Re: [cisco-voip] Best Way To Bulk Update Lines/Directory Numbers?

[NateCCIE]

SQL is the way to go here.Super easy to dump the commands into the cli, “run sql update numplan set fkroutepartition = ‘newguid’ where dnorpattern = ‘extension’”Call forward can be set the same way, but I don’t have that memorized still. Sent from my iPhoneOn Jun 12, 2023, at 6:41 AM, Gary Parker  wrote:







Hi folks, I’m migrating users from CUCM to Teams Phone and need to update:

Route PartitionForward All Destination
for a specific set of directory numbers that cannot be identified with a search based on CUCM data. The partition change will be to identify migrated numbers, the cfwdall will be to send calls out to a voice gateway that then sends the
 call to our Teams tenant via SIP.
 
It looks like I can do the Route Partition with an export/update line appearance job, so that’s good/simple.
 
The call forward details will be different for each line. If the DN is 123456, the cfwdall will be to 901509123456, so this must be done with a custom file.
 
If I do Phones -> Export Phones -> All Details’ I can see the call forward details for the lines on those phones in there, but there doesn’t seem to be a way to import that data back in as an update, only as ‘Phones -> Insert Phones’ for
 new devices.
 
I could delete the existing phones I want to update, then Insert the modified entries back again as new phones, but I’m worried what other interactions that may break.
 
The other option seems to be to do a full ‘Export -> Device Data -> Phone’ for the database tar file, edit the required lines, then Import again, but as you have to do the whole database that is very time consuming and also, I’d imagine,
 service affecting?
 
I explored the option in the past of trying to forward calls placed to a line in the migrated partition using a transformation pattern but could not get this to work (I believe I posted about it on this list).
 
If anyone has an alternative suggestion for achieving the call forwarding I’d love to hear it.
 
Is there a way to forward calls from all the DNs to a kind of pilot number that then forwards again to a new destination based on the forwarding station? So, say for example, I forward calls from 635635 to 22, some logic on 22 forwards
 the call to 901509635635; and for calls to 222333, forward to 22, which then forwards to 901509222333
 

-- 
Gary Parker
Unified Communications Service Manager
Loughborough University, IT Services
Phone - +441509635635
Teams - g.j.par...@lboro.ac.uk
https://www.osx.ninja/pubkey.txt

 



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Re: [cisco-voip] Question about RISDB queries and ATA devices?

[nateccie]

It was broken in newer SUs of 11.5 and 12.5.  I’ve relied on this for 20 years, 
and then it just went away.  Oh well to the cloud we go.

 

From: Jason Aarons  
Sent: Thursday, May 25, 2023 12:40 PM
To: NateCCIE 
Cc: Ryan Ratliff (rratliff) ; Tim Reimers 
; Wes Sisk (wsisk) ; cisco-voip 

Subject: Re: [cisco-voip] Question about RISDB queries and ATA devices?

 

It definitely sucks that CUCM 14 SU2 broke this. It was very helpful. Should be 
an easier way to export Devices } Phone to a csv file for download via GUI. But 
there is not.

 

On Thu, May 25, 2023, 8:09 PM mailto:natec...@gmail.com> > 
wrote:

What is the goal?  I usually get what I want from device/phone, copy it all 
into the clipboard and paste without formatting into excel and continue on.  In 
older versions of CUCM you can change the rows per page, then edit the URL to 
make rows per page all of the devices on the system up to many thousands.

 

From: cisco-voip mailto:cisco-voip-boun...@puck.nether.net> > On Behalf Of Ryan Ratliff 
(rratliff)
Sent: Thursday, May 25, 2023 10:46 AM
To: Tim Reimers mailto:treim...@ashevillenc.gov> >; 
Wes Sisk (wsisk) mailto:ws...@cisco.com> >
Cc: cisco-voip mailto:cisco-voip@puck.nether.net> >
Subject: Re: [cisco-voip] Question about RISDB queries and ATA devices?

 

I’m not surprised the “phone” filter only shows you SEP devices. I was 
expecting RTMT to give you a friendlier way to browse around and find the ATAs. 

 

Do any of the other risdb CLI filters give you those devices? 

Interrogating the API directly is another option. 
https://developer.cisco.com/docs/sxml/#!risport70-api-reference/selectcmdevice

 

-Ryan

 

From: cisco-voip mailto:cisco-voip-boun...@puck.nether.net> > on behalf of Tim Reimers 
mailto:treim...@ashevillenc.gov> >
Date: Thursday, May 25, 2023 at 12:06 PM
To: Wes Sisk (wsisk) mailto:ws...@cisco.com> >
Cc: cisco-voip mailto:cisco-voip@puck.nether.net> >
Subject: Re: [cisco-voip] Question about RISDB queries and ATA devices?

Hi Wes, Ryan, all

 

I'm not seeing any of the registered ATAs showing up in RTMT under a Device 
Search either -- only SEP devices.

 

I gather that you are all expecting that 'show risdb query phone' as well as 
RTMT should be showing the ATAs registered and counted alongside the SEP devices

so long as the ATAs are running in SCCP mode and not something else...

 

Thanks Tim

 

On Thu, May 25, 2023 at 10:17 AM Wes Sisk (wsisk) mailto:ws...@cisco.com> > wrote:

Yes, registration information is in RIS not in SQL(informix). I see some other 
mentions of this, but not clear resolution. 

 

Note that ATA may follow different CM server resolution and 'show risdb' is 
per-node. Aka, have you checked all nodes with CM service activated where ATAs 
might be registered?

 

Oh, and ATAs could be h.323 for a while, so are they registering as SCCP?

 

-w

 

On May 25, 2023, at 9:50 AM, Tim Reimers mailto:treim...@ashevillenc.gov> > wrote:

 

Hi all - 

 

I'm trying to find the ACTIVELY REGISTERED devices on my UCM 9.1 system.

 

I need to find the list of actively registered ATA 186 devices and their DNs.

 

* I'm using "show risdb query phones" command, as documented here among other 
sites

https://getpractical.co.uk/2021/10/11/cisco-cucm-reports-from-sql-show-risdb/

 

That seems to show only the SEPzzz devices, aka my 79XX SCCP phones.

 

I don't see any ATA devices being returned.

Are they not in the "phone" table of the RISDB?

 

Thanks, Tim

 

* my understanding is that any variation on the "run sql select" is

simply querying the Oracle? database for _configured_ devices only, and isn't 
looking 

at the memory table of the Callmanager process to see the _currently 
registered_ devices.

(I've seen a number of other forum posts where people suggested "run sql" 
commands to gather info, but that is statically configured, not necessarily 
"registered", so that does not seem appropriate for the info I want).

 




 

-- 

Quis custodiet  
<https://en.wikipedia.org/wiki/Quis_custodiet_ipsos_custodes%3F> ipsos nexus

Tim Reimers

Network Administrator

I.T Services

City of Asheville 

treim...@ashevillenc.gov <mailto:treim...@ashevillenc.gov> 

(desk) 828-259-5512

(cell)   828-552-1585

"That’s no ordinary rabbit  packet!  That’s the most foul, cruel, and bad 
tempered badly framed packet you ever set eyes on. Listen, that packet’s got a 
vicious streak a mile wide, he’s a killer.He’s got huge sharp MTU…eh, he can 
leap about and cross Vlans…. I warned you, I warned you but did you listen? No… 
ohhh no, it’s just a harmless little packet on the network, isn’t it now"

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https://puck.nether.net/mailman/listinfo/cisco-voip

 




 

-- 

Quis 

Re: [cisco-voip] Question about RISDB queries and ATA devices?

[nateccie]

Just add the filter on the device/phone page to include directory number and 
you’ll be good to go.

 

From: Tim Reimers  
Sent: Thursday, May 25, 2023 12:27 PM
To: natec...@gmail.com
Cc: Ryan Ratliff (rratliff) ; Wes Sisk (wsisk) 
; cisco-voip 
Subject: Re: [cisco-voip] Question about RISDB queries and ATA devices?

 

Sorry, sent before being ready...

 

What I was hoping for was a list of all the registered active ATAs that would 
INCLUDE the DN on them.

As it is -- if I simply use the admin website, as you said, I can get a list of 
all the devices that are registered.

 

I've actually got the individual ATAs downloaded that way, and sorted by 
"Registered" in a Google Sheet.

 

I cannot easily also match those up with the DNs.

I suppose one could probably then write some sort of SQL command that would 
specify a long-ish list of ATA device names to query for DN.

 

It may be time to just ask an Intern to click on each ATA that's registered and 
collect the DN and description.

 

I thought about asking one of our DB people to do that -- 

But they'd want a project set up, and require a complete list of the table 
schema, the IP/TCP port and data connection, etc

to do things as they're used to, with a direct SQL connection to query against.

They're not used to having to operate the way Cisco does with CLI only access 
to SQL in a limited fashion. 

And I get why Cisco does that... not arguing that point.

 

 

 

 

On Thu, May 25, 2023 at 2:18 PM Tim Reimers mailto:treim...@ashevillenc.gov> > wrote:

Nate, I may have to do that.

 

I just thought I was crazy that ATAs don't show up in 

show risdb query phone

 

T

 

On Thu, May 25, 2023 at 2:08 PM mailto:natec...@gmail.com> 
> wrote:

What is the goal?  I usually get what I want from device/phone, copy it all 
into the clipboard and paste without formatting into excel and continue on.  In 
older versions of CUCM you can change the rows per page, then edit the URL to 
make rows per page all of the devices on the system up to many thousands.

 

From: cisco-voip mailto:cisco-voip-boun...@puck.nether.net> > On Behalf Of Ryan Ratliff 
(rratliff)
Sent: Thursday, May 25, 2023 10:46 AM
To: Tim Reimers mailto:treim...@ashevillenc.gov> >; 
Wes Sisk (wsisk) mailto:ws...@cisco.com> >
Cc: cisco-voip mailto:cisco-voip@puck.nether.net> >
Subject: Re: [cisco-voip] Question about RISDB queries and ATA devices?

 

I’m not surprised the “phone” filter only shows you SEP devices. I was 
expecting RTMT to give you a friendlier way to browse around and find the ATAs. 

 

Do any of the other risdb CLI filters give you those devices? 

Interrogating the API directly is another option. 
https://developer.cisco.com/docs/sxml/#!risport70-api-reference/selectcmdevice

 

-Ryan

 

From: cisco-voip mailto:cisco-voip-boun...@puck.nether.net> > on behalf of Tim Reimers 
mailto:treim...@ashevillenc.gov> >
Date: Thursday, May 25, 2023 at 12:06 PM
To: Wes Sisk (wsisk) mailto:ws...@cisco.com> >
Cc: cisco-voip mailto:cisco-voip@puck.nether.net> >
Subject: Re: [cisco-voip] Question about RISDB queries and ATA devices?

Hi Wes, Ryan, all

 

I'm not seeing any of the registered ATAs showing up in RTMT under a Device 
Search either -- only SEP devices.

 

I gather that you are all expecting that 'show risdb query phone' as well as 
RTMT should be showing the ATAs registered and counted alongside the SEP devices

so long as the ATAs are running in SCCP mode and not something else...

 

Thanks Tim

 

On Thu, May 25, 2023 at 10:17 AM Wes Sisk (wsisk) mailto:ws...@cisco.com> > wrote:

Yes, registration information is in RIS not in SQL(informix). I see some other 
mentions of this, but not clear resolution. 

 

Note that ATA may follow different CM server resolution and 'show risdb' is 
per-node. Aka, have you checked all nodes with CM service activated where ATAs 
might be registered?

 

Oh, and ATAs could be h.323 for a while, so are they registering as SCCP?

 

-w

 

On May 25, 2023, at 9:50 AM, Tim Reimers mailto:treim...@ashevillenc.gov> > wrote:

 

Hi all - 

 

I'm trying to find the ACTIVELY REGISTERED devices on my UCM 9.1 system.

 

I need to find the list of actively registered ATA 186 devices and their DNs.

 

* I'm using "show risdb query phones" command, as documented here among other 
sites

https://getpractical.co.uk/2021/10/11/cisco-cucm-reports-from-sql-show-risdb/

 

That seems to show only the SEPzzz devices, aka my 79XX SCCP phones.

 

I don't see any ATA devices being returned.

Are they not in the "phone" table of the RISDB?

 

Thanks, Tim

 

* my understanding is that any variation on the "run sql select" is

simply querying the Oracle? database for _configured_ devices only, and isn't 
looking 

at the memory table of the Callmanager process to see the _currently 
registered_ devices.

(I've seen a number of other forum posts where people suggested "run sql" 
commands to gather info, but that is statically configured, 

Re: [cisco-voip] Question about RISDB queries and ATA devices?

[nateccie]

What is the goal?  I usually get what I want from device/phone, copy it all 
into the clipboard and paste without formatting into excel and continue on.  In 
older versions of CUCM you can change the rows per page, then edit the URL to 
make rows per page all of the devices on the system up to many thousands.

 

From: cisco-voip  On Behalf Of Ryan Ratliff 
(rratliff)
Sent: Thursday, May 25, 2023 10:46 AM
To: Tim Reimers ; Wes Sisk (wsisk) 
Cc: cisco-voip 
Subject: Re: [cisco-voip] Question about RISDB queries and ATA devices?

 

I’m not surprised the “phone” filter only shows you SEP devices. I was 
expecting RTMT to give you a friendlier way to browse around and find the ATAs. 

 

Do any of the other risdb CLI filters give you those devices? 

Interrogating the API directly is another option. 
https://developer.cisco.com/docs/sxml/#!risport70-api-reference/selectcmdevice

 

-Ryan

 

From: cisco-voip mailto:cisco-voip-boun...@puck.nether.net> > on behalf of Tim Reimers 
mailto:treim...@ashevillenc.gov> >
Date: Thursday, May 25, 2023 at 12:06 PM
To: Wes Sisk (wsisk) mailto:ws...@cisco.com> >
Cc: cisco-voip mailto:cisco-voip@puck.nether.net> >
Subject: Re: [cisco-voip] Question about RISDB queries and ATA devices?

Hi Wes, Ryan, all

 

I'm not seeing any of the registered ATAs showing up in RTMT under a Device 
Search either -- only SEP devices.

 

I gather that you are all expecting that 'show risdb query phone' as well as 
RTMT should be showing the ATAs registered and counted alongside the SEP devices

so long as the ATAs are running in SCCP mode and not something else...

 

Thanks Tim

 

On Thu, May 25, 2023 at 10:17 AM Wes Sisk (wsisk) mailto:ws...@cisco.com> > wrote:

Yes, registration information is in RIS not in SQL(informix). I see some other 
mentions of this, but not clear resolution. 

 

Note that ATA may follow different CM server resolution and 'show risdb' is 
per-node. Aka, have you checked all nodes with CM service activated where ATAs 
might be registered?

 

Oh, and ATAs could be h.323 for a while, so are they registering as SCCP?

 

-w

 

On May 25, 2023, at 9:50 AM, Tim Reimers mailto:treim...@ashevillenc.gov> > wrote:

 

Hi all - 

 

I'm trying to find the ACTIVELY REGISTERED devices on my UCM 9.1 system.

 

I need to find the list of actively registered ATA 186 devices and their DNs.

 

* I'm using "show risdb query phones" command, as documented here among other 
sites

https://getpractical.co.uk/2021/10/11/cisco-cucm-reports-from-sql-show-risdb/

 

That seems to show only the SEPzzz devices, aka my 79XX SCCP phones.

 

I don't see any ATA devices being returned.

Are they not in the "phone" table of the RISDB?

 

Thanks, Tim

 

* my understanding is that any variation on the "run sql select" is

simply querying the Oracle? database for _configured_ devices only, and isn't 
looking 

at the memory table of the Callmanager process to see the _currently 
registered_ devices.

(I've seen a number of other forum posts where people suggested "run sql" 
commands to gather info, but that is statically configured, not necessarily 
"registered", so that does not seem appropriate for the info I want).

 




 

-- 

Quis custodiet  
 ipsos nexus

Tim Reimers

Network Administrator

I.T Services

City of Asheville 

treim...@ashevillenc.gov  

(desk) 828-259-5512

(cell)   828-552-1585

"That’s no ordinary rabbit  packet!  That’s the most foul, cruel, and bad 
tempered badly framed packet you ever set eyes on. Listen, that packet’s got a 
vicious streak a mile wide, he’s a killer.He’s got huge sharp MTU…eh, he can 
leap about and cross Vlans…. I warned you, I warned you but did you listen? No… 
ohhh no, it’s just a harmless little packet on the network, isn’t it now"

___
cisco-voip mailing list
cisco-voip@puck.nether.net  
https://puck.nether.net/mailman/listinfo/cisco-voip

 




 

-- 

Quis custodiet  
 ipsos nexus

Tim Reimers

Network Administrator

I.T Services

City of Asheville 

treim...@ashevillenc.gov  

(desk) 828-259-5512

(cell)   828-552-1585

"That’s no ordinary rabbit  packet!  That’s the most foul, cruel, and bad 
tempered badly framed packet you ever set eyes on. Listen, that packet’s got a 
vicious streak a mile wide, he’s a killer.He’s got huge sharp MTU…eh, he can 
leap about and cross Vlans…. I warned you, I warned you but did you listen? No… 
ohhh no, it’s just a harmless little packet on the network, isn’t it now"

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Re: [cisco-voip] CUCM/IM upgrade to v14 - VM Hardware Version Change

[nateccie]

The OS should be detected and updated by vmware tools.  It's super important
to keep your vmware tools up to date.

 

From: cisco-voip  On Behalf Of
Ayoub,Gregory
Sent: Wednesday, May 17, 2023 6:52 PM
To: Sean Riley ; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] CUCM/IM upgrade to v14 - VM Hardware Version
Change

 

Before.

 

From: cisco-voip mailto:cisco-voip-boun...@puck.nether.net> > On Behalf Of Sean Riley
Sent: Friday, May 12, 2023 11:13 AM
To: cisco-voip@puck.nether.net  
Subject: [cisco-voip] CUCM/IM upgrade to v14 - VM Hardware Version Change

 


[External Email] 

I am planning my upgrade from v11.5 SU9 to v14 SU2 in the coming weeks.
Looks like I need to update my vm's to hardware version 13 and OS to CentOS
7 (64-bit).  My question is around the timing of this change to the vm.
Should this be done before the upgrade to 14 or after?  

 

Thank you.

 

Sean

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Re: [cisco-voip] [External] Re: No Custom Background Images after CUCM 14

[NateCCIE]

I did a PCD upgrade years ago to 11.5 and the file permissions were messed up on the tftp directory and TAC rooted in and had to adjust them. Seems crazy that bug would still be around after all of these years. Sent from my iPhoneOn Mar 27, 2023, at 11:01 AM, Adam Pawlowski  wrote:







If trust is broken, I think the phones will not load signed files like the ringlist or background selection, if that is what is going on
 
It doesn’t make any sense that _deleting_ files would fix this
 

Adam Pawlowski
Network Engineer | Network and Communication Services
University at Buffalo Information Technology (UBIT) 
243 Computing Center, Buffalo, NY 14260 


 



From: cisco-voip 
On Behalf Of Bill Talley
Sent: Monday, March 27, 2023 10:58 AM
To: Hunter Fuller 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] [External] Re: No Custom Background Images after CUCM 14


 
Do you have a reference link for that?  I’ve never experienced that, let alone read that in upgrade documentation.   I’ve had that problem with earlier versions of PCD but never on a direct standard or refresh upgrade. 

 


The upgrade documentation says custom files aren’t contained in DRS backup files.  Maybe I’ve just gotten lucky and the background images have all be accidentally copied 路‍♂️

Sent from my iPhone with bery tiny touchscreeb input keys, please excude my typtos.






On Mar 27, 2023, at 9:35 AM, Hunter Fuller  wrote:






Cisco’s upgrade procedure indicates that custom tftp files are never expected to survive an upgrade. 


 


I did notice the files not deleting on my system when I went to replace them after upgrade to 12.5, however, you can upload “on top of” the existing files then delete the .sgn and that seems to work. Unless that changed in 14. 


 


On Mon, Mar 27, 2023 at 09:30 Bill Talley  wrote:



I assume you’re saying the images don’t appear under Wallpapers on the phones?  Have you verified the appropriate Desktops directory on the tftp server?  I assume yes to these questions below since TAC appears to directing you towards the
 sgn files.  Explanation for these questions is below. 

 


- List.xml is present and contains all of the correct file names/references?


- Background images and thumbnail files are present?


 


I recently did a standard upgrade from cucm 12.5 to 14Su2 and on the pub, the upgrade didn’t copy the custom images to the new partition and List.xml was replaced with the default file so only the default images appeared under wallpapers.
    On the sub, the List.xml was copied during the upgrade, but the images were not.   I ended up downloading the original files from the tftp directory on the  inactive  partition the uploading them via OS Admin and restarting tftp.   This happened for two
 different phone models but not all phone models. 


 

 


 



Sent from my iPhone with bery tiny touchscreeb input keys, please excude my typtos.







On Mar 27, 2023, at 7:26 AM, JASON BURWELL  wrote:




 










After we upgraded to CUCM 14 with PCD, our 88XX phones no longer have custom wallpaper images available. The solution from TAC was to delete all the tftpsgn files from the TFTP
 directory on each node and restart tftp services however when we delete the files using the GUI or CLI, they remain and do not delete. TAC has tried twice unsuccessfully to access the directory from root and delete the files but they have not been able to
 do that. Surely others have encountered this issue and I’m wondering what everyone else has done.  I can’t believe it’s really this complicated of an issue to resolve. Thanks Jason
 

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-- 






--
Hunter Fuller (they)
Router Jockey
VBH M-1C
+1 256 824 5331

Office of Information Technology
The University of Alabama in Huntsville
Network Engineering












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Re: [cisco-voip] Specific Off-net Called Numbers Being Dropped When Called From Jabber

[nateccie]

voice-classs sip audio forced is cleaner, but as you said that needs a code
upgrade.  I would rock this until then.

 

 

voice service voip

sip

  sip-profiles inbound

 

voice class sip-profiles 15

request ANY sdp-header Video-Attribute remove 

 request ANY sdp-header Video-Media modify "m=video(.*)" "" 

 request ANY sdp-header Video-Bandwidth-Info remove 

 request ANY sdp-header Video-Session-Info remove 

 request ANY sdp-header Video-Connection-Info remove

 

dial-peer from CUCM

voice-class sip profiles 15 inbound

 

From: cisco-voip  On Behalf Of Gary
Parker
Sent: Friday, January 27, 2023 5:43 AM
To: voip puck ; Telecommunication Managers

Subject: [cisco-voip] Specific Off-net Called Numbers Being Dropped When
Called From Jabber

 

Hi folks, we're using CUCM 12.5.1.14900-63, and CUBE on a pair of 2921
routers running IOS 15.5(3)M2. We've been using this configuration to route
inbound and outbound PSTN calls via our TSP in the UK, Gamma, for a few
years now with no major problems.

 

Recently, however, we changed our travel booking agent to a new company
called Clarity BT and have found that none of our Cisco Jabber softphone
users can call their number, 03330100045, which appears to be hosted with a
company called Redcentric. Calls from any of our Cisco deskphones, and a
small volume of users we have on MS Teams Voice, who also route out via
Gamma using Direct Routing, can connect with no problems, as do calls from
our mobiles.

 

The calls from Jabber fail within a second or two of being placed, with no
message or tone. Looking at the SIP traces, they're rejected with "403
Forbidden"/" Reason: Q.850;cause=57", the CDR records this as a destCause of
57,  "Bearer capability not authorized"

 

I've raised a support case with Gamma and they're focussing on the fact that
calls from Jabber clients appear to be including SDP video information in
the call setup and have asked me if it's possible to stop Jabber sending
this. I've set my Jabber client to not "Always start my calls with video",
but this didn't change anything, and it's notable that I'm successfully
placing calls to the problem number from a Cisco 8865 handset that is also
video enabled and sends similar video SDP information.

 

It's worth mentioning that the SIP sessions are all sending the bare minimum
of g711ulaw, g711alaw and g729 annex b along with whatever else the device
is capable of, so it's not like it's failing to negotiate and audio codec.

 

So, has anyone had similar experience and know a solution? I tried looking
for ways to filter out the SDP video stuff at the CUBE, but my Google-fu
failed me (although I think this is a red herring due to the 8865 always
connecting.

 

Below is a capture of a failed call from a Jabber client to the problem
number with the IP addresses obfuscated:

Jan 23 11:06:03.383: //-1//SIP/Msg/ccsipDisplayMsg:

Received: 

INVITE sip:903330100...@xxx.xxx.xxx.xxx:5060 SIP/2.0

Via: SIP/2.0/TCP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK1ec5d7b7426e8

From: "Gary Parker" sip:+441509635635@ 
xxx.xxx.xxx.xxx ;tag=1460625~6c2496f4-28ae-4afc-bfa9-0620307b8c3e-103494796

To: sip:903330100045@   xxx.xxx.xxx.xxx 

Date: Mon, 23 Jan 2023 11:06:03 GMT

Call-ID: ebb22500-1ee1b910-1cb56-87a27d9e@
  xxx.xxx.xxx.xxx 

Supported: timer,resource-priority,replaces

Min-SE:  1800

User-Agent: Cisco-CUCM12.5

Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
SUBSCRIBE, NOTIFY

CSeq: 101 INVITE

Expires: 180

Allow-Events: presence, kpml

Supported: X-cisco-srtp-fallback,X-cisco-original-called

Call-Info: sip: 
xxx.xxx.xxx.xxx:5060;method="NOTIFY;Event=telephone-event;Duration=500"

Call-Info:
;x-cisco-video-traffic-class=DESKTOP;x-cisco-
qos-tcl=true

Session-ID:
7b390f7500105000a000a860b63b96d1;remote=

Cisco-Guid: 3954320640-065536-002092-2275573150

Session-Expires:  1800

P-Asserted-Identity: "Gary Parker" sip:+441509635635@
  xxx.xxx.xxx.xxx 

Remote-Party-ID: "Gary Parker" sip:+441509635635@
  xxx.xxx.xxx.xxx
;party=calling;screen=yes;privacy=off

Contact: sip:+441509635635@

xxx.xxx.xxx.xxx:5060;transport=tcp;video;audio;+u.sip!devicename.ccm.cisco.c
om="JFWCCGJP";bfcp

Max-Forwards: 69

Content-Type: application/sdp

Content-Length: 1583

v=0

o=CiscoSystemsCCM-SIP 1460625 1 IN IP4 xxx.xxx.xxx.xxx

s=SIP Call

c=IN IP4 xxx.xxx.xxx.xxx

b=TIAS:3968000

b=AS:3968

t=0 0

a=cisco-mari:v1

a=cisco-mari-rate

m=audio 22946 RTP/AVP 114 9 104 105 0 8 18 111 101

a=extmap:14/sendrecv http://protocols.cisco.com/timestamp#100us

a=rtpmap:114 opus/48000/2

a=rtpmap:9 G722/8000

a=rtpmap:104 G7221/16000

a=fmtp:104 bitrate=32000

a=rtpmap:105 G7221/16000

a=fmtp:105 bitrate=24000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:111 X-ULPFECUC/8000

a=fmtp:111  max_esel=1420;m=8;max_n=32;FEC_ORDER=FEC_SRTP

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

Re: [cisco-voip] Extension mobility boot/register to prompt

[nateccie]

Idle URL to the EM service.

 

From: cisco-voip  On Behalf Of Matthew
Huff
Sent: Tuesday, November 22, 2022 8:51 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Extension mobility boot/register to prompt

 

I have extension mobility setup currently with a fake DN so that the phone
will register. If you hit service, you can login and extension mobility
works, but is there anyway to make the phone when it boots up come up with
the login prompt?

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Re: [cisco-voip] CUCM SIP trunk redundancy with multiple CUBES

[NateCCIE]

Split it to two different trunks and add them both to a route group that is 
configured for round robin to the route list.



> On Mar 10, 2022, at 8:13 AM, Matthew Huff  wrote:
> 
> 
> We have two cisco ISR 4331 Cube gateways in two different locations. I want 
> to be able to route calls to both devices (preferably round-robin). I have 
> the router pattern going to a trunk with both cubes defined (with sip options 
> keep-alive configured). The issue we are having is that if the call is made 
> to CUBE1 and the associated outbound dial-peer in in busyout, the CUBE 
> returns 503  Service unavailable and CUCM doesn’t try CUBE2. What am I 
> missing?
>  
> Matthew Huff | Director of Technical Operations | OTA Management LLC
>  
> Office: 914-460-4039
> mh...@ox.com | www.ox.com
> ...
>  
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Re: [cisco-voip] [External] Re: Small business E911 solution

[NateCCIE]

CUCM/CER is not complaint with Ray Baum’s act Phase II.  You will need to 
deploy Intrado or Redsky to be compliant.  I’ve only looked at them for 
existing CER customers, but I believe at least redsky has an option without 
CER.  Don’t worry you have less than a month until Phase II goes into effect on 
January 6th 2022.

 

https://www.fcc.gov/mlts-911-requirements

 

 

From: cisco-voip  On Behalf Of Johnson, Tim
Sent: Thursday, December 9, 2021 12:20 PM
To: Matthew Huff 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] [External] Re: Small business E911 solution

 

Maybe Intrado? Not sure their minimum requirements but I know we started with 
them when they were West, with only a couple hundred DIDs. 

 

On Dec 9, 2021 2:16 PM, Matthew Huff mailto:mh...@ox.com> > 
wrote:

No, hosted solution isn’t an option as we have a number of custom solutions 
like ring downs, etc…

 

We already have CUCM and Expressway working fine, I just need directions on the 
simplest solution for E911 for MRA workers.

 

Matthew Huff | Director of Technical Operations | OTA Management LLC

 

Office: 914-460-4039

mh...@ox.com   | www.ox.com  

...

 

From: Matthew Loraditch mailto:mloradi...@heliontechnologies.com> > 
Sent: Thursday, December 9, 2021 2:00 PM
To: Matthew Huff mailto:mh...@ox.com> >; 
cisco-voip@puck.nether.net  
Subject: RE: Small business E911 solution

 

I’m very curious if you find something. I’m not aware of anything cost 
effective at your size. RedSky’s minimum purchase for a CUCM based system is 
12-14k.

 

Have you looked at moving to a hosted phone system? Almost every vendor I’m 
aware of includes E911 therein

 



 



Matthew Loraditch​



Sr. Network Engineer


(He/Him/His)




p:   443.541.1518



w:   www.heliontechnologies.com

 | 

e:   mloradi...@heliontechnologies.com



  






  


  


  

From: cisco-voip mailto:cisco-voip-boun...@puck.nether.net> > On Behalf Of Matthew Huff
Sent: Thursday, December 9, 2021 1:34 PM
To: cisco-voip@puck.nether.net  
Subject: [cisco-voip] Small business E911 solution

 

[EXTERNAL]

 

We are in the process of moving from legacy ISDN PRI for inbound/outbound 
dialing to SIP, and E911 has hit us in the face. We have less than 50 users, 
where > 90% currently are working from home. They have the same prime dn for 
both the office phone and their home phone. We have users that have phones in 
3-4 locations including in multiple states. What is the simplest solution to 
setup and maintain that doesn’t require a user to have a separate DID in each 
location? Cisco Emergency Responder looks like major overkill. 

 

Our environment is:

CUCM 14.x

Cisco Expressway 14.x for MRA

Cisco 8861 SIP phones (both at home and at work).

 

Matthew Huff | Director of Technical Operations | OTA Management LLC

 

Office: 914-460-4039

mh...@ox.com   | www.ox.com  

...

 

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Re: [cisco-voip] [External] Jabber Users Prompted To Accept Webex Cert

[NateCCIE]

https://bst.cloudapps.cisco.com/bugsearch/bug/CSCvq73203

-Original Message-
From: cisco-voip  On Behalf Of Gary Parker
Sent: Thursday, November 11, 2021 1:45 PM
To: Johnson, Tim 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] [External] Jabber Users Prompted To Accept Webex Cert

Quick follow-up: I’ve heard from another site (off-list) suffering this now, 
too. 

Gary

> On 11 Nov 2021, at 16:13, Gary Parker  wrote:
> 
> Thanks Tim, likewise: glad it’s not just us!
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Re: [cisco-voip] Ported to SIP carrier - some calls still coming in old carrier

[NateCCIE]

A lot of the time the loosing carrier doesn't remove internal translations,
so other customers of the losing carrier still come into the old PRI.
You'll need to ask them to clean it up, I don't think I have been patient
enough to see it be cleaned up by itself.

 

-Nate

 

From: cisco-voip  On Behalf Of Riley,
Sean
Sent: Thursday, July 8, 2021 10:34 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Ported to SIP carrier - some calls still coming in old
carrier

 

We ported about 700 DID's yesterday, from our PRI carrier to a new SIP
carrier. Everything is working great, except I still see some calls coming
into the old H323 gateways connected to the old carrier.  If I make a test
call from my cell phone to the called number seen on the old gateway, the
call routes through our new SIP gateway as expected.  Observing over the
past 2 days it does seem that some of the calls are from the same "calling
ID" and number.  

 

Could it just take time for the port to propagate to other carriers and this
will eventually work itself out over the coming days?  Or could there be
something else causing calls from specific callers to still route through
our old PRI carrier?

 

Thanks for any advice or knowledge you can share on this.

 

Sean.  

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Re: [cisco-voip] Roomkit plus ceiling mics

[nateccie]

Not what you want, and it's MRC, but this seems like the perfect use case

https://help.webex.com/en-us/nffx8kj/Deploy-the-Cisco-Webex-Video-Integration-for-Microsoft-Teams

-Original Message-
From: cisco-voip  On Behalf Of Myron Young
Sent: Wednesday, February 24, 2021 10:22 AM
To: Cisco VoIP Group 
Subject: [cisco-voip] Roomkit plus ceiling mics

Hello,

Has anyone had any experience or ideas on how we could use the roomkit plus 
ceiling mics to connect them to a PC? 

Scenario is that we use MS Teams in our environment and people join the meeting 
from conference rooms with those endpoints, but joining MS teams meetings 
without knowing to call in causes difficulties. So I’m trying to figure out an 
easy way to allow use of the ceiling mics as a PC mic to join these meetings 
with computer audio. 

Any ideas are welcome

Thanks
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Re: [cisco-voip] [External] Re: CUCM 12.5 issue with restore

[NateCCIE]

I feel like it’s old files that gum up the works. Like you’re set to save 3 
copies and you have some files from last year.

I also see It break stuff with old files where the backup looks like it’s ok, 
but reports failed. Clean up the old stuff and it runs fine again.  

Sent from my iPhone

> On Feb 23, 2021, at 2:00 PM, Hunter Fuller via cisco-voip 
>  wrote:
> 
> Surely "try again with an empty directory" defeats the purpose if the
> goal is to restore a backup, no?
> 
> If the directory has to be empty, it isn't much of a backup - right?
> 
> --
> Hunter Fuller (they)
> Router Jockey
> VBH Annex B-5
> +1 256 824 5331
> 
> Office of Information Technology
> The University of Alabama in Huntsville
> Network Engineering
> 
>> On Tue, Feb 23, 2021 at 9:19 AM Lelio Fulgenzi  wrote:
>> 
>> 
>> 
>> I have found, and I didn’t think of this earlier, that sometimes a clean 
>> directory works much better. It has to do with the XML files that remain 
>> explaining what is supposed to be in there.
>> 
>> 
>> 
>> I didn’t notice this as much with the core apps as I did with the express 
>> apps, like CUE. IT drove me nuts for weeks.
>> 
>> 
>> 
>> So, yes, I too have “try a new, empty directory” on my list of things to try 
>> (in my head which I forget all the time).
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> From: cisco-voip  On Behalf Of Andy Carse
>> Sent: Tuesday, February 23, 2021 5:10 AM
>> To: Wes Sisk (wsisk) 
>> Cc: Cisco VoIP List 
>> Subject: Re: [cisco-voip] CUCM 12.5 issue with restore
>> 
>> 
>> 
>> CAUTION: This email originated from outside of the University of Guelph. Do 
>> not click links or open attachments unless you recognize the sender and know 
>> the content is safe. If in doubt, forward suspicious emails to 
>> ith...@uoguelph.ca
>> 
>> 
>> 
>> So I’m now confused, by changing the backup directory to a new directory on 
>> our work system I can now backup and restore.
>> 
>> It still doesn’t like to restore from the original location but will backup 
>> to it.
>> 
>> The permissions are the same for both. The def logs don’t show any issues.
>> 
>> So I’ll add it to our list of things to check in the future.
>> 
>> Case closed
>> 
>> 
>> 
>> 
>> 
>> On Sun, 21 Feb 2021 at 14:21, Andy Carse  wrote:
>> 
>> So,
>> 
>> I configured a new location for the backups to be stored (a subdirectory on 
>> the original)
>> 
>> did a few backups to it from the cluster no problems as usual.
>> 
>> ran the restore wizard it found the recent backups no issues, I even ran a 
>> backup just to make sure as its my lab environment.
>> 
>> then I copied the contents of the original backup directory into the new 
>> location.
>> 
>> Ran the restore wizard expecting it to time out, but no it found the 
>> relevant files.
>> 
>> So I'm a bit stumped.
>> 
>> 
>> 
>> Now as this is my Lab its not exactly the same as the work environment 
>> (network infrastructure etc) So I will raise a change control Monday so I'll 
>> see if I can replicate the issue in Production.
>> 
>> Just to be clear the original issue is with running the Restore Wizard and 
>> not being able to see the backup just taken.
>> 
>> So it's not a cop file mismatch, different versions of the cluster, it's not 
>> file permissions.
>> 
>> So I suggest that don't take it for granted that if you can backup your 
>> cluster that you will be able to restore should you need to, you need to 
>> test it periodically.
>> 
>> Any how I'll update with the outcome.
>> 
>> 
>> 
>> Rgds Andy
>> 
>> 
>> 
>> 
>> 
>> On Sat, 20 Feb 2021 at 14:19, Wes Sisk (wsisk)  wrote:
>> 
>> Andy, sounds like a good start:
>> 
>> https://community.cisco.com/t5/ip-telephony-and-phones/disaster-recovery-problem/td-p/2767755
>> 
>> 
>> 
>> I see 2 other situations that might be relevant:
>> 
>> 1. All the same .cop files not installed
>> 
>> 2. Attempting to restore a backup of a different version
>> 
>> 
>> 
>> 
>> 
>> -Wes
>> 
>> 
>> 
>> On Feb 19, 2021, at 6:46 PM, Andy Carse  wrote:
>> 
>> 
>> 
>> Wes,
>> 
>> I select Restore Wizard then select the backup device
>> 
>> click next
>> 
>> The Ccx then spins the hour glass for 5 mins then says
>> 
>> “Restore request timing out. Either master agent is down or Sftp server is 
>> inaccessible or too slow to respond”
>> 
>> 
>> 
>> It’s the same location all the other UC apps backup to. Could it be that 
>> there are too many files in the directory?
>> 
>> Even though they would have different names etc?
>> 
>> 
>> 
>> It seems to do a couple of hundred new sessions for some reason looking at 
>> the backup server syslog.
>> 
>> It’s keeping 14 versions of cucm cluster backups is that too many, although 
>> I’ve not seen anything to say so.
>> 
>> 
>> 
>> I’m going to change the file path tomorrow and see what happens with that.
>> 
>> 
>> 
>> Andy
>> 
>> 
>> 
>> On Fri, 19 Feb 2021 at 19:16, Wes Sisk (wsisk)  wrote:
>> 
>> What is the exact error? What do DRS logs show?
>> 
>> 
>> 
>> I see one report that after re-install dbreplication is 

Re: [cisco-voip] MRA DR / Resilience

[NateCCIE]

SIP Registration Failover for Cisco Jabber - MRA Deployments

https://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/expressway/release_note/Cisco-Expressway-Release-Note-X12-7.pdf#page16

This is new in x12.7

Sent from my iPhone

> On Jan 13, 2021, at 6:10 AM, Pawlowski, Adam  wrote:
> 
> 
> Hey all,
>  
> I’m playing in this scenario now and trying to figure out what parts of the 
> solution work, and which do not, in a DR “site failover’ kind of scenario 
> with regard to MRA.
>  
> I understand the documentation prescribes there’s no failover for voice and 
> video, but I think that failover is different than the one I’m describing 
> here.
>  
> I know I can take Expressway C and Expressway E nodes out of the cluster at 
> will, and things will heal over time once the Jabber clients catch up.
>  
> I can take a Unity Connection guest down, and it should work, though the 
> Jetty service certainly has load limits. I don’t think I’m hitting those here.
>  
> I can take an IM node down, and, with the exception of pChat services (DB 
> was not deployed HA and merge job just seems to fail but that’s another 
> investigation), clients will eventually fail over and recover.
>  
> Today, we have half the C  cluster, half the E cluster, and one of two CUC 
> nodes down. All IMP are up. One UCM subscriber is down, and things have been 
> going poorly. Jabber customers keep getting punted from the client with “Your 
> session has expired” randomly. The Jabber log looks like this token has 
> expired, but, doesn’t provide enough debugging to know why. It’s possible 
> that the Expressway E is fronting this message, since I understand it sits 
> between Jabber and the rest of the infrastructure for oAuth, and Jabber does 
> not talk to the UCM/CUC directly.
>  
> When we did not have SSO, the worst thing we had to do is make sure that the 
> Jabber client’s device pool had an active UCM as the primary in the CMGroup, 
> as they wouldn’t register properly without that, but, those UCMs are up.
>  
> Does anyone know what might be going on here?
>  
> My best guess is that the Expressway isn’t intelligent enough to mark a UCM 
> out of service when unreachable (or CUC server for that matter) and it is 
> trying to refresh a customer’s token against a server that isn’t up. When 
> this times out, instead of trying another it is telling Jabber the refresh 
> token is expired. If this is the case, there’s no cluster resilience with 
> Jabber, if any nodes are down then things are going to be intermittent.
>  
> Why does Jabber sometimes choose to pop the dialog asking for a new session, 
> and sometimes it just kicks the customer out of the client requiring a new 
> sign in? I see a bug that suggests enabling LegacyOAuthSignout parameter, 
> but, it doesn’t explain what effect that’s going to have on the client.
>  
> Basically, this is just a test but I am trying to learn from it, and would 
> appreciate any thoughts/experiences. If it is the Expressway cluster, then 
> there’s no way around this as far as I can tell. Marking a UCM inactive with 
> xAPI doesn’t work, it just gets pushed back to active.
>  
> Any comments appreciated.
>  
> Best,
>  
> Adam Pawlowski
> SUNYAB NCS
>  
>  
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Re: [cisco-voip] Alternatives for MediaSense simple recording?

[NateCCIE]

There is a record softkey that does what you’re asking Leilo.  The problem with 
Unity Live record is the CUCM BIB calls the number twice, one for TX and one 
for RX, and you get two-one sided voicemails in the inbox.  If that gets fixed, 
it’s an amazing option.

 

From: cisco-voip  On Behalf Of Lelio 
Fulgenzi
Sent: Monday, January 11, 2021 2:00 PM
To: Nick Barnett ; Brian Meade 
Cc: cisco-voip 
Subject: Re: [cisco-voip] Alternatives for MediaSense simple recording?

 

It’s too bad that CUCM doesn’t have a LiveRecord softkey macro that does the 
conferencing and dialing of the live record extension.

 

To ask people to press conference and then dial live record and the conference 
again, is just way to much to ask. I think.

 

Has Live Record support from CUCM side improved at all?

 

 

From: Nick Barnett mailto:nick@barnett.email> > 
Sent: Monday, January 11, 2021 3:50 PM
To: Brian Meade mailto:bmead...@vt.edu> >; Lelio Fulgenzi 
mailto:le...@uoguelph.ca> >
Cc: cisco-voip mailto:cisco-voip@puck.nether.net> >
Subject: Re: [cisco-voip] Alternatives for MediaSense simple recording?

 

CAUTION: This email originated from outside of the University of Guelph. Do not 
click links or open attachments unless you recognize the sender and know the 
content is safe. If in doubt, forward suspicious emails to ith...@uoguelph.ca 
 

 

Unity live record, that's one I haven't thought of yet. Thanks!

 

I'm pretty sure, mediasense is totally dead. We just upgraded to CUCM 12.5 SU3 
in October. MediaSense 11.5 su2 said it was compatible with CUCM 12.x, but in 
this case, it only meant 12.x THRU 12.5 SU2.  The BU's solution was to 
downgrade to SU2. We kind of pushed them and they came back with a fix. 
Apparently between CUCM 12.5 SU2 and 12.5 SU3, CUCM forced HTTPs for AXL 
connections.  to fix it, TAC had to root into my nodes and make a change to the 
haproxy.conf file to stop forcing HTTPS.

 

This whole mess took me right up to the last day of support and I think 
everyone at cisco hated me, but they were clear there would be no more support 
for this monster. meh

 

Thanks,

Nick

 

On Mon, Jan 11, 2021, at 2:24 PM, Brian Meade wrote:

Unity Connection Live Record may be an option you could try and have it 
conference in that number.

 

I think MediaSense is still around for video call handlers/video 
voicemail/video on hold if I remember correctly.  I think they only killed it 
for recording calls.

 

On Mon, Jan 11, 2021 at 1:56 PM Lelio Fulgenzi mailto:le...@uoguelph.ca> > wrote:

I sure was sad when they EOL’ed Media Sense. I really wanted to do video call 
handlers and video voicemail and greetings. 

 

Take a look at https://www.mns.vc/ they might have what you’re looking for.

 

Lelio

 

 

From: cisco-voip mailto:cisco-voip-boun...@puck.nether.net> > On Behalf Of Nick Barnett

Sent: Monday, January 11, 2021 12:54 PM

To: cisco-voip mailto:cisco-voip@puck.nether.net> >

Subject: [cisco-voip] Alternatives for MediaSense simple recording?

 

CAUTION: This email originated from outside of the University of Guelph. Do not 
click links or open attachments unless you recognize the sender and know the 
content is safe. If in doubt, forward suspicious emails to ith...@uoguelph.ca 
 

 

Hey folks. What are people using now that MediaSense is EOL? It was fine for 
what it was. It just recorded anything you threw into it. it weaseled it's way 
into some weird apps we have, and now I'm kinda stuck.  We have an iphone app 
that was developed to work in areas with poor data connectivity. It creates a 
conference call to a PSTN number that routes into our system and is a route 
pattern attached to a SIP trunk directly to MediaSense. 

 

>From there, we use APIs to pull the file down and save it using meta data from 
>the initial call.

 

We aren't using ANY of the recording profiles or advanced features of 
mediasense. Our new recording system is NICE Engage and they don't offer any 
way to record via route patterns.

 

Are there any open source, or really ANYTHING else out there that can do this 
simple procedure? The most basic of requirements are 1) non proprietary audio 
format 2) retrievable with an API or script. My cisco account team can only 
recommend Webex for recording which doesn't look to allow recording with a 
route pattern. Our VAR sells NICE which requires an extra application to kick 
of a recording like this. 

 

What are you guys using? Any suggestions for me?

 

Thanks,

Nick

 

P.S. just to be clear, MeidaSense is not our quality assurance platform. We use 
NICE Engage for that and it's fine for now... just looking for something to 
fill the gap left by a disappearing MediaSense and our route pattern recording 
method.

 

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Re: [cisco-voip] SFTP on CUCM 11.5.su8

[NateCCIE]

Have you tried changing the os to permissive?  I think it’s utils os secure 
permissive. 

Sent from my iPhone

> On Sep 4, 2020, at 4:49 AM, Louis Koekemoer (MEA) 
>  wrote:
> 
>  
> 
> Was wondering if anyone else was experiencing issues with SFTP on 11.5su8. I 
> have 5 clusters that was recently upgraded to 12.5su8 and when I want to 
> collect any files from them via SFTP it fails. I have done this numerous 
> times in my life and also tested with a 12.5su3 instance I have and it works, 
> but none of the 11.5su8 servers allows me. I used various different 
> servers/PC with FreeFTPD, Solarwinds and Mini SFTP.
>  
> Example would be to collect MOH files.
> 12.5su3
> admin:file get activelog mohprep/*
> Please wait while the system is gathering files info ...
> Get file: active/mohprep/CiscoMOHSourceReport.xml
>  
> Get file: active/mohprep/SampleAudioSource.alaw.wav
>  
> Get file: active/mohprep/SampleAudioSource.g729.wav
>  
> Get file: active/mohprep/SampleAudioSource.ulaw.wav
>  
> Get file: active/mohprep/SampleAudioSource.wb.wav
>  
> Get file: active/mohprep/SampleAudioSource.xml
>  
> Get file: active/mohprep/SilenceAudioSource.alaw.wav
>  
> Get file: active/mohprep/SilenceAudioSource.g729.wav
>  
> Get file: active/mohprep/SilenceAudioSource.ulaw.wav
>  
> Get file: active/mohprep/SilenceAudioSource.wb.wav
>  
> Get file: active/mohprep/SilenceAudioSource.xml
>  
> Get file: active/mohprep/ToneOnHold.alaw.wav
>  
> Get file: active/mohprep/ToneOnHold.g729.wav
>  
> Get file: active/mohprep/ToneOnHold.ulaw.wav
>  
> Get file: active/mohprep/ToneOnHold.wb.wav
>  
> Get file: active/mohprep/ToneOnHold.xml
> done.
> Sub-directories were not traversed.
> Number of files affected: 16
> Total size in Bytes: 18537609
> Total size in Kbytes: 18103.133
> Would you like to proceed [y/n]? y
> SFTP server IP: 23.240.48.250
> SFTP server port [22]: 21
> User ID: ccmadmin
> Password: 
> Download directory: /
>  
> ...
> Transfer completed.
> admin:
>  
> 11.5su8
> admin:file get activelog mohprep/*
> Please wait while the system is gathering files info ...
> Get file: active/mohprep/CiscoMOHSourceReport.xml
>  
> Get file: active/mohprep/SampleAudioSource.alaw.wav
>  
> Get file: active/mohprep/SampleAudioSource.g729.wav
>  
> Get file: active/mohprep/SampleAudioSource.ulaw.wav
>  
> Get file: active/mohprep/SampleAudioSource.wb.wav
>  
> Get file: active/mohprep/SampleAudioSource.xml
>  
> Get file: active/mohprep/SilenceAudioSource.alaw.wav
>  
> Get file: active/mohprep/SilenceAudioSource.g729.wav
>  
> Get file: active/mohprep/SilenceAudioSource.ulaw.wav
>  
> Get file: active/mohprep/SilenceAudioSource.wb.wav
>  
> Get file: active/mohprep/SilenceAudioSource.xml
>  
> Get file: active/mohprep/ToneOnHold.alaw.wav
>  
> Get file: active/mohprep/ToneOnHold.g729.wav
>  
> Get file: active/mohprep/ToneOnHold.ulaw.wav
>  
> Get file: active/mohprep/ToneOnHold.wb.wav
>  
> Get file: active/mohprep/ToneOnHold.xml
> done.
> Sub-directories were not traversed.
> Number of files affected: 16
> Total size in Bytes: 18537609
> Total size in Kbytes: 18103.133
> Would you like to proceed [y/n]? y
> SFTP server IP: 23.240.48.250
> SFTP server port [22]: 22
> User ID: ccmadmin
> Password: 
> Download directory: /
>  
> Could not connect to host 23.240.48.250 on port 22. Please verify SFTP 
> settings.
> admin:
> 
> 
> This email and all contents are subject to the following disclaimer:
> "http://www.dimensiondata.com/emaildisclaimer; 
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Re: [cisco-voip] CUBE un-normalizing P-Asserted-Identity on 200 OK after REINVITE for hold

[nateccie]

Tommy,

 

Thank  you for your reply.

 

As it turns out the customer had “midcall-signaling passthru” configured which 
was sending the hold INVITEs to the service provider, and then the CUBE was not 
re-translating the PAI.  I changed it to “midcall-signaling passthru 
media-change” which made the hold INVITEs stop at the SBC which made the caller 
id not update and then CUCM not send the UPDATE message so the REFER worked.

 

Thanks,

-Nate

 

 

From: Schlotterer, Tommy  
Sent: Monday, August 31, 2020 7:27 AM
To: natec...@gmail.com; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] CUBE un-normalizing P-Asserted-Identity on 200 OK 
after REINVITE for hold

 

You can use:

Voice service voip
sip 
no update-callerid

To stop the cube from sending the updated caller ID out.


You can also use the following on the dialpeer side to disable the updates:

no voice-class sip asserted-id

Thanks

Tommy





Tommy Schlotterer

 | 

Engineer



Presidio

 | 

  presidio.com


20 N Saint Clair 3rd Floor, Toledo, OH 43604



D:   419.214.1415

 | 

C:   419.706.0259

 | 

  tschlotte...@presidio.com



  

 

 

-Original Message-
From: cisco-voip mailto:cisco-voip-boun...@puck.nether.net> > On Behalf Of natec...@gmail.com 
 
Sent: Friday, August 28, 2020 7:30 PM
To: cisco-voip@puck.nether.net  
Subject: [cisco-voip] CUBE un-normalizing P-Asserted-Identity on 200 OK after 
REINVITE for hold

EXTERNAL EMAIL



So from the PSTN side I get 10 digit caller id and called number, I transform 
both calling and called with a voice translation profile. Works great, been 
doing it for years.

However I am working on this 3rd party IVR that does some call treatment then 
does a hold with a REINVITE and then a REFER. After the REINVITE happens the 
CUBE responds with a 200 OK, but it changes the P-Asserted-Identity from having 
the +15554441234 to just the 5554441234, CUCM sees this and says oh I need to 
send an UPDATE out the other leg, but the other leg is sending the REFER and 
doesn't process the UPDATE and so CUCM won't process the REFER and the whole 
thing blows up. If I require a MTP on the incoming CUBE leg, it seems like that 
isolates the CUBE from the HOLD reinvite and then it doesn't change the 
P-Asserted-Identity and the flow works fine.

Why would CUBE decide to change the P-Asserted-Identity, and how do I make it 
not? I was thinking I could just do a SIP Profile to change the 
P-Asserted-Identity back to the normalized form on the OK, but I don't want to 
go there first.

Running 16.9.5 IOS, feels like a bug but not finding anything with my googling.

Thanks,
-Nate

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[cisco-voip] CUBE un-normalizing P-Asserted-Identity on 200 OK after REINVITE for hold

[nateccie]

So from the PSTN side I get 10 digit caller id and called number, I
transform both calling and called with a voice translation profile.  Works
great, been doing it for years.

However I am working on this 3rd party IVR that does some call treatment
then does a hold with a REINVITE and then a REFER.  After the REINVITE
happens the CUBE responds with a 200 OK, but it changes the
P-Asserted-Identity from having the +15554441234 to just the 5554441234,
CUCM sees this and says oh I need to send an UPDATE out the other leg, but
the other leg is sending the REFER and doesn't process the UPDATE and so
CUCM won't process the REFER and the whole thing blows up.  If I require a
MTP on the incoming CUBE leg, it seems like that isolates the CUBE from the
HOLD reinvite and then it doesn't change the P-Asserted-Identity and the
flow works fine.

Why would CUBE decide to change the P-Asserted-Identity, and how do I make
it not?  I was thinking I could just do a SIP Profile to change the
P-Asserted-Identity back to the normalized form on the OK, but I don't want
to go there first.

Running 16.9.5 IOS, feels like a bug but not finding anything with my
googling.

Thanks,
-Nate

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Re: [cisco-voip] social miner certificate sign

[NateCCIE]

No. 

Sent from my iPhone

> On Jul 2, 2020, at 8:24 PM, naresh rathore  wrote:
> 
> 
> hi
> 
> 
> 
> we have Cisco communications manager 12.0 setup installed with UCCX12 and 
> Social Miner12. one of my colleague generated CSR previously and customer 
> signed the csr via Public CA and replied and then i took over the project. 
> 
> 
> I wasn't aware about that CSR and generated new CSR. I want to know is there 
> a way that we can discard this new CSR and can use old CSR and signed 
> Certificate? 
> 
> 
> Regards
> 
> 
> Naray
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Re: [cisco-voip] Wildcard certificates

[NateCCIE]

Yeah. In my experience, the cert can have as many extra sans as you want, but 
all of the sans the cucm csr has have to be there, and spelled correctly. 

Sent from my iPhone

> On Jun 19, 2020, at 1:02 AM, James Andrewartha  
> wrote:
> 
> It helps if I spell speeddial instead of speedidal 
> 
>> On 19/6/20 2:21 pm, Anthony Holloway wrote:
>> I've got some thoughts, though, I've never done this before, so it's
>> just guessing.
>> 
>> You don't need *.domain.com  in your SAN.
>> 
>> Just generate your CSR on CUCM as if you were not using wildcard
>> certificates.  Then when you dupe your wildcard on digitcert's site,
>> manually add the exact same SANs in your CSR.
>> 
>> The resulting identity certificate will not have a CN which matches your
>> CSR, but the SANs will match, and according to the thread you linked:
>> 
>> /"The CN doesn't match but CUCM doesn't seem to care as long as the SAN
>> fields line up."/
>> 
>> On Thu, Jun 18, 2020 at 11:58 PM James Andrewartha
>> mailto:jandrewar...@ccgs.wa.edu.au>> wrote:
>> 
>>Hi voipers,
>> 
>>I'm trying to update the wildcard on our CUCM/IMP servers, and am
>>hitting a problem. We have a digicert wildcard, which I used
>>successfully before, but now when generating the certificate the UI
>>complains that *.ccgs.wa.edu.au  isn't a
>>valid certificate name or SAN. I
>>hacked the javascript to ignore this warning, and generated a CSR with
>>*.ccgs.wa.edu.au  in the SAN:
>> 
>>$ openssl req -in tomcat\(8\).csr -text|grep DNS
>>DNS:callmanager1.voip.ccgs.wa.edu.au
>>,
>>DNS:*.ccgs.wa.edu.au , DNS:ccgs.wa.edu.au
>>,
>>DNS:speeddial.voip.ccgs.wa.edu.au
>>,
>>DNS:callmanager2.voip.ccgs.wa.edu.au
>>,
>>DNS:voip.ccgs.wa.edu.au ,
>>DNS:callmanager.voip.ccgs.wa.edu.au
>>,
>>DNS:presence.voip.ccgs.wa.edu.au 
>> 
>>But when I try to upload the certificate to CUCM, it complains "CSR SAN
>>and Certificate SAN does not match". But the SANs on the certificate are
>>the same (albeit in a different order):
>> 
>>$ openssl x509 -in ../ssl/digicert/cucm-star_ccgs_wa_edu_au.crt -text
>>|grep DNS
>>DNS:*.ccgs.wa.edu.au ,
>>DNS:ccgs.wa.edu.au ,
>>DNS:voip.ccgs.wa.edu.au ,
>>DNS:callmanager1.voip.ccgs.wa.edu.au
>>,
>>DNS:callmanager2.voip.ccgs.wa.edu.au
>>,
>>DNS:speedidal.voip.ccgs.wa.edu.au
>>,
>>DNS:callmanager.voip.ccgs.wa.edu.au
>>,
>>DNS:presence.voip.ccgs.wa.edu.au 
>> 
>>I found
>>
>> https://community.cisco.com/t5/unified-communications/wildcard-certificate-on-call-manager-10-5/td-p/2757989
>>from 2016 which says they got it working then, and I also got it working
>>in 2018 when the cert was last renewed, with *.ccgs.wa.edu.au
>> as the
>>common name and a SAN. But I can't get it working now. Anyone got any
>>thoughts? Running CUCM 10.5.2.15900-8
>> 
>>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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>>https://puck.nether.net/mailman/listinfo/cisco-voip
>> 
> 
> 
> -- 
> 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] CUBE Config Dial Peers

[NateCCIE]

Well once Loren speaks up you know it’s an interesting thread.

 

My two cents, I cannot stand DPG.  Its crazy that it completely ignores 
destination pattern.  If you have two destinations in the group, it just 
round-robins them.  I got burned not understanding that DPG didn’t look at 
destination pattern and I swore I would never use them again.

 

I use cor-list to restrict the SP inbound dial-peer to the cucm outbound 
dial-peer, and vice versa.  Matching the inbound dial-peer by URI works great, 
I started with matching “FROM” but that fell apart in some cases, so I use VIA 
by default now, and that has been solid.

 

My numbering is usually 1X for CUCM, with the 0 for inbound in each range, then 
2X for the first SIP provider and 3X for the 2nd, maybe 5X for CVP etc.

 

I always localize on the CUBE, sending a full +E.164 from CUCM and then use 
translation profiles to format to how the specific country/carrier wants to see 
the calls.  The exception is US 11D/10D determination is done by the CUCM 
because I find it easier to load all of the local NPA-NXX into CUCM route 
filters via AXL, but then sometimes I am doing TEHO and have to control which 
outbound dial-peer it chooses.

 

I also never let the CUBE choose the carrier, I think mostly because a long 
time ago I had the same carrier spread over multiple gateways along with 
multiple carriers in each gateway, and I wanted CUCM to re-route to the other 
gateway same carrier before CUBE used a less preferred route because it was 
local.  So when there is multiple carriers I usually will prefix 1#* or 2#* on 
up for each carrier.

 

Anyway, that’s my 2 cents.

 

 

From: cisco-voip  On Behalf Of Loren 
Hillukka
Sent: Tuesday, June 16, 2020 10:26 AM
To: Anthony Holloway 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] CUBE Config Dial Peers

 

Nice to see the examples and explanations - thank you!  I really like the 
naming structure to allow simple a show command to pull everything related to 
one side of the call flow.  

URI matching broke down in UCCE environments as uri match overrides all other 
matching.  I needed to match some ingress numbers from the ITSP to apply CVP 
.tcl scripts too and wasn’t able to when matching all inbound from ITSP via 
URI.  So the config gets a bit longer in UCCE environments due to this. 

I ended up using e164-pattern-maps as another means of collapsing dial-peers, 
with uri match for calls from CUCM, and also server-groups to reduce outbound 
peers. 

Based on the configs from Brian and Anthony, you wouldn’t need 
e164-pattern-maps in those environments.  Curious what direction others have 
taken to simplify dial-peers with UCCE in play. 

 

Loren





On Jun 16, 2020, at 10:55 AM, Anthony Holloway mailto:avholloway+cisco-v...@gmail.com> > wrote:

 

Sorry, transmission failed.  Try disabling NSF and re-sending. 

 

Back to the point of ABC123, it would be so nice if we could add comments to 
the show run.  Second best is to keep a commented copy of the config off box in 
your documentation repository.

 

On Mon, Jun 15, 2020 at 11:31 PM Brian Meade mailto:bmead...@vt.edu> > wrote:

Anthony, 

 

I like the config.  Definitely is nice to have some standardization on the 
dial-peer tags.  I've usually done all my inbound dial-peers in the 1XX range 
but have gone outside of that lately with separating inbound ITSP and inbound 
CUCM dial-peers to make them more obvious but I like the idea of having more 
structure like yours.

 

Using the destination-pattern ABC123 is a great idea to show that's not used as 
mentioned before.

 

I try to always use voice-class codec for every dial-peer even if I've only got 
1 codec configured there just to make it easier to change if ever needed but 
that was in the past when I had separate local/long 
distance/911/international/10-digit dial-peers.  Simplifying it down to a 
single inbound/outbound dial-peer with the ITSP makes it a toss-up if that 
helps anymore.

 

I've tried to keep KPML on my ITSP facing dial-peers just in case they happen 
to support it.  I've found some say they don't but actually do use it if you 
advertise it.  No harm in advertising it from our side.

 

I like the aliases you've got there as well.  I feel like I'm always on some 
random customer's box so not sure I'd remember to always put them in first but 
definitely nice to have when you make the full CUBE config.

 

Looks like all you're missing is your fax config!  I can fax that over to you! 
:)

 

On Fri, Jun 12, 2020 at 8:53 PM Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com> > 
wrote:

All great points, thanks for taking the time to respond. 

 

The only one I think that I need to reply to is the DPG and destination-pattern 
one.  I was actually troubleshooting a customer CUBE wherein this exact 
scenario was in place and the only negative was getting options to work.  
Otherwise, it was routing the call just fine.  Granted, I couldn't tell you 
what 

Re: [cisco-voip] sip 404 not found for incoming calls

[NateCCIE]

Have you been resting the trunk in cucm after each of your changes?

Sent from my iPhone

> On Apr 30, 2020, at 7:08 AM, naresh rathore  wrote:
> 
> 
> hi
> 
> 
> Pls find the attached ccsip messages and cube config. i will send voice ccapi 
> inout config tomorrow. outgoing call works fine but cucm respond with 404 
> message during incoming call attempt
> 
> 
> Regards
> 
> 
> 
> From: Ryan Huff 
> Sent: Thursday, April 30, 2020 10:50 PM
> To: naresh rathore 
> Cc: Amit Kumar ; cisco-voip@puck.nether.net 
> 
> Subject: Re: [cisco-voip] sip 404 not found for incoming calls
>  
> Naresh,
> 
> Any chance you could send a ccsip messages and a voice ccapi inout debug from 
> a failed inbound call?
> 
> Sent from my iPhone
> 
>>> On Apr 30, 2020, at 06:28, naresh rathore  wrote:
>>> 
>> 
>> hi
>> 
>> 
>> I tried both, via translation pattern or directly pointing a particular 
>> number to phone but still the same result.
>> 
>> Regards
>> 
>> 
>> From: Amit Kumar 
>> Sent: Thursday, April 30, 2020 6:41 PM
>> To: naresh rathore 
>> Cc: James B ; cisco-voip@puck.nether.net 
>> 
>> Subject: Re: [cisco-voip] sip 404 not found for incoming calls
>>  
>> Are you having a dn, exact to called number, of you are doing some 
>> translation, then make sure route pattern incoming css shoold have access to 
>> xlate pt. Nd xlate css shoud have access to phones pt. 
>> 
>> On Thu, Apr 30, 2020, 2:02 PM naresh rathore  wrote:
>> hi,
>> 
>> 
>> Thanks for the reply. 
>> 
>> 
>> pls see following snapshot and attached gateway config. outgoing dialpeer 
>> (200,201) is currently matching correctly to cucm (for incoming call)
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> Regards
>> 
>> Naray
>> From: James B 
>> Sent: Thursday, April 30, 2020 5:39 PM
>> To: naresh rathore ; cisco-voip@puck.nether.net 
>> 
>> Subject: RE: [cisco-voip] sip 404 not found for incoming calls
>>  
>> Hello,
>>  
>> Can you send your gateway configuration and a screenshot of your CUCM trunk 
>> configuration? That’d give us more to go off of.
>>  
>> Thanks,
>>  
>> James
>>  
>>  
>>  
>> From: naresh rathore
>> Sent: 30 April 2020 08:36
>> To: cisco-voip@puck.nether.net
>> Subject: [cisco-voip] sip 404 not found for incoming calls
>>  
>> hi,
>>  
>>  
>>  
>> i have cucm version 12.0. outgoing call is working without any issue. but 
>> incoming call is failing. the call request is received by cucm but its 
>> responding with 404 not found. i checked CSS and also pointed call directly 
>> to ip phone using significant digits and incoming css but still the same 
>> issue. also sip uri have called number. not sure why 404 not found msg is 
>> sent by cucm to cube.
>>  
>>  
>> Regards
>>  
>> Naray
>>  
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Re: [cisco-voip] Can MRA work with old UCM configured with IP addresses?

[NateCCIE]

I know I have done one MRA with IPs, back when MRA was new. You still need the 
Cisco Uds srv record to be there and that record needs to be resolvable my the 
C.

That old system isn’t using MRA right now, so I don’t know if it’s changed in 
newer versions of expressway. 

Sent from my iPhone

> On Apr 1, 2020, at 8:13 PM, Dana Tong  wrote:
> 
> 
> Hi all,
> 
> I have a customer who was installed some 9 years ago and the hosts were all 
> configured with IP address instead of FQDN. They’re up on UCM 10.5(2) now and 
> I have spun up Expressway X12.5.7.
> The UC Traversal zone is up and running. The Expressway’s have certificates 
> installed.
> However I am hitting issues with getting the edge config and signing in. I 
> get the usual “cannot communicate with server” when I try to login with 
> Jabber.
>  
> I’ve tried changing some things such as configuring the UCM, IM, Unity 
> servers in Expressway-C by IP or FQDN and have had some varying results on 
> the Collaboration Solution Analyser. Sometimes its downloads the user UDS 
> configuration and sometimes it doesn’t.
>  
> Is MRA do-able with the UCM using IP address? Or am I going to have to bite 
> the bullet and reconfigure all the devices to use FQDN for their clustering? 
> Or is there some kind of easy fix?
>  
> Cheers
> Dana
>  
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Re: [cisco-voip] CCX phone agent over MRA?

[NateCCIE]

I could be confused but I don’t think the phones talk CTI or CTI-QBE, they just 
talk SIP/SCCP.  UCCX talks CTI to CUCM’s CTI manager which then tells the phone 
to do something.  I know CTI isn’t supported over MRA, but that is for 
deskphone control from Jabber, not the jabber softclient.

 

I have people using CCE via thin client talking to a MRA registered jabber.  
Now that I think of it, they said you can’t use Jabber for mobile because as an 
agent device, but I would think that is some other limitation.

 

From: cisco-voip  On Behalf Of Wakelin, 
Frank
Sent: Monday, March 23, 2020 2:49 PM
To: 'Aman Chugh' 
Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net) 

Subject: Re: [cisco-voip] CCX phone agent over MRA?

 

I’m not really sure – I have my doubts as well.  They sited the lack of CTI-QBE 
support as to why the CCX servers could not use CTI to control the phones 
connected over MRA.  The more folks that tell me they had it working with CCX, 
the more I think they just latched on to the phrase in the feature 
configuration guide and went with that.  It certainly wouldn’t be the first 
time TAC has given me a pat answer and been unwilling to escalate/troubleshoot 
with me. :(

 

---

Frank Wakelin – Senior Network Analyst

Information Technology | City of Richmond 

 

Office +16042764190

Mobile +17788394693

  fwake...@richmond.ca

 

From: Aman Chugh mailto:aman.ch...@gmail.com> > 
Sent: March 23, 2020 1:43 PM
To: Wakelin, Frank mailto:fwake...@richmond.ca> >
Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net 
 ) mailto:cisco-voip@puck.nether.net> >
Subject: Re: [cisco-voip] CCX phone agent over MRA?

 

Just curious as to what makes it unsupported with 11.5 or what is added in 12 
which makes it supported.

 

Does SIP phone doing MRA require support for certain sip headers which are only 
supported with CSR 12 or later.

 

I did have it working with CUCM 10.5 , UCCE 11.6 and Expressway 8.11.2

 

 

 

On Mon, Mar 23, 2020 at 4:32 PM Wakelin, Frank mailto:fwake...@richmond.ca> > wrote:

Thanks all – I did get confirmation from TAC that this is not supported – at 
least not with 11.5.  Not sure I’m ready to upgrade everything to 12.x at the 
moment to test but will eat-mark it for later this year.

 

---

Frank Wakelin – Senior Network Analyst

Information Technology | City of Richmond 

 

Office +16042764190

Mobile +17788394693

  fwake...@richmond.ca

 

From: Anthony Holloway mailto:avholloway%2bcisco-v...@gmail.com> > 
Sent: March 23, 2020 11:05 AM
To: James B mailto:james.buchan...@gmail.com> >
Cc: Wakelin, Frank mailto:fwake...@richmond.ca> >; voyp 
list, cisco-voip (cisco-voip@puck.nether.net 
 ) mailto:cisco-voip@puck.nether.net> >
Subject: Re: Re: [cisco-voip] CCX phone agent over MRA?

 

No, I wouldn't think it has anything to do with finesse.  Again the user had 
direct access to Finesse, no VPN or Internet exposure, just simply the user was 
on the network with the PC while the phone was on a public internet circuit.  
I'd guess it has worked in previous versions, as it's been in the UCCX SRND for 
a while now, but perhaps there's some issues with it.  I just wanted to add a 
story of success to this otherwise long thread of failures.

 

On Mon, Mar 23, 2020 at 12:46 PM James B mailto:james.buchan...@gmail.com> > wrote:

Hi Anthony,

 

Do you attribute that to the change in web connectivity for Finesse with 12.x?

 

James

 

 

 

From: Anthony Holloway  
Sent: 23 March 2020 17:44
To: Wakelin, Frank  
Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net) 
 
Subject: Re: [cisco-voip] CCX phone agent over MRA?

 

For whatever it's worth, I just upgraded a customer from CSR 11 to CSR 12.5 
(including UCCX), and testing of an Agent phone registered over MRA with the 
Finesse client directly accessing Finesse server worked.  Clicking call control 
buttons in Finesse was successful in controlling the phone.  So, while the 
documentation and field experiences maybe fuzzy, here's one empirical case of 
evidence that it does work on the latest versions.

 

Frank, what did TAC respond to you with?

 

On Wed, Mar 18, 2020 at 1:07 PM Wakelin, Frank mailto:fwake...@richmond.ca> > wrote:

Thanks for the reply.Finesse is using VPN, but the physical desk phone is 
not – it is connected via MRA.  Standard inbound/outbound calling to the phone 
itself works flawlessly over MRA.  What isn’t working are CCX calls to the 
agent phone; CCX uses CTI to control/monitor the desk phone.

 

I did read the CCX/expressway design guide which generally states that CCX over 
MRA is supported, but features that rely on CTI-QBE are not.  The documentation 
isn’t clear as to what CCX features rely on that.  It does say CCX is supported 
however and in my mind 

Re: [cisco-voip] Jabber mobile calls disconnecting after 15 minutes

[NateCCIE]

When it’s the same time across multiple clients, it’s a timer issue. I first 
would look at the firewall handing a timer expire. 

Sent from my iPhone

> On Mar 22, 2020, at 1:59 PM, James Dust  
> wrote:
> 
> 
> Good evening all,
>  
> I have a reoccurring issues for a handful or users, where their external 
> calls via jabber mobile disconnects after 15 minutes exactly. This isn’t 
> every user only a handful, and can happen over Wi-Fi or 4G networks when they 
> work remotely.
>  
> I have tried allot of trouble shooting, but wondered if anyone else had 
> experienced this very specific issue?
>  
> We are running:
>  
> Cucm 11.5
> Expressway X12.5.3
> 
> Consider the environment - Think before you print
> 
> The contents of this email are confidential to the intended recipient and may 
> not be disclosed. Although it is believed that this email and any attachments 
> are virus free, it is the responsibility of the recipient to confirm this. 
> 
> You are advised that urgent, time-sensitive communications should not be sent 
> by email. We hereby give you notice that a delivery receipt does not 
> constitute acknowledgement or receipt by the intended recipient(s).
> 
> Details of Charles Stanley group companies and their regulators (where 
> applicable), can be found at this URL 
> http://www.charles-stanley.co.uk/contact-us/disclosure/
> 
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Re: [cisco-voip] Jabber and Finesse

[NateCCIE]

Is the UCCX the primary line or the 2nd line on the jabber?  You need newer 
expressway CUCM and jabber to support jabber multi line.

I don’t know about UCCX, but i am told with CCE, the agent line has to be the 
primary line on jabber. 

Sent from my iPhone

> On Mar 18, 2020, at 8:25 PM, Terry Oakley via cisco-voip 
>  wrote:
> 
> 
> Thank you for the quick replies.   I know all of you are undergoing immense 
> pressure so I truly appreciate the assistance.   I have triple checked that 
> the UCCX extension is just on the Jabber Windows client.  When I try and dial 
> the extension I get the nice Cisco lady telling me the number cannot be 
> completed as dialed.   If I dial the primary extension on the Jabber client 
> it works.If I put the UCCX extension on a physical set (8851) it will 
> ring.  
>  
> When I am on the Jabber Windows client I have checked the CSS for the UCCX 
> extension it is fine, same as the primary line.  Double checked to make sure 
> the extension was an active number. Allow Control of Device from CTI is 
> enabled.   There must be some little check box or something that I have 
> missed but I have stared at the page so long it all looks the same.  
>  
> Thanks again
>  
> Terry
>  
>  
>  
> From: Jose Colon II 
> Sent: Wednesday, March 18, 2020 4:38:26 PM
> To: Pawlowski, Adam 
> Cc: Terry Oakley ; cisco-voip@puck.nether.net 
> 
> Subject: Re: [cisco-voip] Jabber and Finesse
>  
> CAUTION: This email is from an external source. Do not click links or open 
> attachments unless you recognize the sender and know the content is safe.
> I think that is the key to the issue. UCCX extension can only be registered 
> to one device. 
> 
> On Wed, Mar 18, 2020 at 5:37 PM Pawlowski, Adam  wrote:
> Hi Terry,
> 
>  
> 
> I had the same problem when I had my CCX extension on multiple items, even 
> when unregistered. Clicking on ready resulted in an error, but the first time 
> I made a call with it by opening the keypad it started working and I could go 
> ready. Since the CCX extension is just an extension, you should be able to 
> dial it regardless of what Finesse is doing, assuming it is in a partition 
> that you can dial but it may not be.
> 
>  
> 
> After I made sure the extension was on nothing but my Jabber client, and I 
> had signed out and back in, it began to work fine.
> 
>  
> 
> I haven’t heard any comments from anyone else and we moved ~75 seats to 
> Jabber MRA and Finesse remote this week.
> 
>  
> 
> Adam
> 
>  
> 
> From: cisco-voip  On Behalf Of Terry 
> Oakley via cisco-voip
> Sent: Wednesday, March 18, 2020 6:29 PM
> To: cisco-voip@puck.nether.net
> Subject: [cisco-voip] Jabber and Finesse
> 
>  
> 
> We have on prem CUCM running 12.5.1.   We also have IM and Presence and UCCX 
> for our phone queues.   I am trying to figure out if I can move our phone 
> queues to Jabber and connect to Finesse via remote access (through VM Ware).  
>   I seem to be able to get part way but when I try to make a call to the 
> queue the Finesse line will not answer and unless I go off hook first on the 
> Jabber app I cannot go to Ready on the Finesse side .   I cannot even dial it 
> just directly.   I can use that line and dial out from Jabber but for some 
> reason I cannot get the line to be recognized on the Finesse side.   I am 
> sure I probably missed something in my haste so if anyone of you have 
> successfully done something like this I would appreciate a simple how to. 
> 
>  
> 
> I hope all of you are safe and your families as well. 
> 
>  
> 
> 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) 343-4034
> 
>  
> 
>  
> 
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[cisco-voip] CMR from Jabber

[NateCCIE]

Has anyone figured out how to get CMR records from Jabber?  I can find lots of 
things saying Jabber doesn’t' support Call Management Records, then it was 
added I believe is sometime in 11.X, were running 12.6/12.8 and not seeing 
records in ISI.

Thanks,
-Nate

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Re: [cisco-voip] migrating mpp phones to callmanager (and back)

[NateCCIE]

MPP to CUCM free, but cannot go back without a paid license, which happens
to be included with cloud flex.

No advantage to buying MPP if you're going to register them to CUCM.

-Original Message-
From: cisco-voip  On Behalf Of Lelio
Fulgenzi
Sent: Monday, March 2, 2020 8:44 AM
To: voyp list, cisco-voip (cisco-voip@puck.nether.net)

Subject: [cisco-voip] migrating mpp phones to callmanager (and back)


Is there anything specially about an mpp phone that wouldn't allow you to
load CUCM software on it? And then go back to MPP when needed?



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Re: [cisco-voip] Field Notice from Cisco making Secure LDAP mandatory

[NateCCIE]

Or the good old days when you could list an IP Address as a SAN. 

Sent from my iPhone

> On Feb 14, 2020, at 9:48 AM, Lelio Fulgenzi  wrote:
> 
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Re: [cisco-voip] CUCM Cluster Expansion

[NateCCIE]

I always do the 7.5k cucm size.  I hate single cpu cucm, ram is usually not a 
problem and I’d rather have the 110GB disk because upgrades about never work on 
the 80gb without clearing some space.  Even 110GB has become a problem lately. 

Sent from my iPhone

> On Feb 13, 2020, at 3:29 PM, Ryan Huff  wrote:
> 
> 
> For 11.x, but I've found this helpful: 
> https://www.cisco.com/web/software/283088407/126036/cucm-11.0.ova.readme.txt
> 
> Thanks,
> 
> Ryan
> From: Matthew Loraditch 
> Sent: Thursday, February 13, 2020 5:24 PM
> To: Ryan Huff ; cisco-voip@puck.nether.net 
> 
> Subject: RE: CUCM Cluster Expansion
>  
> Yeah, I’m just trying to understand (as I read the ovf file) what the actual 
> difference is between the 1000/2500 user OVA. I seem to be missing something 
> (or maybe not). CPU is actually 1 less starting but same reservation, same 
> RAM, same HDD.
>  
>   
> Matthew Loraditch​
> Sr. Network Engineer
> p: 443.541.1518
> w: www.heliontechnologies.com  |  e: mloradi...@heliontechnologies.com
> 
> 
> 
> 
> 
> From: Ryan Huff  
> Sent: Thursday, February 13, 2020 5:21 PM
> To: Matthew Loraditch ; 
> cisco-voip@puck.nether.net
> Subject: Re: CUCM Cluster Expansion
>  
> [EXTERNAL]
>  
> I wouldn't see a reason not to just up-size the two nodes you have now to the 
> 2.5k OVA (use 2 vCPU on each node). For the 15 pieces of flair, I'd then add 
> in a 3rd 2.5k OVA w/o the CCM service enabled and run TFTP.. etc on it and 
> give the pub a break.
>  
> -Ryan
>  
> From: cisco-voip  on behalf of Matthew 
> Loraditch 
> Sent: Thursday, February 13, 2020 5:10 PM
> To: cisco-voip@puck.nether.net 
> Subject: [cisco-voip] CUCM Cluster Expansion
>  
> One of my biggest customers is experiencing issues that appear to be related 
> to resource utilization. I’ve never had a customer who needed more than a 2 
> node 1000 user cluster.
>  
> They are getting close to some of the capacity levels listed in the sizing 
> guides.
>  
> I’m looking for some opinions on what the best way to deal with this. I have 
> the hardware capacity for either method.
>  
> Add a Third 1000 user Subscriber and turn off call processing and tftp on the 
> Pub?
>  
> Rebuild both existing servers to 2500 user OVAs?
>  
> Add a third and do the rebuild also?
>  
> Can I just make the existing server be the 2500 capacity level? I actually 
> don’t understand the difference between the 2500 and 1000 user OVAs, the 2500 
> appears to actually be lesser capacity by default (1 less cpu). So go to 7500?
>  
> I’d appreciate any opinions out there. Going to be doing some reading over 
> the next few days to try and figure this out.
>  
> Thanks all!
>  
> Matthew Loraditch​
> Sr. Network Engineer
> p: 443.541.1518
> w: www.heliontechnologies.com
>  | 
> e: mloradi...@heliontechnologies.com
> 
> 
> 
> 
> 
>  
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Re: [cisco-voip] 8851 - power draw

[NateCCIE]

https://www.cisco.com/c/en/us/products/collateral/collaboration-endpoints/un
ified-ip-phone-6900-series/solution_overview_c22-589129.html


-Original Message-
From: cisco-voip  On Behalf Of Lelio
Fulgenzi
Sent: Thursday, February 6, 2020 4:12 PM
To: voyp list, cisco-voip (cisco-voip@puck.nether.net)

Subject: [cisco-voip] 8851 - power draw


Does anyone have a bunch of 8851s deployed? Can I trouble you for a "show
inline power" output? I'm looking for what normal operating power draw is
for this model. No side cars, no USB, sort of thing.

I've got an 8865 showing 12.9W, and I'm hoping it's less. We're looking at
selecting 8841 or 8851 and power draw is a concern. Startup too, but
apparently, startup will cycle through.

Lelio


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Re: [cisco-voip] ipvmsa service

[NateCCIE]

Glad you were able to restart, but I’d like to say for historical purposes that 
any media resources that are not in any MRG become globally available, like an 
extension that is not in a partition.  If you want to make them not used you 
have to assign to a MRG that is not in a MRGL.

 

-Nate

 

From: cisco-voip  On Behalf Of Scott Voll
Sent: Friday, January 24, 2020 12:55 PM
To: Anthony Holloway 
Cc: Cisco VoIP Group 
Subject: Re: [cisco-voip] ipvmsa service

 

Thanks guys--  I removed the mtp, ann, cfb, and moh resources from the MRG   
waited for resources to go down to 0 and restarted.  Now I can add the services 
back to the MRG.

 

Scott

 

 

On Fri, Jan 24, 2020 at 9:59 AM Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com> > 
wrote:

It depends on what and how you're using those resources. 

 

Eg if you don't use CFB on CM, or if you only use hardware MTP, etc. 

 

On Fri, Jan 24, 2020, 11:00 AM Scott Voll mailto:svoll.v...@gmail.com> > wrote:

Can you restart the IP voice media streaming app service on CM without issues?

 

I'm troubleshooting a MOH issue and would like to restart the service mid day.  
Just want to make sure I won't kill anything.

 

Thanks

 

scott

 

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Re: [cisco-voip] ipvmsa service

[NateCCIE]

If there are active calls on the software conference bridge, those will be 
killed.  But generally a low risk restart.

 

From: cisco-voip  On Behalf Of Scott Voll
Sent: Friday, January 24, 2020 10:00 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] ipvmsa service

 

Can you restart the IP voice media streaming app service on CM without issues?

 

I'm troubleshooting a MOH issue and would like to restart the service mid day.  
Just want to make sure I won't kill anything.

 

Thanks

 

scott

 

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Re: [cisco-voip] Jabber Softphone over WiFi

[NateCCIE]

Cucm 12.5 jabber config xml per user?

Sent from my iPhone

> On Oct 10, 2019, at 8:32 PM, Lelio Fulgenzi  wrote:
> 
> 
> 
> Thanks for that info. I had read about custom tabs for speed dials. 
> 
> I was hoping for the same set of speed dials but at least it’s an option. 
> 
> Thing is, I’d have to enable that custom tab for those who wanted it. 
> 
> I’m thinking of an advanced.feature.config.xml and configure only the handful 
> of clients that need the feature set like hunt group, speed dial, etc. 
> 
> -sent from mobile device-
> 
> Lelio Fulgenzi, B.A. | Senior Analyst
> Computing and Communications Services | University of Guelph
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 
> 2W1
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>  
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>  
> 
> 
>> On Oct 10, 2019, at 9:45 PM, Loren Hillukka  wrote:
>> 
>> Lots to plan out carefully if going that way indeed. Regarding speed dials 
>> one customer settled on using Jabber with a custom tab that loaded a file on 
>> their computer with speed dials in it. The user controlled the file and 
>> updated it as they needed. 
>> 
>> Loren
>> 
>>> On Oct 10, 2019, at 6:19 PM, Tim Smith  wrote:
>>> 
>>> 
>>> Actually good reminder to get some of the new Cisco headsets and give those 
>>> a good run
>>>  
>>> I’ve had issues with support on the other big vendors of headsets
>>> That was only on a few headsets, it would have been a problem on a bigger 
>>> deployment
>>>  
>>> I like the idea of end to end Cisco here
>>>  
>>> From: Lelio Fulgenzi  
>>> Sent: Friday, 11 October 2019 10:15 AM
>>> To: Tim Smith 
>>> Cc: Scott Voll ; voyp list, cisco-voip 
>>> (cisco-voip@puck.nether.net) 
>>> Subject: Re: [cisco-voip] Jabber Softphone over WiFi
>>>  
>>>  
>>> And speaking to that... test them out. And ask people around you. 
>>>  
>>> A headset with audio leak is fine for an office, but not for cubicle land. 
>>> 
>>> -sent from mobile device-
>>> 
>>> 
>>> Lelio Fulgenzi, B.A. | Senior Analyst
>>> Computing and Communications Services | University of Guelph
>>> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 
>>> 2W1
>>> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>>>  
>>> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>>>  
>>> 
>>> 
>>> On Oct 10, 2019, at 6:59 PM, Tim Smith  wrote:
>>> 
>>> Picking good headsets is important too…
>>> Much more hassle troubleshooting some rubbish USB headset
>>> Or even an expensive USB headset with rubbish support
>>>  
>>> From: cisco-voip  On Behalf Of Scott 
>>> Voll
>>> Sent: Friday, 11 October 2019 9:55 AM
>>> To: Lelio Fulgenzi 
>>> Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net) 
>>> 
>>> Subject: Re: [cisco-voip] Jabber Softphone over WiFi
>>>  
>>> We have started migrating our Telecommuters over to Jabber from the old IP 
>>> communicator.  So we are getting the "can I use this in the office"  "do 
>>> you have to leave the phone on the desk?"  I think eventually we will have 
>>> a lot of people over on Jabber.  The question is, does everyone move... 
>>>   I think we will have some people that really want physical phones.   and 
>>> for emergencies you will still want a physical phone available.  then lets 
>>> start the conversation about E911.  Then what happens if they are not in 
>>> the office?  or they are running jabber for iphone / Droid???  how does 
>>> that work?  Lots of things to work through.  Can it be done?  yes, but plan 
>>> ahead.
>>>  
>>> Scott
>>>  
>>> On Thu, Oct 10, 2019 at 11:04 AM Lelio Fulgenzi  wrote:
>>> I get contacts…. But for me (and many others I imagine) a speed dial is a 
>>> button. One click.
>>>  
>>> But again… its just different ways of doing things.
>>>  
>>>  
>>> ---
>>> Lelio Fulgenzi, B.A. | Senior Analyst
>>> Computing and Communications Services | University of Guelph
>>> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 
>>> 2W1
>>> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>>>  
>>> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>>>  
>>> 
>>>  
>>> From: Pawlowski, Adam  
>>> Sent: Thursday, October 10, 2019 2:03 PM
>>> To: Lelio Fulgenzi ; Casper, Steven 
>>> Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net) 
>>> 
>>> Subject: RE: Jabber Softphone over WiFi
>>>  
>>> Call Park should hopefully be there pretty eventually. It is there on 
>>> mobile today. Speed dials sure those would be your contacts, or “pizza 
>>> guys”.
>>>  
>>> I would not deploy an office on it as the only/primary phone without 
>>> knowing if my wireless network was bulletproof. Jabber works fairly well 
>>> and it is not a huge hog on the medium but wireless being what it is, ymmv.
>>>  
>>> Adam
>>>  
>>> From: cisco-voip  On Behalf Of Lelio 
>>> Fulgenzi
>>> Sent: Thursday, October 10, 2019 1:55 PM
>>> To: Casper, Steven 
>>> Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net) 
>>> 

Re: [cisco-voip] SIP Domain substitution

[NateCCIE]

I didn’t see any reference to cloud registered…  I should stop looking at my 
email tonight.

 

From: Anthony Holloway  
Sent: Friday, October 4, 2019 9:11 PM
To: Ryan Huff 
Cc: NateCCIE ; cisco-voip voyp list 

Subject: Re: [cisco-voip] SIP Domain substitution

 

On a Friday night no less.

 

On Fri, Oct 4, 2019 at 10:08 PM Ryan Huff mailto:ryanh...@outlook.com> > wrote:

Come on... we are geeks here we are going to run this down every possible  
avenue regardless :)

Sent from my iPhone





On Oct 4, 2019, at 23:06, Anthony Holloway mailto:avholloway%2bcisco-v...@gmail.com> > wrote:

 

I think Lelio was wondering about a pure cloud registered device, and then 
simply purchasing a vanity domain to overlay on top of the ugly webex one. 

 

You knowlike URL shortening 
<https://nam12.safelinks.protection.outlook.com/?url=https%3A%2F%2Fblog.rebrandly.com%2Fthe-history-of-url-shorteners%2F=02%7C01%7C%7C4df4b81e7de446bcd2fa08d749410008%7C84df9e7fe9f640afb435%7C1%7C0%7C637058415823445677=BK7X5LPEj%2FERkvJhhoKkxhGYbUFaMYDYuHIE3rjPGsE%3D=0>
 . 

 

On Fri, Oct 4, 2019 at 9:57 PM NateCCIE mailto:natec...@gmail.com> > wrote:

Doesn’t cucm have the ability to look at the user portion of the URI only?  For 
like when you’re routing to a DN?  Or I think you can add the short domain to 
the list of the CUCM “owned” domains in enterprise parameters.

 

From: Ryan Huff mailto:ryanh...@outlook.com> > 
Sent: Friday, October 4, 2019 8:51 PM
To: NateCCIE mailto:natec...@gmail.com> >
Cc: Lelio Fulgenzi mailto:le...@uoguelph.ca> >; cisco-voip 
voyp list mailto:cisco-voip@puck.nether.net> >
Subject: Re: [cisco-voip] SIP Domain substitution

 

Hey Nate ... the original ask I think, was to do it all with DNS only and no 
intervention at layer 4, which to my knowledge, DNS alone couldn’t do.  

 

Expressway search rule, CUCM LUA script... etc could all do it in reality.

 

However, the actual goal appears to be dialing a Webex cloud registered codec, 
using a non cloud uri (...@rooms.webex.com <mailto:...@rooms.webex.com> ), and 
for that Webex Hybrid calling with Expressway B2B would get you there, and also 
checks the “no additional transformation needed” box.

 

Sent from my iPhone

 

On Oct 4, 2019, at 22:41, NateCCIE mailto:natec...@gmail.com> > wrote:

 I am not thinking right?  Can’t a dns srv get the call routed to a specific 
host? Then a quick expressway transform to change the domain, and you’re done. 

 

Think of it as a different internal domain va external domain.

 

f...@company.com <mailto:f...@company.com>  does goes to 
expressway.companyinfrastructuredomain.com 
<https://nam12.safelinks.protection.outlook.com/?url=http%3A%2F%2Fexpressway.companyinfrastructuredomain.com=02%7C01%7C%7C4df4b81e7de446bcd2fa08d749410008%7C84df9e7fe9f640afb435%7C1%7C0%7C637058415823445677=26dr97QxoCEhEb11y0lmKT7btioRrDIgcTh7f0lpp%2FQ%3D=0>
  which does a quick trans to foo@internal.local <mailto:foo@internal.local> 

 

 

Sent from my iPhone

 

On Oct 4, 2019, at 8:36 PM, Lelio Fulgenzi mailto:le...@uoguelph.ca> > wrote:

 

 

Interesting. I’ll have to look into that.  Thx. 

-sent from mobile device-

 

Lelio Fulgenzi, B.A. | Senior Analyst

Computing and Communications Services | University of Guelph

Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 2W1

519-824-4120 Ext. 56354   | le...@uoguelph.ca 
<mailto:le...@uoguelph.ca> 

 

www.uoguelph.ca/ccs 
<https://nam12.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.uoguelph.ca%2Fccs=02%7C01%7C%7C4df4b81e7de446bcd2fa08d749410008%7C84df9e7fe9f640afb435%7C1%7C0%7C637058415823455686=6OmIEsttbas7XV2w7vZ6PFeC7tYVDLvVx%2FTGUFyxjbE%3D=0>
  | @UofGCCS on Instagram, Twitter and Facebook

 




On Oct 4, 2019, at 10:32 PM, Ryan Huff mailto:ryanh...@outlook.com> > wrote:

Webex Hybrid Calling (with Expressway B2B), could in theory, help accomplish 
this. The codec is still cloud registered, though Hybrid calling would allow 
for an on-prem URI to be associated with the Webex remote destination of the 
codec.  

 

The call would come into the on-prem URI via B2B like normal, and assuming the 
Hybrid integration was setup correctly, ring the Webex remote destination which 
rings the cloud registered codec.

 

It’s a little bit of an ugly trombone, but it does work..

 

Sent from my iPhone

 

On Oct 4, 2019, at 22:09, Lelio Fulgenzi mailto:le...@uoguelph.ca> > wrote:

 

 

Darn. Double darn.  

 

Let’s hope webex offers up custom domain registration for devices soon. 

 

‘Cause room...@acme.rooms.webex.com <mailto:room...@acme.rooms.webex.com>  is a 
bit much. 

-sent from mobile device-

 

Lelio Fulgenzi, B.A. | Senior Analyst

Computing and Communications Services | University of Guelph

Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N

Re: [cisco-voip] SIP Domain substitution

[NateCCIE]

Doesn’t cucm have the ability to look at the user portion of the URI only?  For 
like when you’re routing to a DN?  Or I think you can add the short domain to 
the list of the CUCM “owned” domains in enterprise parameters.

 

From: Ryan Huff  
Sent: Friday, October 4, 2019 8:51 PM
To: NateCCIE 
Cc: Lelio Fulgenzi ; cisco-voip voyp list 

Subject: Re: [cisco-voip] SIP Domain substitution

 

Hey Nate ... the original ask I think, was to do it all with DNS only and no 
intervention at layer 4, which to my knowledge, DNS alone couldn’t do.  

 

Expressway search rule, CUCM LUA script... etc could all do it in reality.

 

However, the actual goal appears to be dialing a Webex cloud registered codec, 
using a non cloud uri (...@rooms.webex.com <mailto:...@rooms.webex.com> ), and 
for that Webex Hybrid calling with Expressway B2B would get you there, and also 
checks the “no additional transformation needed” box.

 

Sent from my iPhone





On Oct 4, 2019, at 22:41, NateCCIE mailto:natec...@gmail.com> > wrote:

 I am not thinking right?  Can’t a dns srv get the call routed to a specific 
host? Then a quick expressway transform to change the domain, and you’re done. 

 

Think of it as a different internal domain va external domain.

 

f...@company.com <mailto:f...@company.com>  does goes to 
expressway.companyinfrastructuredomain.com which does a quick trans to 
foo@internal.local <mailto:foo@internal.local> 

 

 

Sent from my iPhone





On Oct 4, 2019, at 8:36 PM, Lelio Fulgenzi mailto:le...@uoguelph.ca> > wrote:

 

 

Interesting. I’ll have to look into that.  Thx. 

-sent from mobile device-





Lelio Fulgenzi, B.A. | Senior Analyst

Computing and Communications Services | University of Guelph

Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 2W1 
 

519-824-4120 Ext. 56354   | le...@uoguelph.ca 
<mailto:le...@uoguelph.ca> 

 

www.uoguelph.ca/ccs 
<https://nam12.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.uoguelph.ca%2Fccs=02%7C01%7C%7C9b720a4af69b41bebb5008d7493d7691%7C84df9e7fe9f640afb435%7C1%7C0%7C637058400631419029=Bm%2FOh%2Fp3TDYkOYis27I47D3rrDAJDEp4doaBw5lr9XA%3D=0>
  | @UofGCCS on Instagram, Twitter and Facebook

 




On Oct 4, 2019, at 10:32 PM, Ryan Huff mailto:ryanh...@outlook.com> > wrote:

Webex Hybrid Calling (with Expressway B2B), could in theory, help accomplish 
this. The codec is still cloud registered, though Hybrid calling would allow 
for an on-prem URI to be associated with the Webex remote destination of the 
codec.  

 

The call would come into the on-prem URI via B2B like normal, and assuming the 
Hybrid integration was setup correctly, ring the Webex remote destination which 
rings the cloud registered codec.

 

It’s a little bit of an ugly trombone, but it does work..

 

Sent from my iPhone





On Oct 4, 2019, at 22:09, Lelio Fulgenzi mailto:le...@uoguelph.ca> > wrote:

 

 

Darn. Double darn.  

 

Let’s hope webex offers up custom domain registration for devices soon. 

 

‘Cause room...@acme.rooms.webex.com <mailto:room...@acme.rooms.webex.com>  is a 
bit much. 

-sent from mobile device-





Lelio Fulgenzi, B.A. | Senior Analyst

Computing and Communications Services | University of Guelph

Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 2W1 
 

519-824-4120 Ext. 56354   | le...@uoguelph.ca 
<mailto:le...@uoguelph.ca> 

 

www.uoguelph.ca/ccs 
<https://nam12.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.uoguelph.ca%2Fccs=02%7C01%7C%7C9b720a4af69b41bebb5008d7493d7691%7C84df9e7fe9f640afb435%7C1%7C0%7C637058400631419029=Bm%2FOh%2Fp3TDYkOYis27I47D3rrDAJDEp4doaBw5lr9XA%3D=0>
  | @UofGCCS on Instagram, Twitter and Facebook

 




On Oct 4, 2019, at 9:05 PM, Ryan Huff mailto:ryanh...@outlook.com> > wrote:

What it sounds like you are trying to do to me, is allow the call to ultimately 
setup with a URI different than the URI that was dialed, without the calling 
party being the wiser. 

 

DNS won’t be able to do anything with regards to that I don’t think, because it 
really sounds like you’re trying to manipulate/transform the called URI, and 
you’ll need something to interact with the SIP message stack for that I’d think.

 

You can create a round robin A record, that resolves to multiple IP addresses, 
so when the client looks up the DNS SRV, it receives multiple targets to try 
before considering the SRV target “unreachable” (SRV weights and priorities 
determine the ordering of the target addresses resolved for the client). 
However, this won’t have the ability to change the called URI, which is 
ultimately what I think you’re attempting in the scenario (DNS and SIP messages 
are on different networking layers).

 

As Dave mentioned below, Expressway or a LUA script (sip normalization) in CUCM 
seems to be uniquely qualified for what you’re wanting to do.

Sent from my 

Re: [cisco-voip] SIP Domain substitution

[NateCCIE]

I am not thinking right?  Can’t a dns srv get the call routed to a specific 
host? Then a quick expressway transform to change the domain, and you’re done.

Think of it as a different internal domain va external domain.

f...@company.com does goes to expressway.companyinfrastructuredomain.com which 
does a quick trans to foo@internal.local


Sent from my iPhone

> On Oct 4, 2019, at 8:36 PM, Lelio Fulgenzi  wrote:
> 
> 
> 
> Interesting. I’ll have to look into that.  Thx. 
> 
> -sent from mobile device-
> 
> Lelio Fulgenzi, B.A. | Senior Analyst
> Computing and Communications Services | University of Guelph
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 
> 2W1
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>  
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>  
> 
> 
>> On Oct 4, 2019, at 10:32 PM, Ryan Huff  wrote:
>> 
>> Webex Hybrid Calling (with Expressway B2B), could in theory, help accomplish 
>> this. The codec is still cloud registered, though Hybrid calling would allow 
>> for an on-prem URI to be associated with the Webex remote destination of the 
>> codec. 
>> 
>> The call would come into the on-prem URI via B2B like normal, and assuming 
>> the Hybrid integration was setup correctly, ring the Webex remote 
>> destination which rings the cloud registered codec.
>> 
>> It’s a little bit of an ugly trombone, but it does work..
>> 
>> Sent from my iPhone
>> 
>>> On Oct 4, 2019, at 22:09, Lelio Fulgenzi  wrote:
>>> 
>>> 
>>> 
>>> Darn. Double darn. 
>>> 
>>> Let’s hope webex offers up custom domain registration for devices soon. 
>>> 
>>> ‘Cause room...@acme.rooms.webex.com is a bit much. 
>>> 
>>> -sent from mobile device-
>>> 
>>> Lelio Fulgenzi, B.A. | Senior Analyst
>>> Computing and Communications Services | University of Guelph
>>> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 
>>> 2W1
>>> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>>>  
>>> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>>>  
>>> 
>>> 
>>> On Oct 4, 2019, at 9:05 PM, Ryan Huff  wrote:
>>> 
 What it sounds like you are trying to do to me, is allow the call to 
 ultimately setup with a URI different than the URI that was dialed, 
 without the calling party being the wiser.
 
 DNS won’t be able to do anything with regards to that I don’t think, 
 because it really sounds like you’re trying to manipulate/transform the 
 called URI, and you’ll need something to interact with the SIP message 
 stack for that I’d think.
 
 You can create a round robin A record, that resolves to multiple IP 
 addresses, so when the client looks up the DNS SRV, it receives multiple 
 targets to try before considering the SRV target “unreachable” (SRV 
 weights and priorities determine the ordering of the target addresses 
 resolved for the client). However, this won’t have the ability to change 
 the called URI, which is ultimately what I think you’re attempting in the 
 scenario (DNS and SIP messages are on different networking layers).
 
 As Dave mentioned below, Expressway or a LUA script (sip normalization) in 
 CUCM seems to be uniquely qualified for what you’re wanting to do.
 
 Sent from my iPhone
 
> On Oct 4, 2019, at 20:40, Lelio Fulgenzi  wrote:
> 
> 
> 
> I’ve seen some references to Cisco SIP proxy server. 
> 
> Would that help?
> 
> -sent from mobile device-
> 
> Lelio Fulgenzi, B.A. | Senior Analyst
> Computing and Communications Services | University of Guelph
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | 
> N1G 2W1
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>  
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>  
> 
> 
> On Oct 4, 2019, at 7:46 PM, Ryan Huff  wrote:
> 
>> According to RFC 2782 (https://www.ietf.org/rfc/rfc2782.txt), it does 
>> not, under the “Target Definition”; “there must be one or more address 
>> records for this name, the name must not be an alias”.
>> 
>> However, I can tell you that I have used a CNAME in the SRV target field 
>> before, and it appeared to work at the time. Still, depending on the 
>> application, doing so could potentially cause some weird issue with 
>> regards to PTR or something.
>> 
>> Sent from my iPhone
>> 
>>> On Oct 4, 2019, at 19:10, Brian Meade  wrote:
>>> 
>>> 
>>> I don't think DNS SRV records support CNAME.  Even then, it would only 
>>> change where it was sent to and not the SIP headers.
>>> 
 On Fri, Oct 4, 2019 at 12:26 PM Lelio Fulgenzi  
 wrote:
 Yeah – I’d want this to happen all within DNS. But of course, in a 
 supported fashion. I’m not interested in spending time modifying 
 infrastructure at this time.
 
  
 

Re: [cisco-voip] Single user with multiple device profiles?

[NateCCIE]

You can have lots, especially for different device types , I seem to remember 
if there is more than one profile that would work, a list is presented to the 
user when they log in.

 

From: cisco-voip  On Behalf Of Nick Barnett
Sent: Wednesday, September 18, 2019 9:36 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Single user with multiple device profiles?

 

For some reason I thought I'd seen this before, but it's eluding me. I have a 
need for a single user to have multiple extension mobility device profiles. I 
can't even find where it is supported or unsupported. Anyone have any advice on 
how to get this accomplished, whether  it's a trick or a formal procedure?

 

Thanks,

Nick

 

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Re: [cisco-voip] IM Upgrade Steps during CUCM upgrade

[NateCCIE]

Yeah, I HATE this bug.  Why in the world can’t the docwiki or what ever it’s 
called be updated quicker than a bug be filed/made universally known, and who 
came up with these recommendations TAC or the BU.

 

But I have seem IMP just not start services, and adding resources magic fix it.

 

From: cisco-voip  On Behalf Of Lelio 
Fulgenzi
Sent: Tuesday, May 28, 2019 9:46 AM
To: Bill Talley 
Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net) 

Subject: Re: [cisco-voip] IM Upgrade Steps during CUCM upgrade

 

 

In the words of the immortal Chris Farley….

 

holy schnikes

 

If possible, it is recommended to have 4vCPU and 8 GB RAM as we are seeing more 
cases with high CPU due to resources related.

 

They want 4 vCPU if possible? They think these things grow on trees? 

 

Right now, our two IMP servers are at 2 vCPU and 4GB of RAM. (5000 user OVA).

 

I’ll have to see about coordinating this change as well. We don’t have a lot of 
capacity/activity on these servers, so I think we should be OK for now.

 

Funny thing – Bug updated May 20, 2019, but virtualization docs still show old 
OVA information.

 

---

Lelio Fulgenzi, B.A. | Senior Analyst

Computing and Communications Services | University of Guelph

Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 2W1

519-824-4120 Ext. 56354 |   le...@uoguelph.ca

 

  www.uoguelph.ca/ccs | @UofGCCS on Instagram, 
Twitter and Facebook

 



 

From: Bill Talley mailto:btal...@gmail.com> > 
Sent: Tuesday, May 28, 2019 11:08 AM
To: Lelio Fulgenzi mailto:le...@uoguelph.ca> >
Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net 
 ) mailto:cisco-voip@puck.nether.net> >
Subject: Re: [cisco-voip] IM Upgrade Steps during CUCM upgrade

 

That’s the process I typically follow without issues.

 

Also, I can’t recall if this was posted here, but wanted to make sure you’ve 
seen the recent changes to resource requirements for IM   This may not apply 
to you if you have more than 5000 users.

 

IM VM resource requirements needs to be updated

CSCvk65006

Description

Symptom:
IM version 11.5.1.x or 12.0.1.x installed using one of the following 
configuration:
150 users (Full UC) 1vCPU 2 GB RAM
1,000 users (Full UC) 1vCPU 4 GB RAM
5,000 users (Full UC) 2vCPU 4 GB RAM

Customers with any of the above configuration might notice an increase use of 
CPU and Memory resources.

This can be fixed by manually increasing the resources according to the table 
below:
150 users (Full UC) 2vCPU 8 GB RAM
1,000 users (Full UC) 2vCPU 8 GB RAM
5,000 users (Full UC) 2vCPU 8 GB RAM

Conditions:
Performance Issues

Workaround:
Manually increase resources according to the table below:
150 users (Full UC) 2vCPU 8 GB RAM
1,000 users (Full UC) 2vCPU 8 GB RAM
5,000 users (Full UC) 2vCPU 8 GB RAM

If possible, it is recommended to have 4vCPU and 8 GB RAM as we are seeing more 
cases with high CPU due to resources related.

 

 

 

Sent from an iOS device with very tiny touchscreen input keys.  Please excude 
my typtos.


On May 28, 2019, at 8:47 AM, Lelio Fulgenzi mailto:le...@uoguelph.ca> > wrote:


I'm reading the upgrade guide, specifically, the time sequencing:

https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/upgrade/11_5_1/cucm_b_upgrade-guide-cucm-115/cucm_b_upgrade-guide-cucm-115_chapter_010010.html

and it mentions upgrading the IM publisher (to inactive partition) at the 
same time as upgrading the subscribers. Then doing a switch version on the IMP 
pub at the same time as the CUCM subs.

Anyone do this parallel type upgrade before?

Sure would save a lot of time.

Lelio


---
Lelio Fulgenzi, B.A. | Senior Analyst
Computing and Communications Services | University of Guelph
Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 2W1
519-824-4120 Ext. 56354 | le...@uoguelph.ca  


www.uoguelph.ca/ccs   | 
@UofGCCS on Instagram, Twitter and Facebook

[University of Guelph Cornerstone with Improve Life tagline]



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Re: [cisco-voip] IM Upgrade Steps during CUCM upgrade

[NateCCIE]

Lelio,

 

Are you apposed to upgrading the inactive partition during working hours?  For 
my planning, the upgrade window starts when the bumps in service occur, which 
is usually the switch version.

 

There are some cases where the upgrade is not “online” there is a R1/R2 upgrade 
designation for this, where it is also upgrading the OS, but those are fairly 
rare.

 

Thanks,

-Nate

 

From: cisco-voip  On Behalf Of Lelio 
Fulgenzi
Sent: Tuesday, May 28, 2019 8:23 AM
To: Ryan Huff 
Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net) 

Subject: Re: [cisco-voip] IM Upgrade Steps during CUCM upgrade

 

 

Sweet. Thanks. I need to map out the window I have and see how this all fits 
in. 

 

I really hate how the version compatibility is _so_ tight with the IMP cluster 
now.  One more thing to have to worry about.

 

Oh well.

 

TAC just gave me some advice to see how long the DB replication would take 
after the switch version. Ahead of time, I do a “utils dbreplication status” 
and then “utils dbreplication runtimestate” until all tables show synced and 
take that time.

 

Does that sound about right?

 

---

Lelio Fulgenzi, B.A. | Senior Analyst

Computing and Communications Services | University of Guelph

Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 2W1

519-824-4120 Ext. 56354 | le...@uoguelph.ca  

 

  www.uoguelph.ca/ccs | @UofGCCS on Instagram, 
Twitter and Facebook

 



 

From: Ryan Huff mailto:ryanh...@outlook.com> > 
Sent: Tuesday, May 28, 2019 10:04 AM
To: Lelio Fulgenzi mailto:le...@uoguelph.ca> >
Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net 
 ) mailto:cisco-voip@puck.nether.net> >
Subject: Re: [cisco-voip] IM Upgrade Steps during CUCM upgrade

 

This is what I do for an in-place upgrade with no other changes (hostname, IP 
address.. etc). 

 

I do the cucm pub first (obviously), then imp pub (which is more like a cucm 
sub for upgrade purposes), then all the cucm and imp subs; everything into the 
inactive pt. 

 

I do the pubs individually, then I’ll do a couple subs at a time .. etc.

 

Next I Switch version on the pubs; cucm, imp then the subs.

 

On the switch version, I wait till one node is fully up (tomcat started) before 
switching another.

 

May not be as efficient as it could be, but has kept me out of trouble thus 
far; plan your dive, dive your plan.


On May 28, 2019, at 09:48, Lelio Fulgenzi mailto:le...@uoguelph.ca> > wrote:

 

I’m reading the upgrade guide, specifically, the time sequencing:

 

https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/upgrade/11_5_1/cucm_b_upgrade-guide-cucm-115/cucm_b_upgrade-guide-cucm-115_chapter_010010.html
 

 

 

and it mentions upgrading the IM publisher (to inactive partition) at the 
same time as upgrading the subscribers. Then doing a switch version on the IMP 
pub at the same time as the CUCM subs.

 

Anyone do this parallel type upgrade before? 

 

Sure would save a lot of time.

 

Lelio

 

 

---

Lelio Fulgenzi, B.A. | Senior Analyst

Computing and Communications Services | University of Guelph

Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 2W1

519-824-4120 Ext. 56354 | le...@uoguelph.ca  

 

 

 www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook

 



 

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Re: [cisco-voip] CM routing calls based on facility IE

[NateCCIE]

Friends don’t let friends use MGCP.  You can still use PRI and use SIP between 
the GW and CUCM.


Sent from my iPhone

> On May 24, 2019, at 1:24 PM, Ovidiu Popa  wrote:
> 
> Hi Tommy
> 
> Good suggestion but I forgot to mention that we are using MGCP.
> 
> Thanks,
> Ovidiu
> 
>> On Fri, May 24, 2019 at 9:15 PM Schlotterer, Tommy 
>>  wrote:
>> You could use a sip profile if you are using sip:
>> 
>>  
>> 
>> voice class sip-profiles 100
>> 
>> rule 1 request INVITE sip-header Diversion copy "<(..)" u01
>> 
>>  rule 2 request INVITE sip-header To modify "sip:44085551234(.*)" "\u01\1"
>> 
>>  rule 3 request INVITE sip-header SIP-Req-URI modify "sip: 44085551234(.*)" 
>> "\u01\1"
>> 
>>  rule 4 request INVITE sip-header Diversion remove
>> 
>>  
>> 
>> Thanks
>> 
>> 
>> Tommy
>> 
>>  
>> 
>> From: cisco-voip  On Behalf Of Ovidiu 
>> Popa
>> Sent: Friday, May 24, 2019 2:53 PM
>> To: cisco-voip 
>> Subject: [cisco-voip] CM routing calls based on facility IE
>> 
>>  
>> 
>> EXTERNAL EMAIL
>> 
>>  
>> 
>> Hello everyone
>> 
>>  
>> 
>> Does anyone know if CUCM is capable of routing calls based on the 
>> information in the ISDN Facility IE ?
>> 
>>  
>> 
>> We tested our provider's call failover scenario and we have the following 
>> issue :
>> 
>> - The operator enables the failover 
>> 
>> - All calls from our primary site A to our secondary site B
>> 
>> - All calls received on site B have the same value in the Called Party 
>> Number field - the unique number identifying the site B. The number that was 
>> initially dialed on site A is now placed in the facility IE. 
>> 
>>  
>> 
>> According to  ETS 300 196-1 the Facility Element is used only for 
>> supplementary services not for routing calls. 
>> 
>>  
>> 
>> 11.2.2.1 Facility
>> The purpose of the Facility information element is to carry components. For 
>> the purposes of this ETS these components are used to control supplementary 
>> services.
>> 
>>  
>> 
>> So my question is: can CUCM route calls based on the Facility IE ? 
>> 
>>  
>> 
>> Thanks for any assistance.
>> 
>>  
>> 
>> Ovidiu
>> 
>>  
>> 
>>  
>> 
>> 
>> Tommy Schlotterer | Systems Engineer - Collaboration
>> Presidio (NASDAQ: PSDO) | presidio.com
>> 20 N Saint Clair 3rd Floor, Toledo, OH 43604
>> D: 419.214.1415 | C: 419.706.0259 | tschlotte...@presidio.com
>> 
>> 
>> 
>> 
>> This message w/attachments (message) is intended solely for the use of the 
>> intended recipient(s) and may contain information that is privileged, 
>> confidential or proprietary. If you are not an intended recipient, please 
>> notify the sender, and then please delete and destroy all copies and 
>> attachments. Please be advised that any review or dissemination of, or the 
>> taking of any action in reliance on, the information contained in or 
>> attached to this message is prohibited.
>> 
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Re: [cisco-voip] PUT Tool Bootables - what version?

[NateCCIE]

And use a segmented downloaded like flareget to get it much faster from the 
special file access site. 

Sent from my iPhone

> On May 15, 2019, at 12:34 PM, Charles Goldsmith  wrote:
> 
> I've always opened a TAC case, specified the reason for needing bootable 
> (rebuilding a cluster usually), and they provided it.  I've never had an 
> issue getting them, just takes a bit of time.
> 
>> On Wed, May 15, 2019 at 1:16 PM Evgeny Izetov  wrote:
>> Yeah, CUPS has always been bootable.. CUCM/CUC/CER are still not
>> 
>> So, what is the proper way to obtaining bootable iso's now? Let's say a CUCM 
>> 11.5 SU6 needs to be reinstalled, and there's no bootable because it was 
>> upgraded from an earlier SU. PUT does not have bootable SU6 and neither does 
>> Enterprise Agreement. Is TAC the only way to get the bootable for a specific 
>> SU? I believe there used to be a time when everyone was advised that TAC is 
>> not able to provide bootables?
>> 
>>> On Wed, May 15, 2019 at 12:18 PM Lelio Fulgenzi  wrote:
>>>  
>>> 
>>> Same with CUPS if I’m not mistaken.
>>> 
>>>  
>>> 
>>> ---
>>> 
>>> Lelio Fulgenzi, B.A. | Senior Analyst
>>> 
>>> Computing and Communications Services | University of Guelph
>>> 
>>> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 
>>> 2W1
>>> 
>>> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>>> 
>>>  
>>> 
>>> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>>> 
>>>  
>>> 
>>> 
>>> 
>>>  
>>> 
>>> From: cisco-voip  On Behalf Of Charles 
>>> Goldsmith
>>> Sent: Wednesday, May 15, 2019 12:09 PM
>>> To: Evgeny Izetov 
>>> Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net) 
>>> 
>>> Subject: Re: [cisco-voip] PUT Tool Bootables - what version?
>>> 
>>>  
>>> 
>>> Plus, UCCX is shipping bootables (filename doesn't reflect it).
>>> 
>>>  
>>> 
>>> Description :
>>> 
>>> UCCX 12.0(1) image for fresh install and upgrades.
>>> 
>>> UCSInstall_UCCX_12_0_1_UCOS_12.0.1.1-24.sgn.iso
>>> 
>>>  
>>> 
>>>  
>>> 
>>> On Wed, May 15, 2019 at 11:04 AM Evgeny Izetov  wrote:
>>> 
>>> Wasn't their excuse with not providing bootables that it was based on Red 
>>> Hat? It's CentOS now, and still a struggle..
>>> 
>>>  
>>> 
>>> On Wed, May 15, 2019 at 11:52 AM Brian Meade  wrote:
>>> 
>>> I've given up on trying to get bootables.  I haven't had any issues with 
>>> ones made with UltraISO.
>>> 
>>>  
>>> 
>>> On Wed, May 15, 2019 at 11:39 AM Lelio Fulgenzi  wrote:
>>> 
>>> 
>>> Just wondering what the Put Tool Bootables are at now? We're planning on 
>>> upgrading to v11.5.1 SU6 due to the field notice and I'd like to have the 
>>> bootable available.
>>> 
>>> Otherwise it's opening a case with the TAC, etc.
>>> 
>>> Is it just a matter of submit request and check the filename?
>>> 
>>> Lelio
>>> 
>>> ---
>>> Lelio Fulgenzi, B.A. | Senior Analyst
>>> Computing and Communications Services | University of Guelph
>>> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 
>>> 2W1
>>> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>>> 
>>> www.uoguelph.ca/ccs | @UofGCCS on Instagram, 
>>> Twitter and Facebook
>>> 
>>> [University of Guelph Cornerstone with Improve Life tagline]
>>> 
>>> ___
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>>> 
>>> ___
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>>> 
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>>> 
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Re: [cisco-voip] vg310/320 or ISR4K with analog modules

[NateCCIE]

VG248 was the best one ever made by Cisco.  Not IOS based made the thing
super stable and easy to configure/manage.  Makes me sad when ever I think
about it.



-Original Message-
From: cisco-voip  On Behalf Of Lelio
Fulgenzi
Sent: Wednesday, April 3, 2019 12:35 PM
To: voyp list, cisco-voip (cisco-voip@puck.nether.net)

Subject: [cisco-voip] vg310/320 or ISR4K with analog modules


I see the VG350 has pretty much been EOL'd (February 29, 2024) in favour of
ISR4K with high density voice service modules or a VG450 (which is ISR4K
based).

https://www.cisco.com/c/en/us/products/collateral/unified-communications/vg-
series-gateways/eos-eol-notice-c51-741597.html

Anybody know if the VG310 and VG320 is going that path? It would be hard to
assume so, since the VG400s are very low density and look like they're
replacing the VG202/VG204.

I can't imagine having to fork out for a ISR4K for 24 or 48 analog ports.

Thoughts?


---
Lelio Fulgenzi, B.A. | Senior Analyst
Computing and Communications Services | University of Guelph
Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G
2W1
519-824-4120 Ext. 56354 | le...@uoguelph.ca

www.uoguelph.ca/ccs | @UofGCCS on Instagram,
Twitter and Facebook

[University of Guelph Cornerstone with Improve Life tagline]


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Re: [cisco-voip] SIp Trunk call failing after PBX upgrade

[NateCCIE]

 

Cause No. 65 - bearer capability not implemented.
This cause indicates that the equipment sending this cause does not support
the bearer capability requested.

What it means:



1.  In most cases, the number being called is not an ISDN number but an
analog destination.
2.  The equipment is dialing at a faster rate than the circuitry allows,
for example, dialing at 64K when only 56K is supported.

 

Where is the call going, out a gateway or just a Cisco phone?

 

From: ROZA, Ariel  
Sent: Monday, March 25, 2019 2:03 PM
To: NateCCIE ; 'cisco-voip' 
Subject: RE: [cisco-voip] SIp Trunk call failing after PBX upgrade

 

That was the original setting, and the results is what I included in the
mail

 

De: NateCCIE mailto:natec...@gmail.com> > 
Enviado el: lunes, 25 de marzo de 2019 17:01
Para: ROZA, Ariel mailto:ariel.r...@la.logicalis.com> >; 'cisco-voip'
mailto:cisco-voip@puck.nether.net> >
Asunto: RE: [cisco-voip] SIp Trunk call failing after PBX upgrade

 

I would change preferred codec to 711a and see what happens.

 

From: ROZA, Ariel mailto:ariel.r...@la.logicalis.com> > 
Sent: Monday, March 25, 2019 1:37 PM
To: NateCCIE mailto:natec...@gmail.com> >; 'cisco-voip'
mailto:cisco-voip@puck.nether.net> >
Subject: RE: [cisco-voip] SIp Trunk call failing after PBX upgrade

 

Yes I already looked at that /1. According to the RFC, the /1 denotes the
quantity of channels and it is optional when the codec uses only one
channel.

 

I looked up posible bugs related to that in the Bug Search Tool and did not
find anything suitable.

Already tried changing the Preferred codec to G711U and got the same
results, except the output now shows PCMU/8000 from CUCM side, as expected.

 

Thanks, Nate.

 

De: NateCCIE mailto:natec...@gmail.com> > 
Enviado el: lunes, 25 de marzo de 2019 14:33
Para: ROZA, Ariel mailto:ariel.r...@la.logicalis.com> >; 'cisco-voip'
mailto:cisco-voip@puck.nether.net> >
Asunto: RE: [cisco-voip] SIp Trunk call failing after PBX upgrade

 

Non working call shows G711u and a, working call shows only a.  there is
also a difference of the /1 at the end not sure what that indicates.

 

a=rtpmap:0 PCMU/8000/1

a=rtpmap:8 PCMA/8000

 

 

From: cisco-voip mailto:cisco-voip-boun...@puck.nether.net> > On Behalf Of ROZA, Ariel
Sent: Monday, March 25, 2019 11:17 AM
To: cisco-voip (cisco-voip@puck.nether.net
<mailto:cisco-voip@puck.nether.net> ) mailto:cisco-voip@puck.nether.net> >
Subject: [cisco-voip] SIp Trunk call failing after PBX upgrade

 

Hi, guys and gals.

 

I have a customer with a CUCM 9.0(2) cluster.

It is connected to a SIP PBX via a direct SIP TRUNK (No SBC, CUBE or
otherwise). The PBX has four different nodes, all configured in the SIP
TRUNK

 

They claim it was working fine until last Thursday, where they did an
upgrade to one of the nodes of the PBX. After that, calls going from PBX to
CUCM fail with a 488 Media Not Acceptable error.

They also have tried making calls from one of the not upgraded nodes, with
the same error.

I have been looking into the SIP traces, and I see nothing really telling of
a problem there.

 

We reseted the SIP trunk with no success.

I have looked at the región configuration, and all regions are set to the
System Default (G722, G711)

I also tried changing the preferred codec in the SIP trunk, with no success.

 

Following this, I am pasting the SIP messages of a failed call from PBX ->
CUCM and a successfull call in the reverse, from CUCM -> PBX.

 

Can you see if anything is wrong or odd?

 

Regards,

 

Ariel.

 

Failed Call from PBX



 

INVITE sip:3366@10.4.128.27 SIP/2.0

Via: SIP/2.0/UDP
172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm

From: " " ;tag=2792862

To: 

Call-ID: 501227892-15@172.27.0.15 <mailto:501227892-15@172.27.0.15> 

CSeq: 1 INVITE

Contact: 

Max-Forwards: 70

User-Agent: MitE1x v4.4.5.1062

Expires: 300

Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO

P-Early-Media: Supported

P-Asserted-Identity: " " 

P-Mitrol-idLlamada: 190322160050689_MIT_07437

P-Mitrol-LoginID: 

P-Mitrol-PerfilRuteo: 100

Content-Length: 233

Content-Type: application/sdp

v=0

o=86329 -835641967 1 IN IP4 172.27.0.15

s=MitE1x Call

c=IN IP4 172.27.0.15

t=0 0

m=audio 36112 RTP/AVP 0 8 101

a=sendrecv

a=rtpmap:0 PCMU/8000/1

a=rtpmap:8 PCMA/8000/1

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

 

Reply from CUCM

---

 

SIP/2.0 488 Not Acceptable Media

Via: SIP/2.0/UDP
172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm

From: "Gabriel Querol" ;tag=2792862

To: ;tag=573234994

Date: Fri, 22 Mar 2019 19:00:23 GMT

Call-ID: 501227892-15@172.27.0.15 <mailto:501227892-15@172.27.0.15> 

CSeq: 1 INVITE

Allow-Events: presence

Warning: 304 10.4.128.27 "Media Type(s) Unavailable"

Reason: 

Re: [cisco-voip] Jabber Guest as a “live chat for help” solution

[NateCCIE]

For chat you want social miner connected to UCCX.  Jabber guest is just video, 
and I don’t think has gotten much love.

 

From: cisco-voip  On Behalf Of Lelio 
Fulgenzi
Sent: Saturday, February 16, 2019 9:38 AM
To: cisco-voip voyp list 
Subject: [cisco-voip] Jabber Guest as a “live chat for help” solution

 

 

>From what I gather, Jabber guest is mainly for facilitating live video chat, 
>not live IM chat.  

 

Am I wrong there? A team has been looking at putting a live help chat tool on 
our website. I thought weather Jabber guest could help us with that. 

 

Thoughts?

-sent from mobile device-





Lelio Fulgenzi, B.A. | Senior Analyst

Computing and Communications Services | University of Guelph

Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 2W1 
 

519-824-4120 Ext. 56354   | le...@uoguelph.ca 
 

 

www.uoguelph.ca/ccs   | @UofGCCS on Instagram, 
Twitter and Facebook

 



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Re: [cisco-voip] Hybrid Calendar Connector

[NateCCIE]

I think it’s a global admin to authorize the connection, but the O365 auths the 
connection and it stays up.   I have always had the 0365 admin use their own 
admin account and not a “service account with lots of privileges with a 
password that doesn’t change” like Unity Connection to exchange.

 

From: cisco-voip  On Behalf Of Erick 
Wellnitz
Sent: Friday, February 8, 2019 9:05 AM
To: Cisco VoIP Group 
Subject: [cisco-voip] Hybrid Calendar Connector

 

Has anyone set up the O365 hybrid connector (cloud to cloud) via Control Hub?

 

My client is concerned with the requirement of needing a global admin account.  
Is this only for setup or is it used like a service account?  Documentation 
eludes to the fact it's only for setup but it isn't clear to me.

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Re: [cisco-voip] SIP Fail over

[NateCCIE]

I think the lowest cucm will use is a 3?

Sent from my iPhone

> On Dec 20, 2018, at 3:35 PM, Anthony Holloway 
>  wrote:
> 
> I have never seen that done before.  I like it!
> 
> Just be careful hard coding your stratum to a value of 2 all the time.  
> Instead it should be a relative value, higher than your reference clock.  Or 
> if you do want a one-size-fits-all stratum, 14 maybe?
> 
> Thanks for sharing that tip Ryan!
> 
> 
> 
>> On Thu, Dec 20, 2018 at 3:52 PM Ryan Huff  wrote:
>> I like ntp master 2 as a fallback, to allow synchronization with the local 
>> device clock in a DR/Outage scenario where I fail sync to the actual 
>> reference clock
>> 
>> Sent from my iPhone
>> 
>> On Dec 20, 2018, at 14:51, Anthony Holloway 
>>  wrote:
>> 
>>> It's very interesting to me the kinds of things people take for granted and 
>>> go a long time without ever being corrected, simply because the people who 
>>> know these things, think it's common knowledge.
>>> 
>>> For example, I had a conversation with a senior collab person once, who 
>>> didn't know that region bit rate settings were a ceiling, and that a lower 
>>> bit rate could be negotiated.  
>>> 
>>> And as another example, Engineers who put ntp master on a router because 
>>> they think this makes the router an NTP server.
>>> 
>>> And as one last example, Engineers who use the ^ symbol at the beginning of 
>>> a Dial Peer destination pattern, not knowing that destination patterns are 
>>> left justified implicitly.  Or alternatively, don't use the $ at the end, 
>>> effectively creating a "begins with" clause, when an "is exactly" clause is 
>>> desired.
>>> 
>>> Someone should start a thread titled: What is something you found out that 
>>> you were wrong about for a long time?
>>> 
 On Thu, Dec 20, 2018 at 1:14 PM Lelio Fulgenzi  wrote:
 
 I’ll be honest. I didn’t know there was a difference.
 
 I’m guessing a SIP trunk to a third party app that is reported as down due 
 to to sip option ping really is down and not some silly networking issue 
 where an icmp ping was failing. 
 
 This is good to know. 
 
 And the last thing I will learn this year. ;)
 
 
 
 -sent from mobile device-
 
 Lelio Fulgenzi, B.A. | Senior Analyst
 Computing and Communications Services | University of Guelph
 Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | 
 N1G 2W1
 519-824-4120 Ext. 56354 | le...@uoguelph.ca
  
 www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
  
 
 
 On Dec 20, 2018, at 1:01 PM, Anthony Holloway 
  wrote:
 
> Erik,
> 
> That's an interesting insight.  It kind of sounds like you think ICMP 
> Ping and SIP OPTIONS Ping are related, but they are completely different.
> 
> Just because you cannot ICMP Ping the SIP Peer at L3, doesn't mean you 
> cannot OPTIONs them.
> 
> Am I understanding your thought process correctly?
> 
>> On Thu, Dec 20, 2018 at 11:53 AM Ryan Huff  wrote:
>> 
>> 
>> Thanks,
>> 
>> Ryan Huff, CCDP, CCNP
>> Cisco Certified Network and Design Professional
>> 
>> From: Ryan Huff 
>> Sent: Thursday, December 20, 2018 12:46 PM
>> To: Erik Anderson
>> Subject: Re: [cisco-voip] SIP Fail over
>>  
>> Not sure what kind of code you're working with but if its modern, you 
>> could try server groups. Here is a snippet from one of mine (using AT 
>> admitidly), sanitized for the NSA ...
>> 
>> voice class server-group 100
>>  ipv4 12.x.x.x preference 1
>>  ipv4 12.x.x.x preference 2
>>  ipv4 12.x.x.x preference 3
>>  ipv4 12.x.x.x preference 1
>>  description PSTN SIGNALING PEERS
>> !
>> voice class server-group 200
>>  ipv4 10.x.x.x preference 3
>>  ipv4 10.x.x.x preference 1
>>  ipv4 10.x.x.x preference 2
>>  description CUCM SIGNALING PEERS
>> !
>> voice class sip-options-keepalive 100
>>  description PSTN HEARTBEAT
>> !
>> voice class sip-options-keepalive 200
>>  description CCM HEARTBEAT
>> !
>> { .. other config .. }
>> 
>> dial-peer voice 100 voip
>>  description INGRESS/EGRESS WITH PSTN
>>  translation-profile outgoing PLUS1_STRIP
>>  huntstop
>>  destination-pattern A
>>  session protocol sipv2
>>  session server-group 100
>>  destination dpg 200
>>  incoming uri via PSTN
>>  voice-class codec 1  
>>  voice-class sip options-ping 60
>>  voice-class sip profiles 100
>>  voice-class sip options-keepalive profile 100
>>  voice-class sip bind control source-interface 
>>  voice-class sip bind media source-interface 
>>  dtmf-relay rtp-nte sip-notify
>>  no vad
>> !
>> dial-peer voice 200 voip
>>  description INGRESS/EGRESS WITH CUCM
>>  translation-profile outgoing PLUS1_STRIP
>>  huntstop
>>  

Re: [cisco-voip] SIP Fail over

[NateCCIE]

When you say level3 does not support options ping, do you mean they won't
ping you for failover, or they don't allow you to send it to them?  Only two
messages and the lack of any response will busy the endpoint, anything else
if good enough for CUBE.

 



 

From: cisco-voip  On Behalf Of Ryan Huff
Sent: Thursday, December 20, 2018 10:35 AM
To: Erik Anderson ; cisco-voip voyp list

Subject: Re: [cisco-voip] SIP Fail over

 

Couldn't you just use voice class sip options/keepalives to mark when the
ITSP is down, so CUCM marks the trunk out of service and fails to the next
route group member immediately (ideally, your secondary CUBE)? Seems like
thats a more natural way of doing it versus using IP SLAs...

 

Thanks,

 

- Ryan

  _  

From: cisco-voip mailto:cisco-voip-boun...@puck.nether.net> > on behalf of Erik Anderson
mailto:erik.anderson...@gmail.com> >
Sent: Thursday, December 20, 2018 12:03 PM
To: cisco-voip voyp list
Subject: [cisco-voip] SIP Fail over 

 

Morning Folks, 




We have implemented a new SIP solution with Level 3 and found that we have
outbound calling failover issues. When a CUBE loses its ability to talk to
its Level 3 Peer, but can still talk to CUCM outbound calls will still
connect to the CUBE, but fail connecting to Level 3. In turn CUCM still
thinks the call is connected since the CUCM SIP trunk remains up to the
CUBE.

 

Architecture Notes:

 

4 Locations with 1 CUBE Each

4 CUCM SIP Trunks with each connecting to one of the 4 CUBEs

4 CUCM Route Groups with Various CUBE/SIP Trunks assigned a Distribution
Algorithm of Top Down

Each CUBE has 2 SIP Peers

Each CUBE can only talk to its respective SIP peer via its local Level 3
Transport to reduce call control latency by not allowing it to use the DMVPN
backup network

Level 3 does not support SIP Options Ping

CUCM Trunks have SIP Options Ping enabled

 

Call Flows:

 

Working Flow:

 

Phone > SLRG > Route Group Member #1 > CUBE SIP TRUNK > CUBE
> Level 3 Transport > Level 3 SIP Peer #1/#2 > Call Completes

 

 

CUBE Failure:

 

Phone > SLRG >

 Route Group Member #1 > CUBE SIP TRUNK --X--> CUBE (CUCM Cant
Reach CUBE)

 

CUCM Routes Call to Next Route Group Member

 

  Route Group Member #2 > CUBE SIP TRUNK
> CUBE > Level 3 Transport > Level 3 SIP Peer #1/#2 > Call
Completes

 

Level 3 Transport Failure/SIP Server Failure:

 

Phone > SLRG >

 Route Group Member #1 > CUBE SIP TRUNK > CUBE --X--> Level
3 Transport (CUBE Cant Reach Level 3 SIP Server)

 

CUCM Thinks Call Connects since the CUBE accepts the call, Phone
gets dead air, never tries the next RG Member

 

 

My idea to fix this is to use an IPSLA to ping the pingable address on the
Level 3 SIP Servers. If both address are unreachable then shutdown the CUCM
Dial-Peers. This doesn't sounds like the best way of fixing it, but it
should work. 

 

If any has any other better ideas please let me know.

-- 

Erik Anderson

Telecom Manager

Some Random Corp.

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Re: [cisco-voip] Recovering UCOS Passwords - Round 281

[NateCCIE]

I am pretty sure PCD will grab that file.  I am not sure if its when you do
the cluster discovery or if it's when you start the migration and it pulls
the data.  Both of those actions install/run a COP file on the server that
somehow exports the data to PCD.  Maybe the COP file could be used OOB to
send to your SFTP server of choice?

 

-Nate

 

From: cisco-voip  On Behalf Of Pete
Brown
Sent: Wednesday, December 5, 2018 11:33 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Recovering UCOS Passwords - Round 281

 

I'm sure some of you noticed, but earlier this year Cisco started releasing
patches to kill off the last sanctioned method of getting to
platformConfig.xml.  When you run "utils create report platform" on recent
versions, it's no longer in the report.  Someone in Boxborough really knows
how to put the "cus(s)" in "customers"!

 

https://quickview.cloudapps.cisco.com/quickview/bug/CSCvh62145

 

I'm testing a new version of the UCOS Password Decrypter that acquires the
file for you.  To use this feature, you enable remote support on your UCOS
host then plug in the UCOS host IP, remote support user and remote support
passphrase.  The app decodes the passphrase, pulls the file via SSH and
displays the passwords.

 

Need a few volunteers to test before I update the tools page.  If you're
interested, let me know.  Would post a temp link here but I don't want yet
another dead link floating around.

 

-Pete

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Re: [cisco-voip] connected party information while calling webex

[NateCCIE]

There is a setting on the sip trunks to deliver dn or uri or both. I bet there 
is a difference between your two clusters on that setting. I may not win the 
bet, but that is where I’d check first. 

Sent from my iPhone

> On Nov 2, 2018, at 12:20 PM, Lelio Fulgenzi  wrote:
> 
>  
> Gotcha. All using FQDN as far as I can tell. Very odd. I don’t think I’ve 
> made changes to jabberconfig.xml either.
>  
> ---
> Lelio Fulgenzi, B.A. | Senior Analyst
> Computing and Communications Services | University of Guelph
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 
> 2W1
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>  
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>  
> 
>  
> From: Ryan Huff  
> Sent: Friday, November 2, 2018 2:18 PM
> To: Lelio Fulgenzi 
> Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net) 
> 
> Subject: Re: [cisco-voip] connected party information while calling webex
>  
> I meant UC, Siri assumed I meant UDS. How about the CM server references, all 
> FQDN as well?
> 
> Sent from my iPhone
> 
> On Nov 2, 2018, at 14:15, Lelio Fulgenzi  wrote:
> 
> What is this UDS thoust speak of?
>  
> But seriously, referenced where?
>  
> I checked my UC Services, all of those are FQDN. I checked my Service 
> Profile, and that just has a “Use UDS for Contact Resolution” option that 
> I’ve checked off in both systems.
>  
> What would UDS have to do with replacing @acme.webex.com with 
> @ though? Curious. I was leaning towards connected party 
> information setup.
>  
>  
>  
> ---
> Lelio Fulgenzi, B.A. | Senior Analyst
> Computing and Communications Services | University of Guelph
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 
> 2W1
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>  
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>  
> 
>  
> From: Ryan Huff  
> Sent: Friday, November 2, 2018 1:58 PM
> To: Lelio Fulgenzi 
> Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net) 
> 
> Subject: Re: [cisco-voip] connected party information while calling webex
>  
> UDS services referenced by FQDN or IP?
> 
> Sent from my iPhone
> 
> On Nov 2, 2018, at 13:55, Lelio Fulgenzi  wrote:
> 
>  
> I’ve migrated my development cluster configuration to my production cluster 
> configuration and have compared them as best as possible, but I seem to be 
> missing connected party information when calling webex from Jabber.
>  
> On the development cluster, when I dial coy...@acme.webex.com from Jabber and 
> am connected, the connected party information at the top of Jabber remains 
> coy...@acme.webex.com, however, when I dial the same from production, it 
> changes to coyote@
>  
> I’ve reviewed the SIP trunk and dependencies as much as I could and they all 
> seem the same. I did some comparison of enterprise parameters and ccm service 
> parameters and they too look the same.
>  
> I can’t imagine anything expresway or within webex site config would cause 
> this.
>  
> Thoughts? Pointers?
>  
> ---
> Lelio Fulgenzi, B.A. | Senior Analyst
> Computing and Communications Services | University of Guelph
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 
> 2W1
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>  
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>  
> 
>  
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Re: [cisco-voip] Basic corlist question

[NateCCIE]

Yes. If you have no cor applied to the outbound dial peer, then anything can 
use that dialpeer.

Sent from my iPhone

> On Oct 30, 2018, at 1:11 PM, Ed Leatherman  wrote:
> 
> At least - I think this is basic...
> 
> Can a call with an incoming corlist (based on the inbound dialpeer) match an 
> outbound dialpeer that has no corlist? Basically, following the lock/key 
> idea, the outbound dial-peer has no lock so anything can match it.
> 
> I'm trying to go through a vendor-provided IOS config for a SIP integration 
> between a Cisco router and our public safety's dispatch system, and it keeps 
> matching on a dialpeer we are not expecting, based on how we _think_ it is 
> supposed to work. Essentially the scenario above.
> 
> Thanks!
> 
> -- 
> Ed Leatherman
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Re: [cisco-voip] Migrate HCS to On-Prem

[NateCCIE]

Esxi on dest only, os admin on source, it installs a cop file on source to pull 
the data. 

Sent from my iPhone

> On Oct 23, 2018, at 5:06 PM, Dave Goodwin  wrote:
> 
> I don’t have an easy way to check at the moment, but I think/hope that it’s 
> not required on the source side of the migration, and it just uses OS 
> credentials to send a shutdown. I know on the destination side ESXi is 
> required, because it maps the ISO via NFS, powers up the VM, etc. 
> 
>> On Tue, Oct 23, 2018 at 6:23 PM Matthew Loraditch 
>>  wrote:
>> Good question, because there is no way I’m getting the esxi login.
>> 
>> Get Outlook for iOS
>>  
>> Matthew Loraditch​
>> Sr. Network Engineer
>> p: 443.541.1518
>> w: www.heliontechnologies.com |  e: 
>> mloradi...@heliontechnologies.com
>> 
>> 
>> 
>> 
>> From: Dave Goodwin 
>> Sent: Tuesday, October 23, 2018 6:21:49 PM
>> To: NateCCIE
>> Cc: Brian Meade; Matthew Loraditch; cisco-voip voyp list
>> 
>> Subject: Re: [cisco-voip] Migrate HCS to On-Prem
>>  
>> It’s been a while since I did a PCD migration, but does it also require that 
>> PCD log into ESXi on both the source and destination VMware servers? Or does 
>> it only need ESXi access on destination?
>> 
>>> On Tue, Oct 23, 2018 at 6:08 PM NateCCIE  wrote:
>>> If you can get osadmin credentials, consider PCD migration.  It will change 
>>> the IPs and upgrade all in one nice step.
>>> 
>>>  
>>> 
>>> From: cisco-voip  On Behalf Of Brian 
>>> Meade
>>> Sent: Tuesday, October 23, 2018 3:47 PM
>>> To: Matthew Loraditch 
>>> Cc: cisco-voip voyp list 
>>> 
>>> 
>>> Subject: Re: [cisco-voip] Migrate HCS to On-Prem
>>> 
>>>  
>>> 
>>> They'll do DRS for moving to HCS from on-prem.  They'll setup everything 
>>> with your hostnames/IPs to restore then they'll make all their changes.
>>> 
>>>  
>>> 
>>> Opposite should be technically possible but probably depends on the HCS 
>>> partner if they'll share enough information with you to be able to restore 
>>> and even get the DRS Backups from them at all.
>>> 
>>>  
>>> 
>>> Curious about moving from HCS to on-prem, was cost the main factor or 
>>> anything else?  I've got a bunch of customers trying to go to HCS so could 
>>> use some talking-points if customers are seeing issues.
>>> 
>>>  
>>> 
>>>  
>>> 
>>> On Tue, Oct 23, 2018 at 5:25 PM Matthew Loraditch 
>>>  wrote:
>>> 
>>> Interesting. The UCCX system is only 1 script and 10 agents so if that 
>>> caveat doesn’t apply for CUCM and CUC, I’d be ok with it
>>> 
>>>  
>>> 
>>> Get Outlook for iOS
>>> 
>>>  
>>> 
>>> Matthew Loraditch​
>>> 
>>> Sr. Network Engineer
>>> 
>>> p: 443.541.1518
>>> 
>>> w: www.heliontechnologies.com
>>> 
>>>  | 
>>> 
>>> e: mloradi...@heliontechnologies.com
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> From: Anthony Holloway 
>>> Sent: Tuesday, October 23, 2018 5:20:20 PM
>>> To: Matthew Loraditch
>>> Cc: Cisco VoIP Group
>>> Subject: Re: [cisco-voip] Migrate HCS to On-Prem
>>> 
>>>  
>>> 
>>> Maybe some CDW people watching this mailing list will chime in, because I 
>>> happen to know that they do convert their own HCS customers to on-prem from 
>>> time to time.
>>> 
>>>  
>>> 
>>> From what I understand, you can restore it, but the configuration is all 
>>> jacked up, and you'll have to rename everything and/or reconfigure 
>>> everything anyway.  It might only make sense from a historical reporting 
>>> perspective.
>>> 
>>>  
>>> 
>>> Like, have you ever seen what ASDM does to a FW config?
>>> 
>>>  
>>> 
>>> Or what the CCP does to CME?
>>> 
>>>  
>>> 
>>> Or what CTI Logic was doing with "visual UCCX scripts?"
>>> 
>>>  
>>> 
>>> Example:
>>> 
>>>  
>>> 
>>> 
>>> 
>>>  
>>> 
>>> It's not that it's bad, it's just not a good long term solution to keep the 
>>> conventions the same.
>>> 
>>

Re: [cisco-voip] Migrate HCS to On-Prem

[NateCCIE]

If you can get osadmin credentials, consider PCD migration.  It will change the 
IPs and upgrade all in one nice step.

 

From: cisco-voip  On Behalf Of Brian Meade
Sent: Tuesday, October 23, 2018 3:47 PM
To: Matthew Loraditch 
Cc: cisco-voip voyp list 
Subject: Re: [cisco-voip] Migrate HCS to On-Prem

 

They'll do DRS for moving to HCS from on-prem.  They'll setup everything with 
your hostnames/IPs to restore then they'll make all their changes.

 

Opposite should be technically possible but probably depends on the HCS partner 
if they'll share enough information with you to be able to restore and even get 
the DRS Backups from them at all.

 

Curious about moving from HCS to on-prem, was cost the main factor or anything 
else?  I've got a bunch of customers trying to go to HCS so could use some 
talking-points if customers are seeing issues.

 

 

On Tue, Oct 23, 2018 at 5:25 PM Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com> > 
wrote:

Interesting. The UCCX system is only 1 script and 10 agents so if that caveat 
doesn’t apply for CUCM and CUC, I’d be ok with it

 

Get Outlook for iOS  



 


Matthew Loraditch​



Sr. Network Engineer




p:   443.541.1518



w:   www.heliontechnologies.com

 | 

e:   mloradi...@heliontechnologies.com











  


  


  

  _  

From: Anthony Holloway mailto:avholloway%2bcisco-v...@gmail.com> >
Sent: Tuesday, October 23, 2018 5:20:20 PM
To: Matthew Loraditch
Cc: Cisco VoIP Group
Subject: Re: [cisco-voip] Migrate HCS to On-Prem 

 

Maybe some CDW people watching this mailing list will chime in, because I 
happen to know that they do convert their own HCS customers to on-prem from 
time to time. 

 

>From what I understand, you can restore it, but the configuration is all 
>jacked up, and you'll have to rename everything and/or reconfigure everything 
>anyway.  It might only make sense from a historical reporting perspective.

 

Like, have you ever seen what ASDM does to a FW config?

 

Or what the CCP does to CME?

 

Or what CTI Logic was doing with "visual UCCX scripts?"

 

Example:

 



 

It's not that it's bad, it's just not a good long term solution to keep the 
conventions the same.

 

PS I just noticed the menu prompt says "pPressOneForHookers"  LOL, I'm not sure 
what that's all about, but this image is from Tanner's tweet 
 .

 

On Tue, Oct 23, 2018 at 3:19 PM Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com> > 
wrote:

So we are taking a Customer from a 3rd party’s HCS back to On-Prem.

 

The way I understand HCS is it’s a bunch of management software on top of 
normal VMs for CUCM/CUC/UCCX, etc.

 

Can you restore a DRF of an HCS CUCM to the same version on-prem? Same for the 
other apps?



 


Matthew Loraditch​



Sr. Network Engineer




p:   443.541.1518



w:   www.heliontechnologies.com

 | 

e:   mloradi...@heliontechnologies.com









  


  


  

  
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Re: [cisco-voip] How to handle expired Phone-VPN-trust, phone-SAST-trust, other certificates

[NateCCIE]

The expired certs will throw alarms even if they have been superseded by
newer certs.

 

So during a maintenance window, renew anything that is expired, and just
delete all the old ones.  The newer versions of cucm make this easier by
being able to sort by expiration date.

 

-Nate

 

From: cisco-voip  On Behalf Of ROZA,
Ariel
Sent: Monday, October 22, 2018 11:52 AM
To: cisco-voip (cisco-voip@puck.nether.net) 
Subject: [cisco-voip] How to handle expired Phone-VPN-trust,
phone-SAST-trust, other certificates

 

Hi, guys!

 

I have a customer that is receiving alarms over some expired certificates,
and I would like to know which is the best way to handle them.

The certs are loaded in SERVER1 and all named SERVER2.der, except the CAPF
ones.

.der in phone-vpn-trust. 

 .der in phone-trust

.der in phone-SAST-trust

.der in phone-CTL-trust

And several CAPF-xx.der in Callmanager-trust

 

So far I have dealt with renewing Callmanager, TFTP and TVS cert, but I
always kept clear from those other certs

Shoud I delete them, shoud I keep them, even as they are expired and
throwing alarms?

 

 

Regards.

 

 

Ariel Roza 
Collaboration Support Engineer 

t: +54 11 5282-0458 

c: +54 9 11 5017-4417 webex:

http://logicalis-la.webex.com/join/ariel.roza

Av. Belgrano 955 – Piso 20 – CABA – Argentina – C1092AAJ

  www.la.logicalis.com

_
Business and technology working as one 



 



 

 

  

 

Logicalis Argentina S.A. solo puede ser obligado por sus representantes
legales conforme los límites establecidos en el acto constitutivo y la
legislación en vigor. 

El contenido del presente correo electrónico e inclusive sus anexos
contienen información confidencial. 

El mismo no puede ser divulgado y/o utilizado por cualquiera otro distinto
al destinatario, ni puede ser copiado de cualquier forma.

 

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Re: [cisco-voip] ASR 1k CUBE recommended version

[NateCCIE]

One of the best CUBE resources is Hussain Ali’s box site.  He is one of the 
CUBE TMEs, or maybe the lead TME.

 

http://cisco.box.com/cube

 

 

From: cisco-voip  On Behalf Of Jason Aarons 
(Americas)
Sent: Monday, October 22, 2018 8:39 AM
To: cisco-voip (cisco-voip@puck.nether.net) 
Subject: [cisco-voip] ASR 1k CUBE recommended version

 

 

I’m not finding any guidance on cisco.com for a recommended version of CUBE to 
use on an ASR or 44xx.

 

Is there one?

 

-jason

 



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Re: [cisco-voip] VOIP on AS5400

[NateCCIE]

To do Voice, you will need DSPs, 1 dsp is 64 g711 calls.  That will be the
limiting factor over PSTN connectivity.

 

https://www.cisco.com/c/en/us/products/collateral/unified-communications/as5
400xm-universal-gateway/product_data_sheet0900aecd80458049.html

 

 

From: cisco-voip  On Behalf Of Joseph
Mays
Sent: Tuesday, October 9, 2018 2:04 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] VOIP on AS5400

 

I am recently put in the position of working with some old AS5400s. I put
the following question to the cisco nas list, but I thought it might be
appropriate for here, too

 

The only other question is about the number of voip calls the box can
support. My understanding (possibly wrong) is that there is an upper limit
on the number of voip calls an AS5400 can sustain, depending on the memory,
processor, and codec being used. Is this correct, and if so how do I
calculate it?

 

The codec being used is G.711, which does virtually no compression, so I
would imagine the processor usage is very low as compared to other codecs.

 

ArmoryPl-AS5400#show ver

Cisco Internetwork Operating System Software

IOS (tm) 5400 Software (C5400-JS-M), Version 12.3(3i), RELEASE SOFTWARE
(fc1)

Copyright (c) 1986-2005 by cisco Systems, Inc.

Compiled Fri 12-Aug-05 22:07 by ssearch

Image text-base: 0x6000895C, data-base: 0x6190

 

ROM: System Bootstrap, Version 12.2(1r)1, RELEASE SOFTWARE (fc1)

BOOTLDR: 5400 Software (C5400-BOOT-M), Version 12.1(1)XD1, EARLY DEPLOYMENT
RELEASE SOFTWARE (fc2)

 

ArmoryPl-AS5400 uptime is 1 week, 1 day, 3 hours, 17 minutes

System returned to ROM by bus error at PC 0x6158, address 0x59CCDEC at
11:13:47 EDT Mon Oct 1 2018

System restarted at 11:14:51 EDT Mon Oct 1 2018

System image file is "flash:c5400-js-mz.123-3i.bin"

 

cisco AS5400 (R7K) processor (revision T) with 524288K/131072K bytes of
memory.

Processor board ID JAE09054U8M

R7000 CPU at 250Mhz, Implementation 39, Rev 1.0, 256KB L2, 2048KB L3 Cache

Last reset from warm-reset

Bridging software.

X.25 software, Version 3.0.0.

SuperLAT software (copyright 1990 by Meridian Technology Corp).

TN3270 Emulation software.

Primary Rate ISDN software, Version 1.1.

Manufacture Cookie Info:

EEPROM Type 0x0001, EEPROM Version 0x01, Board ID 0x31,

Board Hardware Version 3.34, Item Number 800-5171-02,

Board Revision C0, Serial Number JAE09054U8M,

PLD/ISP Version 2.2,  Manufacture Date 26-Jan-2005.

Processor 0x14, MAC Address 0x012801C6694

Backplane HW Revision 1.0, Flash Type 5V

2 FastEthernet/IEEE 802.3 interface(s)

444 Serial network interface(s)

432 terminal line(s)

64 Channelized T1/PRI port(s)

2 Channelized T3 port(s)

512K bytes of non-volatile configuration memory.

65536K bytes of processor board System flash (Read/Write)

16384K bytes of processor board Boot flash (Read/Write)

 

 

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Re: [cisco-voip] CUCM 11.5.1SU5 from Pub go to Serviceablity and subscriber error?

[NateCCIE]

Usually IPSEC certs for me when this happens.

 

From: cisco-voip  On Behalf Of Jason Aarons 
(Americas)
Sent: Friday, October 5, 2018 12:03 PM
To: cisco-voip (cisco-voip@puck.nether.net) 
Subject: [cisco-voip] CUCM 11.5.1SU5 from Pub go to Serviceablity and 
subscriber error?

 

 

Any pointers on when Serviceability can't show services status on another sub 
etc? Is that tomcat-trust or ipsec-trust etc?

Seen it on a couple clusters running 11.5.1SU5. Perhaps a known bug.

Db replication status is happy.

Get Outlook for Android  

 



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Re: [cisco-voip] Customizing called party name/number display for route patterns

[NateCCIE]

Where is the call going?  Cucm shouldn’t be updating what the user sees with 
that config, but the thing your sending the call to is probably updating it and 
then cucm shows the update. 

Sent from my iPhone

> On Sep 24, 2018, at 12:49 PM, Lelio Fulgenzi  wrote:
> 
>  
> Hmmm, went through the effort of building a route group and route list and 
> adding the prefix digits I need on the route group member configuration of 
> the route list.
>  
> But still, the prefix digits appear on the display of the dialing device. ☹
>  
> How do I get rid of displaying those prefix digits? I was pretty sure this 
> was the way.
>  
>  
> ---
> Lelio Fulgenzi, B.A. | Senior Analyst
> Computing and Communications Services | University of Guelph
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 
> 2W1
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>  
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>  
> 
>  
> From: cisco-voip  On Behalf Of Lelio 
> Fulgenzi
> Sent: Friday, September 21, 2018 11:20 AM
> To: Florian Kroessbacher ; Brian Meade 
> 
> Cc: cisco-voip voyp list 
> Subject: Re: [cisco-voip] Customizing called party name/number display for 
> route patterns
>  
>  
> Hmmm, looks like another programming option.
>  
> I was hoping for a built-in option. ☹
>  
> ---
> Lelio Fulgenzi, B.A. | Senior Analyst
> Computing and Communications Services | University of Guelph
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 
> 2W1
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>  
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>  
> 
>  
> From: Florian Kroessbacher  
> Sent: Friday, September 21, 2018 10:52 AM
> To: Brian Meade ; Lelio Fulgenzi 
> Cc: cisco-voip voyp list 
> Subject: Re: [cisco-voip] Customizing called party name/number display for 
> route patterns
>  
> Hy out there
> 
> what about CURRI if u were on >10
> Am 20. Sep. 2018, 23:43 +0200 schrieb Lelio Fulgenzi :
> 
>  
> I half want to try the old trick of a CTI route point and use forwarding, but 
> not sure I want to intercept that complexity. 
> 
> -sent from mobile device-
>  
> 
> Lelio Fulgenzi, B.A. | Senior Analyst
> Computing and Communications Services | University of Guelph
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 
> 2W1
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>  
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>  
> 
> 
> On Sep 20, 2018, at 5:32 PM, Brian Meade  wrote:
> 
> For name modification, CUCM doesn't have much.  Probably would need a LUA 
> script to update the name in the SIP messaging.
>  
> On Wed, Sep 19, 2018 at 9:33 PM Lelio Fulgenzi  wrote:
>  
> It’s been a while, and I know I can use route lists and route groups to 
> customize called party number display but I can’t recall what my options are, 
> if any, to customize the called party name display. 
>  
> For example, if I dial 4 and want only 4 to continue to display, I 
> can do the modifications on the route list/group, ie prefix 999, and build 
> appropriate rules on expressway. 
>  
> But what if I’d like to display “WebEx Pilot” on the phone when they call a 
> particular route pattern?
>  
> I can’t seem to find any option for that. 
> 
> -sent from mobile device-
>  
> 
> Lelio Fulgenzi, B.A. | Senior Analyst
> Computing and Communications Services | University of Guelph
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 
> 2W1
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>  
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>  
> 
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Re: [cisco-voip] CUBE setup to Centurylink SIP Trunk

[NateCCIE]

Yeah the customer gets to choose. The nice part about IP TF/LD is you don’t pay 
for sessions or trunks, and it’s a BYOB solution, but there is no local calling 
or DID service.

I am guessing you changed the IPs of the SBC, I believe 6.0.0.0/8 is DOD?  I 
would guess that if it’s broadsoft and not requiring registration it’s a Level3 
platform.

I have heard that internally they are moving to all Level3 backend systems for 
ordering, etc.
Sent from my iPhone

> On Sep 12, 2018, at 8:44 PM, Anthony Holloway 
>  wrote:
> 
> I can confirm that I am talking to broadsoft, since I get a an error in CUBE 
> about the SDP attribute bsoft not being recognized.
> 
> Nate, is there a choice for the customer signing up for new service, or is it 
> dictated by something out of their control?
> 
> 
> 
>> On Wed, Sep 12, 2018 at 8:54 PM NateCCIE  wrote:
>> I don’t see any reason to include the media address in the trusted list.  
>> That would be like including all IP phones in the trusted list.
>> 
>>  
>> 
>> A lot of the time I only route specific IPs to the outside next hop, as a 
>> security measure.  If they didn’t indicate where the media was coming from, 
>> it would be easy to miss that and get one way audio.
>> 
>>  
>> 
>> And centurylink has many SIP plaforms, the registration one with 
>> multi-tennant configs for dual registration is the Broadsoft platform, the 
>> sonos platform isn’t adding new customers, and then there is the IP 
>> TollFree/LD, that one is still current and doesn’t require registration.  
>> There also are at least two Level3 platforms that are now “centurylink”
>> 
>>  
>> 
>> Thanks,
>> 
>> -Nate
>> 
>>  
>> 
>> From: cisco-voip  On Behalf Of Ryan Huff
>> Sent: Wednesday, September 12, 2018 7:31 PM
>> To: Jason Aarons (Americas) ; cisco-voip 
>> (cisco-voip@puck.nether.net) 
>> Subject: Re: [cisco-voip] CUBE setup to Centurylink SIP Trunk
>> 
>>  
>> 
>> Target the signaling address in your dial peers, the media address will be 
>> advertised in the SDP. Make sure to include both in your IP Trusted List ACL 
>> (under the voice service voip configuration) as well as any CUCM signaling 
>> nodes that are not directly targeted by a dial-peer (but I typically add all 
>> the nodes in regardless, just as a measure of safety).
>> 
>>  
>> 
>> Thanks,
>> 
>>  
>> 
>> Ryan
>> 
>> From: cisco-voip  on behalf of Jason 
>> Aarons (Americas) 
>> Sent: Wednesday, September 12, 2018 8:37 PM
>> To: cisco-voip (cisco-voip@puck.nether.net)
>> Subject: [cisco-voip] CUBE setup to Centurylink SIP Trunk
>> 
>>  
>> 
>>  
>> 
>> I have a new CenturyLink SIP Service.  CenturyLink said it is new and 
>> doesn't match the Cisco guides.  (No more of the funky registrar and fixup 
>> headers via SIP profiles!)
>> 
>>  
>> 
>> In short in CUBE they want me to send calls to them per these settings;
>> 
>> SIP Signaling IP 6.6.156.245:5060
>> 
>> RTP IP 6.6.156.244
>> 
>> I'm just drawing a blank on how to setup CUBE to send SIP signaling requests 
>> to CenturyLink with different Signaling and RTP destination addresses.  
>> Don't I just send session target ipv4:X.X.156.245:5060 and the SDP takes 
>> care of the RTP negotiation part?  Do I really care in my CUBE what their 
>> RTP address is?
>> 
>>  
>> 
>>  
>> 
>> -jason
>> 
>>  
>> 
>> 
>> 
>> This email and all contents are subject to the following disclaimer:
>> "http://www.dimensiondata.com/emaildisclaimer;
>> 
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Re: [cisco-voip] CUBE setup to Centurylink SIP Trunk

[NateCCIE]

I don't see any reason to include the media address in the trusted list.
That would be like including all IP phones in the trusted list.

 

A lot of the time I only route specific IPs to the outside next hop, as a
security measure.  If they didn't indicate where the media was coming from,
it would be easy to miss that and get one way audio.

 

And centurylink has many SIP plaforms, the registration one with
multi-tennant configs for dual registration is the Broadsoft platform, the
sonos platform isn't adding new customers, and then there is the IP
TollFree/LD, that one is still current and doesn't require registration.
There also are at least two Level3 platforms that are now "centurylink"

 

Thanks,

-Nate

 

From: cisco-voip  On Behalf Of Ryan Huff
Sent: Wednesday, September 12, 2018 7:31 PM
To: Jason Aarons (Americas) ; cisco-voip
(cisco-voip@puck.nether.net) 
Subject: Re: [cisco-voip] CUBE setup to Centurylink SIP Trunk

 

Target the signaling address in your dial peers, the media address will be
advertised in the SDP. Make sure to include both in your IP Trusted List ACL
(under the voice service voip configuration) as well as any CUCM signaling
nodes that are not directly targeted by a dial-peer (but I typically add all
the nodes in regardless, just as a measure of safety).

 

Thanks,

 

Ryan

  _  

From: cisco-voip mailto:cisco-voip-boun...@puck.nether.net> > on behalf of Jason Aarons
(Americas) mailto:jason.aar...@dimensiondata.com> >
Sent: Wednesday, September 12, 2018 8:37 PM
To: cisco-voip (cisco-voip@puck.nether.net
 )
Subject: [cisco-voip] CUBE setup to Centurylink SIP Trunk 

 

 

I have a new CenturyLink SIP Service.  CenturyLink said it is new and
doesn't match the Cisco guides.  (No more of the funky registrar and fixup
headers via SIP profiles!)

 

In short in CUBE they want me to send calls to them per these settings;

SIP Signaling IP 6.6.156.245:5060

RTP IP 6.6.156.244

I'm just drawing a blank on how to setup CUBE to send SIP signaling requests
to CenturyLink with different Signaling and RTP destination addresses.
Don't I just send session target ipv4:X.X.156.245:5060 and the SDP takes
care of the RTP negotiation part?  Do I really care in my CUBE what their
RTP address is?

 

 

-jason

 



This email and all contents are subject to the following disclaimer:
 
 "http://www.dimensiondata.com/emaildisclaimer; 

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Re: [cisco-voip] PCD 12.1.1 for CUCM 6.5 PCD Migration

[NateCCIE]

I don’t think PCD touches licensing.

 

-Nate

From: cisco-voip  On Behalf Of Jason Aarons 
(Americas)
Sent: Monday, August 20, 2018 3:45 PM
To: cisco-voip (cisco-voip@puck.nether.net) 
Subject: [cisco-voip] PCD 12.1.1 for CUCM 6.5 PCD Migration

 

 

I have a customer with CUCM  6.1.5.11900-1 that I plan to upgrade to CUCM 
11.5.1SU5 (Matrix says it is supported, all 7925 phones/loads are good).  I 
plan to use PCD 12.1.1 with PCD Migration. 

 

Does PCD 12.1.1 need to talk to a PLM/Smart Licensing?  The PLM servers are 
still 11.5

 

-jason

 



This email and all contents are subject to the following disclaimer:
  
"http://www.dimensiondata.com/emaildisclaimer; 

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Re: [cisco-voip] CUBE ignoring SDP responses from ITSP

[NateCCIE]

NAT as in ip nat outside and ip nat inside, etc, would be used for other 
traffic that is flowing through the router.  Address hiding for a SIP 
connection is different.

 

From: Bill Talley  
Sent: Saturday, August 11, 2018 1:08 PM
To: NateCCIE 
Cc: Benjamin Turner ; cisco-voip voyp list 

Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP

 

Is that not essentially what address hiding and binding the telco facing 
dial-peer to the outside interface accomplishes?

Sent from an iOS device with very tiny touchscreen input keys.  Please excude 
my typtos.


On Aug 11, 2018, at 1:24 PM, NateCCIE mailto:natec...@gmail.com> > wrote:

No, if you want to do NAT, it has to be on a different interface than what CUBE 
is using for SIP.

 

From: Benjamin Turner mailto:benmtur...@hotmail.com> > 
Sent: Saturday, August 11, 2018 12:03 PM
To: NateCCIE mailto:natec...@gmail.com> >
Cc: Ryan Huff mailto:ryanh...@outlook.com> >; cisco-voip 
voyp list mailto:cisco-voip@puck.nether.net> >
Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP

 

So should I implement a voice class sip profile to modify the sip headers?





Get Outlook for Android <https://aka.ms/ghei36> 

 

  _  

From: NateCCIE mailto:natec...@gmail.com> >
Sent: Saturday, August 11, 2018 1:46:31 PM
To: Benjamin Turner
Cc: Ryan Huff; cisco-voip voyp list
Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP 

 

You can’t have ip NAT outside on a CUBE. It used to work before the 4K but it 
was never supported according to TAC and the BU. 

 

Inbound traffic goes to the NAT process and the SIP stack never sees it

Sent from my iPhone


On Aug 10, 2018, at 9:38 PM, Benjamin Turner mailto:benmtur...@hotmail.com> > wrote:

My brain is FRIED 

Get Outlook for Android <https://aka.ms/ghei36> 

 

  _  

From: Benjamin Turner mailto:benmtur...@hotmail.com> >
Sent: Friday, August 10, 2018 11:35:15 PM
To: Ryan Huff
Cc: Bill Talley; cisco-voip voyp list
Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP 

 

I did. They see my outside IP (multiple) invites and they are sending 1xx 
responses. And, I see them too on my cube with a monitor capture. I just do not 
see them with a debug ccsip messages

Get Outlook for Android <https://aka.ms/ghei36> 

 

  _  

From: Ryan Huff mailto:ryanh...@outlook.com> >
Sent: Friday, August 10, 2018 11:32:08 PM
To: Benjamin Turner
Cc: Bill Talley; cisco-voip voyp list
Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP 

 

and you’re sure you’re signaling the correct peer address for signaling that 
the carrier gave you when they provisioned the trunk?

You may need to do a live TSHOOT with the carrier so they can tell you if they 
are (and what) they see in their SBC.


On Aug 10, 2018, at 23:29, Benjamin Turner mailto:benmtur...@hotmail.com> > wrote:

Yeah tried both ways. Inside and outside IP on the cucm trunk config

Get Outlook for Android <https://aka.ms/ghei36> 

 

  _  

From: Bill Talley mailto:btal...@gmail.com> >
Sent: Friday, August 10, 2018 11:28:02 PM
To: Benjamin Turner
Cc: Ryan Huff; cisco-voip voyp list
Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP 

 

Just confirming when you changed the bindings in IOS, did you change the SIP 
trunk destination IP address in CUCM to point to the inside address of CUBE and 
reset the trunk?

Sent from an iOS device with very tiny touchscreen input keys.  Please excude 
my typtos.


On Aug 10, 2018, at 9:25 PM, Benjamin Turner mailto:benmtur...@hotmail.com> > wrote:

Acl is not blocking and I tried to set binding on outbound dial-peer to 
outbound interface and inbound dial-peer to inbound interface 

Get Outlook for Android <https://aka.ms/ghei36> 

 

  _  

From: Ryan Huff mailto:ryanh...@outlook.com> >
Sent: Friday, August 10, 2018 10:23:00 PM
To: Benjamin Turner
Cc: Loren Hillukka; cisco-voip voyp list
Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP 

 

I would suggest a couple of things to start with; 

 

1.) Ditch the global bindings on 0/0/0 and bind your individual dial-peers to 
the appropriate interfaces

 

2.) Verify your ACL 101 isn’t interfering (you didn’t include your access lists 
so I can’t tell)

 

Sent from my iPhone


On Aug 10, 2018, at 22:15, Benjamin Turner mailto:benmtur...@hotmail.com> > wrote:

Very basic:

 

 

 

version 15.5

 

voice service voip

ip address trusted list

  ipv4 67.231.8.75

  ipv4 67.231.12.12

  ipv4 192.168.1.20

  ipv4 162.245.36.90

address-hiding

mode border-element license capacity 100

allow-connections sip to sip

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

fax protocol pass-through g711ulaw

sip

  bind control source-interface GigabitEthernet0/0/0

  bind media source-interface GigabitEthernet0/0/0

  registrar server

  no update-callerid

  ear

Re: [cisco-voip] CUBE ignoring SDP responses from ITSP

[NateCCIE]

No, if you want to do NAT, it has to be on a different interface than what
CUBE is using for SIP.

 

From: Benjamin Turner  
Sent: Saturday, August 11, 2018 12:03 PM
To: NateCCIE 
Cc: Ryan Huff ; cisco-voip voyp list

Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP

 

So should I implement a voice class sip profile to modify the sip headers?




Get Outlook for Android <https://aka.ms/ghei36> 

 

  _  

From: NateCCIE mailto:natec...@gmail.com> >
Sent: Saturday, August 11, 2018 1:46:31 PM
To: Benjamin Turner
Cc: Ryan Huff; cisco-voip voyp list
Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP 

 

You can't have ip NAT outside on a CUBE. It used to work before the 4K but
it was never supported according to TAC and the BU. 

 

Inbound traffic goes to the NAT process and the SIP stack never sees it

Sent from my iPhone


On Aug 10, 2018, at 9:38 PM, Benjamin Turner mailto:benmtur...@hotmail.com> > wrote:

My brain is FRIED 

Get Outlook for Android <https://aka.ms/ghei36> 

 

  _  

From: Benjamin Turner mailto:benmtur...@hotmail.com> >
Sent: Friday, August 10, 2018 11:35:15 PM
To: Ryan Huff
Cc: Bill Talley; cisco-voip voyp list
Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP 

 

I did. They see my outside IP (multiple) invites and they are sending 1xx
responses. And, I see them too on my cube with a monitor capture. I just do
not see them with a debug ccsip messages

Get Outlook for Android <https://aka.ms/ghei36> 

 

  _  

From: Ryan Huff mailto:ryanh...@outlook.com> >
Sent: Friday, August 10, 2018 11:32:08 PM
To: Benjamin Turner
Cc: Bill Talley; cisco-voip voyp list
Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP 

 

and you're sure you're signaling the correct peer address for signaling that
the carrier gave you when they provisioned the trunk?

You may need to do a live TSHOOT with the carrier so they can tell you if
they are (and what) they see in their SBC.


On Aug 10, 2018, at 23:29, Benjamin Turner mailto:benmtur...@hotmail.com> > wrote:

Yeah tried both ways. Inside and outside IP on the cucm trunk config

Get Outlook for Android <https://aka.ms/ghei36> 

 

  _  

From: Bill Talley mailto:btal...@gmail.com> >
Sent: Friday, August 10, 2018 11:28:02 PM
To: Benjamin Turner
Cc: Ryan Huff; cisco-voip voyp list
Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP 

 

Just confirming when you changed the bindings in IOS, did you change the SIP
trunk destination IP address in CUCM to point to the inside address of CUBE
and reset the trunk?

Sent from an iOS device with very tiny touchscreen input keys.  Please
excude my typtos.


On Aug 10, 2018, at 9:25 PM, Benjamin Turner mailto:benmtur...@hotmail.com> > wrote:

Acl is not blocking and I tried to set binding on outbound dial-peer to
outbound interface and inbound dial-peer to inbound interface 

Get Outlook for Android <https://aka.ms/ghei36> 

 

  _  

From: Ryan Huff mailto:ryanh...@outlook.com> >
Sent: Friday, August 10, 2018 10:23:00 PM
To: Benjamin Turner
Cc: Loren Hillukka; cisco-voip voyp list
Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP 

 

I would suggest a couple of things to start with; 

 

1.) Ditch the global bindings on 0/0/0 and bind your individual dial-peers
to the appropriate interfaces

 

2.) Verify your ACL 101 isn't interfering (you didn't include your access
lists so I can't tell)

 

Sent from my iPhone


On Aug 10, 2018, at 22:15, Benjamin Turner mailto:benmtur...@hotmail.com> > wrote:

Very basic:

 

 

 

version 15.5

 

voice service voip

ip address trusted list

  ipv4 67.231.8.75

  ipv4 67.231.12.12

  ipv4 192.168.1.20

  ipv4 162.245.36.90

address-hiding

mode border-element license capacity 100

allow-connections sip to sip

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

fax protocol pass-through g711ulaw

sip

  bind control source-interface GigabitEthernet0/0/0

  bind media source-interface GigabitEthernet0/0/0

  registrar server

  no update-callerid

  early-offer forced

  midcall-signaling passthru

  pass-thru content sdp

!

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g729r8

!

!

voice class sip-profiles 30

request ANY sdp-header Audio-Attribute modify "a=ptime:20" "a=ptime:30" 

!

!

!

!

!

voice translation-rule 2

rule 1 /\+1\([2-9].\)/ /\1/

!

voice translation-rule 11

rule 1 /\([2-9]..[2-9]..$\)/ /+1\1/

rule 2 /\(.*\)/ /+\1/

!

voice translation-rule 22

rule 1 /9\(1[2-9]..[2-9]..\)$/ /\1/

rule 2 /9\(911\)$/ /\1/

rule 3 /9\([2-8]11\)$/ /\1/

!

!

voice translation-profile 10DigitTo+1

translate calling 11

translate called 11

!

voice translation-profile LOCALIZE

translate calling 2

!

voice translation-profile OutgoingToBandwidthSIP

translate calling 2

translate called

Re: [cisco-voip] CUBE ignoring SDP responses from ITSP

[NateCCIE]

You can’t have ip NAT outside on a CUBE. It used to work before the 4K but it 
was never supported according to TAC and the BU.

Inbound traffic goes to the NAT process and the SIP stack never sees it

Sent from my iPhone

> On Aug 10, 2018, at 9:38 PM, Benjamin Turner  wrote:
> 
> My brain is FRIED 
> 
> Get Outlook for Android
> 
> From: Benjamin Turner 
> Sent: Friday, August 10, 2018 11:35:15 PM
> To: Ryan Huff
> Cc: Bill Talley; cisco-voip voyp list
> Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP
>  
> I did. They see my outside IP (multiple) invites and they are sending 1xx 
> responses. And, I see them too on my cube with a monitor capture. I just do 
> not see them with a debug ccsip messages
> 
> Get Outlook for Android
> 
> From: Ryan Huff 
> Sent: Friday, August 10, 2018 11:32:08 PM
> To: Benjamin Turner
> Cc: Bill Talley; cisco-voip voyp list
> Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP
>  
> and you’re sure you’re signaling the correct peer address for signaling that 
> the carrier gave you when they provisioned the trunk?
> 
> You may need to do a live TSHOOT with the carrier so they can tell you if 
> they are (and what) they see in their SBC.
> 
> On Aug 10, 2018, at 23:29, Benjamin Turner  wrote:
> 
>> Yeah tried both ways. Inside and outside IP on the cucm trunk config
>> 
>> Get Outlook for Android
>> 
>> From: Bill Talley 
>> Sent: Friday, August 10, 2018 11:28:02 PM
>> To: Benjamin Turner
>> Cc: Ryan Huff; cisco-voip voyp list
>> Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP
>>  
>> Just confirming when you changed the bindings in IOS, did you change the SIP 
>> trunk destination IP address in CUCM to point to the inside address of CUBE 
>> and reset the trunk?
>> 
>> Sent from an iOS device with very tiny touchscreen input keys.  Please 
>> excude my typtos.
>> 
>> On Aug 10, 2018, at 9:25 PM, Benjamin Turner  wrote:
>> 
>>> Acl is not blocking and I tried to set binding on outbound dial-peer to 
>>> outbound interface and inbound dial-peer to inbound interface 
>>> 
>>> Get Outlook for Android
>>> 
>>> From: Ryan Huff 
>>> Sent: Friday, August 10, 2018 10:23:00 PM
>>> To: Benjamin Turner
>>> Cc: Loren Hillukka; cisco-voip voyp list
>>> Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP
>>>  
>>> I would suggest a couple of things to start with;
>>> 
>>> 1.) Ditch the global bindings on 0/0/0 and bind your individual dial-peers 
>>> to the appropriate interfaces
>>> 
>>> 2.) Verify your ACL 101 isn’t interfering (you didn’t include your access 
>>> lists so I can’t tell)
>>> 
>>> Sent from my iPhone
>>> 
>>> On Aug 10, 2018, at 22:15, Benjamin Turner  wrote:
>>> 
 Very basic:
  
  
  
 version 15.5
  
 voice service voip
 ip address trusted list
   ipv4 67.231.8.75
   ipv4 67.231.12.12
   ipv4 192.168.1.20
   ipv4 162.245.36.90
 address-hiding
 mode border-element license capacity 100
 allow-connections sip to sip
 no supplementary-service sip moved-temporarily
 no supplementary-service sip refer
 fax protocol pass-through g711ulaw
 sip
   bind control source-interface GigabitEthernet0/0/0
   bind media source-interface GigabitEthernet0/0/0
   registrar server
   no update-callerid
   early-offer forced
   midcall-signaling passthru
   pass-thru content sdp
 !
 voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8
 !
 !
 voice class sip-profiles 30
 request ANY sdp-header Audio-Attribute modify "a=ptime:20" "a=ptime:30"
 !
 !
 !
 !
 !
 voice translation-rule 2
 rule 1 /\+1\([2-9].\)/ /\1/
 !
 voice translation-rule 11
 rule 1 /\([2-9]..[2-9]..$\)/ /+1\1/
 rule 2 /\(.*\)/ /+\1/
 !
 voice translation-rule 22
 rule 1 /9\(1[2-9]..[2-9]..\)$/ /\1/
 rule 2 /9\(911\)$/ /\1/
 rule 3 /9\([2-8]11\)$/ /\1/
 !
 !
 voice translation-profile 10DigitTo+1
 translate calling 11
 translate called 11
 !
 voice translation-profile LOCALIZE
 translate calling 2
 !
 voice translation-profile OutgoingToBandwidthSIP
 translate calling 2
 translate called 11
 !
 !
 !
 !
 voice-card 0/4
 dsp services dspfarm
 no watchdog
 !
 !
 interface GigabitEthernet0/0/0
 description WAN SIP TRUNK TO BANDWIDTH
 ip address 162.245.36.90 255.255.255.248
 ip nat outside
 ip access-group 101 in
 ip access-group 101 out
 negotiation auto
 no cdp enable
 !
 interface GigabitEthernet0/0/1
 description LAN
 ip address 192.168.1.1 255.255.255.0
 ip nat inside
 negotiation auto
 !
 interface GigabitEthernet0/0/2
 no ip address
 negotiation auto
 !
 interface Service-Engine0/4/0
 !
 interface GigabitEthernet0
 vrf forwarding Mgmt-intf
 no ip address
 

Re: [cisco-voip] ATA190

[NateCCIE]

Ah, but the 2nd port doesn’t have an actual MAC address, just a device name 
based on the mac of the device.

 

From: cisco-voip  On Behalf Of Anthony 
Holloway
Sent: Tuesday, May 22, 2018 10:55 AM
To: Ryan Huff 
Cc: Jon Fox ; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] ATA190

 

Aren't the first few values of a MAC tied to the vendor?  If so, does this 
trick make it look like the second port is a different vendor product?  And if 
so, it would be funny if it was a competitor.

 

Ah, but no such luck today.

 



 

Source: https://macvendors.com/

 

On Tue, May 22, 2018 at 4:30 AM, Ryan Huff  > wrote:

Yes, that is correct. 

The ports are differentiated by the device name. However, the ports themselves 
are registered to CCM and communicate on the network through a single network 
interface on the ATA.

The second port in the ATA will have the first two characters striped from the 
beginning of the MAC address and a “01” appended at the end of the MAC address 
(shown in the device name of the two ports).

Essentially, the ATA is a mini, purpose built media conversion switch. A lot 
going on under the hood of those silly little things when you think about it :).

Thanks,

Ryan

> On May 22, 2018, at 04:57, Jon Fox   > wrote:
> 
> Hello All
> 
> Trying to troubleshoot an issue with a Cisco ATA - CUCM 10.5.2SU3
> 
> I've not had to touch these for some time, so cannot remember if its natural 
> behaviour for Port 1 and Port 2 registering with the same IP address? Is that 
> standard? - Screenshot attached.
> 
> 
> 
> 
> Many thanks
> Jon
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Re: [cisco-voip] EngHouse CTI Route Points

[NateCCIE]

Genesis call center does the same thing. I think it’s just a Cisco thing for 
their apps with respect to CTI ports. 

Sent from my iPhone

> On Mar 22, 2018, at 8:40 AM, Anthony Holloway 
>  wrote:
> 
> Can someone please tell me how EngHouse can get away with using only CTI 
> Route Points and no CTI Ports, and yet, still terminate media?
> 
> If they're aren't doing anything fancy, then why do so many, if not all 
> other, CTI applications, require both CTI Route Points and CTI Ports?
> 
> Signed,
> Confused about CTI
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Re: [cisco-voip] e911

[NateCCIE]

This might be a good time to talk about my favorite way to enable 911.

 

Set the interdigit timeout to a small value, like 3-5 seconds.  Then create a 
911 route pattern, and a 911! Pattern, that does not route to 911.  If the user 
dials 911 and stops, the call connects.  If they keep dialing which usually 
what happens on a miss-dial, they get whatever your 911! Pattern is configured 
to do, usually I like block this pattern.

 

-Nate

 

From: Bill Talley <btal...@gmail.com> 
Sent: Wednesday, March 7, 2018 2:22 PM
To: Matthew Loraditch <mloradi...@heliontechnologies.com>
Cc: NateCCIE <natec...@gmail.com>; Ryan Huff <ryanh...@outlook.com>; 
cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] e911

 

Seems like there's two key aspects we need to be concerned with.  1) As I think 
Matthew is pointing out, notifications are only required if notifications are a 
native feature available "without improvement", i.e. add-on components.  2)  We 
now MUST configure direct 911 access without regard to customer complaints or 
PSAP complaints about accidental 911 calls.

 

To answer your question Matthew, I have only ever used CER and Singlewire for 
notifications, sorry I can't provide more feedback.

 

On Wed, Mar 7, 2018 at 3:06 PM, Matthew Loraditch 
<mloradi...@heliontechnologies.com <mailto:mloradi...@heliontechnologies.com> > 
wrote:

As far as I know that feature doesn’t notify anyone internally.

The part of the law I’m referring to is this:

 

“A person engaged in the business of installing, managing, or operating 
multi-line telephone systems shall, in installing, managing, or operating such 
a system for use in the United States, configure the system to provide a 
notification to a central location at the facility where the system is 
installed or to another person or organization regardless of location, if the 
system is able to be configured to provide the notification without an 
improvement to the hardware or software of the system.”

 

 

 



 


Matthew Loraditch



Sr. Network Engineer




p:   443.541.1518



w:  <http://www.heliontechnologies.com/> www.heliontechnologies.com

 | 

e:  <mailto:mloradi...@heliontechnologies.com> mloradi...@heliontechnologies.com









 <https://facebook.com/heliontech> 


 <https://twitter.com/heliontech> 


 <https://www.linkedin.com/company/helion-technologies> 

 

From: NateCCIE [mailto:natec...@gmail.com <mailto:natec...@gmail.com> ] 
Sent: Wednesday, March 7, 2018 3:58 PM
To: Matthew Loraditch <mloradi...@heliontechnologies.com 
<mailto:mloradi...@heliontechnologies.com> >; 'Ryan Huff' <ryanh...@outlook.com 
<mailto:ryanh...@outlook.com> >; cisco-voip@puck.nether.net 
<mailto:cisco-voip@puck.nether.net> 
Subject: RE: [cisco-voip] e911

 

Um, I thought it did.

 

https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/200452-Usage-of-Native-Emergency-Call-Routing-F.html

 

 

From: cisco-voip <cisco-voip-boun...@puck.nether.net 
<mailto:cisco-voip-boun...@puck.nether.net> > On Behalf Of Matthew Loraditch
Sent: Wednesday, March 7, 2018 1:36 PM
To: Ryan Huff <ryanh...@outlook.com <mailto:ryanh...@outlook.com> >; 
cisco-voip@puck.nether.net <mailto:cisco-voip@puck.nether.net> 
Subject: Re: [cisco-voip] e911

 

To piggy back on this, while Cisco doesn’t have emergency notifications built 
in, as the law mentions, and thus they are not required, does anyone know of 
options beyond Singlewire that they are happy with? The installs would monitor 
up to 1000 or so handsets but the folks that would be notified would probably 
be fewer than 50.

 

 



 


Matthew Loraditch



Sr. Network Engineer




p:   443.541.1518



w:  <http://www.heliontechnologies.com/> www.heliontechnologies.com

 | 

e:  <mailto:mloradi...@heliontechnologies.com> mloradi...@heliontechnologies.com









 <https://facebook.com/heliontech> 


 <https://twitter.com/heliontech> 


 <https://www.linkedin.com/company/helion-technologies> 

 

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan 
Huff
Sent: Wednesday, March 7, 2018 3:11 PM
To: cisco-voip@puck.nether.net <mailto:cisco-voip@puck.nether.net> 
Subject: [cisco-voip] e911

 

I wonder how cloud-based phone system like Cisco spark will answer this?

 

 

https://www.linkedin.com/pulse/karis-law-you-compliant-edgar-salazar

Sent from my iPhone


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Re: [cisco-voip] e911

[NateCCIE]

Um, I thought it did.

 

https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/200452-Usage-of-Native-Emergency-Call-Routing-F.html

 

 

From: cisco-voip  On Behalf Of Matthew 
Loraditch
Sent: Wednesday, March 7, 2018 1:36 PM
To: Ryan Huff ; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] e911

 

To piggy back on this, while Cisco doesn’t have emergency notifications built 
in, as the law mentions, and thus they are not required, does anyone know of 
options beyond Singlewire that they are happy with? The installs would monitor 
up to 1000 or so handsets but the folks that would be notified would probably 
be fewer than 50.

 

 



 


Matthew Loraditch



Sr. Network Engineer




p:   443.541.1518



w:   www.heliontechnologies.com

 | 

e:   mloradi...@heliontechnologies.com









  


  


  

 

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan 
Huff
Sent: Wednesday, March 7, 2018 3:11 PM
To: cisco-voip@puck.nether.net  
Subject: [cisco-voip] e911

 

I wonder how cloud-based phone system like Cisco spark will answer this?

 

 

https://www.linkedin.com/pulse/karis-law-you-compliant-edgar-salazar

Sent from my iPhone

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Re: [cisco-voip] Moving users between OUs

[NateCCIE]

I think the problem here is he has multiple LDAP Sources, and the user is 
moving from one sync agreement to the other.  I wonder if he can reconfigure to 
a higher level in the tree so it doesn’t move between sources.

 

(but is always visible to one of the configured CUCM LDAP Directories),

 

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Malandruccolo, Jim
Sent: Friday, February 16, 2018 9:57 AM
To: Anthony Holloway 
Cc: cisco-voip voip list 
Subject: Re: [cisco-voip] Moving users between OUs

 

I am on CUCM 10.5.2 and see the same behavior as James, so it might be specific 
to this version. If a user is moved out of a watched OU, the RDP is deleted 
from the system.

 

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Anthony Holloway
Sent: Thursday, February 15, 2018 3:11 PM
To: James Andrewartha  >
Cc: cisco-voip voip list  >
Subject: Re: [cisco-voip] Moving users between OUs

 

I agree with Brian, that moving users around in the same search base is no 
issue at all.  In fact, moving users to new domains is no problem either, just 
as long as you're setup correctly to find users in both domains.

 

I'm confused as to what you did exactly.  In any case, I wouldn't think an RDP 
would be deleted from the CUCM config, just because the assigned user went 
missing.  Granted, it's a required field, and unlike a phone, it doesn't have 
an anonymous option; nevertheless, I just tried this on my CUCM 
11.5.1.11900-26, and the RDP remained there.

 



 

 

On Thu, Feb 15, 2018 at 8:45 AM Brian Meade  > wrote:

Moving between OUs should work fine.  CUCM keeps track of the Object ID so 
things like first name/last name changes won't break either.  There was a bug 
where last name changes was creating a new user at one point though.

 

On Thu, Feb 15, 2018 at 1:51 AM, James Andrewartha  > wrote:

Hi list,

We're moving our users to a new OU in AD, and of course I'm the guinea
pig. I moved myself before reconfiguring CUCM, and unsurprisingly all my
permissions and device ownership/associations were stripped. What I
didn't expect was that my remote destination/profile seems to have
disappeared entirely. Is this expected? Is there anything else that
might have gone AWOL? CUCM 10.5.2. Similarly for CUC 10.5.2.

More generally, if a user moves between OUs (but is always visible to
one of the configured CUCM LDAP Directories), will there be any problems?

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] UC server performance and UCCX agent in reserve

[NateCCIE]

And that is why SAS is going to take over.  Systems management is easier to 
justify at larger scales. 

Sent from my iPhone

> On Dec 15, 2017, at 11:42 PM, Anthony Holloway 
>  wrote:
> 
> Out of curiosity, how long had Tomcat been running before you restarted it?
> 
> This isn't at you Terry, but in general.
> 
> Companies will spend a lot of money getting systems in place, but then 
> completely forget that technology has a life cycle; leading towards a better 
> experience.  And no, I don't just mean upgrade to the latest shiny version.  
> I mean, efficiency, features, user experience, stability, scale, shorter MTTR.
> 
> Without being able to quantify it, I have seen more than a comfortable amount 
> of environments without: a pre-production environment, proper analytics, 
> proper change control, a good monitoring solution (emails from RTMT don't 
> count), resource usage monitoring, a good backup strategy, vmtools up to 
> date, and anything other than just MACD work being performed.
> 
> It's like there's this sole effort on "projects," and the old saying: "if 
> isn't broke, don't fix it," wins again. We lose the chance to truly 
> understand our systems, and therefore the chance to optimize them.
> 
> /rant
> 
> Disclaimer: Today was a long cutover, and I'm tired
> 
> PS Ryan amazes me too.
> 
>> On Thu, Dec 14, 2017 at 10:32 PM Terry Oakley  wrote:
>> Thank you again Ryan.   I think I found the issue.   One of the tests showed 
>> a problem with AXL services.  Restarted Tomcat and we appear to be much 
>> better.
>> 
>>  
>> From: Terry Oakley
>> Sent: Thursday, December 14, 2017 5:29:31 PM
>> To: Ryan Huff
>> 
>> Cc: cisco-voip@puck.nether.net
>> Subject: Re: [cisco-voip] UC server performance and UCCX agent in reserve
>> Thanks Ryan.. .I will have a look tonight.. 
>> 
>> PS i don't know how you find all the time to respond to all of us but I am 
>> very thankful that you do.  
>> From: Ryan Huff 
>> Sent: Thursday, December 14, 2017 5:26:53 PM
>> To: Terry Oakley
>> Cc: cisco-voip@puck.nether.net
>> Subject: Re: [cisco-voip] UC server performance and UCCX agent in reserve
>>  
>> Just based on that description alone, I’d say it might be possible you have 
>> some LAN congestion? 
>> Everything you’re talking about here is riding http/https.
>> 
>> - Any recent QoS policy changes?
>> 
>> - Is other non-UC web traffic slower than normal from those PCs?
>> 
>> - Run utils diagnose test on the CLI of each server and see if you find any 
>> goodies ...
>> 
>> -Ryan
>> 
>> On Dec 14, 2017, at 7:18 PM, Terry Oakley  wrote:
>> 
>>> For the past week and a bit I have noticed a decline in UC (Call Manager) 
>>> response time when editing/adding a device.   The message 'loading' stays 
>>> on for 5 to 10 seconds or even longer.   Page refresh is also really slow.  
>>>  In looking at RTMT the CPU/Memory/disk space are all around 50% or less 
>>> with no apparent spikes.   Any suggestions on where this lag could be?
>>> 
>>> On another but may be related , a couple of our agents (but not all) both 
>>> have had their phones restart while in use, and today both had their agent 
>>> go into Reserved state for a couple of minutes before finally connecting 
>>> and allowing them service. Again any suggestions on where one would 
>>> look would be appreciated.
>>> 
>>> UC 11.5 SU3
>>> UCCX 11.5
>>> IMP 11.5 SU3
>>> O365
>>> Unity Connection 11.5
>>> 
>>> Terry
>>> 
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Re: [cisco-voip] 8800 Series firmware upgrade 12.0(1) text color

[NateCCIE]

The firmware has a feature to detect if the background is dark or light, and 
then adjust the labels as needed. But it has to be really dark to be dark. For 
the majority of picture. 

Sent from my iPhone

> On Dec 13, 2017, at 7:50 PM, Haas, Neal  wrote:
> 
> I wanted to push the new 12.0(1) firmware to our 8851/8861 phones. I did a 
> test and you cannot even see the line text on dark backgrounds, which is 90% 
> of the images . Anyone know how to fix line text color? The only image that 
> really works is the one that looks like someone socked the phone.
> 
> Neal
> 
> 
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Re: [cisco-voip] CME phonebook matching (Incoming Caller ID)

[NateCCIE]

You can do it with a custom TCL script.

https://supportforums.cisco.com/t5/ip-telephony/replace-caller-id-by-name-re
questing-an-external-database-using/td-p/2337460

 

 

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
Quenten Grasso
Sent: Monday, December 11, 2017 7:24 PM
To: 'cisco-voip@puck.nether.net' 
Subject: [cisco-voip] CME phonebook matching (Incoming Caller ID)

 

Hi All,

 

Does anyone know if its possible to program the local directory with a bunch
of numbers and when a call comes in, it displays their name as well instead
of just the incoming phone number? In some other systems, I believe its
called Phonebook Matching.

 

Were running c2800nm-adventerprisek9-mz.151-3.T4

 

Cheers,

Quenten

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Re: [cisco-voip] phone model and quantity report...

[NateCCIE]

There is a report in cisco unified reporting, I think device counts summary.

-Original Message-
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
Lelio Fulgenzi
Sent: Wednesday, November 29, 2017 2:04 PM
To: voyp list, cisco-voip (cisco-voip@puck.nether.net)

Subject: [cisco-voip] phone model and quantity report...


Is there any way to get a dump of phone models and the quantity of each. No
Bulk admin report has this information and I can't copy/paste from RTMT.

I'd rather not have to do a screen scrape.

Thoughts?


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Re: [cisco-voip] T1 PRI configuration on 2921 not matching startup config

[NateCCIE]

The router gets the whole config snip it CUCM thinks it needs via TFTP.  If you 
don’t want cucm to configure the ports, then you don’t put a config on the 
ports in CUCM.

 

I agree with Evgeny, it looks like the router is short DSPs for the PRIs.

 

-Nate

 

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Lelio 
Fulgenzi
Sent: Monday, October 2, 2017 6:16 PM
To: Evgeny Izetov 
Cc: Cisco VoIP Group 
Subject: Re: [cisco-voip] T1 PRI configuration on 2921 not matching startup 
config

 

 

That was definitely one issue. I was pretty sure I entered that command 
manually, but apparently not. So that's why the last one did not come up. 

 

Still very weird. I had always assumed that the router asked for the 
configuration for enabled ports only. 

Sent from my iPhone


On Oct 2, 2017, at 6:24 PM, Evgeny Izetov  > wrote:

I wonder if you only had enough DSPs for those 2 and 1/3 PRIs.

 

On Oct 2, 2017 4:02 PM, "Lelio Fulgenzi"  > wrote:

 

Well, apparently, CUCM enabling of ports overrides the router configuration. 

 

I did have all four ports configured. After deleting those two ports, the 
running now matches the startup.

 

This is very weird. Is there a configuration command that says only download 
configuration for the ports I’ve enabled?

 

 

 

---

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: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net 
 ] On Behalf Of Lelio Fulgenzi
Sent: Monday, October 02, 2017 2:29 PM
To: voyp list, cisco-voip (cisco-voip@puck.nether.net 
 )
Subject: [cisco-voip] T1 PRI configuration on 2921 not matching startup config

 

 

OK – Having a weird issue. I was pretty sure that the router config took 
precedence over CUCM (with respect to enabling ports). 

 

I’ve got four of  the four ports configured on CUCM. So not sure why three are 
being configured on the router.

 

Very strange!

 

My startup looks something like this:

 

controller T1 0/0/0

cablelength long 0db

pri-group timeslots 1-24 service mgcp

!

controller T1 0/0/1

cablelength long 0db

pri-group timeslots 1-24 service mgcp

!

controller T1 0/0/2

shutdown

cablelength long 0db

!

controller T1 0/0/3

shutdown

cablelength long 0db

!



!

interface Serial0/0/0:23

no ip address

encapsulation hdlc

isdn switch-type primary-dms100

isdn incoming-voice voice

isdn bind-l3 ccm-manager

no cdp enable

!

interface Serial0/0/1:23

no ip address

encapsulation hdlc

isdn switch-type primary-dms100

isdn incoming-voice voice

isdn bind-l3 ccm-manager

isdn bchan-number-order ascending

no cdp enable



voice-port 0/0/0:23

echo-cancel coverage 64

!

voice-port 0/0/1:23

!

 

 

 

But after a reload, my running config looks something like this:

 

controller T1 0/0/0

cablelength long 0db

pri-group timeslots 1-24 service mgcp

!

controller T1 0/0/1

cablelength long 0db

pri-group timeslots 1-24 service mgcp

!

controller T1 0/0/2

cablelength long 0db

pri-group timeslots 1-8,24 service mgcp

!

controller T1 0/0/3

cablelength long 0db

!



interface Serial0/0/0:23

no ip address

encapsulation hdlc

isdn switch-type primary-dms100

isdn incoming-voice voice

isdn bind-l3 ccm-manager

no cdp enable

!

interface Serial0/0/1:23

no ip address

encapsulation hdlc

isdn switch-type primary-dms100

isdn protocol-emulate network

isdn incoming-voice voice

isdn bind-l3 ccm-manager

no cdp enable

!

interface Serial0/0/2:23

no ip address

encapsulation hdlc

isdn switch-type primary-dms100

isdn incoming-voice voice

isdn bind-l3 ccm-manager

no cdp enable

!



!

voice-port 0/0/0:23

echo-cancel coverage 64

!

voice-port 0/0/1:23

echo-cancel coverage 64

!

voice-port 0/0/2:23

echo-cancel coverage 64

 

---

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] Call Forwarding on secondary lines in 11.5 with 8851?

[NateCCIE]

8851s do it just fine.

 

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Jason 
Aarons (Americas)
Sent: Wednesday, August 9, 2017 10:31 AM
To: cisco-voip (cisco-voip@puck.nether.net) 
Subject: [cisco-voip] Call Forwarding on secondary lines in 11.5 with 8851?

 

 

Any fixes in CUCM 11.5 that allow you to call forward a secondary line from the 
phone, or is that still a feature request from 2001 

 

 

-jason



This email and all contents are subject to the following disclaimer:
  
"http://www.dimensiondata.com/emaildisclaimer; 

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Re: [cisco-voip] um, where'd the scheduled backups go in PLM

[NateCCIE]

I have removed all of the standalone PLM/ELMs from my customers.  It just
didn't make sense to have to worry about another box.  I never found a use
case that having a separate box actually made a difference.

-Original Message-
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
Lelio Fulgenzi
Sent: Tuesday, June 27, 2017 11:41 AM
To: Pawlowski, Adam ; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] um, where'd the scheduled backups go in PLM

Thanks for the feedback. I'll have to look at the CLI scheduling options. 

I see that it's going away in v12 (smart only) but we'll be on v11 for the
next couple of years at least. So I'd like to see this working. We'll have
to watch out for server restarts. I have a script that checks for daily
backups so that would theoretically catch it.


---
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: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
Pawlowski, Adam
Sent: Tuesday, June 27, 2017 1:33 PM
To: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] um, where'd the scheduled backups go in PLM

You can schedule it from the CLI but I find that restarting this machine in
the past has turned this backup off, and I would have to go reschedule it
and turn it on.

If you're using it for usage reporting this is probably more concerning than
if it is just providing licensing. Since we don't often issue new licenses
out into our system the "restore" would just be missing that usage data but
would be fine to restore licensing should we have to rebuild this machine.
(Standalone PLM).

Supposedly it is gone after 11.5 anyways so I am not too concerned about it.
I don't know how the "satellite" works at this point to tell if it is just
the same thing as PLM again or not.

Best,

Adam Pawlowski
SUNYAB NCS




--


Message: 5
Date: Tue, 27 Jun 2017 14:31:22 +
From: Lelio Fulgenzi 
To: Ryan Huff 
Cc: "voyp list, cisco-voip (cisco-voip@puck.nether.net)"

Subject: Re: [cisco-voip] um, where'd the scheduled backups go in PLM
Message-ID:



Content-Type: text/plain; charset="utf-8"


Yes ? we?re looking at a stand alone PLM. We have multiple voicemail and
cucm clusters and liked the idea of separate xLMs.

In the GUI, I just have backup/restore. That?s it. No devices, no schedules,
nothing. Not even # of copies to keep. Docs say that?s limited to 2.



---
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: Ryan Huff [mailto:ryanh...@outlook.com]
Sent: Tuesday, June 27, 2017 10:19 AM
To: Lelio Fulgenzi
Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net)
Subject: Re: [cisco-voip] um, where'd the scheduled backups go in PLM

Are you dealing with a standalone PLM? If it's co-res, you'd just backup the
PLM feature in the normal CUCM DRS.

For standalone, you should be able to use all the normal CLI DRS utilities
(or activate the CUCM application for the purpose of accessing DRS).

Thanks,

Ryan

On Jun 27, 2017, at 10:13 AM, Lelio Fulgenzi
> wrote:

So, just setting up PLM, and I noticed, surprisingly enough, that the
backup/restore menu is a simplified menu of what ELM had.

Basically, just backup/restore.

No scheduled backups, no different backup device configurations.

While I can appreciate not much changing in the licensing manager, it?s
still a drag to see this gone. We built some workflow and recovery
strategies based on this.

Any ideas as to why this happened?


---
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] CUCM 11.5 Tomcat Service SSL Certificate Issue

[NateCCIE]

I don't think you can upload a cert unless there is an active CSR for it.  

Sent from my iPhone

> On May 16, 2017, at 2:12 PM, Brian Meade  wrote:
> 
> You can re-install the same certs. Just make sure to do the trusts, Root then 
> Intermediate then do the server cert and restart services.  Unfortunately, I 
> don't think it shows the root certs anywhere.  Maybe in the certinfo table?
> 
>> On Tue, May 16, 2017 at 4:05 PM, Gary Parker  wrote:
>> 
>> > On 16 May 2017, at 20:42, Brian Meade  wrote:
>> >
>> > Did you make sure to upload those certs in the right order so CUCM was 
>> > able to chain them?
>> 
>> I’ve a feeling that may be the issue. Certs where installed towards the end 
>> of a very long weekend upgrading the cluster and I was losing consciousness 
>> through lack of caffeine :-)
>> 
>> Strange thing is, if that’s my mistake I made it on the publisher and all 
>> four subs, but not the IM nodes and Unity Connection, which seems odd. Is 
>> there any way to check whether CUCM has the certificate relationship right? 
>> I mean, other than creating a CSR and getting and installing new certs.
>> 
>> Gary
> 
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Re: [cisco-voip] UCCx SSO

[NateCCIE]

Going to try one this week. 

Sent from my iPhone

> On May 16, 2017, at 2:19 PM, Scott Voll  wrote:
> 
> Anyone have SSO working with UCCx 11.5su1?
> 
> Have had a TAC case open for a while and can't get it fixed.  Just wondering 
> if anyone else has it working?  CM / UC / CUPS etc went fine.
> 
> Just seeing if anyone has it working.
> 
> Thanks
> 
> scott
> 
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Re: [cisco-voip] CUCM 11.5 Tomcat Service SSL Certificate Issue

[NateCCIE]

Are you using cuplogin or cisco-uds for discovery now?  If your UC services or 
system/server is not fqdn and is IP address then the client will complains 
about the cert unless the ip is listed as a SAN. If cup login make sure your 
tftp server is fqdn over in IM 

Sent from my iPhone

> On May 16, 2017, at 11:57 AM, Brian Meade  wrote:
> 
> Intermediate
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Re: [cisco-voip] Migrating a Cisco 2901 to a Cisco 4321

[NateCCIE]

What do mean by shareable to the backplane?  It is my understanding that unused 
DSPs on a NIM can be used for conferencing/transcoding/MTP.  

Sent from my iPhone

> On May 12, 2017, at 2:00 PM, Ryan Huff <ryanh...@outlook.com> wrote:
> 
> I think that is where my lack of specificity comes into play; the NIM 
> conversation I thought I was participating in was an extension of a convo 
> this AM, regarding a T1 PRI, in which those DSP are reserved for TDM only, 
> and not shareable to the backplane or vice a versa.
> 
> Thanks,
> 
> Ryan
> 
> On May 12, 2017, at 3:46 PM, NateCCIE <natec...@gmail.com> wrote:
> 
>> TDM DSPs on the 4ks have to be on the NIM because there is no shared TDM 
>> clocking backplane like there is on the ISRs/ISR G2. 
>> 
>> Dspfarm stuff can use extra dsps on a NIM and the motherboard DSPs can only 
>> be used for dspfarm tasks. 
>> 
>> Sent from my iPhone
>> 
>> On May 12, 2017, at 1:35 PM, Jose Colon II <jcolon...@gmail.com> wrote:
>> 
>>> I was under the same assumption of why there was a dsp slot on the NIM. I 
>>> know I read a Cisco doc somewhere that lead me that direction.
>>> 
>>>> On May 12, 2017 2:28 PM, "Ryan Huff" <ryanh...@outlook.com> wrote:
>>>> So this is interesting; I was under the impression the backplane DSP could 
>>>> not extend to the NIM (and is the fundamental reason, among others, that 
>>>> the NIM has its own DSP)  looks like I have a new lab task :).
>>>> 
>>>> On May 12, 2017, at 3:09 PM, Anthony Holloway 
>>>> <avholloway+cisco-v...@gmail.com> wrote:
>>>> 
>>>>> Rashmi Patel ( rashmika.pa...@zones.com ) - 12:31 PM
>>>>> Q: Does that mean conference resource will be used from NIM DSP not from 
>>>>> mother board DSP resources
>>>>> Priority: N/A
>>>>> Dolan Spitler - 12:57 PM
>>>>> A: When it comes to IP services (xcoding, conferencing, MTP) The 
>>>>> motherboard DSP can be pooled with the NIM DSPs to increase the DSPfarm 
>>>>> scale
>>>>> 
>>>>> Source: https://communities.cisco.com/docs/DOC-7823 (look for the 4000 
>>>>> event on Oct 16th)
>>>>> 
>>>>>> On Fri, May 12, 2017 at 1:17 PM Brian Meade <bmead...@vt.edu> wrote:
>>>>>> My understanding is any dspfarm resources such as 
>>>>>> conferencing/transcoding use the motherboard resources while the ones on 
>>>>>> the NIM are just for the voice ports themselves.
>>>>>> 
>>>>>>> On Fri, May 12, 2017 at 11:51 AM, Jose Colon II <jcolon...@gmail.com> 
>>>>>>> wrote:
>>>>>>> If you will be using DSP's you will need to decided if you will need 
>>>>>>> them on the motherboard or on the T1 NIM. On the motherboard I believe 
>>>>>>> it can only be used for conferencing. You will need them on the NIM for 
>>>>>>> transcoding.
>>>>>>> 
>>>>>>>> On May 12, 2017 6:42 AM, "Ryan Huff" <ryanh...@outlook.com> wrote:
>>>>>>>> T1 PRI commands are substantially different if that is in play.
>>>>>>>> 
>>>>>>>> Sent from my iPhone
>>>>>>>> 
>>>>>>>> On May 12, 2017, at 7:35 AM, Matthew Loraditch 
>>>>>>>> <mloradi...@heliontechnologies.com> wrote:
>>>>>>>> 
>>>>>>>>> The vast majority of commands are the same. Netflow stuff is changed 
>>>>>>>>> completely if you use that.  Outside of updating interface names, 
>>>>>>>>> most of our templates just worked.
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> Matthew G. Loraditch – CCNP-Voice, CCNA-R, CCDA
>>>>>>>>> Network Engineer
>>>>>>>>> Direct Voice: 443.541.1518
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> Facebook | Twitter | LinkedIn | G+
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> From: cisco-vo

Re: [cisco-voip] Migrating a Cisco 2901 to a Cisco 4321

[NateCCIE]

TDM DSPs on the 4ks have to be on the NIM because there is no shared TDM 
clocking backplane like there is on the ISRs/ISR G2. 

Dspfarm stuff can use extra dsps on a NIM and the motherboard DSPs can only be 
used for dspfarm tasks. 

Sent from my iPhone

> On May 12, 2017, at 1:35 PM, Jose Colon II  wrote:
> 
> I was under the same assumption of why there was a dsp slot on the NIM. I 
> know I read a Cisco doc somewhere that lead me that direction.
> 
>> On May 12, 2017 2:28 PM, "Ryan Huff"  wrote:
>> So this is interesting; I was under the impression the backplane DSP could 
>> not extend to the NIM (and is the fundamental reason, among others, that the 
>> NIM has its own DSP)  looks like I have a new lab task :).
>> 
>> On May 12, 2017, at 3:09 PM, Anthony Holloway 
>>  wrote:
>> 
>>> Rashmi Patel ( rashmika.pa...@zones.com ) - 12:31 PM
>>> Q: Does that mean conference resource will be used from NIM DSP not from 
>>> mother board DSP resources
>>> Priority: N/A
>>> Dolan Spitler - 12:57 PM
>>> A: When it comes to IP services (xcoding, conferencing, MTP) The 
>>> motherboard DSP can be pooled with the NIM DSPs to increase the DSPfarm 
>>> scale
>>> 
>>> Source: https://communities.cisco.com/docs/DOC-7823 (look for the 4000 
>>> event on Oct 16th)
>>> 
 On Fri, May 12, 2017 at 1:17 PM Brian Meade  wrote:
 My understanding is any dspfarm resources such as conferencing/transcoding 
 use the motherboard resources while the ones on the NIM are just for the 
 voice ports themselves.
 
> On Fri, May 12, 2017 at 11:51 AM, Jose Colon II  
> wrote:
> If you will be using DSP's you will need to decided if you will need them 
> on the motherboard or on the T1 NIM. On the motherboard I believe it can 
> only be used for conferencing. You will need them on the NIM for 
> transcoding.
> 
>> On May 12, 2017 6:42 AM, "Ryan Huff"  wrote:
>> T1 PRI commands are substantially different if that is in play.
>> 
>> Sent from my iPhone
>> 
>> On May 12, 2017, at 7:35 AM, Matthew Loraditch 
>>  wrote:
>> 
>>> The vast majority of commands are the same. Netflow stuff is changed 
>>> completely if you use that.  Outside of updating interface names, most 
>>> of our templates just worked.
>>> 
>>>  
>>> 
>>>  
>>> 
>>>  
>>> 
>>> 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 norm.nichol...@kitchener.ca
>>> Sent: Friday, May 12, 2017 7:31 AM
>>> To: cisco-voip@puck.nether.net
>>> Subject: [cisco-voip] Migrating a Cisco 2901 to a Cisco 4321
>>> 
>>>  
>>> 
>>>  
>>> 
>>> We have a base config we use for building our 2901/11’s . Will this 
>>> work on the 4321 or do I have to start from scratch.
>>> 
>>>  
>>> 
>>>  
>>> 
>>>  
>>> 
>>> Thanks
>>> 
>>>  
>>> 
>>>  
>>> 
>>>  
>>> 
>>>  
>>> 
>>> Norm Nicholson
>>> 
>>> Telecom Analyst
>>> 
>>> City of Kitchener
>>> 
>>> (519) 741-2200 x 7000
>>> 
>>>  
>>> 
>>>  
>>> 
>>> ___
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>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>> 
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>> 
> 
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Re: [cisco-voip] using two 3945s for SRST during upgrades - how to create voip dial-peer statements without causing a loop?

[NateCCIE]

You can use COR to limit the inbound dial-peer on the router from seeing the
outbound dial-peer that goes to the other SRST box.  Easy peasy.

-Nate

-Original Message-
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
Lelio Fulgenzi
Sent: Thursday, May 04, 2017 11:30 AM
To: voyp list, cisco-voip (cisco-voip@puck.nether.net)

Subject: [cisco-voip] using two 3945s for SRST during upgrades - how to
create voip dial-peer statements without causing a loop?


Hello folks,

Question regarding voip dial-peers. I've had some experience, but my design
skills are lacking, especially when it comes to something like I'm trying to
do.

Basically, I'd like to take advantage of our two 3945 routers and failover
as many phones as possible.  Problem is, it will be too difficult to fail
them over in ranges.

Can I create a dial-peer that says "5 Pointer to router A" on router B,
and "5 Pointer to router B" on router A and not cause any routing loops?

Is there a built in mechanism that prevents this? Is there something I need
to configure?

If it's too complicated and requires testing and time, I may have to forfeit
the idea of using two routers and use just one and selectively pick those
who failover. I mean, we have to do it anyways, since we have more than the
two routers could handle. And it's a migration model that I might chose to
use anyways, i.e. keep one router connected to the cluster and one not.

Thoughts?

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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Re: [cisco-voip] Automating association devices and profiles to PG_User application user?

[NateCCIE]

You can do a simple SQL insert.  This is a base Query I have used to add all 
hard phones to an app user, that aren’t already associated to the user (for 
apps like phoneview)

 

Add all phones to an app user:

 

INSERT INTO ApplicationUserDeviceMap

(fkdevice, fkApplicationUser,tkuserassociation)

SELECT device.pkid, 'be93a493-3b9b-d0f4-d360-fafb54babb70', '1' from device 
where Device.pkid NOT IN (SELECT fkDevice FROM ApplicationUserDeviceMap AS MAP  
WHERE MAP.fkApplicationUser = 'be93a493-3b9b-d0f4-d360-fafb54babb70') and  
tkclass = '1' and name like 'SEP%'



 

 

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Nick 
Barnett
Sent: Tuesday, April 18, 2017 2:17 PM
To: Cisco VoIP Group 
Subject: [cisco-voip] Automating association devices and profiles to PG_User 
application user?

 

We have automation that builds devices, EM_Profiles, DNs, and just about 
everything else... except for the manual add of the controlled devices to the 
pg user application user for UCCE. We also use Nice for recording and there is 
a nice app user that needs these associations as well.

 

I have figured out how to do this by using a getAppUser AXL call, parsing the 
returned data, inserting my NEW device/profile where appropriate (into a new 
tag) and then submitting the information back as an updateAppUser. I think this 
this the only method we have available to automating this portion.

 

It kind of worries me to do it this way because I can see how the database may 
see it as disassociating all devices from the PG User and then re-associating 
all devices. Depending on processor utilization etc, I can also see where they 
may be a short period of time where the PG user has no associated devices.

 

Are my worries substantiated by any fact? Does anyone else do this? Are there 
better ways to accomplish this task?

 

Thanks,

Nick

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Re: [cisco-voip] CUBE call authorization?

[NateCCIE]

Dial peer groups don't work like you think they would. The round robin between 
all of the dial peers on the outbound side, with no looking at the destination 
pattern.

What I have settled on is using COR to restrict the dial peers.  As with dial 
peers groups above, and always, inbound dial peer matching is paramount.  For 
this I am using incoming URI matching. You can match all sorts of headers like 
from to via etc. 

Sent from my iPhone

> On Mar 9, 2017, at 10:15 AM, Pawlowski, Adam  wrote:
> 
> Carlos,
> 
> I have made use of the dial-peer group feature, available 15.4+, to set this 
> up:
> 
> http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/multiple-outbound-dial-peer.html
> 
> You can more or less use that like an access list to say some particular dial 
> peer can only "see" a list of applicable dial peers to route to (with the 
> listed caveats).
> 
> I did put this together with COR but that was a massive pain. 
> 
> Regards,
> 
> Adam Pawlowski
> SUNYAB NCS
> 
> --
> 
> Message: 9
> Date: Thu, 9 Mar 2017 10:12:21 -0300
> From: Carlos Mendioroz 
> To: cisco-voip@puck.nether.net
> Subject: [cisco-voip] CUBE call authorization ?
> Message-ID: <902f2bde-89bb-df5f-8eb8-78f308c06...@huapi.ba.ar>
> Content-Type: text/plain; charset=utf-8
> 
> Hi,
> I'm trying to migrate a CME install to a CUCM, and was thinking of doing
> it gradually. In the end, the CME should be left as a CUBE, terminating
> the ITSPs trunks.
> 
> Now, I do want to have some sort of call authorization just to be on the
> safe side, and not discovering that one SP ended up making calls accross
> my GW. Thought of COR lists, but I found no easy way
> to link a dial peer to an incoming call from a given SIP trunk.
> 
> I'm currently using a prefix as a enablement "secret", but there has to
> be a better way. I'm embarrased to admit I don't see it.
> Help ?
> 
> -- 
> Carlos G Mendioroz    LW7 EQI  Argentina
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Re: [cisco-voip] Moving calls to/from jabber apps

[NateCCIE]

I would be astonished if hold/resume did not work. 

Sent from my iPhone

> On Feb 18, 2017, at 2:30 PM, Lelio Fulgenzi  wrote:
> 
> So, if someone answers a Jabber call on their mobile phone (via jabber 
> client) and walks into their office and wants to continue the call on their 
> desktop, this is not possible?
> 
> I was hoping at least a hold and resume function was there. 
> 
> I see the use case. And when my manager asks, it's hard for me to argue. 
> 
> Sent from my iPhone
> 
> On Feb 18, 2017, at 4:15 PM, Anthony Holloway 
>  wrote:
> 
>> Why do you think that this is possible at all?  Or are you just seeing if it 
>> is possible?  I haven't heard of anyone ever doing this in the past.
>> 
>>> On Sat, Feb 18, 2017 at 9:49 AM Lelio Fulgenzi  wrote:
>>> 
>>> For the life of me, I can't seem to figure out how to move a jabber (video) 
>>> call from one app to another, i. e. windows to mobile and back.
>>> 
>>> What am I missing?
>>> 
>>> Sent from my iPhone
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Re: [cisco-voip] Teleconferencing question

[NateCCIE]

Look at the shire micro flex advance ceiling mount. 

http://www.shure.com/americas/microflex-advance

Watch the demo videos and be amazed. 

Sent from my iPhone

> On Feb 13, 2017, at 10:37 AM, Ben Amick  wrote:
> 
> Read below for my initial thoughts, but I just did a google since I went to a 
> cisco facility the other day and saw the new cisco Telepresence ceiling mics, 
> which are 4-pin mini jack and are compatible with even the low end video 
> solutions such as the SX10/20, which lowers the price point of your possible 
> video solution, and the audioscience I mention below is now EoL. (link for 
> those curious: 
> http://www.cisco.com/c/en/us/products/collateral/collaboration-endpoints/telepresence-microphones/datasheet-c78-736073.html)
>  
> If it wasn’t for the malleability of the tables, I’d suggest getting a daisy 
> chain unit to extend out both the speaker and the microphones across a longer 
> distance.
>  
> However, if you’re going to look at teleconferencing with the possibility of 
> video, I’d look for an C-series telepresence codec with the XLR based 
> Audioscience ceiling microphones. The configuration of those would allow you 
> to suspend them around the room. This would also enable you to gain much 
> better audio quality around the room, as well as be able to properly 
> integrate wall-mounted displays and the such and multiple video inputs.
>  
> Short of that, an MX 7/800 series (or SX series if you already have displays) 
> unit could enable you for videoconferencing with less facility tie-in, and 
> the higher end units also support the XLR based audioscience mics.
>  
> Ben Amick
> Telecom Analyst
>  
> From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
> Terry Oakley
> Sent: Monday, February 13, 2017 11:55 AM
> To: cisco-voip@puck.nether.net
> Subject: [cisco-voip] Teleconferencing question
>  
> We have a board room 25’ by 40’, that we currently have an 8831 conferencing 
> unit in with the two external mics.   The unit works well but with only the 
> two mics does not adequately cover the room.   Do any of you have suggestions 
> as to how we can add the necessary mic coverage into the room but at the same 
> time keep it neat and tidy?   The room is often use for our College’s Board 
> meeting and will have 18 to 30 guests invited to the meeting.   The setup is 
> 12 tables on castors, that are arranged in an oval but can be separated or 
> arranged differently for other events.
>  
> We do have WebEx, IM available if something there would allow us to 
> provide a better teleconferencing solution.   I am think we should just do a 
> full blown conferencing suite, video and tele but want to know if there are 
> other solutions that I have not seen or more likely thought of.
>  
> 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) 343-4034
>  
> 
> Confidentiality Note: This message is intended for use only by the individual 
> or entity to which it is addressed and may contain information that is 
> privileged, confidential, and exempt from disclosure under applicable law. If 
> the reader of this message is not the intended recipient or the employee or 
> agent responsible for delivering the message to the intended recipient, you 
> are hereby notified that any dissemination, distribution or copying of this 
> communication is strictly prohibited. If you have received this communication 
> in error, please contact the sender immediately and destroy the material in 
> its entirety, whether electronic or hard copy. Thank you
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Re: [cisco-voip] CUCM Upgrade 10.0 to 11.5 only via PCD?

[NateCCIE]

The docs are pretty clear no upgrading with or without PCD from 10.0.  You can 
do a PCD migration from 10.0 to 11.5, but that is a different paradigm.

 

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/compat/11_x/cucm_b_cucm-imp-compatibility-matrix-11x.html#reference_7F05AD16F0D73E4256C4590B3B34F502

 

 

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Gary 
Bates_Command Solutions
Sent: Wednesday, January 18, 2017 4:52 PM
To: Schlotterer, Tommy 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] CUCM Upgrade 10.0 to 11.5 only via PCD?

 

We recently tried an os admin upgrade to 11.5 but wod not work so we had to go 
to 11.0 , 

 

Tommy, how do you do a manual upgrade ?

 

Gary


Sent from my iPhone


On 19 Jan 2017, at 6:13 am, Schlotterer, Tommy  > wrote:

You can do a manual upgrade from 10.0 to 11.5. You don’t have to do it from PCD.

 

Thanks

Tommy

 


Tommy Schlotterer | Systems Engineer - Collaboration
Presidio | www.presidio.com  
20 N Saint Clair 3rd Floor, Toledo, OH 43604
D: 419.214.1415 | C: 419.706.0259 | tschlotte...@presidio.com 
 

 

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Heim, 
Dennis
Sent: Wednesday, January 18, 2017 2:06 PM
To: Ben Amick  >; Bruno 
Takahashi  >; 
cisco-voip@puck.nether.net  
Subject: Re: [cisco-voip] CUCM Upgrade 10.0 to 11.5 only via PCD?

 

Don’t expect PCD to work for the first. Just my experience. It’s just as fickle 
as CUCM is. What I am saying is just plan accordingly from the time and outage 
perspect. Have a plan-B.

 

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ben 
Amick
Sent: Wednesday, January 18, 2017 1:49 PM
To: Bruno Takahashi  >; 
cisco-voip@puck.nether.net  
Subject: Re: [cisco-voip] CUCM Upgrade 10.0 to 11.5 only via PCD?

 

Not to 11.5, but I know 10->11 can go fine via standard OS admin, I’ve 
personally done it. I’d be surprised if it was different to 11.5

Worst comes to worst, you could do the incremental to 10.5, and then do OS 
admin to go 10.5->11.5 (or 10->11, then the incremental from 11->11.5)

 

Ben Amick

Telecom Analyst

 

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Bruno 
Takahashi
Sent: Wednesday, January 18, 2017 1:43 PM
To: cisco-voip@puck.nether.net  
Subject: [cisco-voip] CUCM Upgrade 10.0 to 11.5 only via PCD?

 

Hello all,

Looking at the upgrade docs for 11.5 @ 
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/upgrade/11_5_1/cucm_b_upgrade-guide-cucm-115/cucm_b_upgrade-guide-cucm-115_chapter_01100.html

It states that upgrades from 10.0.x to 11.5 are only supported via PCD upgrade.

Has anyone already done this via standard OS Admin?

As I'm not migrating, I don't see any major advantages on using PCD.

Regards.

--


Bruno


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Re: [cisco-voip] Jabber/CIPC and QoS

[NateCCIE]

Or take the most approach of do nothing.

My personal favorite is to use codecs where QoS matters less, like iLBC, OPUS, 
etc. 

So many business are getting rid of the QoS capable WAN and just doing VPNs, 
even if they have fancy names that make it sound better than public internet.

Sent from my iPhone

> On Jan 3, 2017, at 2:25 PM, Ben Amick  wrote:
> 
> So, I know this is an age old question that’s debated, but I’ve been 
> wondering if anyone here has a perspective here in regards to QoS for 
> softphones. Obviously, with hardphones, you usually partition a separate VLAN 
> with AutoQoS/DSCP tags, but that isn’t applicable with softphones.
>  
> I’ve heard of three different options in the past, neither of which seem to 
> be very simple to deploy, but all seem to be Jabber-centric.
> 1.  Configuring windows to perform DSCP tagging, and do DSCP QoS on the 
> switches they are connected to, as well as trusting the device. Problems: 
> Requires users to be local admins, openings for abuse and network impact due 
> to blind PC trust.
> 2.  Configuring your switches with an access list that recognizes the 
> ports Jabber does outbound to attach DSCP tags to them. Problems: Other 
> programs could theoretically use those ports
> 3.  Installing Medianet services on all jabber clients; Configure all 
> switches for medianet tagging. Problem: (I think?) Requires newer switches to 
> use, maybe needs an additional server (I vaguely remember possibly needing 
> prime collab?)?
>  
> Maybe I’m missing some things, but what approach have you guys taken for 
> softphone/Jabber QoS? And on top of that, what options are there for CIPC (I 
> know there’s the auto qos trust cisco-softphone for cisco switches, but I 
> don’t believe there’s a solution other than #1 for non-cisco switches)?
>  
> Ben Amick
> Telecom Analyst
>  
> 
> Confidentiality Note: This message is intended for use only by the individual 
> or entity to which it is addressed and may contain information that is 
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Re: [cisco-voip] CUCM Patch Insight

[NateCCIE]

I would wait for SU2 at this point.  It will be soon.

 

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Jeffrey McHugh
Sent: Tuesday, January 03, 2017 8:53 AM
To: Tim Franklin ; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] CUCM Patch Insight

 

Just did an SU1 upgrade, no issues reported but look into bug CSCux90747 
depending on your esxi versions

 

I would expect SU2 soon as its named in the Expressway 8.9 release notes for 
some MRA feature preview

 

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Tim 
Franklin
Sent: Tuesday, January 3, 2017 10:44 AM
To: cisco-voip@puck.nether.net  
Subject: [cisco-voip] CUCM Patch Insight

 

Just curious if anyone on this list has any feedback as to the stability of 
CUCM 11.5(1)SU1. I'm planning my upgrades out and I'm a bit leery to deploy it 
given that it's been out since November. While that speaks to no large defects 
to cause a deferral notice I'm also wondering if another SU is on the horizon?

 

Thanks


Jeffrey McHugh | Sr. Collaboration Consulting Engineer | VCP-DCV, CCNP 
Collaboration 

  

Fidelus Technologies, LLC
Named  

 Best UC Provider in the USA

240 West 35th Street, 6th Floor, New York, NY 10001 

+1-212-616-7801 office | +1-212-616-7850 fax |   
www.fidelus.com

   
    
 

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Re: [cisco-voip] Are there any gotchas to watch out for switching to FQDN server names from IP address server names?

[NateCCIE]

A reboot does work. What the deal is the new https version of tftp (port 6972) 
does not restart with the service restart.  So it continues to use the old 
cert. But it does stop and start with a service deactivation and reactivation.  
Before cucm 11 the tftp over http was only plain text (port 6970)


Sent from my iPhone

> On Nov 30, 2016, at 1:12 AM, James Buchanan  wrote:
> 
> Hello,
> 
> If I remember right, it actually has to be deactivated under Service 
> Management. It's not just restarting the service.
> 
> Thanks,
> 
> James
> 
>> On Tue, Nov 29, 2016 at 11:36 PM, Derek Andrew  wrote:
>> Would a simple reboot accomplish the same as deactivating and activating?
>> 
>>> On Mon, Nov 28, 2016 at 2:19 PM, Nick Barnett  
>>> wrote:
>>> I just thought I would share what happened with this, even though it is 
>>> super old. Changing the node names to FQDN was mostly painless. The one 
>>> thing that bit me was bug CSCuy13916. After changing the names of the 
>>> nodes, the TFTP service needs to be DEACTIVATED and then re-activated in 
>>> order to fully update the certificates.  Before taking those steps, I kept 
>>> getting certificate errors from CuciLync, but afterwards, everything worked 
>>> as designed.
>>> 
>>> Other than that, any CTI route points (and any other device as well) that 
>>> exist will fall to another node in the CMG. Not a big deal, just something 
>>> to be aware of.
>>> 
>>> Thanks,
>>> Nick
>>> 
 On Wed, Aug 31, 2016 at 3:13 PM, Nick Barnett  
 wrote:
 We are on 10.0 and this cluster has been upgraded over the years from 8.0 
 to 8.6 to 10.0.  I know it used to be common practice to rip the host name 
 out of a new node and put in the IP address. That's how we are set up... 
 but now that I need to do some work with certs so that jabber and cucilync 
 work properly, it's time to fix this.
 
 Is there anything I should watch out for? Anything that may bite me in 
 rare cases? We have CER, CVP, CUC, UCCE and a rarely used IMP.
 
 I checked that each node has DNS enabled by looking at "show network eth0" 
 on each sub. I also then looked up each FQDN from each node and they all 
 resolve properly. As far as I know, that's about it.
 
 Thanks in advance!
 
 nick
>>> 
>> 
>> 
>> 
>> -- 
>> Copyright 2016 Derek Andrew (excluding quotations)
>> 
>> +1 306 966 4808
>> Communication and Network Services
>> Information and Communications Technology
>> Infrastructure Services
>> University of Saskatchewan
>> Peterson 120; 54 Innovation Boulevard
>> Saskatoon,Saskatchewan,Canada. S7N 2V3
>> Timezone GMT-6
>> 
>> Typed but not read.
>> 
>> 
>> 
>> 
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Re: [cisco-voip] Cisco 8821 registration problems

[NateCCIE]

I do, just tried it on Wednesday, but I have not figured out why it's not
working.  I am running 11.5SU1+ES

 

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
Taylor, Matthew
Sent: Friday, October 28, 2016 4:34 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Cisco 8821 registration problems

 

Anyone had problems with a 8821 not registering to CUCM?

 

I'm running 9.1.2 and have installed device pack 9.1(2.15130) and the newest
firmware for the 8821s.  My phone is joined to the same wireless network as
my current 7925s and can pull it's config file from the TFTP server but
never registers.

 

Thank you,

 

Matthew Taylor

Network Architect II, IT - ISS

Sam Houston State University

matthew.tay...@shsu.edu  
(936) 294-3955

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Re: [cisco-voip] CCX 11.5 Upgrade Issues

[NateCCIE]

Not much of a windows guy anymore, (MCSE Windows 2000), but this looks
pretty easy, but I imagine it's under the control of some other
person/group.

 

https://technet.microsoft.com/en-us/library/ff829847(v=ws.10).aspx

 

-Nate

 

From: Matthew Loraditch [mailto:mloradi...@heliontechnologies.com] 
Sent: Sunday, October 23, 2016 9:16 AM
To: NateCCIE <natec...@gmail.com>; 'Ryan Huff' <ryanh...@outlook.com>
Cc: cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] CCX 11.5 Upgrade Issues

 

We use our AD CA for the certs and setting that up to do EC certs is not a
tiny bit of work. Everything I've read basically indicated I have to rebuild
the thing from scratch. The Cert Management page indicates I can actually
turn them off in Enterprise Parameters. but that's not exposed in UCCX.

I'll probably be just using GPO to push the self signed certs to my agent's
PCs for now.

 

From: NateCCIE [mailto:natec...@gmail.com] 
Sent: Sunday, October 23, 2016 11:04 AM
To: Matthew Loraditch <mloradi...@heliontechnologies.com
<mailto:mloradi...@heliontechnologies.com> >; 'Ryan Huff'
<ryanh...@outlook.com>
Cc: cisco-voip@puck.nether.net <mailto:cisco-voip@puck.nether.net> 
Subject: RE: [cisco-voip] CCX 11.5 Upgrade Issues

 

http://www.cisco.com/c/en/us/support/docs/customer-collaboration/unified-con
tact-center-express/200651-UCCX-Version-11-5-Prerelease-Field-Commu.html

 

All of the 11.5 stuff seems to have the ecdsa certs.  Digicert issues them
just fine on their wildcard cert.

 

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
Matthew Loraditch
Sent: Saturday, October 22, 2016 11:00 PM
To: Ryan Huff <ryanh...@outlook.com <mailto:ryanh...@outlook.com> >
Cc: cisco-voip@puck.nether.net <mailto:cisco-voip@puck.nether.net> 
Subject: Re: [cisco-voip] CCX 11.5 Upgrade Issues

 

Thanks. After another reboot. I've got admin pages on the primary. Also some
finesse service is running on yet another port (12015) and giving me
elliptic curve certs. Need to figure out how to disable them.

 

TAC and Football tomorrow!

 

From: Ryan Huff [mailto:ryanh...@outlook.com] 
Sent: Saturday, October 22, 2016 11:16 AM
To: Matthew Loraditch <mloradi...@heliontechnologies.com
<mailto:mloradi...@heliontechnologies.com> >
Cc: cisco-voip@puck.nether.net <mailto:cisco-voip@puck.nether.net> 
Subject: Re: [cisco-voip] CCX 11.5 Upgrade Issues

 

There is an ES for 11.5 FYI; not sure if any of your upgrade issues are
covered in the ES ... but TAC may lead you down that path.

Sent from my iPhone


On Oct 22, 2016, at 11:09 AM, Matthew Loraditch
<mloradi...@heliontechnologies.com
<mailto:mloradi...@heliontechnologies.com> > wrote:

So I did one of these last night, just a few issues..

1)  None of the admin webpage services will start on the primary server.
Tomcat logs don't show anything I understand or look like obvious errors.

2)  Can't login to CUIC or the new Identity services with any
combination of usernames I've tried. Just plain usernames, the built in
admin account, CCX\username, etc. 

 

I've combed documentation for #2, but I'm either missing something or it
doesn't exist. I read the SSO guide for identity services, but it skips over
the login to it this way section and the configuration guide doesn't seem to
mention it.

 

DB Replication is good for both databases CCX and the platform DBs. CCX is
operating, agents can login and queues are working correctly.

 

I'll be calling TAC tomorrow, but if anyone has any insights or bug IDs that
may save me time, I'd appreciate it.

 

-Matthew

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Re: [cisco-voip] CCX 11.5 Upgrade Issues

[NateCCIE]

http://www.cisco.com/c/en/us/support/docs/customer-collaboration/unified-con
tact-center-express/200651-UCCX-Version-11-5-Prerelease-Field-Commu.html

 

All of the 11.5 stuff seems to have the ecdsa certs.  Digicert issues them
just fine on their wildcard cert.

 

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
Matthew Loraditch
Sent: Saturday, October 22, 2016 11:00 PM
To: Ryan Huff 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] CCX 11.5 Upgrade Issues

 

Thanks. After another reboot. I've got admin pages on the primary. Also some
finesse service is running on yet another port (12015) and giving me
elliptic curve certs. Need to figure out how to disable them.

 

TAC and Football tomorrow!

 

From: Ryan Huff [mailto:ryanh...@outlook.com] 
Sent: Saturday, October 22, 2016 11:16 AM
To: Matthew Loraditch  >
Cc: cisco-voip@puck.nether.net  
Subject: Re: [cisco-voip] CCX 11.5 Upgrade Issues

 

There is an ES for 11.5 FYI; not sure if any of your upgrade issues are
covered in the ES ... but TAC may lead you down that path.

Sent from my iPhone


On Oct 22, 2016, at 11:09 AM, Matthew Loraditch
 > wrote:

So I did one of these last night, just a few issues..

1)  None of the admin webpage services will start on the primary server.
Tomcat logs don't show anything I understand or look like obvious errors.

2)  Can't login to CUIC or the new Identity services with any
combination of usernames I've tried. Just plain usernames, the built in
admin account, CCX\username, etc. 

 

I've combed documentation for #2, but I'm either missing something or it
doesn't exist. I read the SSO guide for identity services, but it skips over
the login to it this way section and the configuration guide doesn't seem to
mention it.

 

DB Replication is good for both databases CCX and the platform DBs. CCX is
operating, agents can login and queues are working correctly.

 

I'll be calling TAC tomorrow, but if anyone has any insights or bug IDs that
may save me time, I'd appreciate it.

 

-Matthew

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Re: [cisco-voip] CUCM CAC Locations Usage Report

[NateCCIE]

Isn't LBM for syncing to VCS?

Sent from my iPhone

> On Aug 30, 2016, at 4:31 PM, Anthony Holloway 
>  wrote:
> 
> Yeah, I'm also aware of the perform logging, and I'm not a fan of it.  From 
> back when I did it on UCCX to trend JVM Heap usage, it was a pain in the ass 
> to collect, consolidate (because it was multiple files), and view (you have 
> to make sense of it).  Unless you know of an easier way, I'm all ears.
> 
> Also, consider the scariness where I have not activated LBM on any nodes (oh 
> my god, what?), but I have been creating locations (effectively unlimited 
> BW), and before I turn on Locations Based CAC, I want to get a base line of 
> traffic patterns.  Well, then Perfmon is not going to help me, as these 
> statistics wont be collected with LBM activated.  I know from experience.
> 
> If I have to roll my own tool, I will, but I just thought...I don't know 
> why...but I just thought there might be a built in report/way/mechanism to 
> get this.
> 
>> On Tue, Aug 30, 2016 at 5:02 PM, Brian Meade  wrote:
>> You can set up RTMT to log the data locally in a CSV- 
>> http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/service/7_1_2/rtmt/RTMT/rtconfpm.html#wp1057823
>> 
>> Unfortunately, the counters you're looking for aren't accessible via the 
>> server-stored Perfmon logs.
>> 
>> You could also collect these counters through the Serviceability XML API and 
>> store them somewhere- 
>> http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/devguide/9_1_1/xmldev-911/serviceability.html
>> 
>>> On Tue, Aug 30, 2016 at 5:02 PM, Ryan Huff  wrote:
>>> You'd probably be looking at a billing application that can parse and graph 
>>> CDRs at that point ... IMHO.
>>> 
>>> Sent from my iPhone
>>> 
>>> On Aug 30, 2016, at 4:47 PM, Anthony Holloway 
>>>  wrote:
>>> 
 Thanks Ryan.  I should have mentioned that I am already in RTMT (Why no 
 "Current Bandwidth Usage" counter Cisco?), but what I'm really looking for 
 is, historical reporting.  I need to answer the question: Have calls 
 between any of our sites began to increase or decrease since the 
 introduction of a new tool?  For example, if we deploy Jabber+CSF, will 
 that increase or decrease our intra-company calls?  One could guess it 
 would decrease, since Instant Messaging is now introduced into the 
 environment, but if there's a CSF for convenience, and now video for rich 
 media communications, maybe it actually increases.  I'd like to report on 
 this, instead of doing the typical thing of pushing my usage statistics on 
 the entire user community.  "I don't think many people are using video, 
 because I don't use video."
 
> On Tue, Aug 30, 2016 at 3:21 PM, Ryan Huff  wrote:
> Anthony,
> 
> 
> Although not likely what you are looking for (at least in the consumption 
> method), there are a couple of performance counters you might be able to 
> look at; Location LBM\BandwidthAvailable and Location 
> LBM\VideoBandwidthAvailable. Subtracted from the configured amount, 
> should yield usage.
> 
> 
> = Ryan =
>  
> From: cisco-voip  on behalf of 
> Anthony Holloway 
> Sent: Tuesday, August 30, 2016 3:52 PM
> To: Cisco VoIP Group
> Subject: [cisco-voip] CUCM CAC Locations Usage Report
>  
> I'm not finding much, and I'll keep looking, but does anyone know how to 
> get a bandwidth usage report between locations?  E.g., Peak 
> audio/video/immersive bandwidth consumption between all sites.
> 
> Thanks.
>>> 
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Re: [cisco-voip] Jabber support for Extension Mobility - Any other Solutions out there

[NateCCIE .]

Cisco doesn't seem to care about shared lines, if they are not registered.
You can login to the EM profile when at the desk, then logout.  Then you
can login to jabber when remote.  Just as long as you're not in both you
should be ok.

Thanks,
-Nate

On Tue, Aug 2, 2016 at 8:35 AM, Max Harmony  wrote:

> Hi All,
>
> I have  remote UCCX agents that are using Jabber when they are remote,
> meanwhile in the office, they have shared 8945 IP Phones
>
> Jabber is associated to UCCX End user
>
> We would like to setup Extension Mobility so users can login when they are
> in the office, and at home login to Jabber
>
> Problem:
> 1- Jabber does not support Extension Mobility
> 2- Cisco does not support multiple devices/not shared lines
>
> Customer does not want to use IP communicator, so we are stuck with Jabber
>
> Has anyone got feed back from the Cisco BU about this requirement, if
> Jabber is going to be the future does anyone know of a work around?
>
>
> Appreciate your feedback
>
>>
>> ***
>>
>
>
>
> --
> --
> Grace Maximuangu
>
> CloudPOP/InvictaCloud
> www.cloudpop.com
>
>  *“Go beyond your limits, push yourself, be the best you can be.*
> *Experience new cultures, broaden your horizons, stay connected.”*
>
>
>
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>
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Re: [cisco-voip] Cisco CUCM 11.5(1) ?

[NateCCIE]

I sent the original ones out a couple of weeks ago.

CSCva09650 is for the device search

 <https://tools.cisco.com/bugsearch/bug/CSCva07788> CSCva07788 is for the BAT 
insert

 

I have another case open that is being looked at by the BU.  “No real time data 
in session trace log view of RTMT CUCM 11.5”

 

Thanks,

-Nate

 

From: Justin Steinberg [mailto:jsteinb...@gmail.com] 
Sent: Monday, July 25, 2016 2:52 PM
To: NateCCIE <natec...@gmail.com>
Cc: Heim, Dennis <dennis.h...@wwt.com>; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Cisco CUCM 11.5(1) ?

 

Nate do you mind providing a list of some of the issues you encountered ?

 

On Mon, Jul 25, 2016 at 2:29 PM, NateCCIE <natec...@gmail.com 
<mailto:natec...@gmail.com> > wrote:

I haven't had problems with Certs on my 11.5 clusters.  But I think I did all 
of the Certs when it was running v11.  Was yours a fresh 11.5?  And what do you 
mean by CUCM certificate?  Call manager or tomcat?

 

I have had plenty of other bugs opened. Way to many, seems like a lot more than 
other releases in recent memory. 

Sent from my iPhone


On Jul 25, 2016, at 9:06 AM, Heim, Dennis <dennis.h...@wwt.com 
<mailto:dennis.h...@wwt.com> > wrote:

11.5 thinks the CUPS server needs a CUCM Certificate. The CUPS server rejects 
it, it generates and error to the user by functionality is the same. However, I 
wonder what else has been omitted from the test plan.

 

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Chris 
Osborne (AM)
Sent: Monday, July 25, 2016 10:56 AM
To: cisco-voip@puck.nether.net <mailto:cisco-voip@puck.nether.net> 
Subject: [cisco-voip] Cisco CUCM 11.5(1) ?

 

 

Hey anybody run into any major issues with 11.5 Train of CUCM etc ?  or even 
11.0.1

 

Got a global IPT install with the works and was wondering if anybody had run 
into any major problems ( jabber, IM, call center stuff) ?

 

 

Thanks

Chris Osborne



This email and all contents are subject to the following disclaimer:
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Re: [cisco-voip] Cisco CUCM 11.5(1) ?

[NateCCIE]

I haven't had problems with Certs on my 11.5 clusters.  But I think I did all 
of the Certs when it was running v11.  Was yours a fresh 11.5?  And what do you 
mean by CUCM certificate?  Call manager or tomcat?

I have had plenty of other bugs opened. Way to many, seems like a lot more than 
other releases in recent memory. 

Sent from my iPhone

> On Jul 25, 2016, at 9:06 AM, Heim, Dennis  wrote:
> 
> 11.5 thinks the CUPS server needs a CUCM Certificate. The CUPS server rejects 
> it, it generates and error to the user by functionality is the same. However, 
> I wonder what else has been omitted from the test plan.
>  
> From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
> Chris Osborne (AM)
> Sent: Monday, July 25, 2016 10:56 AM
> To: cisco-voip@puck.nether.net
> Subject: [cisco-voip] Cisco CUCM 11.5(1) ?
>  
>  
> 
> Hey anybody run into any major issues with 11.5 Train of CUCM etc ?  or even 
> 11.0.1
>  
> Got a global IPT install with the works and was wondering if anybody had run 
> into any major problems ( jabber, IM, call center stuff) ?
>  
>  
> Thanks
> 
> Chris Osborne
> 
> 
> This email and all contents are subject to the following disclaimer:
> "http://www.dimensiondata.com/emaildisclaimer;
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Re: [cisco-voip] Max length for analogue

[NateCCIE]

http://www.sandman.com/LongLoop.html

Sandman is the place to go when you have old school analog telcom problems. 

Sent from my iPhone

> On Jun 22, 2016, at 1:50 PM, Louis Koekemoer (ZA) 
>  wrote:
> 
> 
> 
> Hi all, 
> 
> What would be the maximum length one will be able to run an analogue 
> extension or what device can one use to extend the length? 
> 
> Regards 
> 
> Louis 
> 
> 
> 
> Sent from my Samsung device
> 
> 
> This email and all contents are subject to the following disclaimer:
> "http://www.dimensiondata.com/emaildisclaimer;
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Re: [cisco-voip] UCCX - Need help with SQL CLI command to gather some system info about the UCCX system.

[NateCCIE]

Uplinx?

-Original Message-
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
Brian V
Sent: Tuesday, June 21, 2016 10:19 AM
To: Cisco VOIP 
Subject: [cisco-voip] UCCX - Need help with SQL CLI command to gather some
system info about the UCCX system.

All you SQL gurus out there :)

I'm hoping there is a quick way to accomplish a task.

I recently inherited a UCCX 10.6 install with 39 applications defined.

I need to document the application name, the script assigned to the 
application, and the trigger(s) assigned to the application

I only have WebEx access to a shared desktop at the customer. That 
desktop has a browser and Putty.

I have HTTP and SSH access to the UCCX system via the remote desktop.

Does anyone know how to write a quick SQL CLI statement to gather this 
info in one go ?


Thanks !

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