Re: [OSL | CCIE_Voice] Five-Lab Self Study

[Randall Crumm]

Last I heard was soon.

The labs are very challenging and offer alot more from volume 2. They are well 
worth the money.

Randall




 From: Jurassic Labs 
To: ccie_voice@onlinestudylist.com 
Sent: Saturday, January 7, 2012 5:22 PM
Subject: [OSL | CCIE_Voice] Five-Lab Self Study
 

I've purchased the new Five-Lab package which for sure has some good 
information in it.  However there are only 3 detailed solution guides...where's 
the other two and when does IPX expect to have those completed?   
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[OSL | CCIE_Voice] Five-Lab Self Study

[Jurassic Labs]

I've purchased the new Five-Lab package which for sure has some good
information in it.  However there are only 3 detailed solution
guides...where's the other two and when does IPX expect to have those
completed?
___
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Re: [OSL | CCIE_Voice] New Lab #1

[Rrcrumm]

Chase
Are you talking about you home lab equipment?

Randall

Sent from my iPhone

On Jan 7, 2012, at 3:41 PM, Edgar Feliz  wrote:

> I have done this lab several times and have not had an issue. I have not 
> looked at the PSTN router in a while but in the racks that I have worked on 
> mostly 29, 32 12, 13 I have not seen and issue.
> 
> Edgar
> 
> On Sat, Jan 7, 2012 at 11:54 AM, chase mergenthal  wrote:
> Has anyone noticed the Dial-peers on the PSTN router do not match the Dial 
> plan of the Lab at all?
> 
> The LAB show's
> 
> 911 on line 1
> 408777! on line 2
> 415888! on line 3
> 0207735! on line 4
> ect 
> 
> PSTN-WAN#sho run | section dial-peer
> dial-peer voice 100 pots
>  incoming called-number .
>  direct-inward-dial
> dial-peer voice 101 pots
>  destination-pattern 2025552...
>  port 0/3/0:23
>  forward-digits 10
> dial-peer voice 102 pots
>  destination-pattern 4083873...
>  port 0/3/1:23
>  forward-digits 10
> dial-peer voice 103 pots
>  destination-pattern 02077964...
>  port 0/2/0:15
>  forward-digits 8
> dial-peer voice 104 pots
>  destination-pattern 02077964...
>  port 0/2/0:15
>  prefix +442077964
> 
> 
> What gets me is that I have the dialplan setup on the cucm side, but there is 
> no chance for this to work with the PSTN router the way it is
> 
> -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
> 
> ___
> 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 
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Re: [OSL | CCIE_Voice] MVA dial peers

[George Goglidze]

Yea, good luck with that

Snippet from CUCM 7.x SRND: 
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/mobilapp.html#wp1043827

Mobile Voice Access IVR VoiceXML Gateway URL

The Mobile Voice Access feature requires the Unified CM VoiceXML application to 
reside on the H.323 or SIP gateway. The URL used to load this application is:

http://:8080/ccmivr/pages/IVRMainpage.vxml

where  is the IP address of the Unified CM 
publisher node.



Please paste the exact URL of the guide that you reffer to.


Sent from my iPad

On 7 Jan 2012, at 13:50, datucha123 datucha123  wrote:

> Take a look at:
> "Mobile Voice Access Configuration by Service Parameter" in Feature and 
> Service Guide, where it says:
>  
> To configure Mobile Voice Access without using an H.323 gateway, use the 
> following procedure ...
>  
> Also this kind of configuration does not support IVR prompts.
>  
> But still the MVA is supported without H323 Gateway at all.
>  
> On Sat, Jan 7, 2012 at 4:05 PM, George Goglidze  wrote:
> MVA always needs a voice gateway!!!
> On CUCM 7.0 it supports h323 only
> On CUCM 7.1.3 (not tested in lab) it supports h323 and SIP. 
> 
> To answer initial question, no you do not need dial-peers to reach dialed 
> numbers through MVA. Only MVA dial-peer.
> 
> Cheers,
> 
> Sent from my iPad
> 
> On 7 Jan 2012, at 10:37, datucha123 datucha123  wrote:
> 
>> As I remember, the MVA does not need the H323 dial-peer always. CUCM itself 
>> can also provide the VXML locally, so you do not need an H323 gateway with 
>> dial-peers at all.
>> There is a Service Parameter, which indicates that feature.
>>  
>> I do not remember how it is called, but there it is.
>> 
>> On Sat, Jan 7, 2012 at 12:17 PM, CCIEVoiceKP  wrote:
>> Question here that just struck me ... If you configure Mobile Voice Access 
>> on a router that is a an MGCP gw, obviously you need to also config an H323 
>> gw for MVA (which includes a dial-peer to reach MVA), but do you need 
>> dial-peers to reach numbers you call through MVA.
>> 
>> So if I dial my MVA number, enter my remote destination and pin and then 
>> press 1 to make a call ... Does that call go out of the MGCp gw or will it 
>> go out of the h323 gw and require additional dial-peers?
>> 
>> KP
>> 
>> Sent from my iPad
>> ___
>> 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 
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Re: [OSL | CCIE_Voice] New Lab #1

[Edgar Feliz]

I have done this lab several times and have not had an issue. I have not
looked at the PSTN router in a while but in the racks that I have worked on
mostly 29, 32 12, 13 I have not seen and issue.

Edgar

On Sat, Jan 7, 2012 at 11:54 AM, chase mergenthal wrote:

>  Has anyone noticed the Dial-peers on the PSTN router do not match the
> Dial plan of the Lab at all?
>
> The LAB show's
>
> 911 on line 1
> 408777! on line 2
> 415888! on line 3
> 0207735! on line 4
> ect
>
> PSTN-WAN#sho run | section dial-peer
> dial-peer voice 100 pots
>  incoming called-number .
>  direct-inward-dial
> dial-peer voice 101 pots
>  destination-pattern 2025552...
>  port 0/3/0:23
>  forward-digits 10
> dial-peer voice 102 pots
>  destination-pattern 4083873...
>  port 0/3/1:23
>  forward-digits 10
> dial-peer voice 103 pots
>  destination-pattern 02077964...
>  port 0/2/0:15
>  forward-digits 8
> dial-peer voice 104 pots
>  destination-pattern 02077964...
>  port 0/2/0:15
>  prefix +442077964
>
>
> What gets me is that I have the dialplan setup on the cucm side, but there
> is no chance for this to work with the PSTN router the way it is
>
> -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
>
___
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] Caller ID Update

[btmulg...@gmail.com]

Hi - for called number display use the route pattern level of digit 
manipulation, the rl / xfm level of manipulation will supersede the rp but not 
the display.  

hth

Sent from my iPad

On 6 Jan 2012, at 20:18, datucha123 datucha123  wrote:

> I have the following kind of configuration:
>  
> Called Number are Globalized, and then localized at the GW/Trunk level. The 
> digits that are send to Gateways are without the 9 access code, so that the 
> IP Phone displays the called number without the Access Code of 9.
>  
> But now look, when the Site A phone call National number to Site B 
> destination, and that call is Hop off through the Site B Gateway (so it gets 
> the local cal for Site B) the Called Number is changed. Those changes are 
> made by the Called Party Transformations. Gateway does not update the Caller 
> ID, because it does not make any translations on the inbound VoIP direction.
>  
> So when the user at Site A dials 9 1 XXX XXX, and this call is hop off to 
> Site B gateway the final number is displayed as XXX XXX, where I want it 
> to display the called number with 1, as the user has dialed National Number 
> initially.
>  
> Well if I use the Callmanager Service Parameter "always display dialed 
> digits", then the 9 will also stay on the screen, which I do not want to.

>  
> How to deal with that issue?
>  
> Hope I could describe the problem well :)
> ___
> 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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www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] CUE Viewmail issue

[Rrcrumm]

I configured that already

Sent from my iPhone

On Jan 7, 2012, at 1:57 PM, CCIEVoiceKP  wrote:

> Did you do an:
> 
> ip http server
> 
> That got me with EM service not showing up before.
> 
> KP
> 
> Sent from my iPad
> 
> On Jan 7, 2012, at 12:22 PM, Randall Crumm  wrote:
> 
>> HI,
>> I am having an issue with voiceview and CUE w/ CME.
>> 
>> My phone is not learning the service URL's?
>> 
>> I have reset it, but don't know what else I can do.
>> 
>> Any thoughts?
>> 
>> 
>> thanks,
>> 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

Re: [OSL | CCIE_Voice] CUE Viewmail issue

[CCIEVoiceKP]

Did you do an:

ip http server

That got me with EM service not showing up before.

KP

Sent from my iPad

On Jan 7, 2012, at 12:22 PM, Randall Crumm  wrote:

> HI,
> I am having an issue with voiceview and CUE w/ CME.
> 
> My phone is not learning the service URL's?
> 
> I have reset it, but don't know what else I can do.
> 
> Any thoughts?
> 
> 
> thanks,
> 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

Re: [OSL | CCIE_Voice] CUE MWI issue

[Randall Crumm]

This is working now. I am not sure but some 30-40 minutes later it is working

rc



 From: Randall Crumm 
To: Online Study  
Sent: Saturday, January 7, 2012 1:02 PM
Subject: [OSL | CCIE_Voice] CUE MWI issue
 

Hi,
I have CUE with CME. I have configured MWi for subscribe notify.

According to the DSG my config looks correct. 

When I do debug ccsip messages I see:
Messages-Waiting: yes
Message-Account: sip:4002@10.10.115.2


But the MWI on the phone is not working. 

Any thoughts

I have done the following

checked transcoder is registered

ephone-dn 2
no mwi sip
mwi sip

sup-ua
no mwi-server ipv4:10.10.115.2
mwi-server ipv4:10.10.115.2 


reset ephone 2

thanks,
Rndall



___
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] CUE Viewmail issue

[Randall Crumm]

HI,
I am having an issue with voiceview and CUE w/ CME.

My phone is not learning the service URL's?

I have reset it, but don't know what else I can do.

Any thoughts?


thanks,
Randall
___
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www.ipexpert.com

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

Re: [OSL | CCIE_Voice] MVA dial peers

[CCIEVoiceKP]

Yeah I guess if I'd really stopped to think it through that makes sense.  Any 
calls you make through MVA would need to match a Route Pattern in CUCM and that 
would dictate which gateway would be used ... So in my scenario with an MGCP gw 
no dial-peers needed ... Aside from the one for h323 hairpinning.

Thanks for the replies guys!

Sent from my iPad

KP

On Jan 7, 2012, at 3:05 AM, George Goglidze  wrote:

> MVA always needs a voice gateway!!!
> On CUCM 7.0 it supports h323 only
> On CUCM 7.1.3 (not tested in lab) it supports h323 and SIP. 
> 
> To answer initial question, no you do not need dial-peers to reach dialed 
> numbers through MVA. Only MVA dial-peer.
> 
> Cheers,
> 
> Sent from my iPad
> 
> On 7 Jan 2012, at 10:37, datucha123 datucha123  wrote:
> 
>> As I remember, the MVA does not need the H323 dial-peer always. CUCM itself 
>> can also provide the VXML locally, so you do not need an H323 gateway with 
>> dial-peers at all.
>> There is a Service Parameter, which indicates that feature.
>>  
>> I do not remember how it is called, but there it is.
>> 
>> On Sat, Jan 7, 2012 at 12:17 PM, CCIEVoiceKP  wrote:
>> Question here that just struck me ... If you configure Mobile Voice Access 
>> on a router that is a an MGCP gw, obviously you need to also config an H323 
>> gw for MVA (which includes a dial-peer to reach MVA), but do you need 
>> dial-peers to reach numbers you call through MVA.
>> 
>> So if I dial my MVA number, enter my remote destination and pin and then 
>> press 1 to make a call ... Does that call go out of the MGCp gw or will it 
>> go out of the h323 gw and require additional dial-peers?
>> 
>> KP
>> 
>> Sent from my iPad
>> ___
>> 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

[OSL | CCIE_Voice] CUE MWI issue

[Randall Crumm]

Hi,
I have CUE with CME. I have configured MWi for subscribe notify.

According to the DSG my config looks correct. 

When I do debug ccsip messages I see:
Messages-Waiting: yes
Message-Account: sip:4002@10.10.115.2


But the MWI on the phone is not working. 

Any thoughts

I have done the following

checked transcoder is registered

ephone-dn 2
no mwi sip
mwi sip

sup-ua
no mwi-server ipv4:10.10.115.2
mwi-server ipv4:10.10.115.2 


reset ephone 2

thanks,
Rndall___
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 Calling Name

[Jason Murray]

Yes that is configured too that was pulled over from the shared line 5002. Is 
there anything that affects MVA maybe a Transformation Pattern, Calling Party 
Transformation Pattern CSS, route pattern etc. I've tried playing around with 
those as well. Nothing has changed. What settings are involved, looked at, or 
passed through during MVA? Theres gotta be something other than the Line 
settings involved. Because if I call a number at the same site (5001) or even 
BR2 phones (3...) for calling number I get 10 digits instead of 4 like when I 
call 1002. Still now caller name when answered those on those sites either. 
- Original Message -
From: datucha123 datucha123
Sent: 01/07/12 10:59 AM
To: Jason Murray
Subject: Re: [OSL | CCIE_Voice] MVA Calling Name

 Try to configure the "Alerting Name" and "ASCII Alerting Name" for RDP 
Extensions - just in the screen shot that you haev sent.

 On Sat, Jan 7, 2012 at 7:28 PM, Jason Murray < murr...@usa.com > wrote:
Thanks Amol,

 I read that too but even after I answer it the name still does not appear and 
I do have the RDP line settings set, see attached screenshot.

 Thanks
 Jason



- Original Message -
From: Amol Mittal
Sent: 01/06/12 08:11 PM
To: 'Jason Murray',  ccie_voice@onlinestudylist.com 
Subject: RE: [OSL | CCIE_Voice] MVA Calling Name

Jason,
To my knowledge, it’s working as designed. When you call any internal extension 
after authenticating to the MVA, it shows only the calling number when the 
called phone is ringing. However, when the call gets connected, it shows the 
calling name (or rather the connected name) as well. Calling name would not 
appear after call connect, if you do not have calling name configured in RDP 
line settings.
HTH
Regards, Amol
From: ccie_voice-boun...@onlinestudylist.com  [mailto: 
ccie_voice-boun...@onlinestudylist.com ]  *On Behalf Of *Jason Murray
 *Sent:* Friday, January 06, 2012 3:08 PM
 *To:* ccie_voice@onlinestudylist.com 
 *Subject:* [OSL | CCIE_Voice] MVA Calling Name
I've searched the lists and google until I have no hair left. I am doing Vol1 
5C.12 MVA. Everything works beautifully. I dial in from the PSTN phone on line 
two to 5999. It answers put in my pin. Press 1 to dial a number. Dial 1002. It 
rings and I can answer it. So my problem is there is no Calling name just 
number of 5002. I have Calling name set in the RDP line settings. If I dial 
5001 using MVA the number is shown as 2123945002 tel:2123945002 . So I am 
trying to figure out how it decides what the display name and number is pulled 
from. I see that alot of people in the threads have had this before and most 
people say it should work. Others say its by design. And I havent seen anybody 
have a fix for it. Even during Vic's walkthrough of this section shows he only 
gets 5002 on the display when he calls 1002 with MVA. Anybody have any info on 
this. I cant be the only one. Thanks for your help.


 Jason 

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

Re: [OSL | CCIE_Voice] MVA Calling Name

[datucha123 datucha123]

Try to configure the "Alerting Name" and "ASCII Alerting Name" for RDP
Extensions  -  just in the screen shot that you haev sent.

On Sat, Jan 7, 2012 at 7:28 PM, Jason Murray  wrote:

> Thanks Amol,
>
> I read that too but even after I answer it the name still does not appear
> and I do have the RDP line settings set, see attached screenshot.
>
> Thanks
> Jason
>
>
>
>
>
> - Original Message -
>
> From: Amol Mittal
>
> Sent: 01/06/12 08:11 PM
>
> To: 'Jason Murray', ccie_voice@onlinestudylist.com
>
> Subject: RE: [OSL | CCIE_Voice] MVA Calling Name
>
>  Jason,
>
>
>
>
>
>
>
>
>
>
>
> To my knowledge, it’s working as designed. When you call any internal
> extension after authenticating to the MVA, it shows only the calling number
> when the called phone is ringing. However, when the call gets connected, it
> shows the calling name (or rather the connected name) as well. Calling name
> would not appear after call connect, if you do not have calling name
> configured in RDP line settings.
>
>
>
>
>
>
>
>
>
>
>
> HTH
>
>
>
>
>
> Regards, Amol
>
>
>
>
>
>
>
>
>
>
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Jason Murray
> *Sent:* Friday, January 06, 2012 3:08 PM
> *To:* ccie_voice@onlinestudylist.com
> *Subject:* [OSL | CCIE_Voice] MVA Calling Name
>
>
>
>
>
>
>
>
>
>
>
> I've searched the lists and google until I have no hair left. I am doing
> Vol1 5C.12 MVA. Everything works beautifully. I dial in from the PSTN phone
> on line two to 5999. It answers put in my pin. Press 1 to dial a number.
> Dial 1002. It rings and I can answer it. So my problem is there is no
> Calling name just number of 5002. I have Calling name set in the RDP line
> settings. If I dial 5001 using MVA the number is shown as 2123945002. So
> I am trying to figure out how it decides what the display name and number
> is pulled from. I see that alot of people in the threads have had this
> before and most people say it should work. Others say its by design. And I
> havent seen anybody have a fix for it. Even during Vic's walkthrough of
> this section shows he only gets 5002 on the display when he calls 1002 with
> MVA. Anybody have any info on this. I cant be the only one. Thanks for your
> help.
>
>
> Jason
>
>
>
>
>
>
>
> ___
> 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] New Lab #1

[chase mergenthal]

Has anyone noticed the Dial-peers on the PSTN router do not match the Dial plan 
of the Lab at all?

The LAB show's

911 on line 1
408777! on line 2
415888! on line 3
0207735! on line 4
ect 

PSTN-WAN#sho run | section dial-peer
dial-peer voice 100 pots
 incoming called-number .
 direct-inward-dial
dial-peer voice 101 pots
 destination-pattern 2025552...
 port 0/3/0:23
 forward-digits 10
dial-peer voice 102 pots
 destination-pattern 4083873...
 port 0/3/1:23
 forward-digits 10
dial-peer voice 103 pots
 destination-pattern 02077964...
 port 0/2/0:15
 forward-digits 8
dial-peer voice 104 pots
 destination-pattern 02077964...
 port 0/2/0:15
 prefix +442077964


What gets me is that I have the dialplan setup on the cucm side, but there is 
no chance for this to work with the PSTN router the way it is

-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

[OSL | CCIE_Voice] Hardware Media Resources in same CME Router

[Ken Wyan]

Can we use same ip address for  Media Resource sccp registration & CME
source-address?
(it seems working , but want to know if recommend to use different IP
addresses ; any internal loops etc.. due to same ip address)

sccp local  loop0  <

sccp ccm group 1
bind interface loop0 <--

dspfarm profile ...

interface loopback0
ip address 1.1.1.1 255.255.255.0  <---


telephony-service
ip source-address 1.1.1.1 <---
sdspfarm 
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Re: [OSL | CCIE_Voice] MVA Calling Name

[Jason Murray]

Thanks Amol,

 I read that too but even after I answer it the name still does not appear and 
I do have the RDP line settings set, see attached screenshot.

 Thanks
 Jason



- Original Message -
From: Amol Mittal
Sent: 01/06/12 08:11 PM
To: 'Jason Murray', ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] MVA Calling Name

Jason,
To my knowledge, it’s working as designed. When you call any internal extension 
after authenticating to the MVA, it shows only the calling number when the 
called phone is ringing. However, when the call gets connected, it shows the 
calling name (or rather the connected name) as well. Calling name would not 
appear after call connect, if you do not have calling name configured in RDP 
line settings.
HTH
Regards, Amol
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com]  *On Behalf Of *Jason Murray
 *Sent:* Friday, January 06, 2012 3:08 PM
 *To:* ccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] MVA Calling Name
I've searched the lists and google until I have no hair left. I am doing Vol1 
5C.12 MVA. Everything works beautifully. I dial in from the PSTN phone on line 
two to 5999. It answers put in my pin. Press 1 to dial a number. Dial 1002. It 
rings and I can answer it. So my problem is there is no Calling name just 
number of 5002. I have Calling name set in the RDP line settings. If I dial 
5001 using MVA the number is shown as 2123945002. So I am trying to figure out 
how it decides what the display name and number is pulled from. I see that alot 
of people in the threads have had this before and most people say it should 
work. Others say its by design. And I havent seen anybody have a fix for it. 
Even during Vic's walkthrough of this section shows he only gets 5002 on the 
display when he calls 1002 with MVA. Anybody have any info on this. I cant be 
the only one. Thanks for your help.


 Jason
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[OSL | CCIE_Voice] LAB exam JAN & FEB

[Google]

Hi Dears,
 
Who is appearing in LAB exam in JAN and FEB they can add me.
 
ccievoicedu...@gmail.com
 
in gtalk.
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Re: [OSL | CCIE_Voice] MVA dial peers

[datucha123 datucha123]

Take a look at:
"Mobile Voice Access Configuration by Service Parameter" in Feature and
Service Guide, where it says:

To configure Mobile Voice Access without using an H.323 gateway, use the
following procedure ...

Also this kind of configuration does not support IVR prompts.

But still the MVA is supported without H323 Gateway at all.

On Sat, Jan 7, 2012 at 4:05 PM, George Goglidze  wrote:

>  MVA always needs a voice gateway!!!
> On CUCM 7.0 it supports h323 only
> On CUCM 7.1.3 (not tested in lab) it supports h323 and SIP.
>
> To answer initial question, no you do not need dial-peers to reach dialed
> numbers through MVA. Only MVA dial-peer.
>
> Cheers,
>
> Sent from my iPad
>
> On 7 Jan 2012, at 10:37, datucha123 datucha123 
> wrote:
>
>   As I remember, the MVA does not need the H323 dial-peer always. CUCM
> itself can also provide the VXML locally, so you do not need an H323
> gateway with dial-peers at all.
> There is a Service Parameter, which indicates that feature.
>
> I do not remember how it is called, but there it is.
>
> On Sat, Jan 7, 2012 at 12:17 PM, CCIEVoiceKP wrote:
>
>> Question here that just struck me ... If you configure Mobile Voice
>> Access on a router that is a an MGCP gw, obviously you need to also config
>> an H323 gw for MVA (which includes a dial-peer to reach MVA), but do you
>> need dial-peers to reach numbers you call through MVA.
>>
>> So if I dial my MVA number, enter my remote destination and pin and then
>> press 1 to make a call ... Does that call go out of the MGCp gw or will it
>> go out of the h323 gw and require additional dial-peers?
>>
>> KP
>>
>> Sent from my iPad
>> ___
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>> visit www.ipexpert.com
>>
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>> www.PlatinumPlacement.com 
>>
>
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> visit www.ipexpert.com
>
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>
>
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Re: [OSL | CCIE_Voice] MVA dial peers

[George Goglidze]

MVA always needs a voice gateway!!!
On CUCM 7.0 it supports h323 only
On CUCM 7.1.3 (not tested in lab) it supports h323 and SIP. 

To answer initial question, no you do not need dial-peers to reach dialed 
numbers through MVA. Only MVA dial-peer.

Cheers,

Sent from my iPad

On 7 Jan 2012, at 10:37, datucha123 datucha123  wrote:

> As I remember, the MVA does not need the H323 dial-peer always. CUCM itself 
> can also provide the VXML locally, so you do not need an H323 gateway with 
> dial-peers at all.
> There is a Service Parameter, which indicates that feature.
>  
> I do not remember how it is called, but there it is.
> 
> On Sat, Jan 7, 2012 at 12:17 PM, CCIEVoiceKP  wrote:
> Question here that just struck me ... If you configure Mobile Voice Access on 
> a router that is a an MGCP gw, obviously you need to also config an H323 gw 
> for MVA (which includes a dial-peer to reach MVA), but do you need dial-peers 
> to reach numbers you call through MVA.
> 
> So if I dial my MVA number, enter my remote destination and pin and then 
> press 1 to make a call ... Does that call go out of the MGCp gw or will it go 
> out of the h323 gw and require additional dial-peers?
> 
> KP
> 
> Sent from my iPad
> ___
> 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
> 
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> www.PlatinumPlacement.com
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Re: [OSL | CCIE_Voice] MVA dial peers

[datucha123 datucha123]

As I remember, the MVA does not need the H323 dial-peer always. CUCM itself
can also provide the VXML locally, so you do not need an H323 gateway with
dial-peers at all.
There is a Service Parameter, which indicates that feature.

I do not remember how it is called, but there it is.

On Sat, Jan 7, 2012 at 12:17 PM, CCIEVoiceKP  wrote:

> Question here that just struck me ... If you configure Mobile Voice Access
> on a router that is a an MGCP gw, obviously you need to also config an H323
> gw for MVA (which includes a dial-peer to reach MVA), but do you need
> dial-peers to reach numbers you call through MVA.
>
> So if I dial my MVA number, enter my remote destination and pin and then
> press 1 to make a call ... Does that call go out of the MGCp gw or will it
> go out of the h323 gw and require additional dial-peers?
>
> KP
>
> Sent from my iPad
> ___
> 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] SRST with Unity Connections

[datucha123 datucha123]

Sorry, that will work - inbound gateway css pointing to a translation
pattern

I have just though that you were talking about the Transformation, sorry.

But the Gateway CSS, is a little uncomfortable - becase for that I have to
have the same Translation Pattern as the VM Pilot, just the Gateway has to
see the TP instead of VM Pilot, and TP will have a CSS for VM Pilot to
route the calls, where the TP will tranlsate the calling number.
 Well, that is also a variant.

Thank you very much for your help.

On Sat, Jan 7, 2012 at 5:38 AM, Julien Krieger wrote:

> Alternate Extension definitely works... that's what I am using in real
> life.
>
> Can you explain why inbound gateway css pointing to a translation pattern
> doing calling party transformation would not work ?
>
>
> 2012/1/6 datucha123 datucha123 
>
>> Gateway CSS does not make Ingress Digit Manipulation.
>>
>> As for DTMF patterns, they are used in FXO and other types of analog
>> Connections such as T1 CAS. Those are also used for mailbox Selection
>> during the Forward. That's not an issue, I can also use the ISDN Redirect
>> IE for that feature.
>>
>> My main problem is the direct call to Unity Connection from SRST IP Phone
>> to listen to left Voice Mail Messages.
>>
>> Using Alternate Extension is a variant, I think.
>>
>>
>>
>>
>> On Fri, Jan 6, 2012 at 7:23 PM, Julien Krieger 
>> wrote:
>>
>>> You could also use a translation pattern in gateway css.
>>> The translation pattern do digit manipulation...
>>>
>>>
>>> 2012/1/6 Julien Krieger 
>>>
 Hi,

 Use the Alternate Extension feature available on Unity Connection User
 interface...
 Add an alternate extension with appropriate number.

 Julien
 2012/1/6 datucha123 datucha123 

>
> Hello,
>
> As I gues there are following ways to integrate SRST with Unity
> Connection through PRI:
>
> 1) when the IP phone are in SRST mode, configure the "*isdn outgoing
> ie redirecting-number*", so that the RDNIS will be send over PSTN
> Route to HQ Unity Connection, and it can select the correct Mailbox.
>
> But I have an issue with the following  -  Let the Users press the
> Voicemail Button on their IP Phones in SRST mode and hear their messages.
>
> Well, for that that I have to configure the VM Pilot number in
> call-manager-fallback (or CME SRST) as the PSTN Global number. But when
> that call will come to Unity, (where the Users are present with 4 digit
> extensions), how to strip unnecessary digits from ANI, so that the Unity
> Connection can select the correct users mailbox?
>
> VM Profile does not work in that case any more. And the VM Hunt Pilot,
> setting for Calling Number Translation to  is not a good idea.
>
> As I know, there are several ways:
>
> 1) Configure the Secondary Extension for those users as global PSTN
> Number
> 2)  Or use the Initial Unity Connection with Global Numbers for users,
> and have the VM Profile which will expand the 4 digits extension to
> respective Global PSTN Numbers.
>
> Is there another/better way?
>
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>


>>>
>>
>
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[OSL | CCIE_Voice] MVA dial peers

[CCIEVoiceKP]

Question here that just struck me ... If you configure Mobile Voice Access on a 
router that is a an MGCP gw, obviously you need to also config an H323 gw for 
MVA (which includes a dial-peer to reach MVA), but do you need dial-peers to 
reach numbers you call through MVA.

So if I dial my MVA number, enter my remote destination and pin and then press 
1 to make a call ... Does that call go out of the MGCp gw or will it go out of 
the h323 gw and require additional dial-peers?

KP

Sent from my iPad
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