I'm currently running 15.3(3)M6 on our CUBE HA pairs. I'm looking at upgrading
to 15.4(3)M6, as it looks like the best fit for us. No critical security
advisories, it's acceptable in CVP/UCCE... but more importantly, it gives us
some nice options for routing on URI and other incoming numbers.
Are there any negative issues that arise from increasing the RAM on a CUCM 10.0
or CUC 10.5 VM? I haven't ever seen anyone talk about it. I assume that is
because there is typically an underlying issue that should be resolved rather
than just increasing RAM. Does anyone have any input on this?
Has anyone ever used SiteScope's web service monitor to run axl against CUCM?
I've never used SiteScope, but we have it... and I'm wanting to use it to
schedule period checks on certain things via the CUCM API. I figured if this
would work, I'd rather use it than run my own stuff via curl on
Depending on why a secondary IP is being used… I could see possibly moving that
IP to a loopback address. You could easily bind control to a Loopback in the
dial-peer(s).
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
Anthony Holloway
Sent: Friday, February 19, 2016
Or if you are like me, and didn’t install with those flags and have now screwed
up your Lync presence in Outlook (and several other things), you can follow
these steps to unscrew your installation ☺
http://www.phoney.ninja/2016/01/unscrewing-your-lync-presence-within.html
From: cisco-voip
] On Behalf Of Anthony
Holloway
Sent: Monday, February 08, 2016 9:38 AM
To: Barnett, Nick
Cc: David Jengan; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] SIP sessions
Nick,
Have you looked at the UC Gateway API? I personally, have not, so I cannot say
one way or the other.
https
I went through this with Cisco a few times and it took a while to get an
acceptable answer. It boils down to how many INVITEs your CUBE has sent for
active calls. This includes invites to media sense for media forking. An
inbound forked call to CVP will take 2 licenses. You should be able to
, that configuration will
follow the user no matter which phone they log into.
Also, I typo-ed my steering digits below... just image that they are both the
same length. 01 and 02
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
Barnett, Nick
Sent: Friday, November 06
:06 PM
To: Barnett, Nick
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] solution for multiple sip carriers?
Nick,
I’ve never done it myself but according to
http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/inbound-hdr-for-outbound.html<ht
I'm looking for some advice on how people handle situations with multiple sip
carriers. We have a mix throughout our company of DNs ported from 2 different
sip providers. If only the first provider could have ported everything, but
that didn't happen. I'm trying to NOT use a proxy as I don't
I sure could, but we are going to have over 1000 NPA-NXX combinations after
everything is deployed. I was looking for a way to not have a ton of
translations going on.
From: Scott Voll [mailto:svoll.v...@gmail.com]
Sent: Thursday, October 29, 2015 12:09 PM
To: Barnett, Nick
Cc: cisco-voip
Would this be a good place to use the Enterprise Alternate Extension number
mask on the DN? If I’m following correctly, you could possibly just put in
that field on the DNs… but depending on dial plan, you may have interdigit
timeout issues… but might be something to mess with (you can
along with a new service to track location stats.
Thanks,
Nick
From: Anthony Holloway [mailto:avholloway+cisco-v...@gmail.com]
Sent: Wednesday, September 16, 2015 2:26 PM
To: Barnett, Nick; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] CUCM 10.0.1 location out of resources fix?
Can you
This is the first time this has popped up since our upgrade from 8.6.2 to 10. I
know the location and CAC has changed with 10, but is there still a way to
resync bandwidth when CUCM thinks the link is saturated?
We aren't using any enhanced locations or anything fancy. Our locations have
Are you running a SIP CME? If so, can you post your voice service, gateway and
sip-ua configs?
sh run | sec voice service|gateway|sip-ua
having debugs of the call dropping would be beneficial as well.
Thanks,
Nick
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
To add to this, the PKID is generated using newid(), a function defined in the
Informix database.
At one point, I started digging around to see if I could actually locate the
formula used within that function, but I wasn't able to get very far. I was
able to find a reference to sqlfunctions
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