[OSL | CCIE_Voice] CUE with CCM bug?

[CCIE for Me]

Hi all,

just want to know if I should be approaching this procedure a little different. 
 When I have been integrating CUE with CCM I have been creating the cti ports, 
cti route point, and user ccm BEFORE initializing the CUE.  Is this right 
approach?  I have noticed more than a few times, that even though the system 
says the initialization was good my ports and route point are still not 
registered in CCM.  Only after I delete the ports and route points and recreate 
them, and then do a reload of CUE will they register.

Has anyone else come across this issue?   Is there something I am probably 
doing wrong?

thanks for your help.


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

Re: [OSL | CCIE_Voice] Calling and Called Party Number Type

[Matthew Berry]

For sure, Adam!  "debug isdn q931" was my best friend in the lab.  A 
close runner-up is "debug voip dialpeer" to make sure the correct 
dial-peer is being selected.


Thanks!

Matthew Berry, CCIE #26721 (Voice)

Email: thematthewbe...@gmail.com
Twitter: http://twitter.com/CiscoVoiceGuru
Tech Blog: http://ciscovoiceguru.com


On 3/3/11 6:23 PM, adam compton wrote:
If you run debug isdn q931 on the gateway the call is going out,  you 
can confirm what Call Manager is sending to the PSTN.


On Thu, Mar 3, 2011 at 1:52 PM, Ccie Voice > wrote:


Thank you all for your reply,

I just need to know if the PSTN router in the LAB will accept the
call or no if it is not set to the proper value.

If the PSTN router will not accept the call then it is OK I can
play with these values and solve the problem.

But the problem if the PSTN router accepts all calls based on
called party number and later on the proctor will check if you set
the values correctly or not.

for me what I understood before is the way that Roger sent. (thank
you Roger)

Regards,

*From:* Roger Källberg mailto:roger.kallb...@cygate.se>>
*To:* Ccie Voice mailto:v.c...@yahoo.com>>;
CCIE Study mailto:ccie_voice@onlinestudylist.com>>
*Sent:* Thu, March 3, 2011 6:41:12 PM
*Subject:* SV: [OSL | CCIE_Voice] Calling and Called Party Number Type

Hi,

You need to look at this from the originating endpoint and the
outgoing gateway. For a more detailed explanation see my response
in line with your mail.

Sincerely

*Roger Källberg*
CCIE #26199 (Voice)
Consultant
CygateAB
Eric Perssons väg 21, SE-217 62 MALMÖ

*Från:* Ccie Voice [v.c...@yahoo.com ]
*Skickat:* den 3 mars 2011 02:49
*Till:* CCIE Study
*Ämne:* [OSL | CCIE_Voice] Calling and Called Party Number Type

Hi All,

I am a little bit confused about how to set the value for Calling
and Called Party Number Plan.

let us say HQ Phone 1 Calls local Call in this case I think I have
to set:
Calling Party Number Type to: Subscriber.
Called Party Number Type to: Subscriber.

*/This is correct/
*
What about Long Distance:
Calling Party Number Type to: Subscriberor National

/From the perspective of caller and VGW this is a call that came
from a local site , aka it's subscriber/
Called Party Number Type to: National
/From the perspective of called and VGW this is a call goes
to a remote phone, aka it's national/

//


it will be more complicated if we need to use TEHO, So if HQ Phone
1 calls BR1 Local PSTN number what I should set the values?

Long Distance, using BR1 Router

Calling Party Number Type to: Subscriberor National

/From the perspective of caller and VGW this is a call that came
from a remote site , aka it's national/
Called Party Number Type to: National or Subscriber

/From the perspective of called and VGW this is a call goes
to a local phone, aka it's subscriber/

Long Distance, backup for BR1 using HQ Router

Calling Party Number Type to: Subscriberor National

/From the perspective of caller and VGW this is a call that came
from a local site , aka it's subscriber/
Called Party Number Type to: National or Subscriber I am using BR1
Router

/From the perspective of called and VGW this is a call goes
to a remote phone, aka it's national/

Regards,




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



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

Re: [OSL | CCIE_Voice] Calling and Called Party Number Type

[adam compton]

If you run debug isdn q931 on the gateway the call is going out,  you can
confirm what Call Manager is sending to the PSTN.

On Thu, Mar 3, 2011 at 1:52 PM, Ccie Voice  wrote:

>  Thank you all for your reply,
>
> I just need to know if the PSTN router in the LAB will accept the call or
> no if it is not set to the proper value.
>
> If the PSTN router will not accept the call then it is OK I can play with
> these values and solve the problem.
>
> But the problem if the PSTN router accepts all calls based on called party
> number and later on the proctor will check if you set the values correctly
> or not.
>
> for me what I understood before is the way that Roger sent. (thank you
> Roger)
>
> Regards,
>  --
> *From:* Roger Källberg 
> *To:* Ccie Voice ; CCIE Study <
> ccie_voice@onlinestudylist.com>
> *Sent:* Thu, March 3, 2011 6:41:12 PM
> *Subject:* SV: [OSL | CCIE_Voice] Calling and Called Party Number Type
>
>  Hi,
>
> You need to look at this from the originating endpoint and the outgoing
> gateway. For a more detailed explanation see my response in line with your
> mail.
>
> Sincerely
>  *Roger Källberg*
> CCIE #26199 (Voice)
> Consultant
> Cygate AB
> Eric Perssons väg 21, SE-217 62 MALMÖ
>
>  --
> *Från:* Ccie Voice [v.c...@yahoo.com]
> *Skickat:* den 3 mars 2011 02:49
> *Till:* CCIE Study
> *Ämne:* [OSL | CCIE_Voice] Calling and Called Party Number Type
>
>  Hi All,
>
> I am a little bit confused about how to set the value for Calling and
> Called Party Number Plan.
>
> let us say HQ Phone 1 Calls local Call in this case I think I have to set:
> Calling Party Number Type to: Subscriber.
> Called Party Number Type to: Subscriber.
>
> *This is correct
> *
> What about Long Distance:
> Calling Party Number Type to: Subscriber or National
>
> *From the perspective of caller and VGW this is a call that came from a
> local site , aka it's subscriber*
> Called Party Number Type to: National
> *From the perspective of called and VGW this is a call goes to a remote
> phone, aka it's national*
> **
>
>
> it will be more complicated if we need to use TEHO, So if HQ Phone 1 calls
> BR1 Local PSTN number what I should set the values?
>
> Long Distance, using BR1 Router
>
> Calling Party Number Type to: Subscriber or National
>
> *From the perspective of caller and VGW this is a call that came from a
> remote site , aka it's national*
> Called Party Number Type to: National or Subscriber
>
> *From the perspective of called and VGW this is a call goes to a local
> phone, aka it's subscriber*
>
> Long Distance, backup for BR1 using HQ Router
>
> Calling Party Number Type to: Subscriber or National
>
> *From the perspective of caller and VGW this is a call that came
> from a local site , aka it's subscriber*
> Called Party Number Type to: National or Subscriber I am using BR1 Router
>
> *From the perspective of called and VGW this is a call goes to a remote
> phone, aka it's national*
>
> Regards,
>
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Calling and Called Party Number Type

[givemeccievoice2010]

We technically aren’t allowed to answer your question about the lab.  

 

Don’t stress out though, if the PSTN router won’t accept something or is
expecting something, it’s a safe bet that you will be told the information
you need.

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ccie Voice
Sent: Thursday, March 03, 2011 10:52 AM
To: CCIE Study
Subject: Re: [OSL | CCIE_Voice] Calling and Called Party Number Type

 

Thank you all for your reply,

I just need to know if the PSTN router in the LAB will accept the call or no
if it is not set to the proper value.

If the PSTN router will not accept the call then it is OK I can play with
these values and solve the problem.

But the problem if the PSTN router accepts all calls based on called party
number and later on the proctor will check if you set the values correctly
or not.


for me what I understood before is the way that Roger sent. (thank you
Roger) 

Regards,

  _  

From: Roger Källberg 
To: Ccie Voice ; CCIE Study

Sent: Thu, March 3, 2011 6:41:12 PM
Subject: SV: [OSL | CCIE_Voice] Calling and Called Party Number Type




Hi,

You need to look at this from the originating endpoint and the outgoing
gateway. For a more detailed explanation see my response in line with your
mail. 

Sincerely

Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB
Eric Perssons väg 21, SE-217 62 MALMÖ

 

  _  

Från: Ccie Voice [v.c...@yahoo.com]
Skickat: den 3 mars 2011 02:49
Till: CCIE Study
Ämne: [OSL | CCIE_Voice] Calling and Called Party Number Type

Hi All,

I am a little bit confused about how to set the value for Calling and Called
Party Number Plan.

let us say HQ Phone 1 Calls local Call in this case I think I have to set:
Calling Party Number Type to: Subscriber.
Called Party Number Type to: Subscriber.

This is correct

What about Long Distance:
Calling Party Number Type to: Subscriber or National

>From the perspective of caller and VGW this is a call that came from a local
site , aka it's subscriber
Called Party Number Type to: National
>From the perspective of called and VGW this is a call goes to a remote
phone, aka it's national


it will be more complicated if we need to use TEHO, So if HQ Phone 1 calls
BR1 Local PSTN number what I should set the values?

Long Distance, using BR1 Router

Calling Party Number Type to: Subscriber or National

>From the perspective of caller and VGW this is a call that came from a
remote site , aka it's national
Called Party Number Type to: National or Subscriber 

>From the perspective of called and VGW this is a call goes to a local phone,
aka it's subscriber

Long Distance, backup for BR1 using HQ Router

Calling Party Number Type to: Subscriber or National

>From the perspective of caller and VGW this is a call that came from a local
site , aka it's subscriber
Called Party Number Type to: National or Subscriber I am using BR1 Router 

>From the perspective of called and VGW this is a call goes to a remote
phone, aka it's national

Regards,

 

 

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

Re: [OSL | CCIE_Voice] Calling and Called Party Number Type

[ccieid1ot]

It will be noted in the lab book.  Take a deep breath.  You can breathe
now.  :)

On Thu, Mar 3, 2011 at 12:52 PM, Ccie Voice  wrote:

> Thank you all for your reply,
>
> I just need to know if the PSTN router in the LAB will accept the call or
> no if it is not set to the proper value.
>
> If the PSTN router will not accept the call then it is OK I can play with
> these values and solve the problem.
>
> But the problem if the PSTN router accepts all calls based on called party
> number and later on the proctor will check if you set the values correctly
> or not.
>
> for me what I understood before is the way that Roger sent. (thank you
> Roger)
>
> Regards,
> --
> *From:* Roger Källberg 
> *To:* Ccie Voice ; CCIE Study <
> ccie_voice@onlinestudylist.com>
> *Sent:* Thu, March 3, 2011 6:41:12 PM
> *Subject:* SV: [OSL | CCIE_Voice] Calling and Called Party Number Type
>
>  Hi,
>
> You need to look at this from the originating endpoint and the outgoing
> gateway. For a more detailed explanation see my response in line with your
> mail.
>
> Sincerely
>   *Roger Källberg*
> CCIE #26199 (Voice)
> Consultant
> Cygate AB
> Eric Perssons väg 21, SE-217 62 MALMÖ
>
>  --
> *Från:* Ccie Voice [v.c...@yahoo.com]
> *Skickat:* den 3 mars 2011 02:49
> *Till:* CCIE Study
> *Ämne:* [OSL | CCIE_Voice] Calling and Called Party Number Type
>
>   Hi All,
>
> I am a little bit confused about how to set the value for Calling and
> Called Party Number Plan.
>
> let us say HQ Phone 1 Calls local Call in this case I think I have to set:
> Calling Party Number Type to: Subscriber.
> Called Party Number Type to: Subscriber.
>
> *This is correct
> *
> What about Long Distance:
> Calling Party Number Type to: Subscriber or National
>
> *From the perspective of caller and VGW this is a call that came from a
> local site , aka it's subscriber*
> Called Party Number Type to: National
> *From the perspective of called and VGW this is a call goes to a remote
> phone, aka it's national*
> **
>
>
> it will be more complicated if we need to use TEHO, So if HQ Phone 1 calls
> BR1 Local PSTN number what I should set the values?
>
> Long Distance, using BR1 Router
>
> Calling Party Number Type to: Subscriber or National
>
> *From the perspective of caller and VGW this is a call that came from a
> remote site , aka it's national*
> Called Party Number Type to: National or Subscriber
>
> *From the perspective of called and VGW this is a call goes to a local
> phone, aka it's subscriber*
>
> Long Distance, backup for BR1 using HQ Router
>
> Calling Party Number Type to: Subscriber or National
>
> *From the perspective of caller and VGW this is a call that came
> from a local site , aka it's subscriber*
> Called Party Number Type to: National or Subscriber I am using BR1 Router
>
> *From the perspective of called and VGW this is a call goes to a remote
> phone, aka it's national*
>
> Regards,
>
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>


-- 
duy
CCIE #27737 Voice
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] codec setting for intra and interregion

[ccieid1ot]

Rahul,

Where would the voice class be applied to MOH?  Dial-peers has no relevance
to MOH.

On Wed, Mar 2, 2011 at 1:22 PM, Rahul Kapor  wrote:

> Hi Roger ,
>
>
>
> If Voice class is not configured then MOH will negotiate codec g729  and
> then it that case port should be incremented by 4 . in this case , MOH will
> be heard by PSTN phone , not by BR1 phone . becos still  Br1 phone will try
> listen the MOH on base port. so voice class is mandatory.
>
> what u think ?
>
> thx,
> Rahul
>
>
> On Wed, Mar 2, 2011 at 7:56 PM, Rogers Ochieng wrote:
>
>> If you are sourcing MOH from Flash for a CUCM Branch phone then the
>> multicast MOH server in CUCM will be in a region whose codec you have
>> specified to be used intra and intraregion , this will affect the port or IP
>> you use on the IOS router (increment on IP or port). So if you put it in a
>> G711 only region then you can use the IP you specify and port since G711 is
>> the first priority. I don't see where voice class codec comes in since
>> multicast moh can't be transcoded. Don't forget max-dn and max-ephones
>>
>>
>> On 2 March 2011 16:46, Rahul Kapor  wrote:
>>
>>> Hello Rogers / Wi ,
>>>
>>> thanks for  help.
>>> On top of this i have another question.
>>>
>>> Req is MOH should be played from flash
>>>
>>> gateway is H323
>>>
>>> My config is
>>>
>>> dial-peer voice 3000 voip
>>>  destination-pattern 3...
>>>  voice-class codec 2
>>>  session target ipv4:14.160.110.21
>>>  incoming called-number .
>>>
>>> call-manager-fallback
>>>  max-conferences 12 gain -6
>>>  transfer-system full-consult
>>>  ip source-address 14.160.116.40 port 2000
>>>  moh music-on-hold.au
>>>  multicast moh 239.1.1.1 port 16384 route 1.1.1.1
>>>
>>> voice class codec 2
>>>  codec preference 1 g729br8
>>>  codec preference 2 g729r8
>>>  codec preference 3 g711alaw
>>>  codec preference 4 g711ulaw
>>>
>>> since MOH from flash only supports g711ulaw , voice class is mandatory.
>>>
>>> above config works perfectly. Is this correct approach. I want to make
>>> sure that above  config should not break the codec requirement.
>>>
>>> in this case when br1 phone make a PSTN call , codec g711 will be
>>> negotiated as gateway and phone are in same region and voice class codec is
>>> configured with g711
>>>
>>> i would like to have your feedback.
>>>
>>> thx,
>>> Rahul
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Wed, Mar 2, 2011 at 9:11 AM, Rogers Ochieng 
>>> wrote:
>>>
 When there is a requirement for HQ Phones calls to BR1 area code numbers
 or redundancy through BR1 Gateway

   On 1 March 2011 22:25, Rahul Kapor  wrote:

>   Hi all ,
>
>
>
> For following  question
>
>
>
> For IP Phones and gateways , all calls within same site
>
> should use G711 codec. Also, all calls between the sites to remote IP
> phones and
>
> gateways should use G729 codec.
>
>
>
> My answer would be
>
>
>
> Hq region --à HR
>
>
>
> Br1 region --à BR
>
>
>
> Put the br1 gateway and Br1 phone in BR  through device pool
>
> HQ gateway and Hq phones in HR through device pool
>
>
>
> System parameter
>
>
>
> G722 Codec Enabled [image: Required Field]  disabled
>
>
>
> Intraregion Audio Codec Default  g711/g722
>
>
>
> Interregion Audio Codec Default   g729
>
>
>
>
>
> My doubt is , when IP phone is taking to IP phone (Hq<>Br), gateway is
> not coming in the picture. How can region setting here for gateway matters
> ??
>
>
>
> Please suggest.
>
>
>
> Thx,
>
> Rahul
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> !
>
>
>
>
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training,
> please visit www.ipexpert.com
>
>

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


-- 
duy
CCIE #27737 Voice
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Calling and Called Party Number Type

[Ccie Voice]

Thank you all for your reply,

I just need to know if the PSTN router in the LAB will accept the call or no if 
it is not set to the proper value.

If the PSTN router will not accept the call then it is OK I can play with these 
values and solve the problem.

But the problem if the PSTN router accepts all calls based on called party 
number and later on the proctor will check if you set the values correctly or 
not.


for me what I understood before is the way that Roger sent. (thank you Roger) 

Regards,



From: Roger Källberg 
To: Ccie Voice ; CCIE Study 
Sent: Thu, March 3, 2011 6:41:12 PM
Subject: SV: [OSL | CCIE_Voice] Calling and Called Party Number Type

   
Hi,
You  need to look at this from the originating endpoint and the outgoing 
gateway. For a more detailed explanation see my response in line with your mail.
Sincerely
Roger Källberg
CCIE #26199 (Voice)
Consultant
CygateAB
Eric Perssons väg 21, SE-217 62 MALMÖ

 


 Från: Ccie Voice [v.c...@yahoo.com]
Skickat: den 3 mars 2011 02:49
Till: CCIE Study
Ämne: [OSL | CCIE_Voice] Calling and Called Party Number Type


Hi All,

I am a little bit confused about how to set the value for Calling and Called 
Party Number Plan.

let us say HQ Phone 1 Calls local Call in this case I think I have to set:
Calling Party Number Type to: Subscriber.
Called Party Number Type to: Subscriber.
This is correct

What about Long Distance:
Calling Party Number Type to: Subscriberor National
From the perspective of caller and VGW this is a call that came from a 
local site , aka it's subscriber
Called Party Number Type to: National
From the perspective of called and VGW this is a call goes to a remote phone, 
aka it's national 


it will be more complicated if we need to use TEHO, So if HQ Phone 1 calls BR1 
Local PSTN number what I should set the values?

Long Distance, using BR1 Router
Calling Party Number Type to: Subscriberor National
From the perspective of caller and VGW this is a call that came from a remote 
site , aka it's national
Called Party Number Type to: National or Subscriber 
From the perspective  of called and VGW this is a call goes to a local phone, 
aka it's subscriber

Long Distance, backup for BR1 using HQ Router
Calling Party Number Type to: Subscriberor National
From the perspective of caller and VGW this is a call that came from a local 
site , aka it's subscriber
Called Party Number Type to: National or Subscriber I am using BR1 Router  
From the perspective of called and VGW this is a call goes to a remote phone, 
aka it's national

Regards,


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

Re: [OSL | CCIE_Voice] OT BAT Voicemail Profile

[Ki Wi]

Why not? BAT can handle up to 9 or 10 numbers. Just exclude those unnecessary 
number like voicemail pilot number, uccx... 

Just click on the add sign, select "or" instead of default "and"

Sent from my iPhone
Pls pardon my fat fingers.

On Mar 4, 2011, at 12:00 AM,  wrote:

> Hi All,
> 
>  
> 
> Looking forward for your valuable advice on how to change the VoiceMail 
> Profile for a bulk of users using BAT.
> 
>  
> 
> The standard process is to use Update Lines under the BAT Menu however I 
> doubt if it could be done for the DNs which are non-consecutive.
> 
>  
> 
> Thanks in advance
> 
>  
> 
> Kind Regards
> 
> Wilson Samuel
> 
>  
> 
> ___
> For more information regarding industry leading CCIE Lab training, please 
> visit www.ipexpert.com
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] OT BAT Voicemail Profile

[Wilson.Samuel]

Hi All,

Looking forward for your valuable advice on how to change the VoiceMail Profile 
for a bulk of users using BAT.

The standard process is to use Update Lines under the BAT Menu however I doubt 
if it could be done for the DNs which are non-consecutive.

Thanks in advance

Kind Regards
Wilson Samuel

___
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Re: [OSL | CCIE_Voice] WAN QoS

[Pablo Meneses]

ShinGei,

What DSCP value do you see in the layer 3 part of the RTP packets in the
packet capture?

- Pablo Meneses.

On Thu, Mar 3, 2011 at 6:00 AM, ShinGei Yong  wrote:

> Hi George,
> Thanks for the suggestion,but as Pablo mentioned it doesn't support the mls
> qos command.
>
> Hi Pablo,
> Thanks for you suggestion too.I did an sniffing as you advice, what i seen
> was the udp and rtp ports.
> I'm thinking that,could that because of old legacy of hwic (with old asic
> chip)that doesn't support something new like cos-dscp map table,
> and also the calculation of DSCP value different, so it can only be
> recognize as rtp audio?
>
> In fact i added in additional parameters which try to match the udp ports:
>
> access-list 101 permit udp any any range 16384 38767
> !
> class-map match-any Voice-SIG
>  match ip dscp cs3
>  match ip dscp af31
> class-map match-any Voice-RTP
>  match ip dscp ef
>  match protocol rtp audio
>  match access-group 101
>
> !
>
> Class-map: Voice-RTP (match-any)
>   122680 packets, 7851520 bytes
>
>   5 minute offered rate 25000 bps, drop rate 0 bps
>   Match: ip dscp ef (46)
> 0 packets, 0 bytes
> 5 minute rate 0 bps
>   Match: protocol rtp audio
> 122680 packets, 7851520 bytes
>
> 5 minute rate 25000 bps
>   Match: access-group 101
>
> 0 packets, 0 bytes
> 5 minute rate 0 bps
>
> I believe this should not be an abnormal case as PL Rack did used hwic as
> well.
> Maybe someone can provide the idea?
>
> Shingei
>
> On Thu, Mar 3, 2011 at 3:02 AM, Pablo Meneses wrote:
>
>> Shingei,
>>
>> Your configuration looks fine, however would would need to make sure that
>> the phone is actually marking the packets with the correct DSCP value, you
>> can run a packet capture from spanning the switch port of the 4ESW blade.
>>
>> This is done exactly the same way you do it in a 3750 by using the
>> "monitor session" command.
>>
>> BTW, those 4ESW do not support "mls qos" commands.
>>
>>  -Pablo Meneses.
>>
>>
>> On Wed, Mar 2, 2011 at 10:31 AM, George Goglidze wrote:
>>
>>> have you configured "mls qos trust dscp" on switchports???
>>> if this command is not present, the dscp gets rewritten to default 0.
>>>
>>> Regards,
>>>
>>>  On Wed, Mar 2, 2011 at 2:44 PM, ShinGei Yong wrote:
>>>
 Hi All,
 I've question regarding to the WAN QoS. I've HQ and BR2 sites setup,
 BR2 router equipped with HWIC-4ESW and 2 ipphone connected.

 1. I configured the class-based FRTS on the BR2-RTR as below. Initially
 the RTP traffic didn't match
 the class-map (DSCP) as defined, until i re-configure it with NBAR, then
 it just matched!

 !
 class-map match-any Voice-SIG
  match ip dscp cs3
  match ip dscp af31
 class-map match-any Voice-RTP
  match ip dscp ef
  match protocol rtp audio

 !
 policy-map WAN-EDGE
  class Voice-SIG
 bandwidth 18
  class Voice-RTP
 priority 24
compress header ip rtp
 policy-map MQC-FRTS-768
  class class-default
 shape average 729600 7296 0
   service-policy WAN-EDGE
 !
 !
 interface Serial0/1/0.102 point-to-point
  bandwidth 768
  ip address 10.10.112.2 255.255.255.0
  ip pim sparse-dense-mode
  snmp trap link-status
  frame-relay interface-dlci 102
   class FR-MAP-CLASS-768
 !
 !
 map-class frame-relay FR-MAP-CLASS-768
  frame-relay fragment 960
  frame-relay fair-queue
  service-policy output MQC-FRTS-768
 !
 !
  Class-map: Voice-RTP (match-any)
   104613 packets, 6695232 bytes
   5 minute offered rate 25000 bps, drop rate 0 bps
   Match: ip dscp ef (46)
 0 packets, 0 bytes
 5 minute rate 0 bps
   Match: protocol rtp
 104613 packets, 6695232 bytes
 5 minute rate 25000 bps
   Priority: 24 kbps, burst bytes 1500, b/w exceed drops: 0

 Why the RTP traffic doesn't match the RTP DSCP value?
 How do i configure in such a way the RTP traffic match DSCP EF instead
 of protocol RTP?

 Shingei

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


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

Re: [OSL | CCIE_Voice] Calling and Called Party Number Type

[adam compton]

The best answer is to set it to whatever the lab says to set it to.  In the
reality, Subscriber = local, National = LD, international international for
North American Dial Plan.  In the lab, different country may require all in
country calls to be subscriber, or all  in country calls to be national,
depending on the scenario.  I know of one IPexpert lab that requires that a
UK called number be set to subscriber, and a UK cell phone called number to
be national.

TEHO shouldn't cause to much conflict, because you would normally do this
with a Route Pattern where you can mark the settings for the TEHO number.
You can also set a closer match in called and calling transformations, as
well as translation-rules if you are restricted from using another Route
Pattern.

On Wed, Mar 2, 2011 at 8:49 PM, Ccie Voice  wrote:

>  Hi All,
>
> I am a little bit confused about how to set the value for Calling and
> Called Party Number Plan.
>
> let us say HQ Phone 1 Calls local Call in this case I think I have to set:
> Calling Party Number Type to: Subscriber.
> Called Party Number Type to: Subscriber.
>
> What about Long Distance:
> Calling Party Number Type to: Subscriber. or National
> Called Party Number Type to: National.
>
> it will be more complicated if we need to use TEHO, So if HQ Phone 1 calls
> BR1 Local PSTN number what I should set the values?
>
> Calling Party Number Type to: Subscriber. or National
> Called Party Number Type to: National or Subscriber I am using BR1 Router
>
> Regards,
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] Phones For Voice Lab Prep

[adam compton]

The only difference between the 7962 and 42 is the amount of buttons.  A
couple of the labs with IPexpert require a few more buttons, but it's
something you could test out with less buttons.  I think that would do just
fine.

On Wed, Mar 2, 2011 at 8:08 PM, Rafay Aslam  wrote:

> Hi
> I am thinking about buying 4 7942 and I already have 4 7960.
> What you think will it be enough for my ccie lab prep.
> I have a home lab.
>
>
> On Tuesday, March 1, 2011, adam compton  wrote:
> >
> >
> > On Tue, Mar 1, 2011 at 8:54 PM, adam compton  wrote:
> >
> > The 7961's will do and the 7962 would be the best.  Honestly, I've been
> able to get by pretty with with one 7962, 3 7942s, and and 2 7960s.  I'm
> lucky enough to have a bunch of phones just lying around though :).  If
> you're doing the remote labs through proctor labs, the 7960's are good for
> CME because the phone load is pretty small.  the limitations of the 7960
> aren't as apparent on the CME.
> >
> >
> > Regards,
> > Adam Compton
> >
> >
> >
> > On Tue, Mar 1, 2011 at 5:42 PM, Rafay Aslam  wrote:
> >
> >
> >
> > Hi
> > I have 4 x Cisco 7960 Phones, I am looking to upgrade in order to fully
> comply with Cisco Voice Lab, what do you recommend in terms of Cisco IP
> Phone, 7961, 7962??
> >
> > ___
> > For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com 
> >
> >
> >
> >
>
___
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Re: [OSL | CCIE_Voice] WAN QoS

[ShinGei Yong]

Hi George,
Thanks for the suggestion,but as Pablo mentioned it doesn't support the mls
qos command.

Hi Pablo,
Thanks for you suggestion too.I did an sniffing as you advice, what i seen
was the udp and rtp ports.
I'm thinking that,could that because of old legacy of hwic (with old asic
chip)that doesn't support something new like cos-dscp map table,
and also the calculation of DSCP value different, so it can only be
recognize as rtp audio?

In fact i added in additional parameters which try to match the udp ports:

access-list 101 permit udp any any range 16384 38767
!
class-map match-any Voice-SIG
 match ip dscp cs3
 match ip dscp af31
class-map match-any Voice-RTP
 match ip dscp ef
 match protocol rtp audio
 match access-group 101
!

Class-map: Voice-RTP (match-any)
  122680 packets, 7851520 bytes
  5 minute offered rate 25000 bps, drop rate 0 bps
  Match: ip dscp ef (46)
0 packets, 0 bytes
5 minute rate 0 bps
  Match: protocol rtp audio
122680 packets, 7851520 bytes
5 minute rate 25000 bps
  Match: access-group 101
0 packets, 0 bytes
5 minute rate 0 bps

I believe this should not be an abnormal case as PL Rack did used hwic as
well.
Maybe someone can provide the idea?

Shingei

On Thu, Mar 3, 2011 at 3:02 AM, Pablo Meneses  wrote:

> Shingei,
>
> Your configuration looks fine, however would would need to make sure that
> the phone is actually marking the packets with the correct DSCP value, you
> can run a packet capture from spanning the switch port of the 4ESW blade.
>
> This is done exactly the same way you do it in a 3750 by using the "monitor
> session" command.
>
> BTW, those 4ESW do not support "mls qos" commands.
>
>  -Pablo Meneses.
>
>
> On Wed, Mar 2, 2011 at 10:31 AM, George Goglidze wrote:
>
>> have you configured "mls qos trust dscp" on switchports???
>> if this command is not present, the dscp gets rewritten to default 0.
>>
>> Regards,
>>
>>  On Wed, Mar 2, 2011 at 2:44 PM, ShinGei Yong wrote:
>>
>>> Hi All,
>>> I've question regarding to the WAN QoS. I've HQ and BR2 sites setup,
>>> BR2 router equipped with HWIC-4ESW and 2 ipphone connected.
>>>
>>> 1. I configured the class-based FRTS on the BR2-RTR as below. Initially
>>> the RTP traffic didn't match
>>> the class-map (DSCP) as defined, until i re-configure it with NBAR, then
>>> it just matched!
>>>
>>> !
>>> class-map match-any Voice-SIG
>>>  match ip dscp cs3
>>>  match ip dscp af31
>>> class-map match-any Voice-RTP
>>>  match ip dscp ef
>>>  match protocol rtp audio
>>> !
>>> policy-map WAN-EDGE
>>>  class Voice-SIG
>>> bandwidth 18
>>>  class Voice-RTP
>>> priority 24
>>>compress header ip rtp
>>> policy-map MQC-FRTS-768
>>>  class class-default
>>> shape average 729600 7296 0
>>>   service-policy WAN-EDGE
>>> !
>>> !
>>> interface Serial0/1/0.102 point-to-point
>>>  bandwidth 768
>>>  ip address 10.10.112.2 255.255.255.0
>>>  ip pim sparse-dense-mode
>>>  snmp trap link-status
>>>  frame-relay interface-dlci 102
>>>   class FR-MAP-CLASS-768
>>> !
>>> !
>>> map-class frame-relay FR-MAP-CLASS-768
>>>  frame-relay fragment 960
>>>  frame-relay fair-queue
>>>  service-policy output MQC-FRTS-768
>>> !
>>> !
>>>  Class-map: Voice-RTP (match-any)
>>>   104613 packets, 6695232 bytes
>>>   5 minute offered rate 25000 bps, drop rate 0 bps
>>>   Match: ip dscp ef (46)
>>> 0 packets, 0 bytes
>>> 5 minute rate 0 bps
>>>   Match: protocol rtp
>>> 104613 packets, 6695232 bytes
>>> 5 minute rate 25000 bps
>>>   Priority: 24 kbps, burst bytes 1500, b/w exceed drops: 0
>>>
>>> Why the RTP traffic doesn't match the RTP DSCP value?
>>> How do i configure in such a way the RTP traffic match DSCP EF instead of
>>> protocol RTP?
>>>
>>> Shingei
>>>
>>> ___
>>> For more information regarding industry leading CCIE Lab training, please
>>> visit www.ipexpert.com
>>>
>>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Calling and Called Party Number Type

[Tamer Ismail]

Hello,

I can just answer only one question (Which I confirm).

it will be more complicated if we need to use TEHO, So if HQ Phone 1 calls
BR1 Local PSTN number what I should set the values?
Called Party Number Type to: Subscriber.

 

Best regards,

Tamer Ismail

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ccie Voice
Sent: Thursday, March 03, 2011 3:50 AM
To: CCIE Study
Subject: [OSL | CCIE_Voice] Calling and Called Party Number Type

 

Hi All,

I am a little bit confused about how to set the value for Calling and Called
Party Number Plan.

let us say HQ Phone 1 Calls local Call in this case I think I have to set:
Calling Party Number Type to: Subscriber.
Called Party Number Type to: Subscriber.

What about Long Distance:
Calling Party Number Type to: Subscriber. or National
Called Party Number Type to: National.

it will be more complicated if we need to use TEHO, So if HQ Phone 1 calls
BR1 Local PSTN number what I should set the values?

Calling Party Number Type to: Subscriber. or National
Called Party Number Type to: National or Subscriber I am using BR1 Router 

Regards,

 

___
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Re: [OSL | CCIE_Voice] number of sessions needed in ios media resources

[voice boy]

thanks for your reply
Only I need to know how to calculate the needed sessions and units for xcoder 
and conf-bridge
 
 
thanks
 


Date: Wed, 2 Mar 2011 19:54:54 -0500
Subject: Re: [OSL | CCIE_Voice] number of sessions needed in ios media resources
From: com...@gmail.com
To: voice...@hotmail.com
CC: ccie_voice@onlinestudylist.com


Conference requires dedicated DSPs.  If you still have MIPS left over on a DSP 
because you don't have many T1 channels, you can use them for Transcoding, but 
not for Conference.
 
Adam Compton


2011/3/2 voice boy 


Hi,
 
I'm confused about configuring these parameters when configuring X-coder & 
Conf-Bridge in ios
 
under dspfarm profile:
maximum sessions x [for conf and xcoder]
 
under telephony service:
sdspfarm units y [for conf and xcoder]
sdspfarm sessions z  [for xcoder]
 
 
So please advise how to calculate the needed sessions from the resources ?
Also If I configure conf-Bridge in a resource,, can I use this resources for 
another thing as transcoder or voice termination ?
 
 
thanks

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