Re: [cisco-voip] SRV records for Jabber/Telepresence/MRA

[Charles Goldsmith]

Agreed 100%, customer said they'd get around to it, the priority was
getting jabber rolled out at the time to replace Lync.

On Tue, Mar 3, 2015 at 6:21 PM, NateCCIE  wrote:

> That does work, but it seems the cool service profile stuff doesn't work
> the same when you use Cuplogin.  Getting DNS for the phones should be
> pursued.
>
> Sent from my iPhone
>
> On Mar 3, 2015, at 5:55 PM, Charles Goldsmith 
> wrote:
>
> Nate, we recently had TAC have us put cuplogin back in for a 10.5 setup.
> The customer didn't have DNS on all of their phone vlan's, and we had to go
> back to IP on the CUCM server names.  Because of this, we had issues
> getting Jabber to login and not spit out cert problems.
>
> Removing the uds-cisco srv record and putting cuplogin back in resolved
> the problem.
>
> On Tue, Mar 3, 2015 at 2:44 PM, NateCCIE  wrote:
>
>> Cuplogin is deprecated and only needed with old clients. Get rid
>> Of that.
>>
>> And you are correct, you don't need the uds or Cuplogin externally.
>>
>> Xmpp Federation will not work through expressway with different internal
>> and external domains.
>>
>> Michael white has a blog out there on split domain mra.
>>
>> Sent from my iPhone
>>
>> > On Mar 3, 2015, at 2:15 PM, Andy  wrote:
>> >
>> > I seem to have got myself into a stew and can’t seem to get in my mind
>> what I need to do with SRV records.
>> > I’m trying to deploy Expressway MRA and B2B.
>> >
>> > We currently have an external DNS domain and a different internal
>> domain.
>> >
>> > For our internal domain which I’ll call andy.local we have the
>> following SRV records setup.
>> > _cisco-uds._tcp.andy.local 8443 10 10 sub1.andy.local
>> > _cisco-uds._tcp.andy.local 8443 10 10 sub2.andy.local
>> > _cuplogin._tcp.andy.local 8443 10 10 cups1.andy.local
>> >
>> > For our External domain which I’ll call andy.com
>> > _h323cs._tcp.andy.com 1720 10 10 videoconf.andy.com
>> > _h323ls._udp.andy.com 1719 10 10 videoconf.andy.com
>> > _sip._tcp.andy.com 5060 10 10 videoconf.andy.com
>> > _sip._udp.andy.com 5060 10 10 videoconf.andy.com
>> > _sips._tcp.andy.com 5061 10 10 videoconf.andy.com
>> > _turn._udp.andy.com 3478 10 10 videoconf.andy.com
>> > _collab-edge.tls.andy.com 8443 10 10 videoconf.andy.com
>> >
>> > We have an A Record for videoconf.andy.com which points to our public
>> address.
>> >
>> > As it doesn’t work I raised a TAC case and the engineer changed my
>> errors in the config, but also says I need to add _cisco-uds._tcp and
>> _cuplogin._tcp for our external domain.
>> > now I’m really confused as I thought that Jabber uses the DNS records
>> to determine which side of the firewall it was by the dns records it
>> receives when you boot the app.
>> > Also videoconf.andy.com needs to be changed to match the FQDN of the
>> VCSe or Expressway Edge server.
>> >
>> > We are running CUCM 9.1.2 with a single expressway-c and expressway-e
>> running x8.5
>> >
>> > TIA
>> >
>> > Andy
>> > andy.ca...@gmail.com
>> >
>> >
>> >
>> >
>> > ___
>> > cisco-voip mailing list
>> > cisco-voip@puck.nether.net
>> > https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>> ___
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>>
>
>
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Re: [cisco-voip] SRV records for Jabber/Telepresence/MRA

[NateCCIE]

That does work, but it seems the cool service profile stuff doesn't work the 
same when you use Cuplogin.  Getting DNS for the phones should be pursued.   

Sent from my iPhone

> On Mar 3, 2015, at 5:55 PM, Charles Goldsmith  wrote:
> 
> Nate, we recently had TAC have us put cuplogin back in for a 10.5 setup.  The 
> customer didn't have DNS on all of their phone vlan's, and we had to go back 
> to IP on the CUCM server names.  Because of this, we had issues getting 
> Jabber to login and not spit out cert problems.
> 
> Removing the uds-cisco srv record and putting cuplogin back in resolved the 
> problem.
> 
>> On Tue, Mar 3, 2015 at 2:44 PM, NateCCIE  wrote:
>> Cuplogin is deprecated and only needed with old clients. Get rid
>> Of that.
>> 
>> And you are correct, you don't need the uds or Cuplogin externally.
>> 
>> Xmpp Federation will not work through expressway with different internal and 
>> external domains.
>> 
>> Michael white has a blog out there on split domain mra.
>> 
>> Sent from my iPhone
>> 
>> > On Mar 3, 2015, at 2:15 PM, Andy  wrote:
>> >
>> > I seem to have got myself into a stew and can’t seem to get in my mind 
>> > what I need to do with SRV records.
>> > I’m trying to deploy Expressway MRA and B2B.
>> >
>> > We currently have an external DNS domain and a different internal domain.
>> >
>> > For our internal domain which I’ll call andy.local we have the following 
>> > SRV records setup.
>> > _cisco-uds._tcp.andy.local 8443 10 10 sub1.andy.local
>> > _cisco-uds._tcp.andy.local 8443 10 10 sub2.andy.local
>> > _cuplogin._tcp.andy.local 8443 10 10 cups1.andy.local
>> >
>> > For our External domain which I’ll call andy.com
>> > _h323cs._tcp.andy.com 1720 10 10 videoconf.andy.com
>> > _h323ls._udp.andy.com 1719 10 10 videoconf.andy.com
>> > _sip._tcp.andy.com 5060 10 10 videoconf.andy.com
>> > _sip._udp.andy.com 5060 10 10 videoconf.andy.com
>> > _sips._tcp.andy.com 5061 10 10 videoconf.andy.com
>> > _turn._udp.andy.com 3478 10 10 videoconf.andy.com
>> > _collab-edge.tls.andy.com 8443 10 10 videoconf.andy.com
>> >
>> > We have an A Record for videoconf.andy.com which points to our public 
>> > address.
>> >
>> > As it doesn’t work I raised a TAC case and the engineer changed my errors 
>> > in the config, but also says I need to add _cisco-uds._tcp and 
>> > _cuplogin._tcp for our external domain.
>> > now I’m really confused as I thought that Jabber uses the DNS records to 
>> > determine which side of the firewall it was by the dns records it receives 
>> > when you boot the app.
>> > Also videoconf.andy.com needs to be changed to match the FQDN of the VCSe 
>> > or Expressway Edge server.
>> >
>> > We are running CUCM 9.1.2 with a single expressway-c and expressway-e 
>> > running x8.5
>> >
>> > TIA
>> >
>> > Andy
>> > andy.ca...@gmail.com
>> >
>> >
>> >
>> >
>> > ___
>> > cisco-voip mailing list
>> > cisco-voip@puck.nether.net
>> > https://puck.nether.net/mailman/listinfo/cisco-voip
>> 
>> ___
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>> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
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Re: [cisco-voip] SRV records for Jabber/Telepresence/MRA

[Charles Goldsmith]

Nate, we recently had TAC have us put cuplogin back in for a 10.5 setup.
The customer didn't have DNS on all of their phone vlan's, and we had to go
back to IP on the CUCM server names.  Because of this, we had issues
getting Jabber to login and not spit out cert problems.

Removing the uds-cisco srv record and putting cuplogin back in resolved the
problem.

On Tue, Mar 3, 2015 at 2:44 PM, NateCCIE  wrote:

> Cuplogin is deprecated and only needed with old clients. Get rid
> Of that.
>
> And you are correct, you don't need the uds or Cuplogin externally.
>
> Xmpp Federation will not work through expressway with different internal
> and external domains.
>
> Michael white has a blog out there on split domain mra.
>
> Sent from my iPhone
>
> > On Mar 3, 2015, at 2:15 PM, Andy  wrote:
> >
> > I seem to have got myself into a stew and can’t seem to get in my mind
> what I need to do with SRV records.
> > I’m trying to deploy Expressway MRA and B2B.
> >
> > We currently have an external DNS domain and a different internal domain.
> >
> > For our internal domain which I’ll call andy.local we have the following
> SRV records setup.
> > _cisco-uds._tcp.andy.local 8443 10 10 sub1.andy.local
> > _cisco-uds._tcp.andy.local 8443 10 10 sub2.andy.local
> > _cuplogin._tcp.andy.local 8443 10 10 cups1.andy.local
> >
> > For our External domain which I’ll call andy.com
> > _h323cs._tcp.andy.com 1720 10 10 videoconf.andy.com
> > _h323ls._udp.andy.com 1719 10 10 videoconf.andy.com
> > _sip._tcp.andy.com 5060 10 10 videoconf.andy.com
> > _sip._udp.andy.com 5060 10 10 videoconf.andy.com
> > _sips._tcp.andy.com 5061 10 10 videoconf.andy.com
> > _turn._udp.andy.com 3478 10 10 videoconf.andy.com
> > _collab-edge.tls.andy.com 8443 10 10 videoconf.andy.com
> >
> > We have an A Record for videoconf.andy.com which points to our public
> address.
> >
> > As it doesn’t work I raised a TAC case and the engineer changed my
> errors in the config, but also says I need to add _cisco-uds._tcp and
> _cuplogin._tcp for our external domain.
> > now I’m really confused as I thought that Jabber uses the DNS records to
> determine which side of the firewall it was by the dns records it receives
> when you boot the app.
> > Also videoconf.andy.com needs to be changed to match the FQDN of the
> VCSe or Expressway Edge server.
> >
> > We are running CUCM 9.1.2 with a single expressway-c and expressway-e
> running x8.5
> >
> > TIA
> >
> > Andy
> > andy.ca...@gmail.com
> >
> >
> >
> >
> > ___
> > cisco-voip mailing list
> > cisco-voip@puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
>
> ___
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>
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Re: [cisco-voip] SRV records for Jabber/Telepresence/MRA

[NateCCIE]

Cuplogin is deprecated and only needed with old clients. Get rid
Of that. 

And you are correct, you don't need the uds or Cuplogin externally.

Xmpp Federation will not work through expressway with different internal and 
external domains.

Michael white has a blog out there on split domain mra. 

Sent from my iPhone

> On Mar 3, 2015, at 2:15 PM, Andy  wrote:
> 
> I seem to have got myself into a stew and can’t seem to get in my mind what I 
> need to do with SRV records.
> I’m trying to deploy Expressway MRA and B2B.
> 
> We currently have an external DNS domain and a different internal domain.
> 
> For our internal domain which I’ll call andy.local we have the following SRV 
> records setup.
> _cisco-uds._tcp.andy.local 8443 10 10 sub1.andy.local
> _cisco-uds._tcp.andy.local 8443 10 10 sub2.andy.local
> _cuplogin._tcp.andy.local 8443 10 10 cups1.andy.local
> 
> For our External domain which I’ll call andy.com
> _h323cs._tcp.andy.com 1720 10 10 videoconf.andy.com
> _h323ls._udp.andy.com 1719 10 10 videoconf.andy.com
> _sip._tcp.andy.com 5060 10 10 videoconf.andy.com
> _sip._udp.andy.com 5060 10 10 videoconf.andy.com
> _sips._tcp.andy.com 5061 10 10 videoconf.andy.com
> _turn._udp.andy.com 3478 10 10 videoconf.andy.com
> _collab-edge.tls.andy.com 8443 10 10 videoconf.andy.com
> 
> We have an A Record for videoconf.andy.com which points to our public address.
> 
> As it doesn’t work I raised a TAC case and the engineer changed my errors in 
> the config, but also says I need to add _cisco-uds._tcp and _cuplogin._tcp 
> for our external domain.
> now I’m really confused as I thought that Jabber uses the DNS records to 
> determine which side of the firewall it was by the dns records it receives 
> when you boot the app.
> Also videoconf.andy.com needs to be changed to match the FQDN of the VCSe or 
> Expressway Edge server. 
> 
> We are running CUCM 9.1.2 with a single expressway-c and expressway-e running 
> x8.5
> 
> TIA
> 
> Andy
> andy.ca...@gmail.com
> 
> 
> 
> 
> ___
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> cisco-voip@puck.nether.net
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Re: [cisco-voip] periodic restarts still required for UCCx on RHEL?

[Lelio Fulgenzi]

Thanks everyone!

Sent from my iPhone

> On Mar 3, 2015, at 1:47 PM, "Jason Aarons (AM)" 
>  wrote:
> 
> So I don’t believe in rebooting servers night to clear the cob webs.  However 
> if you have the right tools you could check memory and watch for memory leaks 
> etc.
>  
> However I like to patch/reboot at least once a year (or more often say 
> quarterly) during a maintenance window.  I like to know that a utils system 
> restart will actually work.  I’d rather testing it during a maintenance 
> window then have a data center power loss on a weekday be the testing 
> mechanism.
>  
> Everything you get will be an opinion.   Based on experience you can get 
> burned either way J  It’s really about risk and mitigating risk.  I once saw 
> a trading floor unix server that had red tape 4ft around it, getting near it 
> could cause a reboot costing millions. They were happy to never touch it.
>  
> From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
> Lelio Fulgenzi
> Sent: Tuesday, March 3, 2015 9:53 AM
> To: Cisco VOIP
> Subject: [cisco-voip] periodic restarts still required for UCCx on RHEL?
>  
>  
> 
>  
> My colleague opened up a TAC case for a problem we were having with UCCx. 
> Turned out, a component had to be restarted. The TAC engineer was surprised 
> our system was up and running for 150 days and suggested that we restart the 
> servers periodically, say every 90 days.
>  
> I thought this was going to go away with the upgrade from v7 to v9, moving 
> away from Windows' memory issues, etc.
>  
> Are people still finding they have to periodically reboot servers? Is this 
> the official stance from the TAC?
>  
> I mean, a yearly restart I can appreciate, but every 90 days?
>  
> Yeeesh.
> 
> ---
> 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
>  
> 
> 
> itevomcid
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Re: [cisco-voip] MediaSense Interfaces

[Chris Ward (chrward)]

While it doesn't help you with your current query, I can tell you that we have 
lots of roadmap items on the Search and Play interface.

The MediaSense team never intended the Search and Play application to be used 
on a regular basis. It was originally built as a reference application for 
testing the APIs. We always intended for partners to fill in the gaps. We have 
found, that customers want Search and Play to be a very functional and useful 
interface and so are roadmap is starting to reflect that now.

Stay tuned for 11.0 and 11.X and hopefully you will see some improvements. 
MediaSense probably won't ever tackle analytics or advanced features like that, 
but we do want to build MediaSense to tackle most of our customer's needs.

+Chris
TME - Unity Connection and MediaSense

-Original Message-
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
gen...@ucpenguin.com
Sent: Tuesday, March 03, 2015 10:47 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] MediaSense Interfaces

I'm looking for a recommendation on a MediaSense interface that provides access 
to call recordings.

It would need to be able to support multiple MediaSense clusters, other QM 
features aren't a requirement.

Live monitoring would be a plus, but not a requirement.

It's rather frustrating that MediaSense fails to include a reasonable user 
interface.  I get the protection of existing partner businesses, but there has 
to be a better balance than the cumbersome agent monitoring included with CAD 
and a full blown QM tool. Surely there has to be something out there I'm over 
looking that isn't $200+ per seat for these rather basic features.




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[cisco-voip] "Understanding SIP Traces" document

[Ed Leatherman]

Hello!

One of my guys has been referencing a document on the cisco support forums
called "Understanding SIP Traces":
https://supportforums.cisco.com/document/113271/understanding-sip-traces

but now we are getting "Access Denied", anyone know if this went away on
purpose? I actually never got a chance to read it, he just asked me about
it today, he seemed to find it useful.

Thanks!
Ed

-- 
Ed Leatherman
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Re: [cisco-voip] SRV records for Jabber/Telepresence/MRA

[Heim, Dennis]

Here is a lab guide from Cisco dcloud, that goes through a good portion of it.

https://dcloud-rtp-web-1.cisco.com/dCloud/docs/Documents/pa_for_mm_video_10.5_v1/PA_for_MM_Video_Lab_Guide.pdf


Dennis Heim | Emerging Technology Architect (Collaboration)
World Wide Technology, Inc. | +1 314-212-1814
[twitter]
[chat][Phone][video]
"Innovation happens on project squared" -- 
http://www.projectsquared.com

Click here to join me in my Collaboration Meeting 
Room



From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Charles Goldsmith
Sent: Tuesday, March 03, 2015 1:55 PM
To: Andy
Cc: Cisco VOIP
Subject: Re: [cisco-voip] SRV records for Jabber/Telepresence/MRA

You definitely do not want _cisco-uds and _cuplogin on the outside from my 
understanding.  
http://www.cisco.com/c/en/us/support/docs/unified-communications/expressway-series/117811-configure-vcs-00.html#anc10

Just collab-edge on the outside, plus any b2b/federation records you need.

On Tue, Mar 3, 2015 at 2:15 PM, Andy 
mailto:andy.ca...@gmail.com>> wrote:
I seem to have got myself into a stew and can’t seem to get in my mind what I 
need to do with SRV records.
I’m trying to deploy Expressway MRA and B2B.

We currently have an external DNS domain and a different internal domain.

For our internal domain which I’ll call andy.local we have the following SRV 
records setup.
_cisco-uds._tcp.andy.local 8443 10 10 sub1.andy.local
_cisco-uds._tcp.andy.local 8443 10 10 sub2.andy.local
_cuplogin._tcp.andy.local 8443 10 10 cups1.andy.local

For our External domain which I’ll call andy.com
_h323cs._tcp.andy.com 1720 10 10 
videoconf.andy.com
_h323ls._udp.andy.com 1719 10 10 
videoconf.andy.com
_sip._tcp.andy.com 5060 10 10 
videoconf.andy.com
_sip._udp.andy.com 5060 10 10 
videoconf.andy.com
_sips._tcp.andy.com 5061 10 10 
videoconf.andy.com
_turn._udp.andy.com 3478 10 10 
videoconf.andy.com
_collab-edge.tls.andy.com 8443 10 10 
videoconf.andy.com

We have an A Record for videoconf.andy.com which 
points to our public address.

As it doesn’t work I raised a TAC case and the engineer changed my errors in 
the config, but also says I need to add _cisco-uds._tcp and _cuplogin._tcp for 
our external domain.
now I’m really confused as I thought that Jabber uses the DNS records to 
determine which side of the firewall it was by the dns records it receives when 
you boot the app.
Also videoconf.andy.com needs to be changed to match 
the FQDN of the VCSe or Expressway Edge server.

We are running CUCM 9.1.2 with a single expressway-c and expressway-e running 
x8.5

TIA

Andy
andy.ca...@gmail.com




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Re: [cisco-voip] SRV records for Jabber/Telepresence/MRA

[Charles Goldsmith]

You definitely do not want _cisco-uds and _cuplogin on the outside from my
understanding.
http://www.cisco.com/c/en/us/support/docs/unified-communications/expressway-series/117811-configure-vcs-00.html#anc10


Just collab-edge on the outside, plus any b2b/federation records you need.

On Tue, Mar 3, 2015 at 2:15 PM, Andy  wrote:

> I seem to have got myself into a stew and can’t seem to get in my mind
> what I need to do with SRV records.
> I’m trying to deploy Expressway MRA and B2B.
>
> We currently have an external DNS domain and a different internal domain.
>
> For our internal domain which I’ll call andy.local we have the following
> SRV records setup.
> _cisco-uds._tcp.andy.local 8443 10 10 sub1.andy.local
> _cisco-uds._tcp.andy.local 8443 10 10 sub2.andy.local
> _cuplogin._tcp.andy.local 8443 10 10 cups1.andy.local
>
> For our External domain which I’ll call andy.com
> _h323cs._tcp.andy.com 1720 10 10 videoconf.andy.com
> _h323ls._udp.andy.com 1719 10 10 videoconf.andy.com
> _sip._tcp.andy.com 5060 10 10 videoconf.andy.com
> _sip._udp.andy.com 5060 10 10 videoconf.andy.com
> _sips._tcp.andy.com 5061 10 10 videoconf.andy.com
> _turn._udp.andy.com 3478 10 10 videoconf.andy.com
> _collab-edge.tls.andy.com 8443 10 10 videoconf.andy.com
>
> We have an A Record for videoconf.andy.com which points to our public
> address.
>
> As it doesn’t work I raised a TAC case and the engineer changed my errors
> in the config, but also says I need to add _cisco-uds._tcp and
> _cuplogin._tcp for our external domain.
> now I’m really confused as I thought that Jabber uses the DNS records to
> determine which side of the firewall it was by the dns records it receives
> when you boot the app.
> Also videoconf.andy.com needs to be changed to match the FQDN of the VCSe
> or Expressway Edge server.
>
> We are running CUCM 9.1.2 with a single expressway-c and expressway-e
> running x8.5
>
> TIA
>
> Andy
> andy.ca...@gmail.com
>
>
>
>
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[cisco-voip] SRV records for Jabber/Telepresence/MRA

[Andy]

I seem to have got myself into a stew and can’t seem to get in my mind what I 
need to do with SRV records.
I’m trying to deploy Expressway MRA and B2B.

We currently have an external DNS domain and a different internal domain.

For our internal domain which I’ll call andy.local we have the following SRV 
records setup.
_cisco-uds._tcp.andy.local 8443 10 10 sub1.andy.local
_cisco-uds._tcp.andy.local 8443 10 10 sub2.andy.local
_cuplogin._tcp.andy.local 8443 10 10 cups1.andy.local

For our External domain which I’ll call andy.com
_h323cs._tcp.andy.com 1720 10 10 videoconf.andy.com
_h323ls._udp.andy.com 1719 10 10 videoconf.andy.com
_sip._tcp.andy.com 5060 10 10 videoconf.andy.com
_sip._udp.andy.com 5060 10 10 videoconf.andy.com
_sips._tcp.andy.com 5061 10 10 videoconf.andy.com
_turn._udp.andy.com 3478 10 10 videoconf.andy.com
_collab-edge.tls.andy.com 8443 10 10 videoconf.andy.com

We have an A Record for videoconf.andy.com which points to our public address.

As it doesn’t work I raised a TAC case and the engineer changed my errors in 
the config, but also says I need to add _cisco-uds._tcp and _cuplogin._tcp for 
our external domain.
now I’m really confused as I thought that Jabber uses the DNS records to 
determine which side of the firewall it was by the dns records it receives when 
you boot the app.
Also videoconf.andy.com needs to be changed to match the FQDN of the VCSe or 
Expressway Edge server. 

We are running CUCM 9.1.2 with a single expressway-c and expressway-e running 
x8.5

TIA

Andy
andy.ca...@gmail.com




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Re: [cisco-voip] Alternative Route Pattern - Question

[Amit Kumar]

Hello Andrew,

If I am not mistaken then requirement is to enable site a to dial any
number in site a region with local gateway first, if gateway is down then
use site b gateway with appropriate digit manipulation in place.

This can be achieved with route lists, where SLRG would be first option to
route the call if that fails then move to next route group in list, that
will have digit manipulation to transform site a local number into
international format and route it via site b gateway.
On Mar 3, 2015 6:34 PM, "Andrew Grech"  wrote:

> Hi I'm thinking about a problem and don't know the best way forward.
>
> For a global deployment I'd like to setup a speed dial to a local free
> call number and send the calls out each sites local gateway. I'm unsure how
> to create a backup so that if the local gateway is down the number will be
> re-translated to a global international number and use the standard route
> lists.
>
> Example:
> Site A dials  and route pattern translates to a local free call number
> and sends out local gateway.
>
> Site B dials  and route pattern translates to local free call number
> and sends out local gateway.
>
> If Site A local gateway is down re translate  to a international
> number and dial using standard route list. Resulting in the call going out
> site B.
>
> Obviously I could of just created a global translation using the internal
> partition. However for this example assume that the local gateways should
> be used when available.
>
> Cheers
>
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>
>
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[cisco-voip] MediaSense Interfaces

[gentoo]

I'm looking for a recommendation on a MediaSense interface that provides 
access to call recordings.


It would need to be able to support multiple MediaSense clusters, other 
QM features aren't a requirement.


Live monitoring would be a plus, but not a requirement.

It's rather frustrating that MediaSense fails to include a reasonable 
user interface.  I get the protection of existing partner businesses, 
but there has to be a better balance than the cumbersome agent 
monitoring included with CAD and a full blown QM tool. Surely there has 
to be something out there I'm over looking that isn't $200+ per seat for 
these rather basic features.





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Re: [cisco-voip] errors with IOS transcoder

[Ryan Huff]

So in order to configure the transcoder, you must first initialize the DSP 
Farm. If you plan to use the dspfarm with your T1 card, then yes, you would 
need the card installed first.

If you are not going to use the dspfarm with your T1 card, then you need to 
initialize the dspfarm on the voice-card that the DSP shows up on (doing a 
"show inventory" should reveal that to you).

Thanks,

Ryan

Date: Tue, 3 Mar 2015 12:07:25 -0500
From: bhowser5...@gmail.com
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] errors with IOS transcoder

I am getting ready to add a VWIC3 T1 (and PRI) to my 2900 series ISR router 
running 15.3(2) and I'm trying to configure an IOS transcoder, I have PVDM3-64 
(2 32 modules). When I get into the transcode sub configuration it gives me an 
"Unrecognised Command" error when trying to preference the codecs. What I am 
doing wrong?

router(config)#dspfarm profile 6 transcode 
router(config-dspfarm-profile)#codec ?
% Unrecognized command
router(config-dspfarm-profile)#codec 

I've been reading that I may need to actually have the T1 card in the router 
first but I can't find anything to confirm that. I've been reading here: 
http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cminterop/configuration/15-mt/dia-15-mt-book/vc-enh-confr-vgr.html#GUID-2E7DACC4-090A-4E89-BC1B-006E79BF0BC0
 which seems to suggest that PVDM3 might not be able to support transcoding?

Can anyone shed a little light my way?

thank you


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Re: [cisco-voip] periodic restarts still required for UCCx on RHEL?

[Jason Aarons (AM)]

So I don’t believe in rebooting servers night to clear the cob webs.  However 
if you have the right tools you could check memory and watch for memory leaks 
etc.

However I like to patch/reboot at least once a year (or more often say 
quarterly) during a maintenance window.  I like to know that a utils system 
restart will actually work.  I’d rather testing it during a maintenance window 
then have a data center power loss on a weekday be the testing mechanism.

Everything you get will be an opinion.   Based on experience you can get burned 
either way ☺  It’s really about risk and mitigating risk.  I once saw a trading 
floor unix server that had red tape 4ft around it, getting near it could cause 
a reboot costing millions. They were happy to never touch it.

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Lelio 
Fulgenzi
Sent: Tuesday, March 3, 2015 9:53 AM
To: Cisco VOIP
Subject: [cisco-voip] periodic restarts still required for UCCx on RHEL?



My colleague opened up a TAC case for a problem we were having with UCCx. 
Turned out, a component had to be restarted. The TAC engineer was surprised our 
system was up and running for 150 days and suggested that we restart the 
servers periodically, say every 90 days.

I thought this was going to go away with the upgrade from v7 to v9, moving away 
from Windows' memory issues, etc.

Are people still finding they have to periodically reboot servers? Is this the 
official stance from the TAC?

I mean, a yearly restart I can appreciate, but every 90 days?

Yeeesh.
---
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



itevomcid
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Re: [cisco-voip] errors with IOS transcoder

[Brian Meade]

Shouldn't need the card in there.  Do you have "dsp services dspfarm" under
the voice-card config?

On Tue, Mar 3, 2015 at 12:07 PM, Barry Howser  wrote:

> I am getting ready to add a VWIC3 T1 (and PRI) to my 2900 series ISR
> router running 15.3(2) and I'm trying to configure an IOS transcoder, I
> have PVDM3-64 (2 32 modules). When I get into the transcode sub
> configuration it gives me an "Unrecognised Command" error when trying to
> preference the codecs. What I am doing wrong?
>
> router(config)#dspfarm profile 6 transcode
> router(config-dspfarm-profile)#codec ?
> % Unrecognized command
> router(config-dspfarm-profile)#codec
>
> I've been reading that I may need to actually have the T1 card in the
> router first but I can't find anything to confirm that. I've been reading
> here:
> http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cminterop/configuration/15-mt/dia-15-mt-book/vc-enh-confr-vgr.html#GUID-2E7DACC4-090A-4E89-BC1B-006E79BF0BC0
> which seems to suggest that PVDM3 might not be able to support transcoding?
>
> Can anyone shed a little light my way?
>
> thank you
>
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>
>
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[cisco-voip] SJC-This week

[Heim, Dennis]

Anyone else in San Jose for the Collaboration festivities with Cisco this week?

Dennis Heim | Emerging Technology Architect (Collaboration)
World Wide Technology, Inc. | +1 314-212-1814
[twitter]
[chat][Phone][video]
"Innovation happens on project squared" -- 
http://www.projectsquared.com

Click here to join me in my Collaboration Meeting 
Room



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[cisco-voip] errors with IOS transcoder

[Barry Howser]

I am getting ready to add a VWIC3 T1 (and PRI) to my 2900 series ISR router
running 15.3(2) and I'm trying to configure an IOS transcoder, I have
PVDM3-64 (2 32 modules). When I get into the transcode sub configuration it
gives me an "Unrecognised Command" error when trying to preference the
codecs. What I am doing wrong?

router(config)#dspfarm profile 6 transcode
router(config-dspfarm-profile)#codec ?
% Unrecognized command
router(config-dspfarm-profile)#codec

I've been reading that I may need to actually have the T1 card in the
router first but I can't find anything to confirm that. I've been reading
here:
http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cminterop/configuration/15-mt/dia-15-mt-book/vc-enh-confr-vgr.html#GUID-2E7DACC4-090A-4E89-BC1B-006E79BF0BC0
which seems to suggest that PVDM3 might not be able to support transcoding?

Can anyone shed a little light my way?

thank you
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Re: [cisco-voip] CUCM 11.0 Release Date?

[Matthew Loraditch]

It's due this summer. Watch for Rowan Trollope's keynote at Enterprise Connect 
later this month.

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

Facebook | Twitter | LinkedIn | G+

-Original Message-
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
gen...@ucpenguin.com
Sent: Tuesday, March 03, 2015 11:41 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] CUCM 11.0 Release Date?

Anyone have any guesses for the date when CUCM 11.0 hits CCO?

Its starting to show as the latest version in some places:

http://www.cisco.com/c/en/us/support/unified-communications/unified-communications-manager-version-11-0/model.html


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[cisco-voip] CUCM 11.0 Release Date?

[gentoo]

Anyone have any guesses for the date when CUCM 11.0 hits CCO?

Its starting to show as the latest version in some places:

http://www.cisco.com/c/en/us/support/unified-communications/unified-communications-manager-version-11-0/model.html


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Re: [cisco-voip] periodic restarts still required for UCCx on RHEL?

[Haas, Neal]

I restart once a year, we had UCCX up over 400 days and the only reason we had 
a restart was because we upgraded. That was 8.x to 9.x upgrade.


Neal Haas

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan 
LaFountain (rlafount)
Sent: Tuesday, March 03, 2015 7:01 AM
To: Lelio Fulgenzi; Cisco VOIP
Subject: Re: [cisco-voip] periodic restarts still required for UCCx on RHEL?

Hi Lelio,

Please send me the case number…. I will follow up :)

This is not the recommendation / stance of TAC.

Thank you,

Ryan LaFountain
Unified Contact Center
Cisco Services
Direct: +1 919 392 9898
Hours: M - F 9:00am - 5:00pm Eastern Time

From: Lelio Fulgenzi
Date: Tuesday, March 3, 2015 at 9:52 AM
To: Cisco VOIP
Subject: [cisco-voip] periodic restarts still required for UCCx on RHEL?


My colleague opened up a TAC case for a problem we were having with UCCx. 
Turned out, a component had to be restarted. The TAC engineer was surprised our 
system was up and running for 150 days and suggested that we restart the 
servers periodically, say every 90 days.

I thought this was going to go away with the upgrade from v7 to v9, moving away 
from Windows' memory issues, etc.

Are people still finding they have to periodically reboot servers? Is this the 
official stance from the TAC?

I mean, a yearly restart I can appreciate, but every 90 days?

Yeeesh.
---
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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Re: [cisco-voip] periodic restarts still required for UCCx on RHEL?

[Ryan LaFountain (rlafount)]

Hi Lelio,

Please send me the case number…. I will follow up :)

This is not the recommendation / stance of TAC.

Thank you,

Ryan LaFountain
Unified Contact Center
Cisco Services
Direct: +1 919 392 9898
Hours: M - F 9:00am - 5:00pm Eastern Time

From: Lelio Fulgenzi
Date: Tuesday, March 3, 2015 at 9:52 AM
To: Cisco VOIP
Subject: [cisco-voip] periodic restarts still required for UCCx on RHEL?


My colleague opened up a TAC case for a problem we were having with UCCx. 
Turned out, a component had to be restarted. The TAC engineer was surprised our 
system was up and running for 150 days and suggested that we restart the 
servers periodically, say every 90 days.

I thought this was going to go away with the upgrade from v7 to v9, moving away 
from Windows' memory issues, etc.

Are people still finding they have to periodically reboot servers? Is this the 
official stance from the TAC?

I mean, a yearly restart I can appreciate, but every 90 days?

Yeeesh.

---
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] periodic restarts still required for UCCx on RHEL?

[Lelio Fulgenzi]

My colleague opened up a TAC case for a problem we were having with UCCx. 
Turned out, a component had to be restarted. The TAC engineer was surprised our 
system was up and running for 150 days and suggested that we restart the 
servers periodically, say every 90 days. 


I thought this was going to go away with the upgrade from v7 to v9, moving away 
from Windows' memory issues, etc. 


Are people still finding they have to periodically reboot servers? Is this the 
official stance from the TAC? 


I mean, a yearly restart I can appreciate, but every 90 days? 


Yeeesh. 


--- 
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] R: SIp Phones on CUCM10.5.2 don't revert to CUCM whengoin SRST fallback

[Alessandro Bertacco]

I wait about 10 minuts before shut down the voice register global.



- Messaggio originale -
Da: "Brian Meade" 
Inviato: ‎03/‎03/‎2015 14:22
A: "Alessandro Bertacco" 
Cc: "cisco-voip@puck.nether.net" 
Oggetto: Re: [cisco-voip] SIp Phones on CUCM10.5.2 don't revert to CUCM 
whengoin SRST fallback

You can check the keepalives via packet capture or CallManager traces.


It shouldn't matter if you leave it blank.  It should use the 120second 
enterprise parameter.


How long do you let the phones try to go back to CUCM before shutting down the 
voice register global?


On Tue, Mar 3, 2015 at 3:36 AM, Alessandro Bertacco 
 wrote:

Hi Brian, thank you for the answer.
But if keepalive don't work, phones will unregister automaticalli, and it is 
not my case.

How I can check keepalive from phones?

In connection monitor duration under device pool I've leaved blank, I use the 
default value under Enterprise parameter that is set to 120.

Do you think is better to specify the value also under device pool?

Thank you very much.

Regards

Alessandro.


Da: Brian Meade
Inviato: ‎02/‎03/‎2015 21:57
A: Alessandro Bertacco
Cc: cisco-voip@puck.nether.net
Oggetto: Re: [cisco-voip] SIp Phones on CUCM10.5.2 don't revert to CUCM whengo 
in SRST fallback


Here's what you're probably hitting for the 7800 series- 
https://tools.cisco.com/bugsearch/bug/CSCus18070


For the other models, do you see the phones having successful keepalives with 
CUCM before you disable voice register global?


What do you have set for "Connection Monitor Duration" on the device pool?


On Mon, Mar 2, 2015 at 3:03 PM, Alessandro Bertacco 
 wrote:

 
Hi All,
I’ve this problem:
 
all SIP phone are impacted from my issue.
 
So, when connection to the CUCM is broken, all SIP phone correctly register on 
the SRST gateway version 15.4(3)M2 at the Branch Site, but when communication 
with CUCM go up again no one phone reconnect to the CUCM. To force that I need 
to remove "voice register global" command from the router, and wait about 2 
minutes phones come back to the CUCM.
Only SIP phone are affected from this issue. Phone used in my environment are:
7821 (sip78xx.10-2-1-12SR1-4)
9951 (sip9951.9-4-2-13)
ATA190 (1.1.2 (005) Feb 6 2015)
No Keepalive issue, because phone after removing voice register global connect 
again to the CUCM. 
 
Configuration of the SRST gateway are:
!
voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 supplementary-service h450.12
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 modem passthrough nse codec g711ulaw
 h323
 sip
  bind control source-interface GigabitEthernet0/1.20
  bind media source-interface GigabitEthernet0/1.20
  registrar server expires max 600 min 60
  no silent-discard untrusted
!
!
voice register global
 mode srst
 timeouts interdigit 7
 system message Systema SRST Attivo
 max-dn 116
 max-pool 58
!
voice register pool  1
 registration-timer max 120 min 60
 id network 192.168.101.0 mask 255.255.255.0
 dtmf-relay rtp-nte sip-notify
 voice-class codec 1
!
Can you help me?
 
Thank you regards
 
Alessandro Bertacco

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Re: [cisco-voip] SIp Phones on CUCM10.5.2 don't revert to CUCM whengo in SRST fallback

[Brian Meade]

You can check the keepalives via packet capture or CallManager traces.

It shouldn't matter if you leave it blank.  It should use the 120second
enterprise parameter.

How long do you let the phones try to go back to CUCM before shutting down
the voice register global?

On Tue, Mar 3, 2015 at 3:36 AM, Alessandro Bertacco <
bertacco.alessan...@alice.it> wrote:

> Hi Brian, thank you for the answer.
> But if keepalive don't work, phones will unregister automaticalli, and it
> is not my case.
>
> How I can check keepalive from phones?
>
> In connection monitor duration under device pool I've leaved blank, I use
> the default value under Enterprise parameter that is set to 120.
>
> Do you think is better to specify the value also under device pool?
>
> Thank you very much.
>
> Regards
>
> Alessandro.
> --
> Da: Brian Meade 
> Inviato: ‎02/‎03/‎2015 21:57
> A: Alessandro Bertacco 
> Cc: cisco-voip@puck.nether.net
> Oggetto: Re: [cisco-voip] SIp Phones on CUCM10.5.2 don't revert to CUCM
> whengo in SRST fallback
>
> Here's what you're probably hitting for the 7800 series-
> https://tools.cisco.com/bugsearch/bug/CSCus18070
>
> For the other models, do you see the phones having successful keepalives
> with CUCM before you disable voice register global?
>
> What do you have set for "Connection Monitor Duration" on the device pool?
>
> On Mon, Mar 2, 2015 at 3:03 PM, Alessandro Bertacco <
> bertacco.alessan...@alice.it> wrote:
>
>>
>>
>> Hi All,
>>
>> I’ve this problem:
>>
>>
>>
>> all SIP phone are impacted from my issue.
>>
>>
>>
>> So, when connection to the CUCM is broken, all SIP phone correctly
>> register on the SRST gateway version 15.4(3)M2 at the Branch Site, but when
>> communication with CUCM go up again no one phone reconnect to the CUCM. To
>> force that I need to remove "voice register global" command from the
>> router, and wait about 2 minutes phones come back to the CUCM.
>>
>> Only SIP phone are affected from this issue. Phone used in my environment
>> are:
>>
>> 7821 (sip78xx.10-2-1-12SR1-4)
>>
>> 9951 (sip9951.9-4-2-13)
>>
>> ATA190 (1.1.2 (005) Feb 6 2015)
>>
>> No Keepalive issue, because phone after removing voice register global
>> connect again to the CUCM.
>>
>>
>>
>> Configuration of the SRST gateway are:
>>
>> !
>> voice service voip
>>  allow-connections h323 to h323
>>  allow-connections h323 to sip
>>  allow-connections sip to h323
>>  allow-connections sip to sip
>>  supplementary-service h450.12
>>  fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
>>  modem passthrough nse codec g711ulaw
>>  h323
>>  sip
>>   bind control source-interface GigabitEthernet0/1.20
>>   bind media source-interface GigabitEthernet0/1.20
>>   registrar server expires max 600 min 60
>>   no silent-discard untrusted
>>
>> !
>>
>> !
>> voice register global
>>  mode srst
>>  timeouts interdigit 7
>>  system message Systema SRST Attivo
>>  max-dn 116
>>  max-pool 58
>> !
>> voice register pool  1
>>  registration-timer max 120 min 60
>>  id network 192.168.101.0 mask 255.255.255.0
>>  dtmf-relay rtp-nte sip-notify
>>  voice-class codec 1
>> !
>>
>> Can you help me?
>>
>>
>>
>> Thank you regards
>>
>>
>>
>> Alessandro Bertacco
>>
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>>
>>
>
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[cisco-voip] Alternative Route Pattern - Question

[Andrew Grech]

Hi I'm thinking about a problem and don't know the best way forward.

For a global deployment I'd like to setup a speed dial to a local free call
number and send the calls out each sites local gateway. I'm unsure how to
create a backup so that if the local gateway is down the number will be
re-translated to a global international number and use the standard route
lists.

Example:
Site A dials  and route pattern translates to a local free call number
and sends out local gateway.

Site B dials  and route pattern translates to local free call number
and sends out local gateway.

If Site A local gateway is down re translate  to a international number
and dial using standard route list. Resulting in the call going out site B.

Obviously I could of just created a global translation using the internal
partition. However for this example assume that the local gateways should
be used when available.

Cheers
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[cisco-voip] R: SIp Phones on CUCM10.5.2 don't revert to CUCM whengo in SRST fallback

[Alessandro Bertacco]

Hi Brian, thank you for the answer.
But if keepalive don't work, phones will unregister automaticalli, and it is 
not my case.

How I can check keepalive from phones?

In connection monitor duration under device pool I've leaved blank, I use the 
default value under Enterprise parameter that is set to 120.

Do you think is better to specify the value also under device pool?

Thank you very much.

Regards

Alessandro.

- Messaggio originale -
Da: "Brian Meade" 
Inviato: ‎02/‎03/‎2015 21:57
A: "Alessandro Bertacco" 
Cc: "cisco-voip@puck.nether.net" 
Oggetto: Re: [cisco-voip] SIp Phones on CUCM10.5.2 don't revert to CUCM whengo 
in SRST fallback

Here's what you're probably hitting for the 7800 series- 
https://tools.cisco.com/bugsearch/bug/CSCus18070


For the other models, do you see the phones having successful keepalives with 
CUCM before you disable voice register global?


What do you have set for "Connection Monitor Duration" on the device pool?


On Mon, Mar 2, 2015 at 3:03 PM, Alessandro Bertacco 
 wrote:

 
Hi All,
I’ve this problem:
 
all SIP phone are impacted from my issue.
 
So, when connection to the CUCM is broken, all SIP phone correctly register on 
the SRST gateway version 15.4(3)M2 at the Branch Site, but when communication 
with CUCM go up again no one phone reconnect to the CUCM. To force that I need 
to remove "voice register global" command from the router, and wait about 2 
minutes phones come back to the CUCM.
Only SIP phone are affected from this issue. Phone used in my environment are:
7821 (sip78xx.10-2-1-12SR1-4)
9951 (sip9951.9-4-2-13)
ATA190 (1.1.2 (005) Feb 6 2015)
No Keepalive issue, because phone after removing voice register global connect 
again to the CUCM. 
 
Configuration of the SRST gateway are:
!
voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 supplementary-service h450.12
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 modem passthrough nse codec g711ulaw
 h323
 sip
  bind control source-interface GigabitEthernet0/1.20
  bind media source-interface GigabitEthernet0/1.20
  registrar server expires max 600 min 60
  no silent-discard untrusted
!
!
voice register global
 mode srst
 timeouts interdigit 7
 system message Systema SRST Attivo
 max-dn 116
 max-pool 58
!
voice register pool  1
 registration-timer max 120 min 60
 id network 192.168.101.0 mask 255.255.255.0
 dtmf-relay rtp-nte sip-notify
 voice-class codec 1
!
Can you help me?
 
Thank you regards
 
Alessandro Bertacco

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