[OSL | CCIE_Voice] access to cli in lab

2013-07-08 Thread Ajay Viswanath
Hi,

Do we have access to the server cli in the lab. Its vital for troubleshooting 
dbissues.

Thanks
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Re: [OSL | CCIE_Voice] access to cli in lab

2013-07-08 Thread jainpiyush2022
Yes we have..

Thanks
Piyush Jain

Sent from my android device.



-Original Message-
From: Ajay Viswanath 
To: ccie_voice@onlinestudylist.com
Sent: Mon, 08 Jul 2013 4:50 PM
Subject: [OSL | CCIE_Voice] access to cli in lab

Hi,

Do we have access to the server cli in the lab. Its vital for troubleshooting 
dbissues.

Thanks
___
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___
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[OSL | CCIE_Voice] Connection Monitor Duration

2013-07-08 Thread myfriend myfriend
Hi Guys

I would like to know if we can reduce the connection monitor
Duration for a site to 30 Or 40 sec so that SRST revert back soon.
Is that safe to do in the lab or they would cut mark for that

Thanks
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Re: [OSL | CCIE_Voice] Cue call failed in srst mode

2013-07-08 Thread Karen Johnson
yes thanks, i missed that and now work




From: Rrcrumm 
To: Karen Johnson  
Cc: sanity insanity ; 
"ccie_voice@onlinestudylist.com"  
Sent: Monday, July 8, 2013 12:10:45 AM
Subject: Re: [OSL | CCIE_Voice] Cue call failed in srst mode

Do you have a sip trigger?
On Jul 7, 2013, at 9:21 PM, Karen Johnson  wrote:
dial-peer yes, trigger mean ?
>
>
>
>
>
>From: Rrcrumm 
>To: Karen Johnson  
>Cc: sanity insanity ; 
>"ccie_voice@onlinestudylist.com"  
>Sent: Sunday, July 7, 2013 8:08:42 PM
>Subject: Re: [OSL | CCIE_Voice] Cue call failed in srst mode
>
>Dial peer and trigger
>On Jul 7, 2013, at 4:22 PM, Karen Johnson  wrote:
>Yes 
>>
>>
>>From: Rrcrumm ; To: Karen Johnson 
>>; Cc: sanity insanity 
>>; ccie_voice@onlinestudylist.com 
>>; Subject: Re: [OSL | CCIE_Voice] Cue call 
>>failed in srst mode Sent: Sun, Jul 7, 2013 6:49:41 PM 
>>Do you have the voicemail command under telephony-service?
>>Hath
>>Rc
>>On Jul 7, 2013, at 11:04 AM, Karen Johnson  wrote:
>>Hi folks, My cue is working fine in normal mode. If i switch to srst mode it 
>>failed and busy tone. Dialpeer and codec g711 , sip-ua ip show correct. any 
>>idea what to tshoot and command to check? K 
>>>
>>>
>>>From: sanity insanity ; To: 
>>>ccie_voice@onlinestudylist.com ; Subject: 
>>>[OSL | CCIE_Voice] CME srst best practices.. Sent: Sat, Jul 6, 2013 4:29:43 
>>>PM 
>>>hi Guys,In srst I use the following config... telephony-service srst mode 
>>>auto-provision all srst dn line-mode octo1)Do I also need to configure srst 
>>>dn template & srst ephone template ?2)what are best practices for setting up 
>>>the cue in srst mode ? If possible include details of..-mwi-Xfer to VM - 
>>>button 6 of phone 1 is pressed-MJ 
>>___For more information regarding 
>>industry leading CCIE Lab training, please visit http://www.ipexpert.com/Are 
>>you a CCNP or CCIE and looking for a job? Check out 
>>http://www.platinumplacement.com/ ___
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[OSL | CCIE_Voice] (no subject)

2013-07-08 Thread Dharambirku
Hi
i am going for ccie voice lab in November.
can anyone guide if i get ccie number then after 14th Feb 2014 ,my ccie no will 
be valid or not


Sent from Samsung Mobile___
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Re: [OSL | CCIE_Voice] (no subject)

2013-07-08 Thread Bill Lake
No you will still be a CCIE 

You will just be in a retired track

You will be able to upgrade to CCIE collaboration by passing the CCIE 
collaboration written exam

Good luck on you lab

Sent from my iPhone

On Jul 8, 2013, at 12:22 PM, Dharambirku  wrote:

> Hi
> i am going for ccie voice lab in November.
> can anyone guide if i get ccie number then after 14th Feb 2014 ,my ccie no 
> will be valid or not
> 
> 
> Sent from Samsung Mobile
> ___
> 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] CUPC client to play voicemail

2013-07-08 Thread ccie2k12
check server health in CUPC, to make voice mail is showing green

also have you entered the voice mail user id and password in preferences

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Drake J
Sent: Monday, July 08, 2013 9:46 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] CUPC client to play voicemail

 

hi Guys,

How do we get our CUPC client to play voicemail and display it on the client
as a red Icon?

I tried many times but not see this icon.

-Drake

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[OSL | CCIE_Voice] Meetme Call Handler

2013-07-08 Thread Karen Johnson
Folks,
 
Can we achieve this with One  call handler only ?  
 "Meet Me When joins meet me conference it should announce the participant name 
and when somebody dials meet me number it should ask "who may I say its 
calling".  and when Meetme not avail , go to HQ1 voicemail"
 
Each time I try to use one call handler, Message Setting always override the 
Transfer rule, and as result call always go to HQ1 vm without asking me for 
"Who May I say is calling".
 
tks
K___
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[OSL | CCIE_Voice] H323 Fast Start

2013-07-08 Thread Barrera, Hugo
Guy's,

I used g711 for the codec, on mtp resource, but a peer of mine is stating I 
should have used g729 and by using this I would have to add a transcoder on the 
GW as well. Any thoughts?

Hugo

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Re: [OSL | CCIE_Voice] H323 Fast Start

2013-07-08 Thread Edgar Feliz
I may be wrong but I do not recall g729 being a requirement. for that task,
they are not asking HQ > SB, which would be inter-region, just F/S between
CM > R2 = SB phones dial PSTN.


Edgar


On Mon, Jul 8, 2013 at 8:03 PM, Barrera, Hugo wrote:

>  Guy’s,
>
> ** **
>
> I used g711 for the codec, on mtp resource, but a peer of mine is stating
> I should have used g729 and by using this I would have to add a transcoder
> on the GW as well. Any thoughts? 
>
> ** **
>
> *Hugo *
>
> ** **
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
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[OSL | CCIE_Voice] AAR and uccx

2013-07-08 Thread Ramcharan Arya
Hi,
I have a CTI Route point 4000 and two CTI port 410x and 410y



SiteC Phone 1 and SC Phone 2 are in CSQ which is assign to application and
associated with trigger.

Due to  RSVP when call exceed ip rsvp bandwidth call  from uccx to site
Phones should use AAR and to over PSTN.

My doubts are .

What should be local and device pool of CTI ports so it should work in AAR
when PSTN caller make call to CTI route point number 4000.

Can someone please advice about this.

Thanks,
Ramcharan Arya
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