Re: [cisco-voip] Unity Connection - Exchange permission

[Rob Dawson]

Unless you create a management scope to filter access, and apply it when you 
assign impersonation rights to the service account, then it would have access 
to all Exchange accounts by default.

Rob

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
george.hend...@caci.com
Sent: Monday, July 11, 2016 1:08 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Unity Connection - Exchange permission

Hey Guys,

  I know the service account in Exchange for Unity Connection requires 
application impersonation permissions.  Does this permission allow the service 
account to view a user's email box content even if the user doesn't have an 
account in Unity Connection?

Regards,
Bill

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Re: [cisco-voip] DS3 voice delivery?

[Rob Dawson]

You can still attach service to an AS5400 through January, 2016. Pick up a 
reman and load it with 5 years of smartnet. We never had issues with our 5300s, 
rock solid.

Sent from my iPhone

> On Jun 8, 2016, at 1:29 PM, Doug McIntyre  wrote:
> 
> No, the NM-1T3/E3 is a data only part, packet over T3. 
> 
> Cisco did used to have a product, the VGD-1T3 that did just that,
> which looked like it was mostly a rebranded AS5200 with the T3 interface
> and pushed to voice instead of modems. But it is discontinued, probably
> because they only sold a handful. 
> 
> The M13 MUX to VWIC card solution is probably the best solution still.
> 
> 
>> On Wed, Jun 08, 2016 at 12:45:09PM -0400, Lelio Fulgenzi wrote:
>> There's this... 
>> 
>> http://www.cisco.com/c/en/us/products/collateral/interfaces-modules/2600-3600-3700-series-t3-e3-network-module/product_data_sheet09186a008010fba2.html
>>  
>> 
>> But in CUCM v9.1, I don't see it as a pull down option in a 3900 series 
>> gateway configuration. 
>> 
>> So, not sure it's supported for voice. 
>> 
>> --- 
>> Lelio Fulgenzi, B.A. 
>> Senior Analyst, Network Infrastructure 
>> Computing and Communications Services (CCS) 
>> University of Guelph 
>> 
>> 519‐824‐4120 Ext 56354 
>> le...@uoguelph.ca 
>> www.uoguelph.ca/ccs 
>> Room 037, Animal Science and Nutrition Building 
>> Guelph, Ontario, N1G 2W1 
>> 
>> - Original Message -
>> 
>> From: "Nick Barnett"  
>> To: "Cisco VoIP Group"  
>> Sent: Wednesday, June 8, 2016 9:51:15 AM 
>> Subject: [cisco-voip] DS3 voice delivery? 
>> 
>> So, we have an old MUX that died. We have to either replace the MUX or use 
>> something else. Is it possible to use a DS3 SM on an ISR to terminate the 
>> DS3? Right now, we have a DS3 that hits a mux and breaks out to 28 PRIs... 
>> those PRIs go into a plethora of VWIC interfaces on a SIP router. Would it 
>> be possible to get another module for this router that lets us plug the coax 
>> in and skip all of this VWIC/MUX business? 
>> 
>> Thanks, 
>> Nick 
>> 
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> 
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Re: [cisco-voip] DS3 voice delivery?

[Rob Dawson]

At a previous employer we used AS5300s with DS3 interfaces as the PRI gateways 
for our Broadworks platform, straight from TDM to SIP. I am not sure what the 
modern analogue to that solution would be but I don’t believe you will find a 
DS3 card for an ISR device that supports voice.

Rob

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Nick 
Barnett
Sent: Wednesday, June 08, 2016 9:51 AM
To: Cisco VoIP Group 
Subject: [cisco-voip] DS3 voice delivery?

So, we have an old MUX that died. We have to either replace the MUX or use 
something else. Is it possible to use a DS3 SM on an ISR to terminate the DS3? 
Right now, we have a DS3 that hits a mux and breaks out to 28 PRIs... those 
PRIs go into a plethora of VWIC interfaces on a SIP router. Would it be 
possible to get another module for this router that lets us plug the coax in 
and skip all of this VWIC/MUX business?

Thanks,
Nick
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Re: [cisco-voip] Collab summit keynote simulcast on now...

[Rob Dawson]

I only caught the last half but it was essentially Spark as a voice and video 
call control platform integrated with all the other standard Spark features, 
either as a standalone platform or in a hybrid configuration with on-prem 
infrastructure via Expressway connectors, all accessed through a unified 
client. Endpoints can register directly via QR code, you can move calls 
seamlessly between Spark enabled endpoints and across on-prem and hosted 
infrastructure, ability to use cloud based call term and origination or hybrid 
model with on-prem T, native 25 way multi-party video, self-admin portal, 
etc. To be sold through the partner channel. The Cisco answer to Skype for 
Business, Broadsoft’s UC-One portfolio, etc.

Training materials should be coming out Q1CY2016 to coincide with NA roll-out, 
ROW to follow. Additionally, I think the keynote video will be published today 
at some point.

I liked it, a lot actually, but I come from a hosted/managed background so my 
perspective may be a little different.

Rob
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Anthony Holloway
Sent: Wednesday, December 09, 2015 1:26 AM
To: Brian Meade
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] Collab summit keynote simulcast on now...

Brian, I missed most of it.  Where can I see a replay, or read the cliff notes? 
 Care to give us the highlights?

On Tue, Dec 8, 2015 at 3:17 PM, Brian Meade 
> wrote:
What did everyone think?  Seems fairly polished but I'm sure definitely missing 
a ton of traditional telephony features.

On Tue, Dec 8, 2015 at 12:18 PM, Abhiram Kramadhati (akramadh) 
> wrote:
https://communities.cisco.com/community/technology/collaboration/collaborat
ion_virtual_experience


On 08/12/15, 10:38 PM, "cisco-voip on behalf of Lelio Fulgenzi"
 
on behalf of le...@uoguelph.ca> wrote:

>
>Head on over to the collab user group pages for a link to the simulcast.
>
>Sent from my iPhone
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Re: [cisco-voip] BE7K or not?

[Rob Dawson]

The newest BE7ks are built on C240 M4 hardware. The BE7H-M4-K9 is 20 cores, 128 
GB RAM, 4+ TB of available disk space, the BE7M-M4-K9 is 12 cores, 64 GB RAM, 
and around 2.6 TB available disk.

Cisco states on the data sheet that BE7k is “optimized for enterprise-scale 
organizations with 1000 to 5000 users and 3000 to 15,000 devices” but they also 
note “For more capacity to support larger sized deployments, simply stack 
additional servers. And in smaller sized deployments with less than 1000 users, 
typically more applications can be supported per server.”

I haven’t seen any practical limitations, and as pointed out by someone else 
you can use the VM Placement Tool to see what OVAs fit, and scale to size.

Rob

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Lamont, Joshua
Sent: Thursday, December 03, 2015 10:21 AM
To: Lelio Fulgenzi
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] BE7K or not?

Sorry for the confusion... We had originally planned to install C240 M3s but 
swapped them out for the C240 M4s instead.

Joshua Lamont
Senior Telecommunications Engineer
Brown University
office (401) 863-1003
cell(401) 749-6913

On Thu, Dec 3, 2015 at 10:14 AM, Lelio Fulgenzi 
> wrote:

There are BE7Ks built on the new UCS C240 M4s.

Or by "new" do you mean something else other than the 240M4s?


---
Lelio Fulgenzi, B.A.
Senior Analyst, Network Infrastructure
Computing and Communications Services (CCS)
University of Guelph

519‐824‐4120 Ext 56354
le...@uoguelph.ca
www.uoguelph.ca/ccs
Room 037, Animal Science and Nutrition Building
Guelph, Ontario, N1G 2W1


From: "Joshua Lamont" >
To: "Matthew Loraditch" 
>
Cc: "Lelio Fulgenzi" >, "cisco-voip 
voyp list" >
Sent: Thursday, December 3, 2015 10:02:47 AM
Subject: Re: [cisco-voip] BE7K or not?

We're in the process of converting to Cisco UC and recently during our design 
phase swapped out the BE7Ks in favor of the new UCS C series severs. C240 M4s 
are used for the majority of our cluster.

Joshua Lamont
Senior Telecommunications Engineer
Brown University
office (401) 863-1003
cell(401) 749-6913

On Thu, Dec 3, 2015 at 9:46 AM, Matthew Loraditch 
> 
wrote:
http://tools.cisco.com/ucs
is your friend, pick your server model and the ovas you will use.

In general though BE7K should be fine. No limitations besides overall 
resources. Just gotta figure out how many you need or want

Matthew G. Loraditch – CCNP-Voice, CCNA-R, CCDA
Network Engineer
Direct Voice: 443.541.1518
Facebook | 
Twitter | 
LinkedIn 
| G+

From: cisco-voip 
[mailto:cisco-voip-boun...@puck.nether.net]
 On Behalf Of Lelio Fulgenzi
Sent: Thursday, December 3, 2015 9:38 AM
To: cisco-voip voyp list 
>
Subject: [cisco-voip] BE7K or not?


We're spec'ing out some new UCS servers for our next upgrades.

Still trying to figure out what, if any, limitations going with a BE7K might 
have. Does anyone care to share a reason why we shouldn't go with BE7K, or why 
we shouldn't go with a la cart using the BOM?

There was once an issue with a limitation of users/devices, but I think that's 
gone now. My biggest concern is trying to do something on the BE7K and finding 
out only after that it's not allowed or supported.

Any ideas would be great.

Lelio


---
Lelio Fulgenzi, B.A.
Senior Analyst, Network Infrastructure
Computing and Communications Services (CCS)
University of Guelph

519‐824‐4120 Ext 56354
le...@uoguelph.ca
www.uoguelph.ca/ccs
Room 037, Animal Science and Nutrition Building
Guelph, Ontario, N1G 2W1


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[cisco-voip] Unity Licenses

[Rob Dawson]

Ca you still order Unity licenses? Anyone have a SKU? This is for a DOD 
customer running 8.6 LSC.

Thanks,
Rob
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Re: [cisco-voip] BE7K or not?

[Rob Dawson]

to add memory and/or additional NICs as with the 
other TRCs. Can't really justify it now, but as future versions come up, I'd 
like to have the ability to add more memory if the CUCM v11 or v12 version 
requires it.

It's really only memory and NIC throughput that I can see having to increase 
and only as the apps demand more with each version.

Lelio


---
Lelio Fulgenzi, B.A.
Senior Analyst, Network Infrastructure
Computing and Communications Services (CCS)
University of Guelph

519‐824‐4120 Ext 56354
le...@uoguelph.ca<mailto:le...@uoguelph.ca>
www.uoguelph.ca/ccs<http://www.uoguelph.ca/ccs>
Room 037, Animal Science and Nutrition Building
Guelph, Ontario, N1G 2W1


From: "Rob Dawson" <rdaw...@force3.com<mailto:rdaw...@force3.com>>
To: "cisco-voip voyp list" 
<cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>>
Sent: Thursday, December 3, 2015 11:27:07 AM
Subject: Re: [cisco-voip] BE7K or not?


The newest BE7ks are built on C240 M4 hardware. The BE7H-M4-K9 is 20 cores, 128 
GB RAM, 4+ TB of available disk space, the BE7M-M4-K9 is 12 cores, 64 GB RAM, 
and around 2.6 TB available disk.



Cisco states on the data sheet that BE7k is “optimized for enterprise-scale 
organizations with 1000 to 5000 users and 3000 to 15,000 devices” but they also 
note “For more capacity to support larger sized deployments, simply stack 
additional servers. And in smaller sized deployments with less than 1000 users, 
typically more applications can be supported per server.”



I haven’t seen any practical limitations, and as pointed out by someone else 
you can use the VM Placement Tool to see what OVAs fit, and scale to size.



Rob



From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Lamont, Joshua
Sent: Thursday, December 03, 2015 10:21 AM
To: Lelio Fulgenzi
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] BE7K or not?



Sorry for the confusion... We had originally planned to install C240 M3s but 
swapped them out for the C240 M4s instead.


Joshua Lamont

Senior Telecommunications Engineer

Brown University

office (401) 863-1003

cell(401) 749-6913



On Thu, Dec 3, 2015 at 10:14 AM, Lelio Fulgenzi 
<le...@uoguelph.ca<mailto:le...@uoguelph.ca>> wrote:



There are BE7Ks built on the new UCS C240 M4s.



Or by "new" do you mean something else other than the 240M4s?





---
Lelio Fulgenzi, B.A.
Senior Analyst, Network Infrastructure
Computing and Communications Services (CCS)
University of Guelph



519‐824‐4120 Ext 56354<tel:519%E2%80%90824%E2%80%904120%20Ext%2056354>
le...@uoguelph.ca<mailto:le...@uoguelph.ca>
www.uoguelph.ca/ccs<http://www.uoguelph.ca/ccs>
Room 037, Animal Science and Nutrition Building
Guelph, Ontario, N1G 2W1





From: "Joshua Lamont" <joshua_lam...@brown.edu<mailto:joshua_lam...@brown.edu>>
To: "Matthew Loraditch" 
<mloradi...@heliontechnologies.com<mailto:mloradi...@heliontechnologies.com>>
Cc: "Lelio Fulgenzi" <le...@uoguelph.ca<mailto:le...@uoguelph.ca>>, "cisco-voip 
voyp list" <cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>>
Sent: Thursday, December 3, 2015 10:02:47 AM
Subject: Re: [cisco-voip] BE7K or not?



We're in the process of converting to Cisco UC and recently during our design 
phase swapped out the BE7Ks in favor of the new UCS C series severs. C240 M4s 
are used for the majority of our cluster.


Joshua Lamont

Senior Telecommunications Engineer

Brown University

office (401) 863-1003<tel:%28401%29%20863-1003>

cell(401) 749-6913<tel:%28401%29%20749-6913>



On Thu, Dec 3, 2015 at 9:46 AM, Matthew Loraditch 
<mloradi...@heliontechnologies.com<mailto:mloradi...@heliontechnologies.com>> 
wrote:

http://tools.cisco.com/ucs

is your friend, pick your server model and the ovas you will use.



In general though BE7K should be fine. No limitations besides overall 
resources. Just gotta figure out how many you need or want



Matthew G. Loraditch – CCNP-Voice, CCNA-R, CCDA
Network Engineer
Direct Voice: 443.541.1518

Facebook<https://www.facebook.com/heliontech?ref=hl> | 
Twitter<https://twitter.com/HelionTech> | 
LinkedIn<https://www.linkedin.com/company/helion-technologies?trk=top_nav_home> 
| G+<https://plus.google.com/+Heliontechnologies/posts>



From: cisco-voip 
[mailto:cisco-voip-boun...@puck.nether.net<mailto:cisco-voip-boun...@puck.nether.net>]
 On Behalf Of Lelio Fulgenzi
Sent: Thursday, December 3, 2015 9:38 AM
To: cisco-voip voyp list 
<cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>>
Subject: [cisco-voip] BE7K or not?





We're spec'ing out some new UCS servers for our next upgrades.



Still trying to figure out what, if any, limitations going with a BE7K might 
have. Does anyone ca

Re: [cisco-voip] 3rd Party backup tools

[Rob Dawson]

I have cloned nodes for lab testing with no issue, but during a window when I 
could shut them down first.

On the flip side, I have seen nodes get built erroneously on servers that were 
being backed up by Veeam and a stuck snapshot from a backup job cause call 
processing errors and node restarts.

Rob


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Nathan Reeves
Sent: Tuesday, October 27, 2015 12:42 AM
To: Ryan Huff
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] 3rd Party backup tools

We've always had DRS backups in place given thats the officially supported path 
for restore.  Do keep some Veeam backups in place but tend to run them 
infrequently when we're looking to do work on the cluster and want a quick 
fallback position.  I would tend to power down and backup to ensure consistency 
as I'm not sure how well informix works with running machine snapshots on 
restore.

Nathan

On Tue, Oct 27, 2015 at 1:02 AM, Ryan Huff 
> wrote:
While it is more than NOT supported; I have had success with cloning ucos vm's 
(while powered off). Then powering on the clones with the parents turned off. 
Again, clearly not supported but I have not noticed any behavior differences.

For kick's and curiosity's sake I have vmotioned a lab cluster (with sip pstn) 
before while powered on ... caused cpu spikes and the trunks had to be reset.

However, in practice I'd not wonder outside the bounds of the DocWiki for 
backup purposes.


Sent from my T-Mobile 4G LTE Device


 Original message 
From: Anthony Holloway
Date:10/26/2015 12:20 PM (GMT-05:00)
To: "Heim, Dennis"
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] 3rd Party backup tools
As a Cisco partner, I will be sticking to the book on this one (DRS).

http://docwiki.cisco.com/wiki/Ongoing_Virtualization_Operations_and_Maintenance#Backup.2C_Restore.2C_and_Server_Recovery

However, I am very curious to know what else works, and how well it works.

On Sun, Oct 25, 2015 at 11:12 PM, Heim, Dennis 
> wrote:
Whats the current thought process around 3rd party backup tools to quickly 
restore a CUCM? I’ve played with Veeam in the past. If the VM is shutdown, 
would that be kosher?

Dennis Heim | Emerging Technology Architect (Collaboration)
World Wide Technology, Inc. | +1 314-212-1814
[cid:image001.png@01D10DD2.7FC81F90]
[cid:image002.png@01D10DD2.7FC81F90][cid:image003.png@01D10DD2.7FC81F90][cid:image004.png@01D10DD2.7FC81F90]
“There is a fine line between Wrong and Visionary. Unfortunately, you have to 
be a visionary to see it." – Sheldon Cooper
“The greatest danger for most of us is not that our aim is too high and we miss 
it, but that it is too low and we reach it.” -- Michelangelo Buonarroti
“We should tansform the way we work” -- RowanTrollope

Click here to join me in my Collaboration Meeting 
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Re: [cisco-voip] Video Architecture

[Rob Dawson]

Personally, I am definitely seeing video as a component a lot more. In the last 
year I’ve had to implement a large VTC pilot with all the requisite 
infrastructure (TMS, Telepresence Server, Conductor, etc.), Collaboration 
Meeting rooms, etc. I view it all as just another part of convergence and an 
integral part of both “cisco-voip” and the Cisco Collaboration platform. As you 
said, the architecture is moving to 100% CUCM call control. We have gone from 
bespoke TDM solutions and working a punch down tool to needing to be 
network/systems/application/voice engineers. At the end of the day video is 
just another application on the network.

For a lot of customers video is still just kind of a gimmick and it is not 
ubiquitous enough to drive adoption widely. I think that as those of us in 
pre-sales become better at tying the video use case to a holistic collaboration 
solution then adoption will ramp up and we will start to find some pull through 
for customers that are on the ledge.

As for your follow-up question about resources, I think that the Cisco 
Preferred Architecture guides are pretty good. The latest one for video is here 
- 
http://www.cisco.com/c/dam/en/us/td/docs/solutions/PA/midmarket/11x/video11x.pdf.
 Vidoe is also covered as a component of the mid-market and enterprise 
collaboration architectures too. You can find them, as well as the Cisco 
Validated Design docs here - 
http://www.cisco.com/c/en/us/solutions/enterprise/design-zone-collaboration/index.html.

Rob

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Anthony Holloway
Sent: Saturday, October 17, 2015 10:26 AM
To: Cisco VoIP Group
Subject: [cisco-voip] Video Architecture

Does this list, being labeled "VoIP" and not "Voice", include Video in the "V", 
or is the "V" for Voice only?  A bit of tongue-in-cheek there, but I think you 
know what I mean.

With every day that passes, I feel like, as a Voice Engineer, I am being asked 
to know and implement Video more and more.

From CUCM taking on more of the video call control (just about 100% now, no?), 
to video being a part of the CCIE Collab (to some degree, don't flame me for 
mentioning it), the line between voice and video engineering roles is blurring.

I know there has been quite a bit of Expressway talk on here this year, but I 
don't think I've explicitly read a thread which was 100% focused on video 
technology.  What's the direction you see these two technologies heading, and 
should/could this list serve as both a Voice and Video discussion group?
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Re: [cisco-voip] DRS Backups 10.5

[Rob Dawson]

Concur – the linux option takes ohhh, 20 minutes start to finish and just runs.

[cid:image002.png@01D101BC.50AD0210]<http://www.force3.com/>



Rob Dawson
Solutions Architect
2151 Priest Bridge Dr. Crofton, MD 21114



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M 571-234-2621
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From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan 
Huff
Sent: Thursday, October 08, 2015 11:14 AM
To: Jason Aarons (AM)
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] DRS Backups 10.5

I think the best option is (and has been for awhile, imho) to spin up or box 
your favorite Linux flavor with openssh  (Arch, Ubuntu, Debian) come to mind. 
Not hard to setup at all and they just run, almost a set-it-and-forget-it.

-Ryan

Sent from my iPad

On Oct 8, 2015, at 11:05 AM, Jason Aarons (AM) 
<jason.aar...@dimensiondata.com<mailto:jason.aar...@dimensiondata.com>> wrote:
Being only GlobalScape and Titan are the only two commercial SFTP TAC supported 
products is a source of contention. Non-technical customers don’t want to 
compile anything. I wish FTP and CIFS/SMB was supported. Keep it simple and 
encrypt the file, not the transmission.

I haven’t been able to duplicate the FreeFTPD problem of 1GB file limitation in 
my lab, nor does FreeFTPD indicate they have any file size limitation, it’s 
some sort of Cisco made up release notes limitation that just has TAC wash 
their hands.

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Buchanan, James
Sent: Thursday, October 8, 2015 5:03 AM
To: Heim, Dennis <dennis.h...@wwt.com<mailto:dennis.h...@wwt.com>>; 
norm.nichol...@kitchener.ca<mailto:norm.nichol...@kitchener.ca>; 
cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] DRS Backups 10.5


Actually, it is only SFTP, right? FTP is not supported (a continual pain point 
for customers).


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Heim, 
Dennis
Sent: Wednesday, October 07, 2015 3:17 PM
To: norm.nichol...@kitchener.ca<mailto:norm.nichol...@kitchener.ca>; 
cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] DRS Backups 10.5

SFTP/FTP Is the supported mechanism.

Dennis Heim | Emerging Technology Architect (Collaboration)
World Wide Technology, Inc. | +1 314-212-1814
[twitter]<https://twitter.com/CollabSensei>
<xmpp:dennis.h...@wwt.com>[Phone]<tel:+13142121814>[video]<sip:dennis.h...@wwt.com>
“There is a fine line between Wrong and Visionary. Unfortunately, you have to 
be a visionary to see it." – Sheldon Cooper
“The greatest danger for most of us is not that our aim is too high and we miss 
it, but that it is too low and we reach it.” -- Michelangelo Buonarroti
“We should tansform the way we work” -- RowanTrollope

Click here to join me in my Collaboration Meeting 
Room<https://wwt.webex.com/meet/dennis.heim>

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
norm.nichol...@kitchener.ca<mailto:norm.nichol...@kitchener.ca>
Sent: Wednesday, October 07, 2015 1:52 PM
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: [cisco-voip] DRS Backups 10.5


I have been asked if Cisco supports SMB or NFS for our nightly DRS backups 
instead of SFTP.



Thanks



Norm Nicholson
Telecom Analyst
City of Kitchener
(519) 741-2200 x 7000



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itevomcid
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[cisco-voip] Presence status for third party app

[Rob Dawson]

Doing some tinkering and looking for a way to monitor a particular users status 
from a third party app. I assume I could just do it via XMPP, but is there an 
preferred method for doing this?

Thanks,
Rob
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Re: [cisco-voip] Presence status for third party app

[Rob Dawson]

Thanks. I actually just stumbled on the Jabber web SDK on the Dev site, which 
will probably do what I need –

https://developer.cisco.com/site/jabber-websdk/overview/

The app is non-existent as of yet, just pondering some things – triggering real 
world events based on status, that type of stuff.

Thanks again,
Rob

From: avhollo...@gmail.com [mailto:avhollo...@gmail.com] On Behalf Of Anthony 
Holloway
Sent: Friday, October 02, 2015 3:23 PM
To: Rob Dawson
Cc: voip puck
Subject: Re: [cisco-voip] Presence status for third party app

Have you looked into this:

https://developer.cisco.com/site/unified-presence/overview/

Or are you asking for a ready made solution?  What's the third party app?

On Fri, Oct 2, 2015 at 2:16 PM, Rob Dawson 
<rdaw...@force3.com<mailto:rdaw...@force3.com>> wrote:
Doing some tinkering and looking for a way to monitor a particular users status 
from a third party app. I assume I could just do it via XMPP, but is there an 
preferred method for doing this?

Thanks,
Rob

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[cisco-voip] CSR 11 in the wild

[Rob Dawson]

Anyone have any substantial experience with CSR 11 in the wild? Looking for 
stability issues you may have experienced, etc.

Thanks,
Rob
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Re: [cisco-voip] Nortel 81C / CS1000 SIP Trunk to CUCM 10.5.2

[Rob Dawson]

I was going to mention the FQDN issue as I ran into it last month. Also, we had 
a scenario where the Nortel was sending SRTP keys in the SDP and it would cause 
issues, though I believe the symptom there was one way audio and not a complete 
failure. The fix for that was to disable SRTP in the COS on the Nortel. 

Rob

-Original Message-
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Daniel Pagan
Sent: Tuesday, July 14, 2015 9:30 AM
To: Michael T. Voity; voip puck
Subject: Re: [cisco-voip] Nortel 81C / CS1000 SIP Trunk to CUCM 10.5.2

Is your Nortel PBX sending a FQDN in the Contact header? This is important 
because in 10.5(2) CUCM performs a SRV and A Record lookup on FQDNs contained 
in a SIP Contact header. If this lookup fails, then expect to see a CANCEL 
followed by a BYE. I encountered this a few times over the past few months and 
ended up creating a defect against it. Check out CSCuu84269. If you want, I 
wouldn't mind taking a quick look at your detailed CCM SDL traces offline and 
letting you know if you're experiencing this defect. If you are, I'll send you 
a sample LUA script to use for converting the FQDN to IP address as a 
workaround which you can use for testing and workaround purposes (... and I 
can't guarantee this will work for you).

Hope this helps.

- Dan
 
-Original Message-
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Michael T. Voity
Sent: Monday, July 13, 2015 9:56 AM
To: voip puck
Subject: [cisco-voip] Nortel 81C / CS1000 SIP Trunk to CUCM 10.5.2

Hello,

Before we installed our Cisco CM 10.5.2 system everything here at the 
University is fed from a Nortel Avaya 81c / CS1000 system.   The Telcom 
group has a bunch of systems on it that support SIP and SIP gateways.   
We setup a SIP trunk between the two systems from a guide that Avaya 
provided.   It works fine like 99% of the time.

I am finding that I have to reset the SIP trunk every couple of days because it 
looks like the Nortel is busying out all the channels and it 
can only pass certain traffic.   Example is that someone from Nortel 
land dials a 5 digit extension that has been routed to CUCM, the line on CUCM 
rings once and then discos the call.

Looking at RTMT on the SIP traffic I can tell that the Nortel is sending 
the BYE message on the trunk right when the CUCM sends the RINGING   
The only way that I have found to correct this is to reset the SIP trunk from 
CUCM.

Has anyone see an issue like this and or heard of this?

Any ideas would be helpful!

-Mike

--
Michael T. Voity
Network Engineer
University of Vermont

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Re: [cisco-voip] Expressway ?'s

[Rob Dawson]

I’ll second BIND views, they are a great way to handle split horizon DNS.


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Norton, Mike
Sent: Friday, June 19, 2015 1:29 AM
To: Lelio Fulgenzi; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Expressway ?'s

I don’t know anything about MRA, but firewalling DNS responses does sound icky. 
If your DNS servers happen to be BIND then this sounds suspiciously like a job 
for BIND’s “views” feature, if you haven’t looked into that already.

-mn


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Lelio 
Fulgenzi
Sent: Thursday, June 18, 2015 8:01 PM
To: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Expressway ?'s

I really wish there was another option other than split DNS to get MRA working 
from off-premise. I mean, why rely on DNS response rather than lack of 
connectivity to decide which path to take? A parameter in the jabber-config.xml 
file could help with that.

Anyways, I know it's gonna be fun to use the workaround of configuring our edge 
firewall to filter out DNS responses. ugh.
---
Lelio Fulgenzi, B.A.
Senior Analyst, Network Infrastructure
Computing and Communications Services (CCS)
University of Guelph

519‐824‐4120 Ext 56354
le...@uoguelph.camailto:le...@uoguelph.ca
www.uoguelph.ca/ccshttp://www.uoguelph.ca/ccs
Room 037, Animal Science and Nutrition Building
Guelph, Ontario, N1G 2W1


From: Charles Goldsmith wo...@justfamily.orgmailto:wo...@justfamily.org
To: Scott Voll svoll.v...@gmail.commailto:svoll.v...@gmail.com
Cc: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Sent: Thursday, 18 June, 2015 7:45:14 PM
Subject: Re: [cisco-voip] Expressway ?'s
As said by others, license is free for the MRA part, to get the free license, 
here is a handy blog entry : 
https://ciscocollab.wordpress.com/2014/02/20/how-to-get-expressway-c-and-e-licenses/

He also has entries on helping set it up, but it's pretty simple once you get 
in and start configuring.  Hard part is getting the certs, DNS and firewall in 
line :)


On Thu, Jun 18, 2015 at 4:58 PM, Scott Voll 
svoll.v...@gmail.commailto:svoll.v...@gmail.com wrote:
Im still on UC 8.6.  we are planning an upgrade to 10.x  We currently have 
DLU's for licensing and will be moving to CUWL Standard ( I think).

How does Expressways factor into this?

is it part of CUWL?  Is there a Cost?  What all can you do with Expressway.  
What I believe I understand is that it can get your external voice and video 
internal.  does it replace my lan to lan connections to get an IP phone 
registered to CM?

Does it also do video bridging?  Example. Polycom HDX unit, cisco SX20, jabber 
and skype all in a single call?

TIA

Scott



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[cisco-voip] Where is the SpeechView hosted?

[Rob Dawson]

Anyone know what country the SpeechView transcription service and personnel are 
located in?

Thanks,
Rob
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Re: [cisco-voip] IMP not pinging

[Rob Dawson]

Can you ping it from your gateway? Double check your subnet mask and make sure 
it matches your other hosts.

Verify ICMP and HTTP/S  is allowed - utils firewall ipv4 list

You can disable and test – utils firewall ipv4 disable

Verify services are running - utils service list

[cid:image002.png@01D08CA8.29CED6C0]http://www.force3.com/



Rob Dawson
Solutions Architect
2151 Priest Bridge Dr. Crofton, MD 21114



O 410-774-7153
M 571-234-2621
Check out the new Force3.comhttp://www.force3.com !

[Facebook]https://www.facebook.com/force3inc[Twitter]https://twitter.com/force3[LinkedIn]http://www.linkedin.com/company/force-3



From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Jason 
Aarons (AM)
Sent: Tuesday, May 12, 2015 7:53 AM
To: Techguy; Cisco VoIP Group
Subject: Re: [cisco-voip] IMP not pinging


if you can't ping it and the console isn't coming up, then it sounds like a 
re-install is necessary.  I assume you've tried some reboots etc.






From: cisco-voip [cisco-voip-boun...@puck.nether.net] on behalf of Techguy 
[techguy...@gmail.com]
Sent: Tuesday, May 12, 2015 3:03 AM
To: Cisco VoIP Group
Subject: [cisco-voip] IMP not pinging

Hi all,

Installed an IMP 8.6 in the lab but is not pingable. I can ping other nodes 
from it (using console), but not vice versa. Unable to access the GUI. Any 
known considerations to try?
All nodes are in the same class-c network.

Thanks


itevomcid
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Re: [cisco-voip] Temp Fail since upgrade

[Rob Dawson]

Anytime a vendor fixes an issue and either claims they did nothing or provides 
no detail about the resolution I refer to it as “pulling a Verizon” or say that 
they “Verizoned” the issue. I think it comes from a culture of protecting their 
co-workers and it is pretty commonplace.

I had a customer in NJ that watched a Verizon tech drive up, climb the pole 
outside the facility and after about 5 minutes the sites PRIs dropped. He spoke 
to the tech who said that he didn’t do anything related to the PRIs. The 
customer calls in a ticket and tells them that the tech was working on the 
pole. Of course they have to run automated testing, etc. so a full 24+ hours 
later, he sees another tech pull up, climb the pole,  and within a few minutes 
the PRIs were back up. Verizon closed the ticket as “No Problem Found”.


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Lisa 
Notarianni
Sent: Thursday, May 07, 2015 11:02 AM
To: cisco-voip voyp list
Subject: Re: [cisco-voip] Temp Fail since upgrade

FYI on this issue – some asked me to post once it was resolved.

The problem disappeared and resolved itself once I reported it to Verizon.  I 
believe Verizon made changes on their end they found were incorrect but never 
reported to me what they did.  There are so many people at Verizon buried deep 
in the Central Offices that we do not have direct contact with anymore there is 
no way to find out what the solution was.

Thank you for all of your responses.

[LNsignatureFile]

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Daniel Pagan
Sent: Monday, April 06, 2015 11:13 AM
To: Ryan Ratliff (rratliff); Jason Aarons (AM)
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] Temp Fail since upgrade

Adding to Ryan’s suggestion re: ACKs - I took a snip of some internal 
documentation I wrote re: MGCP debugs that might help. The MGCP transaction ID, 
in a CCM SDL trace or MGCP packet debug output, would look similar to this: 
http://i.imgur.com/O1YGnUO.png. Just showing the transaction ID matching 
between the MGCP command and 200 response. The observed event and Line package 
/hd (off-hook) highlighted can be ignored.

It might also be helpful to debug MGCP packets on the voice router… just make 
sure to disable console logging before leaving something like that running and 
increase the buffer size so you capture a failure. Seeing a missing 200 MGCP 
response in SDL traces doesn’t mean the router didn’t send one, and same the 
other way around where seeing an MGCP command sent from CUCM doesn’t mean it 
arrived at the router. In any case MGCP debugs off the router would come in 
handy in tandem with CCM traces.

Hope this helps.

- Dan


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan 
Ratliff (rratliff)
Sent: Monday, April 06, 2015 10:37 AM
To: Jason Aarons (AM)
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] Temp Fail since upgrade

The first place to look is MGCP packets being lost.  Search for “Retry” in the 
ccm traces similar to:
CCM|MGCPHandler TransId: 1476004 Timeout. Retry#1

That example is from a case in 2008 so what you see may not look exactly the 
same.  This type of message is an indicator that CCM is resending an MGCP 
message that wasn’t Ack’d by the gateway.  After 3 or so of those CCM will 
unregister the gateway and any phones talking to it will get Temp Fail.

Aside from lost messages it could be a RSIP or something that is gateway 
initiated causing the reset.

-Ryan

On Apr 5, 2015, at 10:15 AM, Jason Aarons (AM) 
jason.aar...@dimensiondata.commailto:jason.aar...@dimensiondata.com wrote:

Nothing has changed with MGCP for a long time…Cisco just doesn’t test older IOS.

Pull normal debugs and traces, RTMT alerts, etc.

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Kevin 
Przybylowski
Sent: Monday, March 30, 2015 5:10 PM
To: Lisa Notarianni; Ryan Huff; 
cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Temp Fail since upgrade


Are the mgcp gateways on a supported IOS for 10.5?

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Lisa 
Notarianni
Sent: Monday, March 30, 2015 5:02 PM
To: Ryan Huff; cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Temp Fail since upgrade

2 PRIs one each Communication Media Module– MGCP gateways. 2 Call Managers set 
up as redundant.

Cisco TAC looked at RTMT  with me today.  They want us to set up packet 
captures and try to duplicate the problem tomorrow.  I am just wondering if 
this is an upgrade issue that anyone else may be experiencing since we had no 
problems before the upgrade.



From: Ryan Huff [mailto:ryanh...@outlook.com]
Sent: Monday, March 30, 2015 4:52 PM
To: Lisa Notarianni; 
cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Temp Fail since upgrade

Lisa,
SIP or TDM/PRI?
Have you gandered into RTMT and taken a 

Re: [cisco-voip] QOS - Looking for another set of eyeballs

[Rob Dawson]

The UCCX SRND states that “Unified CCX software does not mark any network 
packet, so ensure that you mark the traffic at the network edge routers”, so 
that traffic is likely not marked EF and would therefore not match the 
class-map, unless you are classifying it somewhere else?

Rob

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan 
Huff
Sent: Monday, May 04, 2015 8:13 AM
To: Ed Leatherman
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] QOS - Looking for another set of eyeballs

Ed,

Thanks for the reply. Yes it is the intent to match ef and ACL 51. The access 
list contains a /24 of servers for UCCE/UCCX products. I thought an ACL to be 
easier than trying to match vXML/CVP traffic inside of HTTP traffic.

Thanks for the second set of eyes.

-r


Date: Mon, 4 May 2015 07:53:32 -0400
Subject: Re: [cisco-voip] QOS - Looking for another set of eyeballs
From: ealeather...@gmail.commailto:ealeather...@gmail.com
To: ryanh...@outlook.commailto:ryanh...@outlook.com
CC: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Ryan,

The match-all keyword in your class-map VOICE is going to cause it to only 
match things that are BOTH EF marked AND match acl-51 - I couldn't tell from 
your initial email if that was your intent. If you want just either type of 
traffic to get the VOICE treatment then you need match-any.

Also agree with John re: using priority instead so that it kicks in LLQ for 
those packets.

On Sat, May 2, 2015 at 9:55 AM, Ryan Huff 
ryanh...@outlook.commailto:ryanh...@outlook.com wrote:
The below OUT map, applied in the output direction on a WAN(mpls) facing 
interface, should put RTP, Signaling and anything from access-list 51 at the 
top of the heap and give everything else best effort.

Not that anything isn't working, I just want to make sure I'm not making 
something up ... etc. Seems basic but I don't get to play with QOS everyday :)

router-3925#sh run | sec class-map|policy-map|access-list 51
!
!
class-map match-all VOICE
 match ip dscp ef
 match access-group 51
class-map match-any CALL-SIGNALING
 match ip dscp cs3
 match ip dscp af31
!
!
policy-map WAN-OUT
 class VOICE
  bandwidth percent 30
 class CALL-SIGNALING
  bandwidth percent 10
 class class-default
  fair-queue
!
!
access-list 51 permit 001.002.003.004 0.0.0.255

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--
Ed Leatherman
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Re: [cisco-voip] T1 Clock source question

[Rob Dawson]

Whichever interface you select in the ‘network-clock-select’ command provides 
signal to the PLL, which controls the on-board clocking domain. The other 
interface will get clock from the line on the RX buffer, but since the TX 
buffer is using the on-board domain for clock you will get slips on that 
interface. You can only have one clock source for the on-board PLL.

It may not cause issues, but I have certainly seen instances of dropped calls 
and other anomalies where the only abnormal diagnostic was clock slips, and 
correcting the slips resolved the issues.

I think that you can use network modules to split the clock domains since they 
have their own onboard PLLs.

Rob


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan 
Huff
Sent: Wednesday, April 22, 2015 3:38 PM
To: Matthew Loraditch; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] T1 Clock source question

That's correct, and that is where I am at. I have different carriers and I am 
wondering if I should/need to use multiple clock sources? I'm in way-back mode 
myself here.

I have a T1 PRI and a T1 CAS (separate providers). Can't I just clock source 
one and it provides that clock on the backplane to everything?

Thanks,

-r

From: 
mloradi...@heliontechnologies.commailto:mloradi...@heliontechnologies.com
To: ryanh...@outlook.commailto:ryanh...@outlook.com; 
cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] T1 Clock source question
Date: Wed, 22 Apr 2015 19:33:26 +
This is one of my foggier areas, but wouldn’t you use multiple when you have 
T1s from multiple carriers? I could be wrong.. very wrong..  I love Ethernet 
circuits! :P
Matthew G. Loraditch – CCNP-Voice, CCNA-RS, CCDA
Network Engineer
Direct Voice: 443.541.1518
Facebookhttps://www.facebook.com/heliontech?ref=hl | 
Twitterhttps://twitter.com/HelionTech | 
LinkedInhttps://www.linkedin.com/company/helion-technologies?trk=top_nav_home 
| G+https://plus.google.com/+Heliontechnologies/posts

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan 
Huff
Sent: Wednesday, April 22, 2015 3:27 PM
To: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: [cisco-voip] T1 Clock source question

I have always subscribed to the idea that you should use one clocking source in 
situations where you have multiple T1's. In fact, I've read (not experienced) 
that when you use multiple clocking domains, that in-fact can sometimes cause 
slips.

Can anyone explain the advantages of using multiple clocking sources?

Thanks,

Ryan
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[cisco-voip] Security audit tool

[Rob Dawson]

Is anyone aware of a security auditing tool for CUCM and its ancillary 
products? Looking for a pass/fail type of tool, or possibly even something that 
could apply/enforce a security template?

Thanks,
Rob
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Re: [cisco-voip] Request for information

[Rob Dawson]

At a previous employer we did this using Phone Services for our multi-tenant 
environment. We had a directory server that ran IIS and served up the XML for 
the directories and used ASP to query AD. We passed a variable to the script 
within the Phone Service URI to limit the scope of the search.

Somewhere there is a site that documents how to do this, but a 2 minute google 
search didn’t turn it up.

Rob


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Terry 
Oakley
Sent: Monday, April 06, 2015 1:03 PM
To: Brian Meade
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Request for information

Thanks.  I may have mislead all of you by not providing a clear statement.  I 
am looking for a phonebook service that would pull the data from MS Exchange 
instead of the Call Manager.   If anyone has that functionality or knows of a 
company that is doing that, that would be helpful.

Thanks though to Ryan as his application is very good.

Terry



From: bmead...@gmail.commailto:bmead...@gmail.com [mailto:bmead...@gmail.com] 
On Behalf Of Brian Meade
Sent: April 6, 2015 9:22 AM
To: Terry Oakley
Cc: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net; ryan huff
Subject: Re: [cisco-voip] Request for information

Ryan Huff on this list created a web front-end for the corporate directory- 
http://ryanthomashuff.com/2015/01/new-web-gui-for-call-manager-corporate-directory/

On Mon, Apr 6, 2015 at 11:11 AM, Terry Oakley 
terry.oak...@rdc.ab.camailto:terry.oak...@rdc.ab.ca wrote:
We are moving to CUCM 10.x and Exchange 2013 (for our voicemail).   As a 
post-secondary education institution we get very good pricing on Microsoft 
products so will continue to use Exchange as our voicemail solution.

The information I am looking for is about an online phonebook solution.   With 
all of the data for the users in Active Directory because of the VM 
requirement,  we can certainly get that information via the phones corporate 
directory option but we would also like to publicize a simple online phonebook 
that we can access via our Intranet.It would be useful to be able to 
include information that is not always in Active Directory such as 
department/college main numbers, FAX numbers etc.   Does anyone have a simple 
and cost effective solution that they are currently using that would also 
require minimal maintenance?   I can visualize us using the AD side and adding 
in the department numbers, FAX numbers and so on but if there is a solution 
that some of you are using that information would be much appreciated.

Thanks

Terry

Terry Oakley
Telecommunications Coordinator | Information Technology Services
Red Deer College |100 College Blvd. | Box 5005 | Red Deer | Alberta | T4N 5H5
work (403) 342-3521   |  FAX (403) 343-4034tel:%28403%29%20343-4034


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Re: [cisco-voip] BE7K Network card issue

[Rob Dawson]

Have you reseated the cards or tried swapping with known working cards?

I’ve definitely seen bad risers in out of box failures before . . . not two in 
one batch though.

Rob

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Tommy 
Schlotterer
Sent: Tuesday, April 07, 2015 2:17 PM
To: Nick; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] BE7K Network card issue

Did you use the Cisco ESXI custom image?

It has custom drivers for the server.

Tommy

Tommy Schlotterer | Systems Engineer
CCNA, CCNA Voice
48325 Alpha Dr. Ste. 150
Wixom, MI 48393
p 248.468.0710
e tschlotte...@netechcorp.commailto:tschlotte...@netechcorp.com
w netechcorp.comhttp://netechcorp.com/
 [cid:DE00F175-A6C9-45A6-B3AC-D658551F1586]
[cid:62EB95BF-20B4-4A1B-98FE-9E042594730A]https://www.linkedin.com/company/75843?trk=tyahtrkInfo=tarId%3A1397760375508%2Ctas%3Anetech%2Cidx%3A3-1-5
 [cid:47C21B6C-578D-4D72-BFF4-8A482CE7A978] 
https://www.facebook.com/NetechCorporation  
[cid:A362BD4D-9EC8-47CC-96A8-8A17AF38C15C] https://twitter.com/netechcorp  
[cid:F52A69B8-DA49-4CD7-91E9-C057917D90C2] 
https://www.youtube.com/user/NetechCorporation

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Nick
Sent: Tuesday, April 07, 2015 2:15 PM
To: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: [cisco-voip] BE7K Network card issue

Hi All

I have four BE7K (UCS C240 M3S) servers that have the two quad port network 
cards, they all came prebuilt with ESXi 5.5 U2 on them, however only the 
on-board Nic's show up anywhere, I cant see the quad card ports in the BIOS, 
CIMC or ESXi?

I have the ports on the quad cards patched and the switchports come up but 
cannot get the ports to appear, have checked BIOS etc but cannot see where they 
woudl be enabled.

Anyone come across anything like this before?

Regards

Nick
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Re: [cisco-voip] OT: 911 App for mobile phones...

[Rob Dawson]

Just an FYI, there may be some NSFW content on that page . . . thankfully no 
one from HR was walking by when I scrolled down and saw the breateses ☺

Rob


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Lelio 
Fulgenzi
Sent: Monday, March 30, 2015 3:12 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] OT: 911 App for mobile phones...

Interesting.

http://mic.com/articles/114046/you-probably-didn-t-know-calling-911-is-a-problem-but-it-is-these-guys-are-fixing-it
---
Lelio Fulgenzi, B.A.
Senior Analyst, Network Infrastructure
Computing and Communications Services (CCS)
University of Guelph

519‐824‐4120 Ext 56354
le...@uoguelph.camailto:le...@uoguelph.ca
www.uoguelph.ca/ccshttp://www.uoguelph.ca/ccs
Room 037, Animal Science and Nutrition Building
Guelph, Ontario, N1G 2W1

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[cisco-voip] 3rd Party PBX integration

[Rob Dawson]

Have a customer looking to integrate a Nortel CS1000 and a few Aastra/Intecom 
PointSpans with Unity Connection. Aside from PIMG/TIMG, SIP via CUCM and SIP 
normalization, and QSIG can anyone come up with any other options?

Thanks,
Rob
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Re: [cisco-voip] ATA190 Random De-Register from CUCM 10.5.2

[Rob Dawson]

There used to be a defect where ATAs with only one line configured/registered 
would deregister due to the second line constantly attempting to download a 
configuration. I am not sure if it only impacted the 18X or not though . . . 
I’ll try to find an article.

Rob













From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan 
Huff
Sent: Monday, March 23, 2015 9:20 AM
To: Alessandro Bertacco; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] ATA190 Random De-Register from CUCM 10.5.2

I've used the ATA190 and it's family of 18X pretty often. In the case of the 
SIP devices like the 190 or 187 I have always been able to use the default SIP 
Profile settings; of course I'm generally dealing with vanilla networks and 
CCM's built very closely to the SRND.

Here is where I would start (if you haven't already):

What do you have the device security mode set to? If you have it set to 
Authenticated or Encrypted but it has not received a CTL file, the phone will 
attempt registration up to four times to make a secure connection.

Are the MAC addresses for Phone 1 and Phone 2 correct (Phone 2 will usually 
loose the first two characters and append a 01 at the end of the device name). 
If you support auto registration, remove the devices and let them auto register 
and see if they stabilize.

Any layer 1/2 issues (change out the patch cable, check the connection into the 
switchport ... etc)? Do you see any interface drops on the switchport? Is the 
switchport full duplex 10/100 or 10/1000? Have you shut/no shut the port just 
to see if clearing the port helps?

Go through the IVR menu on the ATA itself and verify that it reports all the 
same settings that you CCM device configs show. 
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cata/190/1_0/english/administration/guide/sip/ATA190/a190_agBcd.html

Thanks,

Ryan

From: bertacco.alessan...@alice.itmailto:bertacco.alessan...@alice.it
To: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Date: Mon, 23 Mar 2015 11:28:44 +0100
Subject: [cisco-voip] ATA190 Random De-Register from CUCM 10.5.2
Hi Guys,
   ATA190 FW version 1.1.2.(005) with CUCM 10.5.2.1-5, de-register 
randomly, and need to reboot the device to re-register again on the CUCM.

Anyone as the same issue? Do you use a special SIP Device profile changing some 
timings, or you are using standard SIP Device Profile?

Thank you all

Regards


Alessandro Bertacco


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Re: [cisco-voip] Call Manager, Jabber, and Certificates

[Rob Dawson]

That section pretty much sums it up . . . if you a have self-signed certs, a 
private CA, or certs signed by an untrusted CA, then you have to add either the 
server certs themselves, or the root cert/chain for the issuing CA, to the 
_client_ computers/devices. This is not done via CUCM/Jabber though, it would 
be via whatever mechanism is provided by your OS/device manufacturer i.e. group 
policy for Microsoft, mobile device management, etc.

If the certs are untrusted the clients should be getting prompted for them and 
may have the ability to add them themselves based on security policies.

Rob


From: Joe Loiacono [mailto:jloia...@csc.com]
Sent: Friday, March 20, 2015 8:40 AM
To: Rob Dawson; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] Call Manager, Jabber, and Certificates

Rob Dawson rdaw...@force3.commailto:rdaw...@force3.com wrote on 03/19/2015 
10:50:55 AM:

 What document are you looking at?

Cisco Jabber for Windows 9.7 Installation and Configuration Guide

 As far as I know the only certificate “push” would be done via GPO
 or some similar mechanism. During the SSL handshake the server
 certificate is sent to the client and the client will attempt to
 validate either the cert itself, or the signing authority, against
 its trust list. If the certificate is not in the trust list then the
 client will be offered the opportunity to trust/add it to its store,
 but this is the server cert, not the root cert. If however the CA
 root cert (public or private) OR the privately signed cert is
 already in the trust list then it should work with no further
 intervention or prompting. Once the client trusts the certificate
 then the key exchange happens.
 I can’t really think of anytime that it would a solid decision,
 security wise, to allow a piece of software to install a trusted
 root certificate.


I'm thinking the action that we take on CUCM, which we refer to as 'pushing a 
cert to the Jabber client' is the following:



Import Root Certificates on Client Computers

Every server certificate should have an associated root certificate present in 
the trust store on client computers. Cisco Jabber validates the certificates 
that servers present against the root certificates in the trust store. If you 
get server certificates signed by a public CA, the public CA should already 
have a root certificate present in the trust store on the client computer. In 
this case, you do not need to import root certificates on the client computers.

You should import root certificates into the Microsoft Windows certificate 
store if:

• The certificates are signed by a CA that does not already exist in the trust 
store, such as a private CA.

Import the private CA certificate to the Trusted Root Certification Authorities 
store.

• The certificates are self-signed.

Import self-signed certificates to the Enterprise Trust store.




This is driving us nuts, so I'm wondering if we have self-signed server certs 
or we're using our own private CA, etc. I'm inquiring within, of course, just 
was curious what others had done here.

Many thanks,

Joe
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Re: [cisco-voip] Call Manager, Jabber, and Certificates

[Rob Dawson]

Fair enough, those certs are generated/managed via CUCM though. The section he 
quoted below is specifically in relation to Tomcat/XMPP certs.

Same conclusion though, make sure either the cert itself, or the CA root chain 
is trusted by the client computer.

Joe, have a look at this - 
http://www.cisco.com/c/en/us/support/docs/unified-communications/unified-presence/116917-technote-certificate-00.html

Rob


From: Ryan Ratliff (rratliff) [mailto:rratl...@cisco.com]
Sent: Friday, March 20, 2015 9:48 AM
To: Joe Loiacono
Cc: Rob Dawson; cisco-voip voyp list
Subject: Re: [cisco-voip] Call Manager, Jabber, and Certificates

There is the LSC and CTL files that do get transmitted from UCM to the Jabber 
client.
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/jabber/10_5/CJAB_BK_D6497E98_00_deployment-installation-guide-ciscojabber/CJAB_BK_D6497E98_00_deployment-installation-guide-ciscojabber_chapter_0110.html#JABW_RF_U3F30C79_00

The original sentence reads like a dumbed-down description of how trust chains 
work with any cert that UCM presents to Jabber during the establishing of a TLS 
session.

If you aren’t doing encryption then just make sure the trust store on whatever 
platform Jabber is running on (Windows, Mac, etc) can validate the cert that 
UCM, IMP, etc will be presenting to it.

-Ryan

On Mar 20, 2015, at 8:39 AM, Joe Loiacono 
jloia...@csc.commailto:jloia...@csc.com wrote:

Rob Dawson rdaw...@force3.commailto:rdaw...@force3.com wrote on 03/19/2015 
10:50:55 AM:

 What document are you looking at?

Cisco Jabber for Windows 9.7 Installation and Configuration Guide

 As far as I know the only certificate “push” would be done via GPO
 or some similar mechanism. During the SSL handshake the server
 certificate is sent to the client and the client will attempt to
 validate either the cert itself, or the signing authority, against
 its trust list. If the certificate is not in the trust list then the
 client will be offered the opportunity to trust/add it to its store,
 but this is the server cert, not the root cert. If however the CA
 root cert (public or private) OR the privately signed cert is
 already in the trust list then it should work with no further
 intervention or prompting. Once the client trusts the certificate
 then the key exchange happens.
 I can’t really think of anytime that it would a solid decision,
 security wise, to allow a piece of software to install a trusted
 root certificate.


I'm thinking the action that we take on CUCM, which we refer to as 'pushing a 
cert to the Jabber client' is the following:



Import Root Certificates on Client Computers

Every server certificate should have an associated root certificate present in 
the trust store on client computers. Cisco Jabber validates the certificates 
that servers present against the root certificates in the trust store. If you 
get server certificates signed by a public CA, the public CA should already 
have a root certificate present in the trust store on the client computer. In 
this case, you do not need to import root certificates on the client computers.

You should import root certificates into the Microsoft Windows certificate 
store if:

• The certificates are signed by a CA that does not already exist in the trust 
store, such as a private CA.

Import the private CA certificate to the Trusted Root Certification Authorities 
store.

• The certificates are self-signed.

Import self-signed certificates to the Enterprise Trust store.




This is driving us nuts, so I'm wondering if we have self-signed server certs 
or we're using our own private CA, etc. I'm inquiring within, of course, just 
was curious what others had done here.

Many thanks,

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Re: [cisco-voip] CUCM Cluster Redundancy

[Rob Dawson]

You would have to replicate it using something like Veeam or similar which 
isn’t supported. The subs will provide “mostly” normal operation until you can 
rebuild the pub node. I would invest more thought into a solid backup, 
recovery, and test plan.

Have a look at this - 
http://docwiki.cisco.com/wiki/Unified_Communications_VMWare_Requirements

Rob













From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ahmed 
Abd EL-Rahman
Sent: Friday, March 20, 2015 10:03 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] CUCM Cluster Redundancy

Hi All,
  I have a UC cluster V10.5 on 2 UCS Blade servers having 1 Pub and 4 
Subs, and I’d like to ask if there is any need or benefit from having a cold 
standby virtual machine for the Pub server (a replicated VM for the Pub which 
is turned off to backup Pub functionality in case of Pub failure), and also if 
the license will be valid on this cold standby VM for the Pub so that if the 
main Pub fails and we turned on the cold standby Pub VM everything will be fine 
and the operation continues normally.

Waiting your feedback.


Best Regards

Ahmed Abd EL-Rahman
Senior Network Engineer
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Re: [cisco-voip] Extension that hangs up on the user?

[Rob Dawson]

How have I never seen this site before???

Thanks!

Rob


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Mike 
King
Sent: Wednesday, March 18, 2015 10:25 PM
To: James Andrewartha
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Extension that hangs up on the user?

I'm showing my age here, but in the Nortel mailing list I used to be apart of, 
everyone was aware if you needed a phone to do something wacky, you needed to 
look at the sandman.

This might work for you.
http://www.sandman.com/Wizard.html#CPCGenerator



On Wed, Mar 18, 2015 at 9:31 PM, James Andrewartha 
jandrewar...@ccgs.wa.edu.aumailto:jandrewar...@ccgs.wa.edu.au wrote:
On 18/03/15 23:55, Daniel Pagan wrote:
 Definitely makes sense and what I thought you were trying to achieve. If this 
 is the case, then rejecting the call via xlate or route patterns won't do - 
 the call will be rejected instead of connecting and then being disconnected. 
 You'll more than likely receive the standard recording from your call agent 
 or provider for a non-working number.

 I can't think of anything native to CUCM that would answer and then 
 disconnect the call. At the end of the day, for this to happen, the call 
 would not only need to be routed to some endpoint whether its SIP, SCCP, CTI, 
 H.323, or MGCP, but also accepted by the endpoint and only then 
 disconnected... this is basically what CUC is doing for you. In other words, 
 the call would need to go somewhere :) I was thinking maybe call queueing on 
 a hunt pilot with no logged in HG members but even that won't help.

 Do you have UCCX? Is CUBE part of the call-flow? You can do something 
 creative here like Tim Smith mentioned (TCL script in IOS). If UCCX, simply 
 route the call to a trigger, accept it, add a delay step for two seconds, and 
 then disconnect it.

No UCCX, we have 2921s as H.323 E1 gateways, but no CUBE licensing. I've
never done TCL scripting before, so I think the easiest option is to
just bite the bullet and upgrade it via PCD from 8.6 to 10.5 and join
the rest of the now-virtualised servers.

Thanks,

--
James Andrewartha
Network  Projects Engineer
Christ Church Grammar School
Claremont, Western Australia
Ph. (08) 9442 1757
Mob. 0424 160 877
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Re: [cisco-voip] Call Manager, Jabber, and Certificates

[Rob Dawson]

What document are you looking at?
As far as I know the only certificate “push” would be done via GPO or some 
similar mechanism. During the SSL handshake the server certificate is sent to 
the client and the client will attempt to validate either the cert itself, or 
the signing authority, against its trust list. If the certificate is not in the 
trust list then the client will be offered the opportunity to trust/add it to 
its store, but this is the server cert, not the root cert. If however the CA 
root cert (public or private) OR the privately signed cert is already in the 
trust list then it should work with no further intervention or prompting. Once 
the client trusts the certificate then the key exchange happens.
I can’t really think of anytime that it would a solid decision, security wise, 
to allow a piece of software to install a trusted root certificate.
Rob
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Joe 
Loiacono
Sent: Thursday, March 19, 2015 8:29 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Call Manager, Jabber, and Certificates


Jabber documentation indicates that the Certificate that the client may require 
and is 'pushed' from  Call Manager is a 'root certificate' that directs the 
Client to a trusted source that will validate the server's (Call Manager hosts) 
offered certificate.

If the Call Manager certificate is from a public trusted Certificate 
Authority(CA), and that CA is in the Windows certificate store, can the 
Certificate 'push' be avoided altogether?

Thanks,

Joe Loiacono
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Re: [cisco-voip] Migration strategy

[Rob Dawson]

This is the way I handle scenarios like this as well – create a temporary 
partition for the DNs, etc. and use BAT to bulk edit them when it is time to 
move to production. Having the profiles exposed shouldn’t cause any issues as 
long as the DNs are hidden.

Rob


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan 
Huff
Sent: Wednesday, March 18, 2015 8:06 AM
To: kiwi.vo...@gmail.com
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Migration strategy


Any elements that you are going to pre-stage that are a part of the dial plan 
(translations, route patterns, DNs, transformations ...etc) wilk all need to be 
isolated from your currently migrated phones.

So the DNs on your device profiles would likely need to be in an isolated 
partition, but the device profile itself probably doesn't need to be in an  
isolated device pool.

If you do this, remember to use a CLLI or some unique descriptor in the 
description field of all your pre-staged elements, it will make it very easy 
for BAT to find things later on.

Thanks,

Ryan


 Original Message 
From: 秀王 kiwi.vo...@gmail.commailto:kiwi.vo...@gmail.com
Sent: Tuesday, March 17, 2015 11:32 PM
To: Ryan Huff ryanh...@outlook.commailto:ryanh...@outlook.com
Subject: Re: [cisco-voip] Migration strategy
CC: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Hi Ryan,

let's say my phones are on a temp partition not reachable by other CSS. My UDP 
(user device profile) are on a valid partition shared by others ( Ie. 
P_Internal) but they are not logged in anywhere.
Will this confused the CUCM? Or i shall place the UDP on temp partition as 
well. If so, can BAT assist me in migrating from TEMP partition to actual 
partition (P_Internal)?
Cheers,
Ki Wi

On Wed, Mar 18, 2015 at 9:26 AM, Ryan Huff 
ryanh...@outlook.commailto:ryanh...@outlook.com wrote:

I'm not, sure I completely understand your questions but I'll attempt to answer 
based on my understanding.

Yes, you can pre-config devices and users in CCM prior to migration. If they 
are Cisco IP phones, you'll need the MAC address and model of the phone at a 
minimum.

If they are non Cisco IP phones, you'll need to pre configure 3rd party sip 
devices (which is a different license requirement than a Cisco IP phone).

Place the preconfigured dial plan that isnt migrated yet (on CCM), in a temp. 
partition that the already migrated phones cannot access. As you migrate, 
change that partition using BAT, to the correct partition for the portion of 
phones you migrated.

Thanks,

Ryan


 Original Message 
From: 秀王 kiwi.vo...@gmail.commailto:kiwi.vo...@gmail.com
Sent: Tuesday, March 17, 2015 09:15 PM
To: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: [cisco-voip] Migration strategy
Currently the client have avaya and cisco linked together using SIP.
Cisco UCM cluster have users in the production environment.

I'm are going to cutover more sites from avaya to cisco. Is it possible to 
preconfigure the users, extension number (let's say 87XXX range), phones and 
the user device profiles in advance?

I'm thinking that if I preconfigure those information, the cucm will think that 
those extension number (87XXX) are local and unregistered.
Is there a way to make CUCM thinks that in order to reach 87XXX range, it will 
still reach out to Avaya using the SIP trunk? Is there any setting in the route 
pattern can do that?
I thinking that CUCM will always find a more exact match locally instead of 
through other source like translation pattern or route pattern.


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Re: [cisco-voip] CUE 8.6.5 notification to gmail...

[Rob Dawson]

Have you tried a full SMTP AUTH and send test via telnet or just that the port 
responded?

Rob


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Jonathan Charles
Sent: Wednesday, March 11, 2015 11:43 AM
To: Brian Meade
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] CUE 8.6.5 notification to gmail...

Telnet to smtp.gmail.comhttp://smtp.gmail.com on 465 from that subnet works 
fine...


Jonathan

On Wed, Mar 11, 2015 at 9:40 AM, Brian Meade 
bmead...@vt.edumailto:bmead...@vt.edu wrote:
Port 465 blocked?

On Tue, Mar 10, 2015 at 10:19 PM, Jonathan Charles 
jonv...@gmail.commailto:jonv...@gmail.com wrote:
We loaded the gmail settings and we keep getting...

Test Result: Could not connect to SMTP host: 
smtp.gmail.comhttp://smtp.gmail.com, port: 465

We can ping GMAIL from the CUE:

se-192-168-124-2# ping smtp.gmail.comhttp://smtp.gmail.com
PING gmail-smtp-msa.l.google.comhttp://gmail-smtp-msa.l.google.com 
(74.125.193.109) 56(84) bytes of data.
64 bytes from 74.125.193.109http://74.125.193.109: icmp_seq=1 ttl=48 
time=12.5 ms
64 bytes from ig-in-f109.1e100.nethttp://ig-in-f109.1e100.net 
(74.125.193.109): icmp_seq=2 ttl=48 time=12.4 ms
64 bytes from ig-in-f109.1e100.nethttp://ig-in-f109.1e100.net 
(74.125.193.109): icmp_seq=3 ttl=48 time=12.4 ms
64 bytes from ig-in-f109.1e100.nethttp://ig-in-f109.1e100.net 
(74.125.193.109): icmp_seq=4 ttl=48 time=12.3 ms
64 bytes from ig-in-f109.1e100.nethttp://ig-in-f109.1e100.net 
(74.125.193.109): icmp_seq=5 ttl=48 time=12.3 ms

--- gmail-smtp-msa.l.google.comhttp://gmail-smtp-msa.l.google.com ping 
statistics ---
5 packets transmitted, 5 received, 0% packet loss, time 71ms
rtt min/avg/max/mdev = 12.329/12.411/12.550/0.124 ms, ipg/ewma 17.804/12.477 ms
se-192-168-124-2#

We have also verified our credentials... any idea why we keep getting the error?



Jonathan

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