[OSL | CCIE_Voice] unable to start remote desktop - proctorlabs

[CCIE OSL]

Anyone having an issue with proctorlabs pods?
I am able to ping CCM servers but remote desktop will not work.

I am using my own router for eZVPN and able to ping all the interfaces 
and servers, but remote desktop will not connect.
Also I am getting duplex mismatch errors on the HQ3750 for VM server port.
I reset the port and even trying to change the duplex to half,
It;s now back to auto/auto.

I think its something with physical interface on VM servers but 
proctorlab guys think I may have missed something.

I tried loading 2A final config and also tried 12A Final configs.
Anyone seen this??
What am I missing?

HQ-3750#
04:32:23: %CDP-4-DUPLEX_MISMATCH: duplex mismatch discovered on 
FastEthernet1/0/4 (not half duplex), with CUC7-Pub eth0 (half duplex).
HQ-3750#
04:32:38: %CDP-4-DUPLEX_MISMATCH: duplex mismatch discovered on 
FastEthernet1/0/4 (not half duplex), with UCMPub eth0 (half duplex).
HQ-3750#
04:32:54: %CDP-4-DUPLEX_MISMATCH: duplex mismatch discovered on 
FastEthernet1/0/4 (not half duplex), with CUPS-Pub eth0 (half duplex).
HQ-3750#
04:32:57: %CDP-4-DUPLEX_MISMATCH: duplex mismatch discovered on 
FastEthernet1/0/4 (not half duplex), with UCMSub eth0 (half duplex).
HQ-3750#
04:33:23: %CDP-4-DUPLEX_MISMATCH: duplex mismatch discovered on 
FastEthernet1/0/4 (not half duplex), with CUC7-Pub eth0 (half duplex).
HQ-3750#
04:33:38: %CDP-4-DUPLEX_MISMATCH: duplex mismatch discovered on 
FastEthernet1/0/4 (not half duplex), with UCMPub eth0 (half duplex).
HQ-3750#
04:33:54: %CDP-4-DUPLEX_MISMATCH: duplex mismatch discovered on 
FastEthernet1/0/4 (not half duplex), with CUPS-Pub eth0 (half duplex).
HQ-3750#
04:33:57: %CDP-4-DUPLEX_MISMATCH: duplex mismatch discovered on 
FastEthernet1/0/4 (not half duplex), with UCMSub eth0 (half duplex).
HQ-3750#
04:34:23: %CDP-4-DUPLEX_MISMATCH: duplex mismatch discovered on 
FastEthernet1/0/4 (not half duplex), with CUC7-Pub eth0 (half duplex).
HQ-3750#
04:34:38: %CDP-4-DUPLEX_MISMATCH: duplex mismatch discovered on 
FastEthernet1/0/4 (not half duplex), with UCMPub eth0 (half duplex).

 >ping 10.10.210.11

Pinging 10.10.210.11 with 32 bytes of data:

Reply from 10.10.210.11: bytes=32 time=53ms TTL=61
Reply from 10.10.210.11: bytes=32 time=52ms TTL=61
Reply from 10.10.210.11: bytes=32 time=51ms TTL=61
Reply from 10.10.210.11: bytes=32 time=52ms TTL=61

Ping statistics for 10.10.210.11:
Packets: Sent = 4, Received = 4, Lost = 0 (0% los
Approximate round trip times in milli-seconds:
Minimum = 51ms, Maximum = 53ms, Average = 52ms

C:\Documents and Settings\jjung>

 >ping 10.10.210.10

Pinging 10.10.210.10 with 32 bytes of data:

Reply from 10.10.210.10: bytes=32 time=55ms TTL=61
Reply from 10.10.210.10: bytes=32 time=52ms TTL=61
Reply from 10.10.210.10: bytes=32 time=51ms TTL=61
Reply from 10.10.210.10: bytes=32 time=52ms TTL=61

VPN 
Current State: IPSEC_ACTIVE
Last Event: SOCKET_UP
Address: 10.10.0.32
Mask: 255.255.255.255
Save Password: Allowed
Split Tunnel List: 1
   Address: 10.10.0.0
   Mask   : 255.255.0.0
   Protocol   : 0x0
   Source Port: 0
   Dest Port  : 0
Current EzVPN Peer: 74.126.20.247
Backup Gateways
 (0): 12.159.40.185

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

Re: [OSL | CCIE_Voice] V2: IPCC Web link for

[CCIE OSL]

I got it
just in case anyone is interested,,

http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_6_0/installation/guide/cad65ig.pdf







CCIE OSL wrote:


V2
I had this one bookmarked but I cann't seems to find it.

This was under old Cisco Tech docs, but Now with updated web site 
under support, I am not able to get to this page,
Does someone know how to get to IPCC web link for adding ip service 
page under support page.

[OSL | CCIE_Voice] V2: IPCC Web link for

[CCIE OSL]

V2
I had this one bookmarked but I cann't seems to find it.

This was under old Cisco Tech docs, but Now with updated web site under 
support, I am not able to get to this page,
Does someone know how to get to IPCC web link for adding ip service page 
under support page.

[OSL | CCIE_Voice] QOS - Max-reserved-bandwidth needed? Not needed

[CCIE OSL]

I think this question has been asked before,
I have searched achieve but did not find the answer yet.
I thought I know this one, but after spending 3 hours into QOS LFI and 
LLQ now I am more confused.


So I have 768k to BR2,
I am configuring all shaping command to 95% of bandwidth.
729.6K
However, by default I can only use 75% of actual interface bandwidth.
So, Do I need to change the "max-reserved-bandwidth" to 95%?? or even 
100 since class-default can handle the routing and other layer 2 traffic???

I am thinking , YES, but none of solution for IPExPERT has this command in.
So I am guessing that when you configure CIR and MINCIR, this will take 
presence over current max-reserved-bandwidth of 75%

Can someone help??
Thanks...

Re: [OSL | CCIE_Voice] CUE MWI not working -

[CCIE OSL]

Sorry about that ,
The SIP gateway is point to FA interface of CME..

I just change the gateway address to loopback 0 and reloaded CUE.

And now it works

Thanks guys, Cliff, Scott, Tech guy,,
You guys ROCK!!




Tech Guy wrote:
I think Scott was referring to the sip gateway configured on the cue:  
e.g.


ccn subsystem sip
gateway address "10.20.202.1" <---make sure this is correct in cue.
end subsystem


Tech Guy


- Original Message - From: "CCIE OSL" 
To: "Scott ODonnell" 
Cc: "OSL Group" 
Sent: Sunday, April 19, 2009 9:08 PM
Subject: Re: [OSL | CCIE_Voice] CUE MWI not working -




ip route 10.20.202.2  255.255.255.255 Service-Engine1/0

interface Service-Engine1/0
ip unnumbered FastEthernet0/0.302
service-module ip address 10.20.202.2 255.255.255.0
service-module ip default-gateway 10.20.202.1

I am able to leave VM and CUE shows 3 new calls.

I am also running follwing

Installer (Installer application) 2.1.3
Bootloader (Primary) (Service Engine Bootloader) 2.1.2
Infrastructure (Service Engine Infrastructure) 2.1.3
Global (Global manifest) 2.1.3
GPL Infrastructure (Service Engine GPL Infrastructure) 2.1.3
Voice Mail (Voicemail application) 2.1.3
Bootloader (Secondary) (Service Engine Bootloader) 2.1.2
Core (Service Engine OS Core) 2.1.3
Auto Attendant (Service Engine Telephony Infrastructure) 2.1.3






Scott ODonnell wrote:

I had the same problem recently.
It turned out to be the gateway configured under
ccn subsystem sip
 
Make sure it's pointing to your CME address.



 
On Sun, Apr 19, 2009 at 8:15 PM, CCIE OSL <mailto:ccie...@gmail.com>> wrote:


I think I have all the configs, and I can even call manually dial
from the phone 39993001 for MWI on and 39983001 for MWI off. And
it works find.
It seems like the CUE is not sending MWI on/off.
I check the CUE and it's configured with correct numbers.
What am I missing??
This is all within BR2, not from off site.


Here are my current configs,
ccn application ciscomwiapplication
description "ciscomwiapplication"
enabled
maxsessions 8
script "setmwi.aef"
parameter "strMWI_OFF_DN" "3998"
parameter "strMWI_ON_DN" "3999"
parameter "CallControlGroupID" "0"
end application
!
dial-peer voice 3111 voip
destination-pattern 311[126]
session protocol sipv2
session target ipv4:ipofCUE
incoming called-number 399[89]3...
dtmf-relay sip-notify
codec g711ulaw
no vad
!
ephone-dn  4
number 3999 no-reg both
mwi on
ephone-dn  5
number 3998 no-reg both
mwi off
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
 bind control source-interface Loopback0
 bind media source-interface Loopback0
!
ephone-dn  1  dual-line
number 3001 no-reg both
call-forward busy 3111
call-forward noan 3111 timeout 10
ephone-dn  2  dual-line
number 3002 no-reg both
call-forward busy 3111
call-forward noan 3111 timeout 1
!
telephony-service
max-ephones 5
max-dn 5
ip source-address x.x.x.x port 2000
auto assign 1 to 5
sdspfarm units 1
sdspfarm transcode sessions 20
sdspfarm tag 1 mtp0013c3df9b30
create cnf-files version-stamp Jan 01 2002 00:00:00
voicemail 3111
max-conferences 8 gain -6
call-forward pattern .T
web admin system name admin1 password cisco
transfer-system full-consult
transfer-pattern .T
!
ephone  1
username "user1" password null
mac-address 0012.007A.CCD3
type 7940
button  1:3
ephone  2
username "user2" password null
mac-address 0013.19C0.4C81
type 7960
button  1:1

Re: [OSL | CCIE_Voice] CUE MWI not working -

[CCIE OSL]

Did not work!!


Tech Guy wrote:

Try to refresh your mwi from cue:

#mwi refresh all
or
#mwi refresh telephonenumber 

Tech Guy


- Original Message - From: "CCIE OSL" 
To: "OSL Group" 
Sent: Sunday, April 19, 2009 8:15 PM
Subject: [OSL | CCIE_Voice] CUE MWI not working -


I think I have all the configs, and I can even call manually dial 
from the phone 39993001 for MWI on and 39983001 for MWI off. And it 
works find.

It seems like the CUE is not sending MWI on/off.
I check the CUE and it's configured with correct numbers.
What am I missing??
This is all within BR2, not from off site.


Here are my current configs,
ccn application ciscomwiapplication
description "ciscomwiapplication"
enabled
maxsessions 8
script "setmwi.aef"
parameter "strMWI_OFF_DN" "3998"
parameter "strMWI_ON_DN" "3999"
parameter "CallControlGroupID" "0"
end application
!
dial-peer voice 3111 voip
destination-pattern 311[126]
session protocol sipv2
session target ipv4:ipofCUE
incoming called-number 399[89]3...
dtmf-relay sip-notify
codec g711ulaw
no vad
!
ephone-dn  4
number 3999 no-reg both
mwi on
ephone-dn  5
number 3998 no-reg both
mwi off
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
 bind control source-interface Loopback0
 bind media source-interface Loopback0
!
ephone-dn  1  dual-line
number 3001 no-reg both
call-forward busy 3111
call-forward noan 3111 timeout 10
ephone-dn  2  dual-line
number 3002 no-reg both
call-forward busy 3111
call-forward noan 3111 timeout 1
!
telephony-service
max-ephones 5
max-dn 5
ip source-address x.x.x.x port 2000
auto assign 1 to 5
sdspfarm units 1
sdspfarm transcode sessions 20
sdspfarm tag 1 mtp0013c3df9b30
create cnf-files version-stamp Jan 01 2002 00:00:00
voicemail 3111
max-conferences 8 gain -6
call-forward pattern .T
web admin system name admin1 password cisco
transfer-system full-consult
transfer-pattern .T
!
ephone  1
username "user1" password null
mac-address 0012.007A.CCD3
type 7940
button  1:3
ephone  2
username "user2" password null
mac-address 0013.19C0.4C81
type 7960
button  1:1

Re: [OSL | CCIE_Voice] CUE MWI not working -

[CCIE OSL]

ip route 10.20.202.2  255.255.255.255 Service-Engine1/0

interface Service-Engine1/0
ip unnumbered FastEthernet0/0.302
service-module ip address 10.20.202.2 255.255.255.0
service-module ip default-gateway 10.20.202.1

I am able to leave VM and CUE shows 3 new calls.

I am also running follwing

Installer (Installer application) 2.1.3
Bootloader (Primary) (Service Engine Bootloader) 2.1.2
Infrastructure (Service Engine Infrastructure) 2.1.3
Global (Global manifest) 2.1.3
GPL Infrastructure (Service Engine GPL Infrastructure) 2.1.3
Voice Mail (Voicemail application) 2.1.3
Bootloader (Secondary) (Service Engine Bootloader) 2.1.2
Core (Service Engine OS Core) 2.1.3
Auto Attendant (Service Engine Telephony Infrastructure) 2.1.3






Scott ODonnell wrote:

I had the same problem recently.
It turned out to be the gateway configured under
ccn subsystem sip
 
Make sure it's pointing to your CME address.



 
On Sun, Apr 19, 2009 at 8:15 PM, CCIE OSL <mailto:ccie...@gmail.com>> wrote:


I think I have all the configs, and I can even call manually dial
from the phone 39993001 for MWI on and 39983001 for MWI off. And
it works find.
It seems like the CUE is not sending MWI on/off.
I check the CUE and it's configured with correct numbers.
What am I missing??
This is all within BR2, not from off site.


Here are my current configs,
ccn application ciscomwiapplication
description "ciscomwiapplication"
enabled
maxsessions 8
script "setmwi.aef"
parameter "strMWI_OFF_DN" "3998"
parameter "strMWI_ON_DN" "3999"
parameter "CallControlGroupID" "0"
end application
!
dial-peer voice 3111 voip
destination-pattern 311[126]
session protocol sipv2
session target ipv4:ipofCUE
incoming called-number 399[89]3...
dtmf-relay sip-notify
codec g711ulaw
no vad
!
ephone-dn  4
number 3999 no-reg both
mwi on
ephone-dn  5
number 3998 no-reg both
mwi off
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
 bind control source-interface Loopback0
 bind media source-interface Loopback0
!
ephone-dn  1  dual-line
number 3001 no-reg both
call-forward busy 3111
call-forward noan 3111 timeout 10
ephone-dn  2  dual-line
number 3002 no-reg both
call-forward busy 3111
call-forward noan 3111 timeout 1
!
telephony-service
max-ephones 5
max-dn 5
ip source-address x.x.x.x port 2000
auto assign 1 to 5
sdspfarm units 1
sdspfarm transcode sessions 20
sdspfarm tag 1 mtp0013c3df9b30
create cnf-files version-stamp Jan 01 2002 00:00:00
voicemail 3111
max-conferences 8 gain -6
call-forward pattern .T
web admin system name admin1 password cisco
transfer-system full-consult
transfer-pattern .T
!
ephone  1
username "user1" password null
mac-address 0012.007A.CCD3
type 7940
button  1:3
ephone  2
username "user2" password null
mac-address 0013.19C0.4C81
type 7960
button  1:1

[OSL | CCIE_Voice] CUE MWI not working -

[CCIE OSL]

I think I have all the configs, and I can even call manually dial from 
the phone 39993001 for MWI on and 39983001 for MWI off. And it works find.

It seems like the CUE is not sending MWI on/off.
I check the CUE and it's configured with correct numbers.
What am I missing??
This is all within BR2, not from off site.


Here are my current configs,
ccn application ciscomwiapplication
description "ciscomwiapplication"
enabled
maxsessions 8
script "setmwi.aef"
parameter "strMWI_OFF_DN" "3998"
parameter "strMWI_ON_DN" "3999"
parameter "CallControlGroupID" "0"
end application
!
dial-peer voice 3111 voip
destination-pattern 311[126]
session protocol sipv2
session target ipv4:ipofCUE
incoming called-number 399[89]3...
dtmf-relay sip-notify
codec g711ulaw
no vad
!
ephone-dn  4
number 3999 no-reg both
mwi on
ephone-dn  5
number 3998 no-reg both
mwi off
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
 bind control source-interface Loopback0
 bind media source-interface Loopback0
!
ephone-dn  1  dual-line
number 3001 no-reg both
call-forward busy 3111
call-forward noan 3111 timeout 10
ephone-dn  2  dual-line
number 3002 no-reg both
call-forward busy 3111
call-forward noan 3111 timeout 1
!
telephony-service
max-ephones 5
max-dn 5
ip source-address x.x.x.x port 2000
auto assign 1 to 5
sdspfarm units 1
sdspfarm transcode sessions 20
sdspfarm tag 1 mtp0013c3df9b30
create cnf-files version-stamp Jan 01 2002 00:00:00
voicemail 3111
max-conferences 8 gain -6
call-forward pattern .T
web admin system name admin1 password cisco
transfer-system full-consult
transfer-pattern .T
!
ephone  1
username "user1" password null
mac-address 0012.007A.CCD3
type 7940
button  1:3
ephone  2
username "user2" password null
mac-address 0013.19C0.4C81
type 7960
button  1:1

Re: [OSL | CCIE_Voice] Conf bridge per session load balancing

[CCIE OSL]

Folks,
I been playing with soft conference bridge load balancing today.
I done the following steps
1. create a MRG - soft-only-cfb
2. put in soft cfb from SUB first and PUB second
3. Create MRGL - softCFB
4. configured HQ phones to use this MRGL
5. created two meet-me numbers 2700, 2701

test 1: create CFB meet-me number 2700 - goes to SUB CFB
   hang up
test 2: create CFB meet-me number 2701 - goes to SUB CFB again
   hang-up
test 3: create CFB 2700 - goes to SUB, while 2700 is active, create 
meet-me 2701, - this one goes to PUB automatically.


And, I see that the PerfMon is showing 2 resources used out of 48 being 
used on both SUB and PUB.


I just want to confirm that this is the way that the CFB load balancing 
suppose to work?

Can someone confirm this??

Thanks...






Prabahar M wrote:

Hi,
 I had the same issue and posted a request @
http://www.onlinestudylist.com/pipermail/ccie_voice/2009-April/011277.html.

   As for as I understand from the behavior, load balance works only
when an active conference. Put all the SW CFB’s in one MRG. Make a
conference in HQ phones and it uses Pub CFB. Now make a conference in
site B and it uses Sub CFB. Now, stop conference in HQ and restart it
again, it uses Pub CFB.

 Now, you can see 2 CFB resources used in Pub and 1 used in Sub.
Which is what I obsereved after spending lots of time with
configuration and SRND.

 If someone knows better idea, please share it with me. Also let me
know if it works for you.

Best regards,
Prabahar


On Thu, Apr 9, 2009 at 8:10 AM,   wrote:
  

Date: Thu, 9 Apr 2009 14:22:26 +0200
From: "marwa" 
Subject: Re: [OSL | CCIE_Voice] Conf bridge per session load balancing
To: 

hi,

the conf is done but it always goes to the sub , and i am testing using
g711, i even tried to put the hardware conf (hq+ site b)only in the MRG and
put all the rest in another group, but still same issue it goes to one only

has anyone tested this



Eng./ Marwa Ahmed , M.Sc.
Systems Engineering of Egypt  (S.E.E)
CCVP
Cisco IP Communications Support Specialist
CCNP
Senior Customer Support Engineer
Tel : +2 02 22921100
Fax : +2 02 22901673
e-mail : marwa_ah...@seegypt.com
- Original Message -
From: "Linda Mordosky (lmordosk)" 
To: "marwa" ; "Cliff McGlamry"
; "Chris Parker" 
Cc: 
Sent: Thursday, April 09, 2009 1:59 PM
Subject: RE: [OSL | CCIE_Voice] Conf bridge per session load balancing


Remember that the SW conference bridges only support G711.


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of marwa
Sent: Thursday, April 09, 2009 5:50 AM
To: Cliff McGlamry; Chris Parker
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Conf bridge per session load balancing

hi,

i have tried it as Cliff suggested, but i still see all the conf goes
out of
the sub, althought the RMG contains both sub and pub
have it worked with anyone??


- Original Message -
From: "Cliff McGlamry" 
To: "Chris Parker" 
Cc: 
Sent: Monday, April 06, 2009 4:16 PM
Subject: Re: [OSL | CCIE_Voice] Conf bridge per session load balancing




Put them in the same MRG, and it's automatic from there.

Cliff

- Original Message -
From: "Chris Parker" 
To: "OSL Group" 
Sent: Monday, April 06, 2009 10:11 AM
Subject: [OSL | CCIE_Voice] Conf bridge per session load balancing


Hello,

I have an interesting requirement. I need to have the conference
  

bridges


on the pub and sub be available to all phones, but I need each
conference to be set up on each bridge in a round robin fashion. For
example HQ phone 1 starts an ad hoc conference, that conference would
  

be


on the sub conference bridge. The next ad hoc would occur on the pub.
Then after that on the sub again.

Is accomplishing this as simple as having both conf devices in the
  

same


Device Pool / MRG / MRGL? or is there more that needs to be done?

Chris
  

Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 38, Issue 61

[CCIE OSL]

Do you have "max-reserved-bandwidth 100" in you serial interface?

If so, you need to either change this to 95%

Or try to create class-map for routing updates.
You may be running out of bandwidth.
Normally, fair-que on the class default should work.

/Jin Jung...



sean hurricane wrote:

update:
 
Removed QOS config from the interface and what do you know, it did the 
same, Anyone else seeing this?


On Thu, Apr 9, 2009 at 2:19 AM, 
> wrote:


Send CCIE_Voice mailing list submissions to
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When replying, please edit your Subject line so it is more specific
than "Re: Contents of CCIE_Voice digest..."


Today's Topics:

  1. Workbook version 3 lab6 and 10 (sean hurricane)


--

Message: 1
Date: Thu, 9 Apr 2009 01:25:44 -0400
From: sean hurricane mailto:shurric...@gmail.com>>
Subject: [OSL | CCIE_Voice] Workbook version 3 lab6 and 10
To: ccie_voice@onlinestudylist.com

Message-ID:
 
 <29604bef0904082225q176a37a5k9e5dbf8e412dd...@mail.gmail.com

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

In these two labs we were asked to configured SIP trunk over the
WAN and QOS
sections asks to allow 4 G729 calls across the WAN, since the
codec for SIP
trunks has to be G711ulaw, this does not allow enough bandwidth.
my priority
statements are set at 103 for regular frame and 109 for frame
relay with
FRF.12. The calls goes through successfully and supplementary
services work
but while on the call, my ospf neighbor relationship resets and
re-establish
and phones reset especially site B. i have compared with solution will
ipexpert's solution and its the same. is anyone else seeing this?
Vik or
mark can you chime in?

Re: [OSL | CCIE_Voice] bandwidth statement or not

[CCIE OSL]

Bandwidth statement is required if you want to use "auto qos" on a 
frame-relay interface.


Otherwise, below statement is true.

However, I also alway use bandwidth statement myself.

So I do not think I will change just for the lab.
just my 2 cents.

/Jin Jung...

Leigh Harrison wrote:

Hi there Sean,

The "Bandwidth" statement used on interfaces is for reference 
bandwidth for EIGRP and some other protocols, but is only for 
reference purposes.


Check this link:-
http://www.cisco.com/en/US/docs/ios/12_2/interface/command/reference/irfacces.html#wp1120696 



The one that counts on MQC is the bandwidth statement in the policy-map.

LH

Re: [OSL | CCIE_Voice] CCM 4.1 SQL change users for new server

[CCIE OSL]

Thanks Cliff,
I will try your first option a shot.
If not, I am really running out of time and I no longer have my 
installation disks anymore, CD was damaged.

So, I will move on with single CCM for now.
I think I can get most tests done and use proctorlabs for other stuff.

Thanks for your help!!
/Jin Jung...

Cliff McGlamry wrote:

Jin,
 
When SQL Server is installed, it buries the name of the server in the 
system tables as part of the installation.  While it's possible to 
"fix" this, it is not a procedure for the faint at heart.  Even with 
the time it takes to reload, Cisco will openly tell you that they will 
NOT support any other way to fix this in a production environment 
because it's so sticky. 
 
It's not worth wading through the effort on VMWare. 
 
If you can successfully change the IP address and get it running 
(which isn't necessarily easy either), you could probably just put a 
host entry in the host file on the machine to point at the other box 
to avoid issues with name references.  I haven't tried this, but it's 
what I would do.
 
Personally, I keep an image of the OS built up where I can install 
whatever onto it.  Make all the name/ip address/etc. changes and also 
run the newsid.exe tool on the image before installing stuff on top of 
it.  Makes life a lot easier.
 
Cliff
 
 


- Original Message -
*From:* Duy Nguyen <mailto:ccieid...@gmail.com>
*To:* CCIE OSL <mailto:ccie...@gmail.com>
*Cc:* OSL Group <mailto:ccie_voice@onlinestudylist.com>
*Sent:* Monday, April 06, 2009 12:41 AM
*Subject:* Re: [OSL | CCIE_Voice] CCM 4.1 SQL change users for new
server

Are you trying to make a subscriber from the publisher vm?

On Sun, Apr 5, 2009 at 11:36 PM, CCIE OSL mailto:ccie...@gmail.com>> wrote:


I run into little minor problem today.
I am using VM for my home lab, I just copied one VM to another
PC and now I am trying assign it as second CCM.
I change IP and the Name of server.
Well, now the SQL server is complaining.
It looks like I have change all the database users from old server
SUB/ccmCDR, ccmservice,  for all..

Is there a easier way to do this?
Can I just use "sa" for everything?

/Jin Jung...

Re: [OSL | CCIE_Voice] CCM 4.1 SQL change users for new server

[CCIE OSL]

Yep!

Not too fun,,



Duy Nguyen wrote:

Are you trying to make a subscriber from the publisher vm?

On Sun, Apr 5, 2009 at 11:36 PM, CCIE OSL <mailto:ccie...@gmail.com>> wrote:



I run into little minor problem today.
I am using VM for my home lab, I just copied one VM to another PC
and now I am trying assign it as second CCM.
I change IP and the Name of server.
Well, now the SQL server is complaining.
It looks like I have change all the database users from old server
SUB/ccmCDR, ccmservice,  for all..

Is there a easier way to do this?
Can I just use "sa" for everything?

/Jin Jung...

[OSL | CCIE_Voice] CCM 4.1 SQL change users for new server

[CCIE OSL]

I run into little minor problem today.
I am using VM for my home lab, I just copied one VM to another PC and 
now I am trying assign it as second CCM.

I change IP and the Name of server.
Well, now the SQL server is complaining.
It looks like I have change all the database users from old server
SUB/ccmCDR, ccmservice,  for all..

Is there a easier way to do this?
Can I just use "sa" for everything?

/Jin Jung...

Re: [OSL | CCIE_Voice] gatekeeper question - follow up - ACTUAL SOLUTION

[CCIE OSL]

I hope, you are not dreaming about it!

/Jin Jung...

Chris Parker wrote:

Cliff,

This looks great I will try it out tonight. Been thinking about this 
all day too but my pesky job has been keeping me busy ... :-)


Good work

Chris

Cliff McGlamry wrote:
This bugged me.  I kept thinking about it.  
Went back and started googling for some additional gatekeeper 
documentation.  Found what I was looking for.  To summarize the 
problem as I understand it:
 
1.  Trunks coming in from CCM to GK.  Trunk to sub is priority for 
incoming calls from GK.  
2.  If call is in progress over sub trunk, additional call from CCM 
should go to PSTN.
 
3.  If another call from GK comes in and a call is in progress on 
Sub, call should flow to PUB.
 
4.  No Bandwidth command allowed on GK.
 
5.  No additional zones other than the ones defined.
 
6.  No additional tech prefixes allowed.
 
SOLUTION:
 
1.  Create two separate trunks to GK.  Each trunk is in a device pool 
that is either PUB ONLY or SUB ONLY (1 of each).  
2.  On sub trunk ONLY, apply locations bandwidth to limit to one call
 
3.  Build Route groups and route list to allow call to try GK first 
via Sub trunk and fail over to PSTN.  PUB trunk is not in Route list.
 
4.  Adjust CCM service parameters to allow continue attempting to 
route call on busy, etc. (you all are familiare with these 
parameters).  
5.  On Gatekeeper, configuration is defined as sub trunk being 
prioirty 10 for the zone prefix, and pub being priority 9.  So, it 
should look like this (I used the single server in my lab, but it's 
two different trunks:
 
GK#sh gatek gw

GATEWAY TYPE PREFIX TABLE
=
Prefix: 1#*
  Zone CCM master gateway list:
10.0.1.205:49255 GKTrunk1_1
10.0.1.205:49472 GKTrunk2_1
  Zone CCM prefix 1* priority gateway list(s):
   Priority 10:
10.0.1.205:49472 GKTrunk2_1
   Priority 9:
10.0.1.205:49255 GKTrunk1_1
 
Prefix: 2#*

  Zone UCME master gateway list:
10.50.1.2:1720 BR1
 
The trick to make all this work is adding two lines to GK 
configuration (the last two):
 
gatekeeper

 zone local UCME ccm.lab 10.50.1.1
 zone local CCM ccm.lab
 zone prefix CCM 1* gw-priority 10 GKTrunk2_1
 zone prefix CCM 1* gw-priority 9 GKTrunk1_1
 no shutdown
 endpoint resource-threshold<--- Turns on resource tracking at 
default 90% level
 endpoint max-calls h323id GKTrunk2_1 1<--  Assigns max of one 
call to GKTrunk2_1
 
Second command is necessary as CCM doesn't report resource 
utilization to GK.  Once this is applied, it shows up in the show 
gatekeeper endpoints command:
 


GK#sh gatek end
GATEKEEPER ENDPOINT REGISTRATION

CallSignalAddr  Port  RASSignalAddr   Port  Zone Name Type
Flags
--- - --- - - 
-

10.50.1.2   1720  10.50.1.2   58624 UCME  H323-GW
H323-ID: BR1
Voice Capacity Max.=  Avail.=  Current.= 0
10.0.1.205  49255 10.0.1.205  49254 CCM   VOIP-GW
H323-ID: GKTrunk1_1
Voice Capacity Max.=  Avail.=  Current.= 0
10.0.1.205  49472 10.0.1.205  49254 CCM   VOIP-GW
H323-ID: GKTrunk2_1
_Voice Capacity Max.= 1  Avail.= 1  Current.= 0
_Total number of active registrations = 3
 
This shows the trunk has a capacity of 1 call and none currently in 
progress.  When first call is made, it changes to this:
 
GK#sh gatek end GATEKEEPER ENDPOINT REGISTRATION


CallSignalAddr  Port  RASSignalAddr   Port  Zone Name Type
Flags
--- - --- - - 
-

10.50.1.2   1720  10.50.1.2   58624 UCME  H323-GW
H323-ID: BR1
Voice Capacity Max.=  Avail.=  Current.= 1
10.0.1.205  49255 10.0.1.205  49254 CCM   VOIP-GW
H323-ID: GKTrunk1_1
Voice Capacity Max.=  Avail.=  Current.= 0
10.0.1.205  49472 10.0.1.205  49254 CCM   VOIP-GW O
H323-ID: GKTrunk2_1
_Voice Capacity Max.= 1  Avail.= 0  Current.= 1
_Total number of active registrations = 3
 
Now it shows one call in progress.  Show Gatekeeper calls shows the 
call as well:
 
GK#sh gatek calls   Total number of active calls = 1.

 GATEKEEPER CALL INFO
 
LocalCallIDAge(secs)   BW
5-8688 59  128(Kbps)
 Endpt(s): Alias E.164Addr
   src EP: BR1   3001
   CallSignalAddr  Port  RASSignalAddr   Port
   10.50.1.2   1720  10.50.1.2   58624
 Endpt(s): Alias E.164Addr
   dst EP: GKTrunk2_11#1033
   CallSignalAddr  Port  RASSignalAddr   Port
   10.0.1.205  49472 10.0.1.205  49254
 
Now I place another call to force this all into operation:

Re: [OSL | CCIE_Voice] gatekeeper question - follow up - ACTUAL SOLUTION

[CCIE OSL]

Hey, Cliff,
Man!
First, Thanks for all you time on this.
I see your method,  and it's wonderful that this works..

One question,
Let's say, if you are allow to use bandwidth command on Gk, would you 
use same method??


bandwidth interzone 16
/Jin Jung...


Cliff McGlamry wrote:
This bugged me.  I kept thinking about it. 
 
Went back and started googling for some additional gatekeeper 
documentation.  Found what I was looking for.  To summarize the 
problem as I understand it:
 
1.  Trunks coming in from CCM to GK.  Trunk to sub is priority for 
incoming calls from GK. 
 
2.  If call is in progress over sub trunk, additional call from CCM 
should go to PSTN.
 
3.  If another call from GK comes in and a call is in progress on Sub, 
call should flow to PUB.
 
4.  No Bandwidth command allowed on GK.
 
5.  No additional zones other than the ones defined.
 
6.  No additional tech prefixes allowed.
 
SOLUTION:
 
1.  Create two separate trunks to GK.  Each trunk is in a device pool 
that is either PUB ONLY or SUB ONLY (1 of each). 
 
2.  On sub trunk ONLY, apply locations bandwidth to limit to one call
 
3.  Build Route groups and route list to allow call to try GK first 
via Sub trunk and fail over to PSTN.  PUB trunk is not in Route list.
 
4.  Adjust CCM service parameters to allow continue attempting to 
route call on busy, etc. (you all are familiare with these parameters). 
 
5.  On Gatekeeper, configuration is defined as sub trunk being 
prioirty 10 for the zone prefix, and pub being priority 9.  So, it 
should look like this (I used the single server in my lab, but it's 
two different trunks:
 
GK#sh gatek gw

GATEWAY TYPE PREFIX TABLE
=
Prefix: 1#*
  Zone CCM master gateway list:
10.0.1.205:49255 GKTrunk1_1
10.0.1.205:49472 GKTrunk2_1
  Zone CCM prefix 1* priority gateway list(s):
   Priority 10:
10.0.1.205:49472 GKTrunk2_1
   Priority 9:
10.0.1.205:49255 GKTrunk1_1
 
Prefix: 2#*

  Zone UCME master gateway list:
10.50.1.2:1720 BR1
 
The trick to make all this work is adding two lines to GK 
configuration (the last two):
 
gatekeeper

 zone local UCME ccm.lab 10.50.1.1
 zone local CCM ccm.lab
 zone prefix CCM 1* gw-priority 10 GKTrunk2_1
 zone prefix CCM 1* gw-priority 9 GKTrunk1_1
 no shutdown
 endpoint resource-threshold<--- Turns on resource tracking at 
default 90% level
 endpoint max-calls h323id GKTrunk2_1 1<--  Assigns max of one 
call to GKTrunk2_1
 
Second command is necessary as CCM doesn't report resource utilization 
to GK.  Once this is applied, it shows up in the show gatekeeper 
endpoints command:
 


GK#sh gatek end
GATEKEEPER ENDPOINT REGISTRATION

CallSignalAddr  Port  RASSignalAddr   Port  Zone Name Type
Flags
--- - --- - - 
-

10.50.1.2   1720  10.50.1.2   58624 UCME  H323-GW
H323-ID: BR1
Voice Capacity Max.=  Avail.=  Current.= 0
10.0.1.205  49255 10.0.1.205  49254 CCM   VOIP-GW
H323-ID: GKTrunk1_1
Voice Capacity Max.=  Avail.=  Current.= 0
10.0.1.205  49472 10.0.1.205  49254 CCM   VOIP-GW
H323-ID: GKTrunk2_1
_Voice Capacity Max.= 1  Avail.= 1  Current.= 0
_Total number of active registrations = 3
 
This shows the trunk has a capacity of 1 call and none currently in 
progress.  When first call is made, it changes to this:
 
GK#sh gatek end 
GATEKEEPER ENDPOINT REGISTRATION


CallSignalAddr  Port  RASSignalAddr   Port  Zone Name Type
Flags
--- - --- - - 
-

10.50.1.2   1720  10.50.1.2   58624 UCME  H323-GW
H323-ID: BR1
Voice Capacity Max.=  Avail.=  Current.= 1
10.0.1.205  49255 10.0.1.205  49254 CCM   VOIP-GW
H323-ID: GKTrunk1_1
Voice Capacity Max.=  Avail.=  Current.= 0
10.0.1.205  49472 10.0.1.205  49254 CCM   VOIP-GW O
H323-ID: GKTrunk2_1
_Voice Capacity Max.= 1  Avail.= 0  Current.= 1
_Total number of active registrations = 3
 
Now it shows one call in progress.  Show Gatekeeper calls shows the 
call as well:
 
GK#sh gatek calls   
Total number of active calls = 1.

 GATEKEEPER CALL INFO
 
LocalCallIDAge(secs)   BW
5-8688 59  128(Kbps)
 Endpt(s): Alias E.164Addr
   src EP: BR1   3001
   CallSignalAddr  Port  RASSignalAddr   Port
   10.50.1.2   1720  10.50.1.2   58624
 Endpt(s): Alias E.164Addr
   dst EP: GKTrunk2_11#1033
   CallSignalAddr  Port  RASSignalAddr   Port
   10.0.1.205  49472 10.0.1.205  49254
 
Now I place another call to force this 

Re: [OSL | CCIE_Voice] Dial-peer overlapping

[CCIE OSL]

That's good point,

/Jin Jung..

Wesley Lim wrote:

Interesting.

How I would tackle this.

Lets assume we have 2 dialpeers as of the following:

1. 2...
2. 22...

No matter what, both dialpeers are overlapping. Hence for the dialpeer
with 2... , i would put a timeout at the end of the destination
pattern - for eg 2...T

On 3/31/09, CCIE OSL  wrote:
  

I think you need new dial-peer with
diall-peer voice 2400 voip
destination-p 2[23]...

to match 22... and 23...

and may need to create GK prefix match this as well. 23*, 22*
and setup call forward all,

On CCM, need RP for 24xxx and point to GK, may also need GK prefix and
point to CME.
24*

I do not think you can use dial-peer 2300.
Is this a requirement that you must use one dial-peer?
If so, you can use voice translation profile to strip "2", but I do not
think you can make it go directly to VM, it will ring the ext 2001 first
Unless you setup ext 2001 to call forward all directly to VM, Which I do
not think you want to do.

voice translation-rule 1
rule 1 /^2\(2...\)/ /\1/
rule 2 /^2\(3...\)/ /\1/

/Jin Jung...






Duy Nguyen wrote:


How would I achieve this?

User at Site C should press 24XXX, then it will forward to user's
voice mailbox greeting.

E.g. when user dial 24001, then it will be forwarding directly to 4001
VM and leave a message. This call routing should work over the WAN also.

My solution:
ephone-dn 20
number 24...
call-forward all 4111

Problem is:
dial-peer voice 2300 voip
destination-pattern [23]...
session target ras
tech-prefix 1
dtmf-relay h245-alphanumeric
ip qos dscp cs3 signaling
no vad
!

Another issue, it is also asking for the same on CCM side when user
dials 22XXX should go to VM directly and should also work over the
WAN. From CME could not dial 22XXX since it keeps hitting Dial-peer
voice 2300 voip.

If I change the
dial-peer voice 2300
destination-pattern [23][0128]..

Still won't achieve calling to Main site phone's VM.
  



  

Re: [OSL | CCIE_Voice] gatekeeper question - possible solution

[CCIE OSL]

s
 tech-prefix 3#
 dtmf-relay h245-alphanumeric
 no vad
 
 
 


- Original Message -
*From:* kapil atrish <mailto:nice_cha...@yahoo.com>
*To:* CCIE OSL <mailto:ccie...@gmail.com>
*Cc:* ccie_voice@onlinestudylist.com
<mailto:ccie_voice@onlinestudylist.com>
*Sent:* Sunday, March 29, 2009 2:37 AM
*Subject:* Re: [OSL | CCIE_Voice] gatekeeper question

Hi,

I've not tested this since I don't have lab access yet. I can use
"endpoint max'conn" on GK since now I've two trunks towards CCM.
But below I described using CCM locations.  Both should work
depending on what is allowed in GK config snap-shot.
gw-priority config would be straight forward:

zone-prefix GK 1* gw-priority 10 Sub_Trunk_1
zone-prefix GK 1* gw-priority 9 Pub_Trunk_2



--- On *Sun, 3/29/09, CCIE OSL /mailto:ccie...@gmail.com>>/* wrote:


From: CCIE OSL 
Subject: Re: [OSL | CCIE_Voice] gatekeeper question
To: "kapil atrish" 
Cc: "ccie Me" ,
ccie_voice@onlinestudylist.com
Date: Sunday, March 29, 2009, 11:43 AM

kapil,
Actually, I was thinking of using AAR for the first part of
the requirement. - As I said before, I have not tried this, I
am scheculed for a proctorslab Monday.
For the requirement of "from HQ to SiteC via GK it will be
rejected by GK and rerouted to pstn via 6608 on HQ".
I should be able to create AAR group for BR2 and apply it to
the trunk. this way I can reserve the 4 digit HQ ANI as well.
I may have to use "Location" but I think GK will send out a
call reject to CCM.

Have you confirm that your method works. If you got this working,
Can you send me GK end and GK gw-prefix output for this?

Thanks...

/Jin Jung...

kapil atrish wrote:
> You can try this:
>
> Create two trunks between CCM and GK having only one CCM in
each trunk i.e one with Pub and another one with Sub. Create
two set of regions (codec G729), two locations (24kbps to
allow only single call over the trunk), and two DPs. Bind all
this with respective trunks.
>
> On the GK use gw-priority as regular, primary Sub and
secondary Pub.
>
> For HQ to Site C calling: Create two RGs having Sub and
HQGW. Create RL having both these RGs. Create a RP for Site
C>>>Point to this RL. Now if any call is already active over
this H.323 Trunk any subsequent call from HQ side will fall
back (Location on the GK trunk will reject this call). You
might have to turn on the CCM Service parameters (Continue
routing on unallocated number).
>
>
> For Site C to HQ Calling: Since 1 call is already active on
Sub, any subsequent call from Site C will not be allowed due
to b/w limitation over that trunk (Location). Next call should
fall back to Pub trunk which is having gw-priority 9.
>
>
> Thanks...
>
>
> --- On *Sun, 3/29/09, CCIE OSL />/* wrote:
>
>
> From: CCIE OSL >
> Subject: Re: [OSL | CCIE_Voice] gatekeeper question
> To: "ccie Me" >
> Cc: ccie_voice@onlinestudylist.com

> Date: Sunday, March 29, 2009, 9:52 AM
>
> I have some questions for you.
>
> 1. Are you running on 1 tech-prefix or two with on the
gatekeeper?
> 2. Does entire BR2 has to able to call HQ or just a
single phone?
> 3. Does it required to only use 1#, are you allow to use
other
> prefixes?
>
> Your first requirement for HQ to BR2 is fairly easy,
> However, second requirement, is bit confusing.
>
> I think in order to make that work, I will have to use
Hop-off
> prefix  and statically map a another prefix to PUB address.
> But since the CAC requirement of single call, and If I
were to use
> bandwidth interzone, I almost need another zone just for
PUB,
> Which means I may have to use different CAC method.
/? or
> somehow allow calls to PUB work using hop-off prefix not
affected
> by GK CAC.???
>
> I have proctor lab coming up on Monday night, I may have
to lab
> this up.
>
> If you can provide answer to my questions, It may help
me to get
> this done.
>
&

Re: [OSL | CCIE_Voice] Dial-peer overlapping

[CCIE OSL]

I think you need new dial-peer with
diall-peer voice 2400 voip
destination-p 2[23]...

to match 22... and 23...

and may need to create GK prefix match this as well. 23*, 22*
and setup call forward all,

On CCM, need RP for 24xxx and point to GK, may also need GK prefix and 
point to CME.

24*

I do not think you can use dial-peer 2300.
Is this a requirement that you must use one dial-peer?
If so, you can use voice translation profile to strip "2", but I do not 
think you can make it go directly to VM, it will ring the ext 2001 first
Unless you setup ext 2001 to call forward all directly to VM, Which I do 
not think you want to do.


voice translation-rule 1
rule 1 /^2\(2...\)/ /\1/
rule 2 /^2\(3...\)/ /\1/

/Jin Jung...






Duy Nguyen wrote:

How would I achieve this?

User at Site C should press 24XXX, then it will forward to user's 
voice mailbox greeting.


E.g. when user dial 24001, then it will be forwarding directly to 4001 
VM and leave a message. This call routing should work over the WAN also.


My solution:
ephone-dn 20
number 24...
call-forward all 4111

Problem is:
dial-peer voice 2300 voip
destination-pattern [23]...
session target ras
tech-prefix 1
dtmf-relay h245-alphanumeric
ip qos dscp cs3 signaling
no vad
!

Another issue, it is also asking for the same on CCM side when user 
dials 22XXX should go to VM directly and should also work over the 
WAN. From CME could not dial 22XXX since it keeps hitting Dial-peer 
voice 2300 voip.


If I change the
dial-peer voice 2300
destination-pattern [23][0128]..

Still won't achieve calling to Main site phone's VM. 

[OSL | CCIE_Voice] 1760 Router DSP Question,

[CCIE OSL]

I have purchased a 1760 router with PVDM-12 and CME.and FXS
Just so that I can play with it at home.
 I normally use Proctorslab.

Now I am trying to configure Transcoder, 
But it will not work, 

Accouring to the Cisco DOC, 1760 only support PVDM-256k-4,8,12..NOT
PVDM2.
However, when I call from my FXS port to local CME phone it work fine
with DSP farm showing trancording.
 
SPMM  DSPRM  State   Image D-sig  D-sig A-sig  A-sig  Mips
Voice/Xcode
  Dsp   Dsp  allocate   free  allocate   free  Free
Chan
  0/0 0  UP   FIXHC 0  0 2  0   
50    1
  0/1 1  UP   FLEX6 0  6 0  0  
100    0
  0/2 2  UP   FLEX6 0  6 0  0  
100    0

So, Is the PVDM2 working on 1760?? or is not??

In any case, somehow I got it to register sdspfarm for CME, but now my
FXZ phone stops working.
I guess I took the DSP resource away from FXS, 
So I change sdspfarm max-session from 4 to 2 and reload the router,
Now, mu FXS phone works but sdspfarm is no longer registered. 

mtp-1 Device:mtp0014a8202e0f TCP socket:[-1]  UNREGISTERED
actual_stream:0 max_stream 0 IP:0.0.0.0  0  Unknown 0 keepalive 0  

 max-mtps:1, max-streams:4, alloc-streams:0, act-streams::0

Can someone help

/Jin Jung..


Cisco 1751 and Cisco 1760 Routers


Cisco 1751 and Cisco 1760 routers use the following PVDMs:


•PVDM-256K-4 (1 DSP)


•PVDM-256K-8 (2 DSPs)


•PVDM-256K-12 (3 DSPs)


•PVDM-256K-16HD (4 DSPs)


•PVDM-256K-20HD (5 DSPs)


DSP Farms for NM-HDVs: Example 

The following example sets up a DSP farm of 4 DSPs to handle up to 16
sessions (4 sessions per DSP) on a router with an IP address of
10.5.49.160 and a priority of 1 among other servers.

voice-card 1
 dsp services dspfarm
 exit

sccp local FastEthernet 0/0

sccp

sccp ccm 10.5.49.160 priority 1

dspfarm transcoder maximum sessions 16

dspfarm

telephony-service

 ip source-address 10.5.49.200 port 2000

 sdspfarm units 4

 sdspfarm transcode sessions 40

 sdspfarm tag 1 mtp000a8eaca80

Re: [OSL | CCIE_Voice] gatekeeper question

[CCIE OSL]

kapil,
Actually, I was thinking of using AAR for the first part of the 
requirement. - As I said before, I have not tried this, I am scheculed 
for a proctorslab Monday.
For the requirement of "from HQ to SiteC via GK it will be rejected by 
GK and rerouted to pstn via 6608 on HQ".
I should be able to create AAR group for BR2 and apply it to the trunk. 
this way I can reserve the 4 digit HQ ANI as well.
I may have to use "Location" but I think GK will send out a call reject 
to CCM.


Have you confirm that your method works. If you got this working,
Can you send me GK end and GK gw-prefix output for this?

Thanks...

/Jin Jung...

kapil atrish wrote:

You can try this:

Create two trunks between CCM and GK having only one CCM in each trunk 
i.e one with Pub and another one with Sub. Create two set of regions 
(codec G729), two locations (24kbps to allow only single call over the 
trunk), and two DPs. Bind all this with respective trunks.


On the GK use gw-priority as regular, primary Sub and secondary Pub.

For HQ to Site C calling: Create two RGs having Sub and HQGW. Create 
RL having both these RGs. Create a RP for Site C>>>Point to this RL. 
Now if any call is already active over this H.323 Trunk any subsequent 
call from HQ side will fall back (Location on the GK trunk will reject 
this call). You might have to turn on the CCM Service parameters 
(Continue routing on unallocated number).



For Site C to HQ Calling: Since 1 call is already active on Sub, any 
subsequent call from Site C will not be allowed due to b/w limitation 
over that trunk (Location). Next call should fall back to Pub trunk 
which is having gw-priority 9.



Thanks...


--- On *Sun, 3/29/09, CCIE OSL //* wrote:


From: CCIE OSL 
Subject: Re: [OSL | CCIE_Voice] gatekeeper question
To: "ccie Me" 
Cc: ccie_voice@onlinestudylist.com
Date: Sunday, March 29, 2009, 9:52 AM

I have some questions for you.

1. Are you running on 1 tech-prefix or two with on the gatekeeper?
2. Does entire BR2 has to able to call HQ or just a single phone?
3. Does it required to only use 1#, are you allow to use other
prefixes?

Your first requirement for HQ to BR2 is fairly easy,
However, second requirement, is bit confusing.

I think in order to make that work, I will have to use Hop-off
prefix  and statically map a another prefix to PUB address.
But since the CAC requirement of single call, and If I were to use
bandwidth interzone, I almost need another zone just for PUB,
Which means I may have to use different CAC method. /? or
somehow allow calls to PUB work using hop-off prefix not affected
by GK CAC.???

I have proctor lab coming up on Monday night, I may have to lab
this up.

If you can provide answer to my questions, It may help me to get
this done.

Thanks...

/Jin Jung...



ccie Me wrote:
> Gents,
>
> i'm working on this case on gatekeeper:
>
> i need to only allow ONE active call that should be going
through SUB
> now for  any other new call
> if:
>
> -  it is from HQ to SiteC via GK it will be rejected by GK and
rerouted to pstn via 6608 on HQ
>
> -  if it is from SitC  to HQ via GK it will go through PUB
instead of SUB
>
> i tried to play with regions and CAC on gatekepper and CCM. but
i don't think that will lead to solve this case,
>
> does any body have idea about this
>
> thanks
>

Re: [OSL | CCIE_Voice] gatekeeper question

[CCIE OSL]

I have some questions for you.

1. Are you running on 1 tech-prefix or two with on the gatekeeper?
2. Does entire BR2 has to able to call HQ or just a single phone?
3. Does it required to only use 1#, are you allow to use other prefixes?

Your first requirement for HQ to BR2 is fairly easy,
However, second requirement, is bit confusing.

I think in order to make that work, I will have to use Hop-off prefix  
and statically map a another prefix to PUB address.
But since the CAC requirement of single call, and If I were to use 
bandwidth interzone, I almost need another zone just for PUB,
Which means I may have to use different CAC method. /? or somehow 
allow calls to PUB work using hop-off prefix not affected by GK CAC.???


I have proctor lab coming up on Monday night, I may have to lab this up.

If you can provide answer to my questions, It may help me to get this done.

Thanks...

/Jin Jung...



ccie Me wrote:

Gents,

i'm working on this case on gatekeeper:

i need to only allow ONE active call that should be going through SUB
now for  any other new call
if:

-  it is from HQ to SiteC via GK it will be rejected by GK and 
rerouted to pstn via 6608 on HQ


-  if it is from SitC  to HQ via GK it will go through PUB instead of SUB

i tried to play with regions and CAC on gatekepper and CCM. but i 
don't think that will lead to solve this case,


does any body have idea about this

thanks

Re: [OSL | CCIE_Voice] GK no tech-prefix

[CCIE OSL]

It's really not clear what you are asking here.
But, for me, -- This is just for me of course, I am sure other folks may 
have different mathods.


I believe you must use gw-priority command to give each prefix 2*, 1* 7* 
point to SUB first with priority 10 and PUB with priority 9, or anything 
above 5.


And, there are some special cases where, depending on a question, you 
may have to use hop-off.


So you may have to use both,

/Jin Jung...


Christian Hennrich wrote:

Hi,

if we are only allowed to use 1 zone and 1 tech-prefix, let's say 1#, 
then we can use hopoff command with fixing the ports in the CCM 
Service parameter, or we can use gw-priority.


With gw-priority I can select, for example sub first and pub second, 
that is not possible with hopoff.


So is there any difference then what I have mentioned, what is your 
preferred command in this case?


Any answer is very much appreciated.
Christian

[OSL | CCIE_Voice] Just scheduled for June 26, RTP ---

[CCIE OSL]

Must be my lucky day,,
I just scheduled lab date for June 26, at RTP.

/Jin Jung...

Re: [OSL | CCIE_Voice] NM-CUE verses NME-CUE

[CCIE OSL]

Thanks...



Mark Holloway wrote:
Differences Between the AIM-CUE and the NM-CUE and NME-CUE Modules 


Cisco Unity Express is supported on the advanced integration module
(AIM-CUE), the network module and extended capacity network module (NM-CUE
and NM-CUE-EC), and the enhanced network module (NME-CUE). Cisco Unity
Express features work the same way on these modules with the following
exceptions: 

.Physical differences: 


-The AIM-CUE is a 6-port module with 1GB flash memory that stores a maximum
of 50 voice mailboxes and 14 hours of voice messages. 


-The NM-CUE is an 8-port module that stores a maximum of 100 voice mailboxes
and 100 hours of voice messages. 


-The NM-CUE-EC is a 16-port module that stores a maximum of 250 voice
mailboxes and 300 hours of voice messages. 


-The NME-CUE is a 24-port module that stores a maximum of 250 voice
mailboxes and 300 hours of voice messages. 


.A trace or log command used on the NM-CUE, NM-CUE-EC or NME-CUE
automatically saves the data to the disk. On the AIM-CUE, the trace and log
data are not saved to flash memory. A Cisco Unity Express CLI command is
available to save the data to the AIM-CUE flash memory. 







-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of CCIE OSL
Sent: Friday, March 27, 2009 12:49 PM
To: OSL Group
Subject: [OSL | CCIE_Voice] NM-CUE verses NME-CUE



I am trying to find a comparison document between NM0CUE and NME-CUE.

I been looking cisco site without any luck.

I see that the new V3.0 now have NME_CUE.
I need to figure out if I need to purchse NME-CUE or stay with NM-CUE.
This is for the CCIE voice lab. Not for a production.

Can someone tell is if this is absolute required purchurse item that I 
must have and there are much more functionality or testable items on 
NME-CUE that NM-CUE?


Thanks

  

[OSL | CCIE_Voice] NM-CUE verses NME-CUE

[CCIE OSL]

I am trying to find a comparison document between NM0CUE and NME-CUE.

I been looking cisco site without any luck.

I see that the new V3.0 now have NME_CUE.
I need to figure out if I need to purchse NME-CUE or stay with NM-CUE.
This is for the CCIE voice lab. Not for a production.

Can someone tell is if this is absolute required purchurse item that I 
must have and there are much more functionality or testable items on 
NME-CUE that NM-CUE?


Thanks

Re: [OSL | CCIE_Voice] FRF.12+ shaping

[CCIE OSL]

On the same token,
Does anyone one have right way to test FRF.12 LFI, MLP ... ?

I tried wireshark to capture packets, but it only captures from 
Ethernet. and its not showing up correctly on ethennet side.


The only other way sis to use T-bird or smart-bit which I do not have.

Dobugs really does not show me much, unless you tell me denbug command I 
am missing.


How do you guys test this??



Gughan Gug wrote:


Hi,
 
I have configured LLQ and FRF.12 LFI on my router, How to do shaping 
along with FRF.12?
 
Gughan



Add more friends to your messenger and enjoy! Invite them now. 

[OSL | CCIE_Voice] 08-Apr-2009 ( 1 seats available) open RTP ---

[CCIE OSL]

Re: [OSL | CCIE_Voice] CME- Re-route the call when we somehowrejectthe call from GK

[CCIE OSL]

Thanks..

I will lab it up and try it.

/Jin Jung...



Alex wrote:

AFAIK, tech-prefix cannot contain wildcards.
So to use static mapping you have to either:
1/ map every unwanted full DN on GK to a nonexistent IP@
2/ figure out whether your unwanted DN start with common digit string 
and configure shorter tech-prefix(es). Example:
Unwanted DNs are 1212221, 12122216777, 12122218766, 12122218745. 
Then configure the following:

!
gatekeeper
gw-type-prefix 12122216 gw ipaddr 
gw-type-prefix 121222187 gw ipaddr 
!
To make sure you did not blackhole the rest of DNs in the 12122216XXX 
range, prepend them with CCM tech-prefix (e.g. 2#) when sending ARQ to 
the GK.
And don't prepend the 1212221 and 12122216777 when sending ARQ to 
GK. Use specific CME dialpeers to achieve that:

!
dial-peer voice voi 12126 voip
destination-pattern 12122216[67][67][67]
sess target ras
!
dial-peer voice voi 1212 voip
destination-pattern 12122216...
sess target ras
tech-prefix 2#
!
I don't have the lab handy to test this. Would you be able to help?
Rgds
Alex



- Original Message ----- From: "CCIE OSL" 
To: "Alex" 
Cc: "anil batra" ; 
Sent: Friday, March 27, 2009 1:14 PM
Subject: Re: [OSL | CCIE_Voice] CME- Re-route the call when we 
somehowrejectthe call from GK




Hi Alex,
But that seems like its for only for one DN,  Do you think this will 
work for say 121?


I guess, prefix will be 1212,

I am trying to make similar concept work,
I was trying Hop-off  prefix . I guess this is same thing.

Did this work for you??

And I guess, this will work for reverse direction as well, if we were 
to make a call from HQ to BR2.




Alex wrote:
AFAIK, the simplest way would be to configure static mapping on 
gatekeeper:

!
gatekeeper
gw-type-prefix 1212221 gw ipaddr 
!
Then for a DNIS == 1212221 the GK will return nonexisting i...@. 
Remember, there is no fixed separator between tech-prefix and E.164 
number, so the whole DNIS == 1212221 will be regarded as 
tech-prefix.
CME will exhaust its H.225 SETUP timeout (15 secs by default) 
towards nonexisting IP@ and fall back to PSTN dialpeer.

To make it faster, configure voice class h323 on CME:
!
voice class h323 1
h225 timeout setup 3
h225 timeout tcp establish 3
!
and of course assign this voice class to outbound CME dialpeer.
All other DNIS should go through assuming you configure and register 
shorter tech-prefix on called side (I assume it's CCM).

Rgds
Alex

- Original Message -
*From:* anil batra <mailto:anil...@yahoo.com>
*To:* ccie_voice@onlinestudylist.com
<mailto:ccie_voice@onlinestudylist.com> ; Cliff McGlamry
<mailto:cl...@mcglamry.net>
*Sent:* Friday, March 27, 2009 3:59 AM
*Subject:* Re: [OSL | CCIE_Voice] CME- Re-route the call when we
somehowrejectthe call from GK

Cliff, I have calls going thru HQ to CME and vice versa via GK. So
no issues at all.

Now I want to know if there is somehow I can do something on GK(
without shut down or CAC) to reject the call so that call from CME
will go thru PSTN

Hope I myself clear this time :)

--- On *Fri, 3/27/09, Cliff McGlamry /mailto:cl...@mcglamry.net>>/* wrote:


From: Cliff McGlamry mailto:cl...@mcglamry.net>>
Subject: Re: [OSL | CCIE_Voice] CME- Re-route the call when we
somehow rejectthe call from GK
To: "anil batra" ,
ccie_voice@onlinestudylist.com
Date: Friday, March 27, 2009, 8:13 AM

Anil,
 Sorry, but I don't understand your question.  I'll try to
expand a little since that's what it appears you are looking 
for.

 IF everything is perfectly configured and it still doesn't
work, then one of the following may be required:
 Restart of CCM Service (both servers)
 Verify Transcoders are UP and in the correct LOCATION and MRGL
(if they are up but put in the wrong locations, things 
break. I've had this happen, and it's maddening to find!).

 You may possibly need to reboot CME.
 Make sure gatekeeper has been issued a NO SHUT.  Sounds dumb,
but it actually is a problem quite often. Cliff

- Original Message -
*From:* anil batra 
*To:* ccie_voice@onlinestudylist.com
 ; Cliff
McGlamry 
*Sent:* Thursday, March 26, 2009 10:20 PM
*Subject:* Re: [OSL | CCIE_Voice] CME- Re-route the call
when we somehow rejectthe call from GK

Hi Cliff,

Thanks for your kind response. I don't have any confi with
me but what I am sayign is if you supposedly I have
everything working that is Calls passing to and from HQ to
CME via GK.  But now I suddenly want the scenario I
discussed in my last mail...how to ac

Re: [OSL | CCIE_Voice] CME & GK Registration

[CCIE OSL]

Reboot the router,
This happens to me almost every time.

/Jin Jung...

Scott ODonnell wrote:
Has anyone run into the problem where the CME dn's are registering 
even though you have no-reg configured?
I've tried a couple of different IOS versions and I see the problem 
show up randomly.


- Scott

Re: [OSL | CCIE_Voice] CCIE written exam

[CCIE OSL]

You can contact any VUE or Sylvin testing centers for written just like 
your CCVP exams.


Once you passed the written, yon go to this site below to schedule a lab.

https://tools.cisco.com/CCIE/Schedule_Lab/

/Jin Jung...



Santi Cuni wrote:

Hi
I would like to take CCIE Voice certifcation(before written and then 
Lab). I do not know where e ehich tool I have to use in order to take 
the written exam. Could you helpo me? I have already CCVP, and also 
UCCXD, and I have 2 years of experience about UC area, working with a 
Cisco gold partner. Also is it convient for me to do the exam after 
july becasue I read that will be a chenge?

Thanks for your help
Regards 

 
Santi Cuni
 
"Raggio divino al mio pensiero apparve,donna la tua beltà."
"Or questa egli non già,ma quella,ancora nei corporali 
amplessi,inchina ed ama.Alfin l'errore e gli scambiati oggetti 
conoscendo s'adira;e spesso incolpa la donna a torto" (Aspasia G.Leopardi)

Re: [OSL | CCIE_Voice] CME- Re-route the call when we somehowrejectthe call from GK

[CCIE OSL]

Hi Alex,
But that seems like its for only for one DN,  Do you think this will 
work for say 121?


I guess, prefix will be 1212,

I am trying to make similar concept work,
I was trying Hop-off  prefix . I guess this is same thing.

Did this work for you??

And I guess, this will work for reverse direction as well, if we were to 
make a call from HQ to BR2.




Alex wrote:
AFAIK, the simplest way would be to configure static mapping on 
gatekeeper:

!
gatekeeper
gw-type-prefix 1212221 gw ipaddr 
!
Then for a DNIS == 1212221 the GK will return nonexisting i...@. 
Remember, there is no fixed separator between tech-prefix and E.164 
number, so the whole DNIS == 1212221 will be regarded as tech-prefix.
CME will exhaust its H.225 SETUP timeout (15 secs by default) towards 
nonexisting IP@ and fall back to PSTN dialpeer.

To make it faster, configure voice class h323 on CME:
!
voice class h323 1
h225 timeout setup 3
h225 timeout tcp establish 3
!
and of course assign this voice class to outbound CME dialpeer.
All other DNIS should go through assuming you configure and register 
shorter tech-prefix on called side (I assume it's CCM).

Rgds
Alex
 


- Original Message -
*From:* anil batra 
*To:* ccie_voice@onlinestudylist.com
 ; Cliff McGlamry

*Sent:* Friday, March 27, 2009 3:59 AM
*Subject:* Re: [OSL | CCIE_Voice] CME- Re-route the call when we
somehowrejectthe call from GK

Cliff, I have calls going thru HQ to CME and vice versa via GK. So
no issues at all.

Now I want to know if there is somehow I can do something on GK(
without shut down or CAC) to reject the call so that call from CME
will go thru PSTN

Hope I myself clear this time :)

--- On *Fri, 3/27/09, Cliff McGlamry /mailto:cl...@mcglamry.net>>/* wrote:


From: Cliff McGlamry mailto:cl...@mcglamry.net>>
Subject: Re: [OSL | CCIE_Voice] CME- Re-route the call when we
somehow rejectthe call from GK
To: "anil batra" ,
ccie_voice@onlinestudylist.com
Date: Friday, March 27, 2009, 8:13 AM

Anil,
 
Sorry, but I don't understand your question.  I'll try to

expand a little since that's what it appears you are looking for.
 
IF everything is perfectly configured and it still doesn't

work, then one of the following may be required:
 
Restart of CCM Service (both servers)
 
Verify Transcoders are UP and in the correct LOCATION and MRGL
(if they are up but put in the wrong locations, things break. 
I've had this happen, and it's maddening to find!).
 
You may possibly need to reboot CME.
 
Make sure gatekeeper has been issued a NO SHUT.  Sounds dumb,
but it actually is a problem quite often. 
 
Cliff
 


- Original Message -
*From:* anil batra 
*To:* ccie_voice@onlinestudylist.com
 ; Cliff
McGlamry 
*Sent:* Thursday, March 26, 2009 10:20 PM
*Subject:* Re: [OSL | CCIE_Voice] CME- Re-route the call
when we somehow rejectthe call from GK

Hi Cliff,

Thanks for your kind response. I don't have any confi with
me but what I am sayign is if you supposedly I have
everything working that is Calls passing to and from HQ to
CME via GK.  But now I suddenly want the scenario I
discussed in my last mail...how to achive that  please..

--- On *Fri, 3/27/09, Cliff McGlamry />/* wrote:


From: Cliff McGlamry >
Subject: Re: [OSL | CCIE_Voice] CME- Re-route the call
when we somehow rejectthe call from GK
To: "anil batra" ,
ccie_voice@onlinestudylist.com
Date: Friday, March 27, 2009, 7:47 AM

Several things.
 
1.  Either you're missing something in the gatekeeper

config call routing wise
2.  You're missing something in the CallManager config
call routing wise
3.  You're missing something on CME call routing wise
4.  You have a codec mismatch and/or you don't have
transcode resources available where required
5.  You have a problem with locations bandwidth on CCM
6.  You need an MTP and don't have it configured.
 
If you want a more specific answer, start by posting

your gatekeeper config and the output trace of:
 
debug gatekeeper main 10

debug gatekeeper call 10
 
These should serve to isolate whether the issue

Re: [OSL | CCIE_Voice] New LAB Scheduling

[CCIE OSL]

I have 11/16. RTP.
For now, unless I see another June date.



Kirchhof Björn wrote:


Hi,

is there anybody who is planning to take the new lab?

I've scheduled my first try for November.

Regards.

Bjoern

[OSL | CCIE_Voice] LAb date open San Jose,,

[CCIE OSL]

2 seats in San Jose for Voice for April 15 open right now.

Re: [OSL | CCIE_Voice] CME REGISTRATION

[CCIE OSL]

On CCM, Do you have global service option for 150 or do you have one for 
each scope?
You can put 150 in two different places, make sure you have this set for 
each scope if you trying to get IP / TFTP from CCM for all sites.


Also,,
Have you clear / reset phones on BR2?



adefila seun wrote:

Hello,
Pointing ip helper address to the subscriber and publisher is to get 
the ip address(dhcp)
 
option 150 on the publisher points to the CME router



--- On *Tue, 24/3/09, Cliff McGlamry //* wrote:


From: Cliff McGlamry 
Subject: Re: [OSL | CCIE_Voice] CME REGISTRATION
To: "adefila seun" , "Cristobal Priego"

Cc: ccie_voice@onlinestudylist.com
Date: Tuesday, 24 March, 2009, 4:07 PM

Then that's where it will register.  It has to point to CME if you
want it to register there.
 
Do to a Proctor labs issue, it may not change once you point it

back and reboot them.  if so, you'll have to get support to do a
factory reset on the phones since you can't touch the ones in the
rack directly.

- Original Message -
*From:* adefila seun


*To:* Cristobal Priego


*Cc:* ccie_voice@onlinestudylist.com



*Sent:* Tuesday, March 24, 2009 10:53 AM
*Subject:* Re: [OSL | CCIE_Voice] CME REGISTRATION

Hello
 
it points to the subscriber and pubisher
 
Thanks


--- On *Tue, 24/3/09, Cristobal Priego
/http://uk.mc12.mail.yahoo.com/mc/compose?to=cristobalpri...@gmail.com>>/*
wrote:


From: Cristobal Priego http://uk.mc12.mail.yahoo.com/mc/compose?to=cristobalpri...@gmail.com>>
Subject: Re: [OSL | CCIE_Voice] CME REGISTRATION
To: "adefilabi...@yahoo.co.uk

"
http://uk.mc12.mail.yahoo.com/mc/compose?to=adefilabi...@yahoo.co.uk>>
Cc: "ccie_voice@onlinestudylist.com

"
http://uk.mc12.mail.yahoo.com/mc/compose?to=ccie_vo...@onlinestudylist.com>>
Date: Tuesday, 24 March, 2009, 3:00 PM

Did you check your ip helper address?
Where is it poiniting to?



On Mar 24, 2009, at 1:01 AM, adefilabi...@yahoo.co.uk


wrote:

Hello All, 
 
I am using ip expert rack sessions.
 
i configured the dhcp scope on the publisher for the HQ,

brach 1 and branch 2 and set the tftp server  for branch
2 to be the ip address of the CME router but all the
phones registered to the callmanager ncluding the phones
for branch 2
 
Has any one experience this
 
Your support is needed
 
i need to know if i did somehing wrong
 
Regards
 
i will apreciate it if i can get help
 
there is need for technical help during sessions
 

Re: [OSL | CCIE_Voice] New V3 Version

[CCIE OSL]

Thanks ...



Larry Hadrava wrote:

Here is what the blue print tells us:
 
https://cisco.hosted.jivesoftware.com/docs/DOC-3641
 



CCIE Voice Equipment and Software List - v3.0

VERSION 8 Published


  Created on: Nov 23, 2008 8:58 PM by Adrie Finch - Cisco - Last
  Modified:  Dec 1, 2008 11:36 PM by Adrie Finch - Cisco


  CCIE Voice Lab 3.0 Equipment and Software Versions

Passing the CCIE Voice Lab Exam requires a depth of understanding 
difficult to obtain without hands-on experience. Early in your 
preparation, you should arrange access to the equipment listed below:



*Lab Equipment:*

* Cisco MCS-7845 Media Convergence Servers
* Cisco 3825 Series Integrated Services Routers (ISR)
* Cisco 2821 Series Integrated Services Routers (ISR)
* ISR Modules and Interface Cards

+ VWIC2-1MFT-T1/E1
+ PVDM2
+ HWIC-4ESW-POE
+ NME-CUE

* Cisco Catalyst 3750 Series Switches
* IP Phones and Soft Clients


*Software Versions*

Any major software release which has been generally available for six 
months is eligible for testing in the CCIE Voice Lab Exam.


* Cisco Unified Communications Manager 7.0
* Cisco Unified Communications Manager Express 7.0
* Cisco Unified Contact Center Express 7.0
* Cisco Unified Presence 7.0
* Cisco Unity Connection 7.0
* All routers use IOS version 12.4T Train.
* Cisco Catalyst 3750 Series Switches uses 12.2 Main Train


*Network Interfaces*

* Fast Ethernet
* Frame Relay


*Telephony Interfaces*

* T1
* E1

 


Larry Hadrava
CCIE #12203 CCNP CCNA
Sr. Support Engineer – IPexpert, Inc.
URL: http://www.IPexpert.com


On Mon, Mar 23, 2009 at 12:08 AM, CCIE OSL <mailto:ccie...@gmail.com>> wrote:




Does anyone has exact version of new V3.0?

Thanks...

[OSL | CCIE_Voice] Connect router with FXS to Proctorslab??

[CCIE OSL]

I have a simple question,
Say I want to connect my 1760 with FXS port to ProctorLabs Vrack.

Is there a way I can connect this router?

I have 3640 router providing VPN,
I just want to ride the VPN but connect instead of phones, 1760  router 
with FXS.

and test SIP or MGCP.

Anyone done this before?

Please let me know,,

Thanks...

[OSL | CCIE_Voice] New V3 Vesion

[CCIE OSL]

Does anyone has exact version of new V3.0?

Thanks...

Re: [OSL | CCIE_Voice] Gatekeeper CAC - call no longer works

[CCIE OSL]

Yep,,
My Trunk was in HQ.

Thanks...


Christian Hennrich wrote:
hi, your device pool of the GK Trunk needs a g729 only region. Ensure, 
that there is definitively no g711 relationship on the region used on 
your GK Trunk.


HTH

CCIE OSL schrieb:


I just activated bandwidth for my gatekeeper,
bandwidht interzone CME 64,
My calls from HQ to BR2 is no longer working.

Check  location, region, and other  settings, still does not work.

where should I look??

/Jin Jung...

__
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[OSL | CCIE_Voice] Gatekeeper CAC - call no longer works

[CCIE OSL]

I just activated bandwidth for my gatekeeper,
bandwidht interzone CME 64,  


My calls from HQ to BR2 is no longer working.

Check  location, region, and other  settings, still does not work.

where should I look??

/Jin Jung...

Re: [OSL | CCIE_Voice] How to only allow one international or LD call?

[CCIE OSL]

Thanks folks,,



kapil atrish wrote:

Put max-conn under dial-peer.

--- On *Fri, 3/13/09, CCIE OSL //* wrote:


From: CCIE OSL 
Subject: [OSL | CCIE_Voice] How to only allow one international or
LD call?
To: "OSL Group" 
Date: Friday, March 13, 2009, 12:56 AM



Question
How do you only allow one international or LD all any givin time?
This is CAC question or COR,, or both??

[OSL | CCIE_Voice] How to only allow one international or LD call?

[CCIE OSL]

Question
How do you only allow one international or LD all any givin time?
This is CAC question or COR,, or both??

[OSL | CCIE_Voice] how to make CME phone to have same number on button 1 and 2?

[CCIE OSL]

how to make CME phone to have same number on button 1 and 2?

Is there a way to make 3001 appear on both button 1 and 2?
I am not talking about description field.

So it will show up

6727653001  <--description
   3001  <-- button 1
   3001  <-- button 2

/Jin Jung...

Re: [OSL | CCIE_Voice] voice translation to strip8

[CCIE OSL]

Yep,
I must be falling a sleep,,


Thanks...



anil batra wrote:
 
 
 
voice translation-profile strip8

translate calling 37 >>>>>>>>>>>>>>>>>>>Should not this be "Called"
 
 



--- On *Thu, 3/12/09, CCIE OSL //* wrote:

From: CCIE OSL 
Subject: Re: [OSL | CCIE_Voice] voice translation to strip8
To: "OSL Group" 
Date: Thursday, March 12, 2009, 8:45 AM

    It look like I can use num-exp,
but why not voice translation


CCIE OSL wrote:
>
> I must be missing something,
> Why is "8" not getting strip by  voice translation
rule
>
> Debug
> assrv_get_addrinfo: (1#82002) Matched tech-prefix 1#
> Mar 12 04:08:57.692: rassrv_get_addrinfo: (1#82002) unresolved zone 
> prefix, using source zone CME

>
> dial-peer voice 2001 voip
> translation-profile outgoing strip8
> destination-pattern 8[23]...
> session target ras
> tech-prefix 1#
> !
> voice translation-rule 37
> rule 1 /8\(2...\)/ /\1/
> rule 2 /8\(3...\)/ /\1/
> !
> voice translation-profile strip8
> translate calling 37
>
>
>
>

  

Re: [OSL | CCIE_Voice] voice translation to strip8

[CCIE OSL]

It look like I can use num-exp,
but why not voice translation


CCIE OSL wrote:


I must be missing something,
Why is "8" not getting strip by  voice translation rule

Debug
assrv_get_addrinfo: (1#82002) Matched tech-prefix 1#
Mar 12 04:08:57.692: rassrv_get_addrinfo: (1#82002) unresolved zone 
prefix, using source zone CME


dial-peer voice 2001 voip
translation-profile outgoing strip8
destination-pattern 8[23]...
session target ras
tech-prefix 1#
!
voice translation-rule 37
rule 1 /8\(2...\)/ /\1/
rule 2 /8\(3...\)/ /\1/
!
voice translation-profile strip8
translate calling 37

[OSL | CCIE_Voice] voice translation to strip8

[CCIE OSL]

I must be missing something,
Why is "8" not getting strip by  voice translation rule

Debug
assrv_get_addrinfo: (1#82002) Matched tech-prefix 1#
Mar 12 04:08:57.692: rassrv_get_addrinfo: (1#82002) unresolved zone 
prefix, using source zone CME


dial-peer voice 2001 voip
translation-profile outgoing strip8
destination-pattern 8[23]...
session target ras
tech-prefix 1#
!
voice translation-rule 37
rule 1 /8\(2...\)/ /\1/
rule 2 /8\(3...\)/ /\1/
!
voice translation-profile strip8
translate calling 37

Re: [OSL | CCIE_Voice] access to 6509 proctorlabs.com

[CCIE OSL]

I guess, we all have to go thru same amount of pain before reaching the 
final destination.

/Jin Jung..


Mike Brooks wrote:
Yes, its good to see that I am not the only one that experiences 
this.  I do as Vik suggested and configure the 6500 from the CCM. 
 
Mike Brooks

CCIE# 16027 (R&S)

On Wed, Mar 11, 2009 at 11:17 AM, Vik Malhi <mailto:vma...@ipexpert.com>> wrote:


This is a nice feature of catos that affects us all from time to
time. Configure the port by referencing the table. Or try telnet
from the ccm.

Vik Malhi - CCIE#13890
Senior Technical Instructor - IPexpert Inc

Telephone: +1.810.326.1444
Fax: +1.810.454.0130
Mailto: vma...@ipexpert.com <mailto:vma...@ipexpert.com>

Join IPexpert's Free CCIE Peer Groups & Study Communities at
www.IPexpert.com/communities <http://www.ipexpert.com/communities>


    On Mar 10, 2009, at 7:51 PM, CCIE OSL mailto:ccie...@gmail.com>> wrote:

This may sound strange,
For some reason, I was not able to type in a command in 6509.
I was able to log into 6509, but when I put in sh port or sh
vlan, it freezes and will not come back.
This is from my 3640 EZVPN router.

So, I tried via regular Cisco VPN, it work fine.

I am using standard ACL from proctorlabs.com
<http://proctorlabs.com/>
///??

Password:
VOICE-6500-1> (enable)
VOICE-6500-1> (enable)
VOICE-6500-1> (enable)
VOICE-6500-1> (enable)
VOICE-6500-1> (enable)
VOICE-6500-1> (enable)
VOICE-6500-1> (enable)
VOICE-6500-1> (enable)
VOICE-6500-1> (enable)
VOICE-6500-1> (enable)
VOICE-6500-1> (enable)
VOICE-6500-1> (enable) sh vlan

its stuck, no output,

/Jin Jung...

[OSL | CCIE_Voice] strip tech prefix in CCM

[CCIE OSL]

Where is the proper place to strip tech prefix in call manager?

??

[OSL | CCIE_Voice] access to 6509 proctorlabs.com

[CCIE OSL]

This may sound strange,
For some reason, I was not able to type in a command in 6509.
I was able to log into 6509, but when I put in sh port or sh vlan, it 
freezes and will not come back.

This is from my 3640 EZVPN router.

So, I tried via regular Cisco VPN, it work fine.

I am using standard ACL from proctorlabs.com
///??

Password:
VOICE-6500-1> (enable)
VOICE-6500-1> (enable)
VOICE-6500-1> (enable)
VOICE-6500-1> (enable)
VOICE-6500-1> (enable)
VOICE-6500-1> (enable)
VOICE-6500-1> (enable)
VOICE-6500-1> (enable)
VOICE-6500-1> (enable)
VOICE-6500-1> (enable)
VOICE-6500-1> (enable)
VOICE-6500-1> (enable) sh vlan

its stuck, no output,

/Jin Jung...

Re: [OSL | CCIE_Voice] gatekeeper E164 ao-reg both address registration

[CCIE OSL]

Yep,,
It works,

Thanks...

/Jin Jung...


Vik Malhi wrote:

You need no-reg after dialplan pattern within telephony-service.

Vik Malhi - CCIE#13890
Senior Technical Instructor - IPexpert Inc

Telephone: +1.810.326.1444
Fax: +1.810.454.0130
Mailto: vma...@ipexpert.com

Join IPexpert's Free CCIE Peer Groups & Study Communities at 
www.IPexpert.com/communities


On Mar 10, 2009, at 5:29 PM, CCIE OSL  wrote:



I put in "no-reg both" all ephone-dn for BR2.
But I am still getting
E164-ID: 85221314001
  E164-ID: 85221314002
  E164-ID: 85221314003
  E164-ID: 85221314004
  E164-ID: 85221314005
  E164-ID: 852213140
  E164-ID: 85221314007
  E164-ID: 85221314008

registered to my gatekeeper from BR2.

Any idea how to not to register these numbers?
/Jin Jung...

Re: [OSL | CCIE_Voice] gatekeeper E164 ao-reg both address registration

[CCIE OSL]

Here it is,,


!
ephone-dn  1  dual-line
number 4001 no-reg both
call-forward busy 4100
call-forward noan 4100 timeout 12
!
!
ephone-dn  2  dual-line

number 4002 no-reg both
call-forward busy 4100
call-forward noan 4100 timeout 12
!
!
ephone-dn  3  dual-line
number 4003 no-reg both
call-forward busy 4100
call-forward noan 4100 timeout 12
!
!
ephone-dn  4  dual-line
number 4004 no-reg both
call-forward busy 4100
call-forward noan 4100 timeout 12
!
!
ephone-dn  5  dual-line
number 4005 no-reg both
call-forward busy 4100
call-forward noan 4100 timeout 12
!
!

ephone-dn  6  dual-line
number 4006 no-reg both
call-forward busy 4100
call-forward noan 4100 timeout 12
!
!
ephone-dn  7  dual-line
number 4007 no-reg both
call-forward busy 4100
call-forward noan 4100 timeout 12
!
!
ephone-dn  8  dual-line
number 4008 no-reg both
call-forward busy 4100
call-forward noan 4100 timeout 12
!
!
ephone-dn  9  dual-line
number 4009 no-reg both
call-forward busy 4100
call-forward noan 4100 timeout 12
!
!
ephone-dn  10  dual-line
number 4010 no-reg both
call-forward busy 4100
call-forward noan 4100 timeout 12
!
!
ephone-dn  11  dual-line
number 4011 no-reg both
call-forward busy 4100
call-forward noan 4100 timeout 12
!
!
ephone-dn  12  dual-line
number 4012 no-reg both
call-forward busy 4100
call-forward noan 4100 timeout 12
!
!
ephone  1
mac-address 0013.19C0.4C81
type 7960
button  1:2
!
!
!
ephone  2
mac-address 0013.1ADB.3977
type 7940
button  1:1
!
!
!
ephone  3
mac-address 0014.A812.8F2B
type 7960
button  1:3
!
!
!
ephone  4
!
!
!
Pod18-BR2-RTR#sh gateway
H.323 ITU-T Version: 4.0   H323 Stack Version: 0.1
!
H.323 service is up
Gateway  BR2  is registered to Gatekeeper CME

Alias list (CLI configured)
E164-ID 85221314001
E164-ID 85221314002
E164-ID 85221314003
E164-ID 85221314004
E164-ID 85221314005
E164-ID 85221314006
E164-ID 85221314007
E164-ID 85221314008
E164-ID 85221314009
E164-ID 85221314010
E164-ID 85221314011
E164-ID 85221314012
H323-ID BR2
Alias list (last RCF)
E164-ID 85221314001
E164-ID 85221314002
E164-ID 85221314003
E164-ID 85221314004
E164-ID 85221314005
E164-ID 85221314006
E164-ID 85221314007
E164-ID 85221314008
E164-ID 85221314009
E164-ID 85221314010
E164-ID 85221314011
E164-ID 85221314012
H323-ID BR2

H323 resource thresholding is Disabled
Pod18-BR2-RTR#

--More-


Cliff McGlamry wrote:

Try rebooting the BR2 router.

If that still doesn't work, Post the BR2 config and the output of the SHOW 
GATEWAY command on the BR2 router.



- Original Message - 
From: "CCIE OSL" 

To: 
Sent: Tuesday, March 10, 2009 8:29 PM
Subject: [OSL | CCIE_Voice] gatekeeper E164 ao-reg both address registration



I put in "no-reg both" all ephone-dn for BR2.
But I am still getting
E164-ID: 85221314001
E164-ID: 85221314002
E164-ID: 85221314003
E164-ID: 85221314004
E164-ID: 85221314005
E164-ID: 852213140
E164-ID: 85221314007
E164-ID: 85221314008

registered to my gatekeeper from BR2.

Any idea how to not to register these numbers?
/Jin Jung...



  

[OSL | CCIE_Voice] gatekeeper E164 ao-reg both address registration

[CCIE OSL]

I put in "no-reg both" all ephone-dn for BR2.
But I am still getting
E164-ID: 85221314001
   E164-ID: 85221314002
   E164-ID: 85221314003
   E164-ID: 85221314004
   E164-ID: 85221314005
   E164-ID: 852213140
   E164-ID: 85221314007
   E164-ID: 85221314008

registered to my gatekeeper from BR2.

Any idea how to not to register these numbers?
/Jin Jung...

Re: [OSL | CCIE_Voice] FXS restricted without using COR

[CCIE OSL]

CCIE OSL wrote:


Quick question,

I am trying to configure my FXS port to only allow call local and 911.


I know how to do it with COR,

But I do not want to create COR, it seems bit too much for simple 
restriction for PSTN phone.


Is there any other way to restrict this phone with out using COR?

Thanks..


/Jin Jung...

Re: [OSL | CCIE_Voice] IP PIM

[CCIE OSL]

PIM Dence mode is more for local area network or samll size network, 
more of broadcast, push. Not recommended for WAN


PIM sparce-mode is recommended for traffic over WAN. OR LAN.

ip pim sparce-dense mode activates both space and dence mode - sparce 
first and dense,
This command is mainly used for Auto-RP, where you can have multiple RPs 
and selection process to find RP use dense mode, once RP is selected, it 
uses sparce mode for actual multicast.


/Jin Jung...

Ryan Trauernicht wrote:
Dense mode is a push method of doing multicast and sparse dense is a 
pull method of doing multicast.


On Tue, Mar 3, 2009 at 12:56 PM, hasan khalife 
mailto:hasan_khal...@hotmail.com>> wrote:


WHAT IS THE DIFFERENEC BTW IP PIM DENSE-MODE
 
 
IP PIM SPARSE-DENSE-MODE



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