Re: [OSL | CCIE_Voice] Vol2 Lab 7 Question 3.2 MVA

[Roger Källberg]

That is also a known "feature" in CUCM 7.x that must be known :)

** Sent from my iPhone. Excuse brevity and typographical errors. **

18 apr 2011 kl. 04:44 skrev "Cristobal Priego" 
mailto:cristobalpri...@gmail.com>>:

yes i can

the issue at the end was i had to bounce the settings on service parameters 
several times in order to get it to work

2011/4/17 Roger Källberg 
<<mailto:roger.kallb...@cygate.se>roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>>
Can you "see" the pt were you put the MVA dn from the RDP CSS? That was the 
problem for Shingei.

** Sent from my iPhone. Excuse brevity and typographical errors. **

17 apr 2011 kl. 04:47 skrev "Cristobal Priego" 
<<mailto:cristobalpri...@gmail.com>cristobalpri...@gmail.com<mailto:cristobalpri...@gmail.com>>:

Hello Guys,

were you able to find the resolution
i had the same problem on vol 2 lab 7
where my call fails once i'm authenticated . when i press 1 to make a call a 
soon as i dial 1002 my call drops
i double checked all the possible solutions and i have all of those set still 
nothing

do you know what the problem is ?

thank you

2011/3/24 George Goglidze 
<<mailto:gogli...@gmail.com><mailto:gogli...@gmail.com>gogli...@gmail.com<mailto:gogli...@gmail.com>>
Hi Shingei,

Great news then! I'll try it out,

Cheers,


On Thu, Mar 24, 2011 at 10:54 AM, ShinGei Yong 
<<mailto:shingei.y...@gmail.com><mailto:shingei.y...@gmail.com>shingei.y...@gmail.com<mailto:shingei.y...@gmail.com>>
 wrote:
Hi Guys,

Glad to see you all here at this time while now is time for "dinner"for 
asia,are u guys sleepless?
I fire up my home lab after received the suggestion from Roger. Testing out 
now...

20mins later...

Hi Roger,
Your magic works! As you mentioned, i changed the mva# partition from pt-phones 
to pt-snr-3002,
the internal call successful,but just don't understand why RDP CSS can't see 
pt-phones via a translation profile,
and why the mva# should be part of pt-snr-3002 instead of pt-phones? The 
xlation pattern did have a css-phones
which comprised of pt-phones

Any reasons? OR RDP CSS can't view wild card expression?[15]XXX?

ps:George, you may try it out by just change the partition of mva# to 
pt-xlate-mva(in ur case).

Thanks
Shingei.





2011/3/24 George Goglidze 
<<mailto:gogli...@gmail.com><mailto:gogli...@gmail.com>gogli...@gmail.com<mailto:gogli...@gmail.com>>
Hi Roger,

I don't see that as a requirement nowhere... I will try it out though next time 
I do it.

Anyway, just to let you know, that while internal calls didn't work, I had an 
RP to 90014158884343 on PT-XLATE-MVA and that did work!!! so I'm not sure I 
need MVA DN's PT in CSS-MVA.

From what I understand, MVA DN's PT has to be in a CSS of h323 gateway,

Please correct me if I'm wrong.

As I said, I'll try it out anyway!

Thanks,

2011/3/24 Roger Källberg 
<<mailto:roger.kallb...@cygate.se><mailto:roger.kallb...@cygate.se>roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>>
Hi George,
What I ment was that you need to see the MVA DN set under Media Resources in 
the RDP CSS. But I never said that you need to put that DN in PT-INTERNAL, put 
that in PT-XLATE-MVA and try it out.

Sincerely

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


Från: George Goglidze 
[<mailto:gogli...@gmail.com><mailto:gogli...@gmail.com>gogli...@gmail.com<mailto:gogli...@gmail.com>]
Skickat: den 24 mars 2011 11:09
Till: Roger Källberg
Kopia: ShinGei Yong; <mailto:stden...@cisco.com> <mailto:stden...@cisco.com> 
stden...@cisco.com<mailto:stden...@cisco.com>; 
<mailto:ccie_voice@onlinestudylist.com> <mailto:ccie_voice@onlinestudylist.com> 
ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com>

Ämne: Re: [OSL | CCIE_Voice] Vol2 Lab 7 Question 3.2 MVA

Hi Roger,

I think you have mis-read the question. The problem we are having is the 
following:

PT-INTERNAL --- extensions ,500X, 100X
PT-XLATE-MVA - [15]XXX -> translate calling to +44..   -> APPLIED 
CSS-INTERNAL -- contains PT-INTERNAL
CSS-MVA contains PT-XLATE-MVA

CSS-MVA is a CSS applied to RDP, and in service parameters we set MVA uses Line 
CSS and RDP CSS combined.
(left out on purpose but the rest of requierements for MVA is set correctly)

Now user calls in to MVA number, enters PIN#, then 1, then 5002# -> call fails

Now if we change slightly the CSS-MVA and include PT-INTERNAL directly, instead 
of only PT-XLATE-MVA, then the same example succeeds.
but by doing so, we break requirement of the prior task, stating that incoming 
calls from mobile to 500X, 100X should see +44. in missed/received calls 
too.

Regards,


2011/3/24 Roger 

Re: [OSL | CCIE_Voice] Vol2 Lab 7 Question 3.2 MVA

[Roger Källberg]

Can you "see" the pt were you put the MVA dn from the RDP CSS? That was the 
problem for Shingei.

** Sent from my iPhone. Excuse brevity and typographical errors. **

17 apr 2011 kl. 04:47 skrev "Cristobal Priego" 
mailto:cristobalpri...@gmail.com>>:

Hello Guys,

were you able to find the resolution
i had the same problem on vol 2 lab 7
where my call fails once i'm authenticated . when i press 1 to make a call a 
soon as i dial 1002 my call drops
i double checked all the possible solutions and i have all of those set still 
nothing

do you know what the problem is ?

thank you

2011/3/24 George Goglidze 
<<mailto:gogli...@gmail.com>gogli...@gmail.com<mailto:gogli...@gmail.com>>
Hi Shingei,

Great news then! I'll try it out,

Cheers,


On Thu, Mar 24, 2011 at 10:54 AM, ShinGei Yong 
<<mailto:shingei.y...@gmail.com>shingei.y...@gmail.com<mailto:shingei.y...@gmail.com>>
 wrote:
Hi Guys,

Glad to see you all here at this time while now is time for "dinner"for 
asia,are u guys sleepless?
I fire up my home lab after received the suggestion from Roger. Testing out 
now...

20mins later...

Hi Roger,
Your magic works! As you mentioned, i changed the mva# partition from pt-phones 
to pt-snr-3002,
the internal call successful,but just don't understand why RDP CSS can't see 
pt-phones via a translation profile,
and why the mva# should be part of pt-snr-3002 instead of pt-phones? The 
xlation pattern did have a css-phones
which comprised of pt-phones

Any reasons? OR RDP CSS can't view wild card expression?[15]XXX?

ps:George, you may try it out by just change the partition of mva# to 
pt-xlate-mva(in ur case).

Thanks
Shingei.





2011/3/24 George Goglidze 
<<mailto:gogli...@gmail.com>gogli...@gmail.com<mailto:gogli...@gmail.com>>
Hi Roger,

I don't see that as a requirement nowhere... I will try it out though next time 
I do it.

Anyway, just to let you know, that while internal calls didn't work, I had an 
RP to 90014158884343 on PT-XLATE-MVA and that did work!!! so I'm not sure I 
need MVA DN's PT in CSS-MVA.

From what I understand, MVA DN's PT has to be in a CSS of h323 gateway,

Please correct me if I'm wrong.

As I said, I'll try it out anyway!

Thanks,

2011/3/24 Roger Källberg 
<<mailto:roger.kallb...@cygate.se>roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>>
Hi George,
What I ment was that you need to see the MVA DN set under Media Resources in 
the RDP CSS. But I never said that you need to put that DN in PT-INTERNAL, put 
that in PT-XLATE-MVA and try it out.

Sincerely

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

____
Från: George Goglidze 
[<mailto:gogli...@gmail.com>gogli...@gmail.com<mailto:gogli...@gmail.com>]
Skickat: den 24 mars 2011 11:09
Till: Roger Källberg
Kopia: ShinGei Yong; <mailto:stden...@cisco.com> 
stden...@cisco.com<mailto:stden...@cisco.com>; 
<mailto:ccie_voice@onlinestudylist.com> 
ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com>

Ämne: Re: [OSL | CCIE_Voice] Vol2 Lab 7 Question 3.2 MVA

Hi Roger,

I think you have mis-read the question. The problem we are having is the 
following:

PT-INTERNAL --- extensions ,500X, 100X
PT-XLATE-MVA - [15]XXX -> translate calling to +44..   -> APPLIED 
CSS-INTERNAL -- contains PT-INTERNAL
CSS-MVA contains PT-XLATE-MVA

CSS-MVA is a CSS applied to RDP, and in service parameters we set MVA uses Line 
CSS and RDP CSS combined.
(left out on purpose but the rest of requierements for MVA is set correctly)

Now user calls in to MVA number, enters PIN#, then 1, then 5002# -> call fails

Now if we change slightly the CSS-MVA and include PT-INTERNAL directly, instead 
of only PT-XLATE-MVA, then the same example succeeds.
but by doing so, we break requirement of the prior task, stating that incoming 
calls from mobile to 500X, 100X should see +44. in missed/received calls 
too.

Regards,


2011/3/24 Roger Källberg 
<<mailto:roger.kallb...@cygate.se>roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>>
Hi Shingei,
Have you set the MVA number under Media Resources in CUCM and is this number in 
a PT that can be seen by the RDP CSS and possibly also incoming CSS of the MVA 
GW?

You also need to be able to route to this number from the MVA GW with 
appropriate dialpeers, if the MVA DN is in the same number range as other 
internal DN's then you don't need any extra DP's.

I assume that you also already have changed the service parameter “Enable 
Mobile Voice Access” and “Mobile Voice Access Number”. The later should be set 
to the same as MVA DN under Media Resources.

Sincerely

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

__

Re: [OSL | CCIE_Voice] CME SRST: calling name configuration

[Roger Källberg]

Hi Shikhar,
The "known bug" part was said by Vik Malhi at the IPX bootcamp in March 2010 
when I attended it. During the first week we were many that experienced this 
issue, all were on IPX vRacks. So the CME phone firmware should be the one that 
you'd expect to see in the lab.

If memory serves me it was fixed with a reload of the router.

Sincerely

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


Från: |\___sHiKHar___/| [shikhar...@gmail.com]
Skickat: den 11 april 2011 12:44
Till: Roger Källberg
Kopia: Miron Kobelski; ccie_voice@onlinestudylist.com
Ämne: Re: [OSL | CCIE_Voice] CME SRST: calling name configuration

Can you check the firmware versions?
Are they compatible with CME?

I had similar issue which turned out to be a firmware problem

-Shikhar

2011/4/11 Roger Källberg 
mailto:roger.kallb...@cygate.se>>
Hi Miron,
Have you tried to reload the router when this happens? If I remember correctly 
it's a "known bug" that needs a reload to work again.

Sincerely

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


Från: Miron Kobelski [findko...@gmail.com<mailto:findko...@gmail.com>]
Skickat: den 9 april 2011 20:57
Till: ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com>
Ämne: [OSL | CCIE_Voice] CME SRST: calling name configuration

Hello,

I was playing with CME SRST today and I encountered the same issue again. I 
configured CME SRST with srst mode auto-provision all.
Phones reregistered to SRST correctly, ephone and ephone-dn configuration 
appeared in the config. By default, each ephone-dn is configured with CUCM 
external phone mask as a calling name. Is it possible to change ephone's 
calling name to something other in CME SRST?

When I changed the name under ephone-dn and restarted the phone. It 
reregistered, but DN didn't appear on the button (couldn't make any calls).
Is it normal/expected behaviour or I missed something?


best regards
kobel

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

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

Re: [OSL | CCIE_Voice] CME SRST: calling name configuration

[Roger Källberg]

Hi Miron,
Have you tried to reload the router when this happens? If I remember correctly 
it's a "known bug" that needs a reload to work again.

Sincerely

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


Från: Miron Kobelski [findko...@gmail.com]
Skickat: den 9 april 2011 20:57
Till: ccie_voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] CME SRST: calling name configuration

Hello,

I was playing with CME SRST today and I encountered the same issue again. I 
configured CME SRST with srst mode auto-provision all.
Phones reregistered to SRST correctly, ephone and ephone-dn configuration 
appeared in the config. By default, each ephone-dn is configured with CUCM 
external phone mask as a calling name. Is it possible to change ephone's 
calling name to something other in CME SRST?

When I changed the name under ephone-dn and restarted the phone. It 
reregistered, but DN didn't appear on the button (couldn't make any calls).
Is it normal/expected behaviour or I missed something?


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

Re: [OSL | CCIE_Voice] MGCP Gateway registration Issue

[Roger Källberg]

Hi Naoufal!
This is a old known bug in IOS. To fix it you need to do this,

no mgcp
no mgcp bind cont source lo0
no mgcp bind med source lo0

*Wait a few seconds just for good measures

mgcp bind cont source lo0
mgcp bind med source lo0
mgcp

Works for me every time.

Sincerely

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


Från: Naoufal Kerboute [naou...@mhdinfotech.com]
Skickat: den 4 april 2011 05:56
Till: ccie_voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] MGCP Gateway registration Issue

Hi gents,

Please I need your help for the below.
I’m always facing problem to get the MGCP gateway up, always the first time 
register with the serial interface instead of the bind interface (loopback or 
voice vlan int) that I specified in the config. I’ve tried to remove the isdn 
bind-l3 ccm-manager and put it again, no mgcp and mgcp but it doesn’t help 
until I remove sometime the hole config and do it again and again, reset…

I’m missing something??

Thanks a lot.
Naoufal

*
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For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] IPExpert Vol2 Lab 2 Q9.1

[Roger Källberg]

Hi Hannes,
It doesen't. You need to have isdn outgoing ie redirecting-number.

Sincerely

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


Från: Hannes Dippenaar [hanne...@hotmail.com]
Skickat: den 4 april 2011 07:35
Till: gogli...@gmail.com
Kopia: ccie_voice@onlinestudylist.com
Ämne: Re: [OSL | CCIE_Voice] IPExpert Vol2 Lab 2 Q9.1

Hi,

Please see below:

interface Serial0/0/0:23
 description # To Hannes HQ Router #
 no ip address
 encapsulation hdlc
 isdn switch-type primary-net5
 isdn protocol-emulate network
 isdn incoming-voice voice
 no cdp enable
end

I have added "isdn outgoing display-ie"

I will test and let you know. I did not know that "isdn outgoing display-ie" 
affects the redirect number.

thanks for the reply.

regards,

Hannes


From: gogli...@gmail.com
Date: Sun, 3 Apr 2011 14:59:10 -0700
Subject: Re: [OSL | CCIE_Voice] IPExpert Vol2 Lab 2 Q9.1
To: hanne...@hotmail.com
CC: ccie_voice@onlinestudylist.com

Hi,

It must be PSTN Router configuration, looks like when the initial q931 SETUP 
comes in, redirecting IE is there, but on the outgoing isdn q931 message it's 
not there.

can you post the "show run interface Se0/0/0:23" from PSTN router?
I'm sure it's missing "isnd outgoing display-ie" command.

Regards,

On Sun, Apr 3, 2011 at 11:20 AM, Hannes Dippenaar 
mailto:hanne...@hotmail.com>> wrote:
Hi, Thank you for the reply,

HQ = H323
Br1 = MGCP

I have enabled this on both HQ and BR1 GW's.

No Luck.

regards,

Hannes


From: gogli...@gmail.com<mailto:gogli...@gmail.com>
Date: Sun, 3 Apr 2011 06:59:20 -0700
Subject: Re: [OSL | CCIE_Voice] IPExpert Vol2 Lab 2 Q9.1
To: hanne...@hotmail.com<mailto:hanne...@hotmail.com>
CC: ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com>


Hi Hannes,

What is the HQ router MGCP gateway or h323?
If it's MGCP enable redirected number inbound on the MGCP GW configuration.

On Sun, Apr 3, 2011 at 6:30 AM, Hannes Dippenaar 
mailto:hanne...@hotmail.com>> wrote:
Good Day,

I am busy with IPexpert Vol2 Lab2.

I have configured the High Availability with aar and it works perfect except 
for voicemail.

If I dial from ext 5002 to ext 1002 and it transfers to voicemail. I can see on 
my BR1 router that the redirecting number is included.

It also get's to my PSTN router incoming POTS dialpeer.

But when the PSTN router sends it off to HQ router POTS dialpeer the 
redirecting number is no longer included.

This means the Unity server do not recognise the phone I am trying to leave a 
message for.

How can I resolve this issue as I suspect it is caused by my PSTN router.

I am including the debug from BR1,PSTN and HQ:

BR1

*Apr  3 14:43:44.559: ISDN Se0/1/0:23 Q931: RX <- SETUP pd = 8  callref = 0x00B0
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98381
Exclusive, Channel 1
Display i = 'HQph2'
Calling Party Number i = 0x2181, '2123945002'
Plan:ISDN, Type:National
Called Party Number i = 0x80, '16178631002'
Plan:Unknown, Type:Unknown
*Apr  3 14:43:44.767: ISDN Se0/1/0:23 Q931: TX -> CALL_PROC pd = 8  callref = 
0x80B0
Channel ID i = 0xA98381
Exclusive, Channel 1
*Apr  3 14:43:44.771: ISDN Se0/1/0:23 Q931: TX -> ALERTING pd = 8  callref = 
0x80B0
Progress Ind i = 0x8088 - In-band info or appropriate now available
*Apr  3 14:43:56.751: ISDN Se0/1/0:23 Q931: TX -> SETUP pd = 8  callref = 0x000D
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98382
Exclusive, Channel 2
Display i = 'HQph2'
Calling Party Number i = 0x2183, '2123945002'
Plan:ISDN, Type:National
Called Party Number i = 0x80, '12123945600'
Plan:Unknown, Type:Unknown
Redirecting Number i = 0x82, '1002'
Plan:Unknown, Type:Unknown
*Apr  3 14:43:56.775: ISDN Se0/1/0:23 Q931: RX <- CALL_PROC pd = 8  callref = 
0x800D
Channel ID i = 0xA98382
Exclusive, Channel 2
*Apr  3 14:43:59.827: ISDN Se0/1/0:23 Q931: RX <- ALERTING pd = 8  callref = 
0x800D




PSTN:

Apr  3 13:08:39.735: ISDN Se0/3/0:23 Q931: RX <- SETUP pd = 8  callref = 0x000D
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Cir

Re: [OSL | CCIE_Voice] ANI / DNIS ISDN Types

[Roger Källberg]

Hi,
You won't loose points for over configuring things, but you will use time from 
you limited 8 hour budget. That might in the end turn out to be disastrous for 
your score.

But to contradict that, I got an advice from Vik Malhi at his boot camp that 
you always want to take care of TON and NPI for both calling and called number. 
Only time you won't is if the question explicitly restrict you from doing 
either or. Also remember to keep the TON and NPI intact for any eventual SRST 
situation.

Sincerely

Roger Källberg
CCIE # 26199 (Voice)
Unified Communication Consultant
Cygate AB


From: voice boy [mailto:voice...@hotmail.com]
Sent: den 31 mars 2011 16:08
To: OSL Questions
Subject: Re: [OSL | CCIE_Voice] ANI / DNIS ISDN Types

Hi,

I think doing manipulation for both called and calling party won't harm
I they ask for called, the proctor will find

Or will doing manipulation for calling also will loose marks if not required ??


> From: ccie_voice-requ...@onlinestudylist.com
> Subject: CCIE_Voice Digest, Vol 61, Issue 187
> To: ccie_voice@onlinestudylist.com
> Date: Thu, 31 Mar 2011 08:22:30 -0400
>
> 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. ANI / DNIS ISDN Types (Wael Agina)
> 2. Re: ANI / DNIS ISDN Types (Rrcrumm)
> 3. Re: Lab 5 with SIP Trunk (mihal caro)
> 4. Re: Lab 5 with SIP Trunk (Justin Brady)
>
>
> --
>
> Message: 1
> Date: Thu, 31 Mar 2011 09:51:49 +0300
> From: Wael Agina 
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] ANI / DNIS ISDN Types
> Message-ID: 
> Content-Type: text/plain; charset="iso-8859-1"
>
> Dear All,
>
> I always change DNIS to match PSTN requirements [subscriber, national,
> international], But for ANI should we keep it always subscriber or it should
> also match call type ?
> i.e:
> Local call: ANI sub and DNIS sub also ?
> LD Call : ANI Nationat / DNIS National ?
> INTL call : ANI international / DNIS international ?
>
> I am little confused in this point, so kindly correct it for me.
>
> --
>
> Thanks and Best Regards,
> Wael Agina
> -- next part --
> An HTML attachment was scrubbed...
> URL: 
>
> --
>
> Message: 2
> Date: Thu, 31 Mar 2011 00:19:06 -0700
> From: Rrcrumm 
> To: Wael Agina 
> Cc: "ccie_voice@onlinestudylist.com" 
> Subject: Re: [OSL | CCIE_Voice] ANI / DNIS ISDN Types
> Message-ID: 
> Content-Type: text/plain; charset=us-ascii
>
> It should be stated in the question if it is a requirement
>
> Randall
>
> Sent from my iPhone
>
> On Mar 30, 2011, at 11:51 PM, Wael Agina  wrote:
>
> > Dear All,
> >
> > I always change DNIS to match PSTN requirements [subscriber, national, 
> > international], But for ANI should we keep it always subscriber or it 
> > should also match call type ?
> > i.e:
> > Local call: ANI sub and DNIS sub also ?
> > LD Call : ANI Nationat / DNIS National ?
> > INTL call : ANI international / DNIS international ?
> >
> > I am little confused in this point, so kindly correct it for me.
> >
> > --
> >
> > Thanks and Best Regards,
> > Wael Agina
> > ___
> > For more information regarding industry leading CCIE Lab training, please 
> > visit www.ipexpert.com
>
>
> --
>
> Message: 3
> Date: Thu, 31 Mar 2011 10:42:53 +0100
> From: mihal caro 
> To: Matt Buttner 
> Cc: ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] Lab 5 with SIP Trunk
> Message-ID:
> 
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hey guys relax let Cisco make that decision. There is no need to ban users
> he only asked a question. simples :)
>
>
> On 31 March 2011 00:51, Matt Buttner  wrote:
>
> > Everything you just said is against the Cisco NDA. Can we get a list
> > ban on these guys?
> >
> > On Wed, Mar 30, 2011 at 5:20 PM, Jonny Mendas 
> > wrote:
> > > Yes as said i got lab 5 and failed did nt able to remember full lab but
> > made
> > 

Re: [OSL | CCIE_Voice] Remove prefix 9 from called number on phone screen

[Roger Källberg]

To get the called party transformation of the RP to show on the display of the 
phone with a H.323 gw you need to add this to your configuration on the H.323 gw

voice service voip
 no supplementary-service h225-notify cid-update

Sincerely

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


Från: Michael Luo [hout...@gmail.com]
Skickat: den 27 mars 2011 20:07
Till: ccie_voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] Remove prefix 9 from called number on phone screen

I'm using CUCM 7.1.2 with H.323 gateway.

In H323 GW, the dial-peer for local number is 9 with 7-digit.  For example 
"destination-pattern 9...".

When user make a local call, the IP phone will display the called number with 
prefix 9.  For example "To: 95551212".

If we're not allowed to change dial-peer on GW, what's the easiest way to 
remove the prefix 9 from phone screen?

I tried to use route pattern to discard the 9 then prefix 9 in route group.  
But it's still showing up.  Do I have to do called-party transform pattern?  I 
was trying to avoid that.

Thanks!
Michael___
For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] Vol2 Lab 7 Question 3.2 MVA

[Roger Källberg]

Hi George,
What I ment was that you need to see the MVA DN set under Media Resources in 
the RDP CSS. But I never said that you need to put that DN in PT-INTERNAL, put 
that in PT-XLATE-MVA and try it out.

Sincerely

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


Från: George Goglidze [gogli...@gmail.com]
Skickat: den 24 mars 2011 11:09
Till: Roger Källberg
Kopia: ShinGei Yong; stden...@cisco.com; ccie_voice@onlinestudylist.com
Ämne: Re: [OSL | CCIE_Voice] Vol2 Lab 7 Question 3.2 MVA

Hi Roger,

I think you have mis-read the question. The problem we are having is the 
following:

PT-INTERNAL --- extensions ,500X, 100X
PT-XLATE-MVA - [15]XXX -> translate calling to +44..   -> APPLIED 
CSS-INTERNAL -- contains PT-INTERNAL
CSS-MVA contains PT-XLATE-MVA

CSS-MVA is a CSS applied to RDP, and in service parameters we set MVA uses Line 
CSS and RDP CSS combined.
(left out on purpose but the rest of requierements for MVA is set correctly)

Now user calls in to MVA number, enters PIN#, then 1, then 5002# -> call fails

Now if we change slightly the CSS-MVA and include PT-INTERNAL directly, instead 
of only PT-XLATE-MVA, then the same example succeeds.
but by doing so, we break requirement of the prior task, stating that incoming 
calls from mobile to 500X, 100X should see +44. in missed/received calls 
too.

Regards,


2011/3/24 Roger Källberg 
mailto:roger.kallb...@cygate.se>>
Hi Shingei,
Have you set the MVA number under Media Resources in CUCM and is this number in 
a PT that can be seen by the RDP CSS and possibly also incoming CSS of the MVA 
GW?

You also need to be able to route to this number from the MVA GW with 
appropriate dialpeers, if the MVA DN is in the same number range as other 
internal DN's then you don't need any extra DP's.

I assume that you also already have changed the service parameter “Enable 
Mobile Voice Access” and “Mobile Voice Access Number”. The later should be set 
to the same as MVA DN under Media Resources.

Sincerely

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


Från: ShinGei Yong [shingei.y...@gmail.com<mailto:shingei.y...@gmail.com>]
Skickat: den 23 mars 2011 19:01
Till: George Goglidze; stden...@cisco.com<mailto:stden...@cisco.com>
Kopia: ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com>
Ämne: Re: [OSL | CCIE_Voice] Vol2 Lab 7 Question 3.2 MVA

Mates,

I believe quite a number of us hitting this funny issue(if you search thru 
entire list since early last year)
No sure it was caused of softbug or the question itself,unable to get further 
explanation or solution after all posts.

Maybe as Steve suggested, just give it (mobility/ucm service) a lucky 
reset,then the problem gone.
Else,just wait for next unlucky fellow to hit this issue,maybe he'll give us 
the solution.Who know?

Thanks all
Shingei.


On Thu, Mar 24, 2011 at 1:27 AM, George Goglidze 
mailto:gogli...@gmail.com>> wrote:
I had the same problem last time, and I wasn't able to resolve this without 
having an internal partition in the CSS that I applied on RDP.
but then obviously it breaks task 3.1 requierement to show the number: 
+447976852817

If you find out what the problem is, let us know mate,

On Mon, Mar 21, 2011 at 11:33 AM, ShinGei Yong 
mailto:shingei.y...@gmail.com>> wrote:
Hi,

I've gone thru the entire OSL list regarding the problem as i'm facing,
but unfortunately didn't manage to get the answer.

MVA has configured on BR2 gw,the remote destination manage to call in
to MVA# and authenticated successfully. When press option#1, which make a
call to either internal or external, it just failed.

RDP CSS:css-snr-3002/pt-snr-3002
Rerouting CSS:css-br2-unrestricted/pt-uk-emer,pt-uk-national,pt-uk-international

Mobility Service Parameters:
Partial Match 10Digits, RDP + Line CSS

** As per question required,we need xlation rule to display mobile ANI instead 
of internal DN **
Translation Pattern
/pt-snr-3002
CSS:css-phones/pt-phones
Use Calling Party EPNM: Checked

1002/pt-phones
5002/pt-phones

I can call up the internal extension [15]002, IF i include the pt-phones into
RDP CSS,but that will caused ANI display become internal DN instead of mobile 
ANI
because of closest match.

Any idea why internal calling doesn't work?

Thanks
Shingei



___
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___
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Re: [OSL | CCIE_Voice] Vol2 Lab 7 Question 3.2 MVA

[Roger Källberg]

Hi Shingei,
Have you set the MVA number under Media Resources in CUCM and is this number in 
a PT that can be seen by the RDP CSS and possibly also incoming CSS of the MVA 
GW?

You also need to be able to route to this number from the MVA GW with 
appropriate dialpeers, if the MVA DN is in the same number range as other 
internal DN's then you don't need any extra DP's.

I assume that you also already have changed the service parameter “Enable 
Mobile Voice Access” and “Mobile Voice Access Number”. The later should be set 
to the same as MVA DN under Media Resources.

Sincerely

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


Från: ShinGei Yong [shingei.y...@gmail.com]
Skickat: den 23 mars 2011 19:01
Till: George Goglidze; stden...@cisco.com
Kopia: ccie_voice@onlinestudylist.com
Ämne: Re: [OSL | CCIE_Voice] Vol2 Lab 7 Question 3.2 MVA

Mates,

I believe quite a number of us hitting this funny issue(if you search thru 
entire list since early last year)
No sure it was caused of softbug or the question itself,unable to get further 
explanation or solution after all posts.

Maybe as Steve suggested, just give it (mobility/ucm service) a lucky 
reset,then the problem gone.
Else,just wait for next unlucky fellow to hit this issue,maybe he'll give us 
the solution.Who know?

Thanks all
Shingei.


On Thu, Mar 24, 2011 at 1:27 AM, George Goglidze 
mailto:gogli...@gmail.com>> wrote:
I had the same problem last time, and I wasn't able to resolve this without 
having an internal partition in the CSS that I applied on RDP.
but then obviously it breaks task 3.1 requierement to show the number: 
+447976852817

If you find out what the problem is, let us know mate,

On Mon, Mar 21, 2011 at 11:33 AM, ShinGei Yong 
mailto:shingei.y...@gmail.com>> wrote:
Hi,

I've gone thru the entire OSL list regarding the problem as i'm facing,
but unfortunately didn't manage to get the answer.

MVA has configured on BR2 gw,the remote destination manage to call in
to MVA# and authenticated successfully. When press option#1, which make a
call to either internal or external, it just failed.

RDP CSS:css-snr-3002/pt-snr-3002
Rerouting CSS:css-br2-unrestricted/pt-uk-emer,pt-uk-national,pt-uk-international

Mobility Service Parameters:
Partial Match 10Digits, RDP + Line CSS

** As per question required,we need xlation rule to display mobile ANI instead 
of internal DN **
Translation Pattern
/pt-snr-3002
CSS:css-phones/pt-phones
Use Calling Party EPNM: Checked

1002/pt-phones
5002/pt-phones

I can call up the internal extension [15]002, IF i include the pt-phones into
RDP CSS,but that will caused ANI display become internal DN instead of mobile 
ANI
because of closest match.

Any idea why internal calling doesn't work?

Thanks
Shingei



___
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www.ipexpert.com<http://www.ipexpert.com>


___
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Re: [OSL | CCIE_Voice] Unity AA transfer call directly to Voice mail

[Roger Källberg]

You need to change "Transfer incoming calls to a phone?" to "Yes, ring 
extension for:" under Call Transfer on the subscriber.

Sincerely

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


Från: kuldeep tiwari [kuldeep.tiwari1...@gmail.com]
Skickat: den 16 mars 2011 17:08
Till: CCIE for Me
Kopia: ccie_voice@onlinestudylist.com; ccie_voice-requ...@onlinestudylist.com
Ämne: Re: [OSL | CCIE_Voice] Unity AA transfer call directly to Voice mail

still not working

On Wed, Mar 16, 2011 at 5:52 PM, CCIE for Me 
mailto:cciefo...@hotmail.com>> wrote:
Remember too, that whenever a call goes through Unity first, it always will 
search itself for the call routing rules before going to CCM.  In this case, it 
is going to the entity that has the extension you are forwarding to (mailbox or 
call handler) and looks at the call transfer rules.

From: kuldeep tiwari<mailto:kuldeep.tiwari1...@gmail.com>
Sent: Wednesday, March 16, 2011 7:42 AM
To: ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com> ; 
ccie_voice-requ...@onlinestudylist.com<mailto:ccie_voice-requ...@onlinestudylist.com>
Subject: [OSL | CCIE_Voice] Unity AA transfer call directly to Voice mail



Hi All

As i am having the issue with Unity 5.0 Server,if some one calls to AA,after 
pressing the extension call is rolling to directly Voicemail,It is not ringing 
first .

There is no call FWD all apply on DN

Can any body suggests


Regards
Kuldeep



___
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--
Regards
Kuldeep Tiwari
+91-9324390008
+91-9326775437___
For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] Called number display

[Roger Källberg]

I would depend on the question asked if you need to do one or the other.

If you do DM on RL/RG level that won't show on the display of the caller. To 
affect the display in that case you need to also do a DM on the RP level, that 
won't change the actual number sent to the next level, aka RL/RG, but it will 
change the number shown on the display of the caller.

If the VGW is H.323 you also need to add the command stated by Mr Ismail.

voice service voip
 no supplementary-service h225-notify cid-update

Sincerely

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


Från: voice boy [voice...@hotmail.com]
Skickat: den 12 mars 2011 23:54
Till: OSL Questions
Ämne: [OSL | CCIE_Voice] Called number display

Hi,

I need to confirm something that confuse me,,

If I'm calling from BR1-phone to HQ-PSTN
I'll have the call to route first from HQ-GW as local
then if failed,, it'll route from BR1-GW as national

the BR1 gateway is h323

My ask is that in both cases ,, while the call is ringing or connected
BR1-Phone display on the screen  TO  912123942123
 it is because I make the DM on the RL RG level

Is it a must to make it display TO 22123942123 ??? and to remove the "9" ?

if it is a must,, how to do that in this case ?

thanks for your help

___
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Re: [OSL | CCIE_Voice] RES: IPCCX - Debug Script - Lab 12A - Vol I

[Roger Källberg]

Hi Antony,
It could be many differnt resons, the first guess would be wrong name of CSQ in 
the script. But with a reactive debug you should be able to see that.

Sincerely

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


Från: Evandro Nunes [evandrog.nu...@gmail.com]
Skickat: den 14 mars 2011 12:31
Till: 'Anthony Holloway'
Kopia: ccie_voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] RES: IPCCX - Debug Script - Lab 12A - Vol I

Hi Anthony,

I'm using Active Debug but i've already tried reactive Debug but i think that 
i'm making some mistake since everything is fine, CTI ports are registered, CTI 
RP is registered too but when i place a call to 5710 i hear a message that say 
there are somes system problem and is unable to process my call.

Do you have idea what could be it ?

Regards,


De: Anthony Holloway [mailto:avhollo...@gmail.com]
Enviada em: domingo, 13 de março de 2011 3:39
Para: Evandro Nunes
Cc: ccie_voice@onlinestudylist.com
Assunto: Re: [OSL | CCIE_Voice] IPCCX - Debug Script - Lab 12A - Vol I

I would appear that you are doing an active debug, and not the intended 
reactive debug.

The difference being:
Active Debugging: Used to test call flows which do not interact with the 
Triggering Contact (You can create new contacts in an active debug)
Started by simple press F10, or F5 on the keyboard
Reactive Debugging:  Used to test call flows which do interact with the 
Triggering Contact.
Started by setting up a debug listener (timeout value required), and then 
calling a trigger for which the script is associated via an application.

Can you confirm which type of debug you are using when you receive this error?

Anthony Holloway


On Fri, Mar 11, 2011 at 4:50 PM, Evandro Nunes 
mailto:evandrog.nu...@gmail.com>> wrote:
Hello,

I've edited the script using script editor to display ANI for 617 in workbook 
Vol 1 (IPCC)

But when a debug the script it show me an error msg

ANI=Get Call Contact Info (--Triggering Contact --, Calling Number )

Error msg :

Task:1600035   GetCallContactInfoStep (HandledSessionStep) Wrong trigger 
typecom.cisco.app.remote.RemoteApplicationTrigger

Any suggestion ?

Regards

De: 
ccie_voice-boun...@onlinestudylist.com<mailto:ccie_voice-boun...@onlinestudylist.com>
 
[mailto:ccie_voice-boun...@onlinestudylist.com<mailto:ccie_voice-boun...@onlinestudylist.com>]
 Em nome de wilson.sam...@usc-bt.com<mailto:wilson.sam...@usc-bt.com>
Enviada em: sexta-feira, 11 de março de 2011 7:07
Para: com...@gmail.com<mailto:com...@gmail.com>; 
ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com>
Assunto: Re: [OSL | CCIE_Voice] I failed because of MGCP issue- need help

Hi Raj,

To avoid any typos or any confusion I always the info from “sh ccm mananger” 
output

Next time when you lab it, you may want to use the output from the command and 
copy paste (as Ms Ryan suggests :))

Regards
Sam Wilson




From: 
ccie_voice-boun...@onlinestudylist.com<mailto:ccie_voice-boun...@onlinestudylist.com>
 
[mailto:ccie_voice-boun...@onlinestudylist.com<mailto:ccie_voice-boun...@onlinestudylist.com>]
 On Behalf Of adam compton
Sent: Friday, March 11, 2011 9:01 AM
To: mailto:ccie_voice@onlinestudylist.com>>
Subject: Re: [OSL | CCIE_Voice] I failed because of MGCP issue- need help


-- Forwarded message --
From: adam compton mailto:com...@gmail.com>>
Date: Fri, Mar 11, 2011 at 9:00 AM
Subject: Re: [OSL | CCIE_Voice] I failed because of MGCP issue- need help
To: Support mailto:supp...@technisha.com>>

Did you have the MGCP control binded to the loopback or another interface?  If 
so,  you have to remove the binding, register the gateway, and add the binding 
back after the t1 is up.

If you run "show isdn status" you might see TE1 instead of 
Multiple_Frame_Established.
On Fri, Mar 11, 2011 at 1:42 AM, Support 
mailto:supp...@technisha.com>> wrote:
Dear Frineds,

I failed in my first attempt as I could not register HQ and BR1 gateway in CUCM 
as MGCP GW.
My router name for HQ was R1 and for branch R2 and Domain name configured on 
both routers as ccievoice.com<http://ccievoice.com/>

So I added in CUCM as R1.ccievoice.com<http://r1.ccievoice.com/> and 
R2.ccievoice.com<http://r2.ccievoice.com/>, as it is hostname+domainname.

But I could not get registered in CUCM I was getting error like below:

Feb 20 13:00:00.581 : MGCP Packet received from 142.100.64.11:2427--->
A UEP 7 S0/SU0/ds1...@r1.ccievoice.com<mailto:S0/SU0/ds1...@r1.ccievoice.com> 
MGCP 0.1


Thanks
Raj


__ Information from ESET NOD32 Antivirus, version of virus signature 
database 5943 (20110310) __

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com<http://www.eset.com/>

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Re: [OSL | CCIE_Voice] First Attempt...Failed miserably

[Roger Källberg]

Hi Adam,
Don't be to bummed out by this, only a very small percentage pass on the first 
attempt. Use this experience as part of your learning curve. Take a short brake 
and recuperate to get your motivation back, then start to analyze what you need 
to do better next time around. Focus on your identified weak areas and make a 
strong comeback on your next attempt.

Sincerely

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

Från: adam compton [com...@gmail.com]
Skickat: den 10 mars 2011 14:58
Till: 
Ämne: [OSL | CCIE_Voice] First Attempt...Failed miserably

Just giving everybody a status report.  I failed the Voice lab yesterday.  I'm 
really bummed out.  It's not that I failed that bums me out.  It's that a lot 
of areas I though I nailed, I got 0 percent.  It's going to be hard to get back 
on the horse and do it again, but I will probably try again in 30 days.

Adam Compton___
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Re: [OSL | CCIE_Voice] Volume 1 lab 5 (Transformation to 7 and 10 digits) to call 911

[Roger Källberg]

Hi Ahmed,
There are many ways to achieve what you ask for, the most common in my opinion 
would probably be to set and use EPNM (External Phone Number Mask) on the line 
and then manipulate this with calling transform on either the RP, RL/RG, 
calling party transformation level or if H323 GW with voice translation rules 
before sending it out to the PSTN.
Sincerely
Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB
Eric Perssons väg 21, SE-217 62 MALMÖ

Från: Ahmed Ellboudy [ahmed_ellbo...@rayacorp.com]
Skickat: den 12 mars 2011 00:33
Till: ccie_voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] Volume 1 lab 5 (Transformation to 7 and 10 digits) to 
call 911

Dear All,
I am working on this lab and I can call 911, but the PSTN phone  display the 4 
digits only for HQ and BR1 while in the question it was requested to display 7 
digits for HQ and 10 digits for BR1 , so can anyone guide me for the traces I 
need to collect to get my mistake ?

Thanks,


Ahmed Ellboudy | CCNP, CCVP.

Networking Team Leader
Raya IT - Professional Networking Services
Mobile: +20100770837
Tel  : +20238276000 Ext. 2338
Fax : +20238372930
Email  : ahmed_ellbo...@rayacorp.com<mailto:nadia_khal...@rayacorp.com>
Address : El Motamayez District - 6th of October
[cid:image001.jpg@01CBE055.87346130]


Disclaimer: NOTICE The information contained in this message is confidential 
and is intended for the addressee(s) only. If you have received this message in 
error or there are any problems please notify the originator immediately. The 
unauthorized use, disclosure, copying or alteration of this message is strictly 
forbidden. Raya will not be liable for direct, special, indirect or 
consequential damages arising from alteration of the contents of this message 
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Views expressed in this communication are not necessarily those of Raya.If you 
have received this message in error, please notify the sender immediately by 
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Re: [OSL | CCIE_Voice] 8.9 magnitute Earth Quake hit North Japan - CCIE Candidates in Japan please take note.

[Roger Källberg]

I agree with Vik, our thought are with you all. I hope that your safe. The news 
about potential nuclear plant core damage sound horrific and the images pumped 
out on the TV news are really shocking.

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

Från: Vik Malhi [vma...@ipexpert.com]
Skickat: den 11 mars 2011 17:19
Till: Jon 1992
Kopia: 
Ämne: Re: [OSL | CCIE_Voice] 8.9 magnitute Earth Quake hit North Japan - CCIE 
Candidates in Japan please take note.

Jon (and everybody else in Japan)- hope you are safe. The images on TV look 
shocking.

--
Vik Malhi – CCIE #13890
Managing Partner / Instructor - IPexpert, Inc.
Mailto: <mailto:vma...@ipexpert.com> 
vma...@ipexpert.com<mailto:vma...@ipexpert.com>
Telephone: +1.810.326.1444 ext 420
Fax: +1.810.454.0130

On Mar 11, 2011, at 7:24 AM, "Jon 1992" 
mailto:jon1...@hotmail.com>> wrote:

I am local
Give it 3 days and all will be ok, so basically Monday should be ok for any lab 
candidates.

From: ShinGei Yong<mailto:shingei.y...@gmail.com>
Sent: Friday, March 11, 2011 10:01 PM
To: <mailto:ccie_voice@onlinestudylist.com> 
ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com>
Subject: [OSL | CCIE_Voice] 8.9 magnitute Earth Quake hit North Japan - CCIE 
Candidates in Japan please take note.

Hi All,

A gently reminder to all ccie candidates who booked or intend to Japan ccie lab.

Japan major airports Narita and Haneda have been evacuated and stop operation.

Source: CNN 
Asia<http://edition.cnn.com/2011/WORLD/asiapcf/03/11/japan.quake/index.html>

Tsunami warning issued for at least 50 areas after 
quake<http://edition.cnn.com/2011/WORLD/asiapcf/03/11/tsunami.warning/index.html?iref=NS1>
Thanks.
Shingei.


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Re: [OSL | CCIE_Voice] Transcoder requirement for ICD and CUE

[Roger Källberg]

Hi Rahul,
About this "what is you allowed to configure on two gateway", I didn't see it 
was a question reading it first time around.

Based on your original post, with slight modification to spelling and sentence 
structure.
"We are allowed to configure 3 transcoding session per router, also we need to 
configure it on only two router by looking at the requirement.. Marks won't be 
given if transcoder is configured on all IOS router."

The answer is transcoder, or as I wrote xcoder. Same thing just a shorter 
abbreviation.

Sincerely

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

Från: Rahul Kapor [rahul.kapo...@gmail.com]
Skickat: den 11 mars 2011 21:51
Till: Roger Källberg
Kopia: Rogers Ochieng; ccie_voice@onlinestudylist.com
Ämne: Re: [OSL | CCIE_Voice] Transcoder requirement for ICD and CUE

Hi Roger ,

sorry for delay response.

Still one thing is not very clear for me

what is you allowed to configure on two gateway

to answer this question

1. enough DSP resource should be available  on both gateway to configure  3 
session of 3 transcoding session
2. for the xocoder which will be used for CUE , should be configured on BR2 
gateway
3. for the xocoder which will be used for ICD , should be configured on HQ  
gateway

Let me know if i am missing any  thing here

thx,
Rahul

2011/3/10 Roger Källberg 
mailto:roger.kallb...@cygate.se>>
Hi Rahul,
I think your reading too much in to the question, or mis-intrepid it slightly. 
It doesn’t state anything about a restriction for that call to ICD only has to 
be originated from site BR1. The crucial part is to understand were a xcoder 
recourse would be needed, since the requirement say that your allowed to only 
configure it on two gateways.
Apart from this you probably also have would have a codec section that would 
state something similar to this.
Configure IP Phones and gateways in such as way that all calls within same site 
should use G711 codec. Also, all calls between the sites to remote IP phones 
and gateways should use G729 codec.
This in combination with what you wrote should help you answer your question.
Sincerely
Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB
Eric Perssons väg 21, SE-217 62 MALMÖ


Från: Rahul Kapor [rahul.kapo...@gmail.com<mailto:rahul.kapo...@gmail.com>]
Skickat: den 9 mars 2011 19:20
Till: Rogers Ochieng
Kopia: ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com>
Ämne: Re: [OSL | CCIE_Voice] Transcoder requirement for ICD and CUE

Hi Roger ,

I agree with your answer  but let say i am putting HQ and ICD in HQ device pool

if Br1 call ICD , transcoder will be invoked ,
but transcoder will be invoked if BR2 user is calling ICD , is this  breaking 
the answer ? i am not very sure about this.

thx,
Rahul

On Wed, Mar 9, 2011 at 10:32 AM, Rogers Ochieng 
mailto:rogersochi...@gmail.com>> wrote:
Why would you need all those regions? ICD is in HQ Device Pool so traffic 
within the same region will be G711, CUE will be on the Br2 device pool so 
g7711 with thin that region. Since interegion codec is g729 calls from BR1 will 
cause transcoder to be invoked

On 8 March 2011 22:49, Rahul Kapor 
mailto:rahul.kapo...@gmail.com>> wrote:


Site b should be able to call ICD route point number 2400 , using g729

hq and site B IP phone shoud be able to call unity express voice mail usig g729

we are allowed to configure 3 transcoding session per route2
also , we need to configure only one two router by looking at the requirement . 
marks wont be given it
transcoder is configured on all IOS router.

My answer would be

create 5 device pool and hence five diff region

ie HQ , BR , ICD , CUE and TRA and apply to respective device

to full fill the 1st requirement,

I will apply the transcoder to ICD device pool and will make sure that Region 
TRANS and ICD talk over g711
 and Region  BR and ICD talk over g729 (this will invokde the transcoder)

and for HQ and ICD , I will keep g711 so transcoder wont be invoked



same approach for 2nd requirement

ie i will apply the transcoder on the CUE device pool

region setting HQ/BR and CUE will be g729

and Region setting btw CUE and transcoder will be g711


I am not sure about 3rd requirement ie what  will be resource (PVDM) in real lab

Please let me know about approach.

thx,
Rahul








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Re: [OSL | CCIE_Voice] PSTN Phone not able to call BR2

[Roger Källberg]

Do a debug voice dialpeer and debug voice translation and post the out-put from 
these.

Sincerely

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


Från: Support [supp...@technisha.com]
Skickat: den 12 mars 2011 09:30
Till: ccie_voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] PSTN Phone not able to call BR2

Dear All,

I need your help , From my PSTN Phone I am not able to call Br2 IP Phone,

PSTN Phone-PSTN GWH323 -Br2 GW-CUCM7--IP Phone (4001)

Please find my log on H323 GW and config below:

H323 Br2 GW ==

voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 sip
!
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8
!
voice class h323 1
 h225 timeout tcp establish 3
 h225 timeout setup 3
!
!
voice translation-rule 1
 rule 1 // // type any unknown plan any isdn
!
voice translation-rule 2
 rule 1 // // type any subscriber plan any isdn
!
voice translation-rule 4
 rule 1 // // type any international plan any isdn
!
voice translation-rule 5
 rule 1 /.*\(4...\)$/ /\1/
!
voice translation-rule 11
 rule 1 /^4...$/ /2404&/ type any subscriber plan any isdn
!
voice translation-rule 12
 rule 1 /^4...$/ /2404&/ type any subscriber plan any isdn
!
voice translation-rule 14
 rule 1 /^4...$/ /+8522404&/ type any international plan any isdn
!
!
voice translation-profile 999
 translate calling 11
 translate called 1
!
voice translation-profile did
 translate called 5
!
voice translation-profile int
 translate calling 14
 translate called 4
!
voice translation-profile local
 translate calling 12
 translate called 2
!

controller E1 0/0/0
 pri-group timeslots 1-4,16

!
interface Serial0/0/0:15
 no ip address
 encapsulation hdlc
 isdn switch-type primary-net5
 isdn incoming-voice voice
 isdn send-alerting
 isdn sending-complete
 isdn outgoing display-ie
 isdn outgoing ie redirecting-number
 no cdp enable

voice-port 0/0/0:15
 translation-profile incoming did

dial-peer voice 5 pots
 destination-pattern .
 direct-inward-dial
!
dial-peer voice 1 pots
 translation-profile outgoing 999
 destination-pattern 999
 no digit-strip
 port 0/0/0:15
!
dial-peer voice 11 voip
 destination-pattern 4...$
 voice-class codec 1
 voice-class h323 1
 session target ipv4:142.100.64.11
 dtmf-relay h245-alphanumeric
 no vad
!

==ISDN logs on BR2 H323 GW===

ar 12 08:09:15.956: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8  callref = 0x0091

Sending Complete
Sending Complete
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98381
Exclusive, Channel 1
Progress Ind i = 0x8183 - Origination address is non-ISDN
Display i = 'Nisha'
Calling Party Number i = 0x0180, '911'
Plan:ISDN, Type:Unknown
Called Party Number i = 0x81, '24044002'
Plan:ISDN, Type:Unknown
*Mar 12 08:09:15.976: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8  callref = 0x
8091
Channel ID i = 0xA98381
Exclusive, Channel 1
*Mar 12 08:09:15.976: ISDN Se0/0/0:15 Q931: TX -> CONNECT pd = 8  callref = 0x80
91
*Mar 12 08:09:15.992: ISDN Se0/0/0:15 Q931: RX <- CONNECT_ACK pd = 8  callref =
0x0091
*Mar 12 08:09:15.992: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now connected
to 911 N/A
*Mar 12 08:09:21.992: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now connected
to 911 N/A

It is showing connected on IP Phone and this log also but BR2-IP Phone not 
ringing.

Raj



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Re: [OSL | CCIE_Voice] Transcoder requirement for ICD and CUE

[Roger Källberg]

Looks correct to me. Also don't forget to take care of xcoder in any SRST 
situation, or anything else than xcoder for that matter. Easy thing to forget, 
but would be really bad for your score :)

** Sent from my iPhone. Excuse brevity and typographical errors. **

11 mar 2011 kl. 21:51 skrev "Rahul Kapor" 
mailto:rahul.kapo...@gmail.com>>:

Hi Roger ,

sorry for delay response.

Still one thing is not very clear for me

what is you allowed to configure on two gateway

to answer this question

1. enough DSP resource should be available  on both gateway to configure  3 
session of 3 transcoding session
2. for the xocoder which will be used for CUE , should be configured on BR2 
gateway
3. for the xocoder which will be used for ICD , should be configured on HQ  
gateway

Let me know if i am missing any  thing here

thx,
Rahul

2011/3/10 Roger Källberg 
<<mailto:roger.kallb...@cygate.se>roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>>
Hi Rahul,
I think your reading too much in to the question, or mis-intrepid it slightly. 
It doesn’t state anything about a restriction for that call to ICD only has to 
be originated from site BR1. The crucial part is to understand were a xcoder 
recourse would be needed, since the requirement say that your allowed to only 
configure it on two gateways.
Apart from this you probably also have would have a codec section that would 
state something similar to this.
Configure IP Phones and gateways in such as way that all calls within same site 
should use G711 codec. Also, all calls between the sites to remote IP phones 
and gateways should use G729 codec.
This in combination with what you wrote should help you answer your question.
Sincerely
Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB
Eric Perssons väg 21, SE-217 62 MALMÖ


Från: Rahul Kapor 
[<mailto:rahul.kapo...@gmail.com>rahul.kapo...@gmail.com<mailto:rahul.kapo...@gmail.com>]
Skickat: den 9 mars 2011 19:20
Till: Rogers Ochieng
Kopia: <mailto:ccie_voice@onlinestudylist.com> 
ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com>
Ämne: Re: [OSL | CCIE_Voice] Transcoder requirement for ICD and CUE

Hi Roger ,

I agree with your answer  but let say i am putting HQ and ICD in HQ device pool

if Br1 call ICD , transcoder will be invoked ,
but transcoder will be invoked if BR2 user is calling ICD , is this  breaking 
the answer ? i am not very sure about this.

thx,
Rahul

On Wed, Mar 9, 2011 at 10:32 AM, Rogers Ochieng 
<<mailto:rogersochi...@gmail.com>rogersochi...@gmail.com<mailto:rogersochi...@gmail.com>>
 wrote:
Why would you need all those regions? ICD is in HQ Device Pool so traffic 
within the same region will be G711, CUE will be on the Br2 device pool so 
g7711 with thin that region. Since interegion codec is g729 calls from BR1 will 
cause transcoder to be invoked

On 8 March 2011 22:49, Rahul Kapor 
<<mailto:rahul.kapo...@gmail.com>rahul.kapo...@gmail.com<mailto:rahul.kapo...@gmail.com>>
 wrote:


Site b should be able to call ICD route point number 2400 , using g729

hq and site B IP phone shoud be able to call unity express voice mail usig g729

we are allowed to configure 3 transcoding session per route2
also , we need to configure only one two router by looking at the requirement . 
marks wont be given it
transcoder is configured on all IOS router.

My answer would be

create 5 device pool and hence five diff region

ie HQ , BR , ICD , CUE and TRA and apply to respective device

to full fill the 1st requirement,

I will apply the transcoder to ICD device pool and will make sure that Region 
TRANS and ICD talk over g711
 and Region  BR and ICD talk over g729 (this will invokde the transcoder)

and for HQ and ICD , I will keep g711 so transcoder wont be invoked



same approach for 2nd requirement

ie i will apply the transcoder on the CUE device pool

region setting HQ/BR and CUE will be g729

and Region setting btw CUE and transcoder will be g711


I am not sure about 3rd requirement ie what  will be resource (PVDM) in real lab

Please let me know about approach.

thx,
Rahul








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<http://www.ipexpert.com> www.ipexpert.com<http://www.ipexpert.com>




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Re: [OSL | CCIE_Voice] Backup Route Group in a Route List with H323

[Roger Källberg]

Shutdown the voice-port is in my opinion the best way to simulate a PSTN down. 
You also have to change the service parameter "Stop Routing on Unallocated 
Number Flag" to false, it's set to true default.

About this "I am not using a "voice class H323" on my dial-peers", voice class 
H323 only apply to voip DP, so it would have no bearing to pots DP. But you 
really should have one setup for the CUCM redundancy for incoming calls to CUCM.

Sincerely

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


Från: adam compton [com...@gmail.com]
Skickat: den 11 mars 2011 16:00
Till: 
Ämne: [OSL | CCIE_Voice] Backup Route Group in a Route List with H323

I have a Route List setup to go to a BR1 H323 gateway first, and then fall back 
to the HQ h323 gateway if BR1 is unavailable.  I set this up and calls to the 
first gateway work fine.  The problem is, when I shut down the dial-peer for 
that call in the BR1 gateway, the call fails instead of going back to the 
backup gateway.

Is that the proper way to test failover to another gateway in this situation?  
I know if I shut down the Serial interface connecting back to the HQ site the 
failover works.

I am not using a "voice class H323" on my dial-peers.  Is there a certain H323 
class configuration to make this work?

Thanks for you help!

Adam Compton___
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Re: [OSL | CCIE_Voice] CUCM vs QOS SRNDs

[Roger Källberg]

Both source of values would be accepted, this is based on that all calculations 
on the lab typically have a 10% error margin. So if the outcome of your 
calculation is with-in 10% of what the intended answer say you should be good.

Sincerely

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


Från: George Goglidze [gogli...@gmail.com]
Skickat: den 11 mars 2011 12:59
Till: ShinGei Yong
Kopia: ccie_voice@onlinestudylist.com
Ämne: Re: [OSL | CCIE_Voice] CUCM vs QOS SRNDs

As both values are in Cisco's SRNDs which are both available on candidates 
desktop, both values should be accepted in my opinion.
and I mean of course the CCIE Voice lab exam not the real life network.



On Fri, Mar 11, 2011 at 11:44 AM, ShinGei Yong 
mailto:shingei.y...@gmail.com>> wrote:
Hi George,

When you said "both would be an acceptable answer on a lab",you mean real exam 
lab or practice lab?
Personally think there's only one acceptable value,but of course both might 
accepted during exam.

If not wrong (and this topic has been bring up multiple times in OSL or Ask the 
expert)Ben Ng has emphasized that
he would used QoS SRND as a reference guideline during the exam.

I would used different value suggest by various expert for practicing,but for 
real exam, will only use QoS SRND
reference. If not proctor not agree with the value defined in the SRDN, i'll 
just tell him then he shouldn't be left
this on candidate desktop.

Just my opinion,hope i won't kick out by Cisco after all.

Shingei.



On Fri, Mar 11, 2011 at 6:19 PM, George Goglidze 
mailto:gogli...@gmail.com>> wrote:
Hi all,

There is inconsistency between the two SRNDs on how much Layer 2 overhead is to 
be used for voice traffic calculations.

CUCM SRND states the following:

CODEC

Header Type and Size

Ethernet
14 Bytes

PPP
6 Bytes

ATM
53-Byte Cells with a 48-Byte Payload

Frame Relay
4 Bytes

MLPPP
10 Bytes

MPLS
4 Bytes

WLAN
24 Bytes



And QOS SRND states the following:

• 802.1Q Ethernet adds (up to) 32 bytes of Layer 2 overhead.
• Point-to-point protocol (PPP) adds 12 bytes of Layer 2 overhead.
• Multilink PPP (MLP) adds 13 bytes of Layer 2 overhead.
• Frame Relay adds 4 bytes of Layer 2 overhead; Frame Relay with FRF.12 adds 8 
bytes.
• ATM adds varying amounts of overhead, depending on the cell padding 
requirements

Which one do you use? I understand both would be an acceptable answer on a lab, 
but I'm just interested what most of the people use for their calculations.

Thanks for any answers.

George

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Re: [OSL | CCIE_Voice] Transcoder requirement for ICD and CUE

[Roger Källberg]

Hi Rahul,
I think your reading too much in to the question, or mis-intrepid it slightly. 
It doesn’t state anything about a restriction for that call to ICD only has to 
be originated from site BR1. The crucial part is to understand were a xcoder 
recourse would be needed, since the requirement say that your allowed to only 
configure it on two gateways.
Apart from this you probably also have would have a codec section that would 
state something similar to this.
Configure IP Phones and gateways in such as way that all calls within same site 
should use G711 codec. Also, all calls between the sites to remote IP phones 
and gateways should use G729 codec.
This in combination with what you wrote should help you answer your question.
Sincerely
Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB
Eric Perssons väg 21, SE-217 62 MALMÖ


Från: Rahul Kapor [rahul.kapo...@gmail.com]
Skickat: den 9 mars 2011 19:20
Till: Rogers Ochieng
Kopia: ccie_voice@onlinestudylist.com
Ämne: Re: [OSL | CCIE_Voice] Transcoder requirement for ICD and CUE

Hi Roger ,

I agree with your answer  but let say i am putting HQ and ICD in HQ device pool

if Br1 call ICD , transcoder will be invoked ,
but transcoder will be invoked if BR2 user is calling ICD , is this  breaking 
the answer ? i am not very sure about this.

thx,
Rahul

On Wed, Mar 9, 2011 at 10:32 AM, Rogers Ochieng 
mailto:rogersochi...@gmail.com>> wrote:
Why would you need all those regions? ICD is in HQ Device Pool so traffic 
within the same region will be G711, CUE will be on the Br2 device pool so 
g7711 with thin that region. Since interegion codec is g729 calls from BR1 will 
cause transcoder to be invoked

On 8 March 2011 22:49, Rahul Kapor 
mailto:rahul.kapo...@gmail.com>> wrote:


Site b should be able to call ICD route point number 2400 , using g729

hq and site B IP phone shoud be able to call unity express voice mail usig g729

we are allowed to configure 3 transcoding session per route2
also , we need to configure only one two router by looking at the requirement . 
marks wont be given it
transcoder is configured on all IOS router.

My answer would be

create 5 device pool and hence five diff region

ie HQ , BR , ICD , CUE and TRA and apply to respective device

to full fill the 1st requirement,

I will apply the transcoder to ICD device pool and will make sure that Region 
TRANS and ICD talk over g711
 and Region  BR and ICD talk over g729 (this will invokde the transcoder)

and for HQ and ICD , I will keep g711 so transcoder wont be invoked



same approach for 2nd requirement

ie i will apply the transcoder on the CUE device pool

region setting HQ/BR and CUE will be g729

and Region setting btw CUE and transcoder will be g711


I am not sure about 3rd requirement ie what  will be resource (PVDM) in real lab

Please let me know about approach.

thx,
Rahul








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


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

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

[Roger Källberg]

As other have answered, if needed you will be instructed in the lab material 
how these settings should be set. That's about all that anyone can say about 
this, and other stuff covered in the lab, without breaking the NDA.

As a side note, Vik Malhi and Amy Ryan recommendation about this on the 
boot-camp were that you always should do whatever the lab material states, 
pretty obvious :), and if nothing is stated you should set it to the proper 
value for that type of call. This applies to both Type of number (TON) and 
Number plan information (NPI). The later is mostly set to ISDN, if nothing else 
is said.

Sincerely

Roger Källberg
CCIE # 26199 (Voice)
Unified Communication Consultant
Cygate AB


From: Ccie Voice [mailto:v.c...@yahoo.com]
Sent: den 3 mars 2011 19:52
To: CCIE Study
Subject: Re: [OSL | CCIE_Voice] Calling and Called Party Number Type

Thank you all for your reply,

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

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

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

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

Regards,

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


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


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

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

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

This is correct

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

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

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

Long Distance, using BR1 Router

Calling Party Number Type to: Subscriber or National

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

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

Long Distance, backup for BR1 using HQ Router

Calling Party Number Type to: Subscriber or National

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

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

Regards,

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Re: [OSL | CCIE_Voice] Passed CCIE Voice #28291

[Roger Källberg]

Congratulate

Roger Källberg
CCIE #26199
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Re: [OSL | CCIE_Voice] Slips on E1

[Roger Källberg]

Please post appropriate parts of your config, that makes it so much easier to 
help you ;)

** Sent from my iPhone. Excuse brevity and typographical errors. **

24 feb 2011 kl. 19:20 skrev 
"ccie_voice-requ...@onlinestudylist.com"
 
mailto:ccie_voice-requ...@onlinestudylist.com>>:

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..."

Hi,

I have connected two routers Back-to-Back using the E1 Cross over cable, but I 
am unable to get the Multiple_Frame_Established.

I see on one of the side as slips are increasing, I tried to change the Clock 
source to Internal.


E1 0/1/0 is up.
Applique type is Channelized E1 - balanced
  No alarms detected.
  alarm-trigger is not set
  Version info Firmware: 20090113, FPGA: 20, spm_count = 0
  Framing is NO-CRC4, Line Code is HDB3, Clock Source is Internal.
  CRC Threshold is 320. Reported from firmware  is 320.
  Data in current interval (287 seconds elapsed):
 4 Line Code Violations, 1 Path Code Violations
 1 Slip Secs, 0 Fr Loss Secs, 2 Line Err Secs, 0 Degraded Mins
 3 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs


E1 0/2/0 is up.
  Applique type is Channelized E1 - balanced
  No alarms detected.
  alarm-trigger is not set
  Version info Firmware: 20090113, FPGA: 20, spm_count = 0
  Framing is NO-CRC4, Line Code is HDB3, Clock Source is Line Primary.
  CRC Threshold is 320. Reported from firmware  is 320.
  Data in current interval (137 seconds elapsed):
 0 Line Code Violations, 0 Path Code Violations
 29 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
 29 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs


Warm Regards,
Vinay Kumar

___
For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] Display name on CIPC

[Roger Källberg]

Well what I meant was that this is an Expert study "forum". So some effort on 
your own could be expected before posting a question.

You learn more if you try to come up with the solution on your own, than just 
posting a question.

Just my 2 cents

** Sent from my iPhone. Excuse brevity and typographical errors. **

21 feb 2011 kl. 15:44 skrev "Mike Thompson" 
mailto:mthompson...@gmail.com>>:

He probably just likes to hear people talk down to him.  Maybe that's why he 
asked the question in a educational forum.

Sent from my phone, apologies for any typos.

On Feb 21, 2011, at 5:09 AM, Roger Källberg 
<<mailto:roger.kallb...@cygate.se>roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>>
 wrote:

Yes you can use the + in EPNM, if that's what you meant?

As a side note, why don't you just try and see what happens?

** Sent from my iPhone. Excuse brevity and typographical errors. **

21 feb 2011 kl. 09:53 skrev "khaled Saholy" 
<<mailto:khaled_sah...@hotmail.com><mailto:khaled_sah...@hotmail.com>khaled_sah...@hotmail.com<mailto:khaled_sah...@hotmail.com>>:



I mean how can we display the plus sign on Cisco IP Phones?

Regards.

Khaled


From: <mailto:khaled_sah...@hotmail.com> <mailto:khaled_sah...@hotmail.com> 
<mailto:khaled_sah...@hotmail.com> 
khaled_sah...@hotmail.com<mailto:khaled_sah...@hotmail.com>
To: <mailto:roger.kallb...@cygate.se> <mailto:roger.kallb...@cygate.se> 
roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>; 
<mailto:ccie_voice@onlinestudylist.com> <mailto:ccie_voice@onlinestudylist.com> 
<mailto:ccie_voice@onlinestudylist.com> 
ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com>
Date: Sun, 20 Feb 2011 12:31:53 +0300
Subject: Re: [OSL | CCIE_Voice] Display name on CIPC



Is the + sign supported on the CIPC?

Khaled


> From: <mailto:roger.kallb...@cygate.se> <mailto:roger.kallb...@cygate.se> 
> <mailto:roger.kallb...@cygate.se> 
> roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>
> To: <mailto:ccie_voice@onlinestudylist.com> 
> <mailto:ccie_voice@onlinestudylist.com> 
> <mailto:ccie_voice@onlinestudylist.com> 
> ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com>
> CC: <mailto:ccie_voice@onlinestudylist.com> 
> <mailto:ccie_voice@onlinestudylist.com> 
> <mailto:ccie_voice@onlinestudylist.com> 
> ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com>
> Date: Sun, 20 Feb 2011 09:46:59 +0100
> Subject: Re: [OSL | CCIE_Voice] Display name on CIPC
>
> I really hope that this is not a serious question.
>
> Set EPNM, External Phone Number Mask.
>
> Roger Källberg
> CCIE #26199 (Voice)
>
> ** Sent from my iPhone. Excuse brevity and typographical errors. **
>
> 20 feb 2011 kl. 06:21 skrev 
> "<mailto:ccie_voice-requ...@onlinestudylist.com><mailto:ccie_voice-requ...@onlinestudylist.com>ccie_voice-requ...@onlinestudylist.com<mailto:ccie_voice-requ...@onlinestudylist.com>"
>  
> <<mailto:ccie_voice-requ...@onlinestudylist.com><mailto:ccie_voice-requ...@onlinestudylist.com>ccie_voice-requ...@onlinestudylist.com<mailto:ccie_voice-requ...@onlinestudylist.com>>:
>
> > Send CCIE_Voice mailing list submissions to
> > <mailto:ccie_voice@onlinestudylist.com> 
> > <mailto:ccie_voice@onlinestudylist.com> 
> > <mailto:ccie_voice@onlinestudylist.com> 
> > ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com>
> >
> > To subscribe or unsubscribe via the World Wide Web, visit
> > <http://onlinestudylist.com/mailman/listinfo/ccie_voice> 
> > <http://onlinestudylist.com/mailman/listinfo/ccie_voice> 
> > <http://onlinestudylist.com/mailman/listinfo/ccie_voice> 
> > http://onlinestudylist.com/mailman/listinfo/ccie_voice
> > or, via email, send a message with subject or body 'help' to
> > <mailto:ccie_voice-requ...@onlinestudylist.com> 
> > <mailto:ccie_voice-requ...@onlinestudylist.com> 
> > <mailto:ccie_voice-requ...@onlinestudylist.com> 
> > ccie_voice-requ...@onlinestudylist.com<mailto:ccie_voice-requ...@onlinestudylist.com>
> >
> > You can reach the person managing the list at
> > <mailto:ccie_voice-ow...@onlinestudylist.com> 
> > <mailto:ccie_voice-ow...@onlinestudylist.com> 
> > <mailto:ccie_voice-ow...@onlinestudylist.com> 
> > ccie_voice-ow...@onlinestudylist.com<mailto:ccie_voice-ow...@onlinestudylist.com>
> >
> > When replying, please edit your Subject line so it is more specific
> &

Re: [OSL | CCIE_Voice] BUG ????? QoS with MLP LFI

[Roger Källberg]

Please send appropriate parts of the configuration from the GW's.

** Sent from my iPhone. Excuse brevity and typographical errors. **

21 feb 2011 kl. 17:26 skrev 
"ccie_voice-requ...@onlinestudylist.com"
 
mailto:ccie_voice-requ...@onlinestudylist.com>>:

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..."

Hello Guys,

Does anybody  already encountered the problem with Virtual access  ??

Let me explain …..

As I configure qos with MLP LFI between hq and br2 everything was working fine 
!!!

After rebooting, no serial access anymore between hq and br2…..

With  show ip interface brief I have the following protocol state :

HQ
Interface IP-AddressOK? 
  Method   StatusProtocol
Virtual-Access1 10.10.112.1YES
TFTP up   down

BR2
Interface IP-AddressOK? 
  Method   StatusProtocol
Virtual-Access1 10.10.112.2YES
TFTP up   down

It’s not the first time !!! I tried to change the IOS but in vain.  The same 
behavior with IOS 12.4(20)T2,T1  and 12.4(24)T2.

If something like this happens during the lab, what can I do   Just ask the 
proctor and tell him that I have an unexpected behavior and that I’m sure about 
the qos config ?

I have usually this type of behavior after rebooting my routers.

Any comments or solutions would be appreciated.

Regards

Claude.

Claude Friderich
PreSales Support

NETCORE PSF S.A.
49 rue du Baerendall
B.P.65 L-8201 Mamer
Téléphone: 31 33 80-407
Fax: 31 33 80 8-407
GSM: 621 303 616
E-mail: cfrider...@netcore.lu


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privileged. It is intended solely for the addressee. Access to this Internet 
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Re: [OSL | CCIE_Voice] Display name on CIPC

[Roger Källberg]

If you meant for incoming calls, then no it's not supported on a CIPC. To 
display + on a hard phone like, 7965 you need to globalize the incoming calling 
party number on the VGW in CUCM. There are 4 fields for this on each VGW 
entity. If it's a MGCP GW, remember to do a no mgcp/mgcp in IOS to get the 
settings to activate.

** Sent from my iPhone. Excuse brevity and typographical errors. **

21 feb 2011 kl. 09:53 skrev "khaled Saholy" 
mailto:khaled_sah...@hotmail.com>>:



I mean how can we display the plus sign on Cisco IP Phones?

Regards.

Khaled


From: <mailto:khaled_sah...@hotmail.com> 
khaled_sah...@hotmail.com<mailto:khaled_sah...@hotmail.com>
To: roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>; 
<mailto:ccie_voice@onlinestudylist.com> 
ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com>
Date: Sun, 20 Feb 2011 12:31:53 +0300
Subject: Re: [OSL | CCIE_Voice] Display name on CIPC



Is the + sign supported on the CIPC?

Khaled


> From: <mailto:roger.kallb...@cygate.se> 
> roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>
> To: <mailto:ccie_voice@onlinestudylist.com> 
> ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com>
> CC: <mailto:ccie_voice@onlinestudylist.com> 
> ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com>
> Date: Sun, 20 Feb 2011 09:46:59 +0100
> Subject: Re: [OSL | CCIE_Voice] Display name on CIPC
>
> I really hope that this is not a serious question.
>
> Set EPNM, External Phone Number Mask.
>
> Roger Källberg
> CCIE #26199 (Voice)
>
> ** Sent from my iPhone. Excuse brevity and typographical errors. **
>
> 20 feb 2011 kl. 06:21 skrev 
> "ccie_voice-requ...@onlinestudylist.com<mailto:ccie_voice-requ...@onlinestudylist.com>"
>  
> mailto:ccie_voice-requ...@onlinestudylist.com>>:
>
> > Send CCIE_Voice mailing list submissions to
> > <mailto:ccie_voice@onlinestudylist.com> 
> > ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com>
> >
> > To subscribe or unsubscribe via the World Wide Web, visit
> > <http://onlinestudylist.com/mailman/listinfo/ccie_voice> 
> > http://onlinestudylist.com/mailman/listinfo/ccie_voice
> > or, via email, send a message with subject or body 'help' to
> > <mailto:ccie_voice-requ...@onlinestudylist.com> 
> > ccie_voice-requ...@onlinestudylist.com<mailto:ccie_voice-requ...@onlinestudylist.com>
> >
> > You can reach the person managing the list at
> > <mailto:ccie_voice-ow...@onlinestudylist.com> 
> > ccie_voice-ow...@onlinestudylist.com<mailto:ccie_voice-ow...@onlinestudylist.com>
> >
> > When replying, please edit your Subject line so it is more specific
> > than "Re: Contents of CCIE_Voice digest..."
> > 
> > Hi guys,
> > I have been looking for the way a whole day that how to display the name on 
> > top line of CIPC, that CIPC registered on the CUCM. Does anybody know how 
> > to do that? Thanks a million.
> > 
> > 
> ___
> For more information regarding industry leading CCIE Lab training, please 
> visit <http://www.ipexpert.com> www.ipexpert.com<http://www.ipexpert.com>

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

Re: [OSL | CCIE_Voice] Display name on CIPC

[Roger Källberg]

Yes you can use the + in EPNM, if that's what you meant?

As a side note, why don't you just try and see what happens?

** Sent from my iPhone. Excuse brevity and typographical errors. **

21 feb 2011 kl. 09:53 skrev "khaled Saholy" 
mailto:khaled_sah...@hotmail.com>>:



I mean how can we display the plus sign on Cisco IP Phones?

Regards.

Khaled


From: <mailto:khaled_sah...@hotmail.com> 
khaled_sah...@hotmail.com<mailto:khaled_sah...@hotmail.com>
To: roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>; 
<mailto:ccie_voice@onlinestudylist.com> 
ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com>
Date: Sun, 20 Feb 2011 12:31:53 +0300
Subject: Re: [OSL | CCIE_Voice] Display name on CIPC



Is the + sign supported on the CIPC?

Khaled


> From: <mailto:roger.kallb...@cygate.se> 
> roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>
> To: <mailto:ccie_voice@onlinestudylist.com> 
> ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com>
> CC: <mailto:ccie_voice@onlinestudylist.com> 
> ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com>
> Date: Sun, 20 Feb 2011 09:46:59 +0100
> Subject: Re: [OSL | CCIE_Voice] Display name on CIPC
>
> I really hope that this is not a serious question.
>
> Set EPNM, External Phone Number Mask.
>
> Roger Källberg
> CCIE #26199 (Voice)
>
> ** Sent from my iPhone. Excuse brevity and typographical errors. **
>
> 20 feb 2011 kl. 06:21 skrev 
> "ccie_voice-requ...@onlinestudylist.com<mailto:ccie_voice-requ...@onlinestudylist.com>"
>  
> mailto:ccie_voice-requ...@onlinestudylist.com>>:
>
> > Send CCIE_Voice mailing list submissions to
> > <mailto:ccie_voice@onlinestudylist.com> 
> > ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com>
> >
> > To subscribe or unsubscribe via the World Wide Web, visit
> > <http://onlinestudylist.com/mailman/listinfo/ccie_voice> 
> > http://onlinestudylist.com/mailman/listinfo/ccie_voice
> > or, via email, send a message with subject or body 'help' to
> > <mailto:ccie_voice-requ...@onlinestudylist.com> 
> > ccie_voice-requ...@onlinestudylist.com<mailto:ccie_voice-requ...@onlinestudylist.com>
> >
> > You can reach the person managing the list at
> > <mailto:ccie_voice-ow...@onlinestudylist.com> 
> > ccie_voice-ow...@onlinestudylist.com<mailto:ccie_voice-ow...@onlinestudylist.com>
> >
> > When replying, please edit your Subject line so it is more specific
> > than "Re: Contents of CCIE_Voice digest..."
> > 
> > Hi guys,
> > I have been looking for the way a whole day that how to display the name on 
> > top line of CIPC, that CIPC registered on the CUCM. Does anybody know how 
> > to do that? Thanks a million.
> > 
> > 
> ___
> For more information regarding industry leading CCIE Lab training, please 
> visit <http://www.ipexpert.com> www.ipexpert.com<http://www.ipexpert.com>

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

Re: [OSL | CCIE_Voice] Display name on CIPC

[Roger Källberg]

I really hope that this is not a serious question.

Set EPNM, External Phone Number Mask.

Roger Källberg
CCIE #26199 (Voice)

** Sent from my iPhone. Excuse brevity and typographical errors. **

20 feb 2011 kl. 06:21 skrev "ccie_voice-requ...@onlinestudylist.com" 
:

> 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..."
> 
> Hi guys,
>I have been looking for the way a whole day that how to display 
> the name on top line of CIPC, that CIPC registered on the CUCM. Does anybody 
> know how to do that? Thanks a million.
> 
> 
___
For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] SIP trunk to PSTN - CM 8.0 not sending SDP header in SIP INVITE packet

[Roger Källberg]

You will only get SDP if you enable SIP early offer. 

Sent from my iPhone

11 feb 2011 kl. 18:56 skrev "ccie_voice-requ...@onlinestudylist.com" 
:

> 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..."
> 
> Got CUCM 8.0 configured with a SIP trunk for PSTN calls. I've got "Media 
> Termination Point Required" enabled on the trunk but I've confirmed that CM 
> is not sending any SDP information to the PSTN, which is what they are 
> expecting. What am I missing?
>  
> Thanks,
> Mark
> 

___
For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 60, Issue 48

[Roger Källberg]

Read the url that I sent your way, that should be more than enough to get this 
straight. This is basic dial peer stuff that you really need to know if you 
ever will get your IE #.

Sincerely

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

Från: Shaik Dawood [shaik.daw...@cns-me.com]
Skickat: den 10 februari 2011 17:56
Till: Roger Källberg; ccie_voice@onlinestudylist.com
Ämne: RE: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 60, Issue 48

Can you send me the detailed config if possible, it will be a great help

-Original Message-
From: Roger Källberg [mailto:roger.kallb...@cygate.se]
Sent: Thursday, February 10, 2011 8:43 PM
To: Shaik Dawood; ccie_voice@onlinestudylist.com
Subject: SV: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 60, Issue 48

You need to change the dialpeer hunt preference selection.

See this url for more info,
http://cciev.wordpress.com/2006/05/29/dial-peer-hunting-choices/

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

Från: Shaik Dawood [shaik.daw...@cns-me.com]
Skickat: den 10 februari 2011 14:39
Till: ccie_voice@onlinestudylist.com
Ämne: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 60, Issue 48

I have some doubts in CME SRST. If call comes in SRST mode. It works. As
soon as it is configured for SRST . incoming calls hit the CME and busy
out, solution please

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of
ccie_voice-requ...@onlinestudylist.com
Sent: Thursday, February 10, 2011 7:27 AM
To: ccie_voice@onlinestudylist.com
Subject: CCIE_Voice Digest, Vol 60, Issue 48

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. Re: why is vad bad? (Brian Valentine)
   2. Voice Service doesn't show (Stern, Larry)
   3. Re: Phone service doesn't show (Baktha Muralidharan)


--

Message: 1
Date: Wed, 9 Feb 2011 20:57:24 -0500
From: Brian Valentine 
To: CCIE 
Cc: "ccie_voice@onlinestudylist.com" 
Subject: Re: [OSL | CCIE_Voice] why is vad bad?
Message-ID:

Content-Type: text/plain; charset="windows-1252"

Users complain.  It feels like you are using a walkie talkie.  Hard to
have
a conversation that

On Feb 9, 2011 1:34 PM, "CCIE"  wrote:
> You can see clipping at the beginning of people speaking after
silence.
>
> On Feb 9, 2011, at 8:44 AM, "Farkas P?ter" 
wrote:
>
>> Also can source voice quality issues like hissing.
>>
>> Peter
>>
>> - Original Message -
>> From: "matt...@ciscovoiceguru.com" 
>> Date: Wednesday, February 9, 2011 5:16 pm
>> Subject: Re: [OSL | CCIE_Voice] why is vad bad?
>> To: "Stutz, Bernhard" 
>> Cc: ccie_voice@onlinestudylist.com
>>
>>
>>> I've always understood that VAD results in a higher CPU utilization.
For
a site of 10 phones
>>> running a 2921 it wouldn't be an issue. However, if you're running
several hundred (or
>>> thousand) users running off the same pool of devices then you'd run
into
a significant impact
>>> on CPU performance.
>>>
>>> Matthew Berry, CCIE #26721
>>>
>>> Email: matt...@ciscovoiceguru.com
>>> Twitter:
>>> Blog:
>>>
>>> On Feb 9, 2011, at 9:38 AM, Stutz, Bernhard wrote:
>>>
>>>> Hi,
>>>>
>>>> i am just wondering why vad is bad and we all learn as a rule of
thumb
to disable vad on all
>>> voip dial peers?
>>>>
>>>> When you have a look for what vad has been designed for it looks to
me
as a valuable
>>> algorithm (
>>>>
>>>> Whats the reason we disable it all the time?
>>>> Is Cisco not able to support vad correctly or is it user experience
that they want to hear a
>>> noise otherwise they think of that the connection has been lost? But
therefore you have
>>> comfort-noise isn?t it?
>>>>
>>>> Kindly regards,
>>>> Bernhard
>>>>
>>>> ___
>>>> For more informa

Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 60, Issue 48

[Roger Källberg]

You need to change the dialpeer hunt preference selection.

See this url for more info,
http://cciev.wordpress.com/2006/05/29/dial-peer-hunting-choices/

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

Från: Shaik Dawood [shaik.daw...@cns-me.com]
Skickat: den 10 februari 2011 14:39
Till: ccie_voice@onlinestudylist.com
Ämne: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 60, Issue 48

I have some doubts in CME SRST. If call comes in SRST mode. It works. As
soon as it is configured for SRST . incoming calls hit the CME and busy
out, solution please

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of
ccie_voice-requ...@onlinestudylist.com
Sent: Thursday, February 10, 2011 7:27 AM
To: ccie_voice@onlinestudylist.com
Subject: CCIE_Voice Digest, Vol 60, Issue 48

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
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Today's Topics:

   1. Re: why is vad bad? (Brian Valentine)
   2. Voice Service doesn't show (Stern, Larry)
   3. Re: Phone service doesn't show (Baktha Muralidharan)


--

Message: 1
Date: Wed, 9 Feb 2011 20:57:24 -0500
From: Brian Valentine 
To: CCIE 
Cc: "ccie_voice@onlinestudylist.com" 
Subject: Re: [OSL | CCIE_Voice] why is vad bad?
Message-ID:

Content-Type: text/plain; charset="windows-1252"

Users complain.  It feels like you are using a walkie talkie.  Hard to
have
a conversation that

On Feb 9, 2011 1:34 PM, "CCIE"  wrote:
> You can see clipping at the beginning of people speaking after
silence.
>
> On Feb 9, 2011, at 8:44 AM, "Farkas P?ter" 
wrote:
>
>> Also can source voice quality issues like hissing.
>>
>> Peter
>>
>> - Original Message -
>> From: "matt...@ciscovoiceguru.com" 
>> Date: Wednesday, February 9, 2011 5:16 pm
>> Subject: Re: [OSL | CCIE_Voice] why is vad bad?
>> To: "Stutz, Bernhard" 
>> Cc: ccie_voice@onlinestudylist.com
>>
>>
>>> I've always understood that VAD results in a higher CPU utilization.
For
a site of 10 phones
>>> running a 2921 it wouldn't be an issue. However, if you're running
several hundred (or
>>> thousand) users running off the same pool of devices then you'd run
into
a significant impact
>>> on CPU performance.
>>>
>>> Matthew Berry, CCIE #26721
>>>
>>> Email: matt...@ciscovoiceguru.com
>>> Twitter:
>>> Blog:
>>>
>>> On Feb 9, 2011, at 9:38 AM, Stutz, Bernhard wrote:
>>>
>>>> Hi,
>>>>
>>>> i am just wondering why vad is bad and we all learn as a rule of
thumb
to disable vad on all
>>> voip dial peers?
>>>>
>>>> When you have a look for what vad has been designed for it looks to
me
as a valuable
>>> algorithm (
>>>>
>>>> Whats the reason we disable it all the time?
>>>> Is Cisco not able to support vad correctly or is it user experience
that they want to hear a
>>> noise otherwise they think of that the connection has been lost? But
therefore you have
>>> comfort-noise isn?t it?
>>>>
>>>> Kindly regards,
>>>> Bernhard
>>>>
>>>> ___
>>>> For more information regarding industry leading CCIE Lab training,
please visit www.ipexpert.com
>>>
>>> ___
>>> For more information regarding industry leading CCIE Lab training,
please visit www.ipexpert.com
>> ___
>> For more information regarding industry leading CCIE Lab training,
please
visit www.ipexpert.com
> ___
> For more information regarding industry leading CCIE Lab training,
please
visit www.ipexpert.com
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--

Message: 2
Date: Wed, 9 Feb 2011 21:40:48 -0500
From: "Stern, Larry" 
To: 
Subject: [OSL | CCIE_Voice] Voice Service doesn't show
Message-ID: 
C

Re: [OSL | CCIE_Voice] Digit Manipulation on H323

[Roger Källberg]

Hi,
You can get digit manipulation to shown on the display of a IP Phone for calls 
to a H.323 gw just as with an MGCP gw.

Add this to you H.323 config,

voice service voip
 no supplementary-service h225-notify cid-update

Sincerely

Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB

Från: Goran Selthofer [seltho...@gmail.com]
Skickat: den 17 oktober 2010 20:54
Till: Cristobal Priego
Kopia: ccie_voice@onlinestudylist.com
Ämne: Re: [OSL | CCIE_Voice] Digit Manipulation on H323

Hi C.P,

Digit manipulation will be done on CUCM and will be sent to H323 as well, and 
the preference would be on the manipulations done within RL rather than on RP.
So, i.e.
RP is 91608.[2-9]XX and
- if you put pre-dot and prefix 608 under RP,
- and then you also do pre-dot and prefix 9 for specific RG (for your h323 gw) 
under RL,

 then your h323 gw will receive 9[2-9]XX

hence, dial-peer pots on your h323 gw needed to terminate this call should have 
the same/similar destination-pattern configured, i.e:

dial-peer voice 9 pots
destination-pattern 9[2-9]..$
port 0/1/0:23


Now, the real trick comes if you want to actually influence your calling phone 
LCD digit presentations of DNIS (so, not ANI on the receiving end, but the 
actual dialed number on the calling end being presented on your phone from 
which you are dialing those digits - this is where the difference between mgcp 
and h323 gw can be seen).

mgcp will present whatever manipulations you've done using RP (will not present 
back to calling phone LCD what you have done withing RG/RL manipulations though 
it will use those manipulations to send to the GW).

however, in case of h323 gw, manipulations on DNIS done withing RG/RL will be 
also presented back to calling phone LCD.
Now, since that is H323, you can still have one more chance to do your digits 
manipulations and influence back presenting of dialed digits to calling phone - 
voice transformation rules/profiles attached to pots dial-peer (or 
forward-digits under dial-peer but that one will not influence LCD DNIS 
presentation on the calling phone)

i.e. if for above example we want to actually show 9 in front of local number, 
we can just put 'forward-digits 7' under above pots and that's it.
dial-peer voice 9 pots
destination-pattern 9[2-9]..$
port 0/1/0:23
forward-digits 7

But, if we would like to show ONLY local number, without leading 9 back to the 
caller on his ip phone LCD, then we would have to strip that 9 inside voice 
translation-rule, i.e:

voice translation-rule 9
 rule 1 /^9\([2-9]..$\)/ /\1/ p any sub t any sub

voice translation-profile 9
 translate called 9

and then add that to above dp:

dial-peer voice 9 pots
 translation-profile out 9
destination-pattern 9[2-9]..$
port 0/1/0:23

so this will result in showing only 7 digits back to LCD of the calling phone. 
(if dialed number was 91234567, it will show back only 1234567).

here, you can also include forward-digits as well, but translation-profile will 
still have precedence

dial-peer voice 9 pots
 translation-profile out 9
destination-pattern 9[2-9]..$
port 0/1/0:23
forward 7


in both cases you are sending 7 digits to PSTN, just the difference is what you 
will present back to the caller who actually dialed this number.

and that is the difference with mgcp, as you don't have that extra step to 
manipulate DNIS - all needs to be done on the CUCM withing RP, RG of RL or 
CalledPartTransformationPattern attached to outgoing mgcp gw.


hope this will give you some clues how it works...

cheers,
G.



On Sun, Oct 17, 2010 at 8:17 PM, Cristobal Priego 
mailto:cristobalpri...@gmail.com>> wrote:
hello all,

I'm working on the workbook 1 lab 5

and i noticed what when i do digit manipulation either on the RP, RL or by 
using transformation patterns, the changes aren't sent to the GW, if my 
protocol is H.323 usually I need to create some dial rules on the Voice Gateway

when I'm using MGCP i have no problem

i was wondering if there is a setting on the ccm that will allow ccm to send 
the digit manipulation to the GW or does it has to be manually done at the GW 
level ?

could you please explain a bit for me

thank you

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

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

Re: [OSL | CCIE_Voice] Vol2 Lab7 cue aar

[Roger Källberg]

Hi David,
Sounds like your call never gets to the VGW. Have you verified that your RP 
used for AAR will match to send the call to the gateway? You might also want to 
verify that you have reset the RL used, pretty common thing to forget.

What kind of VGW do you use, H.323 or MGCP. With the first you might want to 
run "debug voip dialpeer", to see what ingress dp that is used. That is if the 
call even gets sent to the VGW. If not the problem is within the UCM. Might be 
pretty obvious, but have you activated AAR in the SP?

About AAR group, you should only set that on the line, setting it on the device 
might cause some unpredicted behaviour.

Sincerely
Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB

Från: David A [david.a...@gmail.com]
Skickat: den 17 oktober 2010 16:10
Till: Roger Källberg
Kopia: ccie_voice@onlinestudylist.com
Ämne: Re: [OSL | CCIE_Voice] Vol2 Lab7 cue aar

Hi Roger,

I have the EPNM on the CTI ports set to the CTI-RP/VM pilot ie 3600 and EPMN is 
02077353600. I have no idea why the call doesnt pass thru to the gateway.

Thanks,
DA




2010/10/17 Roger Källberg 
mailto:roger.kallb...@cygate.se>>
You need to set the EPNM on the CTI ports to point to the number of the CTI RP 
for CUE. This is since the call can not go directly to the CTI ports, it has to 
first be sent to the CTI RP, then on to the CTI port.

Sincerely

Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB

Från: David A [david.a...@gmail.com<mailto:david.a...@gmail.com>]
Skickat: den 16 oktober 2010 19:18
Till: ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com>
Ämne: [OSL | CCIE_Voice] Vol2 Lab7 cue aar

Hi all,


I always get a busy when I configure AAR for cue and dial from HQ or SiteB.

I have aar group on all the phones and lines. cue external mask is same as the 
sietc phones. cti ports and cti rp have aar css and aar group. I do not see the 
call go out of any of the gateways.

Anyone face similar issues.


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

Re: [OSL | CCIE_Voice] Vol2 Lab7 cue aar

[Roger Källberg]

You need to set the EPNM on the CTI ports to point to the number of the CTI RP 
for CUE. This is since the call can not go directly to the CTI ports, it has to 
first be sent to the CTI RP, then on to the CTI port.

Sincerely

Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB

Från: David A [david.a...@gmail.com]
Skickat: den 16 oktober 2010 19:18
Till: ccie_voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] Vol2 Lab7 cue aar

Hi all,


I always get a busy when I configure AAR for cue and dial from HQ or SiteB.

I have aar group on all the phones and lines. cue external mask is same as the 
sietc phones. cti ports and cti rp have aar css and aar group. I do not see the 
call go out of any of the gateways.

Anyone face similar issues.


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

Re: [OSL | CCIE_Voice] CME phones and max number of supported calls

[Roger Källberg]

Hi Mark,
I might have misunderstood what you want to do but if I understand your 
question correct you need to use a combination of "huntstop channel x" on the 
ephone-dn and "busy-trigger-per-button x" on the ephone to get the asked for 
behaviour.

For example like this,

ephone-dn  1  dual-line
 number 3001 no-reg both
 label SC Phone 1 - 3001
 description +442077353001
 name SC Phone 1
 call-forward busy 3220
 call-forward noan 3220 timeout 20
!
!
ephone-dn  2  dual-line
 number 3002 no-reg both
 label SC Phone 2 - 3002
 description +442077353002
 name SC Phone 2
 call-forward busy 3220
 call-forward noan 3220 timeout 20
!
!
ephone-dn  3  octo-line
 number 3003 no-reg both
 label SC Share - 3003
 name SC Share
 huntstop channel 5
!
!
ephone  1
 privacy off
 privacy-button
 device-security-mode none
 mac-address 6416.8D50.AD4B
 ephone-template 1
 busy-trigger-per-button 2
 username "SCPh1" password wds85067
 type 7962
 button  1:1 2:3
!
!
!
ephone  2
 privacy off
 device-security-mode none
 mac-address 6416.8D50.AD63
 ephone-template 1
 busy-trigger-per-button 4
 username "SCPh2" password fln05997
 type 7962
 button  1:2 2:3 3w1



Sincerely
Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB

Från: Vik Malhi [vma...@ipexpert.com]
Skickat: den 10 oktober 2010 00:29
Till: Mark Holloway; OSL Group
Ämne: Re: [OSL | CCIE_Voice] CME phones and max number of supported calls

Is that 3 and 6 calls to the shared line or 3/6 calls cumulatively?

The restrictions imposed on the number of calls to a CME phone is done "per
line/button" assigned to a device as opposed to being a blanket "per device"
setting. So I think the cumulative option is not possible.
--
Vik Malhi ­ CCIE #13890
Managing Partner / Instructor - IPexpert, Inc.
Mailto: vma...@ipexpert.com
Telephone: +1.810.326.1444 ext 420
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/>



> From: Mark Holloway 
> Date: Sat, 9 Oct 2010 14:24:05 -0700
> To: OSL Group 
> Subject: [OSL | CCIE_Voice] CME phones and max number of supported calls
>
> If I want limit BR2Ph1 to 3 incoming calls and BR2Ph2 to 6 incoming calls, how
> can I control the total number of incoming calls to each phone if there is
> more than one ephone-dn assign to the phone?  For example, if 6001 is an octo
> line assigned to Ph1, 6002 is an octo line assigned to Ph2, and 6003 is an
> octo line shared on both phones.
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com

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

Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

[Roger Källberg]

Region will overwrite the default, so to work around the "fix" for the bug you 
need to specify the codec to G711 to be used within the region local to the 
phone/VGW/and so on.

Sincerely

Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB

Från: Stutz, Bernhard [st...@pandacom.de]
Skickat: den 5 oktober 2010 16:09
Till: Roger Källberg; David A; ccie_voice@onlinestudylist.com
Ämne: AW: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

If you are changing the IntraAudioRegionDefault to G.729 you will fix that but 
you will then break the requirement to have G.711 for intra region calls. isn't 
it?
Or will in this case the Region Setting overwrite the default setting?

cheers,
Bernhard


Von: ccie_voice-boun...@onlinestudylist.com im Auftrag von Roger Källberg
Gesendet: Di 05.10.2010 14:21
An: David A; ccie_voice@onlinestudylist.com
Betreff: Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

Your hitting bug CSCsl74701. This is a well known bug that you should be really 
familiar with. There are many posts on the OSL about this and also Matthew 
Barry has an excellent post on his blog about this. See this url, 
http://ciscovoiceguru.com/382/cscsl74701-bug-details/

Sincerely

Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB

Från: David A [david.a...@gmail.com]
Skickat: den 4 oktober 2010 22:43
Till: ccie_voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

Hi All,

I am doing the Vol2 Lab2 GK scenario and running into a couple of issues.

issue 1 - Call from CME SCCP phone to CUCM phones is using 128kbps. When I 
check the codec used on the call on both phones it says g729. The gk-tunk is in 
DP GK with region g729 to everyone.

2811-HQ-GW#sh gatekeeper call
Total number of active calls = 1.
 GATEKEEPER CALL INFO
 
LocalCallIDAge(secs)   BW
25-49659   21  128(Kbps) <--- should be 
16kbps as per the requirement
 Endpt(s): Alias E.164Addr
   src EP: SiteC-GW  3003
   CallSignalAddr  Port  RASSignalAddr   Port
   10.10.110.3 1720  10.10.110.3 58555
 Endpt(s): Alias E.164Addr
   dst EP: gk-trunk_21#1002
   CallSignalAddr  Port  RASSignalAddr   Port
   10.137.151.26   1720  10.137.151.26   33447


issue 2 - Call from CUCM to CME SIP phone(g711)  works fine and invokes the 
transcoder and i see a 16kbps GK call. However when I call from CME SIP phone 
to any CUCM phone, CUCM phone rings and I can answer it. However it drops after 
a few seconds and I see no transcoder being used. Here are my configs

Site C -

voice register pool  1
 id mac 0025.4593.0368
 type 7975
 number 1 dn 1
 number 2 dn 2
 template 1
 description 32143002
 codec g711ulaw

!
dial-peer voice 15 voip
 destination-pattern [15]...$
 session target ras
 incoming called-number .
 tech-prefix 1#
 dtmf-relay h245-alphanumeric rtp-nte
!
dial-peer voice 3000 voip
 incoming called-number 3...$
 dtmf-relay h245-alphanumeric
!

Any clues?

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

Re: [OSL | CCIE_Voice] BACD drop through not working on 2821

[Roger Källberg]

You can find the document I referred to on cisco.com -> support - Voice and 
Unified Communication -> IP Telephony -> Call Control -> Cisco Unified 
Communication Manager Express -> Configuration Guides

Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB

Från: Tamer Ismail [tih...@gmail.com]
Skickat: den 5 oktober 2010 12:26
Till: CCIE_Voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] BACD drop through not working on 2821

Hello,
I want to configure BACD drop through on router 2821, when I call the
pilot number, the call is drops.
Can someone check and tell me, may I made something wrong.

Router:
2800 Software (C2800NM-IPVOICEK9-M), c2800nm-ipvoicek9-mz.124-22.YB7.bin

Flash:
10   30421 Oct 04 2010 11:37:44   app-b-acd-3.0.0.2.tcl
11   55599 Oct 04 2010 11:37:46   app-b-acd-aa-3.0.0.2.tcl
12   75650 Oct 04 2010 11:37:48   en_bacd_allagentsbusy.au
13   83291 Oct 04 2010 11:37:50   en_bacd_disconnect.au
14   63055 Oct 04 2010 11:37:50   en_bacd_enter_dest.au
15   37952 Oct 04 2010 11:37:52   en_bacd_invalidoption.au
16  496521 Oct 04 2010 11:38:04  en_bacd_music_on_hold.au
17  123446 Oct 04 2010 11:38:08  en_bacd_options_menu.au
18   42978 Oct 04 2010 11:38:10   en_bacd_welcome.au

Configuration:
!
application
 service aa flash:app-b-acd-aa-3.0.0.2.tcl
  param number-of-hunt-grps 1
  paramspace english index 0
  param drop-through-option 1
  param handoff-string aa
  paramspace english language en
  param max-time-vm-retry 2
  param aa-pilot 6299
  paramspace english location flash:
  param second-greeting-time 30
  param drop-through-prompt en_bacd_welcome.au
  param call-retry-timer 15
  param max-time-call-retry 700
  param service-name callq
 !
 service callq flash:app-b-acd-3.0.0.2.tcl
  param queue-len 10
  param aa-hunt1 6300
  param queue-manager-debugs 1
  param number-of-hunt-grps 1
 !
dial-peer voice 6299 voip
 service aa
 destination-pattern 6299
 session target ipv4:10.5.21.2
 incoming called-number 6299
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad
!
ephone-hunt 10 longest-idle
 pilot 6300
 list 6201, 6202
 timeout 10, 10
!

Thanks for help.
Tamer,

___
For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] BACD drop through not working on 2821

[Roger Källberg]

Your missing some needed parameters, see this config example from the document 
"Cisco Unified CME B-ACD and Tcl Call-Handling Applications". This can be found 
on Cisco support and will be available to you in the lab.


Cisco Unified CME B-ACD with Drop-Through Option: Example
The following example sets parameters for an AA service called aa and a 
call-queue service called callq. The direct-dial number to reach the AA service 
is 800 555-0100. Callers to this number drop through to the ephone hunt group 
that has a pilot number of 5071 after hearing the initial prompt from the file 
en_dt_prompt.au.

ephone-hunt 10 sequential
 pilot 5071
 list 5011, 5012, 5013, 5014, 5015
 timeout 10

application
 service callq tftp://192.168.254.254/user1/CallQ/B-ACD/app-b-acd.tcl
  param queue-manager-debugs 1
  param aa-hunt1 5071
  param number-of-hunt-grps 1
  param queue-len 10

 service aa tftp://192.168.254.254/user1/CallQ/B-ACD/app-b-acd-aa.tcl
  paramspace english location tftp://192.168.254.254/user1/prompts/
  paramspace english index 0
  paramspace english language en
  param aa-pilot 8005550100
  param number-of-hunt-grps 1
  param service-name callq
  param handoff-string aa
  param second-greeting-time 60
  param drop-through-option 1
  param drop-through-prompt _dt_prompt.au
  param call-retry-timer 15
  param max-time-call-retry 700
  param voice-mail 5000
  param max-time-vm-retry 2


Sincerely

Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB

Från: Tamer Ismail [tih...@gmail.com]
Skickat: den 5 oktober 2010 12:26
Till: CCIE_Voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] BACD drop through not working on 2821

Hello,
I want to configure BACD drop through on router 2821, when I call the
pilot number, the call is drops.
Can someone check and tell me, may I made something wrong.

Router:
2800 Software (C2800NM-IPVOICEK9-M), c2800nm-ipvoicek9-mz.124-22.YB7.bin

Flash:
10   30421 Oct 04 2010 11:37:44   app-b-acd-3.0.0.2.tcl
11   55599 Oct 04 2010 11:37:46   app-b-acd-aa-3.0.0.2.tcl
12   75650 Oct 04 2010 11:37:48   en_bacd_allagentsbusy.au
13   83291 Oct 04 2010 11:37:50   en_bacd_disconnect.au
14   63055 Oct 04 2010 11:37:50   en_bacd_enter_dest.au
15   37952 Oct 04 2010 11:37:52   en_bacd_invalidoption.au
16  496521 Oct 04 2010 11:38:04  en_bacd_music_on_hold.au
17  123446 Oct 04 2010 11:38:08  en_bacd_options_menu.au
18   42978 Oct 04 2010 11:38:10   en_bacd_welcome.au

Configuration:
!
application
 service aa flash:app-b-acd-aa-3.0.0.2.tcl
  param number-of-hunt-grps 1
  paramspace english index 0
  param drop-through-option 1
  param handoff-string aa
  paramspace english language en
  param max-time-vm-retry 2
  param aa-pilot 6299
  paramspace english location flash:
  param second-greeting-time 30
  param drop-through-prompt en_bacd_welcome.au
  param call-retry-timer 15
  param max-time-call-retry 700
  param service-name callq
 !
 service callq flash:app-b-acd-3.0.0.2.tcl
  param queue-len 10
  param aa-hunt1 6300
  param queue-manager-debugs 1
  param number-of-hunt-grps 1
 !
dial-peer voice 6299 voip
 service aa
 destination-pattern 6299
 session target ipv4:10.5.21.2
 incoming called-number 6299
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad
!
ephone-hunt 10 longest-idle
 pilot 6300
 list 6201, 6202
 timeout 10, 10
!

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

Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

[Roger Källberg]

Your hitting bug CSCsl74701. This is a well known bug that you should be really 
familiar with. There are many posts on the OSL about this and also Matthew 
Barry has an excellent post on his blog about this. See this url, 
http://ciscovoiceguru.com/382/cscsl74701-bug-details/

Sincerely

Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB

Från: David A [david.a...@gmail.com]
Skickat: den 4 oktober 2010 22:43
Till: ccie_voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls

Hi All,

I am doing the Vol2 Lab2 GK scenario and running into a couple of issues.

issue 1 - Call from CME SCCP phone to CUCM phones is using 128kbps. When I 
check the codec used on the call on both phones it says g729. The gk-tunk is in 
DP GK with region g729 to everyone.

2811-HQ-GW#sh gatekeeper call
Total number of active calls = 1.
 GATEKEEPER CALL INFO
 
LocalCallIDAge(secs)   BW
25-49659   21  128(Kbps) <--- should be 
16kbps as per the requirement
 Endpt(s): Alias E.164Addr
   src EP: SiteC-GW  3003
   CallSignalAddr  Port  RASSignalAddr   Port
   10.10.110.3 1720  10.10.110.3 58555
 Endpt(s): Alias E.164Addr
   dst EP: gk-trunk_21#1002
   CallSignalAddr  Port  RASSignalAddr   Port
   10.137.151.26   1720  10.137.151.26   33447


issue 2 - Call from CUCM to CME SIP phone(g711)  works fine and invokes the 
transcoder and i see a 16kbps GK call. However when I call from CME SIP phone 
to any CUCM phone, CUCM phone rings and I can answer it. However it drops after 
a few seconds and I see no transcoder being used. Here are my configs

Site C -

voice register pool  1
 id mac 0025.4593.0368
 type 7975
 number 1 dn 1
 number 2 dn 2
 template 1
 description 32143002
 codec g711ulaw

!
dial-peer voice 15 voip
 destination-pattern [15]...$
 session target ras
 incoming called-number .
 tech-prefix 1#
 dtmf-relay h245-alphanumeric rtp-nte
!
dial-peer voice 3000 voip
 incoming called-number 3...$
 dtmf-relay h245-alphanumeric
!

Any clues?

Thanks,
DA___
For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] blocking 91900 pattern

[Roger Källberg]

This is the expected behaviour, SIP phones do not support the use of 
annunciator.

Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB

Från: Bhushan Paranjape [paranjape_bhus...@yahoo.com]
Skickat: den 3 oktober 2010 23:21
Till: Randall Saborio
Kopia: ccie_voice@onlinestudylist.com
Ämne: Re: [OSL | CCIE_Voice] blocking 91900 pattern

yes, I dont hear aanunciator from hq ph 2 (sip) ...call drops after 5th digit 
is dialed but i can hear the annunciator from br1 ph 2 (sccp)


--- On Sun, 10/3/10, Randall Saborio  wrote:

From: Randall Saborio 
Subject: Re: [OSL | CCIE_Voice] blocking 91900 pattern
To: "Bhushan Paranjape" 
Cc: "Prashant Patel" , 
ccie_voice@onlinestudylist.com
Date: Sunday, October 3, 2010, 5:08 PM

Is the problem that you don't hear annunciator?

On Sun, Oct 3, 2010 at 1:25 PM, Bhushan Paranjape 
http://us.mc658.mail.yahoo.com/mc/compose?to=paranjape_bhus...@yahoo.com>>
 wrote:

yes, Br1 ph 2 is sccp and it works

but hq ph 2 is sip which doesnt work.
--- On Sun, 10/3/10, Prashant Patel 
http://us.mc658.mail.yahoo.com/mc/compose?to=prashantpatel...@gmail.com>>
 wrote:

From: Prashant Patel 
http://us.mc658.mail.yahoo.com/mc/compose?to=prashantpatel...@gmail.com>>
Subject: Re: [OSL | CCIE_Voice] blocking 91900 pattern

To: 
ccie_voice@onlinestudylist.com<http://us.mc658.mail.yahoo.com/mc/compose?to=ccie_vo...@onlinestudylist.com>
Date: Sunday, October 3, 2010, 1:41 PM


You running a SIP firmware?

I have seen this happen. Try with an SCCP phone with pattern 91900! and 
"Urgent" checked.

HTH

On Sun, Oct 3, 2010 at 1:19 PM, 
http://us.mc658.mail.yahoo.com/mc/compose?to=ccie_voice-requ...@onlinestudylist.com>>
 wrote:
Send CCIE_Voice mailing list submissions to
   
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than "Re: Contents of CCIE_Voice digest..."


Today's Topics:

  1. blocking 91900 pattern (Bhushan Paranjape)
  2. Re: Single Number Reach (Mark Holloway)


--

Message: 1
Date: Sun, 3 Oct 2010 10:00:40 -0700 (PDT)
From: Bhushan Paranjape 
http://us.mc658.mail.yahoo.com/mc/compose?to=paranjape_bhus...@yahoo.com>>
To: 
ccie_voice@onlinestudylist.com<http://us.mc658.mail.yahoo.com/mc/compose?to=ccie_vo...@onlinestudylist.com>
Subject: [OSL | CCIE_Voice] blocking 91900 pattern
Message-ID: 
<633989.54711...@web65807.mail.ac4.yahoo.com<http://us.mc658.mail.yahoo.com/mc/compose?to=633989.54711...@web65807.mail.ac4.yahoo.com>>
Content-Type: text/plain; charset="iso-8859-1"

Hi,
?
I am working on WB1 lab 5.6. I have a 91900 RP that is blocked and annunciator 
set to "precedence level exceeded"
?
Problem is when I call 919004522138 from br1 ph 2 I can hear the annunciatot 
after i dial the 5th digit, but when I call it from hq ph 2 my call gets 
dropped after i dial the 5th digit. (all other calls are working fom hq ph2)
?
I am using all hardware phones (7961)
?
Any idea?
?
TIA,
?
Bhushan.



-- next part --
An HTML attachment was scrubbed...
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--

Message: 2
Date: Sun, 3 Oct 2010 10:18:54 -0700
From: Mark Holloway 
http://us.mc658.mail.yahoo.com/mc/compose?to...@markholloway.com>>
To: Roger K?llberg 
http://us.mc658.mail.yahoo.com/mc/compose?to=roger.kallb...@cygate.se>>
Cc: CCIE Voice Maillist 
http://us.mc658.mail.yahoo.com/mc/compose?to=ccie_vo...@onlinestudylist.com>>
Subject: Re: [OSL | CCIE_Voice] Single Number Reach
Message-ID: 
<0cf714d1-8e56-4478-8b59-9d3104899...@markholloway.com<http://us.mc658.mail.yahoo.com/mc/compose?to=0cf714d1-8e56-4478-8b59-9d3104899...@markholloway.com>>
Content-Type: text/plain; charset="iso-8859-1"

Hmm, for Single Number Reach?  If a call comes in to HQ3 and simultaneously 
rings the PSTN/SNR number, and the PSTN answers the calls, I believe there is a 
way for the HQ3 phone to display on its screen that it is "in use".  Turning 
off Privacy for HQ3 didn't change that behavior. :/



On Oct 3, 2010, at 4:30 AM, Roger K?llberg wrote:

> Turn off privacy
>
> Roger K?llberg
> CCIE # 26199 (Voice)
> Unified Communication Consultant
> Cygate AB
>
>
> From: 

Re: [OSL | CCIE_Voice] Single Number Reach

[Roger Källberg]

Turn off privacy on the RDP.

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

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>

Från: Mark Holloway [...@markholloway.com]
Skickat: den 3 oktober 2010 19:18
Till: Roger Källberg
Kopia: Graham Hopkins; CCIE Voice Maillist
Ämne: Re: [OSL | CCIE_Voice] Single Number Reach

Hmm, for Single Number Reach?  If a call comes in to HQ3 and simultaneously 
rings the PSTN/SNR number, and the PSTN answers the calls, I believe there is a 
way for the HQ3 phone to display on its screen that it is "in use".  Turning 
off Privacy for HQ3 didn't change that behavior. :/



On Oct 3, 2010, at 4:30 AM, Roger Källberg wrote:

Turn off privacy

Roger Källberg
CCIE # 26199 (Voice)
Unified Communication Consultant
Cygate AB


From: Mark Holloway [mailto:m...@markholloway.com]
Sent: den 2 oktober 2010 06:24
To: Graham Hopkins
Cc: CCIE Voice Maillist
Subject: Re: [OSL | CCIE_Voice] Single Number Reach

Is there a specific setting to force the ip phone to display an "in use" 
message in the event the pstn phone answers the incoming call?


On Oct 1, 2010, at 11:42 AM, Graham Hopkins wrote:


Same here , I was beginning to think that no patterns are matched in calling 
number transformations - but I tested with a pattern of ! and a  mask of 12345 
and that works.

So it would appear that there is a mismatch between \+1480.! and the calling 
number, which does seem odd as if you leave it alone it gets sent to the PSTN 
as +1480XXX. It would appear that it should match as the pattern ! with 
XXX works, but as Mark says this doesn't do what he requires


Graham



On 1 Oct 2010, at 19:23, Mark Holloway wrote:


The only issue with this is you don't know if the calling party is Subscriber, 
National, or International, so you can't use XXX because if BR2 or BR1 
calls HQ3 the From number would only show the first 7 digits.


On Oct 1, 2010, at 11:21 AM, sisiaji wrote:


yeah, you are right, I was referring to RP/RL transformations...

i tested it and i got the same in my lab

so i guess, as you already mentioned before, the way to do it is to actually 
put Calling Party Transform Mask to be XXX on the RL (for RG member).


On Fri, Oct 1, 2010 at 7:52 PM, Mark Holloway 
mailto:m...@markholloway.com>> wrote:
When doing it under Call Routing > Transformation Pattern > Calling Party 
Transformation you have to use \+

When doing it on the Calling Party transform mask on a Route Pattern or Route 
List you don't use \


On Oct 1, 2010, at 10:44 AM, sisiaji wrote:


calling party transformation is done without prefix \




On Fri, Oct 1, 2010 at 7:00 PM, Mark Holloway 
mailto:m...@markholloway.com>> wrote:
The crazy thing is I tried this but I couldn't get it to work.

PT_HQ_CALLING_OUT assigned to CSS_HQ_CALLING_OUT assigned to CALLING Number 
Transform on the Outbound portion of the HQ gateway.

Calling Party Transformation: PT_HQ_CALLING_OUT pattern is \+1480.!(replace 
480 with what your HQ area code is)

Strip Predot

That should make the outbound From number +14805552001 appear as 5552001 on the 
PSTN phone. and I should see 5552001 in the isdn q931 debug output.  I'm still 
seeing the full E164 number.


On Oct 1, 2010, at 9:22 AM, Graham Hopkins wrote:


Well I'm just showing the full E.164 as that's what the lab I'm looking at 
looks for. However I guess you could strip the HQ area code at the gateway with 
the calling party transformation.

In the real world  (plan to visit that soon) then the remote destination is 
likely to be a mobile phone which isn't really local to any gateway - at least 
not here in the UK so would be a national call from anywhere.



Graham



On 1 Oct 2010, at 17:10, Mark Holloway wrote:


Sorry, I meant Translation Patterns, not Profiles.  Still working on the From 
number presentation.  I'm assuming that if HQ1 calls HQ3 the PSTN phone should 
show a 7 digit From number, but if BR1 calls 2003 the PSTN should show a 10 
digit From number.  Would you guys agree?



On Oct 1, 2010, at 8:56 AM, Mark Holloway wrote:


Graham, same thing here.

This is a summary of what I've done to get it working correctly. I eliminated 
using Translation Profiles as I didn't find them necessary for this.

Create PT_SNR which is assigned to CSS_SNR

Create a Remote Destination Profile and assign CSS_SNR to both Calling Search 
Space and Rerouting Calling Search Space.  Build/associate your end user with 
this Remote Destination Profile. Build a Route List (RL_SNR) that includes just 
the HQ gateway and set the Calling Party External Phone Mask to On.  Doing this 
in the Route Pattern won't work. Set Called Party to Subscriber (assuming the 
Remote Destination number is a local number).  Lastly, build a Route Pattern 

Re: [OSL | CCIE_Voice] Single Number Reach

[Roger Källberg]

Turn off privacy

Roger Källberg
CCIE # 26199 (Voice)
Unified Communication Consultant
Cygate AB


From: Mark Holloway [mailto:m...@markholloway.com]
Sent: den 2 oktober 2010 06:24
To: Graham Hopkins
Cc: CCIE Voice Maillist
Subject: Re: [OSL | CCIE_Voice] Single Number Reach

Is there a specific setting to force the ip phone to display an "in use" 
message in the event the pstn phone answers the incoming call?


On Oct 1, 2010, at 11:42 AM, Graham Hopkins wrote:


Same here , I was beginning to think that no patterns are matched in calling 
number transformations - but I tested with a pattern of ! and a  mask of 12345 
and that works.

So it would appear that there is a mismatch between \+1480.! and the calling 
number, which does seem odd as if you leave it alone it gets sent to the PSTN 
as +1480XXX. It would appear that it should match as the pattern ! with 
XXX works, but as Mark says this doesn't do what he requires


Graham



On 1 Oct 2010, at 19:23, Mark Holloway wrote:


The only issue with this is you don't know if the calling party is Subscriber, 
National, or International, so you can't use XXX because if BR2 or BR1 
calls HQ3 the From number would only show the first 7 digits.


On Oct 1, 2010, at 11:21 AM, sisiaji wrote:


yeah, you are right, I was referring to RP/RL transformations...

i tested it and i got the same in my lab

so i guess, as you already mentioned before, the way to do it is to actually 
put Calling Party Transform Mask to be XXX on the RL (for RG member).


On Fri, Oct 1, 2010 at 7:52 PM, Mark Holloway 
mailto:m...@markholloway.com>> wrote:
When doing it under Call Routing > Transformation Pattern > Calling Party 
Transformation you have to use \+

When doing it on the Calling Party transform mask on a Route Pattern or Route 
List you don't use \


On Oct 1, 2010, at 10:44 AM, sisiaji wrote:


calling party transformation is done without prefix \




On Fri, Oct 1, 2010 at 7:00 PM, Mark Holloway 
mailto:m...@markholloway.com>> wrote:
The crazy thing is I tried this but I couldn't get it to work.

PT_HQ_CALLING_OUT assigned to CSS_HQ_CALLING_OUT assigned to CALLING Number 
Transform on the Outbound portion of the HQ gateway.

Calling Party Transformation: PT_HQ_CALLING_OUT pattern is \+1480.!(replace 
480 with what your HQ area code is)

Strip Predot

That should make the outbound From number +14805552001 appear as 5552001 on the 
PSTN phone. and I should see 5552001 in the isdn q931 debug output.  I'm still 
seeing the full E164 number.


On Oct 1, 2010, at 9:22 AM, Graham Hopkins wrote:


Well I'm just showing the full E.164 as that's what the lab I'm looking at 
looks for. However I guess you could strip the HQ area code at the gateway with 
the calling party transformation.

In the real world  (plan to visit that soon) then the remote destination is 
likely to be a mobile phone which isn't really local to any gateway - at least 
not here in the UK so would be a national call from anywhere.



Graham



On 1 Oct 2010, at 17:10, Mark Holloway wrote:


Sorry, I meant Translation Patterns, not Profiles.  Still working on the From 
number presentation.  I'm assuming that if HQ1 calls HQ3 the PSTN phone should 
show a 7 digit From number, but if BR1 calls 2003 the PSTN should show a 10 
digit From number.  Would you guys agree?



On Oct 1, 2010, at 8:56 AM, Mark Holloway wrote:


Graham, same thing here.

This is a summary of what I've done to get it working correctly. I eliminated 
using Translation Profiles as I didn't find them necessary for this.

Create PT_SNR which is assigned to CSS_SNR

Create a Remote Destination Profile and assign CSS_SNR to both Calling Search 
Space and Rerouting Calling Search Space.  Build/associate your end user with 
this Remote Destination Profile. Build a Route List (RL_SNR) that includes just 
the HQ gateway and set the Calling Party External Phone Mask to On.  Doing this 
in the Route Pattern won't work. Set Called Party to Subscriber (assuming the 
Remote Destination number is a local number).  Lastly, build a Route Pattern 
that matches your Remote Destination Profile external number and assign it to 
PT_SNR and RL_SNR.

The only thing about this method is that when calls from 2001 ring 2003 which 
rings the PSTN, this method is using the external mask which means HQ1's 
external mask is E164. Typically when a Subscriber call egresses the HQ gateway 
you would want the From number to be 7 digits. Are you guys putting a Calling 
Party Transformation on your HQ gateway to strip off the HQ area code for 
Subscriber calls?  For all other purposes of presenting 7, 10, or E164, I have 
always used the Calling Party Transform in either the Route Pattern or Route 
List's Route Group.


Thanks,
Mark


On Oct 1, 2010, at 7:25 AM, Graham Hopkins wrote:


Just hit the same problem in Vol2 Lab4 and I can confirm that this d

Re: [OSL | CCIE_Voice] Pass

[Roger Källberg]

Congratulations Jeff!

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

Från: Jeff Cotter [jcot...@voxns.com]
Skickat: den 21 september 2010 23:50
Till: ccie_voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] Pass

Finally….took more times than I care to admit!  A big thanks to IPexpert 
(especially Vic) and everybody who has been a part of this list.

Jeff Cotter
CCIE #27033


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

Re: [OSL | CCIE_Voice] CCIE Voice PASSED !!!!!!!

[Roger Källberg]

Congratulations Tom!

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

Från: Tom [tom.c...@gmail.com]
Skickat: den 19 september 2010 10:26
Till: ccie_voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] CCIE Voice PASSED !!!

I wanted to let everyone know that I passed my CCIE Voice Today!!
 I couldnt believe it!
 I wanted to thank everyone from this list that helped out in my studying 
forthe last one Year.
 Thank you all again___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Need opinions / recommendations on Remote Phone Control (UNCLASSIFIED)

[Roger Källberg]

I have used Phone Remote 2.1 from VoIP Intergration without any problems, it 
does what you can expect from a remote control tool.

Currently we use it at a customer that has about 10.000 phones spread across 9 
differnt clusters, and there have been no real issues that I can remember. Only 
problem we have had was that someone, it's alway that person :-), made a change 
to the authentication url on one of the clusters. That completly messed up the 
ability to control any phone in that particular cluster.

Sincerely

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

Från: Jeff Garvas [j...@cia.net]
Skickat: den 19 september 2010 07:14
Till: Girard, Jeffrey T COL RET
Kopia: ccie_voice@onlinestudylist.com
Ämne: Re: [OSL | CCIE_Voice] Need opinions / recommendations on Remote Phone 
Control (UNCLASSIFIED)

I had problems getting Remote Phone 2.1 to work properly.  It required an AXL 
account I believe, and I was in the stage you're in (trying it out) and lost 
focus on testing it.  I'm sure if properly configured it works well, but I 
can't say that for sure.

Singlewire seemed more interested in selling me a corporate license that I 
couldn't justify, but their product looked better to me for some reason (I 
can't recall why).   I specifically asked them if they'd consider selling me a 
single-use copy for study purposes and it seemed to go nowhere.

I look forward to your results because one of my desks keeps moving around and 
I'd rather find a way to not need the phones at my desk at work.

-Jeff


On Fri, Sep 17, 2010 at 1:27 PM, Girard, Jeffrey T COL RET 
mailto:jeffrey.gir...@us.army.mil>> wrote:
Classification: UNCLASSIFIED
Listers -
 I have my own home lab, but spend quite a bit of time away from home on 
travel.  I have been researching 3rd party apps to allow me to remotely control 
my hardphones in my home lab from my hotel room so that I can continue my 
studies.

 I have come across two products - RemotePhoneControl by Singlewire (formerly 
Berbee) and Remote Phone 2.1 from VoIP Integration.

 Im hoping that there is no one on the list who works for either company - as I 
have no intention of this post turning into a vendor debate.

 Looking for anyone with experiences (pros and cons) of either package.

Jeff

Dr. Jeffrey T. Girard (Jeff), PhD
Senior Network Engineer / VoIP Enginner - WireMeHappy.com
Senior Network Engineer / Operations Specialist - EnSync Interactive Solutions
reply to: jeffrey.gir...@wiremehappy.com<mailto:jeffrey.gir...@wiremehappy.com>
  jeffrey.gir...@us.army.mil<mailto:jeffrey.gir...@us.army.mil>
(845)977-0363 (home office)
(845)764-1661 (mobile)
Classification: UNCLASSIFIED
___
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www.ipexpert.com<http://www.ipexpert.com>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] SRST to UC VoiceMail

[Roger Källberg]

Hi all,
You might want to practise to setup this without the use of alternate extension 
in CUC, so that you also have experience with that way of configuration.
Just a friendly tip J
Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB
Eric Perssons väg 21, SE-217 62 MALMÖ

Från: Amy Ryan [ar...@ipexpert.com]
Skickat: den 16 september 2010 17:09
Till: ayman labib; ccie_voice@onlinestudylist.com
Ämne: Re: [OSL | CCIE_Voice] SRST to UC VoiceMail

You will need to add an alternate extension for the voicemail user in Unity 
Connection of the expanded number as it comes in from the PSTN.  If you are 
globalizing numbers, you will need to use the full e164 number as your 
alternate extension.

---
Amy Ryan – CCIE #24677 (Voice)
Technical Instructor - IPexpert, Inc.
Mailto: 
ar...@ipexpert.com<,DanaInfo=.aey4BwynGlxwlu0N645.,SSL+UrlBlockedError.aspx>
Telephone: +1.810.326.1444
Live Assistance, Please visit: www.ipexpert.com/chat 
<http://www.ipexpert.com/chat>
eFax: +1.810.454.0130

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 
www.ipexpert.com/>




From: ayman labib 
>
Date: Thu, 16 Sep 2010 07:53:18 -0700 (PDT)
To: 
>
Cc: 
>
Subject: [OSL | CCIE_Voice] SRST to UC VoiceMail

Greetings,

SRST to Unity Connection VM question.  I put my site in SRST, Everything works 
as expected except when I hit the message button to go to my VM box in UC.  It 
goes to the general UC message.  Under my HQ gateway, I have Redirecting number 
IE inbound and outbound checked.  Am I missing any step?  Any ideas or 
suggestion is greatly appreciated

My config:

interface Serial0/1/0:23
 no ip address
 isdn switch-type primary-ni
 isdn incoming-voice voice
 isdn outgoing display-ie
 isdn outgoing ie redirecting-number
 no cdp enable
!
call-manager-fallback
 secondary-dialtone 9
 max-conferences 8 gain -6
 transfer-system full-consult
 ip source-address 192.168.31.10 port 2000 strict-match
 max-ephones 10
 max-dn 10
 transfer-pattern .T
 voicemail 912123945020
 call-forward pattern .T
 call-forward busy 912123945020
 call-forward noan 912123945020 timeout 20
 moh music-on-hold.au
 multicast moh 239.1.1.1 port 16384 route 192.168.31.10 192.168.31.10
 time-zone 8
!



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

Re: [OSL | CCIE_Voice] One button login - Phone service

[Roger Källberg]

Not sure if this is the only error, but one of your parameters are setup wrong, 
you need to add Pwd with a capital P.

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

Från: KatGuru [gkr2...@yahoo.com]
Skickat: den 15 september 2010 21:09
Till: Tanner Ezell
Kopia: ccie_voice@onlinestudylist.com
Ämne: Re: [OSL | CCIE_Voice] One button login - Phone service

[file:///C:/DOCUME%7E1/KATHIR%7E1.GUR/LOCALS%7E1/Temp/moz-screenshot-1.png][file:///C:/DOCUME%7E1/KATHIR%7E1.GUR/LOCALS%7E1/Temp/moz-screenshot-3.png]Please
 find the attached.

Thanks.
[file:///C:/DOCUME%7E1/KATHIR%7E1.GUR/LOCALS%7E1/Temp/moz-screenshot-2.png]
--- On Thu, 9/16/10, Tanner Ezell  wrote:

From: Tanner Ezell 
Subject: Re: [OSL | CCIE_Voice] One button login - Phone service
To: "KatGuru" 
Cc: ccie_voice@onlinestudylist.com, "Amy Ryan" 
Date: Thursday, September 16, 2010, 2:17 AM

Cool, ok. Would you mind including some screen shots of the configuration pages 
in UCM, so the phone services page, the configuration on the phone itself (or 
the udp, however its configured).

On Wed, Sep 15, 2010 at 11:05 AM, KatGuru 
> 
wrote:
This is what i got in the browser:

This XML file does not appear to have any style information associated with it. 
The document tree is shown below.


Error
Internal error: Invalid input.

−

OK
Key:Services
1



--- On Thu, 9/16/10, Tanner Ezell 
>
 wrote:

From: Tanner Ezell 
>

Subject: Re: [OSL | CCIE_Voice] One button login - Phone service
To: "KatGuru" 
>
Cc: 
ccie_voice@onlinestudylist.com<,DanaInfo=.aey4BwynGlxwlu0N645.,SSL+UrlBlockedError.aspx>,
 "Amy Ryan" 
>
Date: Thursday, September 16, 2010, 1:40 AM


When you visit that URL does it return anything?

On Wed, Sep 15, 2010 at 10:37 AM, KatGuru 
http://mc/compose?to=gkr2...@yahoo.com>> wrote:
Yes I did. I've Extn, ID and password.

--- On Thu, 9/16/10, Amy Ryan 
http://mc/compose?to=ar...@ipexpert.com>> wrote:

From: Amy Ryan http://mc/compose?to=ar...@ipexpert.com>>
Subject: Re: [OSL | CCIE_Voice] One button login - Phone service
To: "KatGuru" http://mc/compose?to=gkr2...@yahoo.com>>, 
ccie_voice@onlinestudylist.com<http://mc/compose?to=ccie_vo...@onlinestudylist.com>
Date: Thursday, September 16, 2010, 12:55 AM


When you added the service did you also add the parameters?

---
Amy Ryan – CCIE #24677 (Voice)
Technical Instructor - IPexpert, Inc.
Mailto: ar...@ipexpert.com
Telephone: +1.810.326.1444
Live Assistance, Please visit: 
www.ipexpert.com/chat<http://www.ipexpert.com/chat> 
<http://www.ipexpert.com/chat>
eFax: +1.810.454.0130

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> 
<http://www.ipexpert.com/communities>  and our public website at 
www.ipexpert.com<http://www.ipexpert.com> 
<http://www.ipexpert.com/<http://www.ipexpert.com/>>




From: KatGuru 
Date: Wed, 15 Sep 2010 09:01:46 -0700 (PDT)
To: 
Subject: [OSL | CCIE_Voice] One button login - Phone service

Folks, i need a help for this simple task (But not now)

I'm configuring Phone Agent One button login and i got the error Internal 
error: Invalid input after i press the services-->One button login in IPC.  I 
checked the user, extn and password everything seems to be correct. I deleted 
the service, recreate, unsubscribe, subscribe but still no luck.

http://xxx.xxx.xxx.xx:6293/ipphone/jsp/sciphonexml/IPAgentLogin.jsp

Please let me know if  i missed something.

Thank you.



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Re: [OSL | CCIE_Voice] I've got a new favorite number!

[Roger Källberg]

Congratulations Mike.

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

Från: groganhockey [groganhoc...@gmail.com]
Skickat: den 15 september 2010 15:16
Till: OSL Group
Ämne: [OSL | CCIE_Voice] I've got a new favorite number!

26966!

I'm not sure how often it'll come up in everyday life, but there it is.

I took the lab Monday in RTP and finally got my score report last night.

IPExpert/Vik/Amy: Thank you for the excellent study guides, walkthroughs, audio 
and everything!

Mike
CCIE #26966 Voice


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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Setting Up the Lab

[Roger Källberg]

AFAIK, you can modify the base configuration files from IPX to suite your 
needs. I know some of my study mates that had their own lab kit did this.

Sincerely

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

Från: Sriharshaa Prabhakar [sriharshaa.prabha...@mannai.com.qa]
Skickat: den 11 september 2010 15:09
Till: Roger Källberg; ccie_voice@onlinestudylist.com
Ämne: RE: [OSL | CCIE_Voice] CCIE_Voice Digest, Setting Up the Lab

Hi Roger,

Thanks a lot for this input, with reference to the first question, is it really 
required to load the Initial Configs for the IP Expert Labs. Yes I understand 
that the PSTN Router has to be configured with the Dial Plan but in my case I 
may not be able to use the Initial Configs on the HQRTR, BR1RTR, BR2RTR due to 
the fact that the IP Addressing Schema is different.

Kindly give me your recommendations.

Sriharshaa Prabhakar
Technical Lead - Cisco Unified Communications
Networks & Telecom Division

-Original Message-
From: Roger Källberg [mailto:roger.kallb...@cygate.se]
Sent: Saturday, September 11, 2010 3:57 PM
To: Sriharshaa Prabhakar; ccie_voice@onlinestudylist.com
Subject: SV: [OSL | CCIE_Voice] CCIE_Voice Digest, Setting Up the Lab

Hi Sriharshaa
See this response I got from Vik Malhi about this.

"Post 12.4(15)T8 the gk license is mandatory (and also very expensive) and we 
are trying to get in touch with Cisco for a GK license they can issue to 
training providers (NFR). Until that point we will not run anything later than 
12.4(15)T8 on HQ-RTR since there are no new GK related commands a later IOS 
release will offer."

I hope this helps you out.

Sincerely

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


Från: Sriharshaa Prabhakar [sriharshaa.prabha...@mannai.com.qa]
Skickat: den 11 september 2010 11:43
Till: ccie_voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] CCIE_Voice Digest, Setting Up the Lab

Hi All,

I am trying to setup my Lab, I have setup the PSTNRTR, HQRTR, BR1RTR & BR2RTR, 
however my configs and the IP Addressing schema has to be little different 
because this Lab is on my production network for accessibility. I cannot use 
the config templates of IP Expert so I may have to redo the templates to my own 
lab or I just would like to know if I can go ahead and practice the lab without 
the Initial Configs from IP Expert given the scenario in each Lab?

The second question is I am running the GK on the PSTNRTR with IOS 12.4.22T 
however this requires a license, could any of you suggest is it really 
mandatory to run 12.4.22T becasue all later versions requires Licenses. I even 
tried with 12.4.24T3 and it is the same.

PSTNSRTR#show license
Index 1 Feature: gatekeeper
Period left: 6  weeks 6  days
License Type: Evaluation
License State: Active, In Use
License Priority: Low

Thanks in advance.

Regards,
Sriharshaa Prabhakar


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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Setting Up the Lab

[Roger Källberg]

Hi Sriharshaa 
See this response I got from Vik Malhi about this.

"Post 12.4(15)T8 the gk license is mandatory (and also very expensive) and we 
are trying to get in touch with Cisco for a GK license they can issue to 
training providers (NFR). Until that point we will not run anything later than 
12.4(15)T8 on HQ-RTR since there are no new GK related commands a later IOS 
release will offer."

I hope this helps you out.

Sincerely

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


Från: Sriharshaa Prabhakar [sriharshaa.prabha...@mannai.com.qa]
Skickat: den 11 september 2010 11:43
Till: ccie_voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] CCIE_Voice Digest, Setting Up the Lab

Hi All,

I am trying to setup my Lab, I have setup the PSTNRTR, HQRTR, BR1RTR & BR2RTR, 
however my configs and the IP Addressing schema has to be little different 
because this Lab is on my production network for accessibility. I cannot use 
the config templates of IP Expert so I may have to redo the templates to my own 
lab or I just would like to know if I can go ahead and practice the lab without 
the Initial Configs from IP Expert given the scenario in each Lab?

The second question is I am running the GK on the PSTNRTR with IOS 12.4.22T 
however this requires a license, could any of you suggest is it really 
mandatory to run 12.4.22T becasue all later versions requires Licenses. I even 
tried with 12.4.24T3 and it is the same.

PSTNSRTR#show license
Index 1 Feature: gatekeeper
Period left: 6  weeks 6  days
License Type: Evaluation
License State: Active, In Use
License Priority: Low

Thanks in advance.

Regards,
Sriharshaa Prabhakar


___
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Re: [OSL | CCIE_Voice] VM config for SIP phones in CME/CUE

[Roger Källberg]

There is an issue with this command in the version of CME that is used in the 
lab, this might not be the case in newer versions, but I've never tried it out 
on anything newer than CME 7.0.1. It simply won't do what it is supposed to do, 
ie enable multiple codec on a CME SIP phone. I have never seen an acctual bug 
id for this, but trust me it's a well known problem.

Look at IPX solution guide, and any other material from them about this, like 
the excellent Walk-trough videos from Vik Malhi or the great new Audio on 
demand that Amy Ryan did.

Simply put you need to hard code your CME SIP phones to use whatever intra site 
codec that is asked for, in most cases that would be G711(ulaw) and then force 
the codec to be used for inter site call to be G729. For this to work you'll 
need to have a transcoder that is registered local to the CME. That would also 
be a requirement for incoming intra site calls that goes to the CUE VM, since 
those calls will use G729 and CUE only supports G711.

Sincerely

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

Från: cisco voip [voip.ccieci...@gmail.com]
Skickat: den 11 september 2010 12:54
Till: Roger Källberg
Kopia: Randall Saborio; ccie_voice@onlinestudylist.com; Avinash Shukla
Ämne: Re: [OSL | CCIE_Voice] VM config for SIP phones in CME/CUE

Hey Roger,

Configuring voice class codec under voice register pool is the only way i could 
think of to make SIP phone to work with both CUE and to talk over wan.

If you don't configure codec under voice register pool, it will take g729 and 
will not work with CUE, if you configure codec g711ulaw, call will not work 
over wan.

and voice class codec always worked for me for SIP CME.

Comments please.


2010/9/11 Roger Källberg 
mailto:roger.kallb...@cygate.se>>
Although the command voice class codec is possible to use on a voice register 
pool it will not take effect. That's one of the small odditiesthat you simply 
need to be aware of when dealing with SIP CME phones.

Sincerely

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

Från: cisco voip [voip.ccieci...@gmail.com<mailto:voip.ccieci...@gmail.com>]
Skickat: den 11 september 2010 09:01
Till: Randall Saborio
Kopia: ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com>; 
Avinash Shukla
Ämne: Re: [OSL | CCIE_Voice] VM config for SIP phones in CME/CUE

Also voice register pool should have a voice class codec defined, which has 
both g711 and g729 in it

On Sat, Sep 11, 2010 at 3:03 AM, Randall Saborio 
mailto:ill2...@gmail.com>> wrote:
Another imoprtant one:

voice service voip
  allow-connections sip to sip




On Fri, Sep 10, 2010 at 7:22 AM, Amy Ryan 
mailto:ar...@ipexpert.com>> wrote:
Avinash,

Here is an example configuration for a SIP phone covering to VM in CUCME.

sip-ua
 mwi-server ipv4:10.10.202.2
!
voice register dn  2
 call-forward b2bua busy 3600
 call-forward b2bua noan 3600 timeout 12
mwi 
!
voice register pool  2
 dtmf-relay rtp-nte
!
voice register global
 voicemail 3600 
create profile
 reset
!
dial-peer voice 3600 voip
 destination-pattern 3600
 session protocol sipv2
 session target ipv4:10.10.202.2
 dtmf-relay sip-notify
 codec g711ulaw
 no vad
 incoming called-number 399[89]….

HTH,
Amy



---
Amy Ryan – CCIE #24677 (Voice)
Technical Instructor - IPexpert, Inc.
Mailto: ar...@ipexpert.com<http://ar...@ipexpert.com>
Telephone: +1.810.326.1444
Live Assistance, Please visit: 
www.ipexpert.com/chat<http://www.ipexpert.com/chat> 
<http://www.ipexpert.com/chat>
eFax: +1.810.454.0130

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> 
<http://www.ipexpert.com/communities>  and our public website at 
www.ipexpert.com<http://www.ipexpert.com> 
<http://www.ipexpert.com/<http://www.ipexpert.com/>>




From: Avinash Shukla 
http://avinashshukla.i...@gmail.com>>
Date: Fri, 10 Sep 2010 18:20:56 +0530
To: http://ccie_voice@onlinestudylist.com>>
Subject: [OSL | CCIE_Voice] VM config for SIP phones in CME/CUE


Hi Experts,

I wanted to know what configs do we need to do on the cme/cue to make VM work 
for SIP phones.

I have added user to CUE with extension and have also mentioned the Voicemail 
DN command in voice register pool. + the dial peer for voicemail.
When i do debug voip dialpeer all. I can see the dialpeer for Voicemail getting 
hit 

Re: [OSL | CCIE_Voice] Trunk group behavior on E1 failure

[Roger Källberg]

Hi Jess,
You need to use this command, "no dial-peer outbound status-check pots". Also 
read this post from Cisco support forum, 
https://supportforums.cisco.com/message/982276

Sincerely

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


Från: Jess Alcantara [jessica.alcant...@gmail.com]
Skickat: den 11 september 2010 09:02
Till: ccie_voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] Trunk group behavior on E1 failure

Hi all!
any advice will be appreciated,  I am trying to find out what would happen in a 
trunk group configuration  if one of the E1 interfaces goes down,  just tried 
to configure it and the gateway continued to deliver the calls to the E1 listed 
in the trunk even if they were down  and I am concern if this is normal or not 
. I have not been able to find any documentation describing this behavior.

This is the config I got..

Thanks in advance!!!



trunk group PSTN



controller E1 0/1/0
 framing NO-CRC4
 line-termination 75-ohm
 ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled ani
 cas-custom 0
  release-guard-time 2
  disconnect-tone
  category 2
  answer-signal group-b 1
  answer-guard-time 500
  timer interdigit incoming 1000
  groupa-callerid-end
  dtmf timer inter-digit 250
  trunk-group PSTN


controller E1 0/1/1
 framing NO-CRC4
 line-termination 75-ohm
 ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled ani
 cas-custom 0
  release-guard-time 2
  disconnect-tone
  category 2
  answer-signal group-b 1
  answer-guard-time 500
  timer interdigit incoming 1000
  groupa-callerid-end
  dtmf timer inter-digit 250
  trunk-group PSTN


Jess Alcántara
jessica.alcant...@gmail.com<mailto:jessica.alcant...@gmail.com>




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Re: [OSL | CCIE_Voice] VM config for SIP phones in CME/CUE

[Roger Källberg]

Although the command voice class codec is possible to use on a voice register 
pool it will not take effect. That's one of the small odditiesthat you simply 
need to be aware of when dealing with SIP CME phones.

Sincerely

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

Från: cisco voip [voip.ccieci...@gmail.com]
Skickat: den 11 september 2010 09:01
Till: Randall Saborio
Kopia: ccie_voice@onlinestudylist.com; Avinash Shukla
Ämne: Re: [OSL | CCIE_Voice] VM config for SIP phones in CME/CUE

Also voice register pool should have a voice class codec defined, which has 
both g711 and g729 in it

On Sat, Sep 11, 2010 at 3:03 AM, Randall Saborio 
mailto:ill2...@gmail.com>> wrote:
Another imoprtant one:

voice service voip
  allow-connections sip to sip




On Fri, Sep 10, 2010 at 7:22 AM, Amy Ryan 
mailto:ar...@ipexpert.com>> wrote:
Avinash,

Here is an example configuration for a SIP phone covering to VM in CUCME.

sip-ua
 mwi-server ipv4:10.10.202.2
!
voice register dn  2
 call-forward b2bua busy 3600
 call-forward b2bua noan 3600 timeout 12
mwi 
!
voice register pool  2
 dtmf-relay rtp-nte
!
voice register global
 voicemail 3600 
create profile
 reset
!
dial-peer voice 3600 voip
 destination-pattern 3600
 session protocol sipv2
 session target ipv4:10.10.202.2
 dtmf-relay sip-notify
 codec g711ulaw
 no vad
 incoming called-number 399[89]….

HTH,
Amy



---
Amy Ryan – CCIE #24677 (Voice)
Technical Instructor - IPexpert, Inc.
Mailto: ar...@ipexpert.com<http://ar...@ipexpert.com>
Telephone: +1.810.326.1444
Live Assistance, Please visit: 
www.ipexpert.com/chat<http://www.ipexpert.com/chat> 
<http://www.ipexpert.com/chat>
eFax: +1.810.454.0130

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> 
<http://www.ipexpert.com/communities>  and our public website at 
www.ipexpert.com<http://www.ipexpert.com> 
<http://www.ipexpert.com/<http://www.ipexpert.com/>>




From: Avinash Shukla 
http://avinashshukla.i...@gmail.com>>
Date: Fri, 10 Sep 2010 18:20:56 +0530
To: http://ccie_voice@onlinestudylist.com>>
Subject: [OSL | CCIE_Voice] VM config for SIP phones in CME/CUE


Hi Experts,

I wanted to know what configs do we need to do on the cme/cue to make VM work 
for SIP phones.

I have added user to CUE with extension and have also mentioned the Voicemail 
DN command in voice register pool. + the dial peer for voicemail.
When i do debug voip dialpeer all. I can see the dialpeer for Voicemail getting 
hit but i get nothin but the call does not completes!

Voice mail works just fine for SCCP phones but not for SIP.

Regards,
Avinash


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--
Randall "da ill" Saborio
CCIE Voice Wannabe #10054675811


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Re: [OSL | CCIE_Voice] CUCM Corporate Directory (BLF Presence Status)

[Roger Källberg]

Hi Mark,
Have you enabled the "BLF For Call Lists" in Enterprice Parameters?

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

Från: Mark Holloway [...@markholloway.com]
Skickat: den 11 september 2010 07:47
Till: osl osl
Ämne: [OSL | CCIE_Voice] CUCM Corporate Directory (BLF Presence Status)

I have HqPh1 and HqPh3 both assigned to the same SUBSCRIBE CSS and HqPh3 has 
BLF Speed Dial assigned to watch HqPh1's primary extension and everything works 
great on HqPh3's line key that watches HqPh1.  However, I am trying to access 
the corporate directory on HqPh3 and expect to see presence status for HqPh1's 
main number.  If HqPh1 goes off hook I am not seeing anything change in the 
corporate directory listing on HqPh3.  I have End Users created for my Hq 
phones and assigned their primary extension. The Hq phones are both in Standard 
Presence Group.  If I go to System > Presence Group and set Allow then restart 
the phone it doesn't make a difference.  Any suggestions?

Thanks,
Mark

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Re: [OSL | CCIE_Voice] Fwd: WB 2 LAB 2 Question 5.1 - Gatekeeper Bandwidth Accounting

[Roger Källberg]

Your hitting a known bug "CSCsl74701". Make a search for that on OSL and have a 
look at Matthew Berrys excellent post about this on his blog, 
http://ciscovoiceguru.com/382/cscsl74701-bug-details/

Sincerely

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

Från: Vccie Vccie [voiceccie2...@gmail.com]
Skickat: den 7 september 2010 03:04
Till: ccie_voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] Fwd: WB 2 LAB 2 Question 5.1 - Gatekeeper Bandwidth 
Accounting


First off, let me say I have looked for this in the past post's but didn't see 
anything so I am sorry if this is redundant.  But this is the problem I am 
having.

Calling between UCM and UCME works perfect for both Skinny and Sip phones. But 
the accounting in the Gatekeepr is showing wrong amounts on the calls coming 
from the UCME to the UCM.  But UCM to UCME calls show correct bandwidth 
accounting (16Kbps).  I know the codec used is G729r8 but for some reason it 
shows g711 bandwidth amounts.

Br2 (version 12.4(22)T)
 dial-peer voice 150 voip
 destination-pattern [51]...
 session target ras
 tech-prefix 1#
 dtmf-relay h245-alphanumeric rtp-nte
 no vad

Gatekeeper (version 12.4.(20).T4)
UCM to UCME = 16Kbps
UCME to UCM  = 128Kbps
 After debuging H225 ans1 messages I can see the following
  admissionRequest - from 3002 to 1002  bandWidth 160
  admissionConfirm - bandWidth 160
  admissionRequest - from 1002 to 3002 bandWidth 1280
  admissionConfirm - bandWidth 1280
  infoRequestResponse - bandWidth 160

UCM - 7.01 (H225 Gatekeeper controlled trunk)
 HQ/BR1/BR2 - each have different Device Pools with G729 intra-device pool 
Codec.

-- So it's the UCM that is responding with a G711 capability's but the call is 
actually using G729 -- so I am stuck..(and phones show g729 being used)  Any 
help is appropriated.






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Re: [OSL | CCIE_Voice] 2 or in some case 3 stage dialing - issue

[Roger Källberg]

You can do this in many different ways, like configure the pots dial peer to 
not strip whatever that is matched, "no digit-strip" under the pots dial-peer. 
Or you could add it back again with this command, "prefix 65". Yet another way 
would be to use a voice translation profile. To add to the confusion you can 
also use num-exp to add the prefix back to the number after it has been removed 
by the incoming pots dial-peer, but before it's matched by the outgoing voip 
dial-peer.

Of course you will need to change the match destination pattern in your other 
dial-peer, the voip dp's, after you have used any of the above ways of keeping 
the prefix.

Sincerely

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

Från: manishankar pandey [manishankar...@yahoo.com]
Skickat: den 2 september 2010 04:16
Till: ccie_voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] 2 or in some case 3 stage dialing - issue

Hello All,

Not sure what wrong I am doing but hope expert can point out the right approach 
here.

Lab Setup
Site A
PBX Phone--->PBX-->PRI GW---> CUBE--->WAN

Site B
WAN--->PRI GW acting as CUBE plus PRI GW>PBX --->PBX Phone

Calling habbit
2 Stage Dialing
User at A dial first access code then hears a 2nd dial-tone and then dial the 5 
digit extension

Example : Access code dialed is 65 and then the number dialed is 22505 site B 
PBX Phone. Site B expect to receive the call with 6522505 and at B end it 
strips out 65 using translation pattern.


I did implemented this but it's not the best way

dial-peer voice 3 pots
 description ## Testing for Incoming Call from Site A PBX Phone##
 incoming called-number 65
 port 0/0/1:23
 forward-digits all
!
dial-peer voice 65 voip
 description ## Testing for Call Going to Site B using Site A CUBE##
 destination-pattern 2
 voice-class codec 11
 voice-class h323 1
 session target ipv4:H323 Interface IP Address of Site A CUBE
 dtmf-relay h245-alphanumeric

Site A CUBE Config
dial-peer voice 46 voip
 description TESTING CARRIED OUT For Call Made FROM PBX PHONE
 destination-pattern 2
 voice-class codec 10
 voice-class h323 1
 session target ipv4:H323 Interface IP Address of Site B CUBE


Issue which I will run into:

In case there is a need for calls which needs to go to extension 2 at any 
other site then dial-peer will create problem.. The only differentiator is that 
access code 65 which needs to be there for all call leg.. But How to keep it 
till the other end CUBE I am not getting any way..

The POTs dial-peer takes out the access code..


Any input..

thanks for your kind attention.

Regards
M


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Re: [OSL | CCIE_Voice] Inbound CSS on GK Trunk not working?

[Roger Källberg]

Hi Ryan,
Have you verifyed that you don't have a db replication problem?

Sincerely

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

Från: Ryan Schwab [schwab...@shaw.ca]
Skickat: den 1 september 2010 06:56
Till: 'Ohamien Uhakheme'
Kopia: 'OSL Group'
Ämne: Re: [OSL | CCIE_Voice] Inbound CSS on GK Trunk not working?

Yep, tried that. Went as far as creating a completely new partition and CSS, 
and same thing…..no matter what, if a directory number is assigned a partition, 
it cannot be reached from the GK trunk….

The moment I place the directory number into a NONE partition, with a CSS 
applied or not to the trunk, it works.

I went as far as rebooting my CUCM cluster with no luck…..very odd.

From: Ohamien Uhakheme [mailto:oham...@gmail.com]
Sent: August-31-10 10:49 PM
To: Ryan Schwab
Cc: OSL Group
Subject: Re: [OSL | CCIE_Voice] Inbound CSS on GK Trunk not working?

Odd..  For S&G, try putting 5001 in the None partition, and placing the same 
CSS on the trunk.  Technically that CSS should be able to see the none 
partition and it should work.  If not, then we most likely have something wrong 
with that CSS...

HTH,

Ohamien
On Wed, Sep 1, 2010 at 12:11 AM, Ryan Schwab 
mailto:schwab...@shaw.ca>> wrote:
Guys,

I am trying to route calls from CME to UCM with a Gatekeeper.

If I place the DN(5001) on the UCM phone in the NONE partition, the call from 
the CME (ext 3001) works like it should.

As soon as I place 5001 into a partition and configure an inbound CSS on the 
Gatekeeper trunk, the call from ext 3001 hears the UCM annunciator “Your call 
can not be completed as dialled”.

I am certain the CSS can see the appropriate partition, the trunk has been 
reset, etc…

Is it just getting late here and I’m missing something blatantly obvious?? I 
should also mention that calls in the reverse direction (5001 -> 3001) work 
with no problems.

Anyone have any ideas?


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Re: [OSL | CCIE_Voice] Lab 4A 4.5 SIP dialing to CME from CUCM

[Roger Källberg]

Hi,
Since the CME SIP phones only supports the use of one codec, hardcoded as you 
also have seen, you need to invoke a transcoder to get the described call flow 
to work. The transcoder should be configured at the side that will need the 
xcoder, ie on the CME side.

Sincerely

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

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se

Från: Ki Wi [kiwi.vo...@gmail.com]
Skickat: den 30 augusti 2010 16:30
Till: ccie_voice@onlinestudylist.com
Ämne: Re: [OSL | CCIE_Voice] Lab 4A 4.5 SIP dialing to CME from CUCM

Opps, i seems to found out the problem. I hardcoded the SIP phones to
used g711. =[

On 8/30/10, Ki Wi  wrote:
> I missed out something important , if the HQ phone try to use g729,
> the call will fails with the following msg .
>
>
> Aug 30 14:01:31.770:
> //325/E815E5408298/SIP/Error/sipSPIProcessAckMedia: Destination SDP
> Body unavailable
> Aug 30 14:01:31.770: //-1//SIP/Error/sipSPIGetContentQSIG:
> No Inbound Container Created !!!
> Aug 30 14:01:31.770: //-1//SIP/Error/sipSPIGetContentQ931:
> No Inbound Container Created !!!
> Aug 30 14:01:31.774: //-1//SIP/Msg/ccsipDisplayMsg:
> Sent:
> BYE sip:5...@10.10.210.11:5060;transport=tcp SIP/2.0
> Reason: Q.850;cause=86
> Date: Mon, 30 Aug 2010 14:01:31 GMT
> From: ;tag=231F278-313
> Timestamp: 1283176891
> Content-Length: 0
> User-Agent: Cisco-SIPGateway/IOS-12.x
> To: "HQ Ph2"
> ;tag=a34694f3-dc35-4336-949a-f860f02a52af-66030191
> Call-ID: 87118e80-c7c11a1a-4b-bd20...@10.10.210.11
> Via: SIP/2.0/TCP 10.10.202.1:5060;branch=z9hG4bK368F1
> CSeq: 101 BYE
> Max-Forwards: 70
>
>
> Aug 30 14:01:31.778: //-1//SIP/Msg/ccsipDisplayMsg:
> Received:
> BYE sip:3...@10.10.202.1:5060;transport=tcp SIP/2.0
> Reason: Q.850;cause=47
> Date: Mon, 30 Aug 2010 20:52:42 GMT
> From: "HQ Ph2"
> ;tag=a34694f3-dc35-4336-949a-f860f02a52af-66030191
> P-Asserted-Identity: "HQ Ph2" 
> Content-Length: 0
> User-Agent: Cisco-CUCM7.0
> To: ;tag=231F278-313
> Call-ID: 87118e80-c7c11a1a-4b-bd20...@10.10.210.11
> Via: SIP/2.0/TCP 10.10.210.11:5060;branch=z9hG4bK425ef07ef8
> CSeq: 102 BYE
> Max-Forwards: 70
>
> On 8/30/10, Ki Wi  wrote:
>> I'm facing this little problem here. If i set the SIP trunk to be on a
>> different region from my originating phone. g729 will be used.
>>
>> It seems like during the 'capability exchange' , the IP phone listed
>> those below :
>> v=0
>> o=Cisco-SIPUA 23883 0 IN IP4 10.10.202.51
>> s=SIP Call
>> t=0 0
>> m=audio 31398 RTP/AVP 0 8 18 116 101
>> c=IN IP4 10.10.202.51
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:18 G729/8000
>> a=fmtp:18 annexb=no
>> a=rtpmap:116 iLBC/8000
>> a=fmtp:116 mode=20
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=sendrecv
>>
>> When the SIP gateway in the CME, listed only
>> v=0
>> o=CiscoSystemsSIP-GW-UserAgent 5981 6082 IN IP4 10.10.202.1
>> s=SIP Call
>> c=IN IP4 10.10.202.1
>> t=0 0
>> m=audio 17266 RTP/AVP 0 19
>> c=IN IP4 10.10.202.1
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:19 CN/8000
>> a=ptime:20
>>
>> Based on my finding above, i set the SIP trunk to CME to be on the
>> same region as my originating phone (HQ phone), it works!
>>
>> How can I make the SIP gateway (on BR2) supports g729? Do i need a
>> xcoder?
>>
>

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Re: [OSL | CCIE_Voice] One Route pattern for all offnet.

[Roger Källberg]

DNA is not a plugin anymore, you reach its by adding /dna after the IP adress 
of the CUCM server.

Like this https://10.10.210.10/dna

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

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se

Från: ccieid1ot [ccieid...@gmail.com]
Skickat: den 25 augusti 2010 20:35
Till: Randall Saborio
Kopia: ccie_voice@onlinestudylist.com
Ämne: Re: [OSL | CCIE_Voice] One Route pattern for all offnet.

> What is the partition for the 9...@?
It's the last partition in the CSS.  PT-HQ-Last

> What pattern do you use?
I've used  911 and 9.[2-9]XX translations.

> What is the called party transform mask?
Didn't seem to work, so did not continue further

> What number do you dial?
911 and 95558805

> Have you used DNA ?
Could not find it in the plugins.

On Wed, Aug 25, 2010 at 12:47 PM, Randall Saborio  wrote:
> How would you configure a translation pattern for this?
> Provide an example:
>
> What is the partition for the 9...@?
> What pattern do you use?
> What is the called party transform mask?
> What number do you dial?
> Have you used DNA ?
>
> Your question doesn't have much info.
>
> On Wed, Aug 25, 2010 at 10:42 AM, ccieid1ot  wrote:
>>
>> Hi,
>>
>> Any way for this to be achieved?  Just having 1 route pattern 9.@ and
>> using all translation-patterns to match the dialed string then it
>> should hit the route pattern.  I can get 911 to work, but not anything
>> else.  What am I missing?
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>
>
>
> --
> Randall "da ill" Saborio
> CCIE Voice Wannabe #10054675811
>
>

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Re: [OSL | CCIE_Voice] I have a Question?

[Roger Källberg]

Congratulations Ken,
Well deserved.

Enjoy the moment.

I like my coffee with a splash of milk :-)

Sincerely

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

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>

Från: Beck, Ken [kb...@vectorusa.com]
Skickat: den 19 augusti 2010 02:45
Till: ccie_voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] I have a Question?

What do I do with my time now that I 
PASSED!!!


Ken Beck CCIE Voice #26726


To Vik & Amy,

It was absolutely invaluable that first OWLE class I took and both of you were 
there.  It truly changed my life.

Amy, I have listened to the audio on demand like 10X.  it’s all I listened to 
in the car and taking walks.

Vik, I can’t express how much of an inspiration you are to us all.


Coffee is on me GUYS!


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[OSL | CCIE_Voice] CCIE #26721 - I PASSED!

[Roger Källberg]

Hi my friend!
I'm so happy for you, almost as happy as I was when I passed actually :-). You 
are very well deserved with this success. Congratulations, enjoy the moment.

Two first time passers in the same study group, it must be some sort of record 
;-) It proves that we must have done something right :-) All those long hours 
payed out in the end.

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

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>
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Re: [OSL | CCIE_Voice] CUBE Issue - Vol2 Lab 8 - BR2 Calls to CUCM

[Roger Källberg]

The call leg from HQ to BR2 should be G729, but call to the SIP CME phone will 
always be G711, or whatever codec you hardcode it to be.

So you should see two call legs with the show sccp connection, one from HQ 
phone to the xcoder and one from the xcoder to the BR2 phone.

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

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>

Från: Kevin Damisch [kevin.dami...@vitalsite.com]
Skickat: den 13 juli 2010 16:29
Till: Kevin Damisch; Roger Källberg; Wael Agina; Angel Perez
Kopia: osl osl
Ämne: RE: [OSL | CCIE_Voice] CUBE Issue - Vol2 Lab 8 - BR2 Calls to CUCM

Calls complete now to and from 5XXX and 3XXX once I setup a transcoder also on 
the BR2 router.  But these calls show up as G711.  Shouldn’t that be G729?

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kevin Damisch
Sent: Monday, July 12, 2010 11:36 PM
To: Roger Källberg; Wael Agina; Angel Perez
Cc: osl osl
Subject: Re: [OSL | CCIE_Voice] CUBE Issue - Vol2 Lab 8 - BR2 Calls to CUCM

Am I just blind or can someone tell me where in the PG that you need a 
transcoders on BR2 on this lab?  From what I’ve read on other posts, this is 
required.  So, this takes a transcoder on HQRTR on the trunk, plus we need a 
transcoder on the BR2RTR.  Is this correct?

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roger Källberg
Sent: Tuesday, May 18, 2010 1:48 PM
To: Wael Agina; Angel Perez
Cc: osl osl
Subject: Re: [OSL | CCIE_Voice] CUBE Issue - Vol2 Lab 8 - BR2 Calls to CUCM

Is the xcoder registered with the CME? Check with sh sdsp units

Brgds,
Roger Källberg

Från: Wael Agina [waelag...@gmail.com]
Skickat: den 18 maj 2010 19:01
Till: Angel Perez
Kopia: osl osl
Ämne: Re: [OSL | CCIE_Voice] CUBE Issue - Vol2 Lab 8 - BR2 Calls to CUCM
Done, but still the same :(

BR2(config)#telephony-service
BR2(config-telephony)#max-dn 25
BR2(config-telephony)#max-ephones 33
BR2(config-telephony)#ip source-address  10.10.202.1
Updating CNF files
CNF files updating complete

BR2(config-telephony)#
BR2(config-telephony)#^Z
BR2#
BR2#
BR2#term mon
BR2#sh deb
BR2#sh debugging

CCSIP SPI: SIP error debug tracing is enabled   (filter is OFF)



BR2#
SIP: (69) Attribute mid, level 1 instance 1 not found.
May 18 17:18:18.416: 
//69/2F9E1026ADB5/SIP/Error/sipSPI_ipip_copy_sdp_to_channelInfo:
failed to update call entry
May 18 17:18:19.568: 
//70/2F9E1026ADB5/SIP/Error/sipSPI_ipip_copy_channelInfo_to_sdp:
filter mis-match, failing call
May 18 17:18:19.568: //-1//SIP/Error/sipSPIGetContentQSIG: No 
Inbound Container Created !!!
May 18 17:18:19.568: //-1//SIP/Error/sipSPIGetContentQ931: No 
Inbound Container Created !!!
May 18 17:18:19.572: //70/2F9E1026ADB5/SIP/Error/sipSPIAddSDPMediaPayload: Call 
Origination Failed: None of the selected codec from CLI is supported by SIP
May 18 17:18:19.572: //70/2F9E1026ADB5/SIP/Error/sipSPIOutgoingCallSDP: Error 
with codec types on media line : 1
May 18 17:18:19.572: //70/2F9E1026ADB5/SIP/Error/sipSPICreateOutboundSDP: Error 
in creating an SDP for the outbound call - Check for supported codecs
May 18 17:18:19.572: //70/2F9E1026ADB5/SIP/Error/preprocessSetup: Error during 
outbound SDP creation
May 18 17:18:19.572: //-1//SIP/Error/sipSPIGetContentQSIG: No 
Inbound Container Created !!!
May 18 17:18:19.572: //-1//SIP/Error/sipSPIGetContentQ931: No 
Inbound Container Created !!!
May 18 17:18:19.572: //-1//SIP/Error/ccsip_spi_process_ccapi_event: 
CCAPI Event Preprocessor Failure
BR2#


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This communication (including any attachments) is intended only for the use of 
the individual or entity to which it is addressed, and may contain information 
that is privileged, confidential and exempt from disclosure under applicable 
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For more information reg

Re: [OSL | CCIE_Voice] Frame-relay fragment question

[Roger Källberg]

Yes, that is correct.

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

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>

Från: Berry, Matthew J. [mjbe...@krollontrack.com]
Skickat: den 29 juni 2010 16:39
Till: Roger Källberg; Bo Gao; OSL
Ämne: RE: [OSL | CCIE_Voice] Frame-relay fragment question

I guess that makes sense.  You’re not actually making the link slower, so the 
fragment size wouldn’t change.

We’d only need to change the minCIR, CIR, and bc?

Matthew Berry, CCVP, Sr. Unified Communications Engineer
mjbe...@kroll.com<mailto:david.ra...@kroll.com>

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roger Källberg
Sent: Tuesday, June 29, 2010 9:33 AM
To: Bo Gao; OSL
Subject: Re: [OSL | CCIE_Voice] Frame-relay fragment question

You shouldn't change the fragment size. Reason being that you want the fragment 
to be of a size that would give you a 10ms transmit delay in the event of 
congestion.

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

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>

Från: Bo Gao [bga...@gmail.com]
Skickat: den 29 juni 2010 16:17
Till: OSL
Ämne: [OSL | CCIE_Voice] Frame-relay fragment question
HQ-BR1 bandwidth is 384K, I have the following config:

map-class frame-relay AutoQoS-FR-Se0/0-201
 frame-relay cir 384000
 frame-relay bc 3840
 frame-relay be 0
 frame-relay mincir 384000
 frame-relay fragment 480
 service-policy output AutoQoS-Policy-Trust


If I were to change the cir to 95% based on the QoS SNRD
Then I would have:

map-class frame-relay AutoQoS-FR-Se0/0-201
 frame-relay cir 364800
 frame-relay bc 3648
 frame-relay be 0
 frame-relay mincir 34800
 frame-relay fragment 480
 service-policy output AutoQoS-Policy-Trust


Question:  Should I also change the frame-realy fragment from 480 to 456?
Why?



Thank you!


Bo


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Re: [OSL | CCIE_Voice] Frame-relay fragment question

[Roger Källberg]

You shouldn't change the fragment size. Reason being that you want the fragment 
to be of a size that would give you a 10ms transmit delay in the event of 
congestion.

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

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>

Från: Bo Gao [bga...@gmail.com]
Skickat: den 29 juni 2010 16:17
Till: OSL
Ämne: [OSL | CCIE_Voice] Frame-relay fragment question

HQ-BR1 bandwidth is 384K, I have the following config:

map-class frame-relay AutoQoS-FR-Se0/0-201
 frame-relay cir 384000
 frame-relay bc 3840
 frame-relay be 0
 frame-relay mincir 384000
 frame-relay fragment 480
 service-policy output AutoQoS-Policy-Trust


If I were to change the cir to 95% based on the QoS SNRD
Then I would have:

map-class frame-relay AutoQoS-FR-Se0/0-201
 frame-relay cir 364800
 frame-relay bc 3648
 frame-relay be 0
 frame-relay mincir 34800
 frame-relay fragment 480
 service-policy output AutoQoS-Policy-Trust


Question:  Should I also change the frame-realy fragment from 480 to 456?
Why?



Thank you!


Bo


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Re: [OSL | CCIE_Voice] Voice Hunt Group to Voicemail Vol2 Lab 9

[Roger Källberg]

Hi Graham,
The big differnce that I can see is that the  first config example will 
actually never even hit voice register dn 3, it will go to VM directly from the 
hunt group.

But the second config will first use the hunt group, then the voice register dn 
and from there it will go to VM.

At some stage in the second call flow the original DNIS=1000, is lost, thats 
why the VM see the call as a direct call from ANI=5002. If DNIS=1000 would have 
been kept it should have been in the form of RDNIS, as per what the first debug 
shows as diversion.

Regards

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

Från: Graham Hopkins [ghopk...@wolf-rock.co.uk]
Skickat: den 25 juni 2010 18:39
Till: CCIE Voice Maillist
Ämne: [OSL | CCIE_Voice] Voice Hunt Group to Voicemail Vol2 Lab 9

Some odd behaviour here that I can't quite get my head around.

Config 1 - Working

voice register dn  3
 number 1003
 call-forward b2bua all 1600
 mwi

 voice hunt-group 1 parallel
 final 1600
 list 1001,1002
 timeout 12
 pilot 1000

Config 2 - As per PG

voice register dn  3
 number 1003
 call-forward b2bua all 1600
 mwi

 voice hunt-group 1 parallel
 final 1003
 list 1001,1002
 timeout 12
 pilot 1000


 Call from 5002 to 1000 and let go to voicemail


with config 1 UC sees the call as redirected and plays the welcome "1000 not 
available"

with config 2  UC see the call as coming directly from 5002 to 1600 and prompts 
for the pin for mailbox 5002

 difference seems to be in the SIP messaging

 for config 1 I see

 Jun 25 13:14:44.778: //-1//SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE 
sip:1...@10.10.210.13:5060<,DanaInfo=.aey4BwynGlxwlu0N645.,SSL+sip:1...@10.10.210.13:5060>
 SIP/2.0
Via: SIP/2.0/UDP 10.10.201.1:5060;branch=z9hG4bK10AD7
Remote-Party-ID: "HQ Ph2" 
>;party=calling;screen=no;privacy=off
From: "HQ Ph2" 
>;tag=82568C-1CF
To: 
>
Date: Fri, 25 Jun 2010 13:14:44 GMT
Call-ID: 74e861cb-7f9211df-836ffb0f-d478...@10.10.201.1
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 1921735851-2140279263-2154205609-2551626315
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, 
NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1277471684
Contact: 
>
Diversion: 
>;privacy=off;reason=no-answer;counter=1;screen=no
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 188

but no such diversion for config 2 just

Jun 25 13:15:34.588: //-1//SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE 
sip:1...@10.10.210.13:5060<,DanaInfo=.aey4BwynGlxwlu0N645.,SSL+sip:1...@10.10.210.13:5060>
 SIP/2.0
Via: SIP/2.0/UDP 10.10.201.1:5060;branch=z9hG4bK110200E
Remote-Party-ID: "HQ Ph2" 
>;party=calling;screen=no;privacy=off
From: "HQ Ph2" 
>;tag=831920-E7
To: 
>
Date: Fri, 25 Jun 2010 13:15:34 GMT
Call-ID: 929827af-7f9211df-8380fb0f-d478...@10.10.201.1
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 2459285263-2140279263-2206071567-222790417
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, 
NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1277471734
Contact: 
>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 187

Any thoughts?

Graham Hopkins



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Re: [OSL | CCIE_Voice] HomeLab Equipments

[Roger Källberg]

Hi Ken,
As long as you can use the proper version of IOS you should be good to go with 
your 3640. Not sure about the DSP module, might be that you need to configure 
that somewhat differnt for conf, mtp, xcode and so on than the one that are 
used in the acctual lab. But as I said I'm not sure.

Brgds,
Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB

Från: Ken Tan [thinkc...@gmail.com]
Skickat: den 26 juni 2010 12:34
Till: ccie_voice
Ämne: [OSL | CCIE_Voice] HomeLab Equipments

Hi,

Can anyone advise if I can build a CCIE Voice homelab based on 3640
instead of 2811.

I checked cisco web site it seems
3640 together with NM-HV PVDM-12 and
VWIC-2MFT-E1/T1 seems workable.

Had too many 3640 lying around do not wish to invest unnecessary.

Any advise is greatly appreciated.

Ken

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Re: [OSL | CCIE_Voice] Presence Issues> Softphone mode in CUPC Lab 5 Volume 2

[Roger Källberg]

Hi Daniel,
It's not always that you can trust the information given by the show perf query 
class "Number of Replicates Created and State of Replication" command.

One easy thing that you can do to verify if you have a db repl problem is to 
put your phones, or any other device, in a pub only enviroment. If all works 
then you know that the sub didn't have the correct info.

And in thet case you need to repair the db replication by utils debreplication 
stop ,1 on sub, then when promtpt returns on the sub put in the same command on 
pub). When the prompt returns on the pub use utils dbreplication repair all on 
the pub. This will take some time to complete.

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

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>

Från: Daniel Berlinski [dberlin...@gmail.com]
Skickat: den 26 juni 2010 23:44
Till: Roger Källberg
Kopia: kobel; osl osl
Ämne: Re: [OSL | CCIE_Voice] Presence Issues> Softphone mode in CUPC Lab 5 
Volume 2

Hi Kobel
Owner was setup for the mobility section to work.  It is in there.

Hi Roger
The way I know how to verify dbReplication is:
admin:show perf query class "Number of Replicates Created and State of 
Replication"
==>query class :

 - Perf class (Number of Replicates Created and State of Replication) has 
instances and values:
ReplicateCount  -> Number of Replicates Created   = 412
ReplicateCount  -> Replicate_State= 2

My reading of this is that is all good.  Am I right?

Well, I have rebooted this many times already so I think I will just upgrade 
the client and see what happens.  Will update you all. Thnaks






2010/6/27 Roger Källberg 
mailto:roger.kallb...@cygate.se>>
Try to verify if db replication is ok, if not, fix that. You might also want to 
restart the CTI Manager on both sub and pub.

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

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>

Från: Daniel Berlinski [dberlin...@gmail.com<mailto:dberlin...@gmail.com>]
Skickat: den 26 juni 2010 23:18
Till: kobel
Kopia: osl osl
Ämne: Re: [OSL | CCIE_Voice] Presence Issues> Softphone mode in CUPC Lab 5 
Volume 2

Thanks for your replies.

Primary extension is assigned to end user and that extension matches with the 
line number of CUPC.
The users are assigned to the Standard CCM End Users, and CTI Enabled groups

What is the version of CUPC you guys use?

Thank you

On Sun, Jun 27, 2010 at 9:03 AM, kobel 
mailto:findko...@gmail.com>> wrote:
See if adding the end user to "Standard CUCM users" group in CUCM helps

regards

On Sat, Jun 26, 2010 at 10:30 PM, Daniel Berlinski 
mailto:dberlin...@gmail.com>> wrote:
Hello all
Out of ideas now after troubleshooting extensively a Presence problem.  I'm 
finalizing lab 5 Vol 2 and I can't get CUPC to download its TFTP configuration 
file from CUCM and for that reason I do not even see the option for selecting 
"softphone control"  Any help is appreciated.  What I have and what I've done 
is the following:

1- Cretaed device named UPC+12alphanumeric characters, in my case 
UPCTERRELLEPRYO, associated its line to the enduser
2- End user configured with primary extension, associated with UPC phone 
device, CTI control of its devices and group association to CTI enabled group.
3- Still in CUCM, Capabilities Assignment was provided for the user.
5- In CUPS, Application-Cisco Unified Personal Comm - Settings I have provided 
the IP addresses for TFTP server primary and secondary

Presence status is working fine and Deskphone control works fine as well.  My 
problem here is that the CUPC SIP phone is not getting in Show Server Health a 
tftp file to download. It displays the IP addres of TFTP primary and seoondary 
but it does not display the UPCTERRELLEPRYO.CNF.XML file to download.

To troubleshoot this I have done the following:
1- Went in DOS and did a tftp -i 10.10.210.10 get UPCTERRELLEPRYO.CNF.XML and 
files downloaded OK so there is no network issues here.  Inside the file I saw 
references to TFTP server as IP addresses so no
name resolution issues either.
2- Ran Wireshark and did not see any attempts from the client machine to 
register with CUCM via SIP so client is not even attempting to register. In 
fact nothing displays when I filter the capture by the CUCM ip addresses.
3- Listing my cupc users by clicking in CUPS, application, Cisco Unified 
personal comm, user settings I see my users listed there but under the column 
"Client Type" nothing displays
4- Created another UPC device for another user with another name and it still 
presents same problem.
5- Tried to enable all phone tracing in CUC

Re: [OSL | CCIE_Voice] Vol1 lab 10- NBAR question

[Roger Källberg]

Not really sure if I fully understand your question, but you don't need to 
configure "ip nbar proctocol-discovery on the ports" for NBAR to work for 
classification of traffic in a class-map.

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

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>

Från: akash patel [akashapa...@yahoo.com]
Skickat: den 26 juni 2010 15:49
Till: ccie_voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] Vol1 lab 10- NBAR question

Lab 10 vol1 ask that not to trust packets from devices, at the same time not to 
configure access-list.

I understood this can be achieved using NBAR.  This works fine for HQ, but on 
BR1/BR2 router you can configure
ip nbar proctocol-discovery on the ports connected to phones.  so if we 
configure class-map matching protocol, how will that work?

Thank you

___
For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] Presence Issues> Softphone mode in CUPC Lab 5 Volume 2

[Roger Källberg]

Try to verify if db replication is ok, if not, fix that. You might also want to 
restart the CTI Manager on both sub and pub.

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

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>

Från: Daniel Berlinski [dberlin...@gmail.com]
Skickat: den 26 juni 2010 23:18
Till: kobel
Kopia: osl osl
Ämne: Re: [OSL | CCIE_Voice] Presence Issues> Softphone mode in CUPC Lab 5 
Volume 2

Thanks for your replies.

Primary extension is assigned to end user and that extension matches with the 
line number of CUPC.
The users are assigned to the Standard CCM End Users, and CTI Enabled groups

What is the version of CUPC you guys use?

Thank you

On Sun, Jun 27, 2010 at 9:03 AM, kobel 
mailto:findko...@gmail.com>> wrote:
See if adding the end user to "Standard CUCM users" group in CUCM helps

regards

On Sat, Jun 26, 2010 at 10:30 PM, Daniel Berlinski 
mailto:dberlin...@gmail.com>> wrote:
Hello all
Out of ideas now after troubleshooting extensively a Presence problem.  I'm 
finalizing lab 5 Vol 2 and I can't get CUPC to download its TFTP configuration 
file from CUCM and for that reason I do not even see the option for selecting 
"softphone control"  Any help is appreciated.  What I have and what I've done 
is the following:

1- Cretaed device named UPC+12alphanumeric characters, in my case 
UPCTERRELLEPRYO, associated its line to the enduser
2- End user configured with primary extension, associated with UPC phone 
device, CTI control of its devices and group association to CTI enabled group.
3- Still in CUCM, Capabilities Assignment was provided for the user.
5- In CUPS, Application-Cisco Unified Personal Comm - Settings I have provided 
the IP addresses for TFTP server primary and secondary

Presence status is working fine and Deskphone control works fine as well.  My 
problem here is that the CUPC SIP phone is not getting in Show Server Health a 
tftp file to download. It displays the IP addres of TFTP primary and seoondary 
but it does not display the UPCTERRELLEPRYO.CNF.XML file to download.

To troubleshoot this I have done the following:
1- Went in DOS and did a tftp -i 10.10.210.10 get UPCTERRELLEPRYO.CNF.XML and 
files downloaded OK so there is no network issues here.  Inside the file I saw 
references to TFTP server as IP addresses so no
name resolution issues either.
2- Ran Wireshark and did not see any attempts from the client machine to 
register with CUCM via SIP so client is not even attempting to register. In 
fact nothing displays when I filter the capture by the CUCM ip addresses.
3- Listing my cupc users by clicking in CUPS, application, Cisco Unified 
personal comm, user settings I see my users listed there but under the column 
"Client Type" nothing displays
4- Created another UPC device for another user with another name and it still 
presents same problem.
5- Tried to enable all phone tracing in CUCM and everything else related to SIP 
under trace settings and nothing displayed with relation to the UPC phone 
attempting to register.

Not sure what to do next, my version of CUPC is 7.0(2.13496) I havem't looked 
for bugs yet.  What version are you guys using? If anyone has any ideas please 
let me know


___
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Re: [OSL | CCIE_Voice] Local Dial Plans (7 digit, 8 digit)

[Roger Källberg]

Hi Mark,
Looks like you over complicate stuff, why do you have your phone DN's in 
different partitions? Just put all of your DN's in pt-internal, put that PT in 
all your routing CSS'es (ie your CSS used on the device level, or CSS used 
after passing trough a TP, or incoming CSS for calls in to a GW/GK and so on).

Then in the different device CSS’es, these are site specific, put your PT’s for 
external call routing. In your case it would be something like this, pt-hq in 
css-dev-hq, pt-br1 in css-dev-br1 and pt-br2 in css-dev-br2. In this way the 
phones at HQ and BR1 will not be able to see the PT used to route calls out 
from the BR2 site, hence it will not experience the post dial delay.


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

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se

Från: Mark [...@markholloway.com]
Skickat: den 22 juni 2010 07:38
Till: Mark
Kopia: OSL osl
Ämne: Re: [OSL | CCIE_Voice] Local Dial Plans (7 digit, 8 digit)

I know why the post dial delay is happening. I was looking for the right 
approach on how build this out when two local dial plans can cause this kind of 
behavior.  The only solution I can come up with is having each Phone (at the 
Device level) be part of a CSS that only contains his site's partition.  This 
prohibits the phones at one site from 4 digit dialing another, but allows 
either Translation Patterns or Route Patterns to be built with similar dial 
plans yet doesn't cause a conflict.  However, to allow the phones to 4 digit 
dial between sites I created another CSS called css-internal which contains 
pt-hq, pt-br1, pt-br2, and I assign the CSS within the Line [1] > Directory 
Settings CSS and all is well.  I'm not sure if this the RIGHT way to do it if 
faced with this kind of task, but it's the best I could come up with.  I'm all 
ears..



On Jun 21, 2010, at 8:43 PM, Mark wrote:

> I'm trying to work through a scenario where all 3 locations (HQ, BR1, BR2) 
> are built in UCM.  HQ and BR1 dial 7 digits for local and BR2 dials 8 digits 
> for local.  When building my number plan I am creating it like this..
>
> pt-hq and pt-br1
> 9.[2-9]XX
>
> pt-br2
> 9.[2-9]XXX
>
> The problem is once pt-br2 is added, all local calls dialed by pt-hq and 
> pt-br1 will experience post dial delay.  I probably just need a big giant 
> slap in the face to see that I am missing the obvious.  So, what is it that 
> I'm overlooking? (shrug)
> ___
> For more information regarding industry leading CCIE Lab training, please 
> visit www.ipexpert.com

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Re: [OSL | CCIE_Voice] CCIE Voice # 26249

[Roger Källberg]

Congratulation.

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

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>

Från: Shadow of Voice [salman.shaik...@gmail.com]
Skickat: den 19 juni 2010 00:06
Till: ccie_voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] CCIE Voice # 26249

I took my lab yesterday, although it was my 2nd attempt but i just got the 
score report and I passed :-) I am now officially CCIE Voice # 26249.

Just like Asher, i got my CCIE # after him and I learned a lot from my first 
attempt and with this study group, who helped me to make better attempt 2nd 
time.

once again i would advise to read Asher comments and strategy as i attempt in 
the same way as he did.

but one thing is exceptional, my lucky number is 9 and i got 9 .. :)
just like Ash got his lucky number 4 and i got mine  :)

Thanks for your help ...

Salman Shaikh
CCIE#26249 (Voice)
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Re: [OSL | CCIE_Voice] CCIE Voice #26244

[Roger Källberg]

Congratulations Ash :-)

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

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>

Från: Ashar Siddiqui [siddas...@gmail.com]
Skickat: den 18 juni 2010 20:46
Till: ccie_voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] CCIE Voice #26244

Hello all,

I went to Brussels yesterday and just an hour before learned that I am now 
officially CCIE Voice. It was my 2nd attempt but it was worth it.
I learned a lot from my first attempt and it helped me build a better strategy 
for the 2nd.

I am thankful to this wonderful list and IPExpert material which I used. 
Special thanks to Amy Ryan for her help whenever I needed.
I am also grateful to my Study Partner Iwan Hoogendoorn, a triple CCIE and I 
was so lucky to have him as Study partner. I will never forget the way he use 
to make daily schedules and strictly made me follow those otherwise I am a lazy 
man..this number is for you Iwan!

Few take home points for all those who will be making an attempt in coming days:

 1 - Read the lab CAREFULLY (I made it Caps for a reason)..every word in a 
question is there for a reason!
 2 - Do not rush! the mistakes you will make in first one hour will haunt you 
in the entire lab (unless you are lucky to figure out what went wrong)
 3 - Do not spend too much time if something is not working - you can always 
come back to it.
 4 - Note down sections and task which you are working and cross them as soon 
as you have completed it
 5 - Call routing - This is how I did it, not necessarily helpful for you, I 
did call routing on a page first as what I am going to do at RL level, Pattern 
level etc..I configured everything first and then tested it one by one..took me 
30 minutes to finish call routing
 6 - Test everything you have done at least twice and as if it was configured 
by someone else and you are the proctor..I found one mistake while doing my 2nd 
check
 7 - Save your config often, make sure before you leave that all gateways are 
up and registered to CUCM.

I joined this list for my CCIE studies when I started my CCIE journey back in 
December 2009 but now I have decided to stick with it as I won't find such a 
nice bunch of people anywhere..

N.B: Above all, I loved my number..Digit '4' is my lucky number and Cisco made 
sure that I have enough of them..  :)

Thank you all. It's party time now ;)

Ashar Siddiqui
CCIE#26244 (Voice)
___
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Re: [OSL | CCIE_Voice] Called Party Transformation

[Roger Källberg]

I don’t think is has to do with the IE part, this is what Cisco write about 
this command. It only mention the called-ID, not Called IE.


supplementary-service h225-notify cid-update (voice-service)
To globally enable the sending of H.225 messages with caller-ID updates, use 
the supplementary-service h225-notify cid-update command in voice-service 
configuration mode. To disable the sending of H.225 messages with caller-ID 
updates, use the no form of this command.

supplementary-service h225-notify cid-update
no supplementary-service h225-notify cid-update

Syntax Description
This command has no arguments or keywords.

Command Default
H.225 messages with caller-ID updates are enabled.

Command Modes
Voice-service configuration

Command History
Release

Modification

12.3(7)T

This command was introduced.



Usage Guidelines
This command globally provides caller ID updates through H.225 notify messages 
when a call is transferred or forwarded between Cisco CallManager Express and 
Cisco CallManager systems. The default is that this behavior is enabled. The no 
form of the command disables caller-ID updates, which is not recommended. Use 
the supplementary-service h225-notify cid-update command in dial peer 
configuration mode to specify this capability for individual dial peers.
If this command is enabled globally and enabled on a dial peer, the 
functionality is enabled for that dial peer. This is the default.
If this command is enabled globally and disabled on a dial peer, the 
functionality is disabled for that dial peer.
If this command is disabled globally and either enabled or disabled on a dial 
peer, the functionality is disabled for that dial peer.


Examples
The following example globally enables the sending of H.225 messages to 
transmit caller-ID updates and then disables that capability on dial peer 24.
Router(config)# voice service voip
Router(config-voi-serv)# supplementary-service h225-notify cid-update
Router(config-voi-serv)# exit
Router(config)# dial-peer voice 24 voip
Router(config-dial-peer)# no supplementary-service h225-notify cid-update
Router(config-dial-peer)# exit


Roger Källberg
CCIE # 26199 (Voice)
Unified Communication Consultant
Cygate AB


From: Moataz Mamdouh [mailto:moataz_m...@yahoo.com]
Sent: den 18 juni 2010 13:35
To: Roger Källberg
Cc: ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] Called Party Transformation



--- On Fri, 6/18/10, Moataz Mamdouh  wrote:

From: Moataz Mamdouh 
Subject: RE: [OSL | CCIE_Voice] Called Party Transformation
To: "Roger Källberg" 
Date: Friday, June 18, 2010, 7:33 AM
Does this will disable the default behavior in ehich  H.225 notify message sent 
from the CM to the Gateway with missing Called # IE in the message body of the 
NOTIFY.


Moataz Mamdouh
CCIE # 26129 (Voice)
Voice Support Engineer
www.seegypt.com

--- On Fri, 6/18/10, Roger Källberg  wrote:

From: Roger Källberg 
Subject: RE: [OSL | CCIE_Voice] Called Party Transformation
To: "Moataz Mamdouh" , "Amy Ryan" , 
"cisco voip" 
Cc: "osl osl" 
Date: Friday, June 18, 2010, 7:19 AM
You can set the H.323 gateway to mimic the functionality of a MGCP GW when it 
comes to display of the called number on the phone display.
Add this to your config
voice service voip
 no supplementary-service h225-notify cid-update
Regards
Roger Källberg
CCIE # 26199 (Voice)
Unified Communication Consultant
Cygate AB

From: Moataz Mamdouh [mailto:moataz_m...@yahoo.com]
Sent: den 18 juni 2010 12:43
To: Amy Ryan; cisco voip
Cc: osl osl
Subject: Re: [OSL | CCIE_Voice] Called Party Transformation

Hi cisco voip
i think the best practice to manipulate the called # for H.323 Gatewys as it's 
Peer to Peer
is to do it on the outgoing dial-peer vy applying voice transaltions

i do not think CM can do something in this case , because the gateway can 
handle it .
--- On Fri, 6/18/10, cisco voip  wrote:

From: 
Subject: Re: [OSL | CCIE_Voice] Called Party Transformation
To: "Amy Ryan" 
Cc: "osl osl" 
Date: Friday, June 18, 2010, 6:32 AM
Hi Amy,

Thanks a lot for quick reply.
I tried this scenario in the lab, and it is working as you mentioned for MGCP 
gateways.

But for H323 gateway, the display on the phone is the destination-pattern of 
outgoing dial-peer, is there a way to manipulate that too.
On Thu, Jun 17, 2010 at 8:36 PM, Amy Ryan  wrote:

When utilizing the called party transformation pattern CSS to manipulate digits 
when conducting outbound calls you are modifying the dialed digits (DNIS) that 
are presented to the destination (in this case the PSTN) and those same digits 
will be reflected on your display.  If you have decided to do called party # 
manipulation at the gateway via a Called Party Transformation Pattern CSS (that 
see a matching pattern) it will always override anything done at the RP/RL 
display or otherwise.  If you run a “debug isdn q931” on the gateway you will 
see this.

If you are trying to force 

Re: [OSL | CCIE_Voice] Called Party Transformation

[Roger Källberg]

You can set the H.323 gateway to mimic the functionality of a MGCP GW when it 
comes to display of the called number on the phone display.
Add this to your config
voice service voip
 no supplementary-service h225-notify cid-update
Regards
Roger Källberg
CCIE # 26199 (Voice)
Unified Communication Consultant
Cygate AB


From: Moataz Mamdouh [mailto:moataz_m...@yahoo.com]
Sent: den 18 juni 2010 12:43
To: Amy Ryan; cisco voip
Cc: osl osl
Subject: Re: [OSL | CCIE_Voice] Called Party Transformation

Hi cisco voip
i think the best practice to manipulate the called # for H.323 Gatewys as it's 
Peer to Peer
is to do it on the outgoing dial-peer vy applying voice transaltions

i do not think CM can do something in this case , because the gateway can 
handle it .
--- On Fri, 6/18/10, cisco voip  wrote:

From: 
Subject: Re: [OSL | CCIE_Voice] Called Party Transformation
To: "Amy Ryan" 
Cc: "osl osl" 
Date: Friday, June 18, 2010, 6:32 AM
Hi Amy,

Thanks a lot for quick reply.
I tried this scenario in the lab, and it is working as you mentioned for MGCP 
gateways.

But for H323 gateway, the display on the phone is the destination-pattern of 
outgoing dial-peer, is there a way to manipulate that too.

On Thu, Jun 17, 2010 at 8:36 PM, Amy Ryan 
> wrote:

When utilizing the called party transformation pattern CSS to manipulate digits 
when conducting outbound calls you are modifying the dialed digits (DNIS) that 
are presented to the destination (in this case the PSTN) and those same digits 
will be reflected on your display.  If you have decided to do called party # 
manipulation at the gateway via a Called Party Transformation Pattern CSS (that 
see a matching pattern) it will always override anything done at the RP/RL 
display or otherwise.  If you run a “debug isdn q931” on the gateway you will 
see this.

If you are trying to force the phone display of the phone dialing 41031000 to 
show 08041031000 on the display only, you could achieve this by doing Called 
Party # Transformations at both Route Pattern (RP) and Route List (RL).

If you perform Called # Transformations at both RP and RL, the called # 
transformations done at the RP level will then affect display on calling phone 
and the called # transformations done at the RL level will affect Called # sent 
to gateway so that the PSTN accepts the format and processes the call.

So in this case you may have both the following:

Route Pattern
Pattern = 9.41031000
Called # Transformation = DDI Predot, Prefix 080

Route List (within route group)
Called # Transformation = DDI Predot

This will invoke 08041031000 to show on the phone display and 41031000 to be 
sent to the PSTN.


In summary, digit manipulation can be done all in RP or all in RL.  The RL will 
always override the RP when determining what is sent as dialed digits to 
destination.  However when doing called party # transformations at this level, 
and utilizing both RL and RP, manipulations done at RP will always affect 
display of calling phone and the manipulation done at the RL will allow for 
appropriate digits to be send to destination.

HTH,
Amy



---
Amy Ryan – CCIE #24677 (Voice)
Technical Instructor - IPexpert, Inc.
Mailto: ar...@ipexpert.com<http://ar...@ipexpert.com>
Telephone: +1.810.326.1444
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From: cisco voip http://voip.ccieci...@gmail.com>>
Date: Thu, 17 Jun 2010 19:12:29 +0530
To: osl osl 
http://ccie_voice@onlinestudylist.com>>
Subject: [OSL | CCIE_Voice] Called Party Transformation


Hi List,

I am trying to understand the use of
Call Routing > Transformation Pattern > Called/Calling

From what i understand Calling party transformation pattern can be used for 
localization by displaying 7 digits while ringing.

Based on this i thought i can use called party transformation mask while 
calling outside.
I used called party transformation CSS on outbound gateway and add a called 
party Xform pattern as
4103 -> Predot prefix digits- 080

i thought, if i dial 41031000 number that will go to gateway is 41031000 but on 
calling phone, number will be displayed as 08041031000,

But CUCM is behaving we

Re: [OSL | CCIE_Voice] Lab 8 - Question 5.5 // RTP Priority Queue

[Roger Källberg]

Yet again 100% correct :)
Roger Källberg
CCIE # 26199 (Voice)
Unified Communication Consultant
Cygate AB


From: Matthew Berry [mailto:ciscovoiceg...@gmail.com]
Sent: den 16 juni 2010 13:41
To: OSL Group; Matthew Berry
Subject: [OSL | CCIE_Voice] Lab 8 - Question 5.5 // RTP Priority Queue

In question 5.5 we are asked to create a priority queue of 128 kbps for RTP 
traffic between HQ and BR2.

The Proctor Guide told me to set:

interface Virtual-Template 200
  no service-policy output AutoQoS-Policy-UnTrust
  ip rtp priority 16384 16383 128

I would have normally set this with a class-map and policy-map.

Are we setting this priority queue under the Virtual-Template because of the 
"no-MQC" restriction in the question?  Now that I am writing this, I am pretty 
sure this is the reason, but I want to verify.
--


Matthew Berry

A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written



Vitals:

GVoice: +1.612.424.5044

Gmail: ciscovoiceg...@gmail.com<mailto:ciscovoiceg...@gmail.com>

Skype: ciscovoiceguru

Twitter: ciscovoiceguru



Cert Stats:

Cisco Cert Journey Began: Jan 1, 2009

1st Lab Attempt: Aug 16, 2010
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Re: [OSL | CCIE_Voice] Lab 8 - iLBC Locactions Setting

[Roger Källberg]

Hi Matthew,
If you set the codec to other regions to g728 and link loss type to lossy it 
will use iLBC, that is if the phone supports that, just as Vik wrote in his 
reply to you. And of course you need to have the proper SP set so that iLBC 
isn't deactivated.

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

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>

Från: Matthew Berry [ciscovoiceg...@gmail.com]
Skickat: den 16 juni 2010 06:41
Till: OSL Group; Vik Malhi; Amy Ryan; Matthew Berry
Ämne: [OSL | CCIE_Voice] Lab 8 - iLBC Locactions Setting

In Lab 8, you are asked to configure iLBC between HQ and BR1 with RSVP CAC on 
top of that.

The Proctor Guide tells me to set the Link Loss Type under CUCM > SYSTEM > 
LOCATIONS to "Lossy."

However, all of my testing to date seems to demonstrate that the lossy setting 
does not affect whether iLBC is used between endpoints.  I am wondering what 
the reason is for setting this option and whether it is necessary to complete 
the requirements of the question.

--

Matthew Berry

A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written



Vitals:

GVoice: +1.612.424.5044

Gmail: ciscovoiceg...@gmail.com<mailto:ciscovoiceg...@gmail.com>

Skype: ciscovoiceguru

Twitter: ciscovoiceguru



Cert Stats:

Cisco Cert Journey Began: Jan 1, 2009

1st Lab Attempt: Aug 16, 2010
___
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Re: [OSL | CCIE_Voice] OSPF Error CUE Module

[Roger Källberg]

Search the OSL. Has been posted before. This is an informal message, nothing to 
pay any attention to.

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

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>

Från: Matthew Berry [ciscovoiceg...@gmail.com]
Skickat: den 15 juni 2010 13:48
Till: OSL Group; Matthew Berry
Ämne: [OSL | CCIE_Voice] OSPF Error CUE Module

I am getting an odd OSPF error after having configured my service-engine for 
the CUE module:

Jun 14 05:46:22.401: %OSPF-4-NO_IPADDRESS_ON_INT: No IP address for interface 
Service-Engine0/0

Everything appeared to function properly even with this error being reported.  
Below is my example config that I use to configure the CUE module's IP and 
connectivity:

interface FastEthernet 0/0.101
 ip address X.X.X.X 255.255.255.0

interface Service-Engine 0/0
 ip unnumered FastEthernet 0/0.101
 service-module ip address X.X.X.X 255.255.255.0
 service-module ip default-gateway Y.Y.Y.Y
 no shut

ip route X.X.X.X 255.255.255.255 Service-Engine 0/0

--

Matthew Berry

A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written



Vitals:

GVoice: +1.612.424.5044

Gmail: ciscovoiceg...@gmail.com<mailto:ciscovoiceg...@gmail.com>

Skype: ciscovoiceguru

Twitter: ciscovoiceguru



Cert Stats:

Cisco Cert Journey Began: Jan 1, 2009

1st Lab Attempt: Aug 16, 2010
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Re: [OSL | CCIE_Voice] Configuring H.323 Call Preserve

[Roger Källberg]

Call preservation is enabled as soon as you enter call-preserve, the warning is 
for a best practis/recommendation for additional configuration. Look at this, 
http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/cisco_ios_h323_configuration_guide/old_archives_h323/4gwconf.html#wp1150861

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

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>

Från: Matthew Berry [ciscovoiceg...@gmail.com]
Skickat: den 15 juni 2010 13:45
Till: OSL Group; Matthew Berry
Ämne: [OSL | CCIE_Voice] Configuring H.323 Call Preserve

When configuring call preservation for an H.323 gateway, I am using the 
following command:

voice service voip
  h323
call-preserve

As soon as I hit ENTER, the IOS spits back this warning/notice to me:

Warning: Configuring media inactivity detection to avoid hung calls is highly 
recommended.

Does anyone know what I need to do in order to configure media inactivity 
detection?  I want to make sure that I am entering the proper commands to 
ensure that H.323 call preservation is enabled.

--

Matthew Berry

A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written



Vitals:

GVoice: +1.612.424.5044

Gmail: ciscovoiceg...@gmail.com<mailto:ciscovoiceg...@gmail.com>

Skype: ciscovoiceguru

Twitter: ciscovoiceguru



Cert Stats:

Cisco Cert Journey Began: Jan 1, 2009

1st Lab Attempt: Aug 16, 2010
___
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Re: [OSL | CCIE_Voice] QOS FRF.12 & MLPP

[Roger Källberg]

Also if you would have class based shaping, stay away from auto qos. Auto will 
configure FRTS, ie not class based shaping. There is an example on how to setup 
cb shape in the QoS SRND. Stick with that one.

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

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>

Från: Angel Perez [gorr...@hotmail.com]
Skickat: den 15 juni 2010 12:40
Till: ciscovoiceg...@gmail.com; osl osl
Ämne: Re: [OSL | CCIE_Voice] QOS FRF.12 & MLPP

Hi:

Just to add something to Matthew's reply, be sure that you set the correct 
compression method either frame relay (activated by default with auto qos voip 
trust in links with 768k bandwith or less) or class based (compress header ip 
rtp at desired class) .

You can't have both at the same time

hth


Date: Tue, 15 Jun 2010 05:28:59 -0500
From: ciscovoiceg...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] QOS FRF.12 & MLPP

Kobel,

In my opinion, you should only retain the "frame-relay ip rtp 
header-compression" under the frame-relay DLCI if you are asked to compress the 
rtp packets.  Because we're dealing with a slow-speed link, auto qos tries to 
be helpful by adding in this command.

My general stance when it comes to answering the QoS lab questions is to only 
configure what they ask you to setup.  Using auto qos is helpful to rough-in a 
configuration, but leaving in unnecessary elements does not demonstrate a 
mastery of the knowledge you are being tested on.  I will provide another 
example:

When you type "auto qos voip" several classes will be created.  One of those 
classes, called something like "remark," will set DSCP values on so-called 
rogue traffic masquerading as media or signaling traffic.  If the question does 
not ask you to perform that task, you'll want to remove the remark class.

I'm not sure if this helps, but it's my take on the subject.  My guess is that 
the lab would be specific whether they wanted class-based cRTP or not.


Matthew Berry

A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written



Vitals:

GVoice: +1.612.424.5044

Gmail: ciscovoiceg...@gmail.com<mailto:ciscovoiceg...@gmail.com>

Skype: ciscovoiceguru

Twitter: ciscovoiceguru



Cert Stats:

Cisco Cert Journey Began: Jan 1, 2009

1st Lab Attempt: Aug 16, 2010

On 6/15/2010 4:23 AM, kobel wrote:

Also, after the Auto QOS generates a lot of classes etc. We do edit few things 
here and there. Just wanted to confirm that is it a good practice to remove rtp 
header compression?
I use to remove it always but now I am getting conflicting feedback that should 
we remove it or not?


interface Serial0/2/0.1 point-to-point
bandwidth 256
frame-relay interface-dlci 301 CISCO
class AutoQoS-FR-Se0/2/0-301
auto qos voip trust
frame-relay ip rtp header-compression

I would appreciate any input in this regard.

you can configure cRTP in two ways. if the task doesn't explicitly ask for CB 
cRTP, I keep auto qos config - why waste time? I'm not aware of any drawback of 
this method.


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Re: [OSL | CCIE_Voice] QOS FRF.12 & MLPP

[Roger Källberg]

My vote on this goes to Matthew. If not clearly asked for, better to remove any 
unwanted parts that quto qos putts in.

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

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>

Från: Matthew Berry [ciscovoiceg...@gmail.com]
Skickat: den 15 juni 2010 12:28
Till: ccie_voice@onlinestudylist.com
Ämne: Re: [OSL | CCIE_Voice] QOS FRF.12 & MLPP

Kobel,

In my opinion, you should only retain the "frame-relay ip rtp 
header-compression" under the frame-relay DLCI if you are asked to compress the 
rtp packets.  Because we're dealing with a slow-speed link, auto qos tries to 
be helpful by adding in this command.

My general stance when it comes to answering the QoS lab questions is to only 
configure what they ask you to setup.  Using auto qos is helpful to rough-in a 
configuration, but leaving in unnecessary elements does not demonstrate a 
mastery of the knowledge you are being tested on.  I will provide another 
example:

When you type "auto qos voip" several classes will be created.  One of those 
classes, called something like "remark," will set DSCP values on so-called 
rogue traffic masquerading as media or signaling traffic.  If the question does 
not ask you to perform that task, you'll want to remove the remark class.

I'm not sure if this helps, but it's my take on the subject.  My guess is that 
the lab would be specific whether they wanted class-based cRTP or not.


Matthew Berry

A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written



Vitals:

GVoice: +1.612.424.5044

Gmail: ciscovoiceg...@gmail.com<mailto:ciscovoiceg...@gmail.com>

Skype: ciscovoiceguru

Twitter: ciscovoiceguru



Cert Stats:

Cisco Cert Journey Began: Jan 1, 2009

1st Lab Attempt: Aug 16, 2010

On 6/15/2010 4:23 AM, kobel wrote:

Also, after the Auto QOS generates a lot of classes etc. We do edit few things 
here and there. Just wanted to confirm that is it a good practice to remove rtp 
header compression?
I use to remove it always but now I am getting conflicting feedback that should 
we remove it or not?


interface Serial0/2/0.1 point-to-point
bandwidth 256
frame-relay interface-dlci 301 CISCO
class AutoQoS-FR-Se0/2/0-301
auto qos voip trust
frame-relay ip rtp header-compression

I would appreciate any input in this regard.

you can configure cRTP in two ways. if the task doesn't explicitly ask for CB 
cRTP, I keep auto qos config - why waste time? I'm not aware of any drawback of 
this method.


___
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Re: [OSL | CCIE_Voice] problem LAB 11 VOL1 CUE-CME license file- working on vrack

[Roger Källberg]

What config have you loaded? The ftp server on the UCCX aren't setup in all 
labs, use UCCX config from volume 2 or 1 week lab experience.

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

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>

Från: amr gaber [amrga...@gmail.com]
Skickat: den 14 juni 2010 02:13
Till: ccie_voice@onlinestudylist.com
Ämne: [OSL | CCIE_Voice] problem LAB 11 VOL1 CUE-CME license file- working on 
vrack

I try to upload CUE -CME license bu the command below
"software install clean url 
ftp://10.10.210.5/cue-vm-license_12mbx_cme_7.0.1.pkg username cisco password 
cisco"
please advise as soon as possible



Logging
se-10-10-202-2# $e_12mbx_cme_7.0.1.pkg username cisco password cisco


WARNING:: This command will install the necessary software to
WARNING:: complete a clean install.  It is recommended that a backup be done
WARNING:: before installing software.

Would you like to continue? [n]y

Downloading ftp cue-vm-license_12mbx_cme_7.0.1.pkg


Error: Download error
 Can not download cue-vm-license_12mbx_cme_7.0.1.pkg
error code 0 : error type 'couldn't connect to host'
se-10-10-202-2#
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Re: [OSL | CCIE_Voice] Spanning-tree portfast IPexpert!

[Roger Källberg]

Yes, that is 100% correct.

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

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>

Från: Ashar Siddiqui [siddas...@gmail.com]
Skickat: den 14 juni 2010 11:38
Till: Roger Källberg
Kopia: kobel; wolfsrudel; ccie_voice@onlinestudylist.com
Ämne: Re: [OSL | CCIE_Voice] Spanning-tree portfast IPexpert!

Roger,

So you mean on BR1 and BR2 switch we can use the following commands on Phone 
ports?

int fa1/0
switchport mode access
swicthport access vlan 130
switchport voice vlan 240
spanning-tree portfast
no shut
!



Roger Källberg wrote:
I always used the access port metod of setting up the HWIC-4ESW ports, or 
acctually not always. I started to use that metod when Vik told us about this 
during my boot camp in March.

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

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>

Från: kobel [findko...@gmail.com<mailto:findko...@gmail.com>]
Skickat: den 13 juni 2010 18:55
Till: wolfsrudel
Kopia: ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com>
Ämne: Re: [OSL | CCIE_Voice] Spanning-tree portfast IPexpert!

AFAIK, the portfast is effective only for access ports. that's why it's not 
visible on HWIC-4ESW ports, which are configured in trunk mode.

regards

On Sun, Jun 13, 2010 at 6:49 PM, wolfsrudel 
mailto:wolfsru...@gmail.com>> wrote:
portfast should be set on any access port where we like to avoid stp
delays (learning and such).
it's part of the de facto port config configuracion, unless were have
specific reasons not to do so. imho

hth

On 6/13/10, Ashar Siddiqui mailto:siddas...@gmail.com>> 
wrote:
> Hi,
>
> In Proctor lab HW-Switch I can see this command:
>
> interface FastEthernet1/0/2
>  switchport access vlan 10
>  switchport mode access
>  switchport voice vlan 20
>  spanning-tree portfast
>
>
> But "Spanning-tree portfast" is not used on BR1/BR2 ports where phones are
> connected. Any specific reason? I thought we will use this command anywhere
> where we want the ports not to come in Election process of Root bridge (STP)
> and we are sure that they won't create ant loops (like access ports or ports
> connected to phone).  Also they quickly go in forwarding state..Why are we
> not using this on Br1 and Br2?
>
> Ash>
>

--
Sent from my mobile device
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Re: [OSL | CCIE_Voice] Spanning-tree portfast IPexpert!

[Roger Källberg]

I always used the access port metod of setting up the HWIC-4ESW ports, or 
acctually not always. I started to use that metod when Vik told us about this 
during my boot camp in March.

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

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>

Från: kobel [findko...@gmail.com]
Skickat: den 13 juni 2010 18:55
Till: wolfsrudel
Kopia: ccie_voice@onlinestudylist.com
Ämne: Re: [OSL | CCIE_Voice] Spanning-tree portfast IPexpert!

AFAIK, the portfast is effective only for access ports. that's why it's not 
visible on HWIC-4ESW ports, which are configured in trunk mode.

regards

On Sun, Jun 13, 2010 at 6:49 PM, wolfsrudel 
mailto:wolfsru...@gmail.com>> wrote:
portfast should be set on any access port where we like to avoid stp
delays (learning and such).
it's part of the de facto port config configuracion, unless were have
specific reasons not to do so. imho

hth

On 6/13/10, Ashar Siddiqui mailto:siddas...@gmail.com>> 
wrote:
> Hi,
>
> In Proctor lab HW-Switch I can see this command:
>
> interface FastEthernet1/0/2
>  switchport access vlan 10
>  switchport mode access
>  switchport voice vlan 20
>  spanning-tree portfast
>
>
> But "Spanning-tree portfast" is not used on BR1/BR2 ports where phones are
> connected. Any specific reason? I thought we will use this command anywhere
> where we want the ports not to come in Election process of Root bridge (STP)
> and we are sure that they won't create ant loops (like access ports or ports
> connected to phone).  Also they quickly go in forwarding state..Why are we
> not using this on Br1 and Br2?
>
> Ash>
>

--
Sent from my mobile device
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Re: [OSL | CCIE_Voice] I Pronounce You "CCIE in Voice"

[Roger Källberg]

Congratulations. Yet another success that can be contributed to IPExperts 
superior material and instructors.

Enjoy your self :-)

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

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>

Från: Tariq Ahmad [...@tariqccie.com]
Skickat: den 12 juni 2010 07:38
Till: OSL Group
Ämne: [OSL | CCIE_Voice] I Pronounce You "CCIE in Voice"

I took Voice lab last Friday.Didn't get the results back Until Monday 
afternoon. Voila , I passed !

I have been following this Study List ever since i started my preparation.It 
has been very useful resource along the way.Even though i didn't use IPExpert's 
Study Material, i have/had great time studying Vik's blog posts & Free 
Vlectures . Vik, you are a great instructor !

Finally, thanks to Wayne for keeping up such an amazing OSL . Now, its time to 
start working on my next track.

Regards,

Tariq Ahmad
CCIE V # 26141
___
For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] I passed CCIE Voice (# 26199)

[Roger Källberg]

Hi A A,
I used the IPExpert End to End program and I read a lot of differnt Cisco Press 
books and other publications from Cisco, like the SRND and so on.

My next goal is to really relay and enjoy time with my family :-)

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

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>

Från: A A [...@hotmail.com]
Skickat: den 12 juni 2010 06:52
Till: Roger Källberg; ccie_voice@onlinestudylist.com
Ämne: RE: [OSL | CCIE_Voice] I passed CCIE Voice (# 26199)

Congrats mate!!

Hope you could share plans/materials you have used for preparations. Whats your 
next goal if i could ask?

Thanks,
Ahmad


From: roger.kallb...@cygate.se
To: ccie_voice@onlinestudylist.com
Date: Fri, 11 Jun 2010 14:13:23 +0200
Subject: [OSL | CCIE_Voice] I passed CCIE Voice (# 26199)

I took my lab yesterday, first attempt, just got the score report. I passed :-)

I will write down my strategy once I have landed from the cloud that I'm 
currently flying on. :-D

Roger Källberg
Consultant
Cygate AB
Eric Perssons väg 21, SE-217 62 MALMÖ

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>


Get a new e-mail account with Hotmail - Free. Sign-up 
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Re: [OSL | CCIE_Voice] Gatekeeper 4 Digit call

[Roger Källberg]

Hi,
Ash already made a really good explanation of what they do, but this is a Cisco 
statement regards to these commands that I got from one of my study mates 
during my preparation,

"Calls placed through a Cisco Unified Border Element may fail to connect when 
the originating or terminating H.323 device is a non-Cisco IOS VoIP device such 
as Cisco Unified Communications Manager.
The default behavior of H.323-to-H.323 calls through a Cisco Unified Border 
Element is to delay sending a H.225 Connect message to the originating H323 
device until the H245 TCS/MSD/OLC negotiation takes place. During this process, 
an H.225 Connect message with an H.245 address present from the terminating 
H.323 device is changed to an H.225 Progress message, followed by an H.225 
Facility message with the embedded H.245 address. This can cause connection 
failures if the originating H.323 device is waiting for the H.225 Connect 
message to begin the H245 TCS/MSD/OLC negotiation.
The h225 connect-passthru command is used to immediately pass H.225 connect 
messages from the trunking gateway to the outgoing gateway via a Cisco Unified 
Border Element.
Configuring the h225 connect-passthru command in H.323 voice-service 
configuration is recommended for all calls passed through the Cisco Unified 
Border Element. This command option will be present only when the 
allow-connections command is configured.
This command is often configured with the h245 passthru tcsnonstd-passthru 
command and emptycapability command when interworking is configured between 
non-Cisco IOS H.323 devices."


This lines summarize and explain the above pretty good

1. the remote phone is picking up . and sending an "H225 connect message" 
back to the CUBE.
2. The CUBE by default is not sending the H225 connect message back to the HQ 
phone because it is waiting for the H245 negotiation to take place.
3. The CUBE changes the H225 connect message to a progress message, followed by 
an H225 facility message and sends to the HQ phone.
4. The HQ phone is waiting for the H225 connect message to begin the H245 
negotiation.



Best of luck

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

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se

Från: ccieid1ot [ccieid...@gmail.com]
Skickat: den 11 juni 2010 21:30
Till: Roger Källberg
Kopia: Dani Bug; Angel Perez; osl osl
Ämne: Re: [OSL | CCIE_Voice] Gatekeeper 4 Digit call

Roger,

Could you explain what those commands do?

2010/6/11 Roger Källberg :
> Add this to your config
>
> voice service voip
>   h323
>   emptycapability
>   h225 connect-passthru
>   h245 passthru tcsnonstd-passthru
>
> Roger Källberg
> CCIE Voice (# 26199)
> Consultant
> Cygate AB
> Eric Perssons väg 21, SE-217 62 MALMÖ
>
> Direkt: +46108787498
> Växel: +46108787400
> roger.kallb...@cygate.se
> 
> Från: Dani Bug [daniyal.vo...@gmail.com]
> Skickat: den 11 juni 2010 17:43
> Till: Angel Perez
> Kopia: Roger Källberg; osl osl
> Ämne: Re: [OSL | CCIE_Voice] Gatekeeper 4 Digit call
>
> Hi Guys
> Thx for your help i reconfigure Router and now its working fine and didn't
> got the solution but trouble moved to in CUBE GW config same behaviour some
> time calls going sometime not now thinking to reconfigure the whole router
> again 
> also i am facing problem with CUBE when i call to UK phone is ringing i
> picked up phone no voice at UK site and HQ phone is still ringing  what
> could be the issue ...
> codec? or something else  ???
>
> any advise regarding CUBE call ?
>
> Thx
> Dani
>
> On Tue, Jun 8, 2010 at 3:56 AM, Angel Perez  wrote:
>>
>> Hi:
>>
>> I would brake your problems in two parts:
>>
>> - First: calls from ucm to ucme and ucme to ucm
>>
>> Delete the gateway command from the cube gateway, this should be enough to
>> make calls working from ucme to ucm,
>> if calls are still not working follow Kobel suggestion:
>>
>> zone prefix GK 2 GW-PRiority 10 GK_Trunk_1
>> zone prefix GK 2 GW-PRiority 0 CUCME
>>
>> zone prefix GK 3 GW-PRiority 10 GK_Trunk_1
>> zone prefix GK 3 GW-PRiority 0 CUCME
>>
>> You will need to add the tech prefix 1 for these calls (as you were doing)
>>
>> then for calls from ucm to ucme let say from 2001 to 4001 add the
>> following:
>>
>> zone prefix GK 4 GW-PRiority 10 CUCME
>> zone prefix GK 4 GW-PRiority 0 GK_Trunk_1
>>
>> I'm assuming that you have the correct route pattern, css and dial-peer
>> configuration on  ucm and ucme.
>>
>> - Second: after that explain a littel bit more your cube call (scenarion
>> number dialed, etc)
>>
>

Re: [OSL | CCIE_Voice] Gatekeeper 4 Digit call

[Roger Källberg]

Add this to your config

voice service voip
  h323
  emptycapability
  h225 connect-passthru
  h245 passthru tcsnonstd-passthru

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

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>

Från: Dani Bug [daniyal.vo...@gmail.com]
Skickat: den 11 juni 2010 17:43
Till: Angel Perez
Kopia: Roger Källberg; osl osl
Ämne: Re: [OSL | CCIE_Voice] Gatekeeper 4 Digit call

Hi Guys
Thx for your help i reconfigure Router and now its working fine and didn't got 
the solution but trouble moved to in CUBE GW config same behaviour some time 
calls going sometime not now thinking to reconfigure the whole router again 
also i am facing problem with CUBE when i call to UK phone is ringing i picked 
up phone no voice at UK site and HQ phone is still ringing  what could be 
the issue ...
codec? or something else  ???

any advise regarding CUBE call ?

Thx
Dani

On Tue, Jun 8, 2010 at 3:56 AM, Angel Perez 
mailto:gorr...@hotmail.com>> wrote:
Hi:

I would brake your problems in two parts:

- First: calls from ucm to ucme and ucme to ucm

Delete the gateway command from the cube gateway, this should be enough to make 
calls working from ucme to ucm,
if calls are still not working follow Kobel suggestion:

zone prefix GK 2 GW-PRiority 10 GK_Trunk_1
zone prefix GK 2 GW-PRiority 0 CUCME

zone prefix GK 3 GW-PRiority 10 GK_Trunk_1
zone prefix GK 3 GW-PRiority 0 CUCME

You will need to add the tech prefix 1 for these calls (as you were doing)

then for calls from ucm to ucme let say from 2001 to 4001 add the following:

zone prefix GK 4 GW-PRiority 10 CUCME
zone prefix GK 4 GW-PRiority 0 GK_Trunk_1

I'm assuming that you have the correct route pattern, css and dial-peer 
configuration on  ucm and ucme.

- Second: after that explain a littel bit more your cube call (scenarion number 
dialed, etc)

hth

From: roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>
To: daniyal.vo...@gmail.com<mailto:daniyal.vo...@gmail.com>
Date: Tue, 8 Jun 2010 08:41:26 +0200

CC: ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com>
Subject: Re: [OSL | CCIE_Voice] Gatekeeper 4 Digit call

Hi Dani,

Have you looked in the terminating gw, with deb voip dialpeer, do you see if 
the call reaches the CME box? To route the call to the end point in CME and in 
the UCM you need to remove the tech prefix. This can be done with either voice 
translation rule or num-exp on the CME and with a TP on the UCM. Just make sure 
that this TP is in a PT that only the GK can see in it's incoming CSS. This is 
not strictly needed, just makes it a lot easier to troubleshoot if needed.


Brgds,
Roger Källberg

Från: Dani Bug [daniyal.vo...@gmail.com<mailto:daniyal.vo...@gmail.com>]
Skickat: den 8 juni 2010 01:08
Till: kobel
Kopia: osl osl
Ämne: Re: [OSL | CCIE_Voice] Gatekeeper 4 Digit call

still no luck but in the debug even call is not hitting to CUBE not a single 
time so i would say it's not issue with load balancing even thu i took out CUBE 
config and try make a normal call but still no luck ..
also i am facing problem with CUBE when i call to UK phone is ringing i picked 
up phone no voice at UK site and HQ phone is still ringing  what could be 
the issue ...
codec? or something else  ???
any idea .. about that advise/question appreciated 

thanks
Dani
On Mon, Jun 7, 2010 at 5:27 PM, kobel 
mailto:findko...@gmail.com>> wrote:
try configuring higher preference for CUCM trunk for prefix 2* . It seems that 
GK performs load balancing between the CUBE and this trunk and only one of them 
works.


regards


On Mon, Jun 7, 2010 at 10:36 PM, Dani Bug 
mailto:daniyal.vo...@gmail.com>> wrote:
I tried without invia/outvia still no luck ...:(

HQ-R1#sh gatekeeper gw
GATEWAY TYPE PREFIX TABLE
=
Prefix: 1*
  Zone GK master gateway list:
142.102.64.254:1720<http://142.102.64.254:1720/> CUBE
172.25.105.101:1720<http://172.25.105.101:1720/> GK_Trunk_1
Prefix: 852*
  Zone GK master gateway list:
142.102.66.254:1720<http://142.102.66.254:1720/> CUCME

Here is debug success call from 4001 to 2001
HQ-R1#
Jun  7 22:07:34.108: ////GK/gk_process: QUEUE_EVENT 
(minor 0) wakeup
Jun  7 22:07:34.172: ////GK/gk_process: QUEUE_EVENT 
(minor 0) wakeup
Jun  7 22:07:34.172: ////GK/gk_rassrv_arq: 
arqp=0x49E870B4,crv=0x59, answerCall=0
Jun  7 22:07:34.172: ////GK/gk_rassrv_sep_arq: ARQ 
Didn't use GK_AAA_PROC
Jun  7 22:07:34.172: //E801AC8F8482/E80248B78484/GK/gk_dns_query: No Name 
servers
Jun  7 22:07:34.172: //E801AC8F8482/E80248B78484/GK/rassrv_get_addrinfo: 
(12001) Matched tech-prefix 1
Jun  7 22:

Re: [OSL | CCIE_Voice] I passed CCIE Voice (# 26199)

[Roger Källberg]

Write down of my strategy going in to the lab.

My strategy was to first put every question and the points on one of the scrap 
papers, even before I read the lab. I then crossed of each off these when I had 
configured it and most important verified that it work as requested. This was 
very much like Pushkar have written in his excellent blog, 
http://pushkarbhatkoti.wordpress.com/category/ccie-voice-lab-strategies/. Thank 
you Push for that very helpful post.
Then I read trough the entire lab, but just browsed trough the dial-plan 
section, to see if there were anything in it that I needed to take into 
consideration in the other section.
Then my strategy was this:

· Do everything with infrastructure, including basic VGW config 
(incoming call and outgoing emergency service call. I also included basic setup 
of CUE in to this section.

· Do QoS for both campus and wan. I didn’t really stick 100% to this, 
but that was my intension going in to it. I did some of the next bullet point 
before QoS, like media and SRST.

· Configure everything, but the dial plan. This included every 
customisation of the phones, setup off VM, SRST, media, and every other think 
that has to do with applications and so on.
My goal was to finish this by lunch, I didn’t quite manage that, I needed about 
1-1½ hours after the break to finish this.
Next part of my strategy was this:

· Configure dial-plan question by question, paying special attention to 
what the question asked for when it comes to type of number and number plan. I 
verified all along when I configured, shutting down the primary path if that 
was the case, and testing the backup path. I then redid the test of the primary 
path after I had brought that up again, just to see that it still worked.
Yet again I had a goal with this section and that was to finish the dial-plan 
with 1-1½ hours to spare. This time I planned to have used for going over each 
of the questions once more and marking them of on the scrap paper one more 
time.  But because of that I didn’t manage to stick to my pre lunch game plan I 
ended up configuring right to the very end off my lab. When the proctor 
announced that it was five minutes left I was just about wrap up my last dial 
plan question. Last thing I did was to verify that all ISDN links were in the 
up state, saved my switch and router config’s for the last time and then I 
removed all my text files that I had stored on the desktop. In the end I had 
1-2 minutes to spare, luckily it ended up to be just enough to pass.

Thank you everyone that have congratulated me.

I also want to thank Vik, Amy and Mark for there hard work with the update of 
the material after the update of the blueprint last year. You guys rock.

Apart from this I would like to send a special thanks to my study mates, 
Matthew Berry, Patrick Fischer, Vishal Preenja, Jody Marshall and Sean 
Hurricane. Your help was invaluable and a monumental help in my preparation.

Regards
Roger Källberg
Consultant
Cygate AB
Eric Perssons väg 21, SE-217 62 MALMÖ

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>

Från: Roger Källberg [roger.kallb...@cygate.se]
Skickat: den 11 juni 2010 14:13
Till: OSL Group
Ämne: [OSL | CCIE_Voice] I passed CCIE Voice (# 26199)

I took my lab yesterday, first attempt, just got the score report. I passed :-)

I will write down my strategy once I have landed from the cloud that I'm 
currently flying on. :-D

Roger Källberg
Consultant
Cygate AB
Eric Perssons väg 21, SE-217 62 MALMÖ

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

[OSL | CCIE_Voice] I passed CCIE Voice (# 26199)

[Roger Källberg]

I took my lab yesterday, first attempt, just got the score report. I passed :-)

I will write down my strategy once I have landed from the cloud that I'm 
currently flying on. :-D

Roger Källberg
Consultant
Cygate AB
Eric Perssons väg 21, SE-217 62 MALMÖ

Direkt: +46108787498
Växel: +46108787400
roger.kallb...@cygate.se<mailto:roger.kallb...@cygate.se>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Gatekeeper 4 Digit call

[Roger Källberg]

Hi Dani,
Have you looked in the terminating gw, with deb voip dialpeer, do you see if 
the call reaches the CME box? To route the call to the end point in CME and in 
the UCM you need to remove the tech prefix. This can be done with either voice 
translation rule or num-exp on the CME and with a TP on the UCM. Just make sure 
that this TP is in a PT that only the GK can see in it's incoming CSS. This is 
not strictly needed, just makes it a lot easier to troubleshoot if needed.

Brgds,
Roger Källberg

Från: Dani Bug [daniyal.vo...@gmail.com]
Skickat: den 8 juni 2010 01:08
Till: kobel
Kopia: osl osl
Ämne: Re: [OSL | CCIE_Voice] Gatekeeper 4 Digit call

still no luck but in the debug even call is not hitting to CUBE not a single 
time so i would say it's not issue with load balancing even thu i took out CUBE 
config and try make a normal call but still no luck ..
also i am facing problem with CUBE when i call to UK phone is ringing i picked 
up phone no voice at UK site and HQ phone is still ringing  what could be 
the issue ...
codec? or something else  ???
any idea .. about that advise/question appreciated 

thanks
Dani
On Mon, Jun 7, 2010 at 5:27 PM, kobel 
mailto:findko...@gmail.com>> wrote:
try configuring higher preference for CUCM trunk for prefix 2* . It seems that 
GK performs load balancing between the CUBE and this trunk and only one of them 
works.


regards


On Mon, Jun 7, 2010 at 10:36 PM, Dani Bug 
mailto:daniyal.vo...@gmail.com>> wrote:
I tried without invia/outvia still no luck ...:(

HQ-R1#sh gatekeeper gw
GATEWAY TYPE PREFIX TABLE
=
Prefix: 1*
  Zone GK master gateway list:
142.102.64.254:1720<http://142.102.64.254:1720/> CUBE
172.25.105.101:1720<http://172.25.105.101:1720/> GK_Trunk_1
Prefix: 852*
  Zone GK master gateway list:
142.102.66.254:1720<http://142.102.66.254:1720/> CUCME

Here is debug success call from 4001 to 2001
HQ-R1#
Jun  7 22:07:34.108: ////GK/gk_process: QUEUE_EVENT 
(minor 0) wakeup
Jun  7 22:07:34.172: ////GK/gk_process: QUEUE_EVENT 
(minor 0) wakeup
Jun  7 22:07:34.172: ////GK/gk_rassrv_arq: 
arqp=0x49E870B4,crv=0x59, answerCall=0
Jun  7 22:07:34.172: ////GK/gk_rassrv_sep_arq: ARQ 
Didn't use GK_AAA_PROC
Jun  7 22:07:34.172: //E801AC8F8482/E80248B78484/GK/gk_dns_query: No Name 
servers
Jun  7 22:07:34.172: //E801AC8F8482/E80248B78484/GK/rassrv_get_addrinfo: 
(12001) Matched tech-prefix 1
Jun  7 22:07:34.176: //E801AC8F8482/E80248B78484/GK/rassrv_get_addrinfo: 
(12001) Matched zone prefix 2 and remainder 001
Jun  7 22:07:34.176: 
////GK/gk_rassrv_get_ingress_network: ARQ non-std 
ingress network = 1
Jun  7 22:07:34.176: //E801AC8F8482/E80248B78484/GK/rassrv_arq_select_viazone: 
about to check the source side, src_zonep=0x49FFA4B8
Jun  7 22:07:34.176: //E801AC8F8482/E80248B78
HQ-R1#484/GK/rassrv_arq_select_viazone: matched zone is GK, and z_invianamelen=2
Jun  7 22:07:34.176: //E801AC8F8482/E80248B78484/GK/rassrv_arq_select_viazone   
   and z_invianamep=GK
Jun  7 22:07:34.176: //E801AC8F8482/E80248B78484/GK/rassrv_arq_select_viazone: 
about to check the destination side, dst_zonep=0x49FFA4B8
Jun  7 22:07:34.176: //E801AC8F8482/E80248B78484/GK/rassrv_arq_select_viazone: 
matched zone is GK, and z_outvianamelen=2
Jun  7 22:07:34.176: //E801AC8F8482/E80248B78484/GK/rassrv_arq_select_viazone   
   and z_outvianamep=GK
Jun  7 22:07:34.176: //E801AC8F8482/E80248B78484/GK/rassrv_arq_select_viazone: 
Received ARQ for a zone (GK) that has an outviazone (GK) specified, but I am 
that viazone.  Continue normal ARQ processing
Jun  7 22:07:34.176: 
////GK/gk_rassrv_get_ingress_network: ARQ non-std 
ingress network = 1
Jun  7 22:07:34.192: ////GK/gk_process: QUEUE_EVENT 
(minor 0) wakeup
Jun  7 22:07:34.192: ////GK/gk_rassrv_arq: 
arqp=0x49E870B4,crv=0x8059, answerCall=1
Jun  7 22:07:34.192: //E801AC8F8482/E80248B78484/GK/gk_rassrv_dep_arq: ARQ 
Didn't use GK_AAA_PROC
Jun  7 22:07:35.268: ////GK/gk_process: QUEUE_EVENT 
(minor 0) wakeup
Jun  7 22:07:35.268: ////GK/gk_process: QUEUE_EVENT 
(minor 0) wakeup
HQ-R1#
=
Here is debug for Failed call from 4001 to 2001
HQ-R1#
Jun  7 22:07:44.888: ////GK/gk_process: QUEUE_EVENT 
(minor 0) wakeup
Jun  7 22:07:44.892: ////GK/gk_rassrv_arq: 
arqp=0x49EB9F60,crv=0x5B, answerCall=0
Jun  7 22:07:44.892: ////GK/gk_rassrv_sep_arq: ARQ 
Didn't use GK_AAA_PROC
Jun  7 22:07:44.892: //EE6A58F08490/EE6A58F08492/GK/gk_dns_query: No Name 
servers
Jun  7 22:07:44.892: //EE6A58F08490/EE6A58F08492/GK/rassrv_get_addrinfo: 
(12001) Matched tech-prefix 1
Jun  7 22:07:44.892: //EE

Re: [OSL | CCIE_Voice] VOL2 LAB2 weird cme to ucm call over gk problem

[Roger Källberg]

You might be hitting this well know bug, CSCsl74701

Search on OSL and you will find many references to this. :-)

Roger Källberg

Från: Hobson Kevin [kevin.hobson2...@ntlworld.com]
Skickat: den 7 juni 2010 13:16
Till: ccie_voice@onlinestudylist.com
Ämne: Re: [OSL | CCIE_Voice] VOL2 LAB2 weird cme to ucm call over gk problem

Hi All,

I have a really werid issue with calls from UCME to UCM.

The issue is that if i call from BR2 to HQ and do a show gatek call it shows 
the bandwidth being usesd as 128k.  See below:

gk-cube#sh gatek call
Total number of active calls = 1.
 GATEKEEPER CALL INFO
 
LocalCallIDAge(secs)   BW
82-43591   21  16(Kbps)
 Endpt(s): Alias E.164Addr
   src EP: BR2-RTR   3001
   CallSignalAddr  Port  RASSignalAddr   Port
   10.10.110.3 1720  10.10.110.3 58865
 Endpt(s): Alias E.164Addr
   dst EP: gk-trunk_21#5001
   CallSignalAddr  Port  RASSignalAddr   Port
   10.10.210.111720  10.10.210.1132786


If i debug h225 asn1 i see that the CME is requesting 16k but the UCM is asking 
for 128k.  See below:

CME

value RasMessage ::= admissionRequest :
{
  requestSeqNum 9277
  callType pointToPoint : NULL
  callModel direct : NULL
  endpointIdentifier {"49887C840001"}
  destinationInfo
  {
dialedDigits : "1#5002"
  }
  srcInfo
  {
dialedDigits : "3001",
h323-ID : {"BR2-RTR"}
  }
  bandWidth 160

UCM

value RasMessage ::= admissionRequest :
{
  requestSeqNum 1393
  callType pointToPoint : NULL
  endpointIdentifier {"4857A5C80003"}
  destinationInfo
  {
dialedDigits : "5002"
  }
  srcInfo
  {
dialedDigits : "3001"
  }
  srcCallSignalAddress ipAddress :
  {
ip ''H
port 20946
  }
  bandWidth 1280



When the call is connected i get no codec sent on the hq phone and g729 on the 
BR2 phone.



If i then enable BRQ on the UCM services when the phone rings it requests 128k 
again:

gk-cube#sh gatek call
Total number of active calls = 1.
 GATEKEEPER CALL INFO
 
LocalCallIDAge(secs)   BW
85-238615  128(Kbps)
 Endpt(s): Alias E.164Addr
   src EP: BR2-RTR   3001
   CallSignalAddr  Port  RASSignalAddr   Port
   10.10.110.3 1720  10.10.110.3 58865
 Endpt(s): Alias E.164Addr
   dst EP: gk-trunk_21#5002
   CallSignalAddr  Port  RASSignalAddr   Port
   10.10.210.111720  10.10.210.1132786


But when it connects this goes down to 16k:

gk-cube#sh gatek call
Total number of active calls = 1.
 GATEKEEPER CALL INFO
 
LocalCallIDAge(secs)   BW
86-24672   8   16(Kbps)
 Endpt(s): Alias E.164Addr
   src EP: BR2-RTR   3001
   CallSignalAddr  Port  RASSignalAddr   Port
   10.10.110.3 1720  10.10.110.3 58865
 Endpt(s): Alias E.164Addr
   dst EP: gk-trunk_21#5001
   CallSignalAddr  Port  RASSignalAddr   Port
   10.10.210.111720  10.10.210.1132786


The phones also show g729 on both of them for the codec in use.

The region is g729 and the dp is assigned this region.

A ucm call the other way only requests 16k.

All help appreciated,

On 7 June 2010 11:46, 
mailto:ccie_voice-requ...@onlinestudylist.com>>
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Today's Topics:

  1. Setting up Voicemail to send Email - CME/CUE 7.0 (Ashar Siddiqui)
  2. Re: Setting up Voicemail to send Email - CME/CUE 7.0 (Angel Perez)
  3. Re: Setting up Voicemail to send Email - CME/CUE 7.0
 (kerboute kerboute)


--

Message: 1
Date: Mon, 7 Jun 2010 11:28:09 +0100
From:

Re: [OSL | CCIE_Voice] Attendant console link ?

[Roger Källberg]

AC is still a testable topic, see this url 
https://supportforums.cisco.com/message/3012407#3012407.

I guess that's why he asked for it.

Roger Källberg
Consultant
Cygate AB

Från: kerboute kerboute [naoufal.kerbo...@cbi.ma]
Skickat: den 1 juni 2010 10:29
Till: Pavan K
Kopia: ccie_voice@onlinestudylist.com
Ämne: Re: [OSL | CCIE_Voice] Attendant console link ?

attendant console is end of life for CUCM 7
You need Cisco unified attendant console server


On 05/31/2010 11:28 PM, Pavan K wrote:
Does anybody have a link to / copy of the attendant console plugin ?

--Thanks in advance.
- Pavan


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Re: [OSL | CCIE_Voice] Vol 2 Lab 1 Question 4.6 - Time Period / Time Schedule in UCM

[Roger Källberg]

Hi Matthew,
When I did this lab I did is as you described it and as far as I could tell 
accomplished the asked for result. This was as you wrote a much easier and 
quicker way off achieving what the question asked for, so I would go down that 
path any day off the week. :-)


Brgds,
Roger Källberg

Från: Matthew Berry [ciscovoiceg...@gmail.com]
Skickat: den 31 maj 2010 18:55
Till: OSL
Ämne: [OSL | CCIE_Voice] Vol 2 Lab 1 Question 4.6 - Time Period / Time Schedule 
in UCM

Question 4.6 asks me to restrict international dialing outside of normal 
business hours.  In the question, there is no mention of a custom "blocked" 
greeting that must be played by the annunciator when a call is blocked.

In the solutions guide to lab one, IPexpert actually configures three time 
periods: "M-F Evening", "M-F Morning", and "Weekends".  They are essentially 
going the route of configuring the OFF periods instead of the ON periods (i.e. 
times when international dialing should be allowed).

They assign the Time Schedule (made up of the three time periods) to a 
partition called PT-TOD.  They then create a TP in the PT-TOD partition with a 
"Block this pattern - No error" action.

I am wondering if the same results could be created by putting the 
international dialing pattern in a partition (PT-INTL-ALLOW) and only setting 
that partition to be active from 7am - 7pm weekdays.  It seems like it would be 
a lot less work and accomplish the same results.

Thoughts?  Can someone confirm my suspicions?

--

Matthew Berry

A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written



Vitals:

GVoice: +1.612.424.5044

Gmail: ciscovoiceg...@gmail.com<mailto:ciscovoiceg...@gmail.com>

Skype: ciscovoiceguru

Twitter: ciscovoiceguru



Cert Stats:

Cisco Cert Journey Began: Jan 1, 2009

1st Lab Attempt: Aug 16, 2010
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Re: [OSL | CCIE_Voice] MIPS per Conference Call and Transcoding Session

[Roger Källberg]

Hi Matthew,
For the conferencing part it will always use an entire DSP module, ie 240 MIPS. 
That is even if you configure max session as 1, ie one half of a DSP-16 module, 
it will anyway reserve the entire module. So it make little sence to set this 
setting to 1, I'd always set it to an even number, if not otherwise stated in 
the question of cource ;-)

For the transcoding part take a look at this tread, 
http://www.mail-rchive.com/ccie_voice@onlinestudylist.com/msg16649.html

Brgds,


Roger Källberg
Consultant
Cygate AB

Från: Matthew Berry [ciscovoiceg...@gmail.com]
Skickat: den 1 juni 2010 03:58
Till: OSL
Ämne: [OSL | CCIE_Voice] MIPS per Conference Call and Transcoding Session

The solutions guide for lab one states that a conference session takes up 240 
MIPS and a transcoder session takes 30-40 MIPS for a high complexity codec.

Where do you find that information?

I found the voice termination MIPS table in the CUCM SRND, but there was no 
mention of transcoding/conferencing MIPS usage.

Help?
--

Matthew Berry

A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written



Vitals:

GVoice: +1.612.424.5044

Gmail: ciscovoiceg...@gmail.com<mailto:ciscovoiceg...@gmail.com>

Skype: ciscovoiceguru

Twitter: ciscovoiceguru



Cert Stats:

Cisco Cert Journey Began: Jan 1, 2009

1st Lab Attempt: Aug 16, 2010
___
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Re: [OSL | CCIE_Voice] Adapt the playback of messages in UC

[Roger Källberg]

Thanks Ash.

Roger Källberg

Från: Ashar Siddiqui [siddas...@gmail.com]
Skickat: den 26 maj 2010 19:24
Till: Roger Källberg
Kopia: OSL Group
Ämne: Re: [OSL | CCIE_Voice] Adapt the playback of messages in UC

Hi Roger,

Go into that user in Unity, go into Edit> Playback messages > " Before Playing 
Each Message, Play" and configure it there.

I hope it helps.

Ash>

Roger Källberg wrote:

I confess, Unity is not one of my strongest areas :-), so here goes.



How would I go about to configure something like this, "For HQ Phone1 make sure 
if PSTN caller leaves a voicemail, the HQ user can hear the calling number of 
the PSTN caller and the message disposition time before playback of the 
message."



Best regards



Roger Källberg



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[OSL | CCIE_Voice] Adapt the playback of messages in UC

[Roger Källberg]

I confess, Unity is not one of my strongest areas :-), so here goes.



How would I go about to configure something like this, "For HQ Phone1 make sure 
if PSTN caller leaves a voicemail, the HQ user can hear the calling number of 
the PSTN caller and the message disposition time before playback of the 
message."



Best regards



Roger Källberg
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Re: [OSL | CCIE_Voice] Not getting "PLUS" on my Phones

[Roger Källberg]

Hi Ash,
For the CME side have a look at this, 
http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg16751.html

The problem is with CME 7.0

Roger Källberg

Från: Ashar Siddiqui [siddas...@gmail.com]
Skickat: den 23 maj 2010 16:29
Till: Ehab Salem
Kopia: ccie_voice@onlinestudylist.com
Ämne: Re: [OSL | CCIE_Voice] Not getting "PLUS" on my Phones

Thanks man!

Solved! I prefixed + for unknown type at GW in CUCM..I actually had this in my 
mind and the reason I didn't try that because I wanted to solve it using 
translation rules...why? because the same problem I have calling from Site B to 
Site C (CME-H323). I am not getting + when dialing from Site B to site C...I 
mean I am getting at the bottom of the phone as From +16178631001 but at the 
top of site C phone it is still From SBPH1 (16178631001).

Is this something to do with CME?

Thanks,
Ash


Ehab Salem wrote:
You can prefix plus to the incoming calling number from Gateway page in CUCM…


Ehab M. Salem

From: Ashar Siddiqui [mailto:siddas...@gmail.com]
Sent: Sunday, May 23, 2010 4:29 PM
To: Wael Agina
Cc: ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com>
Subject: Re: [OSL | CCIE_Voice] Not getting "PLUS" on my Phones

I am glad you mentioned those rules...I have tried all those rules before 
exactly...no joy...this is why I wrote in my last email that I tried all my 
translation rule skills..  :)



voice translation-rule 99
 rule 1 /^34\(.*\)/ /+34\1/ type any unknown plan any unknown
 rule 2 // /+/
 rule 3 // /+/ type any unknown plan any unknown
 rule 4 /^34/ /+\0/


Even after all this...

R2#
May 23 12:27:59.440: ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8  callref = 0x00AB
Bearer Capability i = 0x9090A2
Standard = CCITT
Transfer Capability = 3.1kHz Audio
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98381
Exclusive, Channel 1
Progress Ind i = 0x8583 - Origination address is non-ISDN
Display i = 'SCPH1'
Calling Party Number i = 0x0080, '3432143001'
Plan:Unknown, Type:Unknown
Called Party Number i = 0x80, '6178631001'
Plan:Unknown, Type:Unknown
May 23
R2# 12:27:59.440: ISDN Se0/0/0:23 Q931: Received SETUP  callref = 0x80AB callID 
= 0x0019 switch = primary-ni interface = User
May 23 12:27:59.460: ISDN Se0/0/0:23 Q931: TX -> CALL_PROC pd = 8  callref = 
0x80AB
Channel ID i = 0xA98381
Exclusive, Channel 1
May 23 12:27:59.588: ISDN Se0/0/0:23 Q931: TX -> ALERTING pd = 8  callref = 
0x80AB
R2#
May 23 12:28:01.972: ISDN Se0/0/0:23 Q931: RX <- DISCONNECT pd = 8  callref = 
0x00AB
Cause i = 0x8290 - Normal call clearing
May 23 12:28:01.976: ISDN Se0/0/0:23 Q931: TX -> RELEASE pd = 8  callref = 
0x80AB
May 23 12:28:01.988: ISDN Se0/0/0:23 Q931: RX <- RELEASE_COMP pd = 8  callref = 
0x00AB


Thanks,
Ash>

Wael Agina wrote:
Try This and keep us updated

1-
voice translation-rule 99
 rule 1 // /+\0/

2-
If above working then make it specific for 34* numbers
voice translation-rule 99
 rule 1 /^34/ /+\0/

3- Last resort try num-exp  ===  this will affect both direction calls and any 
calling passing the router !!!
num-exp 3432143... +3432143...


!
Thanks and Best Regards,
Wael Agina


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Re: [OSL | CCIE_Voice] translation rule

[Roger Källberg]

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


Roger Källberg
Consultant
Cygate AB

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

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

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

Ash>

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

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

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

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

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

voice translation-profile Drop-Unknown
  translate calling 1

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

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

Regards,
Wael Agina

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

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

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


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

Ash>

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--

Thanks and Best Regards,
Wael Agina

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