Re: [OSL | CCIE_Voice] CME/CUE DTMF relay from UCM(sip trunk) to CME/CUE

[datucha123 datucha123]

There is a little restriction for SIP Notify DTMF for CUCM.

Juan is correct  - You need to enable the "accept unsollicited notify" in
sip security profile so that CUCM will be able to receive Notify DMTFs.
But if MTP is checked, then the Notify option will not work.
Also for outbound Notify DTMF from CUCM, it is not necessary to enable
"accept unsollicited notify". It will still work.

As for KPML and Notify internetworking  -  it is supported of CUCME (CUBE).

If you are using Inband (RFC2833) on one side and any out of band DTMF on
other side, then you have to configure the following command on RFC2833
side dial-peer, so that the Router will strip out the inband DTMF's and
leave only out of band for outgoing dial-peer:

*dtmf-relay rtp-nte digit-drop*

On Sun, Mar 4, 2012 at 10:23 AM, Juan Lopez
wrote:

> for option 1 below - you could try to set under the sip security profile
> "accept unsollicited notify" so that on the BR2 side, you use sip-notify as
> DTMF relay on both CUE and UCM SIP dialpeers.
> Let us know if thay might help
>
> cheers,
> Juan
>
> 2012/3/4 Juan Lopez 
>
>> Hi Justin,
>>
>> from reading the mail it looks like on the SIP dialpeers on the BR2, you
>> use the rtp-nte (inband) dtmf-relay method?
>>
>> can you try and let us know:
>> 1) use SIP-NOTIFY on both SIP dialpeers at BR2? (not sure if UCM supports
>> this - in SRND it states a UCM SIP trunk uses RTP-NTE or possibly SIP-KPML)
>> if 1 does not work:
>> 2) use sip notify on CUE dialpeer and sip-kpml on sip dialpeer to UCM.
>> Not sure here whether the CUBE at branch 2 supports notify - kpml dtmf
>> interworking...
>>
>> The idea is to have DTMF between UCM and CME out of band...
>>
>> From SRND I read that SIP at UCM uses RTP-NTE or possibly SIP-KPML, so it
>> rules out to use the SIP-NOTIFY on the dialpeer at branch2 pointing to UCM
>> (not tested yet) to keep it all out of band - but this is the way to rule
>> out an MTP
>>
>>
>>
>> 2012/3/3 Justin McIntyre 
>>
>>> Ok.  For those who are interested I have resolved my issue.  By
>>> selecting the Media Termination Point Required option within the SIP trunk
>>> I was able to resolve my media stream to an MTP  prior to connection to the
>>> CME.  This allowed in-band/Out of Band DTMF traversal.  Note that when you
>>> select the MTP required option within your sip trunk to pay special
>>> attention to the device pool and region settings upon with the MTP that you
>>> will resolve to will lie.  The MTP will not inherit the Device Pool
>>> settings from the Sip trunk depending on your configuration.  This was a
>>> really good learning experience and if anyone is curious as to any further
>>> details please let me know.
>>>
>>>
>>> I am however un-clear on one thing and maybe someone can help me out.  I
>>> remember using Sip-Notify within my CUE dial-peer and within CUE
>>> configuration the last time I ran this lab.  For some reason I could not
>>> get SIP-Notify to work in any case at all that I tried this time around.
>>>  If anyone has any clarity on this I would be most appreciative, I'd hate
>>> to see, "please configure a sip trunk between UCM and CME location at to
>>> reach the CUE VM pilot.  Note:  use of an MTP on the SIP trunk is not
>>> allowed" in the lab.  Plus who knows when a customer site may encounter
>>> this situation.  Thanks everyone.
>>>
>>>
>>> *!*!*!*Thanks to Chase and Vik as they were pertinent in my
>>> resolution.*!*!*!*
>>>
>>> Thanks,
>>>
>>> Justin McIntyre
>>>
>>>
>>> This email and any files transmitted with it are confidential and are
>>> intended for the sole use of the individual to whom they are addressed.
>>> Black Box Corporation reserves the right to scan all e-mail traffic for
>>> restricted content and to monitor all e-mail in general. If you are not the
>>> intended recipient or you have received this email in error, any use,
>>> dissemination or forwarding of this email is strictly prohibited. If you
>>> have received this email in error, please notify the sender by replying to
>>> this email.
>>> ___
>>> For more information regarding industry leading CCIE Lab training,
>>> please visit www.ipexpert.com
>>>
>>> Are you a CCNP or CCIE and looking for a job? Check out
>>> www.PlatinumPlacement.com 
>>>
>>
>>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] Voiceview express

[datucha123 datucha123]

Remove the Authentication Credentials. They are not required for VoiceView.

On Fri, Mar 2, 2012 at 5:52 PM, Baktha Muralidharan wrote:

> Hi folks
>
> I configured voiceview express on CME (Branch2 router). here is my config
> on CME-
>
>url services http://http://%3ccue/> IP>/voiceview/common/login.do
>url authentication http://http://%3ccme/>IP>/CCMCIP/authenticate.asp
>authentication credential username pwd (this is same as what's
> configured under CUE GUI-->CallManager page)
>
> On CUE, voiceview is configured/enabled.
>
> On the phone, if I hit the Services button, I see "CME Services URL". If I
> choose it, it is stuck in "Requesting..." state
>
> On CME, debug ip http all shows the  folowing-
>
>  HTTP GET comes in..
>
>   then the following-
>
> * service_url_main_page: CP send failed error=4748*
>
> Any hints on what might be wrong?
>
>I have tried reloading the router, to no avail.
>
> thanks in advance,
> /Baktha
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
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Re: [OSL | CCIE_Voice] saving script in script respoitory

[datucha123 datucha123]

Well, as I remember, the Scipts must not be saved in Repository from Script
Editor.

When you make upload of scripts from UCCX Web interface, it makes the
script saving in repository.

On Thu, Mar 1, 2012 at 4:57 PM, Baktha Muralidharan wrote:

> Hi folks
>
> When I try to save (from CCX script editor) a script into script
> repository, I get an error as follows-
>
> java.lang.NullPointerException Unsupported domain
>
> any hints on what might be wrong?
>
> thanks in advance,
> /Baktha
>
> ___
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> visit www.ipexpert.com
>
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Re: [OSL | CCIE_Voice] Tragic news

[datucha123 datucha123]

My condolences to his friends and family

On Sat, Feb 25, 2012 at 7:59 AM, Emanuel Damasceno wrote:

> Hello brothers,
>
> I just like to let you know that my friend and study partner, Jeferson
> Guardia CCIE #28157, has passed away yesterday.
>
> He was a skateboarder and day before yesterday, he fell, hitting his head.
> He didn't want to go to the hospital, because he thought he was ok.
> According to his father he passed in his sleep due to a blood clot in his
> brain. This is a tragic moment for all his family and friends.
>
> I thought I should share this with you guys because he's been very active
> here on the list, and we were studying together for the CCIE Voice. He was
> a great motivator and helped me get out of my personal problems so I could
> focus on my studies. It's sad how life is, and what shocks everybody the
> most is that he was only 24 years old (soon to be 25 on March 20th).
>
> Mourning, but still on the fight... =(
>
> *Emanuel Damasceno*
> CCNP Voice
>
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] Calling Number of SIP Trunks

[datucha123 datucha123]

Yes, you are sending full E164 number, but pay attention that SIP does not
support Type and Plan for ANI/DNIS numbers, that is part of H323/MGCP and
ISDN.

On Sat, Feb 25, 2012 at 8:05 PM, Ken Wyan  wrote:

> Hi,
>
> If we send a call to PSTN through ISDN  debug isdn q931  shows  the
> number , type , plan  of calling & called numbers.
>
> If we use a SIP ITSP ( using a sip trunk from CUCM) , what's the easiest
> method ?
>
> If SIP Invite message has following text , is it 100% sure that I am
> sending full e.164 number to SIP ITSP?
>
> From: "HQPH2" 
>
> Thanks
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
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> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] FXS incoming "flapping"

[datucha123 datucha123]

FXS is used to connect Analog Phones, not the PSTN.

For PSTN connectivity you need FXO Card.

On Sat, Feb 25, 2012 at 5:52 PM, Steven <
forum.ccie.onlinestudyl...@nocer.net> wrote:

> Hi group,
> i'm playing with FXS ports right now and encountered a strange behaviour.
>
> First the call-flow to get the picture:
> PSTN ---> FXS ---> CME
>
> The callED phone is ringing, but it only rings when the callING phone
> hears a tone.
> When the tone stops, the callED phone stops riniging. And the process
> starts over.
>
> For the callING phone the ringing tone is just fine.
> My guess is i have done something wrong with FXS signaling and/or cabling.
>
> I hope somebody already encountered something similar.
>
> PSTN network is Germany.
> Some config:
> !
> voice-port 0/1/0
>  cptone DE
> !
> dial-peer voice 50 pots
>  translation-profile incoming 54856975
>  incoming called-number .
>  direct-inward-dial
>  port 0/1/0
> !
> voice translation-profile 1000
>  translate called 50
> !
> voice translation-rule 50
>  rule 1 // /1000/
> !
> voice register pool  1
>  id mac E840..
>  type 9971
>  number 1 dn 1
> !
> voice register dn  1
>  number 1000
>  name Steven
>
>
> Thanks in advance,
> Steven
> __**_
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] When does IPMA route point show as registered?

[datucha123 datucha123]

It should appear as registered, as it depends on the IPMA and CallManager
service, where the IPMA service (Route Point) registeres with CUCM Server
on either Publisher or Subscriber (based on CM Groups).
CTI Route Point registration of IPMA does not depend on the End user
configuration for Manager or Assistant.

Hope that helps

On Thu, Feb 23, 2012 at 10:47 AM, Anthony Alba wrote:

> I would like to build-up a step-by-step  IPMA Proxy mode checklist and
> verification.
>
>
> If you configure the IPMA route point (with DN a superset of Managers' DNs
> like 5XXX), configure the IPMA Service Parameters
> on both Pub/Sub and restart he IPMA service, ought the IPMA route point
> appear as registered?
> (at this stage I have no Managers or Assistants configured)
>
> In my brief testing this usually doesn't happen (I suspect my VMs), but I
> would like to confirm whether the IPMA route point should appear registered
> or unregistered without any user configuration (yet).
>
> ___
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> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
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Re: [OSL | CCIE_Voice] Embedded BACD Prompts and DropThrough

[datucha123 datucha123]

Well, based on my knowledge, it is not possible to configure multiple AA's
using embedded TCL.
If you want multiple AAs, then you need to use the TCL files uploaded to
flash or TFTP


2012/2/21 "Farkas Péter" 

>
> Also can we configure multiple AAs using the embeded script itself?
> It seems to me not since usage of 'service app-b-acd-aa' command does not
> allow us to determine a different service name like flash based BACD
> 'service aa flash://', where 'aa' is name of the service.
>
> Peter
>
> - Original Message -
> From: datucha123 datucha123 
> Date: Sunday, February 19, 2012 11:53 am
> Subject: Re: [OSL | CCIE_Voice] Embedded BACD Prompts and DropThrough
> To: AJ BG 
> Cc: ccie_voice@onlinestudylist.com
>
>
> > 1.   Are prompts also embedded in the IOS? Or do they need to be
> copied
> >  in the router’s flash?
> >
> >  No, the Prompts are not embedded in the IOS, you need to manually add
> them
> >  into Flash.
> >
> >
> >  2.   Does drop through mode work with embedded BACD?
> >  Yes, embedded BACD works for Drop Through Mode very well.
> >
> >  You can find the configuration examples here:
> >
> >
> >
> >
> >
> >
>  >  On Sun, Feb 19, 2012 at 7:59 AM, AJ BG  wrote:
> >
> >  > Two questions about embedded BACD.
> >  >
> >  > 1.   Are prompts also embedded in the IOS? Or do they need to be
> >  > copied in the router’s flash?
> >  >
> >  > 2.   Does drop through mode work with embedded BACD?
> >  >  Does anyone have a working copy of embedded BACD configuration?
> >  > Thanks
> >  > AJ
> >  >
> >  > ___
> >  > For more information regarding industry leading CCIE Lab training,
> please
> >  > visit www.ipexpert.com
> >  >
> >  > Are you a CCNP or CCIE and looking for a job? Check out
> >  > www.PlatinumPlacement.com <http://www.platinumplacement.com/> <>
> >  >
> > ___
> >  For more information regarding industry leading CCIE Lab training,
> please visit www.ipexpert.com
> >
> >  Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com <http://www.platinumplacement.com/>
>
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Re: [OSL | CCIE_Voice] Vol 1 Lab 5A 5.2 Calling Party Transformation

[datucha123 datucha123]

Maybe you are using H323 gateways, and + sign is added  on the gateways?

On Tue, Feb 21, 2012 at 12:24 PM, Guoming Zhang wrote:

>  Hi,
>
> When I am doing Vol 1 Lab 5A 5.2 question, the calling number from both HQ
> and BR1 to PSTN should be in full E.164 format such as +1212. and
> +1617  on PSTN phone. The solution uses Calling party Transformation,
> but I would not see +1 on the "Prefix Digit" under Calling Party
> Transformation Parttern Configuration  page for both BR1 and HQ. I need to
> add +1  for both Calling Party Transformation to get +1 on PSTN phone, but
> there is no +1 on DSG. Am I missing something here?
>
> thanks,
>
> guoming
>
> ___
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> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
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Re: [OSL | CCIE_Voice] CUBE registration rejected

[datucha123 datucha123]

Send us the GK configuration. along with CUBE as well.

On Tue, Feb 21, 2012 at 12:30 AM, Chevy  wrote:

> Anyone ever seen this message before?
>
> Gateway CUBE failed to register with Gatekeeper VIA-ZONE even after 2
> retries
>
>
> ___
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> visit www.ipexpert.com
>
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> www.PlatinumPlacement.com 
>
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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 72, Issue 115

[datucha123 datucha123]

Threshold 3 is kind of the "percentage of buffer contributed by this queue
to the overall buffer space" and threshold 4 specifies the 3rd threshold
for the queue in question.

Reserved Threshold is considered for Egress Queues as Threshold 3 (T3). Not
the M (maximum) threshold.

On Mon, Feb 20, 2012 at 2:14 AM, Randall Crumm  wrote:

>  Thank you sir. I appreciate it
>
> Cheers,
> Randall
>
>   --
> *From:* Baktha Muralidharan 
> *To:* ccie_voice@onlinestudylist.com
> *Sent:* Sunday, February 19, 2012 1:13 PM
> *Subject:* Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 72, Issue 115
>
>  Hi Randall
>
>
> 1. I think the solutions guide recommends that you run auto qos on an *
> unused* interface, just to be safe.
> 2.  As you know, the 75% refers to the value of the [drop] threshold.
> Since cos 5 is associated with 3rd threshold, as such, you would configure
> threshold 3 to be 75%. However, the recommended approach is to move the cos
> 5 to threshold 1 and then configure threshold 1 to be 75%. The theory
> behind this is somewhat complex (Vik explains this in detail in the
> bootcamp). It goes something like this-
>
> Threshold 3 is kind of the "percentage of buffer contributed by this queue
> to the overall buffer space" and threshold 4 specifies the 3rd threshold
> for the queue in question.
>
> thanks,
> /Baktha
>
>
>
> --
>
> Message: 2
> Date: Sun, 19 Feb 2012 11:15:30 -0800 (PST)
> From: Randall Crumm 
> To: Online Study 
> Subject: [OSL | CCIE_Voice] New lab #2 - 3750 Qos
> Message-ID:
><1329678930.97124.yahoomail...@web164603.mail.gq1.yahoo.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> HI,
> I have a question on Qos
> 1. The question wants COS 3 i n the 2nd egress queue, 3rd threshold and
> COS 5 in the 1st egress queue?3rd threshold.
> This is done after running auto qos on the phone switchport. No question
>
>
> 2. COS 5 ?traffic sent to SA-GW(queue set 2) should be dropped if queue is
> 75% full.?
> Can someone please explain the answer to me. The DSG is not clear to me.
>
> Thanks,
>
>
> ?
> Cheers,
> Randall
> -- next part --
> An HTML attachment was scrubbed...
> URL:
> 
>
> --
>
>
>
> Hi Emanuel,
>
>   - does the call drop even if you select an internal (on-net) extension?
>   - assume you have configured the CSS on the RDP?
>
> thanks,
> /Baktha
>
>
>
>
> Message: 3
> Date: Sun, 19 Feb 2012 17:17:17 -0200
> From: Emanuel Damasceno 
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] MVA - Phone hangs up when trying to make a
>call.
> Message-ID:
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hey guys,
>
> I've configured everything (I suppose...) RDP, RD, MVA on SP... Everything
> seems to be in order. It welcomes me, asks to dial a password, yadda yadda,
> when I ask to make a call by pressing 1, I dial the numbers and when I
> press # the connection drops.
>
> Here is my config:
> HQ-RTR
> application
>  service mva http://10.10.210.10:8080/ccmivr/pages/IVRMainpage.vxml
> !
> dial-peer voice 5999 pots
>  service mva
>  incoming called-number 5999
> !
> dial-peer voice 5000 voip
>  destination-pattern 5...
>  voice-class codec 1
>  voice-class h323 1
>  session target ipv4:10.10.210.11
>  no vad
> dial-peer voice 5001 voip
>  preference 1
>  destination-pattern 5...
>  voice-class codec 1
>  voice-class h323 1
>  session target ipv4:10.10.210.10
>  no vad
>
> The remote Profile and Remote Destination Profile are correct, otherwise it
> wouldn't ask me to put the PIN straight up. It would ask me first for my
> number then PIN.
> Any ideas?
>
> *Emanuel Damasceno*
> CCNP Voice
> -- next part --
> An HTML attachment was scrubbed...
> URL:
> 
>
> --
>
> Message: 4
> Date: Sun, 19 Feb 2012 11:16:44 -0800 (PST)
> From: Randall Crumm 
> To: romain mullier 
> Cc: Online Study 
> Subject: Re: [OSL | CCIE_Voice] BACD issue new labs #2
> Message-ID:
><1329679004.50928.yahoomail...@web164601.mail.gq1.yahoo.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> lol
>
> You mean load up some more. I have to wake up at 5AM since I am on the
> west coast.
> ?
> Cheers,
> Randall
>
>
>
> 
>  From: romain mullier 
> To: Randall Crumm 
> Cc: Online Study 
> Sent: Sunday, February 19, 2012 11:09 AM
> Subject: Re: [OSL | CCIE_Voice] BACD issue new labs #2
>
>
> Randall,
>
> You defined the name of your sevice as "app-b-acd-aa" yet your dial-peer
> is invoking "service aa" which does not exist.
> Time to take a coffee break ;-)
>
> Romain
>
>
>
> On Sun, Feb 19, 2012 at 12:59 PM, Randall Crumm  wrote:
>
> Hi I am trying to call x4000 and have BACD forward to ephone-hunt list.
> >
> >
> >When i call 4001 it works. When I call 4000 I get a busy signal .
> >
> >
> >Any thoughts are appreciated.
> >
> >
> >application
> >?service app-b-acd-aa
> >? param voi

Re: [OSL | CCIE_Voice] New lab #2 - 3750 Qos

[datucha123 datucha123]

It is also possible to set the T3 (Reserved) threshold to something else
then 100%.

I do not know whether this goes for this particular case, but in general,
Reserved Threshold can be set to other then 100%.

T3 is non-configurable for Ingress SRR Queues, where is it always set to
100%.

But for Egress Queues, you can change the T3 to something other then 100%.

On Mon, Feb 20, 2012 at 1:55 AM, Randall Crumm  wrote:

>  Thanks George,
> I was tired and put 90 90. My mistake. The DSG does have 75 100 100 100
>
> Thanks for your answer sir.It is helpful.
>
> This statement is inccorect n the DSG:
>
>  mls qos srr-queue output cos-map queue 1 threshold 3. It should be T1
>  That's why it was somewhat confusing.
>
>
> Cheers,
> Randall
>
>--
> *From:* George Goglidze 
> *To:* Randall Crumm 
> *Cc:* Online Study 
> *Sent:* Sunday, February 19, 2012 1:39 PM
> *Subject:* Re: [OSL | CCIE_Voice] New lab #2 - 3750 Qos
>
>  I don't know where you got the table that you are referencing below...
> But you can never set threshold 3, it's 100 percent always:
> mls qos queue-set output X threshold X T1 T2 R M
>
> T1 and T2 are self explanatory.
> Then it's R for Reserved and M for Maximum.
>
> So to answer your question, yes you still need to move the needed CoS to
> T1 or T2, and assign 75 percent to it.
>
> Hope this helps,
>
> Sent from my iPad
>
> On 19 Feb 2012, at 20:47, Randall Crumm  wrote:
>
>   Looking at this closer.
>
>  OB queue Queue 2 T1 T2 T3  MAX
> mls qos queue-set output 2 threshold 75 90 90  100
> mls qos srr-queue output cos-map queue 1 threshold 3 (I thing the last
> part of this may be wrong, should be a 1)
>
> So looking at the DSG, (see above) what is confusing, is if COS 5 is
> already in T3, from the auto qos, why move it to T1. I would say keeping it
> in T3 is easier (see below).
>
>   OB queue Queue 2 T1 T2 T3  MAX
> mls qos queue-set output  2  threshold  50 60 75  100
>
> Any thoughts experts?
>
>
> Cheers,
> Randall
>
>   --
> *From:* Randall Crumm 
> *To:* Online Study 
> *Sent:* Sunday, February 19, 2012 11:15 AM
> *Subject:* New lab #2 - 3750 Qos
>
>   HI,
> I have a question on Qos
> 1. The question wants COS 3 i n the 2nd egress queue, 3rd threshold and
> COS 5 in the 1st egress queue 3rd threshold.
> This is done after running auto qos on the phone switchport. No question
>
> 2. COS 5  traffic sent to SA-GW(queue set 2) should be dropped if queue is
> 75% full.
> Can someone please explain the answer to me. The DSG is not clear to me.
>
> Thanks,
>
>
>
> Cheers,
> Randall
>
>
>  ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
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>
>
>
>
> ___
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Re: [OSL | CCIE_Voice] MVA - Phone hangs up when trying to make a call.

[datucha123 datucha123]

That is because the MVA number does not match 5999 configured in CUCM.
please make sure that 5999 is also configured as MVA access number in CUCM,
and no Called Party transformation takes place from Incoming POTS dial-peer
up to CUCM.

On Sun, Feb 19, 2012 at 11:17 PM, Emanuel Damasceno
wrote:

> Hey guys,
>
> I've configured everything (I suppose...) RDP, RD, MVA on SP... Everything
> seems to be in order. It welcomes me, asks to dial a password, yadda yadda,
> when I ask to make a call by pressing 1, I dial the numbers and when I
> press # the connection drops.
>
> Here is my config:
> HQ-RTR
> application
>  service mva http://10.10.210.10:8080/ccmivr/pages/IVRMainpage.vxml
> !
> dial-peer voice 5999 pots
>  service mva
>  incoming called-number 5999
> !
> dial-peer voice 5000 voip
>  destination-pattern 5...
>  voice-class codec 1
>  voice-class h323 1
>  session target ipv4:10.10.210.11
>  no vad
> dial-peer voice 5001 voip
>  preference 1
>  destination-pattern 5...
>  voice-class codec 1
>  voice-class h323 1
>  session target ipv4:10.10.210.10
>  no vad
>
> The remote Profile and Remote Destination Profile are correct, otherwise
> it wouldn't ask me to put the PIN straight up. It would ask me first for my
> number then PIN.
> Any ideas?
>
> *Emanuel Damasceno*
> CCNP Voice
>
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
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Re: [OSL | CCIE_Voice] SRST MoH issue

[datucha123 datucha123]

It works just the same way as Normal CUCME  -  Internal SCCP Phones can get
MoH only though Multicast.

On Sun, Feb 19, 2012 at 10:40 PM, datucha123 datucha123 <
datucha...@gmail.com> wrote:

> I have tested, and Multicast MoH is supported in SRST Mode for SCCP IP
> Phones, I can hear the Music.
>
>
> On Tue, Dec 13, 2011 at 2:54 PM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>> I have not fixed it, I have still the ToH for SRST SCCP IP Phones.
>> It is not possible to get the MoH for SCCP SRST IP Phones
>>
>>   On Tue, Dec 13, 2011 at 2:44 PM, Mohammed Al Baqari <
>> baqari.voic...@gmail.com> wrote:
>>
>>>  Good …  but how you fixed.? Is it matching below statement.
>>>
>>> ** **
>>>
>>> Regards,
>>>
>>> Mohammed Al Baqari
>>>
>>> ** **
>>>
>>> *From:* datucha123 datucha123 [mailto:datucha...@gmail.com]
>>> *Sent:* Tuesday, December 13, 2011 11:58 AM
>>> *To:* Mohammed Al Baqari
>>> *Cc:* ccie_voice@onlinestudylist.com
>>> *Subject:* Re: [OSL | CCIE_Voice] SRST MoH issue
>>>
>>> ** **
>>>
>>> Thank you very much for your answer. 
>>>
>>>  
>>>
>>> I have solved it already :)
>>>
>>> On Tue, Dec 13, 2011 at 1:42 AM, Mohammed Al Baqari <
>>> baqari.voic...@gmail.com> wrote:
>>>
>>> Hi,
>>>
>>>  
>>>
>>> I am not sure whether you have fixed this or not. But in between in case
>>> you are using fixed source feed and MMoH SRST, then the following
>>> restriction is valid.
>>>
>>>  
>>>
>>> MOH is supplied only to PSTN and VoIP G.711 calls. Local IP phone
>>> callers hear a repeating tone on hold for reassurance that they are still
>>> connected.
>>>
>>>  
>>>
>>> Regards,
>>>
>>> Mohammed Al Baqari
>>>
>>>  
>>>
>>> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
>>> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *datucha123
>>> datucha123
>>> *Sent:* Friday, December 02, 2011 10:47 PM
>>> *To:* ccie_voice@onlinestudylist.com
>>> *Subject:* [OSL | CCIE_Voice] SRST MoH issue
>>>
>>>  
>>>
>>> Hi Guys,
>>>
>>> In SRST mode Branch 1 , I cannot hear the MOH from Flash ,but I hear
>>> beep sound. There is no MOH playing for internal calls from BR1 phn 1 to
>>> BR1 Phn 2.
>>>
>>> But when I put the PSTN caller on hold the PSTN phn can hear the MOH. I
>>> tried the debug ephone MOHit says:
>>>
>>>  
>>>
>>> *No MOH entry for DN 1*
>>>
>>>  
>>>
>>> *Dec  2 19:27:37.678: ifs_read flash:MobileConnectOn.ulaw.wav end of
>>> file at 15078 read 6832 = 21910
>>> *Dec  2 19:27:37.682: moh tail fill from 46 at 0x66BBACF4 length 1168
>>> *Dec  2 19:27:38.254: ephone_hold_resume ignored for s2s set on dn=1
>>> chan=1 hold=1 callID=1547
>>> **Dec  2 19:27:39.922: No MOH entry for DN 1**
>>> **Dec  2 19:27:39.926: ephone_hold_resume ignored for s2s set on dn=1
>>> chan=1 hold=0 callID=1547
>>> *Dec  2 19:27:40.678: ifs_read flash:MobileConnectOn.ulaw.wav end of
>>> file at 17214 read 4696 = 21910
>>> *Dec  2 19:27:40.678: moh tail fill from 46 at 0x66BB855C length 3304
>>> *Dec  2 19:27:40.882: MoH route If Vlan11 ETHERNET 177.2.11.1 via ARP
>>> *Dec  2 19:27:40.882: MoH route If Loopback0 46 177.1.254.2 via
>>> 177.1.254.2
>>>
>>>  
>>>
>>>  
>>>
>>> ** **
>>>
>>
>>
>
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Re: [OSL | CCIE_Voice] SRST MoH issue

[datucha123 datucha123]

I have tested, and Multicast MoH is supported in SRST Mode for SCCP IP
Phones, I can hear the Music.

On Tue, Dec 13, 2011 at 2:54 PM, datucha123 datucha123  wrote:

> I have not fixed it, I have still the ToH for SRST SCCP IP Phones.
> It is not possible to get the MoH for SCCP SRST IP Phones
>
>   On Tue, Dec 13, 2011 at 2:44 PM, Mohammed Al Baqari <
> baqari.voic...@gmail.com> wrote:
>
>>  Good …  but how you fixed.? Is it matching below statement.
>>
>> ** **
>>
>> Regards,
>>
>> Mohammed Al Baqari
>>
>> ** **
>>
>> *From:* datucha123 datucha123 [mailto:datucha...@gmail.com]
>> *Sent:* Tuesday, December 13, 2011 11:58 AM
>> *To:* Mohammed Al Baqari
>> *Cc:* ccie_voice@onlinestudylist.com
>> *Subject:* Re: [OSL | CCIE_Voice] SRST MoH issue
>>
>> ** **
>>
>> Thank you very much for your answer. 
>>
>>  
>>
>> I have solved it already :)
>>
>> On Tue, Dec 13, 2011 at 1:42 AM, Mohammed Al Baqari <
>> baqari.voic...@gmail.com> wrote:
>>
>> Hi,
>>
>>  
>>
>> I am not sure whether you have fixed this or not. But in between in case
>> you are using fixed source feed and MMoH SRST, then the following
>> restriction is valid.
>>
>>  
>>
>> MOH is supplied only to PSTN and VoIP G.711 calls. Local IP phone callers
>> hear a repeating tone on hold for reassurance that they are still connected.
>> 
>>
>>  
>>
>> Regards,
>>
>> Mohammed Al Baqari
>>
>>  
>>
>> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
>> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *datucha123
>> datucha123
>> *Sent:* Friday, December 02, 2011 10:47 PM
>> *To:* ccie_voice@onlinestudylist.com
>> *Subject:* [OSL | CCIE_Voice] SRST MoH issue
>>
>>  
>>
>> Hi Guys,
>>
>> In SRST mode Branch 1 , I cannot hear the MOH from Flash ,but I hear beep
>> sound. There is no MOH playing for internal calls from BR1 phn 1 to BR1 Phn
>> 2.
>>
>> But when I put the PSTN caller on hold the PSTN phn can hear the MOH. I
>> tried the debug ephone MOHit says:
>>
>>  
>>
>> *No MOH entry for DN 1*
>>
>>  
>>
>> *Dec  2 19:27:37.678: ifs_read flash:MobileConnectOn.ulaw.wav end of file
>> at 15078 read 6832 = 21910
>> *Dec  2 19:27:37.682: moh tail fill from 46 at 0x66BBACF4 length 1168
>> *Dec  2 19:27:38.254: ephone_hold_resume ignored for s2s set on dn=1
>> chan=1 hold=1 callID=1547
>> **Dec  2 19:27:39.922: No MOH entry for DN 1**
>> **Dec  2 19:27:39.926: ephone_hold_resume ignored for s2s set on dn=1
>> chan=1 hold=0 callID=1547
>> *Dec  2 19:27:40.678: ifs_read flash:MobileConnectOn.ulaw.wav end of file
>> at 17214 read 4696 = 21910
>> *Dec  2 19:27:40.678: moh tail fill from 46 at 0x66BB855C length 3304
>> *Dec  2 19:27:40.882: MoH route If Vlan11 ETHERNET 177.2.11.1 via ARP
>> *Dec  2 19:27:40.882: MoH route If Loopback0 46 177.1.254.2 via
>> 177.1.254.2
>>
>>  
>>
>>  
>>
>> ** **
>>
>
>
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Re: [OSL | CCIE_Voice] Embedded BACD Prompts and DropThrough

[datucha123 datucha123]

1.   Are prompts also embedded in the IOS? Or do they need to be copied
in the router’s flash?

No, the Prompts are not embedded in the IOS, you need to manually add them
into Flash.


2.   Does drop through mode work with embedded BACD?
Yes, embedded BACD works for Drop Through Mode very well.

You can find the configuration examples here:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html#wp1002001




On Sun, Feb 19, 2012 at 7:59 AM, AJ BG  wrote:

> Two questions about embedded BACD.
>
> 1.   Are prompts also embedded in the IOS? Or do they need to be
> copied in the router’s flash?
>
> 2.   Does drop through mode work with embedded BACD?
>  Does anyone have a working copy of embedded BACD configuration?
> Thanks
> AJ
>
> ___
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> visit www.ipexpert.com
>
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>
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Re: [OSL | CCIE_Voice] Unity Connection off net messages

[datucha123 datucha123]

Are you using VMWare for you Unity Server?

If so then that behavour is observed when the Unity Connection is installed
in virtual environment.

I am also running to such issue some times, but the reboot of Unity server
helps me.






On Sun, Feb 19, 2012 at 4:16 AM, chase mergenthal wrote:

>  Something weird, I've run into; After doing the integration between CUCM
> and CUC, I can leave messages from users that have accounts on Unity, but
> not from the PSTN or other phones...
>
> Logs below:
>
> 02/18/2012 23:17:33.300
> |7915,PhoneSystem-PG-001,FBBA1FCBF2B8440982D981A38725901D,Arbiter,-1,Incoming
> Call [callerID='5002' callerName='Ron Paul' calledID='5600'
> redirectingID='' lastRedirectingID='' origin=16=Unknown xferType=16=Unknown
> reason=1=Direct lastReason=1024=Unknown] port=PhoneSystem-PG-001
> portsInUse=1 ansPortsFree=1 callGuid=FBBA1FCBF2B8440982D981A38725901D|
> 02/18/2012 23:22:40.717
> |7916,PhoneSystem-PG-001,DA7997B88186462C8AD2A77DC8265A4E,Arbiter,-1,Incoming
> Call [callerID='911' callerName='Emergency Services' calledID='5002'
> redirectingID='5002' lastRedirectingID='5002' origin=16=Unknown
> xferType=16=Unknown reason=4=FwdNoAnswer lastReason=4=FwdNoAnswer]
> port=PhoneSystem-PG-001 portsInUse=1 ansPortsFree=1
> callGuid=DA7997B88186462C8AD2A77DC8265A4E|
> 02/18/2012 23:22:43.090
> |7916,PhoneSystem-PG-001,DA7997B88186462C8AD2A77DC8265A4E,MiuGeneral,25,GetLicenseStatusLimit
> tag='LicRealspeakSessionsMax'. |
> 02/18/2012 23:22:53.858
> |7916,PhoneSystem-PG-001,DA7997B88186462C8AD2A77DC8265A4E,MiuGeneral,25,GetLicenseStatusLimit
> tag='LicMaxMsgRecLenIsLicensed'. |
> 02/18/2012 23:24:47.859
> |7915,PhoneSystem-PG-001,20C4C8D64C5D4E3892A77F79D6B5846D,Arbiter,-1,Incoming
> Call [callerID='3002' callerName='John Adams' calledID='5002'
> redirectingID='5002' lastRedirectingID='5002' origin=16=Unknown
> xferType=16=Unknown reason=4=FwdNoAnswer lastReason=4=FwdNoAnswer]
> port=PhoneSystem-PG-001 portsInUse=1 ansPortsFree=1
> callGuid=20C4C8D64C5D4E3892A77F79D6B5846D|
> 02/18/2012 23:24:48.594
> |7915,PhoneSystem-PG-001,20C4C8D64C5D4E3892A77F79D6B5846D,MiuGeneral,25,GetLicenseStatusLimit
> tag='LicRealspeakSessionsMax'. |
> 02/18/2012 23:24:58.690
> |7915,PhoneSystem-PG-001,20C4C8D64C5D4E3892A77F79D6B5846D,MiuGeneral,25,GetLicenseStatusLimit
> tag='LicMaxMsgRecLenIsLicensed'. |
> 02/18/2012 23:52:16.695
> |7916,PhoneSystem-PG-001,517F2E6866604E18AF7384BEC27BAC8F,Arbiter,-1,Incoming
> Call [callerID='911' callerName='Emergency Services' calledID='5002'
> redirectingID='5002' lastRedirectingID='5002' origin=16=Unknown
> xferType=16=Unknown reason=4=FwdNoAnswer lastReason=4=FwdNoAnswer]
> port=PhoneSystem-PG-001 portsInUse=1 ansPortsFree=1
> callGuid=517F2E6866604E18AF7384BEC27BAC8F|
> 02/18/2012 23:52:18.722
> |7916,PhoneSystem-PG-001,517F2E6866604E18AF7384BEC27BAC8F,MiuGeneral,25,GetLicenseStatusLimit
> tag='LicRealspeakSessionsMax'. |
> 02/18/2012 23:52:29.195
> |7916,PhoneSystem-PG-001,517F2E6866604E18AF7384BEC27BAC8F,MiuGeneral,25,GetLicenseStatusLimit
> tag='LicMaxMsgRecLenIsLicensed'. |
>
> -Chase
>
>
> --
> If winners never quit and quitters never win, then who coined the phrase,
> "Quit while you’re still ahead."?
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
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Re: [OSL | CCIE_Voice] B-ACD max-time-call-retry timer

[datucha123 datucha123]

I also had such issue some period ago,

And I am also interested in the answer.

On Sat, Feb 18, 2012 at 11:03 PM, Baktha Muralidharan
wrote:

> Hello all,
>
> per the B-ACD guide, the definition of max-time-call-retry is as follows-
>
> *If the amount of time a call is in the queue exceeds the limit set by
> the param max-time-call-retry command, the call is routed to the alternate
> destination set with the param voice-mail command. If the param
> max-time-vm-retry command is set to a number higher than one, the
> call-queue service retries to connect that number of times.*
>
> I have the following B-ACD config-
>
> application
>  service app-b-acd-aa
>   param voice-mail 4220
>   paramspace english index 1
>   param *max-time-call-retry 30*
>   param service-name app-b-acd
>   param number-of-hunt-grps 1
>   param drop-through-option 2
>   paramspace english language en
>   param handoff-string app-b-acd-aa
>   param *max-time-vm-retry 0*
>   paramspace english location flash:
>   param aa-pilot 4019
>   param second-greeting-time 10
>   param welcome-prompt _bacd_welcome.au
>   param call-retry-timer 5
>  !
>  service app-b-acd
>   param queue-len 15
>   param number-of-hunt-grps 1
>   param aa-hunt2 4111
>   param queue-manager-debugs 1
>
> Yet, calls hit voicemail ONLY after a minute (60 secs).
>
> Any hints ? Anybody able to get max-time-call-retry to work?
>
> Thanks in advance,
> /Baktha
>
> ___
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Re: [OSL | CCIE_Voice] RSVP over MLPPP

[datucha123 datucha123]

RSVP may not work over True MLPP (I do not know exactly).

But for MLP Over Frame Relay it will work.

On Sat, Feb 18, 2012 at 6:58 PM, Radhesh Naik  wrote:

>  Hi,
>
> ** **
>
> Came across this statement under SRND.
>
> ** **
>
> “RSVP is currently not available on Bundle Interfaces, including MLPPP,…”*
> ***
>
> ** **
>
> Does this mean we can’t configure RSVP on an interface that is configured
> with MLP ? 
>
> ** **
>
> Because auto qos fr-atm will configure the interface as  MLP LFI.
>
> ** **
>
> I am sure I am missing something here, enlighten please.
>
> ** **
>
> Regards,
>
> ** **
>
> Radhesh
>
> ___
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> visit www.ipexpert.com
>
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Re: [OSL | CCIE_Voice] New Labs Lab 4 7.4

[datucha123 datucha123]

No, as I remember, Paging is not supported on SIP Phones, at least in
version 7.0

On Fri, Feb 17, 2012 at 6:33 AM, chase mergenthal wrote:

>  Is paging supported on SIP phones?
>
> Got it to work on my SCCP phones just fine...
>
> -Chase
>
>
> --
> If winners never quit and quitters never win, then who coined the phrase,
> "Quit while you’re still ahead."?
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
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> www.PlatinumPlacement.com 
>
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Re: [OSL | CCIE_Voice] enabling Chat option in Cisco Unified personal communicator

[datucha123 datucha123]

Well, I think you have to add contacts to you CUPC client.

But here are some restrictions:

1) When you add Contacts from CUPC client directly, those are just
contacts, and they do not support Presence.
2) To support Presence (Chat also will be supported) you need to add
contacts from End User Web page of Presence Server  -
http:///ccmuser    and sign in
with End User username and password, then fill out the Contacts there.
3) The mentioned method (2), will support Presence for other Users who are
also having CUPC. So to configure the Presence for users endpoints (like IP
Phone DN's) you have to associate those DNs with corresponding End Users as
well, along with section 2, for watcher.
4) Integrate CUPS with AD, and let the users to pull out the Contacts from
AD using CUPC. In this case, CUPC will support Presence (Chat) with added
contacts.

Also do not forget to assign the License Capabilites to all End Users, that
must be part of Presence and CUPC, so that they will be visible by Presence
Server. And Chat option will be available for you added users in CUPC.

If that is not the right answer you were expecting, then please right in
more details what are you trying to achieve.

On Thu, Feb 16, 2012 at 12:38 AM, darshan  wrote:

>  Dear Experts;
>
> ** **
>
> I want to enable CHAT option in Cisco Unified Personal communicator for
> Presence question.
>
> ** **
>
> Appreciate your help for this.
>
> ** **
>
> Regards
>
> darsh
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
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Re: [OSL | CCIE_Voice] Query wrt to Cbarge on BR1 router without using the CME for SRST

[datucha123 datucha123]

You can also configure the Privacy Settings globally, at Telephony-service
configuration. with "no privacy" command, so that you will not need to
disable it per ephone.



On Wed, Feb 15, 2012 at 11:10 PM, Vik Malhi  wrote:

> I don't see the restriction in OWLE lab #2.
>
> The HA question 7.1 states "·   You should ensure that all learned
> ephones and ephone-dn’s appear in the running config to achieve this task."
>
> This indicates srst mode auto-provision all which is required for cBarge
> preservation since privacy needs to be disabled at the ephone level.
>
> Let me know the details of the lab and question number and I'll try and
> clear it up- but nonetheless- you are correct in what you have stated- you
> cannot preserve cBarge with call-manager-fallback
>
> Thanks
>
>
>  Vik Malhi – CCIE #13890
> Managing Partner - IPexpert, Inc.
>
> Telephone: +1.810.326.1444 ext 420
> Fax: +1.810.454.0130
> Mailto: vma...@ipexpert.com
>
>
>
>
>   On Feb 15, 2012, at 4:12 PM, Rajasekar Shanmugam wrote:
>
>   Experts -
>
> I`m practicing the scenario 2 from the IP Expert OWLE series & the
> question asks us ,  not to use the CME based SRST for BR1. So we are forced
> to use the call-manager-fallback. There is a requirement later in the lab
> ,asking for Cbarge functionality on SRST. Wondering , if we have an option
> to register the hardware media resources (CFB) with the
> call-manager-fallback to get this working ? The solution guide suggests to
> configure the telephony service in order to do so. Confused here won`t
> that break the original requirement  in the HA section , that asked us not
> to use the CME SRST ? Please advise.
>
> --
> Raj
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
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Re: [OSL | CCIE_Voice] Multicast MOH

[datucha123 datucha123]

I have tested that, and the MMoH is working.

Sorry for incorrect information.

In previous test I had the MMoH binded to incorrect interface.

Now it is working

I am burning with shame, sorry

On Wed, Feb 15, 2012 at 4:32 PM, George Goglidze  wrote:

> Datucha,
>
> I'm hearing the music perfectly :-)
>
> If you can't make it work, that's something wrong with your configuration,
> but don't say it won't work! :-)))
>
> And please tell me where in CUCME admin guide it says such thing I'll
> eat my hat if you show me the section!
>
> Cheers,
>
> On Wed, Feb 15, 2012 at 1:26 PM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>> You will not be able to hear the Music itslef.
>>
>> In my testing the Multicast was also traversed to my Remote Branch Router
>> - I was able to see the Multicast stream and endpoints joined  there with
>> "show ip mroute" and "show ip igmp groups", but the actual stream of Music
>> was not heard, based on CUCME Admin guide.
>>
>>  On Wed, Feb 15, 2012 at 3:08 PM, George Goglidze wrote:
>>
>>> and voila, here's the proof:
>>> sorry but had to remove IP addresses for security reasons.
>>>
>>> I'm using pim sparse-dense mode, with auto RP.
>>>
>>> mmohrtr#show ip pim neighbor
>>> PIM Neighbor Table
>>> Mode: B - Bidir Capable, DR - Designated Router, N - Default DR Priority,
>>>   P - Proxy Capable, S - State Refresh Capable, G - GenID Capable
>>> Neighbor  InterfaceUptime/ExpiresVer   DR
>>> Address
>>> Prio/Mode
>>> x.x.x.x  GigabitEthernet0/1   00:23:38/00:01:24 v21 / G
>>> *this address is the FW! *
>>>
>>> mmohrtr#show ip mroute
>>> IP Multicast Routing Table
>>> Flags: D - Dense, S - Sparse, B - Bidir Group, s - SSM Group, C -
>>> Connected,
>>>L - Local, P - Pruned, R - RP-bit set, F - Register flag,
>>>T - SPT-bit set, J - Join SPT, M - MSDP created entry, E -
>>> Extranet,
>>>X - Proxy Join Timer Running, A - Candidate for MSDP
>>> Advertisement,
>>>U - URD, I - Received Source Specific Host Report,
>>>Z - Multicast Tunnel, z - MDT-data group sender,
>>>Y - Joined MDT-data group, y - Sending to MDT-data group,
>>>V - RD & Vector, v - Vector
>>> Outgoing interface flags: H - Hardware switched, A - Assert winner
>>>  Timers: Uptime/Expires
>>>  Interface state: Interface, Next-Hop or VCD, State/Mode
>>>
>>> (*, 239.1.1.1), 00:06:33/00:02:56, RP x.x.x.x, flags: SP
>>>   Incoming interface: Null, RPF nbr 0.0.0.0
>>>   Outgoing interface list: Null
>>>
>>> (*, 224.0.1.39), 00:30:17/stopped, RP 0.0.0.0, flags: DCL
>>>   Incoming interface: Null, RPF nbr 0.0.0.0
>>>   Outgoing interface list:
>>> GigabitEthernet0/1, Forward/Sparse-Dense, 00:25:58/stopped
>>> Loopback1, Forward/Sparse-Dense, 00:30:17/stopped
>>>
>>> (x.x.x.x, 224.0.1.39), 00:30:16/00:02:42, flags: LT
>>>   Incoming interface: Loopback1, RPF nbr 0.0.0.0
>>>   Outgoing interface list:
>>> GigabitEthernet0/1, Forward/Sparse-Dense, 00:25:58/stopped
>>>
>>> (*, 224.0.1.40), 00:30:16/stopped, RP 0.0.0.0, flags: DCL
>>>   Incoming interface: Null, RPF nbr 0.0.0.0
>>>   Outgoing interface list:
>>> GigabitEthernet0/1, Forward/Sparse-Dense, 00:25:57/stopped
>>> Loopback1, Forward/Sparse-Dense, 00:30:16/stopped
>>>
>>> (x.x.x.x, 224.0.1.40), 00:30:16/00:02:24, flags: LT
>>>   Incoming interface: Loopback1, RPF nbr 0.0.0.0
>>>   Outgoing interface list:
>>> GigabitEthernet0/1, Forward/Sparse-Dense, 00:25:57/stopped
>>>
>>>
>>> And I created a SPAN port, on the switch, and spanned all traffic from
>>> mmohrtr to the Firewall!!!
>>> The part of screenshot attached... which says TTL 255 as I said!
>>>
>>> Regards,
>>>
>>>
>>> On Wed, Feb 15, 2012 at 11:54 AM, George Goglidze wrote:
>>>
>>>> Datucha,
>>>>
>>>> I have it configured right now in the lab... and it works just fine.
>>>>
>>>> VGW -> FW -> L3 Switch -> Phones.
>>>>
>>>> Works just fine...
>>>>
>>>> I will do some wireshark traces and show the packets.
>>>>
>>>>
>>>> On Wed, Feb 15, 2012 at 10:53 AM, da

Re: [OSL | CCIE_Voice] Multicast MOH

[datucha123 datucha123]

I cannot find it any more :(((

But I remember, that I have read about that somewhere.

Ok i will make the test again, (10 minutes) and will post the results.



On Wed, Feb 15, 2012 at 4:32 PM, George Goglidze  wrote:

> Datucha,
>
> I'm hearing the music perfectly :-)
>
> If you can't make it work, that's something wrong with your configuration,
> but don't say it won't work! :-)))
>
> And please tell me where in CUCME admin guide it says such thing I'll
> eat my hat if you show me the section!
>
> Cheers,
>
> On Wed, Feb 15, 2012 at 1:26 PM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>> You will not be able to hear the Music itslef.
>>
>> In my testing the Multicast was also traversed to my Remote Branch Router
>> - I was able to see the Multicast stream and endpoints joined  there with
>> "show ip mroute" and "show ip igmp groups", but the actual stream of Music
>> was not heard, based on CUCME Admin guide.
>>
>>  On Wed, Feb 15, 2012 at 3:08 PM, George Goglidze wrote:
>>
>>> and voila, here's the proof:
>>> sorry but had to remove IP addresses for security reasons.
>>>
>>> I'm using pim sparse-dense mode, with auto RP.
>>>
>>> mmohrtr#show ip pim neighbor
>>> PIM Neighbor Table
>>> Mode: B - Bidir Capable, DR - Designated Router, N - Default DR Priority,
>>>   P - Proxy Capable, S - State Refresh Capable, G - GenID Capable
>>> Neighbor  InterfaceUptime/ExpiresVer   DR
>>> Address
>>> Prio/Mode
>>> x.x.x.x  GigabitEthernet0/1   00:23:38/00:01:24 v21 / G
>>> *this address is the FW! *
>>>
>>> mmohrtr#show ip mroute
>>> IP Multicast Routing Table
>>> Flags: D - Dense, S - Sparse, B - Bidir Group, s - SSM Group, C -
>>> Connected,
>>>L - Local, P - Pruned, R - RP-bit set, F - Register flag,
>>>T - SPT-bit set, J - Join SPT, M - MSDP created entry, E -
>>> Extranet,
>>>X - Proxy Join Timer Running, A - Candidate for MSDP
>>> Advertisement,
>>>U - URD, I - Received Source Specific Host Report,
>>>Z - Multicast Tunnel, z - MDT-data group sender,
>>>Y - Joined MDT-data group, y - Sending to MDT-data group,
>>>V - RD & Vector, v - Vector
>>> Outgoing interface flags: H - Hardware switched, A - Assert winner
>>>  Timers: Uptime/Expires
>>>  Interface state: Interface, Next-Hop or VCD, State/Mode
>>>
>>> (*, 239.1.1.1), 00:06:33/00:02:56, RP x.x.x.x, flags: SP
>>>   Incoming interface: Null, RPF nbr 0.0.0.0
>>>   Outgoing interface list: Null
>>>
>>> (*, 224.0.1.39), 00:30:17/stopped, RP 0.0.0.0, flags: DCL
>>>   Incoming interface: Null, RPF nbr 0.0.0.0
>>>   Outgoing interface list:
>>> GigabitEthernet0/1, Forward/Sparse-Dense, 00:25:58/stopped
>>> Loopback1, Forward/Sparse-Dense, 00:30:17/stopped
>>>
>>> (x.x.x.x, 224.0.1.39), 00:30:16/00:02:42, flags: LT
>>>   Incoming interface: Loopback1, RPF nbr 0.0.0.0
>>>   Outgoing interface list:
>>> GigabitEthernet0/1, Forward/Sparse-Dense, 00:25:58/stopped
>>>
>>> (*, 224.0.1.40), 00:30:16/stopped, RP 0.0.0.0, flags: DCL
>>>   Incoming interface: Null, RPF nbr 0.0.0.0
>>>   Outgoing interface list:
>>> GigabitEthernet0/1, Forward/Sparse-Dense, 00:25:57/stopped
>>> Loopback1, Forward/Sparse-Dense, 00:30:16/stopped
>>>
>>> (x.x.x.x, 224.0.1.40), 00:30:16/00:02:24, flags: LT
>>>   Incoming interface: Loopback1, RPF nbr 0.0.0.0
>>>   Outgoing interface list:
>>> GigabitEthernet0/1, Forward/Sparse-Dense, 00:25:57/stopped
>>>
>>>
>>> And I created a SPAN port, on the switch, and spanned all traffic from
>>> mmohrtr to the Firewall!!!
>>> The part of screenshot attached... which says TTL 255 as I said!
>>>
>>> Regards,
>>>
>>>
>>> On Wed, Feb 15, 2012 at 11:54 AM, George Goglidze wrote:
>>>
>>>> Datucha,
>>>>
>>>> I have it configured right now in the lab... and it works just fine.
>>>>
>>>> VGW -> FW -> L3 Switch -> Phones.
>>>>
>>>> Works just fine...
>>>>
>>>> I will do some wireshark traces and show the packets.
>>>>
>>>>
>>>> On Wed, Feb 15, 2012 at 10:53 AM, datucha123 datucha123 &

Re: [OSL | CCIE_Voice] Multicast MOH

[datucha123 datucha123]

You will not be able to hear the Music itslef.

In my testing the Multicast was also traversed to my Remote Branch Router -
I was able to see the Multicast stream and endpoints joined  there with
"show ip mroute" and "show ip igmp groups", but the actual stream of Music
was not heard, based on CUCME Admin guide.

On Wed, Feb 15, 2012 at 3:08 PM, George Goglidze  wrote:

> and voila, here's the proof:
> sorry but had to remove IP addresses for security reasons.
>
> I'm using pim sparse-dense mode, with auto RP.
>
> mmohrtr#show ip pim neighbor
> PIM Neighbor Table
> Mode: B - Bidir Capable, DR - Designated Router, N - Default DR Priority,
>   P - Proxy Capable, S - State Refresh Capable, G - GenID Capable
> Neighbor  InterfaceUptime/ExpiresVer   DR
> Address
> Prio/Mode
> x.x.x.x  GigabitEthernet0/1   00:23:38/00:01:24 v21 / G
> *this address is the FW! *
>
> mmohrtr#show ip mroute
> IP Multicast Routing Table
> Flags: D - Dense, S - Sparse, B - Bidir Group, s - SSM Group, C -
> Connected,
>L - Local, P - Pruned, R - RP-bit set, F - Register flag,
>T - SPT-bit set, J - Join SPT, M - MSDP created entry, E - Extranet,
>X - Proxy Join Timer Running, A - Candidate for MSDP Advertisement,
>U - URD, I - Received Source Specific Host Report,
>Z - Multicast Tunnel, z - MDT-data group sender,
>Y - Joined MDT-data group, y - Sending to MDT-data group,
>V - RD & Vector, v - Vector
> Outgoing interface flags: H - Hardware switched, A - Assert winner
>  Timers: Uptime/Expires
>  Interface state: Interface, Next-Hop or VCD, State/Mode
>
> (*, 239.1.1.1), 00:06:33/00:02:56, RP x.x.x.x, flags: SP
>   Incoming interface: Null, RPF nbr 0.0.0.0
>   Outgoing interface list: Null
>
> (*, 224.0.1.39), 00:30:17/stopped, RP 0.0.0.0, flags: DCL
>   Incoming interface: Null, RPF nbr 0.0.0.0
>   Outgoing interface list:
> GigabitEthernet0/1, Forward/Sparse-Dense, 00:25:58/stopped
> Loopback1, Forward/Sparse-Dense, 00:30:17/stopped
>
> (x.x.x.x, 224.0.1.39), 00:30:16/00:02:42, flags: LT
>   Incoming interface: Loopback1, RPF nbr 0.0.0.0
>   Outgoing interface list:
> GigabitEthernet0/1, Forward/Sparse-Dense, 00:25:58/stopped
>
> (*, 224.0.1.40), 00:30:16/stopped, RP 0.0.0.0, flags: DCL
>   Incoming interface: Null, RPF nbr 0.0.0.0
>   Outgoing interface list:
> GigabitEthernet0/1, Forward/Sparse-Dense, 00:25:57/stopped
> Loopback1, Forward/Sparse-Dense, 00:30:16/stopped
>
> (x.x.x.x, 224.0.1.40), 00:30:16/00:02:24, flags: LT
>   Incoming interface: Loopback1, RPF nbr 0.0.0.0
>   Outgoing interface list:
> GigabitEthernet0/1, Forward/Sparse-Dense, 00:25:57/stopped
>
>
> And I created a SPAN port, on the switch, and spanned all traffic from
> mmohrtr to the Firewall!!!
> The part of screenshot attached... which says TTL 255 as I said!
>
> Regards,
>
>
> On Wed, Feb 15, 2012 at 11:54 AM, George Goglidze wrote:
>
>> Datucha,
>>
>> I have it configured right now in the lab... and it works just fine.
>>
>> VGW -> FW -> L3 Switch -> Phones.
>>
>> Works just fine...
>>
>> I will do some wireshark traces and show the packets.
>>
>>
>> On Wed, Feb 15, 2012 at 10:53 AM, datucha123 datucha123 <
>> datucha...@gmail.com> wrote:
>>
>>> George,
>>>
>>>  Flash MMoH cannot be traversed through L3.
>>>
>>> So that the MMoH is not supported for remote IP Phones on CUCME.
>>>
>>> CUCME Admin guide says that. Unfortunately I do not remember the exact
>>> page, but its there. Also I have tested that and it is true, the MMoH From
>>> flash does not go to remote IP Phones (for CUCME or SRST even) which are on
>>> subnet separated by L3 from the CUCME Routers interface.
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Wed, Feb 15, 2012 at 11:56 AM, George Goglidze wrote:
>>>
>>>> Hi Boris,
>>>>
>>>> You are wrong, the TTL on multicast music on hold is NOT 1...
>>>> and depending on what you need, you can have pim enabled and deliver
>>>> the mmoh from flash to other remote networks, and by remote I mean across
>>>> any L3 device.
>>>>
>>>> As well, you do not have a way to specify the TTL on the mmoh from
>>>> router flash like in CUCM where you can specify the TTL you need.
>>>> So it will be 255 I believe. I would have to check on that.
>>>>
>>>>
>>>> I hope this answers the original question as well

Re: [OSL | CCIE_Voice] Multicast MOH

[datucha123 datucha123]

George,

 Flash MMoH cannot be traversed through L3.

So that the MMoH is not supported for remote IP Phones on CUCME.

CUCME Admin guide says that. Unfortunately I do not remember the exact
page, but its there. Also I have tested that and it is true, the MMoH From
flash does not go to remote IP Phones (for CUCME or SRST even) which are on
subnet separated by L3 from the CUCME Routers interface.






On Wed, Feb 15, 2012 at 11:56 AM, George Goglidze wrote:

> Hi Boris,
>
> You are wrong, the TTL on multicast music on hold is NOT 1...
> and depending on what you need, you can have pim enabled and deliver the
> mmoh from flash to other remote networks, and by remote I mean across any
> L3 device.
>
> As well, you do not have a way to specify the TTL on the mmoh from router
> flash like in CUCM where you can specify the TTL you need.
> So it will be 255 I believe. I would have to check on that.
>
>
> I hope this answers the original question as well, if you need mmoh from
> router flash to traverse any other layer3 devices, you need to enable PIM.
> otherwise you don't.
>
> Cheers,
>
>
> On Wed, Feb 15, 2012 at 12:29 AM, Boris  wrote:
>
>>  Afaik you never need multicast routing configured for the Moh from
>> flash because this traffic has TTL=1, so it wont traverse Layer 3 boundary.
>>
>> This is why you need the route attribute in the moh statement.
>>
>> Sent from my mobile device, sorry for typos.
>> ---
>> Regards
>> Boris
>>
>> On 15/02/2012, at 8:28, Emanuel Damasceno  wrote:
>>
>>  Hello Experts,
>>
>> I am watching IPExpert's video on demand on Multicast MOH.
>>
>> Did I understand this right?
>>
>> "multicast moh 239.1.1.1 port 16384" < if I have it like this, I have to
>> put "ip pim dense-mode" on my interfaces
>>
>> "multicast moh 239.1.1.1 port 16384 route 10.10.201.1 10.10.110.2" < if I
>> have it like this, I DON'T have to put "ip pim dense-mode" on my interfaces
>>
>> Could somebody please confirm?
>> Thanks*
>> Emanuel Damasceno*
>> CCNP Voice
>>
>>
>>
>>  ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com 
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com 
>>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
___
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Re: [OSL | CCIE_Voice] built-in bridge support for iLBC & g729

[datucha123 datucha123]

Interesting question.

I think that Phones Built in Bridge does support G729 and other codecs,
which are supported by IP Phone.

On Wed, Feb 15, 2012 at 10:30 AM, Juan Lopez  wrote:

> can anyone confirm the phone built-in bridge does support g729? I'm
> running it in my lab, no HW CFB are configured and I put all SW CFB in
> their own MRG, to be sure they are not used:
>
> admin:show perf query class "Cisco SW Conference Bridge Device"
> ==>query class :
>  - Perf class (Cisco SW Conference Bridge Device) has instances and values:
> CFB_2   -> AllocatedResourceCannotOpenPort = 0
> CFB_2   -> OutOfResources = 0
> CFB_2   -> ResourceActive = 0
> CFB_2   -> ResourceAvailable  = 48
> CFB_2   -> ResourceTotal  = 48
> CFB_2   -> SWConferenceActive = 0
> CFB_2   -> SWConferenceCompleted  = 0
> CFB_3   -> AllocatedResourceCannotOpenPort = 0
> CFB_3   -> OutOfResources = 0
> CFB_3   -> ResourceActive = 0
> CFB_3   -> ResourceAvailable  = 48
> CFB_3   -> ResourceTotal  = 48
> CFB_3   -> SWConferenceActive = 0
> CFB_3   -> SWConferenceCompleted  = 0
>
> admin:show perf query class "Cisco HW Conference Bridge Device"
> ==>query class :
>  - Perf class (Cisco HW Conference Bridge Device) has instances and values:
> no values are returned
>
> Nevertheless, the barge function is working: barger and bargee use g722,
> the 3rd party uses g729. same goes is 3rd party would use iLBC.
>
> thanks!
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
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[OSL | CCIE_Voice] MTP and Xcoders

[datucha123 datucha123]

Hello,

Assume that we have G711 and G729 MTP's in first MRG, and then Xcoder in
the next MRG:

MRGL oder:

1) MRG_MTP

2) MRG_Xcoder

So as we know, CUCM can allocate the required MTP's based on codec -  if
call is going to be g711 then the G711 MTP will be allocated, and if the
call is going to be G729, then G729 MTP will be allocated.

But it is also possible for G729 call to use G711 MTP but with Transcoder
support  - IP Phone G729 to Xcoder G729, Xcoder G711 to MTP G711.



So when the G729 call comes to CUCM, will the CUCM allocate G729 MTP, or
will try to allocated G711 with Transacoder?

Here is also another example, with the same question:

MRGL order:

1) MRG_MTP_G711

2)MRG_MTP_G729

3)MRG_Xcoder

In second example, as the G729 call can use G711 MTP with Transcoder
support, will the CUCM allocate G711 MTP along with Transcoder, as the G711
MTP is first in order, or it will look down and allocate G729 MTP?
___
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Re: [OSL | CCIE_Voice] New 5 Labs: Lab 2 UXXC

[datucha123 datucha123]

Prompt  - >  "Create Generated Prompt", where you have to point the
String which you want to play out (in your case it would be String with
value of "1" or "2" or anything you like to announce/play out.

You have to select the Constructor Type as well, as it rules how the Number
is played out gramaitically - 1 as "First", 1 as "One" and etc, (there are
some Types like Number, Telephone Number and etc. You can get the
documentation for each type in UCCX Scripting Guides, or even in UCCX Scipr
Help Files).

Select the Prompt File where you want to save the Generated Prompt.

You can also choose the language, or leave it as it is.

After you have to use the "Play Prompt" step to play previously
generated Prompt.

On Sun, Feb 12, 2012 at 5:45 PM, George Goglidze  wrote:

>  There is a step to generate a number... M not in front of UCCX right now
> so can't tell you exact name of the step, but it is very easy to find it.
>
> In this step you tell it what you want it to gemerate, in your case
> number, and then provide it with the value, and then output prompt.
>
> Sent from my iPad
>
> On 11 Feb 2012, at 21:09, Randall Crumm  wrote:
>
>   HI,
> I know the steps I need to do but I am unsure of the prompts I need to
> record.
>
> Please let know what I need to record.
>
> I believe I need to record:
>
> 1. The number of people ahead of you is
> 2. zero
> 3. one
>
> Please let me know
>
> Cheers,
> Randall
>
>  ___
>
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
___
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[OSL | CCIE_Voice] NTP for UCCX

[datucha123 datucha123]

Hello,

Do we need to set the NTP Server for UCCX Win Server? Or the NTP settings
during the Initial Setup is enough? Well, at least for the exam.

I cannot set the NTP Server for Win2003 (UCCX Server)  -  "net time"
command does not work, as it is looking for Win Server NTP, and is not
conencting to IOS NTP.
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Re: [OSL | CCIE_Voice] New 5 Labs: Lab 2 UXXC

[datucha123 datucha123]

Zero and One are already pre-recorded (by default) in UCCX, so you can use
them.

On Sun, Feb 12, 2012 at 12:09 AM, Randall Crumm  wrote:

>  HI,
> I know the steps I need to do but I am unsure of the prompts I need to
> record.
>
> Please let know what I need to record.
>
> I believe I need to record:
>
> 1. The number of people ahead of you is
> 2. zero
> 3. one
>
> Please let me know
>
> Cheers,
> Randall
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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

[OSL | CCIE_Voice] MMoH RSVP CAC

[datucha123 datucha123]

As we know, the Multicast MoH is not counted against the CAC bandwidth, but
the Priority Queue does. So when the Branch devices (IP Phones/Gateway) are
using the MMoH sourced from the HQ CUCM Servers, we need to take the MoH
bandwidth into account for LLQ Priority queue, but not for CAC, as
mutlicast is not counted for RSVP CAC. And at some point we may get the
Oversubscriptio for the WAN link.

Well, that's ok, but how much bandwidth must be taken into acount for LLQ
Priority queue for Multicast MoH? I mean if using G729 MMoH, then how much
MMoH sessions we have to consider when configuring the LLQ Priority Queue?
What it is based on? Becuase we do not know, how much Holds will be at the
same time on the Branch site.
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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

Re: [OSL | CCIE_Voice] MVA problem

[datucha123 datucha123]

As I have mentioned, the removing CAC is not a solution.

So i will try to continue figuring it out and as soon as I will get
solution, I will definitely inform.

All my Regions setting are correct. I think there is something wrong with
Gateways, as different Gateways are selected in each case.

On Fri, Feb 10, 2012 at 5:25 AM, Vik Malhi  wrote:

>  I found the problem, but another one has arise.
>
> Please inform us of the problem/solution for the benefit of others
> following this thread.
>
>
> Q1: Can you confirm that when you answer the cell phone (working scenario)
> which gateway is being used to get to PSTN.
>
> Q2: Can you confirm that when you answer the deskphone (not working
> scenario) and transfer to cell phone which gateway is being used to get to
> PSTN.
>
> I suspect there is different gateway being used due to you using SLRG
> which could explain the differences.
>
>
>
>  Vik Malhi – CCIE #13890
> Managing Partner - IPexpert, Inc.
>
> Telephone: +1.810.326.1444 ext 420
> Fax: +1.810.454.0130
> Mailto: vma...@ipexpert.com
>
>
>
>
>   On Feb 9, 2012, at 9:24 AM, datucha123 datucha123 wrote:
>
>   I found the problem, but another one has arise.
>
> I am using RSVP as CAC between site, where the RSVP MTP has only G729
> codec enabled. And also the MoH is using Unicast so that it is subject to
> CAC.
>
> So now when the call is picked up by Mobile Phone and then dropped, the
> MoH plays good, as the BR1 site is using G729 to MoH Server. But when the
> HQ Desk Phone picks up the call first and then redirects to Mobile phone,
> after hung up the MoH does not work as the BR1 phone (somehow, why it makes
> so I do not know) is trying to use G711 to MoH Server, where the RSVP MTP
> does not pass the G711 traffic and that is why the ToH is heard.
>
> Now when I have changed the BR1 to use the Local Flash MMoH, everything
> was working fine (MoH was heard always).
>
> But still no idea, why the BR1 is trying to negotiate G711 to MoH server
> after the Mobile has dropped the call (when the Desk had sent the call to
> Mobile).
>
> Also I can see that when the Mobile Phone picks up the call first, the BR1
> Phone is using G711 (call goes through SLRG BR1 local gateway).
> But when the HQ Phone picks up the call first, and then sends the call to
> Mobile, the BR1 phone shows that it is using G729, thus the call is going
> though local Gateway.
>
>
>
>
> On Thu, Feb 9, 2012 at 2:37 AM, Mohammed Al Baqari <
> baqari.voic...@gmail.com> wrote:
>
>>  Hi Datucha,
>>
>> ** **
>>
>> Now when the BR1 phone calls this HQ phone, so that the mobile phone also
>> ring, and when the Mobile Phone picks up the call first, then the MoH works
>> when the mobile is hunged up, on the BR1 phone.
>>
>> But if the HQ phone has picked up the call first and then made the Send
>> Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and
>> instead the ToH is heard when the mobile Phone disconnects. 
>>
>> I suggest to set two different network moh source files. Assign one to
>> RDP and one to HQ Phone. Then repeat your test scenarios above and lets
>> know which MoH file is used in each case.
>>
>> ** **
>>
>> Also, please share the traces regarding the codec part.
>>
>> ** **
>>
>> ** **
>>
>> Regards,
>>
>> Mohammed Al Baqari
>>
>> ** **
>>
>> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
>> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *datucha123
>> datucha123
>> *Sent:* Wednesday, February 08, 2012 11:28 PM
>> *To:* ccie_voice@onlinestudylist.com
>> *Subject:* [OSL | CCIE_Voice] MVA problem
>>
>> ** **
>>
>> I have the following kind of probem:
>>
>> I am using SLRG for Mobile Connect calls, so that that calls to users
>> mobiles are done through local gateway (this is just for test).
>>
>> Now, the HQ phone has the RDP assinged with RD of his mobile phone.
>>
>> Now when the BR1 phone calls this HQ phone, so that the mobile phone also
>> ring, and when the Mobile Phone picks up the call first, then the MoH works
>> when the mobile is hunged up, on the BR1 phone.
>>
>> But if the HQ phone has picked up the call first and then made the Send
>> Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and
>> instead the ToH is heard when the mobile Phone disconnects. 
>>
>>  
>>
>> Also I have noticed that when the Mobile Phone picks up the call faster
>> then the Desk Phone, the codec negotiat

Re: [OSL | CCIE_Voice] MVA problem

[datucha123 datucha123]

I have also tryed to reconfigure the Regions so:

HQ MTP placed into separate Region, which has G729 to everywhere (HQ
devices, included MoH, BR1 devices).

But still not success, the ToH is heard.

On Fri, Feb 10, 2012 at 6:43 AM, Mohd Baqari wrote:

>  Check whats the region relation between HQ MTP RSPV and MoH. It should
> be g729. Also same between BR1 MTP RSVP and HQ MTP RSVP.
>
> If this is done then MoH should negotiate g729 with BR1.
>
> Regards,
> Mohammed Al Baqari
>
> Sent from my iPhone
>
> On Feb 9, 2012, at 9:52 PM, datucha123 datucha123 
> wrote:
>
>  Here is what I get on HQ Router during the call hung up on the Mobile
> Phone:
>
> Feb 10 03:48:06.063: %ISDN-6-DISCONNECT: Interface Serial0/2/0:0
> disconnected from 206501 , call lasted 18 seconds
> Feb 10 03:48:06.187: RSVP-MSG:
> 177.1.254.2_18762->177.1.254.1_16464[0.0.0.0]: Received PathTear message
> from 177.1.254.2 (on Serial0/3/0.1)
> Feb 10 03:48:06.187: RSVP: 177.1.254.2_18762->177.1.254.1_16464[0.0.0.0]:
> Expiring Serial0/3/0.1 PATH state, reason: PathTear arrival
> Feb 10 03:48:06.187: RSVP: 177.1.254.2_18762->177.1.254.1_16464[0.0.0.0]:
> Expiring  RESV state, reason: PathTear arrival
> Feb 10 03:48:06.187: RSVP: 177.1.254.2_18762->177.1.254.1_16464[0.0.0.0]:
> Expiring receiver host RESV state, reason: PathTear arrival (17:18762)
> Feb 10 03:48:06.187: RSVP: 177.1.254.2_18762->177.1.254.1_16464[0.0.0.0]:
> Expiring Serial0/3/0.1 RESV request state, reason: PathTear arrival
> Feb 10 03:48:06.195: RSVP-MSG:
> 177.1.254.1_16464->177.1.254.2_18762[0.0.0.0]: Received ResvTear message
> from 177.0.101.2 (on Serial0/3/0.1)
> Feb 10 03:48:06.195: RSVP: 177.1.254.1_16464->177.1.254.2_18762[0.0.0.0]:
> Expiring Serial0/3/0.1 RESV state, reason: ResvTear arrival
> Feb 10 03:48:06.195: RSVP: 177.1.254.1_16464->177.1.254.2_18762[0.0.0.0]:
> Expiring Serial0/3/0.1 RESV state, reason: ResvTear arrival (17:16464)
> Feb 10 03:48:06.195: RSVP: 177.1.254.1_16464->177.1.254.2_18762[0.0.0.0]:
> Expiring sender host PATH state, reason: Local application requested tear
> Feb 10 03:48:06.195: RSVP: 177.1.254.1_16464->177.1.254.2_18762[0.0.0.0]:
> building hop object with src addr: 177.0.101.1
> Feb 10 03:48:06.215: RSVP: 177.1.254.1_16464->177.1.254.2_18762[0.0.0.0]:
> Sending PathTear message to 177.1.254.2
>
>
> HQ Interface:
>
> interface Serial0/3/0.1 point-to-point
>  description == FR To BR1 (R2)
>  ip address 177.0.101.1 255.255.255.0
>  snmp trap link-status
>  frame-relay interface-dlci 101
>  ip rsvp bandwidth 500
> end
>
> BR1 Interface:
>
> interface Serial0/3/0.1 point-to-point
>  description == FR To HQ (R1)
>  ip address 177.0.101.2 255.255.255.0
>  snmp trap link-status
>  frame-relay interface-dlci 101
>  ip rsvp bandwidth 500
> end
>
> So as I guess, reservation is not taking place after the Mobile Phone has
> hung up.
>
> Also here is the output before the HQ Desk Phone has sent the call to
> Mobile Phone:
>
> Feb 10 03:47:39.587: RSVP: session 177.1.254.1_19398[0.0.0.0]: Received
> Resv message from 127.0.0.1 (on receiver host)
> Feb 10 03:47:39.587: RSVP: 177.1.254.2_17452->177.1.254.1_19398[0.0.0.0]:
> Successfully parsed Resv message from 127.0.0.1 (on receiver host)
> Feb 10 03:47:39.587: RSVP-MSG:
> 177.1.254.2_17452->177.1.254.1_19398[0.0.0.0]: no matching path state for
> Resv
> Feb 10 03:47:39.595: RSVP: 177.1.254.1_19398->177.1.254.2_17452[0.0.0.0]:
> Received Path message from 127.0.0.1 (on sender host)
> Feb 10 03:47:39.595: RSVP: new path message passed parsing, continue...
> Feb 10 03:47:39.595: RSVP: Triggering outgoing Path due to incoming Path
> change or new Path
> Feb 10 03:47:39.595: RSVP: Triggering outgoing Path refresh
> Feb 10 03:47:39.599: RSVP: 177.1.254.1_19398->177.1.254.2_17452[0.0.0.0]:
> Path refresh, Event: rmsg not enabled or ack rcvd, State: trigger to normal
> Feb 10 03:47:39.599: RSVP: 177.1.254.1_19398->177.1.254.2_17452[0.0.0.0]:
> Path refresh (msec), config: 3 curr: 3 xmit: 3
> Feb 10 03:47:39.599: RSVP: Triggering outgoing Path due to incoming Path
> change or new Path
> Feb 10 03:47:39.599: RSVP: Triggering outgoing Path refresh
> Feb 10 03:47:39.603: RSVP: 177.1.254.1_19398->177.1.254.2_17452[0.0.0.0]:
> Path refresh, Event: rmsg not enabled or ack rcvd, State: trigger to normal
> Feb 10 03:47:39.603: RSVP: 177.1.254.1_19398->177.1.254.2_17452[0.0.0.0]:
> Path refresh (msec), config: 3 curr: 3 xmit: 3
> Feb 10 03:47:39.603: RSVP: 177.1.254.1_19398->177.1.254.2_17452[0.0.0.0]:
> Sending Path message to 177.1.254.2
> Feb 10 03:47:39.623: RSVP: 177.1.254.1_19398->177.1.254.2_17452[0.0.0.0]:
> building hop object with

Re: [OSL | CCIE_Voice] MVA problem

[datucha123 datucha123]

Check whats the region relation between HQ MTP RSPV and MoH. It should be
g729. Also same between BR1 MTP RSVP and HQ MTP RSVP

BR1 Phones and MTP are placed in BR1 Region, which has G711 inside and G729
outside to HQ Region and MoH Regions as well.
HQ Phones and MTP are placed in HQ Region, which has G711 inside and G711
to MoH, while G729 to BR1.

Well, I can see the HQ Region (where the HQ MTP is placed) has G711 towards
MoH Regions, but as I know that does not mean that only G711 can be
negotiated between MTP and MoH Server (everything below G711 could be). So
MTP at HQ must negotiate MoH G729.



On Fri, Feb 10, 2012 at 6:43 AM, Mohd Baqari wrote:

>  Check whats the region relation between HQ MTP RSPV and MoH. It should
> be g729. Also same between BR1 MTP RSVP and HQ MTP RSVP.
>
> If this is done then MoH should negotiate g729 with BR1.
>
> Regards,
> Mohammed Al Baqari
>
> Sent from my iPhone
>
> On Feb 9, 2012, at 9:52 PM, datucha123 datucha123 
> wrote:
>
>  Here is what I get on HQ Router during the call hung up on the Mobile
> Phone:
>
> Feb 10 03:48:06.063: %ISDN-6-DISCONNECT: Interface Serial0/2/0:0
> disconnected from 206501 , call lasted 18 seconds
> Feb 10 03:48:06.187: RSVP-MSG:
> 177.1.254.2_18762->177.1.254.1_16464[0.0.0.0]: Received PathTear message
> from 177.1.254.2 (on Serial0/3/0.1)
> Feb 10 03:48:06.187: RSVP: 177.1.254.2_18762->177.1.254.1_16464[0.0.0.0]:
> Expiring Serial0/3/0.1 PATH state, reason: PathTear arrival
> Feb 10 03:48:06.187: RSVP: 177.1.254.2_18762->177.1.254.1_16464[0.0.0.0]:
> Expiring  RESV state, reason: PathTear arrival
> Feb 10 03:48:06.187: RSVP: 177.1.254.2_18762->177.1.254.1_16464[0.0.0.0]:
> Expiring receiver host RESV state, reason: PathTear arrival (17:18762)
> Feb 10 03:48:06.187: RSVP: 177.1.254.2_18762->177.1.254.1_16464[0.0.0.0]:
> Expiring Serial0/3/0.1 RESV request state, reason: PathTear arrival
> Feb 10 03:48:06.195: RSVP-MSG:
> 177.1.254.1_16464->177.1.254.2_18762[0.0.0.0]: Received ResvTear message
> from 177.0.101.2 (on Serial0/3/0.1)
> Feb 10 03:48:06.195: RSVP: 177.1.254.1_16464->177.1.254.2_18762[0.0.0.0]:
> Expiring Serial0/3/0.1 RESV state, reason: ResvTear arrival
> Feb 10 03:48:06.195: RSVP: 177.1.254.1_16464->177.1.254.2_18762[0.0.0.0]:
> Expiring Serial0/3/0.1 RESV state, reason: ResvTear arrival (17:16464)
> Feb 10 03:48:06.195: RSVP: 177.1.254.1_16464->177.1.254.2_18762[0.0.0.0]:
> Expiring sender host PATH state, reason: Local application requested tear
> Feb 10 03:48:06.195: RSVP: 177.1.254.1_16464->177.1.254.2_18762[0.0.0.0]:
> building hop object with src addr: 177.0.101.1
> Feb 10 03:48:06.215: RSVP: 177.1.254.1_16464->177.1.254.2_18762[0.0.0.0]:
> Sending PathTear message to 177.1.254.2
>
>
> HQ Interface:
>
> interface Serial0/3/0.1 point-to-point
>  description == FR To BR1 (R2)
>  ip address 177.0.101.1 255.255.255.0
>  snmp trap link-status
>  frame-relay interface-dlci 101
>  ip rsvp bandwidth 500
> end
>
> BR1 Interface:
>
> interface Serial0/3/0.1 point-to-point
>  description == FR To HQ (R1)
>  ip address 177.0.101.2 255.255.255.0
>  snmp trap link-status
>  frame-relay interface-dlci 101
>  ip rsvp bandwidth 500
> end
>
> So as I guess, reservation is not taking place after the Mobile Phone has
> hung up.
>
> Also here is the output before the HQ Desk Phone has sent the call to
> Mobile Phone:
>
> Feb 10 03:47:39.587: RSVP: session 177.1.254.1_19398[0.0.0.0]: Received
> Resv message from 127.0.0.1 (on receiver host)
> Feb 10 03:47:39.587: RSVP: 177.1.254.2_17452->177.1.254.1_19398[0.0.0.0]:
> Successfully parsed Resv message from 127.0.0.1 (on receiver host)
> Feb 10 03:47:39.587: RSVP-MSG:
> 177.1.254.2_17452->177.1.254.1_19398[0.0.0.0]: no matching path state for
> Resv
> Feb 10 03:47:39.595: RSVP: 177.1.254.1_19398->177.1.254.2_17452[0.0.0.0]:
> Received Path message from 127.0.0.1 (on sender host)
> Feb 10 03:47:39.595: RSVP: new path message passed parsing, continue...
> Feb 10 03:47:39.595: RSVP: Triggering outgoing Path due to incoming Path
> change or new Path
> Feb 10 03:47:39.595: RSVP: Triggering outgoing Path refresh
> Feb 10 03:47:39.599: RSVP: 177.1.254.1_19398->177.1.254.2_17452[0.0.0.0]:
> Path refresh, Event: rmsg not enabled or ack rcvd, State: trigger to normal
> Feb 10 03:47:39.599: RSVP: 177.1.254.1_19398->177.1.254.2_17452[0.0.0.0]:
> Path refresh (msec), config: 3 curr: 3 xmit: 3
> Feb 10 03:47:39.599: RSVP: Triggering outgoing Path due to incoming Path
> change or new Path
> Feb 10 03:47:39.599: RSVP: Triggering outgoing Path refresh
> Feb 10 03:47:39.603: RSVP: 177.1.254.1_19398->177.1.254.2_17452[0.0.0.0]:
> Path refresh, Event: rmsg not enabled or ack r

Re: [OSL | CCIE_Voice] MVA problem

[datucha123 datucha123]

The issue nas been with RSVP CAC, and as soon as I have remove the CAC, the
MoH started to work.

Q1: Can you confirm that when you answer the cell phone (working scenario)
which gateway is being used to get to PSTN
A1:  The BR1 gateway was used.

Q2: Can you confirm that when you answer the deskphone (not working
scenario) and transfer to cell phone which gateway is being used to get to
PSTN.
A2:  HQ Gateway was used.


On Fri, Feb 10, 2012 at 5:25 AM, Vik Malhi  wrote:

>  I found the problem, but another one has arise.
>
> Please inform us of the problem/solution for the benefit of others
> following this thread.
>
>
> Q1: Can you confirm that when you answer the cell phone (working scenario)
> which gateway is being used to get to PSTN.
>
> Q2: Can you confirm that when you answer the deskphone (not working
> scenario) and transfer to cell phone which gateway is being used to get to
> PSTN.
>
> I suspect there is different gateway being used due to you using SLRG
> which could explain the differences.
>
>
>
>  Vik Malhi – CCIE #13890
> Managing Partner - IPexpert, Inc.
>
> Telephone: +1.810.326.1444 ext 420
> Fax: +1.810.454.0130
> Mailto: vma...@ipexpert.com
>
>
>
>
>   On Feb 9, 2012, at 9:24 AM, datucha123 datucha123 wrote:
>
>   I found the problem, but another one has arise.
>
> I am using RSVP as CAC between site, where the RSVP MTP has only G729
> codec enabled. And also the MoH is using Unicast so that it is subject to
> CAC.
>
> So now when the call is picked up by Mobile Phone and then dropped, the
> MoH plays good, as the BR1 site is using G729 to MoH Server. But when the
> HQ Desk Phone picks up the call first and then redirects to Mobile phone,
> after hung up the MoH does not work as the BR1 phone (somehow, why it makes
> so I do not know) is trying to use G711 to MoH Server, where the RSVP MTP
> does not pass the G711 traffic and that is why the ToH is heard.
>
> Now when I have changed the BR1 to use the Local Flash MMoH, everything
> was working fine (MoH was heard always).
>
> But still no idea, why the BR1 is trying to negotiate G711 to MoH server
> after the Mobile has dropped the call (when the Desk had sent the call to
> Mobile).
>
> Also I can see that when the Mobile Phone picks up the call first, the BR1
> Phone is using G711 (call goes through SLRG BR1 local gateway).
> But when the HQ Phone picks up the call first, and then sends the call to
> Mobile, the BR1 phone shows that it is using G729, thus the call is going
> though local Gateway.
>
>
>
>
> On Thu, Feb 9, 2012 at 2:37 AM, Mohammed Al Baqari <
> baqari.voic...@gmail.com> wrote:
>
>>  Hi Datucha,
>>
>> ** **
>>
>> Now when the BR1 phone calls this HQ phone, so that the mobile phone also
>> ring, and when the Mobile Phone picks up the call first, then the MoH works
>> when the mobile is hunged up, on the BR1 phone.
>>
>> But if the HQ phone has picked up the call first and then made the Send
>> Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and
>> instead the ToH is heard when the mobile Phone disconnects. 
>>
>> I suggest to set two different network moh source files. Assign one to
>> RDP and one to HQ Phone. Then repeat your test scenarios above and lets
>> know which MoH file is used in each case.
>>
>> ** **
>>
>> Also, please share the traces regarding the codec part.
>>
>> ** **
>>
>> ** **
>>
>> Regards,
>>
>> Mohammed Al Baqari
>>
>> ** **
>>
>> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
>> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *datucha123
>> datucha123
>> *Sent:* Wednesday, February 08, 2012 11:28 PM
>> *To:* ccie_voice@onlinestudylist.com
>> *Subject:* [OSL | CCIE_Voice] MVA problem
>>
>> ** **
>>
>> I have the following kind of probem:
>>
>> I am using SLRG for Mobile Connect calls, so that that calls to users
>> mobiles are done through local gateway (this is just for test).
>>
>> Now, the HQ phone has the RDP assinged with RD of his mobile phone.
>>
>> Now when the BR1 phone calls this HQ phone, so that the mobile phone also
>> ring, and when the Mobile Phone picks up the call first, then the MoH works
>> when the mobile is hunged up, on the BR1 phone.
>>
>> But if the HQ phone has picked up the call first and then made the Send
>> Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and
>> instead the ToH is heard when the mobile Phone disconnects. 
>>
>>  
>

Re: [OSL | CCIE_Voice] OT - UCM 7x bugs?

[datucha123 datucha123]

That is the Browser issue.



On Fri, Feb 10, 2012 at 5:27 AM, Jeferson Guardia wrote:

> Hi,
>
> Often when doing labs over ucm 7, sometimes I notice specially when
> messing around with CSS/partitions, if you open multiple windows and for
> example:
>
> You have 5 firefox tabs open
>
> CSS-site-A
> css-B
> css-C
> css-D
> css-E
>
> And then you go window by window applying it and saving it, all at once..
> I usually find partitions missing later on that CSS previously assigned. I
> care to check for dbreplication and issues like that, all running fine. is
> this a bug or something? it's terrible to have to go back at something you
> sure you've done correctly it just to fix a bug!
>
> --
> Jeferson Guardia
> CCIE #28157
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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

Re: [OSL | CCIE_Voice] MVA problem

[datucha123 datucha123]

erial0/3/0.1)
Feb 10 03:47:48.703: RSVP: 177.1.254.1_16464->177.1.254.2_18762[0.0.0.0]:
Successfully parsed Resv message from 177.0.101.2 (on Serial0/3/0.1)
Feb 10 03:47:48.703: RSVP: 177.1.254.1_16464->177.1.254.2_18762[0.0.0.0]:
this RESV has a confirm object
Feb 10 03:47:48.703: RSVP: 177.1.254.1_16464->177.1.254.2_18762[0.0.0.0]:
Resv changed: POLICY_DATA, FLOWSPEC,
Feb 10 03:47:48.703: RSVP: 177.1.254.1_16464->177.1.254.2_18762[0.0.0.0]:
process_reservation_change: Resv change requires triggering of Resv upstream
R1#
Feb 10 03:47:48.703: RSVP-RESV: accept_reservation_change: 4A374C78
Feb 10 03:47:48.703: RSVP-RESV: reservation was installed: 4A374C78
Feb 10 03:47:48.707: RSVP: 177.1.254.1_16464->177.1.254.2_18762[0.0.0.0]:
building error_spec object with err-node addr: 177.0.101.1
Feb 10 03:47:48.727: RSVP: 177.1.254.1_16464->177.1.254.2_18762[0.0.0.0]:
Sending ResvConf message to 177.0.101.2




On Thu, Feb 9, 2012 at 9:40 PM, Vik Malhi  wrote:

> Put the CAC back on and increase the "ip rsvp bandwidth" to a high value
> such as 500 (on both routers).
>
> Do a *debug ip rsvp signaling* on HQ and find out how much bandwidth is
> being requested when the problematic call is on hold.
>
> Also what Device Pool is the MOH server in? Place it inside the HQ Device
> Pool / HQ Region.
>
> Vik Malhi – CCIE #13890
> Managing Partner - IPexpert, Inc.
>
> Telephone: +1.810.326.1444 ext 420
> Fax: +1.810.454.0130
> Mailto: vma...@ipexpert.com
>
>
>
>
> On Feb 9, 2012, at 9:27 AM, datucha123 datucha123 wrote:
>
> Also as Vik had asked, I have removed the RSVP CAC, and the MoH begin to
> work and the G729 was negotiated.
>
> Well this is very strange behavior.
>
> While using CAC, BR1 is trying to negotiate G711 to MoH, and when not
> using RSVP CAC, G729 is negotiated.
>
> On Thu, Feb 9, 2012 at 9:16 PM, Vik Malhi  wrote:
>
>> The only difference being you have an existing call in the case of the RD
>> and the MOH would be the second call. Can you try removing Locations CAC
>> altogether to rule this out.
>>
>>
>>   Vik Malhi – CCIE #13890
>> Managing Partner - IPexpert, Inc.
>>
>> Telephone: +1.810.326.1444 ext 420
>> Fax: +1.810.454.0130
>> Mailto: vma...@ipexpert.com
>>
>>
>>
>>
>> On Feb 9, 2012, at 8:52 AM, datucha123 datucha123 wrote:
>>
>> G729 Codec is enable in IP Voice Media Streaming Application Service
>> Parameter. And when HQ Phone calls BR1 Phone and places BR1 on hold, the
>> MoH is played with G729 to BR1 phone without a problem.
>>
>>
>>
>> On Thu, Feb 9, 2012 at 8:31 PM, Vik Malhi  wrote:
>>
>>> My guess is MOH is not working when BR1 phones is placed on hold- and
>>> this has nothing to do with Mobile Connect.
>>>
>>> Enable G729 in the IP Voice Media Streaming Application Service
>>> Parameter.
>>>
>>>   Vik Malhi – CCIE #13890
>>> Managing Partner - IPexpert, Inc.
>>>
>>> Telephone: +1.810.326.1444 ext 420
>>> Fax: +1.810.454.0130
>>> Mailto: vma...@ipexpert.com
>>>
>>>
>>>
>>>
>>> On Feb 8, 2012, at 2:37 PM, Mohammed Al Baqari wrote:
>>>
>>> Hi Datucha,
>>> ** **
>>>
>>> Now when the BR1 phone calls this HQ phone, so that the mobile phone
>>> also ring, and when the Mobile Phone picks up the call first, then the MoH
>>> works when the mobile is hunged up, on the BR1 phone.
>>>
>>> But if the HQ phone has picked up the call first and then made the Send
>>> Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and
>>> instead the ToH is heard when the mobile Phone disconnects.
>>> I suggest to set two different network moh source files. Assign one to
>>> RDP and one to HQ Phone. Then repeat your test scenarios above and lets
>>> know which MoH file is used in each case.
>>> ** **
>>> Also, please share the traces regarding the codec part.
>>> ** **
>>> ** **
>>> Regards,
>>> Mohammed Al Baqari
>>> ** **
>>> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
>>> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *datucha123
>>> datucha123
>>> *Sent:* Wednesday, February 08, 2012 11:28 PM
>>> *To:* ccie_voice@onlinestudylist.com
>>> *Subject:* [OSL | CCIE_Voice] MVA problem
>>> ** **
>>>
>>> I have the following kind of probem:
>>>
>>> I am using SLRG for Mobile Connect calls, so that that calls to users
>>> mobiles are done through 

Re: [OSL | CCIE_Voice] MVA problem

[datucha123 datucha123]

Also as Vik had asked, I have removed the RSVP CAC, and the MoH begin to
work and the G729 was negotiated.

Well this is very strange behavior.

While using CAC, BR1 is trying to negotiate G711 to MoH, and when not using
RSVP CAC, G729 is negotiated.

On Thu, Feb 9, 2012 at 9:16 PM, Vik Malhi  wrote:

> The only difference being you have an existing call in the case of the RD
> and the MOH would be the second call. Can you try removing Locations CAC
> altogether to rule this out.
>
>
> Vik Malhi – CCIE #13890
> Managing Partner - IPexpert, Inc.
>
> Telephone: +1.810.326.1444 ext 420
> Fax: +1.810.454.0130
> Mailto: vma...@ipexpert.com
>
>
>
>
> On Feb 9, 2012, at 8:52 AM, datucha123 datucha123 wrote:
>
> G729 Codec is enable in IP Voice Media Streaming Application Service
> Parameter. And when HQ Phone calls BR1 Phone and places BR1 on hold, the
> MoH is played with G729 to BR1 phone without a problem.
>
>
>
> On Thu, Feb 9, 2012 at 8:31 PM, Vik Malhi  wrote:
>
>> My guess is MOH is not working when BR1 phones is placed on hold- and
>> this has nothing to do with Mobile Connect.
>>
>> Enable G729 in the IP Voice Media Streaming Application Service Parameter.
>>
>>   Vik Malhi – CCIE #13890
>> Managing Partner - IPexpert, Inc.
>>
>> Telephone: +1.810.326.1444 ext 420
>> Fax: +1.810.454.0130
>> Mailto: vma...@ipexpert.com
>>
>>
>>
>>
>> On Feb 8, 2012, at 2:37 PM, Mohammed Al Baqari wrote:
>>
>> Hi Datucha,
>> ** **
>>
>> Now when the BR1 phone calls this HQ phone, so that the mobile phone also
>> ring, and when the Mobile Phone picks up the call first, then the MoH works
>> when the mobile is hunged up, on the BR1 phone.
>>
>> But if the HQ phone has picked up the call first and then made the Send
>> Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and
>> instead the ToH is heard when the mobile Phone disconnects.
>> I suggest to set two different network moh source files. Assign one to
>> RDP and one to HQ Phone. Then repeat your test scenarios above and lets
>> know which MoH file is used in each case.
>> ** **
>> Also, please share the traces regarding the codec part.
>> ** **
>> ** **
>> Regards,
>> Mohammed Al Baqari
>> ** **
>> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
>> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *datucha123
>> datucha123
>> *Sent:* Wednesday, February 08, 2012 11:28 PM
>> *To:* ccie_voice@onlinestudylist.com
>> *Subject:* [OSL | CCIE_Voice] MVA problem
>> ** **
>>
>> I have the following kind of probem:
>>
>> I am using SLRG for Mobile Connect calls, so that that calls to users
>> mobiles are done through local gateway (this is just for test).
>>
>> Now, the HQ phone has the RDP assinged with RD of his mobile phone.
>>
>> Now when the BR1 phone calls this HQ phone, so that the mobile phone also
>> ring, and when the Mobile Phone picks up the call first, then the MoH works
>> when the mobile is hunged up, on the BR1 phone.
>>
>> But if the HQ phone has picked up the call first and then made the Send
>> Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and
>> instead the ToH is heard when the mobile Phone disconnects.
>>
>>  
>>
>> Also I have noticed that when the Mobile Phone picks up the call faster
>> then the Desk Phone, the codec negotiated is g711 from BR1 phone to its
>> local gateway through which the call went out.
>>
>> But if the Desk Phone at HQ  picked up the call first, and then made Send
>> calls to mobile phone, the codec stays at G729 on BR1 phone, even though
>> the call is going out throuhg local BR1 gateway where it should use G711.
>> 
>> ___
>>
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com
>>
>>
>>
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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

Re: [OSL | CCIE_Voice] MVA problem

[datucha123 datucha123]

I found the problem, but another one has arise.

I am using RSVP as CAC between site, where the RSVP MTP has only G729 codec
enabled. And also the MoH is using Unicast so that it is subject to CAC.

So now when the call is picked up by Mobile Phone and then dropped, the MoH
plays good, as the BR1 site is using G729 to MoH Server. But when the HQ
Desk Phone picks up the call first and then redirects to Mobile phone,
after hung up the MoH does not work as the BR1 phone (somehow, why it makes
so I do not know) is trying to use G711 to MoH Server, where the RSVP MTP
does not pass the G711 traffic and that is why the ToH is heard.

Now when I have changed the BR1 to use the Local Flash MMoH, everything was
working fine (MoH was heard always).

But still no idea, why the BR1 is trying to negotiate G711 to MoH server
after the Mobile has dropped the call (when the Desk had sent the call to
Mobile).

Also I can see that when the Mobile Phone picks up the call first, the BR1
Phone is using G711 (call goes through SLRG BR1 local gateway).
But when the HQ Phone picks up the call first, and then sends the call to
Mobile, the BR1 phone shows that it is using G729, thus the call is going
though local Gateway.




On Thu, Feb 9, 2012 at 2:37 AM, Mohammed Al Baqari  wrote:

> Hi Datucha,
>
> ** **
>
> Now when the BR1 phone calls this HQ phone, so that the mobile phone also
> ring, and when the Mobile Phone picks up the call first, then the MoH works
> when the mobile is hunged up, on the BR1 phone.
>
> But if the HQ phone has picked up the call first and then made the Send
> Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and
> instead the ToH is heard when the mobile Phone disconnects. 
>
> I suggest to set two different network moh source files. Assign one to RDP
> and one to HQ Phone. Then repeat your test scenarios above and lets know
> which MoH file is used in each case.
>
> ** **
>
> Also, please share the traces regarding the codec part.
>
> ** **
>
> ** **
>
> Regards,
>
> Mohammed Al Baqari
>
> ** **
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *datucha123
> datucha123
> *Sent:* Wednesday, February 08, 2012 11:28 PM
> *To:* ccie_voice@onlinestudylist.com
> *Subject:* [OSL | CCIE_Voice] MVA problem
>
> ** **
>
> I have the following kind of probem:
>
> I am using SLRG for Mobile Connect calls, so that that calls to users
> mobiles are done through local gateway (this is just for test).
>
> Now, the HQ phone has the RDP assinged with RD of his mobile phone.
>
> Now when the BR1 phone calls this HQ phone, so that the mobile phone also
> ring, and when the Mobile Phone picks up the call first, then the MoH works
> when the mobile is hunged up, on the BR1 phone.
>
> But if the HQ phone has picked up the call first and then made the Send
> Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and
> instead the ToH is heard when the mobile Phone disconnects. 
>
>  
>
> Also I have noticed that when the Mobile Phone picks up the call faster
> then the Desk Phone, the codec negotiated is g711 from BR1 phone to its
> local gateway through which the call went out.
>
> But if the Desk Phone at HQ  picked up the call first, and then made Send
> calls to mobile phone, the codec stays at G729 on BR1 phone, even though
> the call is going out throuhg local BR1 gateway where it should use G711.*
> ***
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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

Re: [OSL | CCIE_Voice] MVA problem

[datucha123 datucha123]

G729 Codec is enable in IP Voice Media Streaming Application Service
Parameter. And when HQ Phone calls BR1 Phone and places BR1 on hold, the
MoH is played with G729 to BR1 phone without a problem.



On Thu, Feb 9, 2012 at 8:31 PM, Vik Malhi  wrote:

> My guess is MOH is not working when BR1 phones is placed on hold- and this
> has nothing to do with Mobile Connect.
>
> Enable G729 in the IP Voice Media Streaming Application Service Parameter.
>
> Vik Malhi – CCIE #13890
> Managing Partner - IPexpert, Inc.
>
> Telephone: +1.810.326.1444 ext 420
> Fax: +1.810.454.0130
> Mailto: vma...@ipexpert.com
>
>
>
>
> On Feb 8, 2012, at 2:37 PM, Mohammed Al Baqari wrote:
>
> Hi Datucha,
> ** **
>
> Now when the BR1 phone calls this HQ phone, so that the mobile phone also
> ring, and when the Mobile Phone picks up the call first, then the MoH works
> when the mobile is hunged up, on the BR1 phone.
>
> But if the HQ phone has picked up the call first and then made the Send
> Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and
> instead the ToH is heard when the mobile Phone disconnects.
> I suggest to set two different network moh source files. Assign one to RDP
> and one to HQ Phone. Then repeat your test scenarios above and lets know
> which MoH file is used in each case.
> ** **
> Also, please share the traces regarding the codec part.
> ** **
> ** **
> Regards,
> Mohammed Al Baqari
> ** **
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *datucha123
> datucha123
> *Sent:* Wednesday, February 08, 2012 11:28 PM
> *To:* ccie_voice@onlinestudylist.com
> *Subject:* [OSL | CCIE_Voice] MVA problem
> ** **
>
> I have the following kind of probem:
>
> I am using SLRG for Mobile Connect calls, so that that calls to users
> mobiles are done through local gateway (this is just for test).
>
> Now, the HQ phone has the RDP assinged with RD of his mobile phone.
>
> Now when the BR1 phone calls this HQ phone, so that the mobile phone also
> ring, and when the Mobile Phone picks up the call first, then the MoH works
> when the mobile is hunged up, on the BR1 phone.
>
> But if the HQ phone has picked up the call first and then made the Send
> Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and
> instead the ToH is heard when the mobile Phone disconnects.
>
>  
>
> Also I have noticed that when the Mobile Phone picks up the call faster
> then the Desk Phone, the codec negotiated is g711 from BR1 phone to its
> local gateway through which the call went out.
>
> But if the Desk Phone at HQ  picked up the call first, and then made Send
> calls to mobile phone, the codec stays at G729 on BR1 phone, even though
> the call is going out throuhg local BR1 gateway where it should use G711.*
> ***
> ___
>
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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

[OSL | CCIE_Voice] MVA problem

[datucha123 datucha123]

I have the following kind of probem:

I am using SLRG for Mobile Connect calls, so that that calls to users
mobiles are done through local gateway (this is just for test).

Now, the HQ phone has the RDP assinged with RD of his mobile phone.

Now when the BR1 phone calls this HQ phone, so that the mobile phone also
ring, and when the Mobile Phone picks up the call first, then the MoH works
when the mobile is hunged up, on the BR1 phone.

But if the HQ phone has picked up the call first and then made the Send
Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and
instead the ToH is heard when the mobile Phone disconnects.



Also I have noticed that when the Mobile Phone picks up the call faster
then the Desk Phone, the codec negotiated is g711 from BR1 phone to its
local gateway through which the call went out.

But if the Desk Phone at HQ  picked up the call first, and then made Send
calls to mobile phone, the codec stays at G729 on BR1 phone, even though
the call is going out throuhg local BR1 gateway where it should use G711.
___
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www.ipexpert.com

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www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] MGCP troubleshooting

[datucha123 datucha123]

debug mgcp packets

Trace in CUCM

On Wed, Feb 8, 2012 at 3:35 PM, Jeferson Guardia wrote:

> Hi,
>
> What are the techniques most used to perform MGCP troubleshooting?
>
> Yesterday I was doing a lab and had a router with pstn integration, it was
> set for a CSS where my phones had visibility and significant digits = 4.
> But whenever I would call out from the PSTN, I would get a second dial
> tone, I would see the call kicking in thru debug isdn q931 but my phone
> would simply not ring. Any ideas how to verify that possible behavior ?
> Articles? tech guides?
>
> Thanks,
>
> --
> Jeferson Guardia
> CCIE #28157
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
___
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www.ipexpert.com

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Re: [OSL | CCIE_Voice] OUTVIA & INVIA

[datucha123 datucha123]

Outvia is more accurate.

Invia, in most cases, is used for incoming LRQs.

On Tue, Feb 7, 2012 at 11:13 PM, mercy forall wrote:

>  Hi All
>
> now in outvia and invia ,,
>
> Are is it deference if i use it in local zone or remote zone ?
>
> As per Doc, outvia for any traffic leave this zone , so are this same if i
> use outvia in local or remote zone
>
>
> I  need to send the call form local zone to remote zone through CUBE as
> local zone ,
>
> what is the correct  [zone remote with outvia OR with invia CUBE ] ?
>
> thanks
> **
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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

Re: [OSL | CCIE_Voice] QOS LFI and BACD files

[datucha123 datucha123]

But that won't go for HQ router, as it is a Hub Router  -- multiple PVCs
per physical interfaces via subinterfaces.

For instance, if you have a T1 frame relay link, and already shaped one PVC
to 500kbps, you cannot shape another PVC to 1536. You have to shape another
PVC to 1536 - 500 = 1036 kbps, as the full interface rate is 1536, while
one PVC is already shaped up to 500, so only 1036 is left for shaping.

On Tue, Feb 7, 2012 at 10:02 PM, Vik Malhi  wrote:

> I agree with the last post.
>
> When you have used FRTS use the command "show traff" to verify Interval
> time and target rate- default to 1536 if PVC speed is not given to you. The
> snippet below shows what happens when FRTS is enabled- both these PVC's
> will need fixing.
>
>
> SiteA-RTR(config)#*interface Serial0/0/1:0*
> SiteA-RTR(config-if)#*frame-relay traff*
> SiteA-RTR#*sh traff*
>
> Interface   Se0/0/1:0.1
>Access TargetByte   Sustain   ExcessInterval  Increment
> Adapt
> VC List   Rate  Limit  bits/int  bits/int  (ms)  (bytes)
> Active
> 201   *56000* 8757000  0 *125*   875
>   -
>
> Interface   Se0/0/1:0.2
>Access TargetByte   Sustain   ExcessInterval  Increment
> Adapt
> VC List   Rate  Limit  bits/int  bits/int  (ms)  (bytes)
> Active
> 202   *56000* 8757000  0 *125*   875
>   -
>
> Vik Malhi – CCIE #13890
> Managing Partner - IPexpert, Inc.
>
> Telephone: +1.810.326.1444 ext 420
> Fax: +1.810.454.0130
> Mailto: vma...@ipexpert.com
>
>
>
>
> On Feb 7, 2012, at 7:58 AM, CCIEVoiceKP wrote:
>
> I personally would set it to a full T1 ... Bandwidth 1536  When in
> doubt, explain what and why to the proctor to make sure it's ok.
>
> KP
>
> Sent from my iPhone and I have big thumbs ... So please excuse the typos.
>
> On Feb 6, 2012, at 9:02 PM, AJ BG  wrote:
>
>  Hello,
>1.   QOS question
> According to Vic, if you configure LFI for a subinterface in a hub and
> spoke environment, Your second sub interface will dopes its CIR to 56k. To
> solve this issue you should configure map-class for the second interface as
> well. I have tested this and confirmed the problem and the solution.
>  But if the interface bandwidth is not given to you, then in what rate do
> you configure the second map-class? What should be your CIR and MinCIR
> bandwidth?
>
>
> 2.   BACD question
>
> will it be possible that the lab requirement will be  to configure BACD
> without giving you direct access to the BACD files? If the above scenario
> happen then how would you copy the files into the router. I am thinking to
> use CUCM. But can you even go to Cisco’s website and download BACD tar file
> during the exam? Any suggestion?
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
___
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www.ipexpert.com

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Re: [OSL | CCIE_Voice] Lab Grading

[datucha123 datucha123]

You are right Emanuel.

I am a native neglish speaker, and sometimes it is hard for me to
understand some task, while rereading them several times

On Tue, Feb 7, 2012 at 8:12 PM, Emanuel Damasceno wrote:

> Hey Vega,
>
> Sorry for hearing you didn't make it. I've been there and I know the
> feeling. I am preparing myself for my second attempt now.
>
> I don't know which strategies you used, but all areas you mentioned there
> are tricks that confuse us. I think the whole point of the exam is
> understanding and doing its requirements. I have Workbooks 1 and 2 of
> IPExpert and the whole "she bang" from INE. I speak fluent English, but I
> am aware that it is not my first language. Sometimes I read a question and
> I understand it some way, when I ask my wife to read it she understands it
> another way, and she explain to me what the Workbooks are actually asking
> of me. Since she is American born and raised, she is also my resource as a
> better understanding simple plain English. I am not trying to say you know
> or don't know the language, it's not that. What I am judging by your last
> name you are not an English native speaker (I might be wrong), but if
> English is not your native language, you may have an understanding that is
> different from what it is required. Thus I mentioned my example, because
> that is my case. If you are an English native speaker, nevermind what I
> just said. But I think that even re-reading it, makes better sense.
>
> In the exam, we don't have time to read and re-read questions until we
> understand. You are not in the exam anymore, but take a question from one
> of your Workbook studies, and pick one you didn't understand it fully. You
> can easily pick that question as you read through your workbook and you
> read a question that you stop and say to yourself "Wait, what is it this
> question is saying?". You automatically re-read it for better
> understanding. As you read it twice or thrice, you realize that you are
> understanding it quite differently from the first time you read it.
>
> My wife is a great support, not only for leaving me alone to study but
> also to help me understand better the English on the workbooks. The exam is
> all about being literal, over configuring won't do you any good. The
> proctor won't think you're an expert because you know a command that only
> few people know it. If the question is specifically saying to do something,
> you need to do it like they ask. If they don't specify it, you can do it
> the way you know. I guess you already know all of this, but I am posting
> this mostly for the CCIE newcomers. But it might work for you too, if you
> don't already know it :)
>
> Best regards to all.
> *Emanuel Damasceno*
> CCNP Voice
>
>
>
>
>
> On Tue, Feb 7, 2012 at 4:33 AM, Vega Wong  wrote:
>
>> Hi all
>>
>> First of all, let me clarify that I am not trying to break NDA or trying
>> to sell anything here, just something I cant get my head around with.
>>
>> I just had my second attmpt yesterday, and I failed. What really
>> demoralising for me is that I thought I was fully prepared. What even more
>> demoralising is that when I looked at the score card, I didnt get 100% in
>> the area I thought I would get a 100%.
>>
>> I dont mean to say I know it all (especially now), but there are areas on
>> my score card that really shocks me. As I read through the exam, I know
>> what the questions were requiring (at least I thought I did). For example,
>> we all know that the topics of DHCP, NTP, VLAN will be in the network
>> infrastructure area. And what could go wrong in those areas? I mean, if you
>> dont setup those area correctly, all your subsequence config will has
>> problem right? But yet, I dont get full marks in those. Similarly the
>> gateway area, QoS area etc.
>>
>> Right now I am pretty lost as to what I can prepare or study on, or at
>> least how to check my config? I know this sounds really bad (or arrogent)
>> but I really have no idea as how to confirm the config I did can get the
>> full mark in those area.
>>
>> I guess I am hoping if someone that share their strategy as to how to
>> confirm their work is good or fulfilled the requirement?
>>
>> hope that makes sense?
>>
>>
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com
>>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] Standard Local RG troubleshooting: RouteGroup :RouteGroup Name= Standard Local Route Group

[datucha123 datucha123]

"stop routing on unallocated number flag"  -  in this particular case, this
parameter has nothing to do with the actual problem. This parameter defines
the rerouting option as William has already mentioned.

Ricardo, try to set the "Digit Analysis Complexity" to "Translation and
Alternate Pattern Analysis". And try to look for CUCM Traces, not the DNA.

On Tue, Feb 7, 2012 at 6:37 AM, William Bell  wrote:

> Ricardo,
>
> IIRC, the "stop routing on unallocated number flag" was actually first
> introduced for ICT call flows. However, it can be applied to other call
> flows. In normal call handling, when the CUCM receives a notification that
> a call failed to complete due to unallocated number it will stop routing
> the call. When you flag this service param to false, CUCM will try the next
> trunk or gateway in the route list/route group.
>
> -Bill
>
>  On Feb 6, 2012, at 8:21 PM, Ricardo Palaver wrote:
>
>  Hi Emanuel !.
>
> No , it does not work .. As far as I know, this is for use AAR, or Am I
> wrong?
>
> Thanks  !
>
>  --
> Date: Mon, 6 Feb 2012 23:01:14 -0200
> Subject: Re: [OSL | CCIE_Voice] Standard Local RG troubleshooting:
> RouteGroup :RouteGroup Name= Standard Local Route Group
> From: aedamasc...@gmail.com
> To: ricardo.pala...@hotmail.com
> CC: ccie_voice@onlinestudylist.com
>
> Hello Ricardo,
>
> You need to go to Service Parameters > Call Manager, and set the option "*Stop
> Routing on Unallocated Number Flag*" to FALSE
>
> I hope this helps :)
> *Emanuel Damasceno*
> CCNP Voice
>
>
>
>
>
> On Mon, Feb 6, 2012 at 10:05 PM, Ricardo Palaver <
> ricardo.pala...@hotmail.com> wrote:
>
>  Hi Folks,
>
> I am facing a problem with Standard local route group ...,  it is not
> working and I have no idea where could I troubleshoot it.  I configured as
> usual ... ,
> RL -> Standard Local RG
> In each DP, I pointed to the respective gateway (using Local  Route Group
> param) and of course each phone with the respective device.
>
> I tried  by using DNA, but it does not go further , the last point I see
> is  "RouteGroup :RouteGroup Name= Standard Local Route Group "..., As far
> as I know there is nothing in the service param to enable ... or there are
> something ?
>
>
> Thanks all !
>
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
>
> ___
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> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
>
>  *
> *
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
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Re: [OSL | CCIE_Voice] Timezone for UCM

[datucha123 datucha123]

That is for production installtion, but what about the LAB Exam?

On Mon, Feb 6, 2012 at 10:47 PM, William Bell  wrote:

>  As noted by Ken, you typically set the timezone during the installation
> process. If you need to adjust the timezone post-install then you can do so
> from the cli:
>
>  admin:set timezone ?
> Syntax:
> set timezone zone
> zone mandatory   This is the new time zone. Enter the appropriate
> string
>  or zone index id to uniquely identify the timezone.
>  A list of valid time zones can be obtained via the
>  following CLI command: show timezone list.
>
>
> Regards,
> Bill
>
>  On Feb 6, 2012, at 11:20 AM, datucha123 datucha123 wrote:
>
>  Hello,
>
> If we are told to synchronize CUCM Pub server with some NTP, do we need to
> set the correct Timezone for CUCM OS as well? Or just Date/Time Groups for
> Phones are enough?
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com <http://www.platinumplacement.com/>
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com <http://www.platinumplacement.com/>
>
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Re: [OSL | CCIE_Voice] Locations CAC

[datucha123 datucha123]

First of all check if the IP Phones are put in correct Locations.

Then for AAR to work, you need to create an AAR Group and assing it to
endpoints.

On Sun, Feb 5, 2012 at 11:03 PM, Emanuel Damasceno wrote:

> Hello Experts,
>
> I am trying to set up an AAR scenario for my studies. I configured 2
> Locations, with unlimited bandwidth, but mandatory RSVP from HQ to BR2. I
> wanna use 5 concurrent calls, and I am also using g729 between sites. I
> added the MTP-HQ, and MTP-BR2 to CUCM, put them in a MRG, followed by MRGL,
> and referenced it in its respective device pool. Reset all the phones.
>
> So here is my config:
> HQ
> dspfarm profile 2 mtp
>  codec g729r8
>  rsvp
>  maximum sessions software 5
>  associate application SCCP
>
> interface Serial0/0/1:0.2 point-to-point
>  description TO BR2
>  bandwidth 768
>  ip address 10.10.112.1 255.255.255.0
>  ip ospf mtu-ignore
>  snmp trap link-status
>  frame-relay interface-dlci 202
>   class AutoQoS-FR-Se0/0/1:0-202
>   auto qos voip trust
>  frame-relay ip rtp header-compression
>  ip rsvp bandwidth 136
>
> BR2
> dspfarm profile 1 mtp
>  codec g729r8
>  rsvp
>  maximum sessions software 5
>  associate application SCCP
>
> interface Serial0/1/0:0.1 point-to-point
>  description to HQ
>  bandwidth 768
>  ip address 10.10.112.2 255.255.255.0
>  ip ospf mtu-ignore
>  snmp trap link-status
>  frame-relay interface-dlci 102 CISCO
>   class AutoQoS-FR-Se0/1/0:0-102
>   auto qos voip
>  frame-relay ip rtp header-compression
>  ip rsvp bandwidth 136
>
> The main problem is that on the FIRST call it already says "Not Enough
> Bandwidth", wasn't that supposed to happen if the 6th caller tried to make
> a call? I already set to TRUE in Service Parameters for Automated Alternate
> Routing, but it's not showing the "Not Enough Bandwidh, Rerouting" message.
> I haven't configured my Partitions and CSSs yet, but what's up with the
> first call not going through? Am I missing something?
>
> *Emanuel Damasceno*
> CCNP Voice
>
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
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Re: [OSL | CCIE_Voice] CFUR does not work

[datucha123 datucha123]

No. it is supported. The destination Phone will just ring a bit later
through PSTN.

On Sun, Feb 5, 2012 at 3:16 AM, John McGaughey (jomcgaug) <
jomcg...@cisco.com> wrote:

>  Hi Vik/All
>
> ** **
>
> I’m working on Lab #4 of the new 5 labs.  Quesiton 9.2.  They are asking
> you to configure CFUR on SiteB phone 2.  However this will not work because
> of the RDP assigned to the phone.
>
> ** **
>
> RDP and CFUR and not supported together.  See CSCtg43998.
>
> ** **
>
> John
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
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Re: [OSL | CCIE_Voice] CUE Live Reply

[datucha123 datucha123]

You have press 44 (twise digit 4) and it will dial out.

On Sun, Feb 5, 2012 at 1:50 AM, John McGaughey (jomcgaug) <
jomcg...@cisco.com> wrote:

>  Hello,
>
> ** **
>
> I’m in lab 4 of the new 5 labs.  Question 6.4.  It’s asking to configure
> Live Reply.  The DSG says to just click the check box “Enable Live Reply.”
> I’ve done so but when I press option 4 it says “this message cannot get a
> reply.”  I’m calling from the SiteC phone 1.
>
> ** **
>
> I added a user and mailbox for SiteC phone 1.  Now when I press option 4 I
> get an option to record a message for SiteC phone.  I cannot figure out how
> to get it to dial the caller back.
>
> ** **
>
> I’ve googled and found nothing.  Any ideas?
>
> ** **
>
> John
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
___
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Re: [OSL | CCIE_Voice] New lab 5 CUCM/CUC SIP Integration

[datucha123 datucha123]

I have the same issue in my own LAB, and as soon as I restart my
CUC server, the MWI and Message start to work from PSTN for a while. but
then again stops.

And I make restart of CUC server every time.

Thus I was using SCCP integration.

On Sun, Feb 5, 2012 at 1:00 AM, Edgar Feliz  wrote:

> Also another issue I had was that it seemed like when I was leaving a VM
> and press # it was not recognizing that from any phone other then SA.
>
> Had most of the lab working except for the SIP/CUC.
>
> Thanks,
>
> Edgar
>
>  On Sat, Feb 4, 2012 at 3:15 PM, Vik Malhi  wrote:
>
>>  Can you successfully leave SAP2 a new VM from any phone ? Another SA
>> phone or PSTN or SB?
>>
>>
>>
>> --
>> Vik Malhi – CCIE #13890
>> Managing Partner / Instructor - IPexpert, Inc.
>> Mailto: vma...@ipexpert.com
>> Telephone: +1.810.326.1444
>> Fax: +1.810.454.0130
>> Live Assistance, Please visit: www.ipexpert.com/chat
>> 
>>
>> IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, 
>> Audio
>> Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE
>> (R&S, Voice, Wireless, Security & Service Provider) certification(s) with 
>> training
>> locations throughout the United States, Europe, South Asia and Australia.
>> Be sure to visit our online communities at www.ipexpert.com/communities <
>> http://www.ipexpert.com/communities>  and our public website at
>> www.ipexpert.com 
>>
>>
>> On Feb 4, 2012, at 12:08 PM, William Affeldt 
>> wrote:
>>
>>   So I am good then?
>>
>> Sent from my iPhone
>>
>> On Feb 4, 2012, at 11:59 AM, Vik Malhi  wrote:
>>
>>   Edgar- make sure that you do not have one way cRTP. Or there is any
>> MTP being used no sccp/sccp.
>>
>> Bill-  there is a CUC bug when you leave a VM  and can press # and hear
>> your message, but this message never gets sent to the mail box (from
>> specific ip addresses). In this case you have to just rely on VM/MWI from
>> an extension /gateway that does work.
>>
>>
>>
>> --
>> Vik Malhi – CCIE #13890
>> Managing Partner / Instructor - IPexpert, Inc.
>> Mailto: vma...@ipexpert.com
>> Telephone: +1.810.326.1444
>> Fax: +1.810.454.0130
>> Live Assistance, Please visit: 
>> www.ipexpert.com/chat
>> < http://www.ipexpert.com/chat>
>>
>> IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, 
>> Audio
>> Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE
>> (R&S, Voice, Wireless, Security & Service Provider) certification(s) with 
>> training
>> locations throughout the United States, Europe, South Asia and Australia.
>> Be sure to visit our online communities at 
>> 
>> www.ipexpert.com/communities < 
>> http://www.ipexpert.com/communities>  and our public website at
>> www.ipexpert.com < 
>> http://www.ipexpert.com/>
>>
>>
>> On Feb 4, 2012, at 11:42 AM, William Affeldt <
>> william.affe...@yahoo.com> wrote:
>>
>>   You are having one way audio issues then.
>>
>> Sent from my iPhone
>>
>> On Feb 4, 2012, at 11:27 AM, Edgar Feliz < 
>> ejzi...@gmail.com> wrote:
>>
>>  I don't hear the message when I press # for "more options" from SC or
>> PSTN nothing is happening but I am getting the options from SA/SB
>>
>> E
>>
>> On Sat, Feb 4, 2012 at 1:44 PM, William Affeldt 
>> <
>> william.affe...@yahoo.com> wrote:
>>
>>>  I am currently having the same problem. I have been troubleshooting
>>> for a hour now. It forwards to the correct VM box and you can even play
>>> the message back to your self after you record it. It just never makes it
>>> to the mailbox.
>>>
>>>   *From:* Edgar Feliz <  
>>> ejzi...@gmail.com>
>>> *To:*  
>>> ccie_voice@onlinestudylist.com
>>> *Sent:* Saturday, February 4, 2012 10:15 AM
>>> *Subject:* [OSL | CCIE_Voice] New lab 5 CUCM/CUC SIP Integration
>>>
>>> I am currently working on new lab 5 and I have my CUCM-CUC integration
>>> working, for voicemail left by SA > SB to SB phone and SB > SA Phone I
>>> get MWI both directions. But for PSTN or SC While I can leave a VM MWIdoes 
>>> not work and the
>>> VM does not show up when I check the inbox for either SA or SB phones
>>> for VM left from PSTN or SC. I have looked at the SIP trunk setting and
>>> do not see anything there CSS/PTs all look correct any Ideas?
>>>
>>> Thanks
>>>
>>> E
>>>
>>> ___
>>> For more information regarding industry leading CCIE Lab training,
>>> please visit  
>>> www.ipexpert.com
>>>
>>> Are you a CCNP or CCIE and looking for a job? Check out
>>>  
>>> www.PlatinumPlacement.com 
>>>
>>>
>>  ___
>> For more information regarding industry lead

Re: [OSL | CCIE_Voice] Reorder tone while + dialing from directories(missed, received)

[datucha123 datucha123]

You can also leave the Urgent Priority for Patterns, but use the SIP
Dial-rules for SIP phones, with \+! and interdigit timeout with 0

On Sun, Feb 5, 2012 at 12:53 AM, Ashwani  wrote:

> Thanks Vik.  Yes now I am seeing inter-digit timeout dialing from missed
> and received calls.  Appreciate your help and pointing me to the right
> direction.
>
> Ashwani
>
>
> On 2/4/2012 3:45 PM, Vik Malhi wrote:
>
>>  From SIP phones calls from the directory are sent digit by digit. This
>> is in contrast to sccp phones which send digits en bloc (as opposed to
>> digit by digit).
>>
>> A route pattern such as : \+! marked as urgent priority would cause a
>> call from the directory from a sip phone to fail. Since the plus would
>> match and since the urgent priority has been selected the call would get
>> sent to the gateway with just a "+" (which would be stripped in the case of
>> an h323 gateway).
>>
>> Remove and plus route patterns marked as urgent priority- if you want to
>> avoid inter digit timeout create route patterns without the "!" and define
>> the exact number of digits ("x's).
>>
>>
>>
>>
>  __**_
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
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>
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Re: [OSL | CCIE_Voice] Help: Configuring static & dynamic ip address assigning in same cisco IOS dhcp server

[datucha123 datucha123]

Yes, you can.

You have to use the DHCP Binding file (put it in flash), so that if the
Client identifier is found in the file, that it will give the static IP,
otherwise it will give the IP addres out of the pool.

Or you can use the DHCP Classes.

You can read about them in DHCP admin guide.

Here is the link:
http://www.cisco.com/en/US/docs/ios-xml/ios/ipaddr_dhcp/configuration/12-4t/dhcp-12-4t-book.pdf

Take a look at - *

Configuring the DHCP Server to Read a Static Mapping Text File
*

On Sat, Feb 4, 2012 at 11:00 AM, Rrcrumm  wrote:

> You need to configure one for the static and one for the rest of the
> dynamic addresses
>
> HTH
> Randall
>
> Sent from my iPhone
>
> On Feb 3, 2012, at 10:29 PM, Prafulla Rangari 
> wrote:
>
> > Hi  Everyone,
> >
> > Can any one tell us can we configure an IOS DHCP Pool which can provide
> static as well as  dynamic address to the clients
> >
> > Thanks
> > ___
> > For more information regarding industry leading CCIE Lab training,
> please visit www.ipexpert.com
> >
> > Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
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>
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Re: [OSL | CCIE_Voice] SLRG with CallForwarding

[datucha123 datucha123]

What are you trying to achieve?

Write it in more details, and we will try to help you

On Sat, Feb 4, 2012 at 7:19 AM, Seifeddine Tlili
wrote:

>  Hi Everyone
>
> ** **
>
> Is there a different workground then creating PT/CSS/RP for Callfowarding
> when using SLRG with Callforwarding on the line?
>
> ** **
>
> ** **
>
> ** **
>
> *Kindly*
>
> * *
>
> *Seifeddine Tlili*  
>
> [image: Description: Description: Long View Systems]
>
> M.Eng CCIE # 26440
> Systems Consultant 
>
> .. *
> Direct:* 403.387.3069 | *Mobile:* 403.973.4840 | *Main:* 403.515.6900
>
> [image: Description: Description: 
> Linkedin]
>  [image: Description: Description: 
> Twitter]
>  [image: Description: Description: 
> Facebook] [image:
> Description: Description: Facebook]
> www.longviewsystems.com
> This message and any attached documents are only for the use of
> the intended recipient(s), are confidential and may contain privileged
> information. Any unauthorized review, use, retransmission, or other
> disclosure is strictly prohibited. If you have received this message in
> error, notify the sender immediately, and delete the original message. ***
> *
>
> ** **
>
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>
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Re: [OSL | CCIE_Voice] load command

[datucha123 datucha123]

you want to say that we DO NOT NEED tftp on CME for our Phones on the LAB
Exam?

On Fri, Feb 3, 2012 at 10:45 PM, Vik Malhi  wrote:

> Correct.
>
> In addition you should be careful NOT to erase factory defaults on the CME
> phone too- this would mean you need the "load" command in order for it to
> boot up (phone would need to TFTP a ".loads" file.
>
>
>
>   On Feb 3, 2012, at 9:22 AM, Ken Wyan wrote:
>
>   For CCME , it's required to use load command as below.
>
> tftp-server flash:PHONES/SCCP.loads  alias  SCCP.loads
>
> telephony-service
> load 7965   SCCPx
>
> But in CCIE lab environment , this may consume lot of time for phone
> firmware upgrades.
>
> I think it's better not to put any of above commands in CCIE lab  unless
> any problem arises. (In Lab all phones should be already having v7
> compatible phone loads)
>
> Do you agree with me or not?
>
>
> ___
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> visit www.ipexpert.com
>
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>
>
>
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Re: [OSL | CCIE_Voice] Codec's, Compression and Bandwidth

[datucha123 datucha123]

You are wong my friend.

L3 codec bandwidth means  -  (Payload + L3) x PPS x 8
L2 Codec bandwidth means -  (Payload + L3 + L2) x PPS x 8

So here is an example for G711 and G729 for L3 BW calculations:

G711 20ms

Payload  -  160 Bytes
L3  -  40 bytes
PPS  - 50

(160 + 40) x 50 x 8 =  8 bps = 80kbps

G729 20ms

Payload  -  20 bytes
L3  -  40 Bytes
PPS - 50

(20 + 40) x 50 x 8 = 24000 bps = 24 kbps

Now copressed RTP

 G711 20ms

Payload  -  160 Bytes
L3  -  2 bytes
PPS  - 50

(160 + 2) x 50 x 8 =  64800 bps = 64.8kbps

 G729 20ms

Payload  -  20 bytes
L3  -  2 Bytes
PPS - 50

(20 + 2) x 50 x 8 = 8800 bps = 8,8 kbps


On Thu, Feb 2, 2012 at 2:51 AM, Nathan Silvers  wrote:

> Whoops started my own private convo with datucha... thought i would copy
> everyone else just to get more eyes on it and hopefully bring some
> clarification...
>
> From my understanding and using the Cisco doc the default g.711 packet
> size is 206bytes(including 2 - 10ms 80 byte voice samples = 160bytes or
> 1028bits)  and using crtp is 168bytes  using the calculations the pps is
> 64kbps/1028 bits = 50pps which using the same calculations of packet size
> is 1648 * 50pps = 82.4kbps total... and the same calculation using crtp
> same voice load so 50 pps bw calculation is total packet 1344bits * 50pps =
> 67.2kbps which is well below your layer 3 base of 80kbps and includes the
> same voice payload as the non-compressed packets at 82.4... the 64 and 8
> kbps are the base layer 3 bandwidth for just the audio, not including any
> headers...
>
> If anyone sees it differently please let me know, i would hate to walk
> into the test and have this whole concept wrong...
>
>
>  On Wed, Feb 1, 2012 at 2:05 PM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>>
>> G711 64kbps and G729 8kbps are not the L3 Bandwidth for those codecs.
>> These are the Payload Bandwidths for those codecs.
>>
>> G711 and G729 on L3 are using 80 and 24 kbps respectavely.
>>
>> Here is the easy way to calcualte the Codec bandwidth:
>>
>> (Payload_Size + L3_Header + L2_Header) x PPS x 8
>>
>> Where
>> *Payload_Size* -  is the Codec Sample Size (Bit Rate). G711 at 20ms has
>> 160 bytes. G729 at 20ms has 20 Bytes and etc (you can find Sampling Rate
>> info on Cisco.com)
>> *L3 Header*  -   is the IP/UDP/RTP, which is 40 bytes, unless compressed
>> RTP is used. If the UDP checksum is used then the L3 Header is compresesd
>> up to 4 bytes, and if not used then up to 2 bytes. (UDP checksum is enabled
>> on a VoIP Dial-peer: "ip udp checksum" command. By default is disabled)
>> *L2 Header*  -  for Ethernet it is 18 bytes, for Frame Relay is 4 bytes
>> (based on QoS SRND).
>> *PPS*  -  Packets per Second. This is based on the Sampling Rate as well
>> in the following way:  PPS = 1000 / Sample_Size in ms.
>> *8*  -   this is just bits. To convert Bytes into bits.
>>
>>
>>
>>  On Thu, Feb 2, 2012 at 1:39 AM, Nathan Silvers wrote:
>>
>>>  Hi Everyone,
>>>
>>> Just starting out on my CCIE path and had a revelation that might not be
>>> huge to anyone else but if it helps one person understand I've done a good
>>> job...
>>>
>>> So I was struggling with Codecs how to calculate bandwidth required and
>>> all the header and crtp if that is involved and then Cisco throws on Layer
>>> 3 Bandwidth vs Overall bandwidth and from a humble voice guy who has
>>> focused mainly on LAN connections I am lost...
>>>
>>> http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml#topic1
>>> Using the above Cisco document which can help and confuse(as they use
>>> median BW when it comes to the CRTP compression as it can be a 2 or 4 byte
>>> header depending on if you are using the UDP checksum) so my exact
>>> calculations were always off a few bytes..
>>>
>>> So I was thinking about this logically and looking at the different
>>> pieces of the puzzle,
>>> Layer 3 is going to ultimately be the voice packets which is defined by
>>> the codec.. 711 = 64kbs, 729 = 8kbs, etc...
>>> Layer two includes my headers and such which add to the needed bandwidth
>>> depending on if they are compressed
>>> So if you use CRTP you only need and additional 3.6kbs for 711 which is
>>> the 67.6kbs bandwidth, or 11.6kbs for 729
>>> Now if uncompressed the headers can add quite a bit more requiring the
>>> need for 18.8kbs more to put 711 at 82.8kbs and 729 at 26.8kbs
>>>
>>> So if you see a question regarding layer 3 bandwidth it is always the
>>> codec kb

Re: [OSL | CCIE_Voice] Codec's, Compression and Bandwidth

[datucha123 datucha123]

G711 64kbps and G729 8kbps are not the L3 Bandwidth for those codecs. These
are the Payload Bandwidths for those codecs.

G711 and G729 on L3 are using 80 and 24 kbps respectavely.

Here is the easy way to calcualte the Codec bandwidth:

(Payload_Size + L3_Header + L2_Header) x PPS x 8

Where
*Payload_Size* -  is the Codec Sample Size (Bit Rate). G711 at 20ms has 160
bytes. G729 at 20ms has 20 Bytes and etc (you can find Sampling Rate info
on Cisco.com)
*L3 Header*  -   is the IP/UDP/RTP, which is 40 bytes, unless compressed
RTP is used. If the UDP checksum is used then the L3 Header is compresesd
up to 4 bytes, and if not used then up to 2 bytes. (UDP checksum is enabled
on a VoIP Dial-peer: "ip udp checksum" command. By default is disabled)
*L2 Header*  -  for Ethernet it is 18 bytes, for Frame Relay is 4 bytes
(based on QoS SRND).
*PPS*  -  Packets per Second. This is based on the Sampling Rate as well in
the following way:  PPS = 1000 / Sample_Size in ms.
*8*  -   this is just bits. To convert Bytes into bits.



On Thu, Feb 2, 2012 at 1:39 AM, Nathan Silvers  wrote:

> Hi Everyone,
>
> Just starting out on my CCIE path and had a revelation that might not be
> huge to anyone else but if it helps one person understand I've done a good
> job...
>
> So I was struggling with Codecs how to calculate bandwidth required and
> all the header and crtp if that is involved and then Cisco throws on Layer
> 3 Bandwidth vs Overall bandwidth and from a humble voice guy who has
> focused mainly on LAN connections I am lost...
>
> http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml#topic1
> Using the above Cisco document which can help and confuse(as they use
> median BW when it comes to the CRTP compression as it can be a 2 or 4 byte
> header depending on if you are using the UDP checksum) so my exact
> calculations were always off a few bytes..
>
> So I was thinking about this logically and looking at the different pieces
> of the puzzle,
> Layer 3 is going to ultimately be the voice packets which is defined by
> the codec.. 711 = 64kbs, 729 = 8kbs, etc...
> Layer two includes my headers and such which add to the needed bandwidth
> depending on if they are compressed
> So if you use CRTP you only need and additional 3.6kbs for 711 which is
> the 67.6kbs bandwidth, or 11.6kbs for 729
> Now if uncompressed the headers can add quite a bit more requiring the
> need for 18.8kbs more to put 711 at 82.8kbs and 729 at 26.8kbs
>
> So if you see a question regarding layer 3 bandwidth it is always the
> codec kbs, ie 711 is 64 and 729 is 8 regardless of any compression, the
> type of connection etc.. more of a no brainer question
> They Layer 2 bandwidth is where the savings are by chopping the 40byte
> IP/UDP/RTP header into 2 byes (or 4 bytes if UDP Checksum is enabled.)
>
> Gotta love white boards and running through a few situations.. the cisco
> doc has the calculations to go through and how changing the payload size
> can adjust the Packets Per Second which affects the required Bandwidth. If
> anyone else is struggling with this I highly recommend just writing it all
> out and trying hypothetical situations.
>
> Hope this helps someone!
>
>
> --
> "The biggest mistake people make in life is not trying to make a living at
> doing what they most enjoy."
>
> - Malcolm Forbes
>
> Nathan Silvers
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
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Re: [OSL | CCIE_Voice] Calling Party Transformation set at the egress GW

[datucha123 datucha123]

That is because of IOS.

IOS detects the US Dialplan, and sets the Types accordingly in H323
gateway.
It is not possible to disable that feature. So you have to use voice
translation rules to change the ANI Type.
On Wed, Feb 1, 2012 at 2:01 PM, Juan Lopez
wrote:

> Hi all,
>
> I'm doing lab 5 Vol1 with some extra things into it and found the
> following:
>
> Has anyone noticed that when using a H323 GW in a backup fashion (example:
> RG-BR1 contains BR1-GW as primary and HQ-GW as secondary), the calling
> number type is ALWAYS set to national, unless you use a prefix in the
> calling party transformation?
>
> For most cases this is ok, but not for all (example: international TEHO
> for calls from the UCME)
>
> any feedback is welcome !
>
> grts,
> Juan
>
> ___
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> visit www.ipexpert.com
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>
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Re: [OSL | CCIE_Voice] CCIE#34310

[datucha123 datucha123]

Congratulations.

On Thu, Jan 26, 2012 at 10:05 PM, Bill Lake  wrote:

> Congratulations, take a well earned break before your work picks up and
> your busier than those of us studying.
>
>   On Thu, Jan 26, 2012 at 10:11 AM, Thomas Koch wrote:
>
>>Congrats!!
>>
>> ** **
>>
>> Regards,
>>
>> ** **
>>
>> Thomas Koch
>>
>> 
>>
>> ** **
>>
>> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
>> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Chris Martin
>> *Sent:* Thursday, January 26, 2012 8:04 AM
>>
>> *To:* ccie_voice@onlinestudylist.com
>> *Subject:* [OSL | CCIE_Voice] CCIE#34310
>>
>>  ** **
>>
>> Well its official, got my results in this morning, CCIE # 34310.  Just
>> want to give a big thanks to everyone here for a great community in which
>> to strive toward this goal over the last year.  Thanks Vik and IPExpert for
>> the great videos, boot camp, labs rack rentals, I can not imagine myself
>> achieving success without your products and ongoing support.  A big shout
>> out to my boot camp  buddies from 11/11/11, keep your head down and don't
>> give up, I am sure you will all make it!
>>
>> ** **
>>
>> Now time for a break.. and maybe a real book, that isn't a SRND or Cisco
>> Press...
>>
>> ** **
>>
>> Chris Martin
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com 
>>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
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Re: [OSL | CCIE_Voice] MGCP gateways

[datucha123 datucha123]

If you configure ccm-manager config, it will try to configure all 24/32 PRI
ports, and if you are not using full PRI, then you have to manually disable
unused PRI channels, and also configure the B channel maintenance in CUCM,
so that after every reload the Router will not try to configure all PRI
channel again.

Bu if you are using manual MGCP configuration, you do not have such problem
in case of fractional PRI.

On Wed, Jan 25, 2012 at 5:08 PM, Chevy  wrote:

> I just thought I'd ping everyone to see what your thoughts are on
> configuring an MGCP gateway manually vs using the ccm-manager config
> commands to download the xml file from the call manager.
>
>
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> visit www.ipexpert.com
>
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>
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Re: [OSL | CCIE_Voice] New Five Labs: UCCX Question

[datucha123 datucha123]

I have tested, and Telecaster user does not need any special User Groups
(except Standard CTI Enabled).

I can tell, that even the Telecaster User is not necessary at all.

On Tue, Jan 24, 2012 at 1:28 PM, Anthony Alba wrote:

> The script populates the variable at runtime with "Set Enterprise Call
> Info" step; this happens in the "Select Resource" step before you connect
> the caller to the agent.
>
> Now, much to my surprise, I actually managed to get this to work and I saw
> the field get updated.
>
> The question I want to ask the list: does the "telecaster" application
> user need any specific User Group (e.g. Standard CTI Enabled)
> ? The 6.6 and 8.x CAD are quite skimpy on this: they state to create an
> application user telecaster/password telecaster and associate all Agent
> phones. They don't mention whether the telecaster user needs specific roles.
> Searches on this list turn up which state that the telecaster user is
> needed for Expanded Call Variables to work.
>
>
>
>
>   On Mon, Jan 23, 2012 at 10:12 PM, John McGaughey (jomcgaug) <
> jomcg...@cisco.com> wrote:
>
>>In lab 1 of the new 5 labs, question 7.3 it asks that the agent be
>> able to see the ani and number of calls in queue.  The DSG says to add a
>> “salesinq” field to the default layout.  The problem is that there is
>> nothing telling IPPA what to populate this field with.
>>
>> ** **
>>
>> So in my lab I see the following when I press CDATA.
>>
>> ** **
>>
>> ANI: 4678124
>>
>> callsinq:
>>
>> ** **
>>
>> But the DSG is showing callsinq: 1.  There must be a step missing from
>> the DSG.  How do we populate the callsinq field?
>>
>> ** **
>>
>> John
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com 
>>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
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> www.PlatinumPlacement.com 
>
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Re: [OSL | CCIE_Voice] New Five Labs: UCCX Question

[datucha123 datucha123]

I cannot understand, how the Agent (IPPA or CAD) can show manually
configured Varibales values, until the call comes to Agents phone.

You can use the Get Reporting Statistics Step to gather info about the
Queue  -  But I do not know, how to pass that variables to Agents layout in
real time  - the call did not get to agents yet, so how that parameter is
dynamically updated on Agents screen?

On Mon, Jan 23, 2012 at 6:12 PM, John McGaughey (jomcgaug) <
jomcg...@cisco.com> wrote:

>  In lab 1 of the new 5 labs, question 7.3 it asks that the agent be able
> to see the ani and number of calls in queue.  The DSG says to add a
> “salesinq” field to the default layout.  The problem is that there is
> nothing telling IPPA what to populate this field with.
>
> ** **
>
> So in my lab I see the following when I press CDATA.
>
> ** **
>
> ANI: 4678124
>
> callsinq:
>
> ** **
>
> But the DSG is showing callsinq: 1.  There must be a step missing from the
> DSG.  How do we populate the callsinq field?
>
> ** **
>
> John
>
> ___
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> visit www.ipexpert.com
>
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> www.PlatinumPlacement.com 
>
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Re: [OSL | CCIE_Voice] IPX voice vol.1 lab #8 - PRI weird behaviour

[datucha123 datucha123]

Check the "Inhibit restarts at PRI initialization" on MGCP gateway in
CUCM configuration page.

That might help

On Mon, Jan 23, 2012 at 6:31 AM, Jeferson Guardia wrote:

> Getting that weird message with my PRI every 30 seconds, I googled, found
> people having the same issue but I didnt find any solution and this was at
> the end of my lab, but anyway it raised my curiosity.. you never know you
> might get the same thing on the lab, gotta be ready to catch any curve ball
> they throw at you :-)
>
>  BR1-RTR#
> Jan 23 02:21:58.590: ISDN Se0/0/0:23 Q931: RX <- RESTART pd = 8  callref =
> 0x
> Restart Indicator i = 0x87
>
> Looking at general status looks ok
>
>  BR1-RTR#sh isdn stat
> Global ISDN Switchtype = primary-ni
>
> %Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output may
> not apply
>
> ISDN Serial0/0/0:23 interface
> dsl 0, interface ISDN Switchtype = primary-ni
> L2 Protocol = Q.921 0x  L3 Protocol(s) = CCM MANAGER 0x0003
> Layer 1 Status:
> ACTIVE
> Layer 2 Status:
> TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
> Layer 3 Status:
> 0 Active Layer 3 Call(s)
> Active dsl 0 CCBs = 0
> The Free Channel Mask:  0x8007
> Number of L2 Discards = 0, L2 Session ID = 11
> Total Allocated ISDN CCBs = 0
> BR1-RTR#
>
> Then my q921 debug tells me the following:
>
>  BR1-RTR#
> Jan 23 02:24:28.589: ISDN Se0/0/0:23 Q921: User TX -> RRp sapi=0 tei=0
> nr=42
> Jan 23 02:24:28.589: ISDN Se0/0/0:23 Q921: User RX <- INFO sapi=0 tei=0,
> ns=42 nr=12
> Jan 23 02:24:28.589: ISDN Se0/0/0:23 Q921: User TX -> RR sapi=0 tei=0 nr=43
> Jan 23 02:24:28.593: ISDN Se0/0/0:23 Q921: User RX <- RRf sapi=0 tei=0
> nr=12
> BR1-RTR#
>
>
> Yes, this was at proctorlabs.. I bet some of you might have faced the same
> issue since this is on the initial config. Somehow I just hate MGCP, yes I
> tried reconfiguring it and also the famous workaround (no mgcp/mgcp), but
> it didnt work.
>
> Cheers!
>
>
> --
> Jeferson Guardia
> CCIE #28157
>
> ___
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> visit www.ipexpert.com
>
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Re: [OSL | CCIE_Voice] No VM recorded to endpoint from PSTN

[datucha123 datucha123]

I have the same issue,

That's because of VMware  -  well I got to this conclusion.

On Mon, Jan 23, 2012 at 6:24 AM, Jurassic Labs wrote:

> I've noticed this lately while loading up the newer 5-lab self study
> vRacks sessions.  Once a phone (say Ext 2002) is defined in Unity
> Connection, if the PSTN phone calls that extension, the greeting is heard
> after the appropriate no-answer timeout and you can leave a message.
> However, NO MWI light shows up on 2002...because there is no "new message"
> for that mailbox!  The same applies if you call the same extension - but
> from another extension that is NOT defined in Unity Connection.  No if you
> have another extension (say 3001) call 2001 and leave a message -
> everything works...MWI, new messages, etc.  because 3001 is defined in
> Unity Connection.
>
> So is there some setting in Unity Connection that will only "record" and
> retain a message from a known subscriber?  I'm scratching my head on this
> one...
>
>
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[OSL | CCIE_Voice] H225 Preservation

[datucha123 datucha123]

Hello,

What is the difference between these two commands:


no h225 timeout keepalive

call preserve limit-media-detection


Well, both commands (separately) can be used for SRST, when the CUCM goes
down, and active calls should be left active.

Also I have read some documentation, and the final usage for those command
are the same.

So can you tell tell me what is the difference between those two commands?
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Re: [OSL | CCIE_Voice] RSVP CAC Sampling Rate

[datucha123 datucha123]

Nicolas, if you do not like my e-mails, just do not read them, (I do not
need your personal comments at all).

There is a great feature in your E-mail client  -  *Delete This E-mail*
or even  - *Unsubscribe from OSL
*
So you can use those "buttons" to get rid of my e-mails.

I will always send any questions that I am interested in and I do not need
your personal comments about them.

Thank you.
On Sun, Jan 22, 2012 at 3:23 AM, Nicolas MICHEL wrote:

> Hey there George.
>
> I agree with your statements, some of the posts are very very interesting
> but I think we are overwhelmed with the number of questions.
>
> Again I dont want to be mean with anyone but I received some private mails
> that agreed with my opinion.
> We could then start a forum that would track any voice topics...
>
> My 2 cts.
>
> Nic
>
>
>
> Nicolas,
>
> I think it would be more constructive if you actually answered the guy.
> this is a very good question, and I believe a lot of people might have this
> wrong.
>
> As I already answered Datucha in private. I'll just put here the correct
> formula for calculations as per Cisco SRND:
>
> For N amount of calls configured for any codec speed with any sampling
> period:
>
> bandwidth = (N-1) * (configured codec speed on configured sampling period)
> + (codec speed on worth case scenario sampling period - 10ms)
> as per this example for g729 configured with 30ms sampling period on CUCM
> for 10 calls:
> bandwidth= 9 * 19kbps + 40kbps
>
> where 19 is g729 speed on 30ms sampling rate... well it's actually
> 18.648bps but I rounded it up.  and 40 is same codec on worth case scenario
> 10ms sampling rate.
>
> Hope this helps,
>
> On Sat, Jan 21, 2012 at 4:38 PM, Nicolas MICHEL wrote:
>
>>  Damn man,
>>
>> Do we need to correct you for everything you test ?
>> Dude make your own assumption sometimes and please don't use this list as
>> your "please Can you verify this and tell me if I'm wrong or not" mailing
>> list.
>>
>> I guess this is a mailing list that is related to CCIE and I'm sure some
>> of your post are very interesting but since we receive maybe 4 or 5 mails
>> from you everyday I feel the need to tell you , man please don't flood us
>> like this !!
>>
>> Have a good day !  ;)
>>
>> Nic
>>
>> Le 21 janv. 2012 à 13:33, datucha123 datucha123  a
>> écrit :
>>
>>   Sorry I make a little mistake here in calculation (at 30ms Sampling
>> Rate G729 needs 40kbps for RSVP CAC), but it does not change the idea of
>> question,
>>
>> So in case of 30ms Sampling Rate, we do not need to add any 16 bkps and:
>>
>> 10 G729 30ms Calls  -  40 x 10  = 400kbps
>>
>> On Sat, Jan 21, 2012 at 4:20 PM, datucha123 datucha123 <
>> datucha...@gmail.com> wrote:
>>
>>>
>>> Hello,
>>>
>>> When we are using default sampling rate for codecs, for example G729 of
>>> 20ms, then during the RSVP CAC caluclation we need to add 16 bkps for sum
>>> of all calls:
>>>
>>> 10 G729 calls  -  24 x 10 + 16 = 256 kbps
>>>
>>> But when using non-default Sampling Rate for codecs, for instance G729
>>> at 30ms, then we do not have to add 16 but add 12 kbps:
>>>
>>> 10 G729 30ms calls  -  28 x 10 +12 = 292
>>>
>>> Is it correct or not?
>>>
>>
>>  ___
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>> visit www.ipexpert.com
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>>
>>
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>>
>
>
>
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[OSL | CCIE_Voice] Type and Plan on H323

[datucha123 datucha123]

Hello,

H323 Router by default (in IOS) sets the Calling Type and Plan for outgoing
calls to PRI. - I think there is something put in IOS so that it recognizes
the US dialplan, and set the Type and Plan automatically.

How can I disable it?

Because when I set the Type and Plan in Callmanager, and send the call to
h323 gateway, if the Calling Number matches the US dialplan, then the H323
gateway overwrites the Type and Plan.
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Re: [OSL | CCIE_Voice] RSVP CAC Sampling Rate

[datucha123 datucha123]

Sorry I make a little mistake here in calculation (at 30ms Sampling Rate
G729 needs 40kbps for RSVP CAC), but it does not change the idea of
question,

So in case of 30ms Sampling Rate, we do not need to add any 16 bkps and:

10 G729 30ms Calls  -  40 x 10  = 400kbps

On Sat, Jan 21, 2012 at 4:20 PM, datucha123 datucha123  wrote:

>
> Hello,
>
> When we are using default sampling rate for codecs, for example G729 of
> 20ms, then during the RSVP CAC caluclation we need to add 16 bkps for sum
> of all calls:
>
> 10 G729 calls  -  24 x 10 + 16 = 256 kbps
>
> But when using non-default Sampling Rate for codecs, for instance G729 at
> 30ms, then we do not have to add 16 but add 12 kbps:
>
> 10 G729 30ms calls  -  28 x 10 +12 = 292
>
> Is it correct or not?
>
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[OSL | CCIE_Voice] RSVP CAC Sampling Rate

[datucha123 datucha123]

Hello,

When we are using default sampling rate for codecs, for example G729 of
20ms, then during the RSVP CAC caluclation we need to add 16 bkps for sum
of all calls:

10 G729 calls  -  24 x 10 + 16 = 256 kbps

But when using non-default Sampling Rate for codecs, for instance G729 at
30ms, then we do not have to add 16 but add 12 kbps:

10 G729 30ms calls  -  28 x 10 +12 = 292

Is it correct or not?
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[OSL | CCIE_Voice] Cisco IP Phone Codecs

[datucha123 datucha123]

please correct if wrong

G711 and G729 variants (g722, iLBC are not part for this email)

Cisco IP phones default codecs  -  G711Ulaw, and G729ar8.

Cisco IP phone supported codecs (SCCP)  -  G711u, G711a, G729r8, G729ar8,
G729br8 and G729abr8
Cisco IP Phone supported codecs (SIP)  -  almost the same Codecs as SCCP
version, except SIP Phone does not support Annex-B variants.
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Re: [OSL | CCIE_Voice] Gatekeeper VIAZONE - Could not find an IPIPGW

[datucha123 datucha123]

Also as I remember, the IPIPGW tech prefix has to match the Destination
Zone prefix.

On Wed, Jan 18, 2012 at 9:39 PM, Steven <
forum.ccie.onlinestudyl...@nocer.net> wrote:

> @Boris
> @Leslie
> @Amit
> I got some other issues too.
> I skipped to check the GK-only functionality (BIG mistake).
> After i fixed the normal (without outvia) GK functions i revisited the
> CUBE issue.
> It turns out i accidently put the allow-connections on the Br2 instead of
> the HQ.
>
> Thanks for your time and help! :D
>
> Regards Steven
>
> __**_
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] CUCM version to study > upgrade to latest 7.0.x release?

[datucha123 datucha123]

well, I do not know the very exact version of CUCM on the LAB Exam, but I
think it is better to study with the one that proctollabs offer. After you
will know the bugs and issues for that UCM/

On Wed, Jan 18, 2012 at 9:03 PM, Juan Lopez
wrote:

> all,
> I found the version on proctorlabs (7.0.1.11002-2) is giving me quite some
> issues with dialrules on the 7962.
> Is it a good thing to upgrade to the latest 7.0.x release to study,
> without being out of sync with the tested UCM version?
> Or should I simply upgrade the phone firmwares instead? what is the best
> way to prepare for the real exam?
>
> thx,
> Juan
>
> ___
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> visit www.ipexpert.com
>
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> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] SRST problem: Phones in srst mode have a different configuration than they were registered to CUCM

[datucha123 datucha123]

try to reload the Router.

On Wed, Jan 18, 2012 at 3:36 PM, The Masterplan wrote:

> The same thing. The only difference is that now I don't see anymore the
> phone in show run.
>
>
> On Wed, Jan 18, 2012 at 12:34 PM, Mohd Baqari wrote:
>
>> Ok ... Try changing the provision mode to none ... Delete ephone-dn 1 and
>> ephone 3  Then do your testing.
>>
>> Regards,
>> Mohammed Al Baqari
>>
>> Sent from my iPhone
>>
>> On Jan 18, 2012, at 1:12 PM, The Masterplan 
>> wrote:
>>
>> The phone does not appear in the output of the command:
>> tftp-server flash:gui/ephone_admin.html
>>  max-ephones 14
>> ephone-dn  11  dual-line
>>  number 1002 no-reg primary
>>  description Cisco IP Communicator
>>  name Cisco IP Communicator
>> ephone-dn  12  dual-line
>>  number 1005 no-reg primary
>>  description IP Blue
>>  name IP Blue
>>  night-service bell
>> ephone  1
>>  description Cisco IP Communicator
>>  mac-address F0DE.F173.03E2
>>  type CIPC
>>  button  1:11
>> ephone  2
>>  description IP Blue
>>  mac-address 0050.56C0.0008
>>  type CIPC
>>  button  1:12
>>
>> After I switch to srst, the output of the command looks like this:
>> tftp-server flash:gui/ephone_admin.html
>>  max-ephones 14
>> ephone-dn  1  dual-line
>>  number 
>>  description 7945 hardware
>>  name 7945 hardware
>> ephone-dn  11  dual-line
>>  number 1002 no-reg primary
>>  description Cisco IP Communicator
>>  name Cisco IP Communicator
>> ephone-dn  12  dual-line
>>  number 1005 no-reg primary
>>  description IP Blue
>>  name IP Blue
>>  night-service bell
>> ephone  1
>>  description Cisco IP Communicator
>>  mac-address F0DE.F173.03E2
>>  type CIPC
>>  button  1:11
>> ephone  2
>>  description IP Blue
>>  mac-address 0050.56C0.0008
>>  type CIPC
>>  button  1:12
>> ephone  3
>>  mac-address 0817.3514.5682
>>  button  1:1
>>
>>
>> On Wed, Jan 18, 2012 at 10:25 AM, Mohd Baqari 
>> wrote:
>>
>>> Post the the output of "show run | sec ephone". Probably the old config
>>> of ephone-dn is saved in running config due to provision all
>>>
>>> Regards,
>>> Mohammed Al Baqari
>>>
>>> Sent from my iPhone
>>>
>>> On Jan 18, 2012, at 11:55 AM, The Masterplan 
>>> wrote:
>>>
>>> Hi,
>>>
>>> I already did no create cnf-files & create cnf-files and reset the phone
>>> to factory defaults and nothing. I'm running srst mode auto provision all.
>>> See below the config:
>>> telephony-service
>>>  srst mode auto-provision all
>>>  srst dn line-mode dual
>>>  em logout 0:0 0:0 0:0
>>>  max-ephones 14
>>>  max-dn 30 no-reg
>>>  ip source-address 10.1.1.25 port 2000
>>>  system message SRST
>>>  max-conferences 8 gain -6
>>>  transfer-system full-consult
>>>  create cnf-files version-stamp 7960 Jan 18 2012 08:09:33
>>>
>>> On Tue, Jan 17, 2012 at 7:13 PM, Ken Wyan  wrote:
>>>
 Did you enter no create cnf-files & create cnf-files on CME Router ?

 Which srst mode are you running? srst mode auto provision all | dn |
 none ?
 Better to post full telephony-service configuration here.

 (In SRST phone may be downloading previous xml configuration file from
 the router. Delete it from flash if so)



 On Tue, Jan 17, 2012 at 6:53 PM, The Masterplan <
 winmasterp...@gmail.com> wrote:

> Hello,
>
> I have a problem regarding srst. The 2811 router than now is a srst
> was a acting as a cme in past in a demo lab and the 7960 phone was
> registered to it with extension . Now, the 7960 phone is registered in
> UCM with extension 5001 and the 2811 router is configured in telephony
> service srst mode. The problem is that although the old configuration of
> the router was erased, when it goes in srst fallback mode, the 7960 gets
> extension  instead of 5001 and the command show telephony-service
> ephone shows the specified phone with message:This is an srst fallback
> phone.
>
> Thank you for your answer
>
> ___
> For more information regarding industry leading CCIE Lab training,
> please visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>


>>> ___
>>> For more information regarding industry leading CCIE Lab training,
>>> please visit www.ipexpert.com
>>>
>>> Are you a CCNP or CCIE and looking for a job? Check out
>>> www.PlatinumPlacement.com
>>>
>>>
>>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] redundancy for SIP dialpeers

[datucha123 datucha123]

Also do not forget to configure "timer trying"  - 150/200 ms is OK

On Wed, Jan 18, 2012 at 12:46 AM, Juan Lopez  wrote:

> When configuring 2 SIP dialpeers for redundancy, together with:
> sip-ua
> retry invite 2
>
> This should generate in total 3 INVITES sent to the primary UCM via the
> first dialpeer, before going over to the second sip dialpeer, right?
> Doing debug ccsip messages only shows 1 invite sent to the primary, and
> then 1 invite to the secondary.
> am I missing something?
> thanks, Juan
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
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Re: [OSL | CCIE_Voice] CUE error

[datucha123 datucha123]

First of all ensure that the Authentication URL is set correctly.

Also you can try to reset the CUE module - just in case.

On Tue, Jan 17, 2012 at 6:30 PM, study buddy  wrote:

> Hi
>
> While accessing my voicemail via VoiceView, I get the following error when
> I click on listen
>
> Unknown error code {0}. Report this error to your system administrator
>
> Has anyone run into this error before? I tried several things but I cant
> get over this error
>
> T&R
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
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Re: [OSL | CCIE_Voice] Unity Connection Integration

[datucha123 datucha123]

You can set it to any server you like initially, but when you go to Server
configuration in Port Group page, there you have to set them as necessary
and required.

On Wed, Jan 18, 2012 at 12:18 AM, datucha123 datucha123 <
datucha...@gmail.com> wrote:

> That parameter actually does not effect anything.
>
>   On Tue, Jan 17, 2012 at 9:22 PM, Hough, Earl <
> earl.ho...@pcmallservices.com> wrote:
>
>>I guess it would determine what your requirements are.  If, for
>> example, the global requirements were that your subscriber were to be the
>> primary server for all call processing, you might want to take that into
>> account when choosing whether to only use one server or two and which
>> should be the primary and secondary.
>>
>> ** **
>>
>> Earl Hough
>>
>> CCIE #16508 (R&S/Security/Voice)
>>
>> ** **
>>
>> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
>> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Ken Wyan
>> *Sent:* Tuesday, January 17, 2012 11:42 AM
>> *To:* ccie_voice@onlinestudylist.com
>> *Subject:* [OSL | CCIE_Voice] Unity Connection Integration
>>
>> ** **
>>
>> Hi,
>>
>>  
>>
>> When adding a new port group to unity connection , it prompts to enter
>> only one CUCM ip address. Should we essentially provide CUCM Publisher here?
>> 
>>
>>  
>>
>> My Integration is CUCM Sub First & CUCM Pub Second as we expect to do in
>> CCIE Exam.
>>
>>  
>>
>> When adding a phone system we can provide Subscriber (first) & Publisher
>> (second) in order (AXL Server)
>>
>>  
>>
>> When adding a new portgroup , it asks for only one ip address. The ip
>> address I provide is displayed under Port Group Configuration (CUCM & TFTP)
>> & we can later add another CUCM address also.
>>
>>  
>>
>> Any guideline?
>>
>>  
>>
>> _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _
>>
>> The information contained in this transmission is confidential. It is
>> intended solely for the use of the individual(s) or organization(s) to
>> whom it is addressed. Any disclosure, copying or further distribution is
>> not permitted unless such privilege is explicitly granted in writing by
>> PC Mall, Inc. Furthermore, PC Mall, Inc. is not responsible for
>> the proper and complete transmission of the substance of this
>> communication, nor for any delay in its receipt.
>>
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com <http://www.platinumplacement.com/>
>>
>
>
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Re: [OSL | CCIE_Voice] Unity Connection Integration

[datucha123 datucha123]

That parameter actually does not effect anything.

On Tue, Jan 17, 2012 at 9:22 PM, Hough, Earl
wrote:

>  I guess it would determine what your requirements are.  If, for example,
> the global requirements were that your subscriber were to be the primary
> server for all call processing, you might want to take that into account
> when choosing whether to only use one server or two and which should be the
> primary and secondary.
>
> ** **
>
> Earl Hough
>
> CCIE #16508 (R&S/Security/Voice)
>
> ** **
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Ken Wyan
> *Sent:* Tuesday, January 17, 2012 11:42 AM
> *To:* ccie_voice@onlinestudylist.com
> *Subject:* [OSL | CCIE_Voice] Unity Connection Integration
>
> ** **
>
> Hi,
>
>  
>
> When adding a new port group to unity connection , it prompts to enter
> only one CUCM ip address. Should we essentially provide CUCM Publisher here?
> 
>
>  
>
> My Integration is CUCM Sub First & CUCM Pub Second as we expect to do in
> CCIE Exam.
>
>  
>
> When adding a phone system we can provide Subscriber (first) & Publisher
> (second) in order (AXL Server)
>
>  
>
> When adding a new portgroup , it asks for only one ip address. The ip
> address I provide is displayed under Port Group Configuration (CUCM & TFTP)
> & we can later add another CUCM address also.
>
>  
>
> Any guideline?
>
>  
>
> _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _
>
> The information contained in this transmission is confidential. It is
> intended solely for the use of the individual(s) or organization(s) to
> whom it is addressed. Any disclosure, copying or further distribution is
> not permitted unless such privilege is explicitly granted in writing by
> PC Mall, Inc. Furthermore, PC Mall, Inc. is not responsible for
> the proper and complete transmission of the substance of this
> communication, nor for any delay in its receipt.
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
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[OSL | CCIE_Voice] Presence on CUPC

[datucha123 datucha123]

Hello,

Based on my testings, CUPC support Presence Status change in its Contacts
only when they are imported from AD.
Manually created contacts does not support Presence Status Change.

Is it correct?
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Re: [OSL | CCIE_Voice] Cant connect to Sub

[datucha123 datucha123]

Also check the DB Replication Status

On Tue, Jan 17, 2012 at 6:17 AM, Ashraf Ayyash  wrote:

> Hello Errol ,
>
> This issue usually mean you have incorrect Host/ processNode files in
> your PUB or the Sub and so the communication is broken between the PUB
> and the SUB and you have to find what table we have corrupted and fix
> it , or you have a big timing difference reported by your NTP ,
>
> can you check what service you have running on both pub and sub
> specially the A Cisco DB and let me know what you will get .
>
> Ash
>
> On Mon, Jan 16, 2012 at 5:15 PM, Errol Abrahams 
> wrote:
> >
> >
> > Hi All,
> >
> > I had a problem with my VMWARE Server and I had to rebuilt the system
> from
> > scratch. I have reloaded PUB,SUB,CUPS,CUC and CUCCX and all virtual
> > addresses are pingable. When I activate the services for the PUB from the
> > Cisco Unified Serviceability screen, it worked. But, when I try to access
> > the SUB from same screen then it displays'Connection to the Server
> cannot be
> > established(unable to access Remote Node).
> >
> > Has anybody had a problem like this and how can I fix this problem. Your
> > help is appreciated..thnx.
> >
> > Chhers
> >
> > EA
> >
>  > ___
> > For more information regarding industry leading CCIE Lab training, please
> > visit www.ipexpert.com
> >
> > Are you a CCNP or CCIE and looking for a job? Check out
> > www.PlatinumPlacement.com 
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
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Re: [OSL | CCIE_Voice] BLF Speed Dials

[datucha123 datucha123]

Thank you George

On Mon, Jan 16, 2012 at 8:09 PM, George Goglidze  wrote:

> Hi Datucha,
>
> Actually for CUPS you need only line association. you do not need to
> specify owner user id.
>
> Misread the question a little bit initially.
>
> the owner user id on the phone is for SNR.
>
> Cheers,
>
> On Mon, Jan 16, 2012 at 4:01 PM, George Goglidze wrote:
>
>> correctisimo  :-)
>>
>>
>>
>>  On Mon, Jan 16, 2012 at 3:25 PM, datucha123 datucha123 <
>> datucha...@gmail.com> wrote:
>>
>>>  When configuring the simple BLF Speed Dial, we need to configure the
>>> Subscribe CSS for watching Device. So that it could the the Watched DN.
>>>
>>> But, the Owner User ID,  Line Association with End User and other kinds
>>> of associations are not required for this BLF, right?
>>>
>>> Even for Call List Presence.
>>>
>>>
>>>
>>> The Owner User ID and Line Association with End User, along with License
>>> Capability Assignements are required only for CUPS Presence method.
>>>
>>> Is it correct?
>>>
>>> ___
>>> For more information regarding industry leading CCIE Lab training,
>>> please visit www.ipexpert.com
>>>
>>> Are you a CCNP or CCIE and looking for a job? Check out
>>> www.PlatinumPlacement.com <http://www.platinumplacement.com/>
>>>
>>
>>
>
___
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[OSL | CCIE_Voice] BLF Speed Dials

[datucha123 datucha123]

When configuring the simple BLF Speed Dial, we need to configure the
Subscribe CSS for watching Device. So that it could the the Watched DN.

But, the Owner User ID,  Line Association with End User and other kinds of
associations are not required for this BLF, right?

Even for Call List Presence.



The Owner User ID and Line Association with End User, along with License
Capability Assignements are required only for CUPS Presence method.

Is it correct?
___
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Re: [OSL | CCIE_Voice] MGCP Registration

[datucha123 datucha123]

What does the "show isdn status" shows up?

On Mon, Jan 16, 2012 at 5:35 PM, mercy forall wrote:

>  Hi,
>
> thanks for your support and good link\
>
> this is my GW configuration , also it is connected to other cisco GW as
> PSTN GW through E1 cross cable
>
>  sh run :
> Current configuration : 15381 bytes
> !
>
> !
> version 15.0
> service timestamps debug datetime msec
> service timestamps log datetime msec
> no service password-encryption
> !
> hostname 
> !
> boot-start-marker
> boot-end-marker
> !
> logging buffered 51200 warnings
>
> no aaa new-model
>
> network-clock-participate wic 2
> !
> dot11 syslog
> ip source-route
> !
> ip cef
> !
> !
> no ipv6 cef
> multilink bundle-name authenticated
> !
> !
> !
> isdn switch-type primary-qsig
> !
> voice-card 0
> !
> !
> voice rtp send-recv
> !
> voice service voip
>  allow-connections h323 to h323
>  allow-connections h323 to sip
>  allow-connections sip to h323
>  allow-connections sip to sip
>  redirect ip2ip
>  h323
>  sip
>   header-passing
>   no call service stop
> !
> voice class codec 1
>  codec preference 1 g711ulaw
> !
> voice class custom-cptone
>  dualtone disconnect
>   frequency 425
>   cadence 250 250
> !
> !
> !
> !
> http client cache memory pool 15000
> http client cache memory file 500
> http client connection timeout 60
> http client connection idle timeout 10
> http client response timeout 30
>
> mrcp client timeout connect 10
> mrcp client timeout message 10
> mrcp client rtpsetup enable
> vxml tree memory 500
> vxml audioerror
> vxml version 2.0
> !
> crypto pki trustpoint TP-self-signed-3307538538
>  enrollment selfsigned
>  subject-name cn=IOS-Self-Signed-Certificate-3307538538
>  revocation-check none
>  rsakeypair TP-self-signed-3307538538
>
> !
> controller E1 0/2/0
>  pri-group timeslots 1-4,16 service mgcp
> !
>
> interface GigabitEthernet0/0
>  no ip address
>  duplex auto
>  speed auto
>  media-type rj45
> !
>
>
> interface GigabitEthernet0/0.1
>  encapsulation dot1Q
>  ip address X.X.X.X 255.255.255.0
> !
>
> !
> interface Serial0/2/0:15
>  no ip address
>  encapsulation hdlc
>  isdn switch-type primary-qsig
>  isdn incoming-voice voice
>  isdn bind-l3 ccm-manager
>  no cdp enable
> !
> ip forward-protocol nd
> !
> !
> ip http server
> ip http access-class 23
> ip http authentication local
> ip http secure-server
> ip http timeout-policy idle 60 life 86400 requests 1
> ip route 0.0.0.0 0.0.0.0 X.X.X.X
> !
>
> !
> !
> control-plane
> !
> call threshold global cpu-5sec low 70 high 85
> !
> voice-port 0/2/0:15
> !
> voice-port 0/3/0
> !
> voice-port 0/3/1
> !
> ccm-manager switchback immediate
> ccm-manager fallback-mgcp
> ccm-manager redundant-host X.X.x.x
> ccm-manager mgcp
> ccm-manager config server x.x.x.x
> !
> mgcp
> mgcp call-agent x.x.x.x service-type mgcp version 1.0
> mgcp bind control source-interface GigabitEthernet0/0.1
> mgcp bind media source-interface GigabitEthernet0/0.1
> !
> mgcp profile default
> !
> !
> gateway
>  timer receive-rtp 1200
> !
> sip-ua
> retry invite 1
>  retry bye 1
>  retry cancel 1
>  timers expires 6
>  reason-header override
> !
> !
> telephony-service
>  max-conferences 12 gain -6
>  transfer-system full-consult
> !
>
>
> thanks
>
>  --
> From: gogli...@gmail.com
> Date: Mon, 16 Jan 2012 13:29:49 +0100
>
> Subject: Re: [OSL | CCIE_Voice] MGCP Registration
> To: mercy_for_...@hotmail.com
> CC: ccie_voice@onlinestudylist.com
>
>
> Hi,
>
> The card is supported:
> http://www.cisco.com/en/US/prod/collateral/routers/ps259/product_data_sheet0900aecd8057f2e0.pdf
>
> Paste your voice gateway configuration, there must be a problem somewhere.
>
> Cheers,
>
> On Sun, Jan 15, 2012 at 8:45 PM, mercy forall 
> wrote:
>
>
> Hi,
>
> I think the issue from card , i need to sure the CUCM support
> VWIC-1MFT-G703  card or not , because i did not fine it in cucm
> mgcp configuration and i use VWIC-1MFT-E1
>
> i made all configuration , and made same configuration on other and it is
> work fine , from where i can check if is it support tor not ?
>
>  > From: ccie_voice-requ...@onlinestudylist.com
> > Subject: CCIE_Voice Digest, Vol 71, Issue 54
> > To: ccie_voice@onlinestudylist.com
> > Date: Sun, 15 Jan 2012 06:01:19 -0500
> >
> > Send CCIE_Voice mailing list submissions to
> > ccie_voice@onlinestudylist.com
> >
&

Re: [OSL | CCIE_Voice] Privacy & Calling Number

[datucha123 datucha123]

Calling Number will be visible to all other Sharing users, until you answer
the call. As soon as you will pickup the call, the other IP phone users wil
not be able to see active call on you phone any more.

On Sun, Jan 15, 2012 at 5:49 PM, Ken Wyan  wrote:

> According to cisco docs , when Privacy Button is pressed , calling number
> & time should not be visible to other phones  sharing my number.
>
> My phone is registered to CUCM 7.
>
> I have a line shared with another phone.
>
> If I answer a call ringing on shared line , others can barge into my call.
>
> But if I press Privacy button after answering call , others cannot barge
> into my call. But still calling number is visible to other phone sharing
> this number.
>
> What can be the reason?
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
___
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Re: [OSL | CCIE_Voice] MGCP Registration

[datucha123 datucha123]

first of all make sure that the L2 is up for the PRI, otherwise the MGCP
gateway in CUCM will show unregistere status.

On Sun, Jan 15, 2012 at 12:16 PM, mercy forall wrote:

>  Hi,
>
>
> i have 3845 with VWIC-1MFT-G703
>
> i configured mgcp but it is bending in registering on sh ccm-manager
>
> i did not find VWIC-1MFT-G703 on CUCM 7 , i use VWIC-1MFT-E1 ,
>
> Are this card supported in mgcp ?
>
> thanks
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
___
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Re: [OSL | CCIE_Voice] AAR for UCCX

[datucha123 datucha123]

I have not testeed AAR with EM, but for CTI RP I have made it and it works.

Maybe I made the wrong test with CTI RP, but the fact is that I could call
the UCCX IVR from Branch 1 IP phones when there was no enough bandwidth on
the WAN link, and the call to UCCX was routed through PSTN.

On Sun, Jan 15, 2012 at 2:24 AM, Mohammed Al Baqari <
baqari.voic...@gmail.com> wrote:

>  Hi,
>
> ** **
>
> I haven’t tested this. But my info is based on CUCM SRND. They haven’t
> given any exception on AAR and CTI inter-operability. Can you please
> highlight further.
>
> ** **
>
> “AAR does not support CTI route points as the origin or the destination
> of calls. Also, AAR is incompatible with the Extension Mobility feature
> when users roam across different sites.”
>
> ** **
>
> Regards,
>
> Mohammed Al Baqari
>
> ** **
>
> *From:* datucha123 datucha123 [mailto:datucha...@gmail.com]
> *Sent:* Sunday, January 15, 2012 1:03 AM
> *To:* Mohd Baqari
> *Cc:* ccie_voice@onlinestudylist.com
> *Subject:* Re: [OSL | CCIE_Voice] AAR for UCCX
>
> ** **
>
> AAR is supported with CTI, when it is registered. For instance the CTI
> Route Point supports AAR, when it is registered. But as soon as it will
> unregister (or some Dummy CTI Route Point, which never gets registered)
> does not support AAR.
>
>  
>
>
>  
>
> On Sat, Jan 14, 2012 at 10:29 PM, Mohd Baqari 
> wrote:
>
> Hi Datucha,
>
> In your scenario whats confusing me is how aar is triggered. As far as I
> know AAR isn't supported with CTI
>
> Regards,
> Mohammed Al Baqari
>
> Sent from my iPhone
>
>
> On Jan 8, 2012, at 9:11 PM, datucha123 datucha123 
> wrote:
>
> > Hello,
> >
> > I have been making a test and find out the following issue:
> >
> > So, I set the AAR Mask to CTI Route Points Line Settings, and the calls
> from Branch1 to UCCX when there was no WAN bandwidth available were routed
> through PSTN successfully.
> >
> > I have not set any AAR Group and AAR CSS for CTI ports, but the HQ
> gateway has them.
> >
> > So when I called the UCCX IVR (simpla AA script) through HQ Gateway and
> dialed the Branch 1 Phones extensions, the call has been routed through
> PSTN based on CAC restrictions. Well that's ok with CAC. But I cannot
> understand how the AAR got worked, when neither CTI ports nor the CTI Route
> Point had any AAR configuration (AAR Mask has been also removed from the
> CTI RP Line).
> > HQ Gateway had the AAR Group and CSS set. So based on that, UCCX took
> the HQ gateway setttings of AAR and made a call?
> >
> >
> >
> >
> >
>
> > ___
> > For more information regarding industry leading CCIE Lab training,
> please visit www.ipexpert.com
> >
> > Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com <http://www.platinumplacement.com/>
>
> ** **
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] incoming called-number .

[datucha123 datucha123]

Outgoing dial-peer matching:

The only way to match outgoing dial-peer is destination-pattern.

Incoming dial-peer matching:

1)incoming called-number
2)answer address
3) destination-pattern (matches ANI)
4) Incoming Port (only for Pots dial-peers)
5) Default Dial-peer

now about your question Ken  -  you are right, it is possible to make a
partial match. To overcome it, use the ^ and $ signs to indicate the start
and end of the string, so that there would not be partial match any more.

Mercy, dial-peer 1 will be matched, as the router first tries to match the
dial-peer based on the incoming called number, and as soon as it matches,
the further matching is stopped. In your case . (dot) will be matched for
dial-peer 1, so that dial-peer 2 will not be even considered.


On Sat, Jan 14, 2012 at 11:37 PM, mercy forall wrote:

>  if there are 2 dial peer for incoming mach
>
> dial-peer voice 1 pots
>  incoming called-number .
>
> and
>
>
> dial-peer voice 2 pots
>
>  *answer-address* 12345
>
>
>
> and the ani is 12345
>
>
> which dial peer will mach ??
>
>
>
>
>  > From: ccie_voice-requ...@onlinestudylist.com
> > Subject: CCIE_Voice Digest, Vol 71, Issue 50
> > To: ccie_voice@onlinestudylist.com
> > Date: Fri, 13 Jan 2012 20:49:33 -0500
> >
> > Send CCIE_Voice mailing list submissions to
> > ccie_voice@onlinestudylist.com
> >
> > To subscribe or unsubscribe via the World Wide Web, visit
> > http://onlinestudylist.com/mailman/listinfo/ccie_voice
> > or, via email, send a message with subject or body 'help' to
> > ccie_voice-requ...@onlinestudylist.com
> >
> > You can reach the person managing the list at
> > ccie_voice-ow...@onlinestudylist.com
> >
> > When replying, please edit your Subject line so it is more specific
> > than "Re: Contents of CCIE_Voice digest..."
> >
> >
> > Today's Topics:
> >
> > 1. CUE with CUCM (datucha123 datucha123)
> > 2. Re: RSVP CAC (Ashraf Ayyash)
> > 3. Re: incoming called-number . (Bill Lake)
> > 4. Re: Caller ID Update (Mohd Baqari)
> > 5. vol 2 lab 7 ---uccx script (donny f)
> >
> >
> > --
> >
> > Message: 1
> > Date: Fri, 13 Jan 2012 21:12:34 +0400
> > From: datucha123 datucha123 
> > To: ccie_voice@onlinestudylist.com
> > Subject: [OSL | CCIE_Voice] CUE with CUCM
> > Message-ID:
> > 
> > Content-Type: text/plain; charset="iso-8859-1"
> >
> > Hello,
> >
> > Do we need to configure the CUCME-SRST (or call-manager-fallback) as the
> > Third Callmanager in JTAPI Subsystem configuration in CUE for SRST
> fallback?
> >
> > I think we do not need to configure the CUCME-SRST as the Third
> > Callmanager, as CUCME-SRST does not use the JTAPI, but is integrated with
> > SIP, and we have to configure the SIP Subsystem for that. Right?
> > -- next part --
> > An HTML attachment was scrubbed...
> > URL:
> 
> >
> > --
> >
> > Message: 2
> > Date: Fri, 13 Jan 2012 12:00:54 -0600
> > From: Ashraf Ayyash 
> > To: study buddy 
> > Cc: ccie_voice@onlinestudylist.com
> > Subject: Re: [OSL | CCIE_Voice] RSVP CAC
> > Message-ID:
> > 
> > Content-Type: text/plain; charset=ISO-8859-1
> >
> > did you run qos on sc as well ? you know that you have to otherwise
> > you will kill sc sub interface , do show traff and you will see it
> > getting 56 K only , so the solution is to run the QOS on both sub
> > interfaces on HQ and on sc as well
> >
> > Ash
> >
> > On Fri, Jan 13, 2012 at 12:09 AM, study buddy 
> wrote:
> > > Hi All,
> > >
> > > For the Lab topology, if I run QoS between HQ & SB & RSVP CAC between
> HQ &
> > > SC & set the ip rsvp band to 112 for 4 calls, I can actually make only
> two
> > > calls the third calls get re-routed. Now if I run qos between HQ & SC
> > > routers as well with a BW of 1536 I can make 3 calls, but the 4th call
> still
> > > fails & I get the following RSVP error
> > >
> > > *Jan 10 12:58:29.883:?? QoS Module: RESV ERROR received : Remote IP:
> > > 142.102.64.254 | Local IP: 142.102.66.254
> > >
> > > *Jan 10 12:58:29.883: qos_rsvp_resv_notify_events: errCode 1, errVal 2,
> > > errFlag 0, errNode 10.10.112.1
> > >
> > > *Jan 10 12:58:29.883: qos_rsvp_remove_reservation:? Removing RESV
> state for
> > > CallId? : 0xFFC5
> > 

Re: [OSL | CCIE_Voice] AAR for UCCX

[datucha123 datucha123]

AAR is supported with CTI, when it is registered. For instance the CTI
Route Point supports AAR, when it is registered. But as soon as it will
unregister (or some Dummy CTI Route Point, which never gets registered)
does not support AAR.



On Sat, Jan 14, 2012 at 10:29 PM, Mohd Baqari wrote:

> Hi Datucha,
>
> In your scenario whats confusing me is how aar is triggered. As far as I
> know AAR isn't supported with CTI
>
> Regards,
> Mohammed Al Baqari
>
> Sent from my iPhone
>
> On Jan 8, 2012, at 9:11 PM, datucha123 datucha123 
> wrote:
>
> > Hello,
> >
> > I have been making a test and find out the following issue:
> >
> > So, I set the AAR Mask to CTI Route Points Line Settings, and the calls
> from Branch1 to UCCX when there was no WAN bandwidth available were routed
> through PSTN successfully.
> >
> > I have not set any AAR Group and AAR CSS for CTI ports, but the HQ
> gateway has them.
> >
> > So when I called the UCCX IVR (simpla AA script) through HQ Gateway and
> dialed the Branch 1 Phones extensions, the call has been routed through
> PSTN based on CAC restrictions. Well that's ok with CAC. But I cannot
> understand how the AAR got worked, when neither CTI ports nor the CTI Route
> Point had any AAR configuration (AAR Mask has been also removed from the
> CTI RP Line).
> > HQ Gateway had the AAR Group and CSS set. So based on that, UCCX took
> the HQ gateway setttings of AAR and made a call?
> >
> >
> >
> >
> >
> > ___
> > For more information regarding industry leading CCIE Lab training,
> please visit www.ipexpert.com
> >
> > Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com <http://www.platinumplacement.com/>
>
___
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www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] lab 4a shouldn't take too much time, right?

[datucha123 datucha123]

First of all you can look at the CUCM Traces filtered by particular Phone.

Go to Trace, then enable Device Based Tracing and choose the necessary
Phone/device

After ssh into CUCM Server and issue the command "file tail activelog
/cm/trace/ccm/sdi recent",  this will give you the real time traces.



On Sat, Jan 14, 2012 at 5:41 PM, Juan Lopez
wrote:

> someone of you guys/girls walked into following weird behaviour ? I for
> sure wasted too much time with lab 4a, 2 remote rack days :(( , where a SIP
> HQ phone (5002) is supposed to dial 3...@ipxcme.com (SIP CME phone)
>
> The issue was when picking up the call ath the CME phone, it disconnected.
> So a codec issue. Per lab 4a, the call at HQ phone leaves the trunk, both
> are in HQ device pool.
>
> Inbound dialpeer at CME has voice-class codec with g711 and g729.
>
> Registered SIP CME phone has g711 configured on voice register pool.
>
> So call should negotiate to g711: supported by the region configuration at
> UCM, on the inbound call leg and the final CME phone.
>
> But it fails.
>
> Putting g729 on voice register pool > call succeeds as g729... > So in
> g729 it works > Suspicion: is the correct inbound dialpeer matched? YES,
> the one with the voice-class codec. So that cannot be the cause why the
> call MUST be g729.
>
> What is? debug ccsip mess is not very descriptive: the CME router sends
> BYE to the UCM, with *some* cause code, when the CME phone is configured to
> have g711. Not familiar zith UCM traces unfortunately (hopefully this will
> improve, quickly...)
>
> After a *very long* journey in the dark, I reset the UCM SIP phone.
> Problem automagically gone :-(
>
> Question:
>
> - does this happen often with sip phones?
>
> - what trace should I have looked into on the UCM, to find more UCM info
> that the phone ONLY tried to negotiate the g729? Can someone point me in a
> good direction what trace to enable, and what text files to investigate?
> Is there some good info/examples available on this topic :S ?
>
> thanks
> /juan
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
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[OSL | CCIE_Voice] CUE with CUCM

[datucha123 datucha123]

Hello,

Do we need to configure the CUCME-SRST (or call-manager-fallback) as the
Third Callmanager in JTAPI Subsystem configuration in CUE for SRST fallback?

I think we do not need to configure the CUCME-SRST as the Third
Callmanager, as CUCME-SRST does not use the JTAPI, but is integrated with
SIP, and we have to configure the SIP Subsystem for that. Right?
___
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Re: [OSL | CCIE_Voice] incoming called-number .

[datucha123 datucha123]

Incoming called number matching is done based on more specific pattern
(Incoming called-number ).

So it depends what Number is received as the Called Number.

2012/1/13 "Farkas Péter" 

> Good to know:
>
> "When the Cisco IOS router or gateway receives a call setup request, a
> dial peer match is made for the incoming call in order to facilitate
> routing the call to different session applications. This is not a
> digit-by-digit match, rather the full digit string received in the setup
> request is used to match against configured dial peers."
>
> So I vote dp 2.
>
> Peter
>
> - Original Message -
> From: Ken Wyan 
> Date: Friday, January 13, 2012 3:10 pm
> Subject: Re: [OSL | CCIE_Voice] incoming called-number .
> To: Emanuel Damasceno 
> Cc: ccie_voice@onlinestudylist.com
>
>
> > If we had 2 incoming dial-peers with same incoming called-number , I
> think
> >  selection should be random.
> >
> >  Anybody found a cisco document for this?
> >
> >  On Fri, Jan 13, 2012 at 7:05 PM, Emanuel Damasceno <
> aedamasc...@gmail.com>wrote:
> >
> >  > Right. =)
> >  >
> >  > *Emanuel Damasceno*
> >  > CCNP Voice
> >  >
> >  >
> >  >
> >  >
> >  >
> >  >
> >  > On Fri, Jan 13, 2012 at 10:41 AM, datucha123 datucha123 <
> >  > datucha...@gmail.com> wrote:
> >  >
> >  >> First of all Preference command works only for Outgoing calls. It
> does
> >  >> not make any sense for incoming dial-peer matching.
> >  >>
> >  >> also in that particular case, the preference command will not make
> any
> >  >> sense, because those dial-peers are having a differenct destination
> >  >> patterns.
> >  >>
> >  >>  On Fri, Jan 13, 2012 at 4:31 PM, Emanuel Damasceno <
> >  >> aedamasc...@gmail.com> wrote:
> >  >>
> >  >>> Wouldn't the command "preference X" work in that situation?
> >  >>>
> >  >>> *Emanuel Damasceno*
>  >  >>> CCNP Voice
> >  >>>
> >  >>>
> >  >>>
> >  >>>
> >  >>>
> >  >>>
> >  >>> On Fri, Jan 13, 2012 at 6:36 AM, datucha123 datucha123 <
> >  >>> datucha...@gmail.com> wrote:
> >  >>>
> >  >>>> For incoming calls, the 23548 is more specific match for Called
> Number
> >  >>>> of 235482345 then 235. And that is why the Dial-peer 2 is matched.
> >  >>>>
> >  >>>> For outgoing calls, if you place a call to the same number
> (235482345)
> >  >>>> and the destination patterns are the same (235 and 23548) then the
> >  >>>> dial-peer 2 would be matched as again, it is more specific match
> and if the
> >  >>>> Enblock is used. If you will use Overlap Dialing, (UCME for
> instance) then
> >  >>>> dial-peer 1 would be matched
> >  >>>>
> >  >>>>  On Fri, Jan 13, 2012 at 10:58 AM, Ken Wyan 
> wrote:
> >  >>>>
> >  >>>>>  Is there a cisco doc for incoming called-number selection order?
> >  >>>>>
> >  >>>>> What I mean is say
> >  >>>>>
> >  >>>>> dial-peer voice 1
> >  >>>>> incoming called-number  235
> >  >>>>> xxx
> >  >>>>>
> >  >>>>>
> >  >>>>> dial-peer voice 2
> >  >>>>> incoming called-number 23548
> >  >>>>> xx
> >  >>>>>
> >  >>>>> If a call arrives with DNIS  235482345 which incoming dial-peer
> will
> >  >>>>> be matched?
> >  >>>>>
> >  >>>>> As per my testing it seems dial-peer 2 is selected.
> >  >>>>>
> >  >>>>> If this was an outgoing dial-peer (with default dial-peer hunt 0)
> ,
> >  >>>>> dial-peer 1 should be the match & this is well documented.
> >  >>>>>
> >  >>>>> For incoming calls ??
> >  >>>>>
> >  >>>>> ___
> >  >>>>> For more information regarding industry leading CCIE Lab training,
> >  >>>>> please visit www.ipexpert.com
> >  >>>>>
> >  >>>>> Are you a CCNP or CCIE and looking for a job? Check out
> >  >>>>> www.PlatinumPlacement.com <http://www.platinumplacement.com/> <>
> >  >>>>>
> >  >>>>
> >  >>>>
> >  >>>> ___
> >  >>>> For more information regarding industry leading CCIE Lab training,
> >  >>>> please visit www.ipexpert.com
> >  >>>>
> >  >>>> Are you a CCNP or CCIE and looking for a job? Check out
> >  >>>> www.PlatinumPlacement.com <http://www.platinumplacement.com/> <>
> >  >>>>
> >  >>>
> >  >>>
> >  >>
> >  >
> > ___
> >  For more information regarding industry leading CCIE Lab training,
> please visit www.ipexpert.com
> >
> >  Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com <http://www.platinumplacement.com/>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com <http://www.platinumplacement.com/>
>
___
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Re: [OSL | CCIE_Voice] Preference within policy-map

[datucha123 datucha123]

That is a great question.

Based on my knowledge, the Set DSCP command is executed first, because
otherwise the exceeding traffic will become EF.

You can also refer to this Linke, where the Auto QoS is duscussed:

http://ieoc.com/forums/t/17680.aspx?PageIndex=1

On Fri, Jan 13, 2012 at 5:47 PM, Ken Wyan  wrote:

> When I want to mark all voip packets to ef & mark exceeding packets to 8 I
> do following in a Catalyst Switch.
>
> policy-map AutoQos-Policy-Untrust
>  class AutoQos-VoIP-RTP-Untrust
>police 128 exceed-action policed-dscp-transmit
>set dscp ef
>
> mls qos map policed-dscp  46 to 8
>
>
> Here will the set dscp ef command will be executed before police command?
> Otherwise exceeding packets (which are remarked to 8 by police statement )
> can again marked with dscp ef.
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
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Re: [OSL | CCIE_Voice] incoming called-number .

[datucha123 datucha123]

First of all Preference command works only for Outgoing calls. It does not
make any sense for incoming dial-peer matching.

also in that particular case, the preference command will not make any
sense, because those dial-peers are having a differenct destination
patterns.

On Fri, Jan 13, 2012 at 4:31 PM, Emanuel Damasceno wrote:

> Wouldn't the command "preference X" work in that situation?
>
> *Emanuel Damasceno*
> CCNP Voice
>
>
>
>
>
>
> On Fri, Jan 13, 2012 at 6:36 AM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>> For incoming calls, the 23548 is more specific match for Called Number of
>> 235482345 then 235. And that is why the Dial-peer 2 is matched.
>>
>> For outgoing calls, if you place a call to the same number (235482345)
>> and the destination patterns are the same (235 and 23548) then the
>> dial-peer 2 would be matched as again, it is more specific match and if the
>> Enblock is used. If you will use Overlap Dialing, (UCME for instance) then
>> dial-peer 1 would be matched
>>
>>  On Fri, Jan 13, 2012 at 10:58 AM, Ken Wyan  wrote:
>>
>>>  Is there a cisco doc for incoming called-number selection order?
>>>
>>> What I mean is say
>>>
>>> dial-peer voice 1
>>> incoming called-number  235
>>> xxx
>>>
>>>
>>> dial-peer voice 2
>>> incoming called-number 23548
>>> xx
>>>
>>> If a call arrives with DNIS  235482345 which incoming dial-peer will be
>>> matched?
>>>
>>> As per my testing it seems dial-peer 2 is selected.
>>>
>>> If this was an outgoing dial-peer (with default dial-peer hunt 0) ,
>>> dial-peer 1 should be the match & this is well documented.
>>>
>>> For incoming calls ??
>>>
>>> ___
>>> For more information regarding industry leading CCIE Lab training,
>>> please visit www.ipexpert.com
>>>
>>> Are you a CCNP or CCIE and looking for a job? Check out
>>> www.PlatinumPlacement.com <http://www.platinumplacement.com/>
>>>
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com <http://www.platinumplacement.com/>
>>
>
>
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[OSL | CCIE_Voice] CUE MWI types

[datucha123 datucha123]

Hello,

Please correct me if I am wrong.

CUE has three types for MWI notification:

1) Outcall
2) Sub-notify (Subscribe)
3) Unsolicited

CME does support all three MWI methods:

When using Subscribe option in CME, we have to configure the "mwi sip"
commands on SCCP Ephone-DNs, and "MWI" on SIP Voice Register DNs.

When using Unsolicited option in CME, we do not have to configure the "mwi
sip" command for SCCP Phones any more, while the SIP Voice Register DN
still needs the "MWI" command.

So, the CME SCCP Phones need the "MWI SIP" command only when using
Subscriber option (NOT FOR UNSOLICITED). And the SIP Phones needs the "MWI"
command with both types - Subscribe and Unsolicited.



Now as for SRST:

SIP phones in SRST does not support MWI at all -  There is a bug in IOS
version for the exam.

SCCP phones in either CME-SRST or Traditional SRST DO NOT needs "mwi sip"
command, even for Subscribe (Call-manager-fallback does not support
ephone-dn command).

We have to configure the SIP-UA with "mwi-server" command (also in case of
CME) and it should work

please correct me if I am wrong.

Thank you very much.
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Re: [OSL | CCIE_Voice] incoming called-number .

[datucha123 datucha123]

For incoming calls, the 23548 is more specific match for Called Number of
235482345 then 235. And that is why the Dial-peer 2 is matched.

For outgoing calls, if you place a call to the same number (235482345) and
the destination patterns are the same (235 and 23548) then the dial-peer 2
would be matched as again, it is more specific match and if the Enblock is
used. If you will use Overlap Dialing, (UCME for instance) then dial-peer 1
would be matched

On Fri, Jan 13, 2012 at 10:58 AM, Ken Wyan  wrote:

> Is there a cisco doc for incoming called-number selection order?
>
> What I mean is say
>
> dial-peer voice 1
> incoming called-number  235
> xxx
>
>
> dial-peer voice 2
> incoming called-number 23548
> xx
>
> If a call arrives with DNIS  235482345 which incoming dial-peer will be
> matched?
>
> As per my testing it seems dial-peer 2 is selected.
>
> If this was an outgoing dial-peer (with default dial-peer hunt 0) ,
> dial-peer 1 should be the match & this is well documented.
>
> For incoming calls ??
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
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[OSL | CCIE_Voice] SRST Auto Provision

[datucha123 datucha123]

Hello,

Can you please confirm the following?

When Using Auto Provision DN (for CME SRST), then we can manually create
the DNs, so that the CME will try to match the manually created DN with
learned DN, and if it matches, then assing to that Ephone.
When using Auto Provision ALL, then we can manually create DN's and Ephone
also, and in that case, the CME will try to match the manually created
Ephones and Ephone-dn with learned ones.
And when using NONE option, then we cannot create DNs or Ephone manually,
and the CME will not make any matching, and will create learned DNs and
Ephones.
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[OSL | CCIE_Voice] Phone Label

[datucha123 datucha123]

Hello,

I have noticed, that the Phone Line Label is not passed with SCCP Phone,
when it goes to SRST.

But when the SIP Phones goes to SRST, then the Line Label is passed on over
the SRST.

Is it normal?
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Re: [OSL | CCIE_Voice] RSVP and Multicast MoH

[datucha123 datucha123]

I have tested today, and Boris is right. The multicast MoH can travers
through RSVP MTPs fine.

While Mutlicast does not work for H323 and SIP gateway, when they are using
MTP

On Thu, Jan 12, 2012 at 12:14 AM, datucha123 datucha123 <
datucha...@gmail.com> wrote:

> I will test that tomorrow. Because the MMoH does not work for Gateways
> (H323 and SIP) when MTP is checked for those Gateways/Trunks (Multicast
> cannot traverse the MTP device, it cannot be teared down and re-originated)
>
>
> On Wed, Jan 11, 2012 at 11:43 PM, Boris  wrote:
>
>> AFAIK MMoH will work together with RSVP because it is not taken into
>> consideration for CAC. The multicast streams flow around MTP/RSVP
>> reservations, but you need to cater for this traffic in priority queue
>> towards the stream destination.
>>
>> Sent from my mobile device, sorry for typos.
>> -------
>> Regards
>> Boris
>>
>> On 12/01/2012, at 5:31, datucha123 datucha123 
>> wrote:
>>
>> > Hello,
>> >
>> > When using RSVP CAC between HQ and Branch 1 Sites, the Multicast MoH
>> should not work for Branch 1 IP Phones from UCM Server, as the Muticast
>> cannot traverse the MTP, right?
>> >
>> > But when sourcing Mutlicast MoH from the Branch Router, it will work.
>> >
>> >
>> > ___
>> > For more information regarding industry leading CCIE Lab training,
>> please visit www.ipexpert.com
>> >
>> > Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com <http://www.platinumplacement.com/>
>>
>
>
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Re: [OSL | CCIE_Voice] Local Route Groups

[datucha123 datucha123]

No it is not bound to one Route List, it can be used by any Route List in
the system



On Thu, Jan 12, 2012 at 4:21 PM, Spence, Paul wrote:

>  Ok that’s good to know I though it may have been bound to one RL only
>
> Many Thanks
>
> ** **
>
> *From:* datucha123 datucha123 [mailto:datucha...@gmail.com]
> *Sent:* 12 January 2012 12:10
> *To:* Spence, Paul
> *Cc:* ccie_voice@onlinestudylist.com
> *Subject:* Re: [OSL | CCIE_Voice] Local Route Groups
>
> ** **
>
> Different Route Lists may use the LRG
>
> On Thu, Jan 12, 2012 at 4:26 AM, Spence, Paul 
> wrote:
>
> Guys,
> Need a little advise on LRG, I have it set up in a live environment and
> associated with a CUCM group. We recently added a new subscriber to a
> remote site and wanted to include it the LRG set up, but that meant adding
> the subscriber to the CUCM group at the cluster site and as such the
> devices would not register with the local server.  Took the remote site out
> of the LRG set up and built its own set of route patterns etc.
> My question is and not had the time to test in a lab can I use the
> Standard local Route Group in more than one RL and therefore have another
> set of common dial patterns.
> E.g RL1 uses standard local route group and CUCM group1 and RL2 uses
> standard local route group and CUCM group 2
>
> Thanks
> Paul
>
>
> Sent from mobile device, grammar mistakes are purely accidental.
>
> NOTICE - This communication may contain confidential and privileged
> information that is for the sole use of the intended recipient. Any
> viewing, copying or distribution of, or reliance on this message by
> unintended recipients is strictly prohibited. If you have received this
> message in error, please notify us immediately by replying to the message
> and deleting it from your computer.
>
> Jacobs Engineering U.K. Limited
> 1180 Eskdale Road, Winnersh, Wokingham RG41 5TU
> Registered in England and Wales under number 2594504
> ___
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> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
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>
> ** **
>
> --
> NOTICE - This communication may contain confidential and privileged
> information that is for the sole use of the intended recipient. Any
> viewing, copying or distribution of, or reliance on this message by
> unintended recipients is strictly prohibited. If you have received this
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>
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Re: [OSL | CCIE_Voice] Local Route Groups

[datucha123 datucha123]

Different Route Lists may use the LRG

On Thu, Jan 12, 2012 at 4:26 AM, Spence, Paul wrote:

> Guys,
> Need a little advise on LRG, I have it set up in a live environment and
> associated with a CUCM group. We recently added a new subscriber to a
> remote site and wanted to include it the LRG set up, but that meant adding
> the subscriber to the CUCM group at the cluster site and as such the
> devices would not register with the local server.  Took the remote site out
> of the LRG set up and built its own set of route patterns etc.
> My question is and not had the time to test in a lab can I use the
> Standard local Route Group in more than one RL and therefore have another
> set of common dial patterns.
> E.g RL1 uses standard local route group and CUCM group1 and RL2 uses
> standard local route group and CUCM group 2
>
> Thanks
> Paul
>
>
> Sent from mobile device, grammar mistakes are purely accidental.
>
> NOTICE - This communication may contain confidential and privileged
> information that is for the sole use of the intended recipient. Any
> viewing, copying or distribution of, or reliance on this message by
> unintended recipients is strictly prohibited. If you have received this
> message in error, please notify us immediately by replying to the message
> and deleting it from your computer.
>
> Jacobs Engineering U.K. Limited
> 1180 Eskdale Road, Winnersh, Wokingham RG41 5TU
> Registered in England and Wales under number 2594504
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
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