Re: [OSL | CCIE_Voice] CME After Hours Override

[Cristi Radescu]

How did you configure the block pattern?
If you defined it with option "7-24" (after-hours block pattern pattern-tag 
pattern [7-24]) that pattern will not be overrided by code.


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jason Granat
Sent: 22 august 2009 04:30
To: 'OSL Group'
Subject: [OSL | CCIE_Voice] CME After Hours Override

Hi all,

I am having trouble with what I thought should be a simple task. In CME I 
created an after hours block pattern, which works. I also configured an after 
hours override but this is not working. What is the best way to troubleshoot 
this?

Thanks,

Jason




http://slash128.com
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Re: [OSL | CCIE_Voice] auto qos

[Cristi Radescu]

Hi Mick,



Under your Virtual template you will have the following commands:

- "ppp multilink" - enables MLP;

- "ppp multilink interleave" - enables real-time packet interleaving;

- "ppp multilink fragment-delay 10 " - configures a maximum fragment delay of 
10 ms;

In this manner Auto QoS configure  LFI for MLP.



Hth,

Cristian



-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mick Vaites
Sent: 29 iulie 2009 20:29
To: Vik Malhi; Jonathan Charles
Cc: OSL Group; ccie_voice-boun...@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] auto qos



Hi Vik,



I saw the "auto qos voip fr-atm" in the proctors guide and was wondering

about this.



I'm probably doing something wrong but my solution was to create

Virtual-template then apply multilink PPP + fragment/inter afterwards. -

along with map-class/policy-map/class-map with the defined parameters

applied.



Well I've just cleaned the config back to initial configs and then tried the

"auto qos voip fr-atm" within the interface-dlci and it created a

map-class/class-map/policy-map/rmon -- but I'm not sure how MLP-LFI comes

in.



Sorry if I'm being thick here ;-)



Best regards



Mick



E: m...@pobox.net.uk







> From: Vik Malhi 

> Date: Wed, 29 Jul 2009 10:20:33 -0700

> To: Jonathan Charles 

> Cc: OSL Group ,

> 

> Subject: Re: [OSL | CCIE_Voice] auto qos

>

> If you have FRTS enabled on the physical interface (WAN) at HQ which is used

> for the 2 x PVC's to each of the branch sites then the default CIR is 56kbps

> and the default MINCIR is 28kbps. This will have MAJOR repercussions for

> your branch sites.

>

> You should ensure a frame-relay map-class is used to set CIR/MINCIR/BC/BE

> for BOTH PVC's. The auto-qos macros will only attach a map-class to the DLCI

> where you are configuring auto-qos.

>

> Use "show frame-relay pvc XXX" to confirm cir/mincir. |

>

>

> --

> Vik Malhi ­ CCIE #13890

> Senior Technical Instructor - IPexpert, Inc.

>

> Telephone: +1.810.326.1444

> Fax: +1.810.454.0130

> Mailto: vma...@ipexpert.com

>

>

> Join our free online support and peer group communities:

> http://www.IPexpert.com/communities

> IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand

> and Audio Certification Training Tools for the Cisco CCIE R&S Lab, CCIE

> Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage

> Lab Certifications.

>

>

>

>

>

>

>

>> From: Jonathan Charles 

>> Date: Wed, 29 Jul 2009 11:58:39 -0500

>> To: Vik Malhi 

>> Cc: c george , OSL Group

>> , 

>> Subject: Re: [OSL | CCIE_Voice] auto qos

>>

>> On this same subject, is it possible for the auto qos voip fr-atm to

>> create a queue that will starve out all other traffic and destroy your

>> network?

>>

>>

>> Jonathan

>>

>> On Wed, Jul 29, 2009 at 9:11 AM, Vik Malhi wrote:

>>> Yep- ³auto qos voip fr-atm² within the DLCI will configure MLP LFI.

>>> --

>>> Vik Malhi ­ CCIE #13890

>>> Senior Technical Instructor - IPexpert, Inc.

>>>

>>> Telephone: +1.810.326.1444

>>> Fax: +1.810.454.0130

>>> Mailto: vma...@ipexpert.com

>>>

>>>

>>> Join our free online support and peer group communities:

>>> http://www.IPexpert.com/communities

>>> IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand

>>> and Audio Certification Training Tools for the Cisco CCIE R&S Lab, CCIE

>>> Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage

>>> Lab Certifications.

>>>

>>>

>>>

>>>

>>>

>>>

>>>

>>> 

>>> From: c george 

>>> Date: Tue, 28 Jul 2009 20:12:34 +

>>> To: OSL Group ,

>>> 

>>> Subject: [OSL | CCIE_Voice] auto qos

>>>

>>>

>>> If you are instructed in the lab not to use fr.12, is there any way you

>>> could still use auto qos?

>>>

>>> Respectfully Charles George

>>>

>>>

>>>

>>> 

>>> NEW mobile Hotmail. Optimized for YOUR phone. Click here.

>>> 

>>> 

>>> ___

>>> 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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For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com
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Re: [OSL | CCIE_Voice] Media resource registration error

[Cristi Radescu]

You may verify "Run Flag" under Service parameters. It should be True by 
default but you can give it a try. If this will not work try restarting all 
server. There can be strange things under Vmware.


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ravindra Lakpriya
Sent: 29 iulie 2009 20:13
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Media resource registration error

Greetings guys,

i have setup CUCM 7cluster top of vmware. and i have activate Cisco IP Voice 
Media Streaming App on the new subscriber. then under media resource i have 
checked the registration and it says unknown. i tried to restart the media 
streaming app. no luck. Then i disable the service and enable it again.

results still the same.

any idea guys

Thanks

--
Ravindra Lakpriya
+94 773 532 094
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Re: [OSL | CCIE_Voice] CM to GK BRQ behavior

[Cristi Radescu]

Did you put Gatekeeper trunk in a G729-only region?
You only configured CAC mechanism until now.



From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Scott ODonnell
Sent: Wednesday, July 08, 2009 11:04 PM
To: OSL Group
Subject: [OSL | CCIE_Voice] CM to GK BRQ behavior

I'm seeing something strange in making calls from CM to CME via GK.

I've enabled the BRQ service parameter in CM.
I've included "bandwidth total default 16" in my gk config and did a shut/no 
shut

When I make calls from CM to CME the deb h225 asn1 shows (I think) that 128k is 
being requested.

Am I missing something obivous here ?
Currently all my calls get rejected from the GK and go via the HQ GW.
If I remove the bandwidth command from the Gatekeeper config, the call works 
using g729.


- Scott


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Re: [OSL | CCIE_Voice] Question about route lists and mixing MGCP +H323 route groups

[Cristi Radescu]

Only document I found on this is here:

http://www.ciscotaccc.com/kaidara-advisor/voice/showcase?case=K20653428




From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Michael Ciarfello
Sent: Friday, July 10, 2009 10:48 PM
To: Nara Shikamaru
Cc: OSL Group; GRAFL Philipp
Subject: Re: [OSL | CCIE_Voice] Question about route lists and mixing MGCP 
+H323 route groups

Did you wait enough time for the h323 call to fail before it kicked over to the 
MGCP gateway entry?  I remember it taking a while.  I can try it later tonight. 
 Also, what do your trace files indicate is happening?

From: Nara Shikamaru [mailto:shikam...@kagadis.com]
Sent: Friday, July 10, 2009 2:27 PM
To: Michael Ciarfello
Cc: GRAFL Philipp; OSL Group
Subject: Re: [OSL | CCIE_Voice] Question about route lists and mixing MGCP 
+H323 route groups

The detail for this setting states that it's for ICT calls, so I don't know if 
it will work for this particular issue.  I've worked through the issue by 
adding voip dial peer pointed at an alternate gateway with lower preferences 
that are used in case the pots dial peers fail.  Of course, the side effect is 
that I have a ton of dial peers now.  Maybe Vik or Mark have an opinion?

It seems to me that this would be a problem for both H323 and SIP gateways.
On Fri, Jul 10, 2009 at 10:57 AM, Michael Ciarfello 
mailto:mciarfe...@iplogic.com>> wrote:

You bring up an interesting question that would be good to get down solid.  I 
haven't researched it FULLY yet, but try this as an answer.  I'm sure someone 
has a more exacting answer or knows where the answer is.



I think because the call could not go out the H323 connection (for whatever 
reason), the error returned will be unallocated number and cause us to go to 
the route list member.  You should be able to look at a trace file to see what 
happens as your H323 call fails and the error that returns.  You should also 
see it try the next member in the route list (MGCP this time.)



From: 
ccie_voice-boun...@onlinestudylist.com
 
[mailto:ccie_voice-boun...@onlinestudylist.com]
 On Behalf Of Nara Shikamaru
Sent: Friday, July 10, 2009 12:56 PM
To: GRAFL Philipp
Cc: OSL Group
Subject: Re: [OSL | CCIE_Voice] Question about route lists and mixing MGCP 
+H323 route groups



Sorry, I have no idea how this flag is related to my original question.  Can 
you explain?





Stop Routing on Unallocated Number Flag: Error! Filename not specified.


This parameter determines routing behavior for intercluster trunk calls to an 
unallocated number. An unallocated number represents a dialed directory number 
that does not exist in a Cisco cluster. Valid values specify True or False. 
When the parameter is set to True and a call that is being routed to a remote 
Cisco cluster through a route list is released by a remote Cisco CallManager 
because of the unallocated number, a local Cisco CallManager will stop routing 
the call to a next device in the route list. When the parameter is set to 
False, the local Cisco CallManager will route the call to the next device.




On Fri, Jul 10, 2009 at 1:10 AM, GRAFL Philipp 
mailto:philipp.gr...@nextiraone.at>> wrote:

Try in CCM-Service parameter: Stop routing on unallocated number flag = false.







Von: 
ccie_voice-boun...@onlinestudylist.com
 
[mailto:ccie_voice-boun...@onlinestudylist.com]
 Im Auftrag von Nara Shikamaru
Gesendet: Donnerstag, 09. Juli 2009 18:08
An: OSL Group
Betreff: [OSL | CCIE_Voice] Question about route lists and mixing MGCP +H323 
route groups



I'm working on a situation whereby a route list has two route groups, each 
contain end points with different call controls.  Primary is H323, secondary is 
MGCP.  I've run into an unfortunate shortcoming that I hope I'm wrong abount.  
In order to get calls routed to the second route group in the list, it LOOKS 
like the H323 end point has to be completely inaccessible.  In other words, 
it's not enough that the PRI is down; the whole device needs to be unreachable. 
 I suspect this is due to the fact that on an H323 gateway the PRI is not being 
backhauled to CUCM so the cluster has no way of knowing that the circuit is 
down, so it continues to be engaged.  I would LIKE to set up a situation 
whereby if the PRI is not functional, the second route group in the list is 
used.  Does this mean that I would have to configure the first gateway as MGCP 
instead?

--
-Shikamaru



--
-Shikamaru



--
-Shikamaru
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Re: [OSL | CCIE_Voice] IPMA Questions

[Cristi Radescu]

Exaclty. Thanks Jose.
Sorry  it's my way of naming. I am doing IPMA manually and PT-IPMA is 
manager's line partition.



From: Jose Gregorio Linero (jlinero) [mailto:jlin...@cisco.com]
Sent: Tuesday, June 30, 2009 5:27 PM
To: Cristi Radescu; OSL Group
Subject: RE: [OSL | CCIE_Voice] IPMA Questions

Hi:

Just a little correction IPMA CSS should contain manager and internal 
partitions.

Regards,

Jose

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cristi Radescu
Sent: Martes, 30 de Junio de 2009 09:04 a.m.
To: OSL Group
Subject: Re: [OSL | CCIE_Voice] IPMA Questions

Hi Scott,

-  IPMA RP will need CSS-IPMA. CSS-IPMA will contain partitions PT-IPMA 
and PT-Internal if you use an internal partition.
-  There are 3 places where you need CSS-IPMA: CTI RP, Proxy Line  and 
MWI Ports;
-  Your IPMA must work after restarting only Tomcat Service.

Hope this helps,
Cristi


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Scott ODonnell
Sent: Tuesday, June 30, 2009 3:05 PM
To: OSL Group
Subject: [OSL | CCIE_Voice] IPMA Questions

I'm not sure if my lab setup is flaky or IPMA sucks (well, I'm pretty sure IPMA 
sucks anyway )

I've gone through Vik's vlecture and I follow the same config process.

My problem is I consistently end up getting fast busy when I call into the IPMA 
RP/extension.

A couple of questions

- I was surprised to see that the IPMA RP doesn't need a CSS assigned to it . 
I'm wondering why?

- I've tried to resolve the above mentioned problem by restarting the processes 
"Call Manager", "CTI Manager" and Tomcat but it doesn't help. The only way to 
get things working is to reboot the pub & sub.
Is there another process to consider restarting?

Any opinions are greatly appreciated.
- Scott

Re: [OSL | CCIE_Voice] IPMA Questions

[Cristi Radescu]

Hi Scott,

-  IPMA RP will need CSS-IPMA. CSS-IPMA will contain partitions PT-IPMA 
and PT-Internal if you use an internal partition.
-  There are 3 places where you need CSS-IPMA: CTI RP, Proxy Line  and 
MWI Ports;
-  Your IPMA must work after restarting only Tomcat Service.

Hope this helps,
Cristi


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Scott ODonnell
Sent: Tuesday, June 30, 2009 3:05 PM
To: OSL Group
Subject: [OSL | CCIE_Voice] IPMA Questions

I'm not sure if my lab setup is flaky or IPMA sucks (well, I'm pretty sure IPMA 
sucks anyway )

I've gone through Vik's vlecture and I follow the same config process.

My problem is I consistently end up getting fast busy when I call into the IPMA 
RP/extension.

A couple of questions

- I was surprised to see that the IPMA RP doesn't need a CSS assigned to it . 
I'm wondering why?

- I've tried to resolve the above mentioned problem by restarting the processes 
"Call Manager", "CTI Manager" and Tomcat but it doesn't help. The only way to 
get things working is to reboot the pub & sub.
Is there another process to consider restarting?

Any opinions are greatly appreciated.
- Scott

Re: [OSL | CCIE_Voice] San Jose start time?

[Cristi Radescu]

I'll have mine same building same hour as Michael:

LAB LOCATION and START TIME:
Cisco Systems
150 West Tasman Drive
Bldg C, 1st floor
San Jose, CA  USA  95134

Hours:  8:15am to 5:10pm



From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mike Thompson
Sent: Tuesday, June 30, 2009 7:46 AM
To: 'Michael Ciarfello'; 'Cliff McGlamry'; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] San Jose start time?

If memory serves, SJ is a little after 8 (8:15 or 8:30, but be early).  RTP is 
earlier, I think a 7:30 start (get there about 7am).

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Michael Ciarfello
Sent: Monday, June 29, 2009 11:33 PM
To: Cliff McGlamry; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] San Jose start time?

Mine in February was 8:15am.
I think I got there at about 7:30 and sat in the car listening to music.
I think your confirmation e-mail has the time.  Here's a snippet from mine.  
Title of the e-mail was "Online Lab Scheduling Confirmation"


LAB LOCATION and START TIME:

Cisco Systems

150 West Tasman Drive

Bldg C, 1st floor

San Jose, CA  USA  95134



Hours:  8:15am to 5:10pm



GENERAL EXAM INFORMATION:

1. Please arrive on time for the exam.  If you arrive late, you will be 
expected to finish with the group.  If you arrive more than 2 hours late, you 
will not be allowed to start.  The hands-on lab exam runs for 8 hours with 
additional time scheduled for the initial exam briefing and a lunch break.



From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cliff McGlamry
Sent: Monday, June 29, 2009 11:29 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] San Jose start time?

I'm looking at Cisco's web site for start time and directions for the lab exam 
in San Jose.  I thought they started at 7 AM, but the web site says 8:15 AM 
start time.

I'm a little confused.  Anyone been over there lately that can comment on this?

Cliff

Re: [OSL | CCIE_Voice] CME web access disable

[Cristi Radescu]

telephony-service
  service phone webAccess 1


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ahmed Elnagar
Sent: Wednesday, June 24, 2009 10:38 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] CME web access disable

Hello all;

Anyway know a way to disable phone web access for CME phones?


check out the rest of the Windows Live(tm). More than mail-Windows Live(tm) 
goes way beyond your inbox. More than 
messages

[OSL | CCIE_Voice] V2: Strange message from Unity/CUE -"Wait while I transfer your call"

[Cristi Radescu]

Hi all,

Last night I had for the first time this strange message from CUE and Unity: 
"Wait while I transfer your call".
This message came after I was listening messages from mailbox.
The call flow is:
-  I call CUE/Unity pilot --> I am listening all the messages --> I 
delete all messages --> I am hearing "You have no new messages" and immediately 
I have this message "wait while I transfer your call" followed by busy tone.
I was on a remote rack and I didn't have enough time to investigate this.
Anybody faced this problem before? What could be a possible cause for 
this(especially on CUE side)?

Thanks,
Cristi

Re: [OSL | CCIE_Voice] Integrating Cisco CallManager and Cisco SRST to Use Cisco SRST As a Multicast MOH Resource

[Cristi Radescu]

"moh Virgin-on-hold.wav"

You should have "virgin-on-hold.au".
File format is not correct.



-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Azeem ahamed
Sent: Thursday, June 18, 2009 2:24 PM
To: Michael Ciarfello
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Integrating Cisco CallManager and Cisco SRST to 
Use Cisco SRST As a Multicast MOH Resource

Hi all

Well i tried my second implementation of MOH on multicast and this
time i didnt enable multicast on the router and its not working

The file is working as i have tested it on unity express which
requires the same format and it works also with multile formats

But when i do a sh ephone summ

i see this Skinny Music on hold Status
No MOH file loaded

I have the files on the router flash. I deleted it multiple times and
put it back again still not working

I also made the loopback address which again is on a different subnet
than ccm and VG and added on the route

CCM:- 10.0.12.X, GW:- 10.0.19.X, Loopback:- 10.0.0.1

call-manager-fallback
moh Virgin-on-hold.wav
multicast moh 239.1.1.1 port 16384 route 10.0.19.10 10.0.0.1

Also the time when i take multicasting off from Media resource group
settings it works perfectly.

i also have the HOP count set to 1 on the MOH server..

So what do u think seems to be the issue


On Thu, Jun 18, 2009 at 6:49 AM, Michael
Ciarfello wrote:
> Just wanted to re-confirm for my notes that multicast Moh from flash can
> only use G.711, so the multicast address should only be a x.x.x.1 (for first
> source.)  That the Multicast Moh server on CCM should be in a G711 only
> region, etc.
>
> Multicast Moh from CCM to remote site can be g729 and use 239.1.1.3 as in
> the example below.
>
> 
> From: kapil atrish [nice_cha...@yahoo.com]
> Sent: Friday, June 12, 2009 4:40 AM
> To: Michael Ciarfello; Azeem ahamed
> Subject: Re: [OSL | CCIE_Voice] Integrating Cisco CallManager and Cisco SRST
> to Use Cisco SRST As a Multicast MOH Resource
>
> To Play the MOH from flash, you won't need Mcast enabled anywhere, not even
> on the local router.
>
> I would check what codec I am trying MOH to be played in, if G729, u've to
> select G729 in IPVMS Service parameters and set the base address to
> 239.1.1.3 in CCM and in SRST Router.
>
> If this router is going to serve as SRST gateway, the file has to be G711
> only.
>
> Also, to extend the MOH to PSTN, you've to have loopback interface on the
> router and add that loopback address in the: multicast moh command.
> Recommended is to remove the command completely and entry new one rather
> then trying to modify it on the fly.
>
> multicast moh 239.1.1.1 port 16384 route SUB_IF IP LOOPBACK_IP
>
>
>
>
> --- On Thu, 6/11/09, Azeem ahamed  wrote:
>
> From: Azeem ahamed 
> Subject: Re: [OSL | CCIE_Voice] Integrating Cisco CallManager and Cisco SRST
> to Use Cisco SRST As a Multicast MOH Resource
> To: "Michael Ciarfello" 
> Cc: "ccie_voice@onlinestudylist.com" 
> Date: Thursday, June 11, 2009, 1:40 PM
>
> Also do try putting in the ip pim sparse mode when you enable the
> Multicasting
>
>
>
> On Thu, Jun 11, 2009 at 7:30 AM, Michael
> Ciarfello wrote:
>> Hi and welcome.
>>
>> Did you upload the MOH file to all the other servers in the cluster?
>> Multicast routing enabled on the remote gateway?
>> You have 239.1.1.1 which implies FIRST music source, G.711.  Are you using
>> G.711 over the WAN?
>> What does show ccm-manager music-on-hold output when the phone is on hold?
>> What does show ip mroute display when on hold?
>>
>> There is a Cisco made document describing this feature.  Probably search
>> for
>> Moh from router flash on the Cisco web site.
>>
>> The remote phone config file should have in it 239.1.1.1 for Moh.  You can
>> get the phone.xml file from the TFTP directory if the appropriate TFTP
>> service parameters are set or since you are a production system I wouldn't
>> mess with them and would sniff the phone bootup packets to read that file
>> from the sniffer.  The phone might be listening to another IP Address for
>> some reason.
>>
>> Multicast MOH from remote site flash is like the phone turning a radio
>> channel.  If it doesn't tune to the proper channel (defined in the config
>> file downloaded to the phone) you don't hear the radio program.
>>
>> 
>> From: ccie_voice-boun...@onlinestudylist.com
>> [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cristobal Priego
>> [cristobalpri...@gmail.com]
>> Sent: Wednesday, June 10, 2009 4:54 PM
>> To: ccie_voice@onlinestudylist.com
>> Subject: [OSL | CCIE_Voice] Integrating Cisco CallManager and Cisco SRST
>> to
>> Use Cisco SRST As a Multicast MOH Resource
>>
>> Hello Experts,
>>
>> This is not related to a lab or a pod,  sometimes i use my customer's
>> networks to study
>>
>> I have centralized deployment, the cluster has 2 servers running CCM 7.x,
>> the MOH Se

Re: [OSL | CCIE_Voice] g729ar8 on dial-peer not showing up

[Cristi Radescu]

On Cisco IOS gateways, the variant to use (G.729 or G.729A) is related to the 
codec complexity configuration on the voice card. It does not show up 
explicitly in the Cisco IOS command line interface (CLI) codec choice. For 
example, the CLI does not show g729ar8 ("a" code) as a codec option. However, 
if the voice-card is defined as medium-complexity, then the g729r8 option is 
the G.729A codec.
http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a00800b6710.shtml




From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Michael Ciarfello
Sent: Thursday, June 18, 2009 8:28 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] g729ar8 on dial-peer not showing up

What would cause the g729ar8 codec not to show up on a voip dial-peer?  Only 
g729r8 is showing up (in that g729 family) and I want more practice on 
understanding codec compatibiltities, etc.  For example G729ar8 is supposed to 
be compatible with g729r8.

voice-card 0 is set to flex mode.  If no answer, I'll try unconfiguring a bunch 
of stuff and switch the complexity to medium then high and see how the list 
changes. But I think it should show up in flex.



Thanks

Re: [OSL | CCIE_Voice] Integrating Cisco CallManager and Cisco SRST to Use Cisco SRST As a Multicast MOH Resource

[Cristi Radescu]

Hi Cristobal,

You don't need to match the two names(in CCM and flash).
Probably you had some problems with file loaded on flash. When you uploaded it 
once again you loaded file in supported format.
You can verify that everything is ok with file with "sh ephone summ".

Cristi




From: Cristobal Priego [mailto:cristobalpri...@gmail.com]
Sent: Tuesday, June 16, 2009 11:13 PM
To: Cristi Radescu
Cc: Michael Ciarfello; Azeem ahamed; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Integrating Cisco CallManager and Cisco SRST to 
Use Cisco SRST As a Multicast MOH Resource

Thank you to all of the people that answered my question. we got this problem 
resolved our problem was that the Name on the MOH audio source in CCM didn't 
match with the file name stored in Flash, once we changed the name on the file 
and uploaded it again to the flash it worked like charm

thanks again
2009/6/16 Cristi Radescu 
mailto:cristian.rade...@crescendo.ro>>
Hi Michael,

You're welcome.
You will need this command:

multicast moh 239.1.1.x port 16384 route VOICE_VLAN_IP LOOPBACK_IP

Basically, when you configure this, the router will start to transmit music to 
multicast ip address 239.1.1.x port 16384 on interfaces VOICE_VLAN and 
LOOPBACK. Loopback is needed there because you will need MoH for the PSTN 
phones. After that you must configure phones to "listen" to that address from 
call-manager-fallback(239.1.1.x, where x depends on codec used).
When a phone is on hold it will listen to the address from it's configuration 
file. You can verify what address CCM configured for the phones by verifying 
configuration files on TFTP Server.
Also, if you did a mistake in "multicast moh" command you must delete it and 
put it again in order to work with the new configurations.

Hope this helps,
Cristi





-Original Message-
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 Michael Ciarfello
Sent: Tuesday, June 16, 2009 7:29 AM
To: Azeem ahamed
Cc: ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com>
Subject: Re: [OSL | CCIE_Voice] Integrating Cisco CallManager and Cisco SRST to 
Use Cisco SRST As a Multicast MOH Resource

Did the original poster get this working?

I did verify multicast routing is not needed on any router.  (Thanks Cristi). 
Also no PIM is needed.

I DID have to reboot the router becasue the multicast packets kept coming from 
the loopback interface on my last configuration even though SRST source-address 
was voice VLAN interface.

Seemed to need the route command in my last configuration.  IP of loopback AND 
IP of voice VLAN. I think MGCP was bound to loopback and SRST was bound to 
voice vlan.  I'll re-try Christ's route meaning, but think if I didn't also 
have loopback music stopped.

Never configured the route command for a customer.  Still working on perfecting 
the configuration and meanings of everything.  I can get it to work, but that's 
not expert enough for me.


From: Azeem ahamed [azeemo...@gmail.com<mailto:azeemo...@gmail.com>]
Sent: Thursday, June 11, 2009 4:10 AM
To: Michael Ciarfello
Cc: Cristobal Priego; 
ccie_voice@onlinestudylist.com<mailto:ccie_voice@onlinestudylist.com>
Subject: Re: [OSL | CCIE_Voice] Integrating Cisco CallManager and Cisco SRST
to Use Cisco SRST As a Multicast MOH Resource

Also do try putting in the ip pim sparse mode when you enable the Multicasting



On Thu, Jun 11, 2009 at 7:30 AM, Michael
Ciarfellomailto:mciarfe...@iplogic.com>> wrote:
> Hi and welcome.
>
> Did you upload the MOH file to all the other servers in the cluster?
> Multicast routing enabled on the remote gateway?
> You have 239.1.1.1 which implies FIRST music source, G.711.  Are you using
> G.711 over the WAN?
> What does show ccm-manager music-on-hold output when the phone is on hold?
> What does show ip mroute display when on hold?
>
> There is a Cisco made document describing this feature.  Probably search for
> Moh from router flash on the Cisco web site.
>
> The remote phone config file should have in it 239.1.1.1 for Moh.  You can
> get the phone.xml file from the TFTP directory if the appropriate TFTP
> service parameters are set or since you are a production system I wouldn't
> mess with them and would sniff the phone bootup packets to read that file
> from the sniffer.  The phone might be listening to another IP Address for
> some reason.
>
> Multicast MOH from remote site flash is like the phone turning a radio
> channel.  If it doesn't tune to the proper channel (defined in the config
> file downloaded to the phone) you don't hear t

Re: [OSL | CCIE_Voice] DSP Calculation

[Cristi Radescu]

Hi all,

I don't find right now that good doc but I'll tell you how you can test this:

1. On a router with PVDM2, when you configure dspfarm profile(let's assume you 
have PVDM2-16):
dspfarm profile 1 transcode
  codec g711u
  maxx session ? <-- you will see 16(assure only that codec is present in 
config)

dspfarm profile 1 transcode
 codec g711u
 codec g729a
 maxx sess ? <-- you will see here: 8 (assure only those codecs are present in 
config)

dspfarm profile 1 transcode
  codec g711u
  codec g729a
  codec g729r
  max sess ? <-- you will see here 6
 After each case you can see DSP sharing with: sh voice dsp group all.

 2. On DSP Calculator choose an ISR (e.q. 2801) and configure only transcoding 
sessions. First thing you will see that you will have 3 colums:
- G.711 a-law to/from u-law;
- G.711 to G.729a;
- G.711 to G.729(b).

To test what I said simply put 6 session in the last column. You will see that 
you need only one DSP PVDM2-16(6x40=240 MIPS). Then put in the same column 7 
sessions. You will see the system add a PVDM2-8.
If you will test G729a, configure 8 sessions in the second column and after 
that 9 sessions. The results will confirm our formula.


Also you must "see" voice termination and transcoding separately.
For example, if you have 1 x PVDM2-16 and you will have 6 E1 timeslots the 
calculations are like following:
-  6 x 15 = 90 MIPS;
-  240 - 90 = 150 MIPS remaining for transcoding.
-  With 150 MIPS we can have a maximum of 5 transcoding sessions to 
G729a or 3 transcoding sessions to G729b(in this case it will remain 30 MIPS 
unused); Try this in DSP calculator: configure 6 timeslots(G711u) and after 
that configure in transcoding section 5/6 transcoding sessions to G729a or 3/4  
transcoding sessions to G729b. The results will confirm formula.

Hth,
Cristi

From: Michael Ciarfello [mailto:mciarfe...@iplogic.com]
Sent: Tuesday, June 16, 2009 5:10 PM
To: Cristi Radescu; Art Sandborgh; ccie list
Subject: RE: [OSL | CCIE_Voice] DSP Calculation

Want to add the "D" channel doesn't count as a signaling channel.  So If you 
have "pri-group timeslots 1-4,24" then only 4 channels are needed.

Also how does one use g729 on an FXO, FXS or PRI "B" channel?

The CCM 7.x SRND, Table 6-2 says:

At 15 MIPS per call:
*G.711 (a-law, mu-law)
*Fax/modem passthrough
*Clear channel

At 30 MIPS per call:
*G.726 (32K, 24K, 16K)
*Fax relay
*G.729
*G.729 (a, b, ab)

At 40 MIPS per call:
*G.728
*G.723.1 (32K, 24K, 16K)
*G.723.1a (5.3K, 6.3K)
*Modem relay

What document did you get the g729's are 40 MIPS per call?  We'll have to 
validate this ourselves if there is a documentation inconsistency (no 
surprise.)  But need to know how to test this.  I think it might be incoming 
PRI channel (or FXO port) TERMINATING to a g729r8 only device will use 30 MIPS. 
 There is no codec on an FXO.

I'll post an updated doc sheet soon after we validate some of these and give 
people a chance to add / update.

From: Cristi Radescu [mailto:cristian.rade...@crescendo.ro]
Sent: Tuesday, June 16, 2009 5:31 AM
To: Michael Ciarfello; Art Sandborgh; ccie list
Subject: RE: [OSL | CCIE_Voice] DSP Calculation

Hi Michael,

Very nice doc. Thanks for that. I'll do some corrections on it from my point of 
view.
Please see below with blue.

Hope this helps,
Cristi


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Michael Ciarfello
Sent: Tuesday, June 16, 2009 7:02 AM
To: Art Sandborgh; ccie list
Subject: Re: [OSL | CCIE_Voice] DSP Calculation

Here are my DSP notes as promised:
Maybe we can finalize this and have IPexpert post it on their web site.

PVDM2-16 is one DSP chip (32=2, 48=3, 64=4)
-  PVDM2-16 can manage 16 voice termination channels
-  has one DSP C5510(new one); DSP C5510 can be shared between 
transcoding and voice termination. Can not be shared if it's used for 
conferencing!
PVDM2-8 is one DSP chip but less processing capacity than DSP on the 16
- PVDM2-8 can manage 8 voice termination channels

PVDM2-16 - Signaling
- 8 calls per DSP (medium complexity codecs)
- 6 calls per DSP (high complexity codecs)
- 240 MIPS in flex mode
  - G711 uses 15 MIPS per call (240 / 15 = 16 calls per DSP)
  - G729a, G729ab  uses 30 MIPS per call (not all g729 variants) => 240 MIPS/30 
= 8 medium complexity calls
   - G729, G729b uses 40 MIPS per call => 240 MIPS/40 = 6 high complexity 
calls(ATT: g729r8 is a high complexity codec!)

PVDM-12
- can manage 12 voice termination channels
- has 3 x DSP C549(the old one); each of these DSPs can manage 4 
calls/transcoder sessions/voice termination channels no matter what codec is 
used;
- one DSP C549 can not be shared(not even between voice termination and 
tr

Re: [OSL | CCIE_Voice] DSP Calculation

[Cristi Radescu]

Hi Michael,

Very nice doc. Thanks for that. I'll do some corrections on it from my point of 
view.
Please see below with blue.

Hope this helps,
Cristi


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Michael Ciarfello
Sent: Tuesday, June 16, 2009 7:02 AM
To: Art Sandborgh; ccie list
Subject: Re: [OSL | CCIE_Voice] DSP Calculation

Here are my DSP notes as promised:
Maybe we can finalize this and have IPexpert post it on their web site.

PVDM2-16 is one DSP chip (32=2, 48=3, 64=4)
-  PVDM2-16 can manage 16 voice termination channels
-  has one DSP C5510(new one); DSP C5510 can be shared between 
transcoding and voice termination. Can not be shared if it's used for 
conferencing!
PVDM2-8 is one DSP chip but less processing capacity than DSP on the 16
- PVDM2-8 can manage 8 voice termination channels

PVDM2-16 - Signaling
- 8 calls per DSP (medium complexity codecs)
- 6 calls per DSP (high complexity codecs)
- 240 MIPS in flex mode
  - G711 uses 15 MIPS per call (240 / 15 = 16 calls per DSP)
  - G729a, G729ab  uses 30 MIPS per call (not all g729 variants) => 240 MIPS/30 
= 8 medium complexity calls
   - G729, G729b uses 40 MIPS per call => 240 MIPS/40 = 6 high complexity 
calls(ATT: g729r8 is a high complexity codec!)

PVDM-12
- can manage 12 voice termination channels
- has 3 x DSP C549(the old one); each of these DSPs can manage 4 
calls/transcoder sessions/voice termination channels no matter what codec is 
used;
- one DSP C549 can not be shared(not even between voice termination and 
transcoding);
- e.g. if you have 5 x timeslots E1/T1 you will have 2 X DSPs blocked 
for voice termination => it remains only one DSP(4 sessions) for transcoding;


MTP
- CCM SW MTP is G711 only (all versions including CCM7.x)
- IOS SW MTP
  - Supports G711 and any G729 variant.  But can choose only one
codec on the dspfarm profile at a time.
  - Need the capacities.
- IOS HW MTP
  - 16 G711 sessions per DSP
  - 6 G729 sessions per DSP

PVDM2-16 - Conferencing
- Each DSP accomodates 8 conference participants
- IOS 12.4(15)T has new capability for 32 participants (needs verification)
- Each DSP supports 8 conferences if G.711 is only configrued codec on the 
dspfarm
  profile.
- Each DSP supports 2 conferences (of 8 participants each) if G.729 is 
CONFIGURED on the dspfarm.
  (even if all participants on all conferences are using G.711.)
- I think you have to turn off GSM to get 8 conferences (needs verification)
- Can't share conference on DSP with xcode or voice signaling.
- Config max-sessions in multiples of 2 or 8 (depending on configured codec).  
Doesn't
  make sense to configure less--wasting resources.
PVDM2-16 - Transcoding
- Can share Transcoding with voice signaling.
- 8 Sessions - G711a/u to G729a/ab (must turn off g729 and g729b to get 8 
sessions) in dspfarm profile.
- 6 sessions - G711a/u to G729 / G729b


Need to verify formula for mixed transcoding and signaling sessions on same 
DSP.  Don't think we can trust max-sessions.

Please feel free to correct or amend this document.

Re: [OSL | CCIE_Voice] Integrating Cisco CallManager and Cisco SRST to Use Cisco SRST As a Multicast MOH Resource

[Cristi Radescu]

Hi Michael,

You're welcome.
You will need this command:

multicast moh 239.1.1.x port 16384 route VOICE_VLAN_IP LOOPBACK_IP

Basically, when you configure this, the router will start to transmit music to 
multicast ip address 239.1.1.x port 16384 on interfaces VOICE_VLAN and 
LOOPBACK. Loopback is needed there because you will need MoH for the PSTN 
phones. After that you must configure phones to "listen" to that address from 
call-manager-fallback(239.1.1.x, where x depends on codec used).
When a phone is on hold it will listen to the address from it's configuration 
file. You can verify what address CCM configured for the phones by verifying 
configuration files on TFTP Server.
Also, if you did a mistake in "multicast moh" command you must delete it and 
put it again in order to work with the new configurations.

Hope this helps,
Cristi





-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Michael Ciarfello
Sent: Tuesday, June 16, 2009 7:29 AM
To: Azeem ahamed
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Integrating Cisco CallManager and Cisco SRST to 
Use Cisco SRST As a Multicast MOH Resource

Did the original poster get this working?

I did verify multicast routing is not needed on any router.  (Thanks Cristi). 
Also no PIM is needed.

I DID have to reboot the router becasue the multicast packets kept coming from 
the loopback interface on my last configuration even though SRST source-address 
was voice VLAN interface.

Seemed to need the route command in my last configuration.  IP of loopback AND 
IP of voice VLAN. I think MGCP was bound to loopback and SRST was bound to 
voice vlan.  I'll re-try Christ's route meaning, but think if I didn't also 
have loopback music stopped.

Never configured the route command for a customer.  Still working on perfecting 
the configuration and meanings of everything.  I can get it to work, but that's 
not expert enough for me.


From: Azeem ahamed [azeemo...@gmail.com]
Sent: Thursday, June 11, 2009 4:10 AM
To: Michael Ciarfello
Cc: Cristobal Priego; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Integrating Cisco CallManager and Cisco SRST
to Use Cisco SRST As a Multicast MOH Resource

Also do try putting in the ip pim sparse mode when you enable the Multicasting



On Thu, Jun 11, 2009 at 7:30 AM, Michael
Ciarfello wrote:
> Hi and welcome.
>
> Did you upload the MOH file to all the other servers in the cluster?
> Multicast routing enabled on the remote gateway?
> You have 239.1.1.1 which implies FIRST music source, G.711.  Are you using
> G.711 over the WAN?
> What does show ccm-manager music-on-hold output when the phone is on hold?
> What does show ip mroute display when on hold?
>
> There is a Cisco made document describing this feature.  Probably search for
> Moh from router flash on the Cisco web site.
>
> The remote phone config file should have in it 239.1.1.1 for Moh.  You can
> get the phone.xml file from the TFTP directory if the appropriate TFTP
> service parameters are set or since you are a production system I wouldn't
> mess with them and would sniff the phone bootup packets to read that file
> from the sniffer.  The phone might be listening to another IP Address for
> some reason.
>
> Multicast MOH from remote site flash is like the phone turning a radio
> channel.  If it doesn't tune to the proper channel (defined in the config
> file downloaded to the phone) you don't hear the radio program.
>
> 
> From: ccie_voice-boun...@onlinestudylist.com
> [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cristobal Priego
> [cristobalpri...@gmail.com]
> Sent: Wednesday, June 10, 2009 4:54 PM
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] Integrating Cisco CallManager and Cisco SRST to
> Use Cisco SRST As a Multicast MOH Resource
>
> Hello Experts,
>
> This is not related to a lab or a pod,  sometimes i use my customer's
> networks to study
>
> I have centralized deployment, the cluster has 2 servers running CCM 7.x,
> the MOH Servers are enabled for multicast, the ip address and port are
> configured, the multicast increment is set to "Ip address"
> On the selected audio source field on the option 1 (which is the MOH file
> that we want to use) the hop count is set to 1
> The MOH audio source has the box check for Allow multicasting.
> I created an MRG that include both serves and is enabled for Multicast, the
> MRGL has the Moh MRG. the MRGL is assigned to the phones and the remote
> gateway. the phones have the user/network audio source selected to option 1
> the remote gateway has the wav file loaded in flash.
> the call-manager-fallback has the
> ccm-manager music-on-hold
> moh 
> multicast moh 239.1.1.1 port 16384 route  dedicated for voice>
>
> If I put a call on hold on the remote branch, the phone is not playing the
> music from the

Re: [OSL | CCIE_Voice] Integrating Cisco CallManager and Cisco SRST to Use Cisco SRST As a Multicast MOH Resource

[Cristi Radescu]

Hi Cristobal,

Just a starting point in your debug:
-  enable multicasting only on a server not both; if you will have both 
you must configure different starting IP addresses on each server and this will 
not work in Moh from flash scenario;
-  239.1.1.1 is for audio source 1, codec g711u; make sure you are 
using this one for MoH. If you have g729 between regions you should have there 
239.1.1.3 and G729 enabled for MoH in service parameters;
-  also you can use "debug ephone moh" for debugging on router.
-  Also in SRST condition, ip phones will receive only beeps. PSTN 
phones will receive music.

Hope this helps,
Cristian



From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cristobal Priego
Sent: Wednesday, June 10, 2009 11:54 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Integrating Cisco CallManager and Cisco SRST to Use 
Cisco SRST As a Multicast MOH Resource

Hello Experts,

This is not related to a lab or a pod,  sometimes i use my customer's networks 
to study

I have centralized deployment, the cluster has 2 servers running CCM 7.x, the 
MOH Servers are enabled for multicast, the ip address and port are configured, 
the multicast increment is set to "Ip address"
On the selected audio source field on the option 1 (which is the MOH file that 
we want to use) the hop count is set to 1
The MOH audio source has the box check for Allow multicasting.
I created an MRG that include both serves and is enabled for Multicast, the 
MRGL has the Moh MRG. the MRGL is assigned to the phones and the remote 
gateway. the phones have the user/network audio source selected to option 1
the remote gateway has the wav file loaded in flash.
the call-manager-fallback has the
ccm-manager music-on-hold
moh 
multicast moh 239.1.1.1 port 16384 route 

If I put a call on hold on the remote branch, the phone is not playing the 
music from the local gateway. i get beeps
either internal or external calls. what am i missing?

If i place a call internally i get the beeps, however in RTMT i can see that a 
multicast request becomes active

If i issue a show ccm-manager music-on-hold i don't see anything

your help is greatly appreciated

Re: [OSL | CCIE_Voice] GDM configuration with notification on 2phones

[Cristi Radescu]

No. First thing with your config, on second buttons on phones you will have 
number "3005".
Also envelope don't work with secondary numbers.
Try it. You will see.


From: wilfred d'souza [mailto:wilfred_the...@yahoo.co.in]
Sent: Wednesday, June 10, 2009 9:57 AM
To: ccie_voice@onlinestudylist.com; Cristi Radescu
Subject: RE: [OSL | CCIE_Voice] GDM configuration with notification on 2phones

Guys,

How about this...
Phone 1
Line1 : 3001
Line2 : 3101

Phone 2
Line1 : 3002
Line2 : 3102

If GDM extn is 3005, then

ephone-dn 3
number 3005 secondary 3101

ephone-dn 4
number 3005 secondary 3102

ephone 1
button 1:1 2m3

ephone 2
button 1:2 2m4

I guess this would work correct?

Thanks,
Wilfred



--- On Wed, 10/6/09, wilfred d'souza  wrote:

From: wilfred d'souza 
Subject: RE: [OSL | CCIE_Voice] GDM configuration with notification on 2phones
To: "ccie_voice@onlinestudylist.com" , "Cristi 
Radescu" 
Date: Wednesday, 10 June, 2009, 2:45 PM
ooh yes... my bad...



--- On Wed, 10/6/09, Cristi Radescu  wrote:

From: Cristi Radescu 
Subject: RE: [OSL | CCIE_Voice] GDM configuration with notification on 2phones
To: "wilfred d'souza" , 
"ccie_voice@onlinestudylist.com" 
Date: Wednesday, 10 June, 2009, 2:25 PM
Wilfred, you can not configure 2 extensions after "m".
You can monitor only one extension.


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of wilfred d'souza
Sent: Wednesday, June 10, 2009 6:43 AM
To: Michael Ciarfello; Cliff McGlamry ; ccie_voice@onlinestudylist.com ; Cristi 
Radescu
Subject: Re: [OSL | CCIE_Voice] GDM configuration with notification on 2phones

Guys,

How about monitor? For example
Phone 1
Line1 : 3001
Line2 : 3101

Phone 2
Line1 : 3002
Line2 : 3102

If GDM extn is 3005, then

ephone 1
button 1:1 2m5,3

ephone 2
button 1:2  2m5,4

Lemme know if this works... I dont have it setup right now else would have 
tested it but if i remember it I think I had accomplished this

Thanks,
Wilfred


--- On Wed, 10/6/09, Cristi Radescu  wrote:

From: Cristi Radescu 
Subject: Re: [OSL | CCIE_Voice] GDM configuration with notification on 2phones
To: "Michael Ciarfello" , " Cliff McGlamry " 
, " ccie_voice@onlinestudylist.com " < 
ccie_voice@onlinestudylist.com >
Date: Wednesday, 10 June, 2009, 11:05 AM
I will say "NO" too.
I tested many scenarios that came in my mind without a positive result.
If you cofigure overlay as Cliff suggested, you must put it as first extension 
after "o" in order for envelope to show on both phones. But if you will do 
this, you will modify the look of the phone(second line will show this 
extension).

____________
From: Michael Ciarfello [mailto:mciarfe...@iplogic.com]
Sent: Wednesday, June 10, 2009 5:45 AM
To: Cliff McGlamry ; Cristi Radescu; ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] GDM configuration with notification on 2phones

Did this ever get solved?  Am I correct in saying the answer so far is NO?


From: ccie_voice-boun...@onlinestudylist.com 
[ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cliff McGlamry 
[cl...@mcglamry.net]
Sent: Thursday, May 28, 2009 3:47 PM
To: Cristi Radescu; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] GDM configuration with notification on 2phones
Actually, you might be able to do it if it is on CME.

Assign button 2 as an overlay, and put the mailbox number DN on the overlay 
ephone-dn.  It should be hard forwarded so it will never ring, but I bet that 
would make the envelope appear the way being discussed.

From: Cristi Radescu
Sent: Thursday, May 28, 2009 5:22 AM
To: 'ccie_voice@onlinestudylist.com'
Subject: Re: [OSL | CCIE_Voice] GDM configuration with notification on 2phones


I think this is not possible. With "secondary number" or "overlay" it will not 
work.


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of David Corbeil
Sent: 27 May, 2009 8:42 PM
To: 'ccie_voice@onlinestudylist.com'
Subject: [OSL | CCIE_Voice] GDM configuration with notification on 2 phones

Hi,

I want to know if it's possible to have the voicemail letter on the second line 
of 2 phones without changing the phone facing.

Example:

Phone 1
Line1 : 3001
Line2 : 3101

Phone 2
Line1 : 3002
Line2 : 3102

Each line need to be access to GDM mailbox, and when a message is left on GDM I 
need to have VM Letter on both Line2 Phone.
Can't have line 3
Can't modify the look of the phone

It's possible? If yes, how ?

Thanks

David Corbeil
Consultant en technologie | Technology Consultant
Tel. 514-798-4206 | Fax. 514-748-5333
Membre de l'équipe TELUS



__

Re: [OSL | CCIE_Voice] GDM configuration with notification on 2phones

[Cristi Radescu]

Wilfred, you can not configure 2 extensions after "m".
You can monitor only one extension.


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of wilfred d'souza
Sent: Wednesday, June 10, 2009 6:43 AM
To: Michael Ciarfello; Cliff McGlamry; ccie_voice@onlinestudylist.com; Cristi 
Radescu
Subject: Re: [OSL | CCIE_Voice] GDM configuration with notification on 2phones

Guys,

How about monitor? For example
Phone 1
Line1 : 3001
Line2 : 3101

Phone 2
Line1 : 3002
Line2 : 3102

If GDM extn is 3005, then

ephone 1
button 1:1 2m5,3

ephone 2
button 1:2  2m5,4

Lemme know if this works... I dont have it setup right now else would have 
tested it but if i remember it I think I had accomplished this

Thanks,
Wilfred


--- On Wed, 10/6/09, Cristi Radescu  wrote:

From: Cristi Radescu 
Subject: Re: [OSL | CCIE_Voice] GDM configuration with notification on 2phones
To: "Michael Ciarfello" , "Cliff McGlamry" 
, "ccie_voice@onlinestudylist.com" 

Date: Wednesday, 10 June, 2009, 11:05 AM
I will say "NO" too.
I tested many scenarios that came in my mind without a positive result.
If you cofigure overlay as Cliff suggested, you must put it as first extension 
after "o" in order for envelope to show on both phones. But if you will do 
this, you will modify the look of the phone(second line will show this 
extension).


From: Michael Ciarfello [mailto:mciarfe...@iplogic.com]
Sent: Wednesday, June 10, 2009 5:45 AM
To: Cliff McGlamry; Cristi Radescu; ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] GDM configuration with notification on 2phones

Did this ever get solved?  Am I correct in saying the answer so far is NO?


From: ccie_voice-boun...@onlinestudylist.com 
[ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cliff McGlamry 
[cl...@mcglamry.net]
Sent: Thursday, May 28, 2009 3:47 PM
To: Cristi Radescu; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] GDM configuration with notification on 2phones
Actually, you might be able to do it if it is on CME.

Assign button 2 as an overlay, and put the mailbox number DN on the overlay 
ephone-dn.  It should be hard forwarded so it will never ring, but I bet that 
would make the envelope appear the way being discussed.

From: Cristi Radescu
Sent: Thursday, May 28, 2009 5:22 AM
To: 
'ccie_voice@onlinestudylist.com'
Subject: Re: [OSL | CCIE_Voice] GDM configuration with notification on 2phones


I think this is not possible. With "secondary number" or "overlay" it will not 
work.


From: 
ccie_voice-boun...@onlinestudylist.com
 [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of David Corbeil
Sent: 27 May, 2009 8:42 PM
To: 
'ccie_voice@onlinestudylist.com'
Subject: [OSL | CCIE_Voice] GDM configuration with notification on 2 phones

Hi,

I want to know if it's possible to have the voicemail letter on the second line 
of 2 phones without changing the phone facing.

Example:

Phone 1
Line1 : 3001
Line2 : 3101

Phone 2
Line1 : 3002
Line2 : 3102

Each line need to be access to GDM mailbox, and when a message is left on GDM I 
need to have VM Letter on both Line2 Phone.
Can't have line 3
Can't modify the look of the phone

It's possible? If yes, how ?

Thanks

David Corbeil
Consultant en technologie | Technology Consultant
Tel. 514-798-4206 | Fax. 514-748-5333
Membre de l'équipe TELUS




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Re: [OSL | CCIE_Voice] GDM configuration with notification on 2phones

[Cristi Radescu]

I will say "NO" too.
I tested many scenarios that came in my mind without a positive result.
If you cofigure overlay as Cliff suggested, you must put it as first extension 
after "o" in order for envelope to show on both phones. But if you will do 
this, you will modify the look of the phone(second line will show this 
extension).


From: Michael Ciarfello [mailto:mciarfe...@iplogic.com]
Sent: Wednesday, June 10, 2009 5:45 AM
To: Cliff McGlamry; Cristi Radescu; ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] GDM configuration with notification on 2phones

Did this ever get solved?  Am I correct in saying the answer so far is NO?


From: ccie_voice-boun...@onlinestudylist.com 
[ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cliff McGlamry 
[cl...@mcglamry.net]
Sent: Thursday, May 28, 2009 3:47 PM
To: Cristi Radescu; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] GDM configuration with notification on 2phones
Actually, you might be able to do it if it is on CME.

Assign button 2 as an overlay, and put the mailbox number DN on the overlay 
ephone-dn.  It should be hard forwarded so it will never ring, but I bet that 
would make the envelope appear the way being discussed.

From: Cristi Radescu<mailto:cristian.rade...@crescendo.ro>
Sent: Thursday, May 28, 2009 5:22 AM
To: 'ccie_voice@onlinestudylist.com'
Subject: Re: [OSL | CCIE_Voice] GDM configuration with notification on 2phones


I think this is not possible. With "secondary number" or "overlay" it will not 
work.


From: 
ccie_voice-boun...@onlinestudylist.com<mailto:ccie_voice-boun...@onlinestudylist.com>
 [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of David Corbeil
Sent: 27 May, 2009 8:42 PM
To: 'ccie_voice@onlinestudylist.com'
Subject: [OSL | CCIE_Voice] GDM configuration with notification on 2 phones

Hi,

I want to know if it's possible to have the voicemail letter on the second line 
of 2 phones without changing the phone facing.

Example:

Phone 1
Line1 : 3001
Line2 : 3101

Phone 2
Line1 : 3002
Line2 : 3102

Each line need to be access to GDM mailbox, and when a message is left on GDM I 
need to have VM Letter on both Line2 Phone.
Can't have line 3
Can't modify the look of the phone

It's possible? If yes, how ?

Thanks

David Corbeil
Consultant en technologie | Technology Consultant
Tel. 514-798-4206 | Fax. 514-748-5333
Membre de l'équipe TELUS

Re: [OSL | CCIE_Voice] VPIM Broadcast CUE-->UNITY

[Cristi Radescu]

Hi Art,

You was right. I found the message in "future broadcasts". I shall listen to 
that lady. I allways ignored her :)
After I am loging to Broadcast Administrator I am presssing "3" and the lady is 
telling me "the system has x future broadcast messages".
I tested this today in my personal lab. I will test it once again, tommorow, on 
ProctoLabs Rack.
Thank you so much. If I could rate you I will give 5 stars for this answer. 
This is the normal setup(HQ in PST and CME in GMT +8 timezone) but yestarday 
was the first time when I set timezones correctly:)

Many thanks once again,
Cristian

P.S: I will let you know about my tests on same Rack.


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Art Sandborgh
Sent: Monday, June 08, 2009 10:19 PM
To: ccie list
Subject: Re: [OSL | CCIE_Voice] VPIM Broadcast CUE-->UNITY


Christian,



I have seen this symptom before, but I do not know if you are experiencing the 
same cause.



I fought for hours trying to figure out why the messages did not arrive, when 
in reality they had

(but had not been distributed to the mailboxes yet). If your NTP for the 852 
site is set to an earlier timezone

( say +8) and you send the message to a later timezone (say -5) marked 
"immediate" it gets stored in the "future broadcasts".

The only way I know of to check is to dial into the Broadcast Administrator and 
check for pending messages (I think that is the

menu selection.  You should find it there if you are experiencing what I did.



Good Luck!



Art



Message: 5

Date: Mon, 8 Jun 2009 14:57:27 +0300

From: Cristi Radescu 

Subject: [OSL | CCIE_Voice] VPIM Broadcast CUE-->UNITY

To: "ccie_voice@onlinestudylist.com" 

Message-ID:

<21c0dae06d627846bbfeebed4c4da65622595b9...@cre-exch.csc.crescendo.ro>

Content-Type: text/plain; charset="us-ascii"



Hi all,



Today I had a strange problem on a ProctorLabs rack.

I couldn't nail VPIM Broadcast from CUE to Unity. I did this many times before 
but today it didn't work.

Normal VPIM message is working perfect. Please see below related config on CUE:



network location id "104"

 email domain voip.lab

 name "UNITY-LAB"

 voicemail broadcast vpim-id 

 end location



network location id "852"

 email domain cciev.com

 name "cue"

 voicemail broadcast vpim-id 

 end location



network local location id 852



In Unity I had normal stuff for VPIM configured. Normal VPIM works in both 
directions. Broadcast VPIM from Unity to CUE works.

I have ping in unity-lab.voip.lab from CUE.



Debug from CUE looks ok:



961 06/08 15:16:52.293 netw vpim 3 VPIM: To: <1...@voip.lab>

8961 06/08 15:16:52.297 netw vpim 3 VPIM: From: cme<3...@cciev.com>

8961 06/08 15:16:52.303 netw vpim 3 VPIM: Date: Mon, 08 Jun 2009 15:16:51 +0100 
(BST)

8961 06/08 15:16:52.304 netw vpim 3 VPIM: MIME-Version: 1.0 (Voice 2.0)

8961 06/08 15:16:52.305 netw vpim 3 VPIM: Content-Type: 
Multipart/Voice-Message; Version=2.0;

8961 06/08 15:16:52.306 netw vpim 3 VPIM:   
Boundary="==VpimMsg==1244470612233"

8961 06/08 15:16:52.307 netw vpim 3 VPIM: Content-Transfer-Encoding: 7bit

8961 06/08 15:16:52.308 netw vpim 3 VPIM: Message-ID: 


8961 06/08 15:16:52.311 netw vpim 3 VPIM: Subject: Broadcast Message from CUE 
Location 852

8961 06/08 15:16:52.311 netw vpim 3 VPIM: X-CISCO-SBM-ID: 
FHK0849F1CY-AIM-FOC11200C37-1244468200983-NBCM

8961 06/08 15:16:52.317 netw vpim 3 VPIM: X-CISCO-SBM-START-TIME: Mon, 08 Jun 
2009 15:16:51 +0100 (BST)

8961 06/08 15:16:52.324 netw vpim 3 VPIM: X-CISCO-SBM-END-TIME: Wed, 08 Jul 
2009 15:16:44 +0100 (BST)

8961 06/08 15:16:52.325 netw vpim 3 VPIM: X-CISCO-SBM-CUSTOM1: 
61EF9907D84E295ADB9E2F9951F79543

8961 06/08 15:16:52.326 netw vpim 3 VPIM:

8961 06/08 15:16:52.327 netw vpim 3 VPIM: --==VpimMsg==1244470612233

8961 06/08 15:16:52.328 netw vpim 3 VPIM: Content-Type: Audio/32KADPCM

8961 06/08 15:16:52.329 netw vpim 3 VPIM: Content-Transfer-Encoding: Base64

8961 06/08 15:16:52.331 netw vpim 3 VPIM: Content-Description: VPIM Message

8961 06/08 15:16:52.331 netw vpim 3 VPIM: Content-Disposition: inline; 
voice=Voice-Message

8961 06/08 15:16:52.332 netw vpim 3 VPIM: Content-ID: 
FHK0849F1CY-AIM-FOC11200C37-1244468200983-NBCM

8961 06/08 15:16:52.333 netw vpim 3 VPIM:

...



Anyone faced it before? I changed "broadcast message verify interval" at 1 
minute, I waited 10 minutes probably but ... nothing.



Any idea would be highly appreciated,

Cristian



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--

[OSL | CCIE_Voice] VPIM Broadcast CUE-->UNITY

[Cristi Radescu]

Hi all,

Today I had a strange problem on a ProctorLabs rack.
I couldn't nail VPIM Broadcast from CUE to Unity. I did this many times before 
but today it didn't work.
Normal VPIM message is working perfect. Please see below related config on CUE:

network location id "104"
 email domain voip.lab
 name "UNITY-LAB"
 voicemail broadcast vpim-id 
 end location

network location id "852"
 email domain cciev.com
 name "cue"
 voicemail broadcast vpim-id 
 end location

network local location id 852

In Unity I had normal stuff for VPIM configured. Normal VPIM works in both 
directions. Broadcast VPIM from Unity to CUE works.
I have ping in unity-lab.voip.lab from CUE.

Debug from CUE looks ok:

961 06/08 15:16:52.293 netw vpim 3 VPIM: To: <1...@voip.lab>
8961 06/08 15:16:52.297 netw vpim 3 VPIM: From: cme<3...@cciev.com>
8961 06/08 15:16:52.303 netw vpim 3 VPIM: Date: Mon, 08 Jun 2009 15:16:51 +0100 
(BST)
8961 06/08 15:16:52.304 netw vpim 3 VPIM: MIME-Version: 1.0 (Voice 2.0)
8961 06/08 15:16:52.305 netw vpim 3 VPIM: Content-Type: 
Multipart/Voice-Message; Version=2.0;
8961 06/08 15:16:52.306 netw vpim 3 VPIM:   
Boundary="==VpimMsg==1244470612233"
8961 06/08 15:16:52.307 netw vpim 3 VPIM: Content-Transfer-Encoding: 7bit
8961 06/08 15:16:52.308 netw vpim 3 VPIM: Message-ID: 

8961 06/08 15:16:52.311 netw vpim 3 VPIM: Subject: Broadcast Message from CUE 
Location 852
8961 06/08 15:16:52.311 netw vpim 3 VPIM: X-CISCO-SBM-ID: 
FHK0849F1CY-AIM-FOC11200C37-1244468200983-NBCM
8961 06/08 15:16:52.317 netw vpim 3 VPIM: X-CISCO-SBM-START-TIME: Mon, 08 Jun 
2009 15:16:51 +0100 (BST)
8961 06/08 15:16:52.324 netw vpim 3 VPIM: X-CISCO-SBM-END-TIME: Wed, 08 Jul 
2009 15:16:44 +0100 (BST)
8961 06/08 15:16:52.325 netw vpim 3 VPIM: X-CISCO-SBM-CUSTOM1: 
61EF9907D84E295ADB9E2F9951F79543
8961 06/08 15:16:52.326 netw vpim 3 VPIM:
8961 06/08 15:16:52.327 netw vpim 3 VPIM: --==VpimMsg==1244470612233
8961 06/08 15:16:52.328 netw vpim 3 VPIM: Content-Type: Audio/32KADPCM
8961 06/08 15:16:52.329 netw vpim 3 VPIM: Content-Transfer-Encoding: Base64
8961 06/08 15:16:52.331 netw vpim 3 VPIM: Content-Description: VPIM Message
8961 06/08 15:16:52.331 netw vpim 3 VPIM: Content-Disposition: inline; 
voice=Voice-Message
8961 06/08 15:16:52.332 netw vpim 3 VPIM: Content-ID: 
FHK0849F1CY-AIM-FOC11200C37-1244468200983-NBCM
8961 06/08 15:16:52.333 netw vpim 3 VPIM:
...

Anyone faced it before? I changed "broadcast message verify interval" at 1 
minute, I waited 10 minutes probably but ... nothing.

Any idea would be highly appreciated,
Cristian

Re: [OSL | CCIE_Voice] GDM configuration with notification on 2 phones

[Cristi Radescu]

Hi Sergio,

Are you sure? I was sure MWI doesn't work with overlay lines.
I tested it once again right now, after your answer, and it seems to me that it 
doesn't work.
If you have 2 ephone-dn (e.g 5,6 in your example) it will send MWI/Envelope 
only to first defined extension(e.g.5);
If you have only one ephone-dn and you will configure it on two phones with 
overlay, in that case MWI will show up. But, you can not have two labels for 
the second line in that case.

Cristi




From: Sergio Polizer [mailto:spoli...@hotmail.com]
Sent: 28 May, 2009 4:55 PM
To: cyrus@gmail.com
Cc: Cristi Radescu; ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] GDM configuration with notification on 2 phones

Yeah! Its really depend on the requirements.

In this case for e.g. the System Message will be override by "2:Forward to VM"



> Date: Thu, 28 May 2009 23:43:17 +1000
> Subject: Re: [OSL | CCIE_Voice] GDM configuration with notification on 2 
> phones
> From: cyrus@gmail.com
> To: spoli...@hotmail.com
> CC: cristian.rade...@crescendo.ro; ccie_voice@onlinestudylist.com
>
> I guess it should be ,
>
> ephone-dn 6
> number 3100 (GDN)
> label Support 2
> call-forward all VM
>
> Sine u call GDM directly or u call other numbers then if they have
> noan set then they would forward to GDM (3100)
>
> Just my thoughts here though! :)
>
> On Thu, May 28, 2009 at 11:23 PM, Sergio Polizer  wrote:
> > Hi, If you could associate a Label for the secondary lines,  a possible
> > solution could be:
> >
> > Face Requirement:
> >
> > Phone 1
> >
> > Line1 : 3001
> >
> > Line2 : Support 1
> >
> > Phone 2
> >
> > Line1 : 3002
> >
> > Line2 : Support 2
> >
> > ephone-dn 1
> > number 3001
> >
> > ephone-dn 2
> > number 3002
> >
> > ephone-dn 3
> > number 3101
> >
> > ephone-dn 4
> > number 3102
> >
> > ephone-dn 5
> > number 3100 (GDN)
> > label Support 1
> > call-forward noan VM timeout 12
> >
> > ephone-dn 6
> > number 3100 (GDN)
> > label Support 2
> > call-forward noan VM timeout 12
> >
> > ephone 1
> > button 1:2 2o5,3
> >
> > ephone 2
> > button 1:2 2o5,4
> >
> >
> > In this case, both line will ring together. I don't know if it will break
> > any other requirement like ring line 1 and after ring line 2, etc.
> >
> > 
> > From: cristian.rade...@crescendo.ro
> > To: ccie_voice@onlinestudylist.com
> > Date: Thu, 28 May 2009 12:22:16 +0300
> > Subject: Re: [OSL | CCIE_Voice] GDM configuration with notification on 2
> > phones
> >
> >
> >
> > I think this is not possible. With "secondary number" or "overlay" it will
> > not work.
> >
> >
> >
> > 
> >
> > From: ccie_voice-boun...@onlinestudylist.com
> > [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of David Corbeil
> > Sent: 27 May, 2009 8:42 PM
> > To: 'ccie_voice@onlinestudylist.com'
> > Subject: [OSL | CCIE_Voice] GDM configuration with notification on 2 phones
> >
> >
> >
> > Hi,
> >
> >
> >
> > I want to know if it's possible to have the voicemail letter on the second
> > line of 2 phones without changing the phone facing.
> >
> >
> >
> > Example:
> >
> >
> >
> > Phone 1
> >
> > Line1 : 3001
> >
> > Line2 : 3101
> >
> >
> >
> > Phone 2
> >
> > Line1 : 3002
> >
> > Line2 : 3102
> >
> >
> >
> > Each line need to be access to GDM mailbox, and when a message is left on
> > GDM I need to have VM Letter on both Line2 Phone.
> >
> > Can't have line 3
> >
> > Can't modify the look of the phone
> >
> >
> >
> > It's possible? If yes, how ?
> >
> >
> >
> > Thanks
> >
> > David Corbeil
> > Consultant en technologie | Technology Consultant
> > Tel. 514-798-4206 | Fax. 514-748-5333
> > Membre de l'équipe TELUS
> >
> >
> >
> > 
> > Conheça os novos produtos Windows Live. Clique aqui!
>
>
>
> --
> Sirus Moghadasian
> CCIE #21862 (R&S)

Quer uma internet mais segura? Baixe agora o novo Internet Explorer 8. É 
grátis!<http://brasil.microsoft.com.br/IE8/mergulhe/?utm_source=MSN%3BHotmail&utm_medium=Tagline&utm_campaign=IE8>

Re: [OSL | CCIE_Voice] VPIM CUE.message & MWI takes a long time to show up

[Cristi Radescu]

Hi,

It's a normal behavior. It's default value for check frequency.
Pleasee see below. You can change this in "Advanced Settings".

Conversation - System Broadcast Message Check Frequency

Specify how often Cisco Unity checks for new broadcast messages. By default, 
Cisco Unity checks for new broadcast messages every 5 minutes.


By the way. Are you sure you are seeing MWI for Unity Broadcasts? How did you 
configure this?

Cristi R



From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of john D
Sent: 28 May, 2009 3:39 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] VPIM CUE.message & MWI takes a long time to show up

Hi,
on CME router use VPIM to record message to HQ router.

Vmail is present on HQ & MWI seen
But this takes a long time to happen. Atleast 5 mins..

Anyone seen this situation before. ? Any suggestions how to fix this?


Thx
John

Re: [OSL | CCIE_Voice] GDM configuration with notification on 2 phones

[Cristi Radescu]

I think this is not possible. With "secondary number" or "overlay" it will not 
work.


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of David Corbeil
Sent: 27 May, 2009 8:42 PM
To: 'ccie_voice@onlinestudylist.com'
Subject: [OSL | CCIE_Voice] GDM configuration with notification on 2 phones

Hi,

I want to know if it's possible to have the voicemail letter on the second line 
of 2 phones without changing the phone facing.

Example:

Phone 1
Line1 : 3001
Line2 : 3101

Phone 2
Line1 : 3002
Line2 : 3102

Each line need to be access to GDM mailbox, and when a message is left on GDM I 
need to have VM Letter on both Line2 Phone.
Can't have line 3
Can't modify the look of the phone

It's possible? If yes, how ?

Thanks

David Corbeil
Consultant en technologie | Technology Consultant
Tel. 514-798-4206 | Fax. 514-748-5333
Membre de l'équipe TELUS

Re: [OSL | CCIE_Voice] Unity Call Transfer with Alternate Vs Standardgreetings.

[Cristi Radescu]

Restriction table in Unity?


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Josmar Ramirez
Sent: 26 May, 2009 8:43 PM
To: john D; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Unity Call Transfer with Alternate Vs 
Standardgreetings.

Check the CSS on the voicemail ports.. probably can't dial out to the PSTN 
number due to CSS restrictions.



From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of john D
Sent: Tuesday, May 26, 2009 1:38 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Unity Call Transfer with Alternate Vs 
Standardgreetings.

Hi,
I have setup call transfer for a subscriber (when noan/busy) to a PSTN no.
When a call is made is to this subscriber it seems to hit the "Alternate 
greetings "
and then gets stuck there
Hear message "Wait while i transfer your call" many times but call never 
succeeds in transfering to pstn no..
Here is the setup:

1) Subscriber->Standard greetings
  source: Blank
  After greeting:
  send call to subscriber(attempt transfer to same subscriber)
2) Alternate & Busy greetings same.

3) Call transfer : Yes ring subscriber at : PSTN No

The pstn no is accessible when  it is called directly from that subscriber.

Any suggestions how to fix this ? Tried rebooting unity , ccm . no luck!
I have tried this same steps many times before and it worked..

-Regards
J.

Re: [OSL | CCIE_Voice] call classification, how to determine if it is offnet?

[Cristi Radescu]

You can put offnet on gateway.


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of jeremy co
Sent: 25 May, 2009 3:41 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] call classification, how to determine if it is 
offnet?

if calls come from PSTN ,how ccm figure out that it is a offnet call not on net 
?

for calls that going out of ccm, it is determined by option exist in RP.


Jeremy

Re: [OSL | CCIE_Voice] caller id block code on ccme to pstn via r2

[Cristi Radescu]

For that case you would have two dial-peers, let's say for International:

dial-peer voice 100 pots
  destination-pattern 900T
  port 0/0/0:23
  prefix 00
dial-peer voice 101 pots
  destination-pattern *11900T
  port 0/0/0:23
  prefix 00
  clid restrict

It's just an example. I don't know what do you want to achieve exactly.

Hth,
Cristian



From: zamuel del Toro [mailto:sdelto...@hotmail.com]
Sent: 23 May, 2009 7:18 PM
To: Cristi Radescu; wantedtobec...@yahoo.in; ccie foro ipexpert
Subject: RE: [OSL | CCIE_Voice] caller id block code on ccme to pstn via r2


   caller-id block code *11, work fine only for internal call or voip call but 
when dialing code + pstn number the pstn still see the caller id no matter if 
you press or not the code.

From: cristian.rade...@crescendo.ro
To: sdelto...@hotmail.com; wantedtobec...@yahoo.in; 
ccie_voice@onlinestudylist.com
Date: Sun, 17 May 2009 11:05:06 +0300
Subject: RE: [OSL | CCIE_Voice] caller id block code on ccme to pstn via r2
telephony-service
   caller-id block code *11



From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of zamuel del Toro
Sent: 16 May, 2009 5:51 PM
To: wantedtobec...@yahoo.in; ccie foro ipexpert
Subject: Re: [OSL | CCIE_Voice] caller id block code on ccme to pstn via r2

thanks for help, just this command will rstrict all caller id  from all phones 
and not just when press the code.

Date: Sat, 16 May 2009 20:05:19 +0530
From: wantedtobec...@yahoo.in
Subject: Re: [OSL | CCIE_Voice] caller id block code on ccme to pstn via r2
To: sdelto...@hotmail.com; ccie_voice@onlinestudylist.com
Put on the dial-peer command CLID restrict.

Thks


From: zamuel del Toro 
To: ccie foro ipexpert 
Sent: Saturday, 16 May, 2009 8:02:36 PM
Subject: [OSL | CCIE_Voice] caller id block code on ccme to pstn via r2

How can restrict the caller id on ccme to pstn. this work fine on voip and same 
ccme extension but is not working on pstn site

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Re: [OSL | CCIE_Voice] V2- CME on hook tranfer

[Cristi Radescu]

Hi Sergio,

I have 12.4(5b)(c2800nm-ipvoice_ivs-mz.124-5b.bin) on my router.
I don't have access at lab right now and I can't tell you exactly what version 
is on IP Phones.

Hth,
Cristian


From: Sergio Polizer [mailto:spoli...@hotmail.com]
Sent: 22 May, 2009 8:16 PM
To: Cristi Radescu
Cc: ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] V2- CME on hook tranfer

Cristian,

It sounds good. Which version do you have at CME and IP Phone?

Thank you, Sergio.


From: cristian.rade...@crescendo.ro
To: spoli...@hotmail.com
CC: ccie_voice@onlinestudylist.com
Date: Fri, 22 May 2009 17:08:26 +0300
Subject: RE: [OSL | CCIE_Voice] V2- CME on hook tranfer
Hi Sergio,

I just tested it and it works as you said. When I put Phone B On Hook, phone A 
is connected with C.
I have CME configured with full-consult.
If you will configure as "full-blind" when you press transfer on phone B it 
will connect phones A and C directly.
But, even in this case you must put phone B Off Hook because you don't have 
"Transfer" key on "alerting" mode.


Hth,
Cristian





From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Sergio Polizer
Sent: 22 May, 2009 4:46 PM
To: cyrus@gmail.com; lhadr...@ipexpert.com
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] V2- CME on hook tranfer

Thank you for your answers.

Right, if we have  "transfer-system full-consult"  or "local-consult" 
configured we can make a transfer with consult so, B can Talk to C before 
transfer A to C.

But the only way to do that is when B press the transfer button to proceed with 
the transfer.

At CM, as you probally know, there is a service parameters that we can provide 
on hook transfer, so when B hangs up, A is connected to C.

I was just wondering if it is possible at CME.

Regards, Sergio.

Date: Fri, 22 May 2009 17:16:29 +1000
Subject: Re: [OSL | CCIE_Voice] V2- CME on hook tranfer
From: cyrus@gmail.com
To: lhadr...@ipexpert.com
CC: spoli...@hotmail.com; ccie_voice@onlinestudylist.com

Hi Larry,

I think this way after pressing transfer it rings other party and as soon as he 
picks up the phone, call would transfer to him,

Neither B can talk to C after he picked up the phone nor this requirement would 
satisfy "B hung up and A is connected to C."


Anyway interesting scenario.


Cyrus

On Fri, May 22, 2009 at 11:20 AM, Larry Hadrava 
mailto:lhadr...@ipexpert.com>> wrote:
Sounds like you are looking to do a consultative transfer.

telephonyservice
transfer transfer-system full-consult

This should be the default action.
correct me if my understanding of the requirement is incorrect.

Larry Hadrava
CCIE #12203 CCNP CCNA
Sr. Support Engineer - IPexpert, Inc.
URL: http://www.IPexpert.com<http://www.ipexpert.com/>
On Thu, May 21, 2009 at 9:05 PM, Sergio Polizer 
mailto:spoli...@hotmail.com>> wrote:
Hi,

I'm trying to enable on hook transfer at CME.
Does someone know a way to do that?

E.g.

A calls B
B answers
B press transfer and calls C
B talks to C
B hung up and A is connected to C.


Thanks in advance, Sergio.

Descubra uma nova internet. Internet Explorer 8. 
Mergulhe.<http://brasil.microsoft.com.br/IE8/mergulhe/?utm_source=MSN;Hotmail&utm_medium=Tagline&utm_campaign=IE8>




--
Sirus Moghadasian
CCIE #21862 (R&S)

Conheça os novos produtos Windows Live. Clique 
aqui!<http://www.windowslive.com.br/>


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Re: [OSL | CCIE_Voice] V2- CME on hook tranfer

[Cristi Radescu]

Hi Sergio,

I just tested it and it works as you said. When I put Phone B On Hook, phone A 
is connected with C.
I have CME configured with full-consult.
If you will configure as "full-blind" when you press transfer on phone B it 
will connect phones A and C directly.
But, even in this case you must put phone B Off Hook because you don't have 
"Transfer" key on "alerting" mode.


Hth,
Cristian





From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Sergio Polizer
Sent: 22 May, 2009 4:46 PM
To: cyrus@gmail.com; lhadr...@ipexpert.com
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] V2- CME on hook tranfer

Thank you for your answers.

Right, if we have  "transfer-system full-consult"  or "local-consult" 
configured we can make a transfer with consult so, B can Talk to C before 
transfer A to C.

But the only way to do that is when B press the transfer button to proceed with 
the transfer.

At CM, as you probally know, there is a service parameters that we can provide 
on hook transfer, so when B hangs up, A is connected to C.

I was just wondering if it is possible at CME.

Regards, Sergio.

Date: Fri, 22 May 2009 17:16:29 +1000
Subject: Re: [OSL | CCIE_Voice] V2- CME on hook tranfer
From: cyrus@gmail.com
To: lhadr...@ipexpert.com
CC: spoli...@hotmail.com; ccie_voice@onlinestudylist.com

Hi Larry,

I think this way after pressing transfer it rings other party and as soon as he 
picks up the phone, call would transfer to him,

Neither B can talk to C after he picked up the phone nor this requirement would 
satisfy "B hung up and A is connected to C."


Anyway interesting scenario.


Cyrus


On Fri, May 22, 2009 at 11:20 AM, Larry Hadrava 
mailto:lhadr...@ipexpert.com>> wrote:
Sounds like you are looking to do a consultative transfer.

telephonyservice
transfer transfer-system full-consult

This should be the default action.
correct me if my understanding of the requirement is incorrect.

Larry Hadrava
CCIE #12203 CCNP CCNA
Sr. Support Engineer - IPexpert, Inc.
URL: http://www.IPexpert.com

On Thu, May 21, 2009 at 9:05 PM, Sergio Polizer 
mailto:spoli...@hotmail.com>> wrote:
Hi,

I'm trying to enable on hook transfer at CME.
Does someone know a way to do that?

E.g.

A calls B
B answers
B press transfer and calls C
B talks to C
B hung up and A is connected to C.


Thanks in advance, Sergio.

Descubra uma nova internet. Internet Explorer 8. 
Mergulhe.




--
Sirus Moghadasian
CCIE #21862 (R&S)

Conheça os novos produtos Windows Live. Clique 
aqui!

Re: [OSL | CCIE_Voice] Voice problems on ProctorLabs Rack

[Cristi Radescu]

Resolved: It was a routing problem with my old VPN Client. Strange one because 
everything else worked perfect(e.g. I could login on every equipment).
I installed last version(5.0.5.xx) and everything is fine.


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cristi Radescu
Sent: 22 May, 2009 12:14 PM
To: ccie foro ipexpert
Subject: [OSL | CCIE_Voice] Voice problems on ProctorLabs Rack


Hi all,

Does anybody have problems with voice stream from rack?
I am on a remote rack, everything is configured ok, my calls hit the CUE and 
Unity pilot numbers, the call is connected but I can't hear anything.
I can't hear prompts from Unity or CUE.

Any thought will be highly appreciated,
Cristian

[OSL | CCIE_Voice] Voice problems on ProctorLabs Rack

[Cristi Radescu]

Hi all,

Does anybody have problems with voice stream from rack?
I am on a remote rack, everything is configured ok, my calls hit the CUE and 
Unity pilot numbers, the call is connected but I can't hear anything.
I can't hear prompts from Unity or CUE.

Any thought will be highly appreciated,
Cristian

Re: [OSL | CCIE_Voice] caller id block code on ccme to pstn via r2

[Cristi Radescu]

telephony-service
   caller-id block code *11



From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of zamuel del Toro
Sent: 16 May, 2009 5:51 PM
To: wantedtobec...@yahoo.in; ccie foro ipexpert
Subject: Re: [OSL | CCIE_Voice] caller id block code on ccme to pstn via r2

thanks for help, just this command will rstrict all caller id  from all phones 
and not just when press the code.

Date: Sat, 16 May 2009 20:05:19 +0530
From: wantedtobec...@yahoo.in
Subject: Re: [OSL | CCIE_Voice] caller id block code on ccme to pstn via r2
To: sdelto...@hotmail.com; ccie_voice@onlinestudylist.com
Put on the dial-peer command CLID restrict.

Thks


From: zamuel del Toro 
To: ccie foro ipexpert 
Sent: Saturday, 16 May, 2009 8:02:36 PM
Subject: [OSL | CCIE_Voice] caller id block code on ccme to pstn via r2

How can restrict the caller id on ccme to pstn. this work fine on voip and same 
ccme extension but is not working on pstn site

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[OSL | CCIE_Voice] REG-G729only and REG-G711only Dependencies

[Cristi Radescu]

Hi List,

I have the following scenario:

- we have GK in a G729only region;
- we have MOH Server or SIP Trunk in G711only region;

How do you configure these Regions? What Region has priority?
If "G711only" region has priority and we configure G711 between "REG-G729only" 
and "REG-G711only", when a call is made from  CME side we will see BW:128kbps 
on GK. This way we will violate an eventual G729 codec selection condition 
between sites.
If "G729only" region has priority and we configure G729 between "REG-G729only" 
and "REG-G711only" is it possible to have problems with calls between SIP FXS 
and CME Phones(sure I am assuming we have a hardware transcoder in SIP Trunk 
MRGL)?

Please share your thoughts.

Many thanks in advance,
Cristian

Re: [OSL | CCIE_Voice] Windows & VMware Combination

[Cristi Radescu]

You can try Vmware ESXi. It's free and you will have more free resources for 
Virtual Machines.
http://www.vmware.com/products/vi/esx/



From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Fawad asad
Sent: 14 May, 2009 11:44 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Windows & VMware Combination


Team -

I am in process to install windows on my 7825 servers and I wanted to know what 
is the right combination of windows and VMware I need in order to install CUCM 
7.0,Presence 7.0 and IPCC 7.0 on VMware.

Thanks in advance.

Kind Regards,
Fawad

Re: [OSL | CCIE_Voice] Gatekeeper controller trunk & MTP ( 7.X)

[Cristi Radescu]

As a result  could we say, as a best practice, to uncheck "MTP Required" and 
"Wait for Far End H.245 Terminal Capability Set" on Gatekeeper Trunk?


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Michael Ciarfello
Sent: 14 May, 2009 8:05 PM
To: vineet sanghi; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Gatekeeper controller trunk & MTP ( 7.X)

SRND 7.0.  See the section: H.323 Trunks with Media Termination Points
Your case 1 looks like it complies with the SRND.  There is also other 
discussion in the SRND for when to use MTP vs when not to.  Later on there is a 
good discussion on MTP and SIP and DTMF-Relay.


All-in-All the SRND should be read COVER TO COVER 100 times during your 
journey.  It's something I have to do more myself.


-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of vineet sanghi
Sent: Thursday, May 14, 2009 10:42 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Gatekeeper controller trunk & MTP ( 7.X)


Hi,

I am testing Gatekeeper controlled trunk with BR2.

BR2 dialpeer is configured for G729.
HQ GK Controller trunk is in G729 region with all other regions. "Wait for ECS" 
is not checked on GK trunk.
BR1 phones are G729 with HQ and GK controlled trunk.


G729
HQGK--BR2
  |  
Default codec(G729)
  |
  | G729
 BR1


Now BR2 calls HQ and HQ phone transfers to BR1. The end result is to connect 
BR2 phone to BR1 phone.

Case 1. If "MTP required" is not enabled then every thing works without 
transcoders. the codec on phones is G729 in this case.

Case 2. If "MTP required" is enabled then BR2 phones shows G729 codec. HQ 
phones shows G711 codec with transcoder on HQ router but transfer to BR1 phone 
fails. In this case direct call from BR2 to BR1 also fails. In this case BR1 
phone rings but doesn't negotiate Codec and drops.

 Inter-site BW is configured for 500K. All devices has enough transcoders. I 
tried looking into software MTP utilization via RTMT with any success.


Please help me to understand the reasons of failure in second case and what can 
be the possible solutions?

Vineet

Re: [OSL | CCIE_Voice] unable to cfw

[Cristi Radescu]

Sorry for that typo.

You can configure preference in the following way:

Call-manager-fallback
 max-dn 4 dual-line preference 2

I tested it and it worked for me.


-Original Message-
From: ccieid1ot [mailto:ccieid...@gmail.com]
Sent: 14 May, 2009 7:42 PM
To: Cristi Radescu
Cc: Kumar, Narinder; ccie_voice@onlinestudylist.com; Jenny Chris
Subject: Re: [OSL | CCIE_Voice] unable to cfw

Now that you mention it.

dialplan-pattern 1 5303073... extension-length 4 extension-pattern
3... <with a lower preference then alias (or higher memory loss)


You can not configure preference on the ephones since it's a SRST gw.


On Thu, May 14, 2009 at 1:35 AM, Cristi Radescu
 wrote:
> What preference do you have on ephone-dns?
> Try to configure "ephone-dn 4 dual-line preference 2" and let alias command 
> as is.
>
> Hth,
> Cristian
>
> -Original Message-
> From: ccie_voice-boun...@onlinestudylist.com 
> [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kumar, Narinder
> Sent: 14 May, 2009 4:05 AM
> To: ccieid1ot
> Cc: ccie_voice@onlinestudylist.com; Jenny Chris
> Subject: Re: [OSL | CCIE_Voice] unable to cfw
>
> Yes you can call the HQ Phone from BR1 Phone in SRST mode
>
> -Original Message-
> From: ccieid1ot [mailto:ccieid...@gmail.com]
> Sent: Thursday, 14 May 2009 10:58 AM
> To: Kumar, Narinder
> Cc: Jenny Chris; ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] unable to cfw
>
> Can you call the HQ phone while in SRST mode?
>
> On Wed, May 13, 2009 at 6:35 PM, Kumar, Narinder
>  wrote:
>> Under call-manager-fallback do you have "call-forward pattern .T " if not
>> then add it in, also when you perform debug voice dialpeer what do you see ?
>>
>>
>>
>> From: ccie_voice-boun...@onlinestudylist.com
>> [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jenny Chris
>> Sent: Thursday, 14 May 2009 8:14 AM
>> To: ccie_voice@onlinestudylist.com
>> Subject: [OSL | CCIE_Voice] unable to cfw
>>
>>
>>
>> Hi,
>> Sorry if this question has been posted before.
>> Trying to test cfw to a HQ phone under SRST mode on BR1
>> Scenario:
>> call comes into BR1 Phone via PSTN (BR1 in SRST), busy/noans forward to HQ
>> Phone
>> under call-manager-fallback tried using alias command but doesnt work .Have
>> the dialplan-pattern configured for incoming calls like this:
>>
>> !
>> call-manager-fallback
>>  dialplan-pattern 1 5303073... extension-length 4 extension-pattern 3...
>>   alias 1 5303073003 to 3003 cfw HQPHONE timeout 6
>> !
>> A dp to route this exists. a Normal call to HQ Phone works.
>>
>> any way to solve this problem?
>> Thx
>>
>> J
>>
>> 
>> CONFIDENTIALITY - The information contained in this electronic mail message
>> is confidential and is intended solely for the addressee(s). If you are not
>> an authorised recipient of this message please contact UXC Getronics
>> Australia immediately by reply email and destroy/delete this message from
>> your computer. Any unauthorised form of reproduction of this message, or
>> part thereof, is strictly prohibited.
>> DISCLAIMER - Unless specifically indicated otherwise, the views and opinions
>> expressed in this email are those of the sender and not UXC Getronics
>> Australia. While we endeavour to protect our network from computer viruses,
>> UXC Getronics Australia does not warrant that this email or any attachments
>> are free of viruses or any other defects or errors. It is the duty of the
>> recipient to virus scan and otherwise test any information contained in this
>> email before loading onto any computer system.
>>
>
>
> CONFIDENTIALITY - The information contained in this electronic mail message 
> is confidential and is intended solely for the addressee(s). If you are not 
> an authorised recipient of this message please contact UXC Getronics 
> Australia immediately by reply email and destroy/delete this message from 
> your computer.  Any unauthorised form of reproduction of this message, or 
> part thereof, is strictly prohibited.
> DISCLAIMER - Unless specifically indicated otherwise, the views and opinions 
> expressed in this email are those of the sender and not UXC Getronics 
> Australia.  While we endeavour to protect our network from computer viruses, 
> UXC Getronics Australia does not warrant that this email or any attachments 
> are free of viruses or any other defects or errors.  It is the duty of the 
> recipient to virus scan and otherwise test any information contained in this 
> email before loading onto any computer system.
>
>

Re: [OSL | CCIE_Voice] unable to cfw

[Cristi Radescu]

What preference do you have on ephone-dns?
Try to configure "ephone-dn 4 dual-line preference 2" and let alias command as 
is.

Hth,
Cristian

-Original Message-
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kumar, Narinder
Sent: 14 May, 2009 4:05 AM
To: ccieid1ot
Cc: ccie_voice@onlinestudylist.com; Jenny Chris
Subject: Re: [OSL | CCIE_Voice] unable to cfw

Yes you can call the HQ Phone from BR1 Phone in SRST mode

-Original Message-
From: ccieid1ot [mailto:ccieid...@gmail.com]
Sent: Thursday, 14 May 2009 10:58 AM
To: Kumar, Narinder
Cc: Jenny Chris; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] unable to cfw

Can you call the HQ phone while in SRST mode?

On Wed, May 13, 2009 at 6:35 PM, Kumar, Narinder
 wrote:
> Under call-manager-fallback do you have "call-forward pattern .T " if not
> then add it in, also when you perform debug voice dialpeer what do you see ?
>
>
>
> From: ccie_voice-boun...@onlinestudylist.com
> [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jenny Chris
> Sent: Thursday, 14 May 2009 8:14 AM
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] unable to cfw
>
>
>
> Hi,
> Sorry if this question has been posted before.
> Trying to test cfw to a HQ phone under SRST mode on BR1
> Scenario:
> call comes into BR1 Phone via PSTN (BR1 in SRST), busy/noans forward to HQ
> Phone
> under call-manager-fallback tried using alias command but doesnt work .Have
> the dialplan-pattern configured for incoming calls like this:
>
> !
> call-manager-fallback
>  dialplan-pattern 1 5303073... extension-length 4 extension-pattern 3...
>   alias 1 5303073003 to 3003 cfw HQPHONE timeout 6
> !
> A dp to route this exists. a Normal call to HQ Phone works.
>
> any way to solve this problem?
> Thx
>
> J
>
> 
> CONFIDENTIALITY - The information contained in this electronic mail message
> is confidential and is intended solely for the addressee(s). If you are not
> an authorised recipient of this message please contact UXC Getronics
> Australia immediately by reply email and destroy/delete this message from
> your computer. Any unauthorised form of reproduction of this message, or
> part thereof, is strictly prohibited.
> DISCLAIMER - Unless specifically indicated otherwise, the views and opinions
> expressed in this email are those of the sender and not UXC Getronics
> Australia. While we endeavour to protect our network from computer viruses,
> UXC Getronics Australia does not warrant that this email or any attachments
> are free of viruses or any other defects or errors. It is the duty of the
> recipient to virus scan and otherwise test any information contained in this
> email before loading onto any computer system.
>


CONFIDENTIALITY - The information contained in this electronic mail message is 
confidential and is intended solely for the addressee(s). If you are not an 
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Re: [OSL | CCIE_Voice] DND feature ring and Hunt-group problem

[Cristi Radescu]

Hi Jeremy,

In fact pressing DND on that phone logout the phone from hunt.
So, when you are calling hunt number that phone is logged out and it will not 
ring. It seems like DND is active but is not. Actuallly the phone is logout 
from hunt.
You can see this with "sh ephone-hunt".
You can not change - as far as I found - this behavior on CME 3.3.

Hth,
Cristian


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of jeremy co
Sent: 11 May, 2009 11:21 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] DND feature ring and Hunt-group problem

Hi,

I want to disable DND on DN  6001 while it's a member of hunt-group (so it 
receive calls when DND enables on ephone)

DND button pressed on ephone 1 .what happened is DND disabled on 6001 DN on 
ephone 1 but when calling to Hunt group pilot 6100, Hunt-group treat it as DND 
is enabled on it.

!
ephone-dn  1  dual-line
 number 6001 no-reg primary

ephone  1
 no dnd feature-ring
 button  1f1 2o11,3

ephone-hunt 1 sequential
 pilot 6100
 list 6001, 6002
 timeout 12
 no-reg


any workaround for this?

Jeremy

[OSL | CCIE_Voice] V2: SWAP lab date in San Jose with one in Brussels

[Cristi Radescu]

Hi,

I have a lab date on July 2nd in San Jose and I wanna change it with one in 
Brussels, same period, before change to V3.
I'm not interested in selling date.
If someone is interested please answer me asap.


Many thanks in advance,
Cristian