Re: [OSL | CCIE_Voice] translation rule

[Ashar Siddiqui]

Thanks Roger! It's a bit clear now..appreciate that.

Ash>

Roger Källberg wrote:

  
  
  
  Hi Ash,
  There are
time when you would need to use that match pattern, for example see
this tread on NetPro,
  https://supportforums.cisco.com/thread/2013151
   
   
  
  
  Roger Källberg
  Consultant
  Cygate AB
  
  
  
  
  Från: Ashar
Siddiqui [siddas...@gmail.com]
  Skickat: den 21 maj 2010 22:58
  Till: David Holman
  Kopia: ccie_voice@onlinestudylist.com
  Ämne: Re: [OSL | CCIE_Voice] translation rule
  
  
  Thanks David I have already gone through that document many
times :)
  
Thanks Wael for your explanation. I was actually thinking that what
would a null number be in my example. I have this customer router which
has this specific rule for inbound calls. Will it work when PSTN will
send "null digits"? doesn't make sense to me. What is a null/unknown
digit?
  
Ash>
  
David Holman wrote:
  I keep this link handy for voice translation
questions:  

http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080325e8e.shtml

On Fri, May 21, 2010 at 4:15 PM, Wael
Agina 

wrote:

  Dear Ashar,
  
  The ^$ is catching null, which could be used to catch calls from
unkown.
example usage, drop any calls from PSTN that has ANI of unkown type.
On H323 you could use following rule to do this
  
voice translation-rule 1
 rule 1 reject /^$/ 
  
voice translation-profile Drop-Unknown
  translate calling 1
  
dial-peer voice 1 pots
direct-inward-dial
incom called .
  call-block translation-profile incoming Drop-Unknown
  
For you example may be it i setting unknown ANI to be 42000 for example, bu not sure, need to be tested.
  
Regards,
Wael Agina
  
  
  
  On Fri, May 21, 2010 at 11:02 PM, Ashar Siddiqui 

wrote:
  
  
  


Hi,

I know I may sound stupid to some but I really want to know the purpose
of ^$ in a translation rule for e.g:

voice translation-rule 100
 rule 1 /^$/ /42000/
!


^$ is null...what does it mean? what is a null number?

Ash>




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-- 
  
Thanks and Best Regards,
Wael Agina
  
  
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Re: [OSL | CCIE_Voice] DHCP in CCM on Proctor lab

[Matthew Berry]

Can't say I've ever tried. You might be able to get away using the ip  
helper-address on your local interfaces.



Matthew Berry

**Sent from my iPhone**
Skype/Twitter: ciscovoiceguru
Google Voice: +1 612 424 5044

On May 22, 2010, at 2:14 PM, Erwan Erwan  wrote:


hi,

Is anybody know how to activate DHCP in proctolab CCM  for our home  
IP phones ?

What need to condig in VPN router and Switches ?

tks

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[OSL | CCIE_Voice] DHCP in CCM on Proctor lab

[Erwan Erwan]

hi,
 
Is anybody know how to activate DHCP in proctolab CCM  for our home IP phones ?
What need to condig in VPN router and Switches ?
 
tks


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Re: [OSL | CCIE_Voice] Location based RSVP over dual Frame Relay Links

[Graham Hopkins]

I think we mean the same thing although my use of the term call setup is 
probably not a good one - when the request for bandwidth for call setup is made 
with G729 then 40 kbps is requested - worse case bandwidth for a 10ms sample 
rate. After call established this drops to 24kbps leaving 40kbps available for 
the bandwidth request of the second call. 

I think this is more likely to be a routing issue as the router makes no 
attempt to request bandwidth on the second link


Gateways and debug follow - btw the configs have some legacy stuff from other 
testing - this is the dynamips  version rather than the physical one so are 
7200s


HQ-GW#sh run
Building configuration...

Current configuration : 4800 bytes
!
! Last configuration change at 15:55:33 BST Sat May 22 2010
! NVRAM config last updated at 15:56:06 BST Sat May 22 2010
!
upgrade fpd auto
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname HQ-GW
!
boot-start-marker
boot-end-marker
!
logging message-counter syslog
!
no aaa new-model
clock timezone GMT 0
clock summer-time BST recurring last Sun Mar 1:00 last Sun Oct 1:00
clock calendar-valid
ip source-route
ip cef
!
!
ip dhcp excluded-address 192.168.60.1 192.168.60.9
ip dhcp excluded-address 192.168.60.21 192.168.60.254
!
ip dhcp pool PHONES
   network 192.168.60.0 255.255.255.0
   default-router 192.168.60.1
   option 150 ip 192.168.60.2
!
!
no ip domain lookup
no ipv6 cef
!
multilink bundle-name authenticated
!
!
!
voice service voip
 fax protocol cisco
 h323
  ras rrq dynamic prefixes
!
!
!
voice class codec 1
 codec preference 1 g729r8
 codec preference 2 g711ulaw
!
!
archive
 log config
  hidekeys
!
!
class-map match-all VOIP
 match ip dscp ef
class-map match-any CONTROL
 match ip dscp cs3  af31
class-map match-any AutoQoS-VoIP-RTP-Trust
 match ip dscp ef
class-map match-any AutoQoS-VoIP-Control-Trust
 match ip dscp cs3
 match ip dscp af31
!
!
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
policy-map WAN
 class VOIP
priority percent 25
   compress header ip rtp
 class CONTROL
bandwidth percent 30
 class class-default
fair-queue
!
!
!
!
!
interface Loopback0
 ip address 11.11.11.11 255.255.255.255
 h323-gateway voip interface
 h323-gateway voip id GK1 ipaddr 20.20.20.20 1719
 h323-gateway voip tech-prefix 1#
 h323-gateway voip bind srcaddr 11.11.11.11
!
interface FastEthernet0/0
 ip address 192.168.60.1 255.255.255.0
 duplex half
 speed 100
!
interface FastEthernet0/1
 description to GK
 no ip address
 shutdown
 duplex auto
 speed auto
!
interface Serial1/0
 no ip address
 encapsulation frame-relay
 serial restart-delay 0
 frame-relay traffic-shaping
 frame-relay lmi-type ansi
 ip rsvp bandwidth
!
interface Serial1/0.1 point-to-point
 bandwidth 384
 ip address 10.10.10.1 255.255.255.252
 frame-relay interface-dlci 101
 ip rsvp bandwidth 96
!
interface Serial1/0.2 point-to-point
 bandwidth 768
 ip address 10.10.10.5 255.255.255.252
 frame-relay interface-dlci 102
  class AutoQoS-FR-Se1/0-102
  auto qos voip trust
 frame-relay ip rtp header-compression
!
interface Serial1/0.3 point-to-point
 bandwidth 384
 ip address 10.10.10.9 255.255.255.252
 frame-relay interface-dlci 111
 ip rsvp bandwidth 96
!
interface Serial1/1
 no ip address
 shutdown
 serial restart-delay 0
!
interface Serial1/2
 no ip address
 shutdown
 serial restart-delay 0
!
interface Serial1/3
 no ip address
 shutdown
 serial restart-delay 0
!
interface Serial1/4
 no ip address
 shutdown
 serial restart-delay 0
!
interface Serial1/5
 no ip address
 shutdown
 serial restart-delay 0
!
interface Serial1/6
 no ip address
 shutdown
 serial restart-delay 0
!
interface Serial1/7
 no ip address
 shutdown
 serial restart-delay 0
!
router ospf 1
 log-adjacency-changes
 redistribute connected metric 10 subnets
 network 10.10.10.0 0.0.0.255 area 0
 network 11.11.11.11 0.0.0.0 area 0
 maximum-paths 2
!
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 192.168.200.1
no ip http server
no ip http secure-server
!
!
!
!
map-class frame-relay AutoQoS-FR-Se1/0-102
 frame-relay cir 768000
 frame-relay bc 7680
 frame-relay be 0
 frame-relay mincir 768000
 frame-relay fragment 960
 service-policy output AutoQoS-Policy-Trust
!
map-class frame-relay 384K
 frame-relay cir 364800
 frame-relay bc 3684
 frame-relay be 0
 frame-relay mincir 364800
logging alarm informational
!
!
!
!
!
tftp-server disk0:P00308000500.loads alias P00308000500.loads
tftp-server disk0:P00308000500.sb2 alias P00308000500.sb2
tftp-server disk0:P00308000500.sbn alias P00308000500.sbn
!
control-plane
!
rmon event 3 log trap AutoQoS description "AutoQoS SNMP traps for Voice 
Drops" owner AutoQoS
!
!
!
sccp local Loopback0
sccp ccm 192.168.60.2 identifier 1 version 7.0
sccp
!
sccp ccm group 1
 associate ccm 1 priority 1
 associate profile 1 register HQ-RSVP
!
dspfarm

[OSL | CCIE_Voice] CME Globalization ?

[Mike Brooks]

Is it possible to globalize a "calling" number inbound to a SCCP CME phone ?

On BR2 phone 1, I want the calling number to show up as +12123945001.
Currently it just shows up on the screen as 12123945001.  In the bottom left
of the screen it displays "From: +12123945001", but is it possible for it to
be on the top of the screen and show up in the missed calls directory in the
globalized format ?

By the way in the debug isdn, I am seeing the calling number in globalized
format:

Display i = 'HQ PH1'
Calling Party Number i = 0x0081, *'+12123945001'*
Plan:Unknown, Type:Unknown
Called Party Number i = 0x91, '32143001'
Plan:ISDN, Type:International

I know that an H323 gateway strips the plus sign, but can I add it back as
the calling number gets delivered to the phone ?  I can't find a way to do
it.

Thanks,
Mike
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Re: [OSL | CCIE_Voice] Location based RSVP over dual Frame Relay Links

[Matthew Berry]

Graham,

According to my understanding, the 64 Kbps does not equal 24 Kbps for  
the call and 40 Kbps for setup. Instead, the RSVP reservation always  
calculates the incoming call at the worst-case scenario of 40 Kbps for  
a g.729 call. The remaining 24 Kbps is for call #2.

I am not familiar with lab 5 so I can't speak to the load balanced  
links. Could you send your gateway configs and the "debug ip RSVP  
messages"?

Happy labbing!

Matthew Berry

**Sent from my iPhone**
Skype/Twitter: ciscovoiceguru
Google Voice: +1 612 424 5044

On May 22, 2010, at 10:06 AM, Graham Hopkins  wrote:

> Matthew - the two interfaces S1/0.1 and S1/0.3 are parallel links to  
> the same remote site 96 K is allocated on each of the two links,  
> enough for one call per link. This is based  on Vol 2 Lab 5  
> scenario, according to the proctor guide the first call should use  
> S1/0.1 and the second S1/0.3 but I never get a call on the second  
> link - even if the bandwidth is set to 500K !
>
> The actual example in Vol2 Lab 5 was to allow 4 calls at G.729 and  
> the solution allowed 64K per sub interface ( i.e. 24K plus 40K for  
> call setup) however I could not get more than two calls between the  
> sites in this instance
>
> Regards
>
> Graham Hopkins
>
>
>
> On 22 May 2010, at 15:45, Matthew Berry wrote:
>
>> You are not allocating enough bandwidth for two G711 calls with  
>> RSVP. One at 96 (worst case) and one at 64.
>>
>>
>> Matthew Berry
>>
>> **Sent from my iPhone**
>> Skype/Twitter: ciscovoiceguru
>> Google Voice: +1 612 424 5044
>>
>> On May 22, 2010, at 8:48 AM, Graham Hopkins > rock.co.uk> wrote:
>>
>>> Has anyone got this working/had problems etc. I have two links  
>>> with 96k allocated per link but the second call (both G711) gets  
>>> Not Enough Bandwidth.
>>>
>>> routing is load-sharing
>>>
>>> O E2 192.168.50.0/24 [110/20] via 10.10.10.10, 00:23:33, Serial1/0.3
>>>   [110/20] via 10.10.10.2, 00:23:33, Serial1/0.1
>>>
>>> RSVP call agents are up and registered to CUCM.
>>>
>>> Any ideas ?
>>>
>>> interface Serial1/0.1 point-to-point
>>> bandwidth 384
>>> ip address 10.10.10.1 255.255.255.252
>>> frame-relay interface-dlci 101
>>> ip rsvp bandwidth 96
>>> end
>>>
>>> HQ-GW#sh run int s1/0.3
>>> Building configuration...
>>>
>>> Current configuration : 155 bytes
>>> !
>>> interface Serial1/0.3 point-to-point
>>> bandwidth 384
>>> ip address 10.10.10.9 255.255.255.252
>>> frame-relay interface-dlci 111
>>> ip rsvp bandwidth 96
>>> end
>>>
>>> HQ-GW#
>>>
>>> HQ-GW#sh ip rsvp interface
>>> interfacersvp  allocated  i/f max  flow max sub max
>>> Se1/0ena   80K1158K1158K0
>>> Se1/0.1  ena   80K96K  96K  0
>>> Se1/0.3  ena   0  96K  96K  0
>>>
>>>
>>> dspfarm profile 1 mtp
>>> codec pass-through
>>> codec g711ulaw
>>> rsvp
>>> maximum sessions software 8
>>> associate application SCCP
>>>
>>> O E2 192.168.50.0/24 [110/20] via 10.10.10.10, 00:23:33, Serial1/0.3
>>>   [110/20] via 10.10.10.2, 00:23:33, Serial1/0.1
>>>
>>> Graham Hopkins
>>>
>>>
>>>
>>>
>>> ___
>>> For more information regarding industry leading CCIE Lab training,  
>>> please visit www.ipexpert.com
>
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Re: [OSL | CCIE_Voice] Location based RSVP over dual Frame Relay Links

[Graham Hopkins]

Matthew - the two interfaces S1/0.1 and S1/0.3 are parallel links to the same 
remote site 96 K is allocated on each of the two links, enough for one call per 
link. This is based  on Vol 2 Lab 5 scenario, according to the proctor guide 
the first call should use S1/0.1 and the second S1/0.3 but I never get a call 
on the second link - even if the bandwidth is set to 500K !

The actual example in Vol2 Lab 5 was to allow 4 calls at G.729 and the solution 
allowed 64K per sub interface ( i.e. 24K plus 40K for call setup) however I 
could not get more than two calls between the sites in this instance

Regards

Graham Hopkins



On 22 May 2010, at 15:45, Matthew Berry wrote:

> You are not allocating enough bandwidth for two G711 calls with RSVP. One at 
> 96 (worst case) and one at 64.
> 
> 
> Matthew Berry
> 
> **Sent from my iPhone**
> Skype/Twitter: ciscovoiceguru
> Google Voice: +1 612 424 5044
> 
> On May 22, 2010, at 8:48 AM, Graham Hopkins  wrote:
> 
>> Has anyone got this working/had problems etc. I have two links with 96k 
>> allocated per link but the second call (both G711) gets Not Enough Bandwidth.
>> 
>> routing is load-sharing
>> 
>> O E2 192.168.50.0/24 [110/20] via 10.10.10.10, 00:23:33, Serial1/0.3
>>[110/20] via 10.10.10.2, 00:23:33, Serial1/0.1
>> 
>> RSVP call agents are up and registered to CUCM.
>> 
>> Any ideas ?
>> 
>> interface Serial1/0.1 point-to-point
>> bandwidth 384
>> ip address 10.10.10.1 255.255.255.252
>> frame-relay interface-dlci 101
>> ip rsvp bandwidth 96
>> end
>> 
>> HQ-GW#sh run int s1/0.3
>> Building configuration...
>> 
>> Current configuration : 155 bytes
>> !
>> interface Serial1/0.3 point-to-point
>> bandwidth 384
>> ip address 10.10.10.9 255.255.255.252
>> frame-relay interface-dlci 111
>> ip rsvp bandwidth 96
>> end
>> 
>> HQ-GW#
>> 
>> HQ-GW#sh ip rsvp interface
>> interfacersvp  allocated  i/f max  flow max sub max
>> Se1/0ena   80K1158K1158K0
>> Se1/0.1  ena   80K96K  96K  0
>> Se1/0.3  ena   0  96K  96K  0
>> 
>> 
>> dspfarm profile 1 mtp
>> codec pass-through
>> codec g711ulaw
>> rsvp
>> maximum sessions software 8
>> associate application SCCP
>> 
>> O E2 192.168.50.0/24 [110/20] via 10.10.10.10, 00:23:33, Serial1/0.3
>>[110/20] via 10.10.10.2, 00:23:33, Serial1/0.1
>> 
>> Graham Hopkins
>> 
>> 
>> 
>> 
>> ___
>> For more information regarding industry leading CCIE Lab training, please 
>> visit www.ipexpert.com

___
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Re: [OSL | CCIE_Voice] Location based RSVP over dual Frame Relay Links

[Matthew Berry]

You are not allocating enough bandwidth for two G711 calls with RSVP.  
One at 96 (worst case) and one at 64.


Matthew Berry

**Sent from my iPhone**
Skype/Twitter: ciscovoiceguru
Google Voice: +1 612 424 5044

On May 22, 2010, at 8:48 AM, Graham Hopkins   
wrote:

> Has anyone got this working/had problems etc. I have two links with  
> 96k allocated per link but the second call (both G711) gets Not  
> Enough Bandwidth.
>
> routing is load-sharing
>
> O E2 192.168.50.0/24 [110/20] via 10.10.10.10, 00:23:33, Serial1/0.3
> [110/20] via 10.10.10.2, 00:23:33, Serial1/0.1
>
> RSVP call agents are up and registered to CUCM.
>
> Any ideas ?
>
> interface Serial1/0.1 point-to-point
> bandwidth 384
> ip address 10.10.10.1 255.255.255.252
> frame-relay interface-dlci 101
> ip rsvp bandwidth 96
> end
>
> HQ-GW#sh run int s1/0.3
> Building configuration...
>
> Current configuration : 155 bytes
> !
> interface Serial1/0.3 point-to-point
> bandwidth 384
> ip address 10.10.10.9 255.255.255.252
> frame-relay interface-dlci 111
> ip rsvp bandwidth 96
> end
>
> HQ-GW#
>
> HQ-GW#sh ip rsvp interface
> interfacersvp  allocated  i/f max  flow max sub max
> Se1/0ena   80K1158K1158K0
> Se1/0.1  ena   80K96K  96K  0
> Se1/0.3  ena   0  96K  96K  0
>
>
> dspfarm profile 1 mtp
> codec pass-through
> codec g711ulaw
> rsvp
> maximum sessions software 8
> associate application SCCP
>
> O E2 192.168.50.0/24 [110/20] via 10.10.10.10, 00:23:33, Serial1/0.3
> [110/20] via 10.10.10.2, 00:23:33, Serial1/0.1
>
> Graham Hopkins
>
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training,  
> please visit www.ipexpert.com
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[OSL | CCIE_Voice] AGENT CAD CUSTOMIZATION!!

[Fatai Adekunle]

Hello Guys,

I am working on a contact center solution and need to achieve a task with an 
agent Cisco Agent Desktop(CAD) such that at 7:01 pm, agent is automatically 
logged out of CAD. This require customizing agent CAD through the Desktop 
administration. I have done this but it is not just working. I am working on 
UCCX 7.0.

Any help will be highly appreciated.

Thanks.


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[OSL | CCIE_Voice] Location based RSVP over dual Frame Relay Links

[Graham Hopkins]

Has anyone got this working/had problems etc. I have two links with 96k 
allocated per link but the second call (both G711) gets Not Enough Bandwidth. 

routing is load-sharing

O E2 192.168.50.0/24 [110/20] via 10.10.10.10, 00:23:33, Serial1/0.3
 [110/20] via 10.10.10.2, 00:23:33, Serial1/0.1 

RSVP call agents are up and registered to CUCM.

Any ideas ?

interface Serial1/0.1 point-to-point
 bandwidth 384
 ip address 10.10.10.1 255.255.255.252
 frame-relay interface-dlci 101
 ip rsvp bandwidth 96
end

HQ-GW#sh run int s1/0.3
Building configuration...

Current configuration : 155 bytes
!
interface Serial1/0.3 point-to-point
 bandwidth 384
 ip address 10.10.10.9 255.255.255.252
 frame-relay interface-dlci 111
 ip rsvp bandwidth 96
end

HQ-GW#  



HQ-GW#sh ip rsvp interface
interfacersvp  allocated  i/f max  flow max sub max
Se1/0ena   80K1158K1158K0
Se1/0.1  ena   80K96K  96K  0
Se1/0.3  ena   0  96K  96K  0  


dspfarm profile 1 mtp
 codec pass-through
 codec g711ulaw
 rsvp
 maximum sessions software 8
 associate application SCCP  

O E2 192.168.50.0/24 [110/20] via 10.10.10.10, 00:23:33, Serial1/0.3
 [110/20] via 10.10.10.2, 00:23:33, Serial1/0.1 

Graham Hopkins




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Re: [OSL | CCIE_Voice] translation rule

[Roger Källberg]

Hi Ash,
There are time when you would need to use that match pattern, for example see 
this tread on NetPro, https://supportforums.cisco.com/thread/2013151


Roger Källberg
Consultant
Cygate AB

Från: Ashar Siddiqui [siddas...@gmail.com]
Skickat: den 21 maj 2010 22:58
Till: David Holman
Kopia: ccie_voice@onlinestudylist.com
Ämne: Re: [OSL | CCIE_Voice] translation rule

Thanks David I have already gone through that document many times :)

Thanks Wael for your explanation. I was actually thinking that what would a 
null number be in my example. I have this customer router which has this 
specific rule for inbound calls. Will it work when PSTN will send "null 
digits"? doesn't make sense to me. What is a null/unknown digit?

Ash>

David Holman wrote:
I keep this link handy for voice translation questions:

http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080325e8e.shtml

On Fri, May 21, 2010 at 4:15 PM, Wael Agina 
mailto:waelag...@gmail.com>> wrote:
Dear Ashar,

  The ^$ is catching null, which could be used to catch calls from unkown.
example usage, drop any calls from PSTN that has ANI of unkown type.
On H323 you could use following rule to do this

voice translation-rule 1
 rule 1 reject /^$/

voice translation-profile Drop-Unknown
  translate calling 1

dial-peer voice 1 pots
direct-inward-dial
incom called .
call-block translation-profile incoming Drop-Unknown

For you example may be it i setting unknown ANI to be 42000 for example, bu not 
sure, need to be tested.

Regards,
Wael Agina

On Fri, May 21, 2010 at 11:02 PM, Ashar Siddiqui 
mailto:siddas...@gmail.com>> wrote:
Hi,

I know I may sound stupid to some but I really want to know the purpose of ^$ 
in a translation rule for e.g:

voice translation-rule 100
 rule 1 /^$/ /42000/
!


^$ is null...what does it mean? what is a null number?

Ash>

___
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--

Thanks and Best Regards,
Wael Agina

___
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___
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