from:"Hesham Abdelkereem"

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

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 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

___
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] 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 javascript:_e({}, 'cvml',
 '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 aman.i...@gmail.comjavascript:_e({}, 
 '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




___
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] 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 aa.mend...@icloud.com 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 heshamcentr...@gmail.com
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

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

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

[OSL | CCIE_Voice] Generate a report for number of calls into PRI

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
___
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] 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 wilsonc...@gmail.com 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,

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

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

[OSL | CCIE_Voice] 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,
___
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] 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 karen.johnson...@yahoo.ca 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

 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] 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=fetchBugDetailsbugId=CSCuf24788


The fix was upgrading the CUCM to a different version.


Many Thanks,

Hesham


On 28 July 2013 20:27, Ashok Boinpally ping.as...@gmail.com 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.

___
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] 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 ping.as...@gmail.com 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=fetchBugDetailsbugId=CSCuf24788


 The fix was upgrading the CUCM to a different version.


 Many Thanks,

 Hesham


 On 28 July 2013 20:27, Ashok Boinpally ping.as...@gmail.com 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.

___
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] HWIC-4ESW

2013-07-25 Thread Hesham Abdelkereem

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 ccievoice2013.2...@gmail.comwrote:

 Hello list

 I am looking for 2 HWIC-4ESW cards for my lab, does anyone have spares I
 can buy?

 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

Re: [OSL | CCIE_Voice] HWIC-4ESW

2013-07-25 Thread Hesham Abdelkereem

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
___
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] HWIC-4ESW

2013-07-25 Thread Hesham Abdelkereem

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


On 25 July 2013 11:28, Hesham Abdelkereem heshamcentr...@gmail.com 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



___
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] 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 inht...@yahoo.co.uk 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



 ___
 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] 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 justin.mcint...@blackbox.com 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

 This email and any files transmitted with it are confidential and are
 intended for the sole use of the individual to whom they are addressed.
 Black Box Corporation reserves the right to scan all e-mail traffic for
 restricted content and to monitor all e-mail in general. If you are not the
 intended recipient or you have received this email in error, any use,
 dissemination or forwarding of this email is strictly prohibited. If you
 have received this email in error, please notify the sender by replying to
 this email.
 ___
 For more information regarding industry leading CCIE Lab training, please
 visit www.ipexpert.com

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

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

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

[OSL | CCIE_Voice] 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
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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

[OSL | CCIE_Voice] 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.
___
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] 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:(.*)@(.*) 
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 ping.as...@gmail.com 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.

___
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] 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 lorenzl...@gmail.com 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 jainpiyush2...@ymail.comwrote:

 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


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

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



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

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

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

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

[OSL | CCIE_Voice] 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
___
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] 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 heshamcentr...@gmail.com 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

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 whl...@gmail.com 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 heshamcentr...@gmail.com
 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

___
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] 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 whl...@gmail.com 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
 ATT 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 heshamcentr...@gmail.com
 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 whl...@gmail.com 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 heshamcentr...@gmail.com
 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



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

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

[OSL | CCIE_Voice] 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

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 khaled_sah...@hotmail.com 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

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 khaled_sah...@hotmail.com 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.auparam
 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 khaled_sah...@hotmail.com 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

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 somp...@gmail.com 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 
 khaled_sah...@hotmail.comwrote:

 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.auparam
 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 khaled_sah...@hotmail.com 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



___
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] 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 heshamcentr...@gmail.com 
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 lorenzl...@gmail.com 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
 

___
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] 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 lorenzl...@gmail.com 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

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

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

[OSL | CCIE_Voice] 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
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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

[OSL | CCIE_Voice] 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
___
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] 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 regis_r...@yahoo.com.br 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 heshamcentr...@gmail.com
 *Para:* ccie_voice@onlinestudylist.com 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

 ___
 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] 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 somp...@gmail.com 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 somp...@gmail.comwrote:

 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 somp...@gmail.com 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 
 heshamcentr...@gmail.com 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

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 somp...@gmail.com 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
heshamcentr...@gmail.com 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 somp...@gmail.com 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 somp...@gmail.com wrote:
Hi Hesham,

Thanks for the detail explanation and well thanks for sharing the case. I
find it very intriguing.

I'm working on some idea, but for now, I just want to forward your reply to
the group, in case anyone else can help too.

--Somphol

On Sun, Jun 23, 2013 at 4:44 PM, Hesham Abdelkereem
heshamcentr...@gmail.com wrote:
Hi Somphol,

I have to give you details as much as I can for better assistance not to
tackle some of the information.
Ok let me tell you the call flow
In my scenario
HQ and SB are registered to CUCM

[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
___
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] 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 tho...@gmail.com 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

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

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

[OSL | CCIE_Voice] 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
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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

[OSL | CCIE_Voice] 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
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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

[OSL | CCIE_Voice] 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
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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

[OSL | CCIE_Voice] 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
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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

[OSL | CCIE_Voice] 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
___
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] 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 whl...@gmail.com 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

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

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



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

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

[OSL | CCIE_Voice] 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 ATT 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
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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

[OSL | CCIE_Voice] 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
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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

Re: [OSL | CCIE_Voice] CCIE 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 m...@markholloway.com 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 vma...@ipexpert.com 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

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

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

[OSL | CCIE_Voice] 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
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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

[OSL | CCIE_Voice] 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
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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

[OSL | CCIE_Voice] 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
___
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] 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 jainpiyush2...@ymail.com 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 heshamcentr...@gmail.com
 To: ccie_voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice] Cisco evils erase configurations needs
 magicianto explain how?
 Message-ID:
 caa-uhvfss6yfv-tas-hcs1p3hszc_+eyogwpt9cgvcyh06h...@mail.gmail.com
 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:
 /archives/ccie_voice/attachments/20130518/b689de9e/attachment-0001.html

 --

 Message: 3
 Date: Sun, 19 May 2013 06:44:35 +0530
 From: sanity insanity networksanitytoinsan...@gmail.com
 To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] BACD DOUBTS...
 Message-ID:
 cag4zmyv+xmyonfj-8vni0u9fwd6wvk+7beqffa5d-4ufnnu...@mail.gmail.com
 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:
 /archives/ccie_voice/attachments/20130519/5c23e516/attachment-0001.html

 --

 Message: 4
 Date: Sun, 19 May 2013 07:19:17 +0530
 From: singh singh8...@in.com
 To:
 ccie_voice-requ...@onlinestudylist.com;ccie_voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice] Unity connections - Unassigned Numbers
 question
 Message-ID: 1368928157.4b55df75e2e804bab559aa885be40...@mail.in.com
 Content-Type: text/plain; charset=utf-8

 Hi Guys,For a block of unassigned numbers . I am able to play a prompt
 ofThe number which you are trying reach is not a valid number

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

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

[OSL | CCIE_Voice] 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
___
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] 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 hugo.barr...@nexusis.com 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 http://www.platinumplacement.com/

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

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

[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
___
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] 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 tho...@gmail.com 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

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

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

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

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

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

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

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

[OSL | CCIE_Voice] How to debug H323 inbound fast start?

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

Re: [OSL | CCIE_Voice] How to configure ringlist for a specific phone in CUCM?

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 b...@ucguerrilla.com 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/



___
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] h323 fast start

 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 whl...@gmail.com 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 
 hugo.barr...@nexusis.comwrote:

   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 http://www.platinumplacement.com/



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

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

___
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] 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 ramcharan.a...@gmail.com 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

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

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

[OSL | CCIE_Voice] 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
___
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] 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 whl...@gmail.com 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 tho...@gmail.com 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 whl...@gmail.com 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 aa.mend...@icloud.com
 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 tho...@gmail.com 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.PlatinumPlacement.com

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

Are you a

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 heshamcentr...@gmail.com 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 whl...@gmail.com 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 tho...@gmail.com 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 whl...@gmail.com 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 aa.mend...@icloud.com
 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 tho...@gmail.com 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

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

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

[OSL | CCIE_Voice] 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
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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

[OSL | CCIE_Voice] 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
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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

[OSL | CCIE_Voice] 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*
___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com

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

[OSL | CCIE_Voice] 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?
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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

[OSL | CCIE_Voice] 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

interface

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

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

[OSL | CCIE_Voice] 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
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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

[OSL | CCIE_Voice] 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
___
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] 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 whl...@gmail.com 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



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

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

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

[OSL | CCIE_Voice] 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
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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

[OSL | CCIE_Voice] 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
___
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

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 bring...@gmail.com 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 
 heshamcentr...@gmail.com 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 skeller...@gmail.com 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 
 heshamcentr...@gmail.com 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 corygray22...@hotmail.com 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 
  heshamcentr...@gmail.com wrote:
 
  I did that and allow connections as well
 
  On Feb 17, 2013, at 3:21 PM, Cory Gray corygray22...@hotmail.com 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 
  heshamcentr...@gmail.com 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

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

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

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

I did that and allow connections as well

On Feb 17, 2013, at 3:21 PM, Cory Gray corygray22...@hotmail.com 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 heshamcentr...@gmail.com 
 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

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 corygray22...@hotmail.com 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 heshamcentr...@gmail.com 
 wrote:
 
 I did that and allow connections as well
 
 On Feb 17, 2013, at 3:21 PM, Cory Gray corygray22...@hotmail.com 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 
 heshamcentr...@gmail.com 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

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 corygray22...@hotmail.com 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 corygray22...@hotmail.com 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
 heshamcentr...@gmail.com wrote:
 
 I did that and allow connections as well
 
 On Feb 17, 2013, at 3:21 PM, Cory Gray corygray22...@hotmail.com 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
 heshamcentr...@gmail.com 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 Type
 Flags
 --- - --- - - 
 -
 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

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 corygray22...@hotmail.com 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 corygray22...@hotmail.com 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 corygray22...@hotmail.com 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
 heshamcentr...@gmail.com wrote:
 
 I did that and allow connections as well
 
 On Feb 17, 2013, at 3:21 PM, Cory Gray corygray22...@hotmail.com
 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
 heshamcentr...@gmail.com 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

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