[cisco-voip] solution support (SSSNT) vs product support (SNT)

[Lelio Fulgenzi]

anyone care to comment on solution support vs product support? 

list price over 60 months is not that big of a difference. it also seems to 
include onsite engineer as well according to cisco service finder. 

this would be for a group of C240 servers. 

i'm looking over the video portfolio as we speak... 

http://www.cisco.com/web/services/portfolio/solutions-support/index.html 



--- 
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] ExtMob Application window remains

[Scott Voll]

OK, I'm on CM 10.5.2.13038-1

I have 8861 phone running both 10.3.1 and 11.0.1.  Both phone loads after
logging into or out of extension Mobility leave the application windows on
the screen.  Is this a configuration issue?  I don't see a bug ID for it.
But users are tired of hitting the exit button.

TIA

Scott
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Re: [cisco-voip] VCS Checker

[Daniel Pagan]

Had no idea this XML generator was available. I’ve been grabbing sample scripts 
off Github and customizing them in the lab for testing purposes so this should 
supplement that setup very well. I just tested it for a few minutes and it 
seems like it replaces device-specific fields (via, contact header, 
request-uri, connection info field in the SDP, etc.) with SIPp-specific 
variables…

Very neat. Thanks for the heads up.

In case you’re into (or getting into) SIPp, here’s a list of scripts I 
reference whenever I needed a SIPp call-flow that are not difficult to 
customize.

https://github.com/saghul/sipp-scenarios

- Dan

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Tim 
Smith
Sent: Wednesday, December 09, 2015 6:45 PM
To: Matthew Loraditch ; Ryan Huff 
; cisco voip 
Subject: Re: [cisco-voip] VCS Checker

They are pretty much all under
http://www.cisco.com/c/en/us/support/web/tools-catalog.html

SIPp XML generator looks good

Cheers,

Tim

From: cisco-voip 
> 
on behalf of Matthew Loraditch 
>
Date: Thursday, 10 December 2015 at 08:18
To: Ryan Huff >, cisco voip 
>
Subject: Re: [cisco-voip] VCS Checker

Awesome, I wonder what else is hiding on this site... they have 
https://cway.cisco.com/sncheck/  (check smartnet and eol info) and 
https://cway.cisco.com/csc (upload attachments to tac cases via html5) as well

Seems to be the place for neat little tools…

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 Ryan 
Huff
Sent: Wednesday, December 9, 2015 4:13 PM
To: cisco voip >
Subject: [cisco-voip] VCS Checker


Not sure when Cisco made this tool public but it is a welcomed edition! I knew 
TAC had it but didn't realize Cisco opened it up.



https://cway.cisco.com/tools/SrvRecord


= Ryan =





Email: ryanthomash...@outlook.com

Spark: ryanthomash...@outlook.com

Twitter: @ryanthomashuff

LinkedIn: ryanthomashuff

Web ryanthomashuff.com
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Re: [cisco-voip] VCS destination outbound issue

[Brian Meade]

If you want to do it on your side, you pretty much have to do a SIP
Normalization Script in LUA.  It's definitely easier to have them do a
transform on the VCS side.

On Wed, Dec 9, 2015 at 3:10 PM, Andy Carse  wrote:

> Hi, I'm trying to get my head around an issue with VCS.
>
> Its to do with destination aliases I think, and I'm not sure if its my
> problem or the 3rd party we are trying to connect to.
>
>
>
> We try to dial another company's video endpoint as endpoints seem to work
> and others don't.
> We have SIP Route patterns pointing from Callmanager the the VCSc we can
> see the connection going out but we seem to be adding :5060 to the dest
> alias.
> My question is can I configure my end not to append the :5060 or is it for
> the receiving end the create a transform to chop of the :5060.
>
> Dst-alias-type="SIP" Dst-alias="sip:1234...@xxx.xxx.xxx.xxx:5060"
>
> If I can do it at my end how would I achieve it?
>
> TIA
>
> --
> Rgds Andy
>
>
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>
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