>  Hi Hesham,
>
> Are you using ccm-manager config and ccm-manager config server commands on
> gateway ??
>
> Whenever you have to configure Partial PRI (like 12 channels) then don't
> use ccm-manager config commands. If you use, then everytime your router
> reboots or you change anything in configuration then it will download the
> configuration from call manager.. Call manager always create full PRI (i.e.
> 31 channels). I believe thats the reason you are facing this issue..
>
>
> Thanks and Regards,
> Piyush Jain
>
>
>   --
>
>
> Message: 2
> Date: Sat, 18 May 2013 16:01:55 -0700
> From: Hesham Abdelkereem 
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] Cisco evils erase configurations needs
> magicianto explain how?
> Message-ID:
> 
> Content-Type: text/plain; charset="iso-8859-1"
>
>
> Dear Experts,
>
> I was practicing labs on my homelab yesterday.
> I have noticed 2 things happened knowing that I always save my configs
> every short bit of time.
> 1- On my SC router I have configured the PRI Channels to use 12 channels
> only.
> I have hardcoded it with the manual configs as well as GUI Config and it
> was working and when i was seeing show isdn service all the unused channels
> were busied out and it was fine for all day.
> 2-On my SC router , I have configured SRST and All dial-peer and everything
> were working perfectly.
>
> I save all configs
>
> I have safely shutdown my lab and saved all configs before
> second day
> I have found
> 1-SiteC MGCP became 31 Channels and not 12
> 2-All dial-peer are missing port 0/0/0:15
>
>
> can someone explain me the reason why that happened to me and how to avoid?
> I am saving my configs always whenever i config any thing even if its
> small
> It's something severe and I was wondering why i scored 20% on my SRST as
> well as low score on my VG sections.
> so I believe it has something to do with that.
>
>
> Thank you very much in advance
>
> Best Regards,
> Hesham
> -- next part --
> An HTML attachment was scrubbed...
> URL:
> 
>
> --
>
> Message: 3
> Date: Sun, 19 May 2013 06:44:35 +0530
> From: sanity insanity 
> To: "ccie_voice@onlinestudylist.com" 
> Subject: Re: [OSL | CCIE_Voice] BACD DOUBTS...
> Message-ID:
> 
> Content-Type: text/plain; charset="iso-8859-1"
>
> hi Guys,
> Any update?
>
>
> On Wed, May 15, 2013 at 11:15 AM, sanity insanity <
> networksanitytoinsan...@gmail.com> wrote:
>
> > Hi Guys,
> >
> > If there is a  requirement using BACD that If agent did not answer the
> > call in 10 s , the call should be routed to next
> > agent. If agents are busy they should hear " All agents are currently
> > busy..."
> > The following prompts are available on the tftp serer  ( ip address
> > X.Y.X.X) and this needs to
> > taken/downloaded to the flash of router. The prompts available are...
> >
> > 3) en_bacd_music-on-hold.au
> > 4) en_bacd_options_menu.au
> > 5) en_bacd_xferto_operater.au
> > 6) en_bacd_afag.au
> > 7) en_bacd_disconnect.au
> > 8) en_bacd_enter_dest.au
> > 9) en_bacd_invalidoption.au
> > 10) en_bacd_welcome.au
> >
> > ==
> > Questions:
> > ==
> >
> > 1) Does this mean that we  need to be using the BACD embedded scripts for
> > bacd?
> >
> > 2) Also which one of the above prompts do we download ? The standard cco
> > doc for the embedded script shows
> > the following as the welcome prompt "welcome-prompt _bacd_welcome.au "  .
> > Are we requried to rename the prompt?
> >
> >
> > -MJ
> >
> -- next part --
> An HTML attachment was scrubbed...
> URL:
> 
>
> --
>
> Message: 4
> Date: Sun, 19 May 2013 07:19:17 +0530
> From: "singh" 
> To:
> ccie_voice-requ...@o

[OSL | CCIE_Voice] How to send calling name with TEHO?

2013-05-20 Thread Hesham Abdelkereem

Dear Experts,

How can i send the calling name when I configure TEHO?
in normal Route Pattern there is a field for calling name to make it
allowed when you do it with SLRG but when You configure Route List for TEHO
you don't have this option to be enabled and disabled however, If you did
it on the Route Pattern level and you've used a TEHO route list everything
is controlled on the Route List Level.

Please advise me


Thanks,
Hesham
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[OSL | CCIE_Voice] FastStart fast busy signal when enabled with MTP

2013-05-19 Thread Hesham Abdelkereem

Dear Experts,

I have configured h323 gateway and enabled outbound fast start.
It gives me fast busy signal when I enable it.
However , I have tried to configure on SiteB router Hardware and Software
MTP as well as Hardware Transcoder associated to an MRG then to MRGL then
to the DP and still didn't work.

I have made it work before many times successfully and I just don't know
why is not working?

Media Termination Point is checked as well as H245 Wait for TCS is
unchecked then Outbound fast start checked

Inbound fast start is working perfectly but outbound gives fast busy signal
and when i disable it then the call works.

I believe the gateway is unable to invoke the MTP thats why it result for
fast busy signal for outbound.

I tried to reload the router didn't work.
tried to make NO SCCP then SCCP
Tried to restart the gateway on cucm gateway section
restarted the Device Pool

Don't know what to do please share ur ideas

Thanks,
Hesham
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[OSL | CCIE_Voice] Cisco evils erase configurations needs magician to explain how?

2013-05-18 Thread Hesham Abdelkereem

Dear Experts,

I was practicing labs on my homelab yesterday.
I have noticed 2 things happened knowing that I always save my configs
every short bit of time.
1- On my SC router I have configured the PRI Channels to use 12 channels
only.
I have hardcoded it with the manual configs as well as GUI Config and it
was working and when i was seeing show isdn service all the unused channels
were busied out and it was fine for all day.
2-On my SC router , I have configured SRST and All dial-peer and everything
were working perfectly.

I save all configs

I have safely shutdown my lab and saved all configs before
second day
I have found
1-SiteC MGCP became 31 Channels and not 12
2-All dial-peer are missing port 0/0/0:15


can someone explain me the reason why that happened to me and how to avoid?
I am saving my configs always whenever i config any thing even if its
small
It's something severe and I was wondering why i scored 20% on my SRST as
well as low score on my VG sections.
so I believe it has something to do with that.


Thank you very much in advance

Best Regards,
Hesham
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[OSL | CCIE_Voice] Please remove this email from this group kar...@naver.com

2013-05-14 Thread Hesham Abdelkereem

Attention To:- Administrator of CCIE Voice Study Group.

Kindly , Please remove this e-mail from your e-mail distribution group
kar...@naver.com.
Whenever we reply to the Study Group we always get a message from this
email that his e-mail is full or invalid.


Your Prompt action would be highly appreciated

Thanks in advance,

Hesham Abdelkereem
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Re: [OSL | CCIE_Voice] ssh client

2013-05-14 Thread Hesham Abdelkereem

I think it should be v2 however I am not quite sure

On 14 May 2013 15:07, Barrera, Hugo  wrote:

>  Anybody know what version of ssh client that is in the real lab on the
> CUPC Test PC? 
>
> ** **
>
> - Hugo
>
> ** **
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
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Re: [OSL | CCIE_Voice] How to debug H323 inbound fast start?

2013-05-08 Thread Hesham Abdelkereem

I think this is what we are looking for only

debug h225 asn 1
h323-message-body connect :
{
  protocolIdentifier { 0 0 8 2250 0 5 }
  h245Address ipAddress :
  {
ip '8E64400C'H
port 33671
  }
  destinationInfo
  {
vendor
{
  vendor
  {
t35CountryCode 181
t35Extension 0
manufacturerCode 18
  }
  productId '436973636F43616C6C4D616E61676572'H
  versionId '31'H
}
terminal
{
}
mc FALSE
undefinedNode FALSE
  }
  conferenceID '140B17D3B6D611E28007002699A4A0C0'H
  callIdentifier
--
*  {
guid '140C5023B6D611E2801EDA9F86EAC63F'H
  }
  fastStart
  {
'000C60138011140001008E66411E59E4008E...'H,
'40060401004C60138011140001008E6641FE...'H
  }
*
On 8 May 2013 07:27, Robert Thomas  wrote:

> I would debug the H225 plane on the GW. Debug h225 asn1 should do it.  I
> don't have a sample debug now, but you should look for OpenLogicalChannels
> and H245 characteristics like Codecs on the same initial SETUP message the
> GW is sending/receiving.
>
>
>  On Mon, May 6, 2013 at 11:30 PM, Hesham Abdelkereem <
> heshamcentr...@gmail.com> wrote:
>
>>  Dear Experts,
>>
>> I would like to know whats the debug command that will prove me that I
>> have enabled they inbound fast start on the H323 gateway?
>>
>> for example give me what should I look for in the debug command.
>>
>> Thanks,
>> Hesham
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com <http://www.platinumplacement.com/>
>>
>
>
>
> --
> Robert Thomas Zamora
> tho...@gmail.com +50689389544
> http://cr.linkedin.com/pub/robert-thomas-zamora/29/913/8a8
> CCNP, CCNP Voice
>
___
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Re: [OSL | CCIE_Voice] How to configure ringlist for a specific phone in CUCM?

2013-05-07 Thread Hesham Abdelkereem

Bill I totally agree with you if it's a big hassle for a CCIE test then
definitely I'd rather skip 2-3 points the whole test is front of me to
score points while the required passing score is 80% and I bet you this
point will never worth over 2% of the test

On 7 May 2013 17:37, Bill Lake  wrote:

>  Could you manually set the phone to an alternative tftp server with all
> the files copied over and do it that way?
>
> OK so who would want to do that in a lab?  Not me, would it be possible,
> yeah maybe, would it be worth a few points, um I don't think it would be
> worth the points it takes to configure.
>
> Now I got one for you, completely off the subject, does anyone know some
> good tools for isolating network jitter and or setting up software agents
> for doing VOIP testing for QOS?  I want to start seeing what differences I
> see with different configs.
>
>
>
> On Tue, May 7, 2013 at 7:25 PM, William Bell  wrote:
>
>>   Ok , now for that thing you mentioned below to point to a different
>> TFTP server and then have a different Ringlist.xml
>> do you mean by that for example I make the universal on Publisher and let
>> all phones register to Publisher?
>>
>>
>> No. Phone registration and TFTP are completely separate aspects of the
>> phone integration to CUCM. That said, you could have a different
>> Ringlist.xml on the publisher than you have on the subscriber and you can
>> have DHCP scopes set different Option 150 addresses. Assuming that meets
>> your requirements and doesn't conflict with other requirements.
>>
>> It may also be possible to leverage the TFTP service on an IOS device.
>> But this approach is convoluted. Especially if we are talking about a lab
>> scenario.
>>
>>  Now , the question where is the parameter where can I apply an external
>> link for the ringlist.xml?
>>
>>
>> No such parameter exists as far as I know. All phones look for the same
>> basic path and file name for Ringlist.xml. The only difference is
>> introduced by the IP address of the TFTP server.
>>
>> -Bill
>>
>>   --
>> William Bell, CCIE #38914
>> blog: http://ucguerrilla.com
>> Follow me on twitter @ucguerrilla
>>
>>
>>
>>
>>  On May 7, 2013, at 11:58 AM, Hesham Abdelkereem <
>> heshamcentr...@gmail.com> wrote:
>>
>>  Hi William,
>>
>> Thanks a lot for your great input.
>> Yes I am aware of the universal ringlist.xml which is located at
>> http://cucmip:6970/ringlist.xml.
>> I know how to change and edit that very well for all the phones.
>> Ok , now for that thing you mentioned below to point to a different TFTP
>> server and then have a different Ringlist.xml
>> do you mean by that for example I make the universal on Publisher and let
>> all phones register to Publisher?
>> and make the other ringlist on the subscriber and let that specific phone
>> register with the subscriber likewise I should configure the first option
>> 150 ip for the phone to subscriber and publisher is the second.
>> I think I can let the UCCX publish the ringlist.xml as it has an IIS as
>> webserver but I don't know how to apply this file on that specific phone on
>> which tab or parameter I am able to do that.
>> In Directories menu , I can create a custom Directories.xml and publish
>> it via UCCX server then I apply the link on the service provisioning
>> enterprise parameters. Then I make service provisioning both inernal and
>> external.
>> Now , the question where is the parameter where can I apply an external
>> link for the ringlist.xml?
>> I am sure that it has something to do with the original phone file
>> configuration which can be tweaked for that.
>>
>> Thanks,
>> Hesham
>>
>> On 7 May 2013 05:18, William Bell  wrote:
>>
>>> There is a specific config file for each phone, this is true. However,
>>> that config file does not contain the ring list. That is a separate config
>>> file, as I am sure you are aware. As far as I know the ringlist file is
>>> universal. The only way you could specify a custom ringlist for one phone
>>> would be to point that phone to a different TFTP server and then have a
>>> different Ringlist.xml on that TFTP server (along with all of the other
>>> files you would need).
>>>
>>> -BIll
>>>
>>>
>>>   --
>>> William Bell, CCIE #38914
>>> blog: http://ucguerrilla.com
>>> twitter: @ucguerrilla
>>>
>>>
>>>
>>>
>>>   On May 7, 2

Re: [OSL | CCIE_Voice] h323 fast start

2013-05-07 Thread Hesham Abdelkereem

 WW
I just got that link earlier this morning as well as Suresh really helped.
Bill , Thank you so much you've been such a great helpful man.
I appreciate all your great efforts.

Many Thanks,
Hesham

On 7 May 2013 15:35, Bill Lake  wrote:

>  try this
>
> http://ciscovoip-amitr.blogspot.com/2011/04/fast-start-vs-slow-start.html
>
>
>  On Tue, May 7, 2013 at 1:44 PM, Barrera, Hugo 
> wrote:
>
>>   Hi,
>>
>> ** **
>>
>> Anyone guide me in the right direction how to read the debugs for h323
>> fast start? 
>>
>> ** **
>>
>> *thanks*
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com 
>>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
___
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Re: [OSL | CCIE_Voice] How to configure ringlist for a specific phone in CUCM?

2013-05-07 Thread Hesham Abdelkereem

Hi William,

Thanks a lot for your great input.
Yes I am aware of the universal ringlist.xml which is located at
http://cucmip:6970/ringlist.xml.
I know how to change and edit that very well for all the phones.
Ok , now for that thing you mentioned below to point to a different TFTP
server and then have a different Ringlist.xml
do you mean by that for example I make the universal on Publisher and let
all phones register to Publisher?
and make the other ringlist on the subscriber and let that specific phone
register with the subscriber likewise I should configure the first option
150 ip for the phone to subscriber and publisher is the second.
I think I can let the UCCX publish the ringlist.xml as it has an IIS as
webserver but I don't know how to apply this file on that specific phone on
which tab or parameter I am able to do that.
In Directories menu , I can create a custom Directories.xml and publish it
via UCCX server then I apply the link on the service provisioning
enterprise parameters. Then I make service provisioning both inernal and
external.
Now , the question where is the parameter where can I apply an external
link for the ringlist.xml?
I am sure that it has something to do with the original phone file
configuration which can be tweaked for that.

Thanks,
Hesham

On 7 May 2013 05:18, William Bell  wrote:

> There is a specific config file for each phone, this is true. However,
> that config file does not contain the ring list. That is a separate config
> file, as I am sure you are aware. As far as I know the ringlist file is
> universal. The only way you could specify a custom ringlist for one phone
> would be to point that phone to a different TFTP server and then have a
> different Ringlist.xml on that TFTP server (along with all of the other
> files you would need).
>
> -BIll
>
>
>   --
> William Bell, CCIE #38914
> blog: http://ucguerrilla.com
> twitter: @ucguerrilla
>
>
>
>
>   On May 7, 2013, at 2:28 AM, Hesham Abdelkereem wrote:
>
>   Dear Experts,
>
> I would like to know how can i edit the ringlist for a specific phone only
> and not for all?
> I believe there is a specific configuration file sepxx.cnf is
> available somewhere in the CUCM but I don't know how to get hold of it.
> Please share your ideas.
>
>
> Thanks,
> Hesham
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com <http://www.platinumplacement.com/>
>
>
>
___
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[OSL | CCIE_Voice] How to debug H323 inbound fast start?

2013-05-06 Thread Hesham Abdelkereem

Dear Experts,

I would like to know whats the debug command that will prove me that I have
enabled they inbound fast start on the H323 gateway?

for example give me what should I look for in the debug command.

Thanks,
Hesham
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[OSL | CCIE_Voice] How to configure MOH Ringback for UCCX?

2013-05-06 Thread Hesham Abdelkereem

Dear Experts,

I'd like to configure MOH for UCCX Ringback tone.
Please share your thoughts and inputs for how we configure this.

Thanks,
Hesham
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[OSL | CCIE_Voice] How to configure ringlist for a specific phone in CUCM?

2013-05-06 Thread Hesham Abdelkereem

Dear Experts,

I would like to know how can i edit the ringlist for a specific phone only
and not for all?
I believe there is a specific configuration file sepxx.cnf is available
somewhere in the CUCM but I don't know how to get hold of it.
Please share your ideas.


Thanks,
Hesham
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Re: [OSL | CCIE_Voice] 5 Lab Handbook Lab 4 task 7.3 : Caller hear Ringback when agent phone ringing

2013-05-05 Thread Hesham Abdelkereem

Hi Ramarchan,

I wonder if you have solved the issue as I am looking forward how we
accomplish that.
If you solved it please share your solution if you don't mind.


Thanks,
Hesham


On 27 April 2013 18:19, Ramcharan Arya  wrote:

> Hi,
>
> I have configure ringback as per solutions guide when agent phone is
> ringing caller hear tone of hold. I am using music on hold audio source 2
> ringback2.wav file
>
> ringback file is upload in PUB and SUB. IP voice streaming media app
> service restarted
>
> Network on hold is set as per solutions guide.PSTN caller and HQ caller
> hear tone on hold.
>
> Can someone please suggest approach how to troubleshoot this issue.
>
> Thanks & Regards,
> Ramcharan Arya
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
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[OSL | CCIE_Voice] Unity Connection & Unity Express Ports Region Interregion Relationship

2013-05-01 Thread Hesham Abdelkereem

Dear Experts,

I'd like to ask when I configure the Regions between HQ , SB and SC
Usually for Interregion relationship is G729 Codec is used while for
Intraregion we use G711 Codec.
So , In case of the Unity Connection and Unity Express. I wonder if i
should apply the same rule on them?
On Unity Connection it has Device Pool and usually you apply HQ for it.
So When SB communicates with unity then is it should be G729
What is your recommendation of how to make it in the test?

Thanks,
Hesham
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Re: [OSL | CCIE_Voice] Buying home lab or Waiting for Cisco Live in June

2013-04-30 Thread Hesham Abdelkereem

Also , It's good to mention that if you've decided to wait for a new
version.
You have to wait for a very long time at least a year after the new version
released.
As  you must wait for a new workbook versions as well as you need to
consult people who took and passed the new version so that will never be in
a couple of months and you talkin' at least a year for things to get
cleared to you if you will wait for a new version so you talkin' about 2
years from now to make your first attempt.
I see go ahead look for the equipment right this minute , Practice and
Study and good luck


On 30 April 2013 19:12, Hesham Abdelkereem  wrote:

> Robert,
>
> Robert my good advice for you.
> But the lab with the highest end Server something like DELL POWER EDGE V3
> which has 64 gb rams and 4 TB hard drives the server will be valid for any
> other application you want.
> CUCM Cluster V9 , Microsoft Server 2012 , Exchange 2013 and LYNC 2013.
> Make sure your server is powerful in the first place and that will not
> cost you more than $950 from ebay.com.
> For the routers and switches If you bought it as components and you would
> like to sell it as component it will still sell slightly more or less than
> the price you get.
> I see don't hesitate just buy your homelab nowadays from ebay.com and
> Never give up.
> If you have only this year for this current version then keep calm and
> carry on.
> Study and If you didn't pass schedule a test every 30 days exactly till
> the abolishment date.
> You will find a spot somewhere for sure whether in SJC , RTP , Europe ,
> Dubai and it worth to travel if needed.
> We have to struggle for something which will give us the lifeline and CCIE
> is a big lifeline for many people not necessarily R/S as it lost it's value
> in the market already but CCIE Voice is unique and demanded and if you
> invested 20K for lab equipment , lab attempts , workbooks and etc. believe
> me you will get compensated right away as soon as you achieve it.
> If you value in the market now 80K - 120K with your current experience +
> CCVP then when you become CCIE V you will get a job for 150K or 150K+ then
> its worth it its lifetime investment do it mate.
>
> Thanks,
> Hesham
>
> On 30 April 2013 18:35, Bill Lake  wrote:
>
>> I know this can be true.  When I took my lab, another version was
>> updating and I talked to one of the guys taking it.  he said that all the
>> seats were full for the next 6 weeks.  So if he did not pass, he most
>> likely would not get another chance.  Not sure what people do, book extra
>> labs just in case?  Then what do you do with the extra one if you pass?  Oh
>> well, don't have to worry about that.
>>
>>
>> On Tue, Apr 30, 2013 at 8:26 PM, Robert Thomas  wrote:
>>
>>>
>>> Eventhough they refresh the HW I don't expect any major changes. I mean
>>> I think people will continue to use 28XX series and PVDM2 for a while...
>>>
>>> Most likely they will upgrade to CUCM 8.6, 9 is still too early on the
>>> field.  10 wont be comming out until next year to the customers.
>>>
>>> The only thing that concerns me, is if they anounce a change, everyone
>>> is going to book their last attempts and buyout any remaining spots for the
>>> year.
>>>
>>> Cisco might go for a refresher just to "increase demand" for spots
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Sun, Apr 28, 2013 at 12:02 PM, Bill  wrote:
>>>
>>>> If they announce a new version in June you will have about 6 months
>>>> from then to pass your lab.
>>>>
>>>> My recommendation from there is if you have plenty of study/lab time
>>>> then go for it.  If you don't and have to squeeze it in then you might be
>>>> better off waiting to see.
>>>>
>>>>  My thought is that 600 to 1200 hours of lab time is needed, more if
>>>> you spread it out and less if you can focus solely on Cisco voice stuff.
>>>>
>>>> Sent from my iPad
>>>>
>>>> On Apr 28, 2013, at 12:38 PM, Alex Mendoza 
>>>> wrote:
>>>>
>>>> Did you know the official date for new version?
>>>>
>>>> I assume that I'll be ready for Sep/oct 2013
>>>>
>>>> Best Regards
>>>>
>>>> AA Mendoza
>>>> Sent from my iPhone 
>>>>
>>>> On 28/04/2013, at 10:33, Robert Thomas  wrote:
>>>>
>>>> Hi,
>>>>
>>>> I'm thinking on buying a home lab to start my stud

Re: [OSL | CCIE_Voice] Buying home lab or Waiting for Cisco Live in June

2013-04-30 Thread Hesham Abdelkereem

Robert,

Robert my good advice for you.
But the lab with the highest end Server something like DELL POWER EDGE V3
which has 64 gb rams and 4 TB hard drives the server will be valid for any
other application you want.
CUCM Cluster V9 , Microsoft Server 2012 , Exchange 2013 and LYNC 2013.
Make sure your server is powerful in the first place and that will not cost
you more than $950 from ebay.com.
For the routers and switches If you bought it as components and you would
like to sell it as component it will still sell slightly more or less than
the price you get.
I see don't hesitate just buy your homelab nowadays from ebay.com and Never
give up.
If you have only this year for this current version then keep calm and
carry on.
Study and If you didn't pass schedule a test every 30 days exactly till the
abolishment date.
You will find a spot somewhere for sure whether in SJC , RTP , Europe ,
Dubai and it worth to travel if needed.
We have to struggle for something which will give us the lifeline and CCIE
is a big lifeline for many people not necessarily R/S as it lost it's value
in the market already but CCIE Voice is unique and demanded and if you
invested 20K for lab equipment , lab attempts , workbooks and etc. believe
me you will get compensated right away as soon as you achieve it.
If you value in the market now 80K - 120K with your current experience +
CCVP then when you become CCIE V you will get a job for 150K or 150K+ then
its worth it its lifetime investment do it mate.

Thanks,
Hesham

On 30 April 2013 18:35, Bill Lake  wrote:

> I know this can be true.  When I took my lab, another version was updating
> and I talked to one of the guys taking it.  he said that all the seats were
> full for the next 6 weeks.  So if he did not pass, he most likely would not
> get another chance.  Not sure what people do, book extra labs just in
> case?  Then what do you do with the extra one if you pass?  Oh well, don't
> have to worry about that.
>
>
> On Tue, Apr 30, 2013 at 8:26 PM, Robert Thomas  wrote:
>
>>
>> Eventhough they refresh the HW I don't expect any major changes. I mean I
>> think people will continue to use 28XX series and PVDM2 for a while...
>>
>> Most likely they will upgrade to CUCM 8.6, 9 is still too early on the
>> field.  10 wont be comming out until next year to the customers.
>>
>> The only thing that concerns me, is if they anounce a change, everyone is
>> going to book their last attempts and buyout any remaining spots for the
>> year.
>>
>> Cisco might go for a refresher just to "increase demand" for spots
>>
>>
>>
>>
>>
>>
>> On Sun, Apr 28, 2013 at 12:02 PM, Bill  wrote:
>>
>>> If they announce a new version in June you will have about 6 months from
>>> then to pass your lab.
>>>
>>> My recommendation from there is if you have plenty of study/lab time
>>> then go for it.  If you don't and have to squeeze it in then you might be
>>> better off waiting to see.
>>>
>>>  My thought is that 600 to 1200 hours of lab time is needed, more if you
>>> spread it out and less if you can focus solely on Cisco voice stuff.
>>>
>>> Sent from my iPad
>>>
>>> On Apr 28, 2013, at 12:38 PM, Alex Mendoza 
>>> wrote:
>>>
>>> Did you know the official date for new version?
>>>
>>> I assume that I'll be ready for Sep/oct 2013
>>>
>>> Best Regards
>>>
>>> AA Mendoza
>>> Sent from my iPhone 
>>>
>>> On 28/04/2013, at 10:33, Robert Thomas  wrote:
>>>
>>> Hi,
>>>
>>> I'm thinking on buying a home lab to start my studies.
>>> It would run around 3K investment according to my amazon shopping list.
>>>
>>> However I'm thinking to wait for June and Cisco Live
>>> for announcement about the new version.
>>>
>>> I don't expect major changes on the setup, perhaps some new phone models
>>> like 99XX, or 89XX on the phones. And upgrade to the routers 29XX.
>>>
>>> However I don't expect major new features from the 29XX roll out on the
>>> exam.
>>>
>>> I would appreciate your opinions on this.
>>>
>>> --
>>> Robert Thomas Zamora
>>> tho...@gmail.com +50689389544
>>> http://cr.linkedin.com/pub/robert-thomas-zamora/29/913/8a8
>>> CCNP, CCNP Voice
>>>
>>> ___
>>> 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
>>>
>>>
>>
>>
>> --
>> Robert Thomas Zamora
>> tho...@gmail.com +50689389544
>> http://cr.linkedin.com/pub/robert-thomas-zamora/29/913/8a8
>> CCNP, CCNP Voice
>>
>
>
> ___
> 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.Pl

[OSL | CCIE_Voice] CUPS User is not authorized to access this page

2013-04-28 Thread Hesham Abdelkereem

Dear Experts,

I'd like to add contacts to CUPC Client.
However i go to https://CUPSSERVERIP/ccmuser
I login with HQ2 and SB2 first it gives me hard time to login
sometimes it logins right away sometimes gives me an error and then i go
back its logined.

The most important thing is when I am succesfully logged in
when I go to any other page such as Preferences , Contacts or etc.
I get User is not authorized to access this page

Knowing that in CUCM user has the CCM Super Users , Standard CTI Enabled ,
Standard AXL API access , ALLO CONTROL from CTI Devices and Standard CCM
USER.

Also , All users are associated to phones as well as the DN's.

What could be the problem then?

Thanks in advance,
Hesham
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[OSL | CCIE_Voice] CUE installation via boothelper

2013-04-27 Thread Hesham Abdelkereem

Dear Experts,

The current version of CUE is 2.1.3 something like that and I was trying to
install CUE 7.0.3
However , I have downloaded all the package from CCO.
I have used the following document
http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/rel3_0/installation/guide/upg3boot.pdf

for my installation process
Everything went quite well except the last step
after I have choosed the language installation instead of reloading it told
me that installation was failed.
Any advice.

Is there are any pre requisits such as formatting the flash? or anything

Thanks,
Hesham
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[OSL | CCIE_Voice] The best way to restore routers to base configs HOMELAB

2013-04-26 Thread Hesham Abdelkereem

Dear Experts?

I wonder whats the best and most efficient way to restore all the
routers/switches of the homelab to the base configs?

Should I just write erase on all devices and then paste the base configs?

Please give me some advice

Thanks in Advance,

Hesham
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[OSL | CCIE_Voice] How to restore CUPS , UCCX and Unity Connection to Post Installation Wizard State?

2013-04-24 Thread Hesham Abdelkereem

Dear Experts,

I have deleted the CUCM PUB and SUB and did a fresh installation.
However , I would like to restore the CUPS and UCCX to the Post
Installation State as they were integrated with the old CUCM nodes.
How can I restrore the CUPS to Post Installation Wwizard?
Ok regarding the UCCX ,
I have followed the below web pages

http://ccie4fun.wordpress.com/2011/11/07/password-recovery-for-uccx-4-to-7/

http://www.cisco.com/en/US/products/sw/custcosw/ps1846/products_tech_note09186a00805a7acc.shtml

In the UCCX

1) Go to Start, run, type ‘cet’ on the UCCX Server. This will launch the
Configuration Object Editor.

2) Browse to: com.cisco.crs.cluster.config.AppAdminSetupConfig in the left
hand pane.

3) Right click the row on the right and hit modify. Then select the
‘com.cisco.crs.cluster.config.AppAdminSetupConfig’ tab.

4) Change the setup state to: FRESH_INSTALL and hit OK

5) Log into the CRA App Admin page with the default username (may be case
sensitive): Administrator and password: ciscocisco

When I open the CET I just see the left list only and on the right its
blank white page.

Now , for the appadmin its not directly in wfavvid its in \wfavvid\system

and I am unable to find the line that contains *
com.cisco.wf.admin.installed=false*

*Also , Please let me know If I need to do anything with Unity Connection
as I have deleted the PUB and SUB nodes.*

*Thanks in advance,*

*Best Regards,*

*Hesham*
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[OSL | CCIE_Voice] UTILS DBREPLICATION REPAIRREPLICATE

2013-04-23 Thread Hesham Abdelkereem

Dear Experts,

I have been running UTILS DBREPLICATION REPAIRREPLICATE FOR 1 DAY and still
the replicaiton is running in the background.
Is it normal that this command takes over 12 hours now and still working?
How long it usually it takes to finish process?
It's just a new 2 Pub and Sub not in production servers which takes forever.
Please advise me?
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[OSL | CCIE_Voice] Help with building a homelab

2013-04-21 Thread Hesham Abdelkereem

Dear Experts,

I am building my homelab now and I have bought an external hard drive that
contains all VMDK and VMX files for all clusters.
I have Dell Power Edge 1950 III so I have installed VMWare ESXi v5.1 on the
server and for the cluster.
I have connected the external hard drive into my laptop and then copied to
the local store of ESXi server.
The storage is just local hard drives not SAN.
After I have finished copying all the cluster to local datastore and then I
have added them to the inventory.
When I try to power up the machine I get this error

Error message on CUPS7: Cannot open the
configuration file
/vmfs/volumes/51718924-525819d8-8c27-001d-
09644fc2/Datastore1/CUPS7/CUPS7.vmx.
error
4/20/2013 10:20:52 PM
CUPS7
root

Error message on UCXN7: Cannot open the
configuration file
/vmfs/volumes/51718924-525819d8-8c27-001d-
09644fc2/Datastore1/UCXN7/UCXN7.vmx.
error
4/20/2013 10:21:13 PM
UCXN7
root


I am unable to open it and always getting error message cannot open the
config file.

Any ideas guys please


Thanks,
Hesham
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[OSL | CCIE_Voice] Unable to reach my voicemail to CUE under SRST with CUE TRANSFER

2013-04-07 Thread Hesham Abdelkereem

Dear Experts,

I am integrating CUE with CUCM and I am doing a feature called CUE TRANSFER.
Which is during an active call , I can transfer the caller to Voicemail by
pressing Transfer + *4XXX + Transfer.
The feature is working and everything under CUCM.
Know I want to make it work under SRST mode.
However, I made it work under SRST but when I transfer it say's no mailbox
setup for the user.Basically, When I press the envelope button from
SCPHONE1 or SCPHONE2 it works and I hear my mailbox greetings but when I do
it by CUE transfer it unable to recognize my mailbox and I configured E164
number in the mailbox but still didn't work.
 The alternate number is working with CUCM integration but it doesn't look
like its working under SRST mode.
Howeve , My testing was the following

Call from SC2 to SC1 and SC1 hit Transfer + Xfer-To-VM + Transfer
>result no mailbox setup from user
Call from HQ1 to SC1 and SC1 hit Transfer + Xfer-To-VM + Transfer
>result no mailbox setup from user
Call from SC1 to VM Pilot --->Reaching personal greeting
Call from SC2 to VM Pilot --->Reaching personal greeting

So it looks like this configs

 CUE(config)# username SiteC1 phonenumberE164 85224044001
CUE(config)# username SiteC2 phonenumberE164 85224044002

is not working under SRST because I am able to make the same thing when its
registered to CUCM

Please let me know what to do to reach my mailbox


Here you are my configurations below:-


voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to sip
allow-connections sip to h323
sip
bind all source-interface loopback0

sip-ua
mwi-server ipv4:142.1.66.253 unsolicited

dial-peer voice 1 pots
incoming called-number .
direct-inward-dial
voice translation-rule 1
rule 1 /^2404/ //
voice translation-profile STRIP
translate called 1
voice-port 0/0/0:15
translation-profile in STRIP

voice translation-rule 8
rule 1 /^\*/ //

voice translation-profile vmredirect
translate called 8

dial-peer voice 4220 voip
destination-pattern 42..$
session protocol sipv2
session target ipv4:142.1.66.253
dtmf-relay sip-notify
codec g711ulaw
vad
translation-profile out vmredirect


voice translation-rule 2
rule 1 /^4...$/ /2404&/ type any subscriber plan any isdn
rule 2 // // type any unknown plan any isdn

voice translation-profile 999
translate calling 2
translate called 2

dial-peer voice 999 pots
translation-profile outgoing 999
destination-pattern 999
port 0/0/0:15
forward-digits all
clid strip name
!
voice translation-rule 3
rule 1 /^4...$/ /2404&/ type any subscriber plan any isdn
rule 2 // // type any subscriber plan any isdn
voice translation-profile LOCAL
translate calling 3
translate called 3
dial-peer voice 98 pots
translation-profile outgoing LOCAL
destination-pattern 9[2-9]...
port 0/0/0:15

voice translation-rule 4
rule 1 /^4...$/ /+8522404&/ type any international plan any isdn
rule 2 // // type any international plan any isdn
voice translation-profile INT
translate calling 4
translate called 4
dial-peer voice 900 pots
translation-profile outgoing INT
destination-pattern 900T
port 0/0/0:15

voice translation-rule 6
rule 1 /^4...$/ /+852404&/ type any international plan any isdn
rule 2 /^2...$/ /1408202&/ type any international plan any isdn
rule 3 /^3...$/ /1972303&/ type any international plan any isdn
!
voice translation-profile 4digits
translate calling 6
translate called 6

dial-peer voice 2300 pots
translation-profile outgoing 4digits
destination-pattern [23]...$
port 0/0/0:15


ephone-dn 10
number 1998 no-reg both
mwi on
number 1999 no-reg both
mwi off
ephone-dn 11 octo-line
number *4...
call-forward all 4220
telephony-service
srst mode auto-provision all
srst dn template 1
srst dn line-mode octo
max-ephones 15
max-dn 15
ip source-address 142.102.66.254 port 2000 strict-match
time-zone 42
date-format dd-mm-yy
voicemail 4220
mwi relay
max-conferences 8 gain -6
call-forward pattern .T
moh music-on-hold.au
transfer-system full-consult
transfer-pattern .T
secondary-dialtone 9
moh music-on-hold.au
multicast moh 239.1.1.1 port 16384 route 142.1.65.254 142.102.65.254
create cnf-files

ephone-dn-template 1
call-forward busy 4220
call-forward noan 4220 timeout 20
mwi sip
huntstop channel 1

ephone-dn 1 octo-line
number 4001 no-reg both
description +85224044001
name SCPHONE1
ephone-dn-template 1
ephone-dn 2 octo-line
number 4002 no-reg both
description +85224044002
name SCPHONE2
ephone-dn-template 1
ephone-dn 3 octo-line
number 4101
description +85224043101
name sc ph1 icd
ephone-dn-template 1
ephone-dn 5 octo-line
number *4001
call-forward all 4220

ephone-dn 4 octo-line
number 4102
description +85224043102
name sc ph2 icd
ephone-dn-template 1

ephone 1
device-security-mode none
mac-address 0024.14B3.8341
speed-dial 4 *4001 label "Xfer-to-VM"
type 7965
button 1:1 2:3
!
!
!
ephone 2
device-security-mode none
mac-address 001A.2F83.3616
type 7970
button 1:2 2:4

application
global
service alternate default

ccm-manager fallback-mgcp

i

[OSL | CCIE_Voice] How to configure CUE as a Backup of Unity Connecitons?

2013-04-02 Thread Hesham Abdelkereem

Dear Experts,

I'd like to know how to configure CUE to work as a backup in case of Unity
Connection failure.
It's very important question as It could come in the new CCIE Voice Labs
all over the world.

Best Regards,
Hesham
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[OSL | CCIE_Voice] Subscriber failure

2013-04-02 Thread Hesham Abdelkereem

Dear Experts,

I was working yesterday on one of the online Rack Rentals.
I have registered all Phones , Gateways and everything to the Subscriber.
Something is very odd.
I was unable to make any calls from the phone at all and the calls were not
reaching the gateway.
I have deleted the SLRG and Recreated, Delete all Route Patterns and then
Recreated them again.
Deleted all Route Groups and recreated them again.
Disassociated LRG from Device Pool and Recreated them Again never worked.
Restarted all Device Pool , Phones and Gateways never worked.
However, When I shut down the subscriber and when it was restarting and
everything fails over on Publisher then everything works perfectly and as
soon as the Subscriber comes back everything is ruined.
However , NTP Server is configured properly , Checked DB replication in
Unified Reporting and it's good status.
All Endpoints shows registered successfully but I am unable to perform
calls.
All Devices are configured with the correct Device Pool and Correct CSS.
So what's likely other problem that makes the subscriber fail?
I restarted it and as soon as it comes back nothing works.

Thanks a lot for your great efforts.

Hesham
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[OSL | CCIE_Voice] MOH Music On Hold source from local router issue

2013-03-28 Thread Hesham Abdelkereem

Dear Experts,

I am trying to configure MOH in order to make SB Router source its music on
hold from the local router.

thats my configs

A-Enable Multicast MOH for the audio stream:
Go to CUCM --->Media Resources--->Music On Hold Audo Source
Tick play continuously , Allow Multicasting

B-Enable Multicast MOH for the MOH server:
Publisher will be unicast MOH Server for HQ
Subscriber will be multicast for SB Site

Go to Media Resources ---> MOH Server--->MOH_3(Subscriber)
Make MOH Device Pool
Enable Multicast Audo Source on this MOH Server

C-Create a Media Resource Group (MRG) for unicast MOH:
Media Resources > Media Resource Group > Add New
Name: MOH_UNICAST
Selected Media Resources: MOH_2 (MOH)
Make sure use multicast in unticked

D-Create a Media Resource Group (MRG) for multicast MOH:
Media Resources > Media Resource Group > Add New
Name: MOH_MCAST
Selected Media Resources: MOH_3 (MOH)
Make sure use multicast in ticked

E-Assign the newly created MRGs to appropriate Media Resource Group List
(MRGL):
Under HQ MRGL Add MOH_UNICAST
Under SB MRGL Add MOH_MCAST

Then Reset Device Pool to take effect

On your SRST make sure that you make multicast
SB
call-manager-fallback
moh music-on-hold.au
multicast moh 239.1.1.1 port 16384 route 142.1.65.254 142.102.65.254
loopback -to  voice vlan
exit

ccm-manager music-on-hold
ip multicast-routing
int vlan 302
ip pim dense-mode
int lo0
ip pim dense-mode

I created a Region called MOH and it's G711 with all sites HQ , SB , SC


When I call and place on hold and try to issue
show ccm-manager music-on-hold
i see 0 active calls
knowing that also on the HQ Phone 1 and SB Phone 1 I put a music on hold
source and without.
All phones are in the correct Device pool , region and location.

I have noticed it's beeping while on hold that means it's unable to invoke
the moh
knowing that i checked the flash: of the router it has the moh file
correclty
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Re: [OSL | CCIE_Voice] BBGK Gatekeeper issues

2013-03-26 Thread Hesham Abdelkereem

Experts,
Thank you so much Suresh and Bill for your great efforts.
I think you both right because i was trying to troubleshoot for one of my
friends.
I didn't look at all for the the Route List thing thats my bad.
Thanks a lot for your great efforts.

Hesham

On 26 March 2013 05:16, Bill Lake  wrote:

> So the call is completing even without showing in GK calls?  That could mean
> you are using an alternative path so check if the call is completing:
>
>- debug isdn q931
>- check for alternate dial peer/trunk it could be using
>
>
> On Mon, Mar 25, 2013 at 11:47 PM, Hesham Abdelkereem <
> heshamcentr...@gmail.com> wrote:
>
>> Dear Experts,
>>
>> I'd like to ask you a very quick question.
>> I have setup a BBGK gatekeeper CUBE between HQ and PSTN (Belgium).
>> However, I see very odd behaviors
>> 1-The call is connecting  whether the Gatekeeper trunk has h245 TCS
>> waiting for capability set is ticked or untucked. I want to know why it is
>> working with the checkbox ticked.
>> 2- when i debug and write the following debugs
>>
>> *debug cch323 h225 *
>>
>> *debug cch323 h245 *
>>
>> *show gatekeeper-calls*
>>
>> *it never shows anything *
>>
>> *whether if i call from HQ - PSTN  OR HQ-SC on both sides*
>>
>> *
>> *
>>
>> *also it doesn't show anything on show gatekeeper calls it comes up with
>> no active call results.*
>>
>> *
>> *
>>
>> *3-i have attached here SDL traces from RTMT to look on.*
>>
>> *
>> *
>>
>> *Thank you in advance and I appreciate all your efforts,*
>>
>> *
>> *
>>
>> *Hesham*
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com
>>
>
>
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Re: [OSL | CCIE_Voice] Gatekeeper Issue (RESOLVED)

2013-03-14 Thread Hesham Abdelkereem

Dear Guys,

I have solved this issue yesterday night.
Thank you very much Vignesh and Lesile for your support.
Ok , Here is a very odd thing happens when you apply a translation profile
on the gatekeeper.
I'm aware of many people who were complaining about Gatekeeper
configuration on CCIE Voice Test.
Ok here what I did

Voice translation-rule 852
rule 1 /^8522404/ //

voice translation-profile STRIP
translate called 852

dial-peer voice 852 voip
incoming called-number 852.
voice-translation in STRIP


that way really didn't work
how did i fix it?

I made it Voice translation-rule 852
rule 1 /^85224044/ /4/

this way it worked but it's very odd and strange they should be same
results.

rule 1 /^8522404/ // means strip this digit and make it null so the
result will be 4001/4002
rule 2 /$85224044/ /4/ means replace/mask 85224044 with 4 so the result
will be 4001/4002
it's like you say 1 + 1 = 2   and  3-1 = 2 so they are same results exactly.

Thanks guys for your great efforts


On 12 March 2013 04:07, vignesh sethuraman  wrote:

>  Thanks Hesham for your kind words. I will definitely check with you
> whenever I have questions related to my preparation.
>
> Take care, All the best.
>
>   ----------
> *From:* Hesham Abdelkereem 
> *To:* vignesh sethuraman 
> *Sent:* Tuesday, 12 March 2013 12:38 AM
>
> *Subject:* Re: [OSL | CCIE_Voice] Gatekeeper Issue
>
>  Wow nice over there:)
> I like Switzerland, I've been always dreaming to go there to Geneva and
> Zurich.
> I got the visa before it becomes Schengen but I was unable to go.
> Good Luck man if you need anything please let me know.
> I guess you will do your lab in Brussels , Belguim if i am not mistaken.
>
>
> Take care of youresef.
>
> On 11 March 2013 17:11, vignesh sethuraman wrote:
>
>   Hesham,
>
> As of now am in Switzerland. I just started my CCIE voice preparation.
> Working on vol 1 labs using ipexpert.
>
> For racks am using my office lab.
>
> Thanks,
> Viki
>
>  --
> *From: *Hesham Abdelkereem ;
> *To: *;
> *Subject: *Re: [OSL | CCIE_Voice] Gatekeeper Issue
> *Sent: *Mon, Mar 11, 2013 5:55:02 PM
>
>   Yes , I will update you tonight.
> r u living in India?
> I work with a brother from India who actually has the lab.
> I am not sure what time is it in your place?
> I live in U.S(SAN JOSE) and my partner in Melbourne , Australia.
> we should be meeting today at 8:00 pm San Jose-Pacific time.
>
> Thanks,
> Hesham
>
> On 11 March 2013 11:52,  wrote:
>
>   U r welcome Hesham. Will wait for your update.
>
> Thanks,
> Vignesh
>
>
> Sent from Yahoo! Mail for iPhone
>
>  --
> *From: *Hesham Abdelkereem ;
> *To: *vignesh sethuraman ;
> *Subject: *Re: [OSL | CCIE_Voice] Gatekeeper Issue
> *Sent: *Mon, Mar 11, 2013 5:22:19 PM
>
>   Ok Vignesh,
>
> That make sense , I am not a geek in gatekeepers.
> I really made it work with the CUBE and with default-technology-prefix
> very easily without any issues.
> But my monster now is not using CUBE or default-technology-prefix.
> But a CCIE Level Engineer is expected to do anything anytime
> loool.
> Thank you for all your great and valuable notes.
> I will test and definitely i will let you know whether works or not.
>
> Best Regards,
> Hesham
>
> On 11 March 2013 11:19, vignesh sethuraman wrote:
>
>
> Adding to that, can you put a secondary DN like "85224044001" on the
> Ephone 1 and try to call from HQ. If the call is working, then I hope it is
> due the E164 registration of SCCP phones with GK.
>
> Or to isolate this issue further, try adding a SIP phone and assign a DN
> with 4003 and try to call from HQ.
>
> Thanks,
> Vignesh
>
> --
> *From:* vignesh sethuraman 
> *To:* Hesham Abdelkereem 
> *Sent:* Monday, 11 March 2013 6:14 PM
>
> *Subject:* Re: [OSL | CCIE_Voice] Gatekeeper Issue
>
>   Hi Hesham,
>
> CUCME should not receive 85224044001 as the called number.
>
> can you see " no E164 registration" on the "show telephony-service dail
> peer" output. If you are not seeing then reload the router and check if no
> E164 registration" is shown on the output.
>
> Thanks,
> Vignesh
>
>
>
>   --
> *From:* Hesham Abdelkereem 
> *To:* vignesh sethuraman 
> *Sent:* Monday, 11 March 2013 5:18 PM
> *Subject:* Re: [OSL | CCIE_Voice] Gatekeeper Issue
>
>  Hi Vignesh,
>
> I'd like to thank you so much for your help and assistance.
>
> That was my telephony-service configs
>
> telephony-service
> no auto-reg-ephone
> em l

Re: [OSL | CCIE_Voice] all incoming calls to HQ phones failing

2013-03-09 Thread Hesham Abdelkereem

Hi Farooq,

You should provide the group with the PSTN configuration here in order to help 
you solve your issues.

Thanks,
Hesham
On Mar 9, 2013, at 11:12 PM, Jaleel ccie  wrote:

> Hi,
> 
> I am having problem with incoming calls to HQ phones, outgoing calls from HQ 
> phones are working. I can't even call HQ phone from HQ-PSTN number.
> I have gone through PSTN config several times but I couldn't find any thing 
> wrong with it. HQ Router is a mgcp gateway and I'm using only 6 Channels for 
> T1 controller.
> 
> What else can I do or check to fix this problem.
> 
> 
> Farooq
> ___
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> visit www.ipexpert.com
> 
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> www.PlatinumPlacement.com

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[OSL | CCIE_Voice] How to edit and overwrite .cnf.xml file on CUCM?

2013-02-26 Thread Hesham Abdelkereem

Dear All,


Hi All, 

I would like to download .cnf.xml for a specific phone so that I can edit it's 
Directories button for that particular phone only.
However , I can do the following http://cucmip:6970/phonemac.cnf.xml
I click on it ---> refresh save as and I can get it and edit it fine.
But there is a big problem when i edit it and upload it back to CUCM nothing 
happens
I did the following
Service Parameters > CISCO TFTP --->Advanced --> Build CNF Files (Build 
All) then Enable Caching of constant and bin (false).

Then I have went to OS Admin ---> TFTP upload then i uploaded with / directory
Then i restarted TFTP and restarted the phone then nothing happened.

Please give me some advice this is very important for you to beat the CCIE lab 
phone customization so quickly and efficiently
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[OSL | CCIE_Voice] IP PHONE CUSTOMIZATION best answer

2013-02-26 Thread Hesham Abdelkereem

Dear All,

I want to solve this case study with the lowest time I can for the test

SB PHONE 1 user is alleging an unauthorized access of his corporate directory 
services from his phone and has asked to disable access to his IP Phone 
corporate Directory.
You have management approval to disable the corporate directory for this phone 
only. When directory button is depressed for the other phones it should display 
services in below order
1) Missed Call 2) Received Calls 3) Placed Calls 4)Corporate Directory.

Kindly , Please guide me step by step for the quickest best way to make it 
easily in less than 15 minutes as this may take with me about 30 minutes in my 
way.
Also , Whats the best way of writing xml files to be in a working format that's 
my tricky part?
I can change ringlist.xml easily but this one is kind of difficult to make the 
thing working properly with IIS of the UCCX Server?

I want the best practice/optimal way to do it from A-Z so that I can beat that 
question as soon as I can.


Best Regards,
Hesham
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Re: [OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUCM and CME issue

2013-02-18 Thread Hesham Abdelkereem

Ok Suresh thank you so much sir for adding that point as well.

On Feb 18, 2013, at 10:37 AM, Suresh Bhandari  wrote:

> Two more things from my side:
> 
> 1. If you have the output of "sh gatek end" and "sh gatek gw" as mentioned, 
> why you used the second zone local command for UCME?
> 
> 2. On you SC router, under your dial-peer to UCM,you need a tech-prefix 
> command.
> dial-peer 85
>  tech-prefix 1*
>  
> Even then the call to CUCM fails, look at / paste the output of "debug 
> gatekeeper main 10". 
> 
> HTH
> 
> 
> On Mon, Feb 18, 2013 at 6:09 AM, Hesham Abdelkereem 
>  wrote:
> I agree with you and and it does make sense.
> I have nothing now I just do that for my CCIE Voice lab preparation and I 
> just try that during the rack rental. I have to do all that over again.
> As soon as I do it , I will let you know.
> I appreciate all your valuable information and thanks so much
> On Feb 17, 2013, at 5:18 PM, Steve Keller  wrote:
> 
>> Since you have 2 zones i believe you must rely on zone prefix to determine 
>> which zone to select a gw from in order to route the call. In your config 
>> your zone prefix is 3... which seems incorrect by glancing at it.
>>  
>> To route calls to CME via GK i would have a RP in CUCM like 4XXX and then 
>> prefix whatever the zone prefix is to it in the pattern. In your case prefix 
>> 31* to match your gateway registration to GK. Thus, my GK config would say 
>> zone prefix CUCME 31*
>>  
>> The ARQ would come into GK with dialed digits of 31*4XXX , Then the 
>> gatekeeper would match tech prefix of 31*, and route to the gw registered in 
>> that zone (your CUCME). I would expect the call setup to arrive on CME with 
>> digits 31*4XXX and try to hit an inbound voip dialpeer, then you would need 
>> the inbound voip dialpeer to strip down to the last 4 digits, or 4XXX in 
>> this case, to match a registered ephone-dn. My inbound voip dialpeer on CME 
>> would only allow the g729 if my GK trunk was set to use g729. Apply a voice 
>> translation rule to the dialpeer to strip down to last 4 digits. If that 
>> ephone-dn is registered then it should ring.
>>  
>> just my 2 cents...
>>  
>> When you make the call from the CUCM phone, what output do you see on the 
>> CME with debug voip dialpeer? Do you see anything?
>> 
>> On Sun, Feb 17, 2013 at 2:56 PM, Hesham Abdelkereem 
>>  wrote:
>> Yes i am using g729 and i configured them from both sides CUCM side as 
>> region and location /devicepool and voice class codec as cme side.
>> I am able to send calls from CME to CUCM but cucm unable to place calls to 
>> CME
>> 
>> On Feb 17, 2013, at 3:51 PM, Cory Gray  wrote:
>> 
>> > Should not have allow connections either unless you are doing cube but 
>> > that should not break it.  Debug h22r ans1 and look to see if there is 
>> > detail on why the call is failing.  Make sure you are using g729 as well
>> >
>> > Sent from my iPhone
>> >
>> > On Feb 17, 2013, at 5:39 PM, "Hesham Abdelkereem" 
>> >  wrote:
>> >
>> >> I did that and allow connections as well
>> >>
>> >> On Feb 17, 2013, at 3:21 PM, Cory Gray  wrote:
>> >>
>> >>> Not sure if this is what is breaking it but you should not have voice 
>> >>> class h323 1 on your ras dialpeer on site c
>> >>>
>> >>> Sent from my iPhone
>> >>>
>> >>> On Feb 17, 2013, at 4:59 PM, "Hesham Abdelkereem" 
>> >>>  wrote:
>> >>>
>> >>>> Dear All,
>> >>>>
>> >>>>
>> >>>> I have tried to configure a  gatekeeper between HQ-SC for 
>> >>>> interoperability between CME and HQ
>> >>>> The issue is I am just able to call from CME to CUCM but Unable to call 
>> >>>> from CUCM to CME.
>> >>>> Knowing that I have created a Device Pool , Route Pattern , Gatekeeper 
>> >>>> info , Gatekeeper controlled trunk for Gatekeeper to call CME from CUCM 
>> >>>> Side
>> >>>> when I debug i always get ARJ Admission Rejection.
>> >>>> I don't want to change anything in the technology prefix or anything.
>> >>>> I don't want to use default technology prefix.
>> >>>> I want show gatekeeper endpoints and show gatekeeper gw-type-prefix to 
>> >>>> be the same exactly.
>> >>>> I just want to troublesho

Re: [OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUCM and CME issue

I agree with you and and it does make sense.
I have nothing now I just do that for my CCIE Voice lab preparation and I just 
try that during the rack rental. I have to do all that over again.
As soon as I do it , I will let you know.
I appreciate all your valuable information and thanks so much
On Feb 17, 2013, at 5:18 PM, Steve Keller  wrote:

> Since you have 2 zones i believe you must rely on zone prefix to determine 
> which zone to select a gw from in order to route the call. In your config 
> your zone prefix is 3... which seems incorrect by glancing at it.
>  
> To route calls to CME via GK i would have a RP in CUCM like 4XXX and then 
> prefix whatever the zone prefix is to it in the pattern. In your case prefix 
> 31* to match your gateway registration to GK. Thus, my GK config would say 
> zone prefix CUCME 31*
>  
> The ARQ would come into GK with dialed digits of 31*4XXX , Then the 
> gatekeeper would match tech prefix of 31*, and route to the gw registered in 
> that zone (your CUCME). I would expect the call setup to arrive on CME with 
> digits 31*4XXX and try to hit an inbound voip dialpeer, then you would need 
> the inbound voip dialpeer to strip down to the last 4 digits, or 4XXX in this 
> case, to match a registered ephone-dn. My inbound voip dialpeer on CME would 
> only allow the g729 if my GK trunk was set to use g729. Apply a voice 
> translation rule to the dialpeer to strip down to last 4 digits. If that 
> ephone-dn is registered then it should ring.
>  
> just my 2 cents...
>  
> When you make the call from the CUCM phone, what output do you see on the CME 
> with debug voip dialpeer? Do you see anything?
> 
> On Sun, Feb 17, 2013 at 2:56 PM, Hesham Abdelkereem 
>  wrote:
> Yes i am using g729 and i configured them from both sides CUCM side as region 
> and location /devicepool and voice class codec as cme side.
> I am able to send calls from CME to CUCM but cucm unable to place calls to CME
> 
> On Feb 17, 2013, at 3:51 PM, Cory Gray  wrote:
> 
> > Should not have allow connections either unless you are doing cube but that 
> > should not break it.  Debug h22r ans1 and look to see if there is detail on 
> > why the call is failing.  Make sure you are using g729 as well
> >
> > Sent from my iPhone
> >
> > On Feb 17, 2013, at 5:39 PM, "Hesham Abdelkereem" 
> >  wrote:
> >
> >> I did that and allow connections as well
> >>
> >> On Feb 17, 2013, at 3:21 PM, Cory Gray  wrote:
> >>
> >>> Not sure if this is what is breaking it but you should not have voice 
> >>> class h323 1 on your ras dialpeer on site c
> >>>
> >>> Sent from my iPhone
> >>>
> >>> On Feb 17, 2013, at 4:59 PM, "Hesham Abdelkereem" 
> >>>  wrote:
> >>>
> >>>> Dear All,
> >>>>
> >>>>
> >>>> I have tried to configure a  gatekeeper between HQ-SC for 
> >>>> interoperability between CME and HQ
> >>>> The issue is I am just able to call from CME to CUCM but Unable to call 
> >>>> from CUCM to CME.
> >>>> Knowing that I have created a Device Pool , Route Pattern , Gatekeeper 
> >>>> info , Gatekeeper controlled trunk for Gatekeeper to call CME from CUCM 
> >>>> Side
> >>>> when I debug i always get ARJ Admission Rejection.
> >>>> I don't want to change anything in the technology prefix or anything.
> >>>> I don't want to use default technology prefix.
> >>>> I want show gatekeeper endpoints and show gatekeeper gw-type-prefix to 
> >>>> be the same exactly.
> >>>> I just want to troubleshoot the issue of calling from CUCM to CME.
> >>>> Thank you so much for all your efforts
> >>>>
> >>>>
> >>>> However, here you are my configs
> >>>>
> >>>> GATEKEEPER HQ Router - SIDE
> >>>>
> >>>> voice service voip
> >>>> allow-connections h323 to h323
> >>>> allow-connections h323 to sip
> >>>> allow-connections sip to h323
> >>>> allow-connections sip to sip
> >>>>
> >>>> interface Loopback0
> >>>> ip address 177.1.254.1 255.255.255.255
> >>>> h323-gateway voip bind srcaddr 177.1.254.1
> >>>>
> >>>> gatekeeper
> >>>> zone local CUCM cisco.com 177.1.254.1
> >>>> zone local CUCME cisco.com
> >>>> zone prefix CUCM 1...
> >>>>

Re: [OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUCM and CME issue

Yes thanks a lot I believe that's the whole issue of the prefix.
That make sense and yes I believe you do understand what I am getting at 
totally and yes all what you've said are correct.
I thank you so much for all your efforts.
I will test it and feed you back but It may take with me a week or so to test 
but I have put it in my consideration.
Many Thanks for all your efforts and it's highly appreciated.

On Feb 17, 2013, at 4:25 PM, Cory Gray  wrote:

> With CUBE, there is no tech prefix so that is why you don't need it here.
> Based on your config, I am assuming your CUCME phones are 3XXX.  That strip
> pattern (taught by IPexpert) will take the last 4 digits of any inbound
> call.  
> H323 has two legs.
> 1.  Inbound Call - which reminds me... needs to be ^313...$ because Site A
> GK will send the tech-prefix to Site C Gateway (your output shows 31 as the
> tech prefix for Site C)
> 2.  Outbound Call - now that you have accepted the call on dial peer 3000
> (or whatever you decided to use) Site C Gateway will look to make another
> call out based on destination-pattern.  Normally the call would be made to
> 313 but we will use the stip translation rule to make it 3XXX before
> trying to make the call
> 
> Where is destination pattern 3XXX?
> You hidden CUCME dial-peers is where.
> Show voice dial-peer summary will show your hidden CUCME dial-peer which I
> am assuming have destination patter 3001 and 3002
> 
> Hope this helps.
> 
> 
> -Original Message-
> From: Hesham Abdelkereem [mailto:heshamcentr...@gmail.com] 
> Sent: Sunday, February 17, 2013 6:16 PM
> To: Cory Gray
> Cc: 'ccie_voice'
> Subject: Re: [OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUCM and CME issue
> 
> Thank you so much for your efforts.
> I believe it may need a strip but i don't know exactly what or how to strip
> the prefix as with CUBE it works without need for translation rule.
> 
> Thanks for info i will try and feed you back.
> 
> Thanks,
> Hesham
> On Feb 17, 2013, at 4:08 PM, Cory Gray  wrote:
> 
>> I am sorry.  I had it backwards.  I thought you had an issue routing 
>> to CUCM.  For call into CUCME, you need this Dial peer voice 3000 voip 
>> Incoming called ^3...$ Dtmf-r h245a No vad Translation-profile in 
>> STRIP !
>> Voice translation-rule 1
>> Rule 1 /.+\(\)$/ /\1/
>> !
>> Voice translation-profile STRIP
>> Translate called 1
>> 
>> -Original Message-
>> From: Hesham Abdelkereem [mailto:heshamcentr...@gmail.com]
>> Sent: Sunday, February 17, 2013 5:56 PM
>> To: Cory Gray
>> Cc: ccie_voice
>> Subject: Re: [OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUCM and CME 
>> issue
>> 
>> Yes i am using g729 and i configured them from both sides CUCM side as 
>> region and location /devicepool and voice class codec as cme side.
>> I am able to send calls from CME to CUCM but cucm unable to place 
>> calls to CME
>> 
>> On Feb 17, 2013, at 3:51 PM, Cory Gray  wrote:
>> 
>>> Should not have allow connections either unless you are doing cube 
>>> but that should not break it.  Debug h22r ans1 and look to see if 
>>> there is detail on why the call is failing.  Make sure you are using 
>>> g729 as well
>>> 
>>> Sent from my iPhone
>>> 
>>> On Feb 17, 2013, at 5:39 PM, "Hesham Abdelkereem"
>>  wrote:
>>> 
>>>> I did that and allow connections as well
>>>> 
>>>> On Feb 17, 2013, at 3:21 PM, Cory Gray 
> wrote:
>>>> 
>>>>> Not sure if this is what is breaking it but you should not have 
>>>>> voice class h323 1 on your ras dialpeer on site c
>>>>> 
>>>>> Sent from my iPhone
>>>>> 
>>>>> On Feb 17, 2013, at 4:59 PM, "Hesham Abdelkereem"
>>  wrote:
>>>>> 
>>>>>> Dear All,
>>>>>> 
>>>>>> 
>>>>>> I have tried to configure a  gatekeeper between HQ-SC for 
>>>>>> interoperability between CME and HQ The issue is I am just able to 
>>>>>> call
>> from CME to CUCM but Unable to call from CUCM to CME.
>>>>>> Knowing that I have created a Device Pool , Route Pattern , 
>>>>>> Gatekeeper info , Gatekeeper controlled trunk for Gatekeeper to 
>>>>>> call
>> CME from CUCM Side when I debug i always get ARJ Admission Rejection.
>>>>>> I don't want to change anything in the technology prefix or anything.
>>>>>> I don't want to use default technology

Re: [OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUCM and CME issue

Thank you so much for your efforts.
I believe it may need a strip but i don't know exactly what or how to strip the 
prefix as with CUBE it works without need for translation rule.

Thanks for info i will try and feed you back.

Thanks,
Hesham
On Feb 17, 2013, at 4:08 PM, Cory Gray  wrote:

> I am sorry.  I had it backwards.  I thought you had an issue routing to
> CUCM.  For call into CUCME, you need this
> Dial peer voice 3000 voip
> Incoming called ^3...$
> Dtmf-r h245a
> No vad
> Translation-profile in STRIP
> !
> Voice translation-rule 1
> Rule 1 /.+\(\)$/ /\1/
> !
> Voice translation-profile STRIP
> Translate called 1
> 
> -----Original Message-
> From: Hesham Abdelkereem [mailto:heshamcentr...@gmail.com] 
> Sent: Sunday, February 17, 2013 5:56 PM
> To: Cory Gray
> Cc: ccie_voice
> Subject: Re: [OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUCM and CME issue
> 
> Yes i am using g729 and i configured them from both sides CUCM side as
> region and location /devicepool and voice class codec as cme side.
> I am able to send calls from CME to CUCM but cucm unable to place calls to
> CME
> 
> On Feb 17, 2013, at 3:51 PM, Cory Gray  wrote:
> 
>> Should not have allow connections either unless you are doing cube but 
>> that should not break it.  Debug h22r ans1 and look to see if there is 
>> detail on why the call is failing.  Make sure you are using g729 as 
>> well
>> 
>> Sent from my iPhone
>> 
>> On Feb 17, 2013, at 5:39 PM, "Hesham Abdelkereem"
>  wrote:
>> 
>>> I did that and allow connections as well
>>> 
>>> On Feb 17, 2013, at 3:21 PM, Cory Gray  wrote:
>>> 
>>>> Not sure if this is what is breaking it but you should not have 
>>>> voice class h323 1 on your ras dialpeer on site c
>>>> 
>>>> Sent from my iPhone
>>>> 
>>>> On Feb 17, 2013, at 4:59 PM, "Hesham Abdelkereem"
>  wrote:
>>>> 
>>>>> Dear All,
>>>>> 
>>>>> 
>>>>> I have tried to configure a  gatekeeper between HQ-SC for 
>>>>> interoperability between CME and HQ The issue is I am just able to call
> from CME to CUCM but Unable to call from CUCM to CME.
>>>>> Knowing that I have created a Device Pool , Route Pattern , 
>>>>> Gatekeeper info , Gatekeeper controlled trunk for Gatekeeper to call
> CME from CUCM Side when I debug i always get ARJ Admission Rejection.
>>>>> I don't want to change anything in the technology prefix or anything.
>>>>> I don't want to use default technology prefix.
>>>>> I want show gatekeeper endpoints and show gatekeeper gw-type-prefix to
> be the same exactly.
>>>>> I just want to troubleshoot the issue of calling from CUCM to CME.
>>>>> Thank you so much for all your efforts
>>>>> 
>>>>> 
>>>>> However, here you are my configs
>>>>> 
>>>>> GATEKEEPER HQ Router - SIDE
>>>>> 
>>>>> voice service voip
>>>>> allow-connections h323 to h323
>>>>> allow-connections h323 to sip
>>>>> allow-connections sip to h323
>>>>> allow-connections sip to sip
>>>>> 
>>>>> interface Loopback0
>>>>> ip address 177.1.254.1 255.255.255.255 h323-gateway voip bind 
>>>>> srcaddr 177.1.254.1
>>>>> 
>>>>> gatekeeper
>>>>> zone local CUCM cisco.com 177.1.254.1 zone local CUCME cisco.com 
>>>>> zone prefix CUCM 1...
>>>>> zone prefix CUCM 2...
>>>>> zone prefix CUCME 3...
>>>>> gw-type-prefix 1*
>>>>> no shutdown
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> SC Side
>>>>> 
>>>>> interface Loopback0
>>>>> ip address 177.1.254.3 255.255.255.255 h323-gateway voip interface 
>>>>> h323-gateway voip id CUCM ipaddr 177.1.254.1 1719 h323-gateway voip 
>>>>> h323-id CUCME h323-gateway voip tech-prefix 31 h323-gateway voip 
>>>>> bind srcaddr 177.1.254.3
>>>>> 
>>>>> 
>>>>> dial-peer voice 85 voip
>>>>> destination-pattern [12]...$
>>>>> voice-class h323 1
>>>>> session target ras
>>>>> dtmf-relay h245-alphanumeric
>>>>> 
>>>>> 
>>>>> CorpHQ(config-dial-peer)#do show gatekeeper end
>>>>>GATEKE

Re: [OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUCM and CME issue

Yes i am using g729 and i configured them from both sides CUCM side as region 
and location /devicepool and voice class codec as cme side.
I am able to send calls from CME to CUCM but cucm unable to place calls to CME

On Feb 17, 2013, at 3:51 PM, Cory Gray  wrote:

> Should not have allow connections either unless you are doing cube but that 
> should not break it.  Debug h22r ans1 and look to see if there is detail on 
> why the call is failing.  Make sure you are using g729 as well
> 
> Sent from my iPhone
> 
> On Feb 17, 2013, at 5:39 PM, "Hesham Abdelkereem"  
> wrote:
> 
>> I did that and allow connections as well
>> 
>> On Feb 17, 2013, at 3:21 PM, Cory Gray  wrote:
>> 
>>> Not sure if this is what is breaking it but you should not have voice class 
>>> h323 1 on your ras dialpeer on site c
>>> 
>>> Sent from my iPhone
>>> 
>>> On Feb 17, 2013, at 4:59 PM, "Hesham Abdelkereem" 
>>>  wrote:
>>> 
>>>> Dear All,
>>>> 
>>>> 
>>>> I have tried to configure a  gatekeeper between HQ-SC for interoperability 
>>>> between CME and HQ
>>>> The issue is I am just able to call from CME to CUCM but Unable to call 
>>>> from CUCM to CME.
>>>> Knowing that I have created a Device Pool , Route Pattern , Gatekeeper 
>>>> info , Gatekeeper controlled trunk for Gatekeeper to call CME from CUCM 
>>>> Side
>>>> when I debug i always get ARJ Admission Rejection.
>>>> I don't want to change anything in the technology prefix or anything.
>>>> I don't want to use default technology prefix.
>>>> I want show gatekeeper endpoints and show gatekeeper gw-type-prefix to be 
>>>> the same exactly.
>>>> I just want to troubleshoot the issue of calling from CUCM to CME.
>>>> Thank you so much for all your efforts
>>>> 
>>>> 
>>>> However, here you are my configs
>>>> 
>>>> GATEKEEPER HQ Router - SIDE
>>>> 
>>>> voice service voip
>>>> allow-connections h323 to h323
>>>> allow-connections h323 to sip
>>>> allow-connections sip to h323
>>>> allow-connections sip to sip
>>>> 
>>>> interface Loopback0
>>>> ip address 177.1.254.1 255.255.255.255
>>>> h323-gateway voip bind srcaddr 177.1.254.1
>>>> 
>>>> gatekeeper
>>>> zone local CUCM cisco.com 177.1.254.1
>>>> zone local CUCME cisco.com
>>>> zone prefix CUCM 1...
>>>> zone prefix CUCM 2...
>>>> zone prefix CUCME 3...
>>>> gw-type-prefix 1*
>>>> no shutdown
>>>> 
>>>> 
>>>> 
>>>> 
>>>> SC Side
>>>> 
>>>> interface Loopback0
>>>> ip address 177.1.254.3 255.255.255.255
>>>> h323-gateway voip interface
>>>> h323-gateway voip id CUCM ipaddr 177.1.254.1 1719
>>>> h323-gateway voip h323-id CUCME
>>>> h323-gateway voip tech-prefix 31
>>>> h323-gateway voip bind srcaddr 177.1.254.3
>>>> 
>>>> 
>>>> dial-peer voice 85 voip
>>>> destination-pattern [12]...$
>>>> voice-class h323 1
>>>> session target ras
>>>> dtmf-relay h245-alphanumeric
>>>> 
>>>> 
>>>> CorpHQ(config-dial-peer)#do show gatekeeper end
>>>> GATEKEEPER ENDPOINT REGISTRATION
>>>> 
>>>> CallSignalAddr  Port  RASSignalAddr   Port  Zone Name TypeFlags
>>>> --- - --- - - -
>>>> 177.1.10.10 1720  177.1.10.10 32811 CUCM  VOIP-GW
>>>> H323-ID: CUCM_TRUNK_1
>>>> Voice Capacity Max.=  Avail.=  Current.= 0
>>>> 177.1.10.20 1720  177.1.10.20 32788 CUCM  VOIP-GW
>>>> H323-ID: CUCM_TRUNK_2
>>>> Voice Capacity Max.=  Avail.=  Current.= 0
>>>> 177.1.254.3 1720  177.1.254.3 63360 CUCM  H323-GW
>>>> H323-ID: CUCME
>>>> Voice Capacity Max.=  Avail.=  Current.= 0
>>>> Total number of active registrations = 3
>>>> 
>>>> CorpHQ(config-dial-peer)#
>>>> 
>>>> 
>>>> CorpHQ(config-dial-peer)#do show gatekeeper gw-type-prefix
>>>> GATEWAY TYPE PREFIX TABLE
>>>> =
>>>> Prefix: 31*
>>>> Zone CUCM master gateway list:
>>>> 177.1.254.3:1720 CUCME
>>>> 
>>>> Prefix: 1*
>>>> Zone CUCM master gateway list:
>>>> 177.1.10.10:1720 CUCM_TRUNK_1
>>>> 177.1.10.20:1720 CUCM_TRUNK_2
>>>> 
>>>> 
>>>> CorpHQ(config-dial-peer)#
>>>> ___
>>>> 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] HQ-SC Gatekeeper Between CUCM and CME issue