Re: [cisco-voip] MRA DR / Resilience

[Pawlowski, Adam]

For SIP yes, but I can’t tell if it also works with UDS or not.  I’ve a 
question in to find out, but that may be dependent on future releases.

I’ve been going over what happened, and SSO introduces a new layer. A very 
helpful gentleman from TAC spend a bunch of time with me going over how Jabber 
and Expressway sort of handle this.

Expressway, at least in X12.6 where I’m at, has no understanding that a UCM 
node is down as far as UDS is concerned.
Requests can forward to downed UCM which will return a HTTP error status code 
to Jabber
HTTP transaction failures will cause re-auth to be triggered
Jabber can also want to talk to a UCM that isn’t there, sometimes repeatedly 
for some reason instead of choosing a new one

So, there are a number of reasons that may trigger a cycle where Jabber wants 
to verify it’s token validity, tries to talk to nothing a few times, then after 
about 3 retries it will give up and punt the user out.

It’s not clear from looking at the Jabber log (and going cross eyed in the 
process) if Jabber is aware that it can try a different UCM or not. It shows 
the URL being put on a block list, but, then it just uses it again anyway.

Still waiting to learn more about what happened.

Best,

Adam


From: ROZA, Ariel 
Sent: Monday, January 18, 2021 1:59 PM
To: ROZA, Ariel ; NateCCIE ; 
Pawlowski, Adam 
Cc: cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] MRA DR / Resilience

I just reread the release notes, and it includes the case where CUCM is down.

De: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
En nombre de ROZA, Ariel
Enviado el: lunes, 18 de enero de 2021 15:53
Para: NateCCIE mailto:natec...@gmail.com>>; Pawlowski, Adam 
mailto:aj...@buffalo.edu>>
CC: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Asunto: Re: [cisco-voip] MRA DR / Resilience

But will this include the scenario were one of the CUCMs  is down? Don´t see 
explicitly in the notes…

De: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
En nombre de NateCCIE
Enviado el: miércoles, 13 de enero de 2021 10:56
Para: Pawlowski, Adam mailto:aj...@buffalo.edu>>
CC: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Asunto: Re: [cisco-voip] MRA DR / Resilience

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<https://nam10.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwww.cisco.com%2Fc%2Fdam%2Fen%2Fus%2Ftd%2Fdocs%2Fvoice_ip_comm%2Fexpressway%2Frelease_note%2FCisco-Expressway-Release-Note-X12-7.pdf%23page16=04%7C01%7Cariel.roza%40la.logicalis.com%7C7102b260f7c543fc5d8c08d8bbe27944%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C637465928819010016%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C1000=2hhQwkNYTqiqc6wDDUwV%2B%2BZUcfpKc%2Bpg3otGhRX5ePw%3D=0>

This is new in x12.7
Sent from my iPhone

On Jan 13, 2021, at 6:10 AM, Pawlowski, Adam 
mailto:aj...@buffalo.edu>> 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

Re: [cisco-voip] MRA DR / Resilience

[Pawlowski, Adam]

Hi Nate, we’re still on X12.6.5 so I’ll have to scope this out.

It looks like, if I read that right, the Expressway will finally flag servers 
as inactive instead of … just not.

It’s unclear if this improves anything with Jabber’s behavior.

My customers have gifted my inbox with Jabber PRT logs this morning, and in 
reading through them, it looks like most of the issues are:


  *   Jabber trying to hit the CUC node that’s down for SSO auth, which results 
in a sign in failure
  *   Jabber trying to hit the UCM node that’s down for UDS, which results in a 
sign in failure

Both things would be resolved if the servers are marked inactive and not 
presented to the Jabber client, but the Jabber client also has to handle this 
better if it tries to reach to something it cannot, instead of just bombing 
out. That’s probably a pipe dream with Jabber at this point.

Thanks again,

Adam

From: NateCCIE 
Sent: Wednesday, January 13, 2021 8:56 AM
To: Pawlowski, Adam 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] MRA DR / Resilience

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 
mailto:aj...@buffalo.edu>> 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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[cisco-voip] MRA DR / Resilience

[Pawlowski, Adam]

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?

[Pawlowski, Adam]

One of the flavors of a wrapped Asterisk product I played with some years ago 
included this functionality, where you could key *1 or something while on a 
call, and it would begin a recording. It emailed you the wav after you were 
finished.

I don’t know if that can be accomplished in just the base Asterisk or not, and 
maintaining such a system isn’t as nice as something with a support contract, 
but a trunked in Asterisk box can provide some fun and utility.

Adam

From: cisco-voip  On Behalf Of Lelio 
Fulgenzi
Sent: Monday, January 11, 2021 4:00 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] List still active?

[Pawlowski, Adam]

Merry Christmas , and happy new year to all

From: cisco-voip  On Behalf Of Anthony 
Holloway
Sent: Friday, December 25, 2020 3:51 PM
To: Ryan Huff 
Cc: cisco-voip 
Subject: Re: [cisco-voip] List still active?

Chat services like Webex Teams and Discord have killed the list, IMO.

Also, Merry Christmas, all you VoIP Heads out there.

On Fri, Dec 25, 2020 at 1:38 PM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
Yes, it has declined in volume.

Sent from my iPhone

> On Dec 25, 2020, at 14:30, Bill Talley 
> mailto:btal...@gmail.com>> wrote:
>
> Thanks for the confirmation Ryan.  Are you also seeing a significant decline 
> in volume from the group?
>
> Hope all the usual (and even casual) participants are staying healthy and 
> employed.  Hope those aren’t reasons for the decline in forum usage.
>
> Sent from an iPhone mobile device with very tiny touchscreen input keys.  
> Please excude my typtos.
>
>> On Dec 25, 2020, at 1:28 PM, Ryan Huff 
>> mailto:ryanh...@outlook.com>> wrote:
>>
>> I still see you.
>>
>> Sent from my iPhone
>>
 On Dec 25, 2020, at 14:28, Bill Talley 
 mailto:btal...@gmail.com>> wrote:
>>>
>>> I stopped receive list emails.   Is the list dead or was I banned? 
>>>
>>> Sent from an iPhone mobile device with very tiny touchscreen input keys.  
>>> Please excude my typtos.
>>> ___
>>> cisco-voip mailing list
>>> cisco-voip@puck.nether.net
>>> https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fpuck.nether.net%2Fmailman%2Flistinfo%2Fcisco-voipdata=04%7C01%7C%7C862cd952c60f4cb8ebb608d8a90b8c18%7C84df9e7fe9f640afb435%7C1%7C0%7C637445214339013495%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C1000sdata=TOQxb7cMkrbb3akfFF2LdEwcWyveeqpl6PJ%2Bk9AMIlg%3Dreserved=0
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Re: [cisco-voip] Deprecated phones in CUCM 14

[Pawlowski, Adam]

Sure, you could fence them off if creating an isolated or appropriately gapped 
network is in your wheelhouse. Since the devices are usually used with pass 
through, they’re potentially exposed to some issues regardless of how well you 
try and secure the address space of the phones.

In my case I ran into issues with 79x1 series devices exposed to either some 
type of traffic or too high of a volume, which caused the phones to become 
unusable. It is not fixed and never will be as there is no longer support. If 
that were to happen to someone else, it becomes I think a problem to try and 
patch around phones where cabling isn’t present, replace devices rapidly, etc.

But, yes absolutely you could bolt a 7940 to the wall on a private network by 
itself with the PC Port off, only allowed to speak to the UCM and maybe relayed 
to gateways or other devices, and then it could be okay.

Best,

Adam

From: cisco-voip  On Behalf Of James 
Buchanan
Sent: Thursday, September 24, 2020 10:39 AM
To: James Andrewartha 
Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net) 

Subject: Re: [cisco-voip] Deprecated phones in CUCM 14

Hello,

I know the security argument is valid in some respects. But, if I have an 
insecure IoT device on the network, like a temperature control system, I don't 
have the option of replacing it. So, I ring fence it. I isolate it so that it 
can't be exploited, etc. Maybe I'm missing something, but isn't that what we 
can do to workaround the security issues?

Thanks,

James

On Thu, Sep 24, 2020 at 3:36 PM James Andrewartha 
mailto:jandrewar...@ccgs.wa.edu.au>> wrote:
We’re still running 10 year old 79x5s (despite their cold boot RAM failure 
issue) and won’t be replacing them until next year at the earliest, I think 
most organisations would want to sweat at least 10 years out of their handsets. 
I mean what features does an 8800 have over an old 7940 if all you want is 
dialtone? Yes security etc, but why pay for new handsets when you get nothing 
for it? Plus I bet COVID is making business reconsider upgrading handsets in 
empty offices.

--
James Andrewartha
Network & Projects Engineer
Christ Church Grammar School
Claremont, Western Australia
Ph. (08) 9442 1757
Mob. 0424 160 877

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Anthony Holloway
Sent: Thursday, 24 September 2020 10:30 PM
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] Deprecated phones in CUCM 14

This statement from the link is interesting to me:

"...opportunity to move to newer phone models and clients at a pace that is 
reasonable."


  *   8800 series was released 8 years ago
  *   7940's have been end of support since 5 years ago

On Thu, Sep 24, 2020 at 8:20 AM Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:
Yeah, there was some talk about this in the forums. Someone from Cisco said, 
“watch the page for some changes we think you’ll like”.

Wish they would update the “updated” date.

Lelio


From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of James Buchanan
Sent: Thursday, September 24, 2020 7:49 AM
To: voyp list, cisco-voip 
(cisco-voip@puck.nether.net) 
mailto:cisco-voip@puck.nether.net>>
Subject: [cisco-voip] Deprecated phones in CUCM 14

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

Hello,

So, Cisco changed their mind for now:

https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/trouble/14_0_1/fieldNotices/cucm_b_deprecated-phones-14.html

Thanks,

James
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Re: [cisco-voip] Implementing 802.1x for IP Phones - issue with UCM timeout

[Pawlowski, Adam]

I've not implemented this with the phones, but, I would check the phone log to 
see if it thinks a VLAN change is taking place, and the device is dropping off 
the network at a session renewal time. Depending on the volume of devices you 
may want session renewal times to be in the order of days, if ever, if it's set 
to 45 minutes or something.

Best,

Adam

From: cisco-voip  On Behalf Of Riley, Sean
Sent: Thursday, September 17, 2020 6:00 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Implementing 802.1x for IP Phones - issue with UCM timeout

We are working on implementing 802.1x with our 8851 ip phones.  We have 
installed the LSC cert and enabled 802.1x on a few phones for testing.  We are 
using Cisco ISE and the switch is configured for host mode multi domain.  
Everything seemed to be working fine, until we noticed the phones were 
resetting about every 48 minutes.  Looking at the logs on the phone it seems it 
is being reset due to a timeout with CUCM.

ReasonForOutOfService=10 followed by a ReasonForOutOfService=23

My Cisco ISE admin didn't see anything on that side that he thinks is causing 
the timeout, and we only seem to see this issue after registering the phone 
with 802.1x enabled.

Any thoughts before I start collecting traces and wireshark captures.

CUCM v 11.5(1) SU6
IP phones are on latest firmware 12.7 or 12.8

Thanks.

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

[Pawlowski, Adam]

While we haven’t run 11.5SU8 ourselves, we did have similar issues with SFTP 
and DRF that were related to cipher issues. I caused it myself at one point by 
running a machine in FIPS mode which … was broken but not certified so its not 
supported (?).  It was kind of hard to see to find it in the right logs or get 
TAC in to try and run a sftp session from shell and see the complaints.

Not sure if that helps or not.

Adam

From: cisco-voip  On Behalf Of Louis 
Koekemoer (MEA)
Sent: Friday, September 4, 2020 6:48 AM
To: cisco-voip 
Subject: [cisco-voip] SFTP on CUCM 11.5.su8


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:
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Re: [cisco-voip] [External] Re: Remote Phone Control

[Pawlowski, Adam]

That’s pretty neat.

Back before COVID when we had more people in those terrible open office spaces 
I wanted to set up a couple of RTP streams to “dial” into white noise, or ocean 
sounds or some such thing for a bit more privacy. Rather than having SURL 
buttons to do it that would be a neat way to go about it.

Adam

From: Anthony Holloway 
Sent: Friday, August 21, 2020 4:39 PM
To: Pawlowski, Adam 
Cc: Hunter Fuller ; Cisco VoIP Group 

Subject: Re: [cisco-voip] [External] Re: Remote Phone Control

Yeah, neat idea.  Speaking of unconventional ideas with joining RTP streams, 
when I worked at a larger company which held all employee conference calls, I 
hooked into the CURRI API for the toll free number for the audio bridge, and 
caused the call to be rejected but at the same time, I joined the calling phone 
to the RTP stream, because the production was also streaming the audio over the 
network enterprise wide.  Saved on trunks, and conferencing costs.

On Fri, Aug 21, 2020 at 3:36 PM Pawlowski, Adam 
mailto:aj...@buffalo.edu>> wrote:
Have had way too much fun with that in the past with various media files, the 
Play function, and yeah the RTP. VLC can be a source but so can a cheapo 7941. 
We used them for audio for some holiday parties years ago, just stuck some 
phones under the tables to boost the ones already in the room and viola.



From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Anthony Holloway
Sent: Friday, August 21, 2020 4:27 PM
To: Hunter Fuller mailto:hf0...@uah.edu>>
Cc: Cisco VoIP Group 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] [External] Re: Remote Phone Control

And you would never upload a rick roll audio clip to your CUCM as a mutlitcast 
audio source, and then abuse the join rtp stream function on your co-workers 
phones either.  No...no you wouldn't.  And neither have I.  ;)

On Fri, Aug 21, 2020 at 3:07 PM Hunter Fuller 
mailto:hf0...@uah.edu>> wrote:
Im pretty sure this just completely changed the way we provide remote
help in the COVID era. Since I can just add a customer's phone to my
controlled devices, help them fix/show me some problem remotely, and
then remove it.

--
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 Fri, Aug 21, 2020 at 3:04 PM Erick Bergquist 
mailto:erick...@gmail.com>> wrote:
>
> It’s a great tool.
>
>
> On Fri, Aug 21, 2020 at 9:13 AM Anthony Holloway 
> mailto:avholloway%2bcisco-v...@gmail.com>> 
> wrote:
>>
>> My add-on was approved to be in the add-on store: 
>> https://addons.mozilla.org/addon/cisco-phone-controller/
>>
>> On Fri, Aug 14, 2020, 11:38 AM Anthony Holloway 
>> mailto:avholloway%2bcisco-v...@gmail.com>> 
>> wrote:
>>>
>>> I published a phone control firefox add-on that I've been sitting on for 
>>> over a decade now: it's not super polished, because it's just for me and my 
>>> friends to use, but I thought, what the hell, make it available publicly. I 
>>> might even polish it up and list it in the add-on store one day. Until 
>>> then: https://github.com/avholloway/cisco-phone-controller
>>>
>>>
>>
>>
>> ___
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Re: [cisco-voip] [External] Re: Remote Phone Control

[Pawlowski, Adam]

Have had way too much fun with that in the past with various media files, the 
Play function, and yeah the RTP. VLC can be a source but so can a cheapo 7941. 
We used them for audio for some holiday parties years ago, just stuck some 
phones under the tables to boost the ones already in the room and viola.



From: cisco-voip  On Behalf Of Anthony 
Holloway
Sent: Friday, August 21, 2020 4:27 PM
To: Hunter Fuller 
Cc: Cisco VoIP Group 
Subject: Re: [cisco-voip] [External] Re: Remote Phone Control

And you would never upload a rick roll audio clip to your CUCM as a mutlitcast 
audio source, and then abuse the join rtp stream function on your co-workers 
phones either.  No...no you wouldn't.  And neither have I.  ;)

On Fri, Aug 21, 2020 at 3:07 PM Hunter Fuller 
mailto:hf0...@uah.edu>> wrote:
Im pretty sure this just completely changed the way we provide remote
help in the COVID era. Since I can just add a customer's phone to my
controlled devices, help them fix/show me some problem remotely, and
then remove it.

--
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 Fri, Aug 21, 2020 at 3:04 PM Erick Bergquist 
mailto:erick...@gmail.com>> wrote:
>
> It’s a great tool.
>
>
> On Fri, Aug 21, 2020 at 9:13 AM Anthony Holloway 
> mailto:avholloway%2bcisco-v...@gmail.com>> 
> wrote:
>>
>> My add-on was approved to be in the add-on store: 
>> https://addons.mozilla.org/addon/cisco-phone-controller/
>>
>> On Fri, Aug 14, 2020, 11:38 AM Anthony Holloway 
>> mailto:avholloway%2bcisco-v...@gmail.com>> 
>> wrote:
>>>
>>> I published a phone control firefox add-on that I've been sitting on for 
>>> over a decade now: it's not super polished, because it's just for me and my 
>>> friends to use, but I thought, what the hell, make it available publicly. I 
>>> might even polish it up and list it in the add-on store one day. Until 
>>> then: https://github.com/avholloway/cisco-phone-controller
>>>
>>>
>>
>>
>> ___
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>>
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Re: [cisco-voip] Jabber via MRA failover from one CUCM node to another

[Pawlowski, Adam]

They occasionally leak things through there but they’ve been pretty good about 
that lately.

I wanted to test and confirm the MRA failover between UCM nodes but we are 
rebuilding a part of our lab and MRA isn’t running right now.

Maybe later today.

I know for a fact that if you shut down the primary CM in the CMGroup the 
Jabber clients are assigned to, they won’t register. My guess is that this is 
not so much a Jabber limitation but one with the way the registration proxy 
works and how it configures itself for the client based on data it receives. 
The Expressway E doesn’t know about the state of the zones on the C back 
towards the UCM cluster.

But, like we’ve all figured out from trial and many, many errors, these things 
don’t always work in a way that makes sense, and then they’ve been developed 
into a hole that is hard to get out of.

That being said with Jabber’s development pretty much done, are they using 
JabberWerx with Webex Teams or are they spending more time on that code? The 
latter implies that this is why we don’t have multiline there yet, but who 
knows.

From: cisco-voip  On Behalf Of Lelio 
Fulgenzi
Sent: Thursday, July 30, 2020 10:22 PM
To: Anthony Holloway 
Cc: cisco-voip voyp list 
Subject: Re: [cisco-voip] Jabber via MRA failover from one CUCM node to another


What document is that?

Title?

Don’t share the confidential footer!

;)
Sent from my iPhone


On Jul 30, 2020, at 9:32 PM, Anthony Holloway 
mailto:avholloway+cisco-v...@gmail.com>> wrote:

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

Do you have access to this document?  It's limited to who can view it.

https://salesconnect.cisco.com/#/content-detail/8da1f8a8-1f29-4e56-bc09-eb43175a3bba

On Thu, Jul 30, 2020 at 7:20 PM Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>> 
wrote:
" This is a surprise to me that this fundamental feature is still not supported 
after many years."

Are you new to working with Cisco solutions?

On Wed, Jul 29, 2020 at 12:02 AM Gerence Guan 
mailto:cisco.g...@gmail.com>> wrote:
Hi All,

I found this old post from Cisco community.
https://community.cisco.com/t5/ip-telephony-and-phones/jabber-softphone-via-mra-does-not-fall-over-to-secondary-cucm/td-p/2807099

Is this still true for Jabber 12.8, MRA 12.5 and CUCM 12.5?

In the MRA 12.6 deployment guide it says
Cisco Jabber clients support IM and Presence Service failover over MRA. 
However, they don't support any other type of MRA-related redundancy or 
failover—including SIP, voicemail, and User Data Services (UDS). Clients use a 
single UDS server only.
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/expressway/config_guide/X12-6/exwy_b_mra-expressway-deployment-guide/exwy_b_mra-expressway-deployment-guide_chapter_01.html


In the Jabber 12.8 planning guide, it says High Availability(failover) of audio 
and video service is not supported
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/jabber/12_8/cjab_b_planning-guide-cisco-jabber-128/cjab_b_planning-guide-cisco-jabber-128_chapter_010.html

If anyone has a MRA lab, can you please help to test it buy just disconnect the 
first cucm node in the jabber's ccm group and verify whether jabber can 
register to the second node in the ccm group?

This is a surprise to me that this fundamental feature is still not supported 
after many years.

Best Regards
Guan

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Re: [cisco-voip] [External] Re: how to disable backup warning message on cucm

[Pawlowski, Adam]

Any sort of IOWait spike will cause processes to core , phones and resources to 
lose registration , etc . 



-Original Message-
From: cisco-voip  On Behalf Of Hunter Fuller
Sent: Wednesday, July 29, 2020 7:17 PM
To: Ryan Huff 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] [External] Re: how to disable backup warning message 
on cucm

If Veeam works the same way as NetBackUp, aka, takes a snapshot, then you are 
going to start hearing stutter or maybe even dropped calls during the snapshot. 
Worst case your pub/sub sync could become broken.
Don't do it!

--
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 Wed, Jul 29, 2020 at 2:41 AM Ryan Huff  wrote:
>
> No way to disable the alert message that I’m aware of. I think DRS is an 
> unavoidable assumption (and by extension m, the alert) in the modern versions.
>
> As you know, this isn’t a great strategy. It’s a little more than, 
> “not recommended”, it’s actually not supported by Cisco to backup this 
> way. Veeam has been known to cause CPU spikes, kernel panics.. etc in 
> CUCM (while powered on)... not a good strategy at all. DRS is the path 
> to reinforce ;)
>
> From my understanding, it’s a pretty simplistic check... just looking for a 
> backup device, and the the XML file for the backup set in the backup device’s 
> location.
>
> They might be able to run one manual DRS, and then just keep modifying the 
> dates in the XML for the backup set. Seems like something that could be 
> scripted fairly easily too.
>
> To me though, that’s a lot of work to intentionally do it the wrong way. It’s 
> been my experience that when customers invite the Devil to dinner, he usually 
> shows up.
>
> - Ryan
>
> On Jul 29, 2020, at 03:12, naresh rathore  wrote:
>
> 
> hi,
>
>
> One of our Customer running version cucm 12.5.1.12900-115 (upgraded from 10). 
> they had backup enabled, they decided to do veem backup (even though not 
> recommended by Cisco). they deleted backup device and schedule configuration 
> and also disabled DRF Master and DRF local and restarted tomcat but still we 
> see  message of 32 days without backup. is there a way to disable this 
> warning?
>
> 
>
>
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Re: [cisco-voip] uccx12.0, social miner (for webchat)

[Pawlowski, Adam]

I know the answer to #3 is no ,w hen we asked it's not anything they were 
planning to offer with socailminer, you had to send a URL to some external 
service.



From: cisco-voip  On Behalf Of naresh 
rathore
Sent: Sunday, July 26, 2020 11:17 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] uccx12.0, social miner (for webchat)

hi,


i have implemented social miner and using webchat. my customer have following 
requirements.


  1.  problem statement selected during chat initiation not displayed on agent 
window. is there a way for agent to see problem statement selected during chat 
initiation.
  2.  also when agent send any webpage url to chat user (chat initiator) he/she 
cant click and open the link. but when chat user send webpage url to agent, he 
can open url without any issue.
  3.  customer also wants to have an file attachment option, which he can use 
to send files to agent. is there option to activate attachment capability?
Regards

Naray
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Re: [cisco-voip] digitized voice quality

[Pawlowski, Adam]

That and what the regional bandwidth/codec lists are set to

Perhaps some calls are using g729 or something which would sound pretty gross 
on a PC headset in comparison to opus or even g711



From: cisco-voip  On Behalf Of Anthony 
Holloway
Sent: Monday, July 13, 2020 11:08 AM
To: Scott Voll 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] digitized voice quality

Who is hearing the poor audio, the jabber user?  the person the jabber user is 
speaking to?  is this jabber to jabber only?  is this over wlan or lan?

One idea for why audio sucks but audio+video doesn't is that your priority 
queue is dropping packets and when video is added both audio and video are af41 
and therefore circumvents the priority queue.

Can you confirm the dscp values on the voice stream for a voice only versus 
voice+video call?

On Mon, Jul 13, 2020 at 9:51 AM Scott Voll 
mailto:svoll.v...@gmail.com>> wrote:
All--

we are using Jabber 12.8 with CM 12.5 in a voice only mode.  we have users 
reporting digitized voice call quality.  has anyone else been dealing with this?

we have had multiple users say if they add Video to the call it makes it 
better.  Anyone else seeing that?  Any ideas why that would be better?

what else can we do to make this work better?  TAC has not been able to help 
with this after multiple weeks of engagement.

any thoughts?

TIA

Scott

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Re: [cisco-voip] Creating Jabber for non-existent phones

[Pawlowski, Adam]

> However, getting that good data on a brownfield that's 10 years old with a 
> lot garbage in it is painful to say the least.

This is what I’ve been doing for the last several years and I’m still not done. 
 It’s much better than it was, but there is still some “just get it out there” 
mentality. We know how easy it is to take things away or change them once 
they’re set. This is where tooling and proficiency really help, I’ve been able 
to use various data sources and APIs to clean things up, where I can now make 
presumptions for things, like changing button layouts or provisioning. The 
payoff was well worth the pain. If/when we end up shifting off of premise UCM 
to some other solution or vendor, not having the ducks in a row would be brutal.
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Re: [cisco-voip] Creating Jabber for non-existent phones

[Pawlowski, Adam]

B would be the way to go yes if you mean provision people for Jabber service 
for telephony but they do not have a hardware phone. If they don't need phone 
service at all and just IM then there are other ways.

C is what we do here. I have line templates for 
campus/local/national/international for each of our gateway locations, and a 
couple for internal contact centers or similar that we grind through enough 
that it is worth putting them in here. Anything more complicated does not 
follow our SOP and gets pushed to someone with more hours on the system to work 
through. This is combined with User Profile, which by default it setup for a 1 
line 5 button 8800 series device and is rarely changed. I built those for 1 and 
2 line 8800 and 7900 series (largely just common phone profile and button 
template between them) which can be used when building something new or when 
moving devices. In practice it doesn't usually get used as most devices are 
either bog standard 1 line, or some mess of buttons and need intervention. 

Adam





> -Original Message-
> From: cisco-voip  On Behalf Of Lelio
> Fulgenzi
> Sent: Wednesday, July 1, 2020 2:06 PM
> To: cisco-voip voyp list 
> Subject: [cisco-voip] Creating Jabber for non-existent phones
> 
> 
> Hello all. Looking for feedback and opinions and caveats.
> 
> Right now, we’re deploying Jabber only to those with phones/DNs. But, we need
> to start deploying Jabber for those individuals without phones/DNs.
> 
> Our SOPs include using Quick Add feature. (Thanks a million time Brian Meade
> for the pointer).
> 
> My choices so far, to address Jabber for new those without phones:
> 
> (a) Create a fake hardware phone first. This has many benefits, namely, all
> SOPs remain the same. Hardware phone would be deleted afterwards.
> 
> (b) Use Directory Number admin page to create/update a DN first, then use
> Quick Add page to assign DN to user accordingly and then click manage devices
> and follow remaining SOP steps.
> 
> (c) create line templates and use those when creating new extensions under
> quick add. The issue with this is we have so many combinations, I’d need a lot
> of templates.
> 
> I’m leaning towards (b), since it gives me the best of both worlds.
> 
> Thoughts?
> 
> Lelio
> 
> Sent from my iPhone
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Re: [cisco-voip] [External] Re: Cellphone with pararell calls

[Pawlowski, Adam]

Yes, that is what I got from it with Hybrid, but, my guess is this is not an 
iPhone we are talking about?

From: Hunter Fuller 
Sent: Tuesday, June 16, 2020 9:09 PM
To: Pawlowski, Adam 
Cc: ROZA, Ariel ; cisco-voip 
(cisco-voip@puck.nether.net) 
Subject: Re: [External] Re: [cisco-voip] Cellphone with pararell calls

Honestly it may be an issue with the app. If I recreate this scenario with 
Jabber, I get this prompt.

[cid:image001.jpg@01D6449A.9AA1DDC0]

It’s clearly an iPhone prompt. And if I select “hold and accept” it does work, 
and doesn’t leak audio.

There’s nothing I could select that would leak audio, as far as I can tell.

Does Webex Teams not use the system call prompts? I’m not really familiar.

On Tue, Jun 16, 2020 at 19:58 Pawlowski, Adam 
mailto:aj...@buffalo.edu>> wrote:
Sure, it does not sound like an ideal situation to be in, but, the issue I 
believe is with the platform more than the application.

Hybrid Calling is done for users anyways, and no new customers can activate it. 
If you can configure additional or existing Expressway for MRA, I would hope it 
would have a better experience.


From: ROZA, Ariel 
mailto:ariel.r...@la.logicalis.com>>
Sent: Tuesday, June 16, 2020 8:45 PM
To: Pawlowski, Adam mailto:aj...@buffalo.edu>>; cisco-voip 
(cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Subject: RE: Cellphone with pararell calls

The downside is that is very prone to human error, and it may end. up in calls 
being evasdropped. I can´t think of any webex user fond of something like that ☹

De: Pawlowski, Adam mailto:aj...@buffalo.edu>>
Enviado el: martes, 16 de junio de 2020 21:41
Para: ROZA, Ariel 
mailto:ariel.r...@la.logicalis.com>>; cisco-voip 
(cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Asunto: RE: Cellphone with pararell calls

This sounds like an interaction with the type of phone or phone OS, and not 
related to Webex Teams.

I do not have this issue with hybrid calling on Teams on iOS, it appears to 
hook the native dialer, at least for now, which will not allow this scenario to 
occur.

Whoever is operating the cell will have to … well end the Teams call since 
there is no hold in hybrid, or not answer the cellular call ?



From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of ROZA, Ariel
Sent: Tuesday, June 16, 2020 5:59 PM
To: cisco-voip (cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Subject: [cisco-voip] Cellphone with pararell calls

I have this situation with Webex Teams running on a cellphone:
A user has his cellular phone line and Webex Teams with Hybrid calling enabled.
He receives a call at his cellular phone line.
Then he gets a second call via Webex Teams. When he answers the second call, 
the first one keeps running in the background, still listening to the user 
having the second conversation.
Is there a way to better control this scenario? The first call should be ended 
or put on hold, but as those are two different apps with two unrelated audio 
streams, I don´t know if there is a proper way to handle them

Does anyone have any suggestions?

Thanks,

Ariel.

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--
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+1 256 824 5331

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The University of Alabama in Huntsville
Network Engineering
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Re: [cisco-voip] Cellphone with pararell calls

[Pawlowski, Adam]

Sure, it does not sound like an ideal situation to be in, but, the issue I 
believe is with the platform more than the application.

Hybrid Calling is done for users anyways, and no new customers can activate it. 
If you can configure additional or existing Expressway for MRA, I would hope it 
would have a better experience.


From: ROZA, Ariel 
Sent: Tuesday, June 16, 2020 8:45 PM
To: Pawlowski, Adam ; cisco-voip 
(cisco-voip@puck.nether.net) 
Subject: RE: Cellphone with pararell calls

The downside is that is very prone to human error, and it may end. up in calls 
being evasdropped. I can´t think of any webex user fond of something like that ☹

De: Pawlowski, Adam mailto:aj...@buffalo.edu>>
Enviado el: martes, 16 de junio de 2020 21:41
Para: ROZA, Ariel 
mailto:ariel.r...@la.logicalis.com>>; cisco-voip 
(cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Asunto: RE: Cellphone with pararell calls

This sounds like an interaction with the type of phone or phone OS, and not 
related to Webex Teams.

I do not have this issue with hybrid calling on Teams on iOS, it appears to 
hook the native dialer, at least for now, which will not allow this scenario to 
occur.

Whoever is operating the cell will have to … well end the Teams call since 
there is no hold in hybrid, or not answer the cellular call ?



From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of ROZA, Ariel
Sent: Tuesday, June 16, 2020 5:59 PM
To: cisco-voip (cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Subject: [cisco-voip] Cellphone with pararell calls

I have this situation with Webex Teams running on a cellphone:
A user has his cellular phone line and Webex Teams with Hybrid calling enabled.
He receives a call at his cellular phone line.
Then he gets a second call via Webex Teams. When he answers the second call, 
the first one keeps running in the background, still listening to the user 
having the second conversation.
Is there a way to better control this scenario? The first call should be ended 
or put on hold, but as those are two different apps with two unrelated audio 
streams, I don´t know if there is a proper way to handle them

Does anyone have any suggestions?

Thanks,

Ariel.

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Re: [cisco-voip] Consolidating access to Cisco CUCM APIs via Service Mesh

[Pawlowski, Adam]

This is more or less what I had been doing internally at a very primitive 
level, and its been very valuable but quickly consuming of a ton of time to try 
and account for the items and interactions between them.  I have to more or 
less abandon Ruby at this point given that gems are not being updated and 
things are starting to be a problem to maintain for me, oh well.

The Ruby gem I coked up abstracts a user object with details pulled from LDAP, 
UCM, Unity Connection, and lets me interface with local resources or write 
simple reports. It is what I used to write a basic CSV Jabber import (stripped 
way down to just basically AXL) tool, though no one took me up on it for import 
over BAT. I think they’re nuts to use that.

I am still unclear on what a service mesh is, but, my overall vision was to 
have some sort of abstracted layer that monitors some level of object history, 
but allowed me to tether business rules and systems to actions in the data 
systems. It became pretty quickly obvious a job engine, call brokering and 
buffering, intermediate database with reconciliation and clean up jobs, etc 
would be needed on top of trying to define the way the objects actually 
interact with each other. I don’t have the time.

I did not look into AutomationFX, and I’ll earn some ire for this, but, we have 
used both PhoneView and MigrationFX. PhoneView I have largely gotten past 
internally with my own tooling but it is great for the rest of the team looking 
to do a quick data dump or get some data for reporting and with it’s built in 
data cache it’s pretty good. MigrationFX seems to be built on top of 
AutomatioFX, but, while better than going to each phone and configuring a new 
one, has had a clunky interface that doesn’t appear to be possible to secure 
when using older phones, at least per my coworkers, and I don’t want to invest 
my efforts into that.

I’d love to see something like this, but I wonder how complicated it would have 
to be to be ubiquitously useful.

Adam

From: cisco-voip  On Behalf Of Anthony 
Holloway
Sent: Tuesday, June 16, 2020 8:34 PM
To: Pete Brown 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Consolidating access to Cisco CUCM APIs via Service 
Mesh

Yeahstill lost.  Possibly even more than I was before.  I'll just see 
myself out now.

On Tue, Jun 16, 2020 at 6:44 PM Pete Brown 
mailto:j...@chykn.com>> wrote:
Thanks for mentioning AutomationFX.  That’s exactly the sort of API 
consolidation I was thinking of.  Should’ve guessed the guys at UnifiedFX had 
already done something along these lines!  I’ll probably still build a CUCM 
mesh agent just to demo the marrying of access to objects & their associated 
data streams.  But it won’t be near as complete as AutomationFX.

Wouldn’t say ignorant at all.  Service meshes (Istio, etc) in their current 
form have only been around a few years.  Even most developers I talk to haven’t 
really touched them beyond POCs.  Mainly due to the complexity since they all 
require the use of Kubernetes.  The concept has never really been applied to 
infrastructure sources before, especially in a vendor agnostic way.  In 
infrastructure we’re dealing with a federation of loosely coupled data sets 
instead of something that a single architecture group came up with.

To answer the question of “why” regarding the mesh I’m working on, it’s to 
eliminate a bunch of the common pitfalls in traditional integrations.  Instead 
of finding and calling each source directly using different client libraries, 
the mesh provides a single method of executing RPC & pub/sub operations against 
backend services.  You can even navigate the resources like a directory 
structure.

In this mesh, the backend services register themselves and declare their 
functions, object schemas, etc.  That’s why I say it’s sort of a hybrid between 
a service mesh and a data mesh; you can call service functions or search on 
object class attributes regardless of source.  Clients who need to access 
sources make calls to the Brokers.  Very roughly analogous to a “meshified” 
version of SNMP.

Here’s a before and after of what an Ansible script might look like calling 
difference sources.


[cid:image001.png@01D6441F.07C40490]


Sent from Mail for Windows 10

From: Anthony Holloway
Sent: Tuesday, June 16, 2020 3:02 PM
To: Pete Brown
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Consolidating access to Cisco CUCM APIs via Service 
Mesh

You lost me there, as I'm too ignorant to understand what an API mesh is, but 
this sounds very familiar to what UnifiedFX is/was doing with AutomationFX, is 
it not?  Or am I again showing my ignorance?

On Tue, Jun 16, 2020 at 12:32 PM Pete Brown 
mailto:j...@chykn.com>> wrote:
TLDR – Developing a hybrid service/data mesh for interacting with 
infrastructure services.  Thinking about 

Re: [cisco-voip] Cellphone with pararell calls

[Pawlowski, Adam]

This sounds like an interaction with the type of phone or phone OS, and not 
related to Webex Teams.

I do not have this issue with hybrid calling on Teams on iOS, it appears to 
hook the native dialer, at least for now, which will not allow this scenario to 
occur.

Whoever is operating the cell will have to ... well end the Teams call since 
there is no hold in hybrid, or not answer the cellular call ?



From: cisco-voip  On Behalf Of ROZA, Ariel
Sent: Tuesday, June 16, 2020 5:59 PM
To: cisco-voip (cisco-voip@puck.nether.net) 
Subject: [cisco-voip] Cellphone with pararell calls

I have this situation with Webex Teams running on a cellphone:
A user has his cellular phone line and Webex Teams with Hybrid calling enabled.
He receives a call at his cellular phone line.
Then he gets a second call via Webex Teams. When he answers the second call, 
the first one keeps running in the background, still listening to the user 
having the second conversation.
Is there a way to better control this scenario? The first call should be ended 
or put on hold, but as those are two different apps with two unrelated audio 
streams, I don´t know if there is a proper way to handle them

Does anyone have any suggestions?

Thanks,

Ariel.

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Re: [cisco-voip] 7900 Factory Reset w/out connecting to the network

[Pawlowski, Adam]

I had assumed the working state of the phone when finished wasn’t important 
here given scrapping them out.

Otherwise the second one will leave you with a bad time if you’re not setup to 
repair the phone when you do that. The first one shouldn’t but its been a long 
time since I’ve been near one of these to know if it retains any last good 
configuration

From: Anthony Holloway 
Sent: Monday, June 8, 2020 5:10 PM
To: Pawlowski, Adam 
Cc: Riley, Sean ; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] 7900 Factory Reset w/out connecting to the network

Adam,  I've always been under the impression that either of these options 
require DHCP Option 150 with a working TFTP and files to work properly.  Are 
you saying they don't?  I've never tested it though.


On Mon, Jun 8, 2020 at 3:26 PM Pawlowski, Adam 
mailto:aj...@buffalo.edu>> wrote:
Sean,

The two codes for it, are to hold the pound key when powering it on.

When the line keys flash, key go of pound and enter: 123456789*0# , which 
should erase the configuration.

If you observe that it is still showing internal information, try 3491672850*# 
, which in my experience rips it down to a bootstrap, but, I’ve never loaded 
the term default on it to see if there’s anything left by the time it recovers.

Best,

Adam

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Riley, Sean
Sent: Monday, June 8, 2020 3:59 PM
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: [cisco-voip] 7900 Factory Reset w/out connecting to the network

We are finally refreshing our 7970’s.  We wanted to factory reset the old 
phones before sending out for scrap.  Is there a way to factory reset without 
having to connect to the network?

Thanks.

Sean.
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Re: [cisco-voip] 7900 Factory Reset w/out connecting to the network

[Pawlowski, Adam]

Sean,

The two codes for it, are to hold the pound key when powering it on.

When the line keys flash, key go of pound and enter: 123456789*0# , which 
should erase the configuration.

If you observe that it is still showing internal information, try 3491672850*# 
, which in my experience rips it down to a bootstrap, but, I've never loaded 
the term default on it to see if there's anything left by the time it recovers.

Best,

Adam

From: cisco-voip  On Behalf Of Riley, Sean
Sent: Monday, June 8, 2020 3:59 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] 7900 Factory Reset w/out connecting to the network

We are finally refreshing our 7970's.  We wanted to factory reset the old 
phones before sending out for scrap.  Is there a way to factory reset without 
having to connect to the network?

Thanks.

Sean.
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Re: [cisco-voip] Jabber and shared (primary) lines

[Pawlowski, Adam]

I have been pushing my shop to reduce the number of shared lines that are in 
use in general, and have been banging on that drum for a while. Having your own 
line and voicemail meshes better with other services, and keeps your data 
separate as far as records, messages, etc as privacy continues to be a thing 
that needs shoring up. That being said while a lot of things are now "person 
centric" this doesn't account for business processes.

We have deployed Jabber as the primary extension on BOT/TAB/TCT devices for 
customers who want to operate mobile and primarily cover a shared line, given 
this situation. Desktop we stack shared lines on as additional - I have a "bot" 
I can IM that just tosses them on there upon request.  I have not tested it 
with Teams / UCM calling, but, I don't see why it wouldn't work. 

My experiences have been:

- The call limits for a line on BOT/TCT/TAB are lower than CSF, which is lower 
than a desk phone. If you used a shared line to cover a busy line instead of an 
ACD, then this is where you could feel this pain. We all know what happens in 
these situations - "my phone didn't ring", "I couldn't transfer", "People got a 
busy signal", etc. Wrong tool for the job in those situations.

We've been trying to deploy hunt groups, but, in a system built to keep the 
company "flat", this has been challenging without large administrative 
overhead. You'd end up boxing people in to partitions where they are 
transformed calling out on/off premise to meet the requirement of being able to 
call from the shared line, and that can be a lot of work and doesn't scale well.

- If you're SSO enabled, setting the voicemail to "No Credentials" doesn't work 
anymore. So, regardless of what you do, you do not seem to be able to have 
these customers sign in to  voicemail via Jabber for the shared line, which 
creates its own set of problems on how to cover voicemail. Jabber still does 
not work with dispatch messaging for whatever reason, so you're left with other 
not great options.

- Jabber multi-line I'm beginning to suspect is less tightly integrated into 
Jabber's code as it would seem. Since we've deployed the solution, we've run 
into issues with partial registration on a handful of customers. The right set 
of interruptions to connectivity can render lines beyond line one unregistered, 
and Jabber doesn't seem to have any awareness of this. It just monitors the 
state of the primary line and considers itself happy. In the meanwhile, calls 
to and from fail. From going blind looking at Jabber's PRT logs, it's become 
clear that there's a large soup of things in there, and managing it must be an 
absolute nightmare. Given other development priorities I don't know if this 
will ever get fixed unless they move on to something else. 

I don't understand your question about deskphones with the lines in different 
order. That's an annoyance when trying to manage or maintain the sets, yes. 
We've moved to deploying the user's/station DN as primary, only copying this if 
the customer has multiple seats. We will not deploy a set any more with shared 
lines as line one all around an office. There are vague prohibitions on this in 
the UCM documentation, but it also makes identifying the set awful.

Best,

Adam

-Original Message-
From: cisco-voip  On Behalf Of Lelio 
Fulgenzi
Sent: Thursday, June 4, 2020 11:41 AM
To: cisco-voip voyp list 
Subject: [cisco-voip] Jabber and shared (primary) lines


What are people’s experiences with deploying Jabber with a shared line as the 
primary extension?

So, it would be two sets of four devices: botuser1, botuser2, csfuser1, 
csfuser2, etc. 

But they would all have the same DN. 

I doubt that Teams unified client in either CUCM or webex calling mode will 
support this, but still interesting in hearing stories. 

Also, how about associating a hardware phone that has the DNs in a different 
order? 

Go!

Sent from my iPhone
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Re: [cisco-voip] vCUBE Experiences

[Pawlowski, Adam]

Lelio,

Well – maybe. They rescinded video conferencing (and transcoding?) using the 
DSPs at some point. Audio transcoding is done in software all day – but the CPU 
of the ISR G2 platform at least is not its strong point, it is quickly snowed 
in by enough feature processing, records processing, debugging, etc. In a 
virtual environment there should be a stack of CPU available to do audio 
transcoding and get around that, but, then you don’t sell hardware.

Also yeah, 2.5 years thanks. I’ve lost track of time with all going on.

Best,

Adam



From: Lelio Fulgenzi 
Sent: Thursday, June 4, 2020 12:45 PM
To: Pawlowski, Adam 
Cc: Anthony Holloway ; UC Penguin 
; Cisco VoIP Group 
Subject: Re: [cisco-voip] vCUBE Experiences


I’m guessing DSPs fall into the custom silicon branch of things. But I hear so 
much about software being able to use GPUs to do magic.

I could see the requirement of vDSP being a robust GPU installed on the chassis.

P S. 3900 eol dec 31 2022 so 2 1/2 years.
Sent from my iPhone


On Jun 4, 2020, at 12:38 PM, Pawlowski, Adam 
mailto:aj...@buffalo.edu>> wrote:

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<mailto:ith...@uoguelph.ca>

Oh and to keep this on vCUBE – the ISR G2 boxes that we run will be done for 
support in another … year and a half or so ?

Unfortunately there’s no “vPRI” that will help with DSP, until we change 
transport.

From: Pawlowski, Adam
Sent: Thursday, June 4, 2020 10:55 AM
To: 'Anthony Holloway' 
mailto:avholloway+cisco-v...@gmail.com>>; UC 
Penguin mailto:gen...@ucpenguin.com>>
Cc: Cisco VoIP Group 
mailto:cisco-voip@puck.nether.net>>
Subject: RE: [cisco-voip] vCUBE Experiences

I’m not immersed in the industry by any means to say, but yes the cloud thing 
seems to be a moving target of opportunity to gain actual ROI or benefit from 
it, amidst changing budgets, feature demands, and licensing models.

I consider where my shop is to be on a bit of a lag on some trends which has 
its ups and downs. In this case we’re being asked to look at “cloud” now, but, 
narrowly focused to telephony. On one hand we may get shoved there by vendors, 
and cloud telephony is everywhere at this point, but, on the other hand the ROI 
ship has sort of sailed a bit.

When this appeared that you can run in AWS I … just don’t get it. I bet it has 
its applications if you don’t have space to run premise servers, if the whole 
city/state/country/world is your “WAN” as far as your business goes, sure. Or 
if you’re one of the shops still running old PBX and finally looking to bite 
the bullet, but, hasn’t that well run dry at this point?

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Anthony Holloway
Sent: Thursday, June 4, 2020 10:34 AM
To: UC Penguin mailto:gen...@ucpenguin.com>>
Cc: Cisco VoIP Group 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] vCUBE Experiences

I was just hearing from a Cisco person, who was saying something like 
"Everybody said they had to have it, but when we finally had an offer, there 
were literally ZERO people who did it."

And here I was thinking that my customer base was just cloud adverse and 
everyone else was jumping on the AWS band wagon.  Guess not.

On Thu, Jun 4, 2020 at 9:28 AM UC Penguin 
mailto:gen...@ucpenguin.com>> wrote:
Shared resources like AWS on the surface seem like a great idea for lab stuff. 
Looks like a great solution for on demand scaling etc though.

It just doesn’t seem to that useful for UC purposes and even if it were it 
would still be cheaper to buy one server and run it all on one box.

It’s interesting to watch management push for cloud everything and then slowly 
back away when they see the increased cost.



On Jun 4, 2020, at 09:11, Tim Smith 
mailto:tim.sm...@enject.com.au>> wrote:

Back to CUCM on prem in lab via VPN.

The CUCM AWS deployment is out of reach for lab purposes.
That one is basically CUCM on VMWare in AWS (which is like the dedicated 
resources) - it's not AWS AMI format.

That said, I've got a great provider here in Australia that does VMWare based 
cloud (NSX).
That would be good for lab.

Cheers,

Tim



From: Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>>
Sent: Thursday, 4 June 2020 11:20 AM
To: Tim Smith mailto:tim.sm...@enject.com.au>>
Cc: Cisco VoIP Group 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] vCUBE Experiences



EXTERNAL SENDER WARNING. This message was sent from outside your organisation. 
Please do not click links or open attachments unless you recognise the source 
of this email and know the content is safe.

Was that trunk to Twilio for CME?  If not, what was on the backside of your 
gateway?  CUC

Re: [cisco-voip] Resolving Sectigo root expiration affecting MRA

[Pawlowski, Adam]

This is the boat we were in as well, and I’ve learned some lessons here.

The bug that I posted about for Jabber mobile devices got me – since we’re MRA 
only I thought I broke it again and it took a while to figure out why. The bugs 
in Expressway  On Behalf Of Derek Andrew
Sent: Wednesday, June 3, 2020 10:20 AM
To: Anthony Holloway 
Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net) 

Subject: Re: [cisco-voip] Resolving Sectigo root expiration affecting MRA

If you had previously installed the certs on CUCM CUP CUC and CER as we did, 
they would also have expired.

On Wed, Jun 3, 2020 at 7:34 AM Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>> 
wrote:
CAUTION: This email originated from outside of the University of Saskatchewan. 
Do not click links or open attachments unless you recognize the sender and know 
the content is safe. If in doubt, please forward suspicious emails to 
phish...@usask.ca

Hunter,

I might be exposing a gap in my knowledge here, but why did you need these 
certs on CUCM?

Cisco has now published a troubleshooting guide for this issue, and the article 
does not mention modifying CUCM cert store.

https://www.cisco.com/c/en/us/support/docs/unified-communications/expressway/215561-troubleshooting-expressway-mra-login-and.html

On Sat, May 30, 2020 at 7:02 PM Hunter Fuller 
mailto:hf0...@uah.edu>> wrote:
All,

If you use certs whose trust is derived from the Sectigo root that expired 
today, and your MRA isn’t working, I’ll try to save you a call to TAC.

Do all of these things:

 - Load the new intermediates and root into callmanager-trust and tomcat-trust 
on all your UCMs
 - restart tomcat, tftp, and callmanager on those boxes
 - load the new intermediates and root into the CA trust store on all 
expressways
 - reboot the Expressway-Es

If you need more detail or help, let me know, we just got off the phone with 
TAC. Hope it helps.

--

--
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
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--
Copyright 2020 Derek Andrew (excluding quotations)

+1 306 966 4808
Communication and Network Services
Information and Communications Technology
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] call forward is not working in this scenario

[Pawlowski, Adam]

Just to throw at this since I like to do so, that a hunt will not forward is 
not obvious when you’re looking at the configuration at its face, but, when you 
think about how that’s supposed to work with what happens to a call when you 
forward it, it makes more sense. SNR can be configured for 0 second wait and it 
more or less immediately rings out but there is still a bit of a delay.

We use this as a way to “cover” critical hunts such as safety ringdowns or 
things that need to operate when the system is up, but, the phones are down. A 
blind device is a member with SNR enabled, which will begin ringing a backup 
device after a period of time, sort of like a CFUR but this covers the “whoops 
building is on fire” scenario as well.

Adam

From: cisco-voip  On Behalf Of Anthony 
Holloway
Sent: Tuesday, June 2, 2020 11:52 AM
To: Arun Kumar 
Cc: Cisco VoIP Group 
Subject: Re: [cisco-voip] call forward is not working in this scenario

Ok, well, if your base configuration is hunt groups, and you want mobility, the 
answer is likely SNR.  Any reason you wanted to stay away from SNR based 
solutions?

On Tue, Jun 2, 2020 at 10:50 AM Arun Kumar 
mailto:arunraodhu...@gmail.com>> wrote:
agents are not at the office because of COVID situation and they want to attend 
the calls, hence they requested to forward the calls to their mobile phones

On Tue, Jun 2, 2020 at 9:14 PM Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>> 
wrote:
Nope.  The forward no coverage is for the following scenario only:

  1.  Caller calls phone DN directly
  2.  DN is set to forward to a Hunt Pilot
  3.  Hunt Group (HP+HL+LG) cannot cover to a member for some reason (e.g., RNA)
  4.  Call is now forwarded to the DN's Forward No Coverage destination
Seems like a strange scenario, one in which I have never needed nor been asked 
to configure.  I wonder who the hell uses that and why?

On Tue, Jun 2, 2020 at 10:22 AM Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:
Could you not use “forward no coverage” for this ?



From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Anthony Holloway
Sent: Tuesday, June 2, 2020 10:52 AM
To: Arun Kumar mailto:arunraodhu...@gmail.com>>
Cc: Cisco VoIP Group 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] call forward is not working in this scenario

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

Call forwarding is ignored for members of hunt groups.

If you actually configure SNR (like with RDP/RD) then that does work.  In fact, 
you don't even need a Cisco phone in the membership of the LG to ring a cell 
phone.  Once you build the DN on the RDP you can add the DN to the LG.

On Tue, Jun 2, 2020 at 7:27 AM Arun Kumar 
mailto:arunraodhu...@gmail.com>> wrote:
Hi

can anybody please help to fix this call flow issue

the call flow is like this

Hunt pilot number --> linegroup --> algorithm set as broadcast and LG - Phone 
ext() set CFA To Mobile number(X). Please let me know is there any 
alternate for apart from SNR ?
--
Thanks,
Arun
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Thanks,
Arun
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Re: [cisco-voip] Migration from CUCM/UCXN/UCCX 8.5 to 12.5

[Pawlowski, Adam]

Sure I can fire this up


  1.  part_ , like part_local, part_ld, part_ld_privacy , etc.
  2.  FQDN, but, make sure your DNS/NTP/etc works with resiliency.
  3.  Depends on if you’re distributed, using hardware conf, transcoder cause 
you have some people on some sort of twizzler based connection using g729, etc
  4.  Yes, unless you have something on the other side that can’t handle these 
requests coming from the whole group, or again a distributed system.
  5.  SIP. MGCP is nice in a set it and forget it way, but if you want to use 
the gateway to do anything else like custom intercepts, redirection, 
hairpinning, it won’t help you. There are some features that don’t work when 
you go to SIP but whatever.
  6.  Why would you give anything an upper case hostname

From: cisco-voip  On Behalf Of James B
Sent: Thursday, May 28, 2020 2:28 PM
To: Anthony Holloway ; Matthew Loraditch 

Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Migration from CUCM/UCXN/UCCX 8.5 to 12.5

I was thinking of the community configuration approach Anthony suggested and 
was thinking of the debates we’d have if we did that:


  1.  Do you use “_PT”, “PT_”, or just the site name? Same for “CSS”, “LOC”, 
and the ever-debated “RGN” or “REG”?.
  2.  FQDN or IP addresses?
  3.  Do all the media resources go into a single MRG or not?
  4.  Do we click “Run on all Nodes” for route lists and trunks or not?
  5.  MGCP, SIP, or H323 (if using PRIs)?
  6.  Can UCCX have upper-case hostnames or not?

The debates would take us so long, version 14.0 would be out, and then we’d 
have to debate about whether a “.0” versoin is stable or not or should we wait 
for “.5”? Still, could be fun!



From: Anthony Holloway
Sent: 28 May 2020 19:15
To: Matthew Loraditch
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Migration from CUCM/UCXN/UCCX 8.5 to 12.5

Keep in mind that PCD network migrations, while awesome for CUCM, do not work 
for other products.

Typically with a project like this, you'll likely have a different approach for 
each app, and not a one size fits all solution.

With the app upgrades, you will also have to change OVA sizes (or want to in 
some cases), and at that point, it might be better to install fresh, and use 
tools like COBRAS, BAT, AXL, ADMIN API, stare & compare, etc. to get data 
exported/imported from old to new.

Or like Kent said, tell yourself it's a toshiba system, and treat it like a 
greenfield.

On Thu, May 28, 2020 at 11:26 AM Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>> 
wrote:
Here’s a fun one. We have taken over support of these ancient servers hosted on 
Esxi 4.1 on UCS-C200-M2s!

Exact Versions are:
8.5 SU2 for CUCM/UCXN
8.5 FCS for UCCX

Each  is a pair of servers.

Have new M5s and flex licensing… need to get to 12.5..  8.5 docs are dead for 
CUCM/UCXN and 8 and 9 docs are dead for UCCX. ISOs I may need are not available 
publicly.

Also fun wrinkle the new host are across the WAN and for many logistical 
reasons are staying there. The migration to the new hosts will have to be via 
DRS or maybe PCD somehow? Not sure if the bandwidth available to get data 
across will be fast enough to finish in the allotted time period. My plan was a 
change freeze window and copy/restore the backups and then activate the new 
servers and move the subnet to the new location.

As best I can tell I need to get UCCX to SU4 of 8.5.1 then I can go to 10.6 SU3 
and then to 12x (with the fun of two CAD upgrades and then a migration to 
Finesse!)

For CUCM/UCXN I need to go to 8.6 anything and then I can go to 11.5 and then 
to 12x

I think my plan is to do the upgrades to the interim versions on the old hosts 
then migrate to the new and then finish the upgrades. The old hosts will need 
ESXi upgrades to an interim version.

Anyone have thoughts on what they would do here? This is partially depending on 
TAC being able to provide me the ISOs I will need, but I presume there is an 
archive.

In theory some sort of PCD migration is also an option, but I’ve never done one 
and not sure how it could handle the subnet situation that will have to exist.

Welcome to my fun life!















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[cisco-voip] Jabber Bulk Provisioning and CSCur73925

[Pawlowski, Adam]

Hey all,

I ran into this one late last night so I wanted to share.

https://bst.cloudapps.cisco.com/bugsearch/bug/CSCur73925

This bug says : " All jabber mobile devices(BOT/TCT) are created statically at 
the time of system start up, so they are counted toward Max Registered Device 
limit. Same as gateway configuration.

This is confusing to customers because it does not line up with what they see 
as registered devices. "

Notable here is "System Startup". I changed cert chains for the AddTrust root 
expiry and restarted the CM service which triggered this. CM nodes with 5000 
max registrations suddenly had 0 devices registered per RTMT, no alarm, and 
nothing would register. Jabber log shows a 503 Service Unavailable to the 
REGISTER with:

Server: Cisco-CUCM11.5
Retry-After: 35
Date: Wed, 27 May 2020 02:50:10 GMT
Warning: 399 sub1234 "Max Configured Devices Registered"

I had to increase the max registrations to get anything to register and come on 
board. I don't know if this is in the documentation, and the bug shows a couple 
dozen cases so I'm not the first to hit this.

The trick is, I bulk provision via AXL for my users, and had recently done this 
to get BOT/TCT up for COVID-19 prep. I wedged 5053 mobile devices in each pool 
that went to these subs which was just over the limit, but, nothing happened 
until CM was restarted. If the server had restarted the same thing would happen 
I would imagine. Since the AXL insert doesn't trigger this, and I believe quick 
user/phone add also uses AXL, there's a chance you could have this waiting to 
bite you as well.

Happy . Wednesday?

Regards

Adam Pawlowski
SUNY Buffalo
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Re: [cisco-voip] Field Notice: FN - 70555 - Legacy VoIP Mode and iOS 13 Apple Push Notification Service Changes That Affect Cisco Jabber, Cisco Webex Teams, Cisco Unified Communications Manager and IM

[Pawlowski, Adam]

Yeah that was a good one.

Back when I did support on a selection of lucent legend, partner acs, etc  
people were on our case the day DST flipped over to fix that clock on the phone 
right away.

Used to have to drive around and plug a console into some of them because the 
modem didn’t work or there wasn’t one.

I believe we’ve used prime provisioning to some success on our clusters as well.

Jabber client upgrades …. Uhhh well I hope you have a management system like 
SCCM or something. The administrator requirement really stings for these things.




From: cisco-voip  On Behalf Of Lelio 
Fulgenzi
Sent: Friday, May 15, 2020 11:32 AM
To: Anthony Holloway 
Cc: cisco-voip voyp list 
Subject: Re: [cisco-voip] Field Notice: FN - 70555 - Legacy VoIP Mode and iOS 
13 Apple Push Notification Service Changes That Affect Cisco Jabber, Cisco 
Webex Teams, Cisco Unified Communications Manager and IM & Presence Service, 
and Cisco Expressway - Software...


We’ve moved to APNs. But we’re still going to have to upgrade all our severs. :(

And clients. At least mobile clients upgrade auto.

This reminds me more of daylight savings time and all those cop files and 
restarts.

Sent from my iPhone


On May 15, 2020, at 11:11 AM, Anthony Holloway 
mailto:avholloway+cisco-v...@gmail.com>> wrote:

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

What's funny is, this has been going on for like 4 years now.

But just like we, as kids, always waited until the last day to complete our 
science fair projects, we wait on securing LDAP, moving to O365's oauth, and 
enabling APNs.

On Fri, May 15, 2020 at 9:49 AM Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:

Like we didn’t have enough to worry about.


https://www.cisco.com/c/en/us/support/docs/field-notices/705/fn70555.html


Sent from my iPhone
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Re: [cisco-voip] Field Notice: FN - 70555 - Legacy VoIP Mode and iOS 13 Apple Push Notification Service Changes That Affect Cisco Jabber, Cisco Webex Teams, Cisco Unified Communications Manager and IM

[Pawlowski, Adam]

Yes, this has been a thing for us in the past, if we didn't schedule SU 
upgrades and try and get them in there, moving forward a few SUs would take a 
ton more time to go through notes, updates, etc. 

This also got us though when you're expected to use a device pack to enable a 
device or a feature (say, turning off the call blacklist feature in the 12.8 
firmware releases for the 8800 series) and you're behind where that wants you 
to be to run it.

I feel like you almost have to update to these things, and we're trying to get 
a better cadence for it to feel out the org for opportunity. Downtimes aren't 
too bad on SU updates, assuming you don't hit the VMWare bug or run out of 
partition space, but there are always some number of phones that wander off and 
don't return, or Jabber quirks.



-Original Message-
From: Lelio Fulgenzi  
Sent: Friday, May 15, 2020 10:53 AM
To: Pawlowski, Adam 
Cc: cisco-voip voyp list 
Subject: Re: Field Notice: FN - 70555 - Legacy VoIP Mode and iOS 13 Apple Push 
Notification Service Changes That Affect Cisco Jabber, Cisco Webex Teams, Cisco 
Unified Communications Manager and IM & Presence Service, and Cisco Expressway 
- Software Upg...


I guess depending what version you’re running. 

But it looks like unless you’re at the latest and greatest SU, you’re upgrading 
quite a bit. 



Sent from my iPhone

> On May 15, 2020, at 10:49 AM, Pawlowski, Adam  wrote:
> 
> 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
> 
> 
> This one is on easy mode compared to some of these things, at least from my 
> reading of it.
> 
> 
> 
> -Original Message-
> From: cisco-voip  On Behalf Of Lelio 
> Fulgenzi
> Sent: Friday, May 15, 2020 10:48 AM
> To: cisco-voip voyp list 
> Subject: [cisco-voip] Field Notice: FN - 70555 - Legacy VoIP Mode and iOS 13 
> Apple Push Notification Service Changes That Affect Cisco Jabber, Cisco Webex 
> Teams, Cisco Unified Communications Manager and IM & Presence Service, and 
> Cisco Expressway - Software Upg...
> 
> 
> Like we didn’t have enough to worry about. 
> 
> 
> https://www.cisco.com/c/en/us/support/docs/field-notices/705/fn70555.html
> 
> 
> Sent from my iPhone
> ___
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Re: [cisco-voip] Field Notice: FN - 70555 - Legacy VoIP Mode and iOS 13 Apple Push Notification Service Changes That Affect Cisco Jabber, Cisco Webex Teams, Cisco Unified Communications Manager and IM

[Pawlowski, Adam]

I always struggle with the competing schools of thought on keeping things up to 
date.

The network guys don’t want to update anything because of more bugs, but I 
still feel that’s old school where we weren’t talking about the same level of 
exposure and frequency of updates. I like to keep up to date because I don’t 
like digging a hole for myself on features, user doc, bug resolutions, compat, 
device packs etc. And the number of severe issues has rounded off, at least for 
UC where it seems the platforms are largely mature.

The 11.5 train has been great as far as not having too many issues, and having 
a good set of features and such backported into them from 12/12.5.

APNs are a usability thing with Jabber for sure, I don’t know how anyone lives 
with that app and not having that. It would be like going back in time when I 
had my StarTAC and had to look at the flashing LED to see if I’d wandered out 
of service or into an expensive “Roaming” area just to make sure I could 
receive a phone call. No thanks.

Looking forward to future support for push on Android too so we can catch those 
up.



From: cisco-voip  On Behalf Of Anthony 
Holloway
Sent: Friday, May 15, 2020 11:11 AM
To: Lelio Fulgenzi 
Cc: cisco-voip voyp list 
Subject: Re: [cisco-voip] Field Notice: FN - 70555 - Legacy VoIP Mode and iOS 
13 Apple Push Notification Service Changes That Affect Cisco Jabber, Cisco 
Webex Teams, Cisco Unified Communications Manager and IM & Presence Service, 
and Cisco Expressway - Software...

What's funny is, this has been going on for like 4 years now.

But just like we, as kids, always waited until the last day to complete our 
science fair projects, we wait on securing LDAP, moving to O365's oauth, and 
enabling APNs.

On Fri, May 15, 2020 at 9:49 AM Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:

Like we didn’t have enough to worry about.


https://www.cisco.com/c/en/us/support/docs/field-notices/705/fn70555.html


Sent from my iPhone
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Re: [cisco-voip] Field Notice: FN - 70555 - Legacy VoIP Mode and iOS 13 Apple Push Notification Service Changes That Affect Cisco Jabber, Cisco Webex Teams, Cisco Unified Communications Manager and IM

[Pawlowski, Adam]

This one is on easy mode compared to some of these things, at least from my 
reading of it.



-Original Message-
From: cisco-voip  On Behalf Of Lelio 
Fulgenzi
Sent: Friday, May 15, 2020 10:48 AM
To: cisco-voip voyp list 
Subject: [cisco-voip] Field Notice: FN - 70555 - Legacy VoIP Mode and iOS 13 
Apple Push Notification Service Changes That Affect Cisco Jabber, Cisco Webex 
Teams, Cisco Unified Communications Manager and IM & Presence Service, and 
Cisco Expressway - Software Upg...


Like we didn’t have enough to worry about. 


https://www.cisco.com/c/en/us/support/docs/field-notices/705/fn70555.html


Sent from my iPhone
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Re: [cisco-voip] UCCX Flex Licensing

[Pawlowski, Adam]

It was not the last time I looked into it (a few months ago), there were still 
suites for EA.

From: cisco-voip  On Behalf Of Matthew 
Loraditch
Sent: Monday, May 11, 2020 10:15 PM
To: Lelio Fulgenzi 
Cc: Cisco VoIP Group 
Subject: Re: [cisco-voip] UCCX Flex Licensing

I don’t know anything about EAs, don’t have anyone big enough to sell them to.



Matthew Loraditch​

Sr. Network Engineer


p: 443.541.1518



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

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From: Lelio Fulgenzi mailto:le...@uoguelph.ca>>
Sent: Monday, May 11, 2020 8:19 PM
To: Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>>
Cc: Anthony Holloway 
mailto:avholloway+cisco-v...@gmail.com>>; 
Cisco VoIP Group mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] UCCX Flex Licensing

[EXTERNAL]

After all that I want to ask...

Is UCCx included in EA yet? Or still separate.


Sent from my iPhone

On May 11, 2020, at 8:03 PM, Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>> 
wrote:

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<mailto:ith...@uoguelph.ca>

A port is a port.

Chat/Email and Advanced Outbound Campaigns (predictive/progressive dialing) 
require Flex Premium so agents doing those functions would need premium.





Matthew Loraditch​

Sr. Network Engineer


p: 443.541.1518



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<https://facebook.com/heliontech>



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From: Anthony Holloway 
mailto:avholloway+cisco-v...@gmail.com>>
Sent: Monday, May 11, 2020 7:11 PM
To: Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>>
Cc: Pawlowski, Adam mailto:aj...@buffalo.edu>>; Cisco VoIP 
Group mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] UCCX Flex Licensing

[EXTERNAL]

Why did they have to borrow the same names for the licensing levels?  It's like 
when Cisco decided to call UCCX CAD+Finesse Mixed mode, while on CUCM mixed 
mode already meant secure communications.  Anyway.

Ok, so, a port is a port in Flex?  There is no concept of a premium port or a 
standard port then?  Any kind of inbound port can do any kind of feature?  The 
only licensing levels are for Agent/Supervisor capabilities?

On Mon, May 11, 2020 at 6:07 PM Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>> 
wrote:
You need to dissociate flex std/prem from L-CCX std/pre. Every feature on flex 
is the equivalent of the L-CCX premium level from a capabilities standpoint.

Does that make sense?

Get Outlook for iOS<https://aka.ms/o0ukef>


Matthew Loraditch​

Sr. Network Engineer


p: 443.541.1518



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From: Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>>
Sent: Monday, May 11, 2020 7:03:47 PM
To: Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>>
Cc: Pawlowski, Adam mailto:aj...@buffalo.edu>>; Cisco VoIP 
Group mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] UCCX Flex Licensing

[EXTERNAL]

My man!  Always coming through!  So, the Supervisor one is true but the admin 
one is bogus, right?  I mean, about the Premium requirement for each.

So, what is still confusing to me is, in the past, the Premium seat also got 
you 2 premium IVR ports.  Does a standard flex seat get you 2 standard ivr 
ports?  Thus, a mized std/pre felx deal is going to net you a mixed std/pre 
port solution?  How does that work?

On Mon, May 11, 2020 at 4:36 PM Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>> 
wrote:

A-FLEX-CC has Standard and Premium Licenses.



These are different from non flex licensing.



Standard is inbound agent licensing essentially



Premium is supervisor licensing,   email/chat agents, outbound campaign 
licensing.



2 CTI ports per agent/license.



Admin sti

Re: [cisco-voip] UCCX Flex Licensing

[Pawlowski, Adam]

This was the information I heard as well, and the purchase quantities are based 
on feature utilization and concurrency.



From: cisco-voip  On Behalf Of Brian Meade
Sent: Monday, May 11, 2020 5:07 PM
To: Anthony Holloway 
Cc: Cisco VoIP Group 
Subject: Re: [cisco-voip] UCCX Flex Licensing

Pretty sure when buying as A-Flex-CC that it always just gives you Premium 
licensing on the CCX side.  Had this cause an issue with a customer that was 
staying on Enhanced for the extra CTI ports for many years.

On Mon, May 11, 2020 at 4:23 PM Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>> 
wrote:
All,

Anyone already deal with this themselves?  I am reading/being told something I 
cannot swallow as the truth, because it seems so ridiculous.

I am being told that you need a Premium license to even login as a Supervisor 
at all.  Like, not for extra functionality (silent monitoring), but just as a 
basic license requirement to even sign in.

Also, I am being told a Premium license is required for Administrative users 
too.  Like, even the app admin account.  So what, completing a fresh install 
now requires a Premium license?

Are either of these true?  Can you confirm from your own tests that this is in 
fact how Flex works in UCCX on-prem?
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Re: [cisco-voip] Cisco moth-balling CUE - Is Connection SRSV the answer?

[Pawlowski, Adam]

In my experience with it, message replication or something can break, and TAC 
can fix it, but that’s pretty rare.

I’ve yet to have any other sort of database issues with it, and it only has 
been upset by overloading it, or resources issues in VMWare.

If you’re going to play the “if it doesn’t shut down clean then rebuild” game 
then the restore can take quite a while. I guess it depends what sort of 
services you need to deliver, and when.

I can fall back with a TCL on the gateway that plays an announcement and hangs 
up on the caller if I need to or whatever, if it came down to a DR scenario. 
Given the number of times the ILEC’s voicemail product also decided to stop 
answering, or whoops your greetings are all gone, this product has a pretty 
good track record, if not better. All depends on its care and feeding though.

Adam

From: cisco-voip  On Behalf Of Lelio 
Fulgenzi
Sent: Monday, May 4, 2020 4:11 PM
To: Charles Goldsmith 
Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net) 

Subject: Re: [cisco-voip] Cisco moth-balling CUE - Is Connection SRSV the 
answer?

All valid questions. No offense taken. Unless of course, you complain about me 
primarily using the @ macro plus route filters in all my route patterns. Then, 
them’s fighting words. 

The great thing about CUE was that it covered all scenarios with one solution. 
Every other scenario will need at least another fall-back meaning two 
solutions. I did this in my head a while back, never got it down on paper.

While I can appreciate the idea of a UNTCNXN cluster (is that the right acronym 
Anthony?), I’m not sold that there will never be a scenario where the second 
node will always work during whatever maintenance we’re planning. I’ve read 
document after document after scenario after scenario and have found we always 
seem to fit in that one exception to the rule for whatever reason.

I’m not saying that we won’t eventually move to a CUXN cluster (we’re not there 
yet) – but I was hoping to have a bit more time to delve into a proper design 
of both what the cluster can and can’t give us and what options we have for 
fall-back.

Let’s say, for whatever reason, a database corruption is replicated across the 
cluster. Then what? What do I do? I have to restore services from backup, 
rebuild the cluster, etc. All the while, having an unreliable AA going around 
because SRSV is trying to connect? (again, I don’t know the ins and outs of 
SRSV and CNXN clusters).

Having CUE available let me sleep at night and gave me a quick get out of jail 
free card I could use for almost any maintenance requirement, including those 
outside my control.



From: Charles Goldsmith mailto:w...@woka.us>>
Sent: Monday, May 4, 2020 1:53 PM
To: Lelio Fulgenzi mailto:le...@uoguelph.ca>>
Cc: Eric Pedersen 
mailto:peders...@bennettjones.com>>; voyp list, 
cisco-voip (cisco-voip@puck.nether.net) 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Cisco moth-balling CUE - Is Connection SRSV the 
answer?

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

Lelio, just curious why you would have scheduled downtime for the entire CUC 
cluster?  I can appreciate downtime for a node for maintenance, but even during 
an upgrade, your cluster should be up, one node or the other.

If it's more DC / network outage, why not have the 2nd node of your CUC cluster 
where ever you have your CUE for "backup".

No offense intended on your design, just wanting to know and possibly learn if 
it's something I'm overlooking.

Thanks


On Mon, May 4, 2020 at 12:48 PM Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:

Ok. Thanks. This might work.

What I’m hoping to be able to do is to manually redirect calls from Connection 
to SRSV (for AA and voicemail) and still allow calls to be transferred 
accordingly to phones registered to CUCM, not SRST.

This was easily done with CUE, since it would register to both CUCM and SRST.

If SRSV has similar functionality, we’re golden.
Sent from my iPhone

On May 4, 2020, at 1:43 PM, Eric Pedersen 
mailto:peders...@bennettjones.com>> wrote:

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

Yes, from what I remember it can operate while CUCM and CUCX are both up.

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Lelio Fulgenzi
Sent: Monday, May 4, 2020 9:37 AM
To: voyp list, cisco-voip 
(cisco-voip@puck.nether.net) 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Cisco moth-balling CUE - Is Connection 

Re: [cisco-voip] Cisco moth-balling CUE - Is Connection SRSV the answer?

[Pawlowski, Adam]

I use UCXN , the “Cisco” part I guess implied. Feels like one too many letters 
though.

As for cluster downtime, the only time we really had the system completely down 
had been for Unity -> Unity Connection migration, and if we have to grow the 
cluster again to support more Jabber clients and rebuild, that would also do 
it. COBRAS took about a billion years.

Adam

From: cisco-voip  On Behalf Of Anthony 
Holloway
Sent: Monday, May 4, 2020 1:59 PM
To: Eric Pedersen 
Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net) 

Subject: Re: [cisco-voip] Cisco moth-balling CUE - Is Connection SRSV the 
answer?

"CUCX"

I find it interesting the different ways we Engineers write that.  I have also 
seen CUXN, CUCXN and CUC.  I'm team CUC, but I think we can all agree that 
simply "Unity" is wrong.

On Mon, May 4, 2020 at 12:45 PM Eric Pedersen 
mailto:peders...@bennettjones.com>> wrote:
Yes, from what I remember it can operate while CUCM and CUCX are both up.

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Lelio Fulgenzi
Sent: Monday, May 4, 2020 9:37 AM
To: voyp list, cisco-voip 
(cisco-voip@puck.nether.net) 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Cisco moth-balling CUE - Is Connection SRSV the 
answer?

Do you know if SRSV can operate while CUCM is up?

The great thing about CUE, is that it operated while CUCM was up. Completely 
independent of Unity Connection.

This means, I could schedule downtime for Connection and have an almost fully 
operational AA working.

From: Eric Pedersen 
mailto:peders...@bennettjones.com>>
Sent: Monday, May 4, 2020 11:35 AM
To: Lelio Fulgenzi mailto:le...@uoguelph.ca>>; voyp list, 
cisco-voip (cisco-voip@puck.nether.net) 
mailto:cisco-voip@puck.nether.net>>
Subject: RE: Cisco moth-balling CUE - Is Connection SRSV the answer?

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

I used SRSV a while ago for one of our remote sites. I found it much simpler to 
get up and running than CUE and you can use your centralized Exchange.  IIRC 
you can send your voicemail pilot back to the gateway SRSV is registered to so 
all calls go to it. But it's been a really long time…

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Lelio Fulgenzi
Sent: Sunday, May 3, 2020 11:38 AM
To: voyp list, cisco-voip 
(cisco-voip@puck.nether.net) 
mailto:cisco-voip@puck.nether.net>>
Subject: [cisco-voip] Cisco moth-balling CUE - Is Connection SRSV the answer?


Looks like Cisco is moth-balling CUE. I liked that product. I’ll miss it.

It looks like Connection SRSV is the answer. Although I’m not sure it will 
offer everything we used (and planned to use) CUE for. For example, our 
voicemail ports forwarded to CUE which was always registered to CUCM. This way, 
calls would continue to work. It’s looking like SRSV will only work if the 
router is in SRST mode and all phones are registered to SRST.

Has anyone successfully deployed SRSV? How about using it during voicemail 
maintenance?

Lelio



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

[Pawlowski, Adam]

What does the dialed number analyzer tool tell you?

Is there some digit manipulation going on ? You're not somehow using this 
trunk/port for registrations ? Messaging inbound is using a domain or address 
the UCM is going to respond to ?

Adam

From: cisco-voip  On Behalf Of naresh 
rathore
Sent: Thursday, April 30, 2020 6:27 AM
To: Amit Kumar 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] sip 404 not found for incoming calls

hi


I tried both, via translation pattern or directly pointing a particular number 
to phone but still the same result.

Regards



From: Amit Kumar mailto:amit3@gmail.com>>
Sent: Thursday, April 30, 2020 6:41 PM
To: naresh rathore mailto:nare...@hotmail.com>>
Cc: James B mailto:james.buchan...@gmail.com>>; 
cisco-voip@puck.nether.net 
mailto: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 
mailto:nare...@hotmail.com>> 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)

[cid:9c8082fb-a47a-4859-8d73-bd22200523c5]
[cid:f721c217-c6cf-4fd5-bbc6-10962f7f7269]
[cid:2300f744-b399-4d90-ab4b-133bb37a3611]
[cid:a2643290-7308-42d9-b6bc-699c2df07a6f]


Regards

Naray

From: James B mailto:james.buchan...@gmail.com>>
Sent: Thursday, April 30, 2020 5:39 PM
To: naresh rathore mailto:nare...@hotmail.com>>; 
cisco-voip@puck.nether.net 
mailto: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] ng911

[Pawlowski, Adam]

Hi Mark,

You caught me replying to the latest post on this one.

Yes - they do let you define remote sites and locations, which I mentally put 
akin to building subnet based / L3 tracking for a remote site in to the UCM to 
handle fixed locations. I wouldn't assume that is tenable to support from an 
administrative side of the shop for a home user, or somewhere that does not 
have a static IP. Sure, I suppose you could try it over VPN if that's possible 
with the application.

I agree with you on the need for these items to be available even in cloud 
softphone products, and wish that there was a better way to geo-locate with 
precision, without the customer's intervention, with plus or minus on that 
location data being available to applications. Today I believe some products 
would allow the customer to supply an address, which can be verified. As for 
the cellphones we've had customers want to use old or deactivated phones as 
mobile phones, so even there who knows if it will be able to place that 
emergency call or not. It should.

I also get the impression on the can-kicking in a way. Jabber's location 
services haven't improved much lately, and CER sure hasn't. But, there's money 
to be made so I'm sure something will become available. I too try and make sure 
that we can do the best we can, and thus provide room level granularity to 
responding agencies using CER, which required quite a bit of work to get that 
right. We had an XML app that popped the deskphones to confirm port maps, and 
have scripting to update the port/erl/etc files for us to import to CER. The 
need to shuttle all the calls to a PSAP will blow that up without buying a 
service, as I don't think anyone is going to want to have thousands of DIDs 
sitting there to be ELINs (which we find are also vectors for spoofed calls 
which create some unfortunate PSAP call back scenarios).

It will come around, I'm sure of that. If we can wait for that piece to be 
there before we end up shoving into "cloud" and wholesale nomadic softphone 
products, especially in light of the current situation, that I don't know.

Best,

Adam

From: Mark H. Turpin 
Sent: Monday, April 20, 2020 2:28 PM
To: Pawlowski, Adam ; cisco-voip@puck.nether.net
Subject: Re: ng911

Adam,

Microsoft's Dynamic Location Routing & trusted IP architecture gets them to 
being NG911-compliant. It isn't just looking at the IP's geolocation, but 
rather an administrator can have a workflow that requires the user's external 
IP to be in the list of defined IPs, and that IP must have a dispatchable 
location associated to it.

For our current products, CER+CUCM/Jabber/Hybrid Teams calling, these things 
are all easily solved when the call is going through call control you can 
control, or PSTN you can control. But when you take away PSTN and call control, 
you need the software creator to provide the knobs and switches. Presently, 
Webex Teams + Cloud Calling has no knobs for us engineers to turn.

On one hand, Cisco isn't required to solve this problem and customers don't 
need to be compliant for nomadic users until 2022. I'm sure this is on their 
radar as Cisco has dedicated quite the landing page to the topic: 
https://www.cisco.com/c/en/us/products/unified-communications/next-gen-karis-law.html
 but I suspect it is a can that is being kicked down the road, maybe?

On the other hand, despite it not being required, I morally feel like we should 
provide the best information about a 911-caller to get the first responders to 
the right place, as quickly as possible. But, however it is accomplished, it 
can't be so burdensome that admins don't deploy it and users won't update it. 
It is my understanding that a 911 call through a Webex Calling cloud provider 
will only dispatch to the main corporate address. Perhaps that's fine for a 
retail store that's wide open and less than 5000 square feet. But I want to do 
better for facilities like a 40-story office building.




____
From: Pawlowski, Adam mailto:aj...@buffalo.edu>>
Sent: Monday, April 20, 2020 8:56 AM
To: Mark H. Turpin mailto:mtur...@covene.com>>; 
cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> 
mailto:cisco-voip@puck.nether.net>>
Subject: RE: ng911

*** EXTERNAL EMAIL - DO NOT CLICK LINKS ***

It is on the list of considerations for sure.



If I'm reading (skimmed) Microsoft's document you linked, this works more or 
less how the traditional E911 systems work, and do nothing for you for home or 
roaming users. In a way I am surprised there aren't some more solutions on the 
table, but, at the same time these disclaimers and the wording in a lot of the 
documentation is to try and shy away from any sort of liability. If I go to 
Bing to figure out how to order a pizza, both it and the pizza places can tell 
where I am pretty darn close just based on my IP address. You'd think that sort 
of thing would be a start to a 911 solution, but

Re: [cisco-voip] ng911

[Pawlowski, Adam]

It is on the list of considerations for sure.

If I'm reading (skimmed) Microsoft's document you linked, this works more or 
less how the traditional E911 systems work, and do nothing for you for home or 
roaming users. In a way I am surprised there aren't some more solutions on the 
table, but, at the same time these disclaimers and the wording in a lot of the 
documentation is to try and shy away from any sort of liability. If I go to 
Bing to figure out how to order a pizza, both it and the pizza places can tell 
where I am pretty darn close just based on my IP address. You'd think that sort 
of thing would be a start to a 911 solution, but, it is imperfect, imprecise, 
and can be wrong - and the liability there may force some to just say, look, we 
can't do it - implement at your own risk.

Absolutely waiting on this one to be more well developed, but, I hazard that 
the business address solutions are just fine for SMB and not Enterprise, which 
holds true for whole hog implementations of these systems in general at the 
moment. (Costs, ROI, feature set, etc)

Adam

From: cisco-voip  On Behalf Of Mark H. 
Turpin
Sent: Saturday, April 18, 2020 3:15 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] ng911

Is anyone thinking about NG911 compatibility for pure Teams/Webex cloud 
calling? I understand Intrado/RedSky offerings for CER/on-prem/Jabber/hybrid 
calling.

The Webex Calling terms 
(https://www.cisco.com/c/dam/en_us/about/doing_business/legal/OfferDescriptions/cisco_collaboration_flex_plan.pdf)
 state pretty clearly Cisco isn't supporting it today.

Emergency Response Disclaimer
YOUR EMERGENCY RESPONSE LOCATION FOR PURPOSES OF EMERGENCY CALLS IS LIMITED TO
YOUR COMPANY ADDRESS. IT IS YOUR RESPONSIBILITY TO ADVISE YOUR AUTHORIZED USERS
TO ALWAYS PROVIDE THEIR CURRENT LOCATION WHEN CALLING EMERGENCY SERVICES.

That disclaimer is fine except when a user is calling 911 and can't speak to 
provide their address.

While Ray Baum's Act isn't in effect yet, it seems like Microsoft might have a 
leg up on this already with Dynamic Location Routing capabilities via their LIS 
and trusted IP architecture.
https://docs.microsoft.com/en-us/microsoftteams/configure-dynamic-emergency-calling

I don't have an answer yet, just starting the conversation.
-Mark
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Re: [cisco-voip] Jabber audio issues?

[Pawlowski, Adam]

Hello all

I wrote some scripting and bulk deployed Jabber to any of our users with 
telephony so they can grab and go, and we’ve seen (if control hub analytics are 
to be believed) about 25% of people use it at one point or another. I have not 
heard any complaints about audio – you can tell when someone is at home on a 
call and they decide to email you that attachment you’re talking about and 
their upload is congested. We have some customers that for whatever reason are 
using RDP to connect to a machine in their office to run Jabber there. That 
works through RDP gateway but not Anyconnect VPN, the audio is choppy and 
broken up there. Someone else reported that on the Community forums but there 
was no resolution that I could see.

The only real trouble we’ve had is maybe a couple of customers report that the 
calls drop off after a few minutes, or Jabber will disconnect from services. 
Universally, this seems to be related to their home routers or access points 
but we haven’t gathered enough data to say anything about that. Unfortunately, 
in this case there is not much that we can do for anyone unless they can use 
mobile data.

Adam



From: cisco-voip  On Behalf Of Scott Voll
Sent: Monday, April 13, 2020 6:15 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Jabber audio issues?

How has everyone's deployment of Jabber been doing since everyone has went home?

we were about half way through migrating from IPC to Jabber when we were all 
sent home to Telecommute.

we have had some people complaining about there audio issues.  robotic or 
missing.

CMR data shows low latency, low jitter, and low packet loss. but they still 
complain.

we did find a couple PRI's that had a clocking issue and fixed them.  we also 
opened a TAC case and they asked us to remove the OPUS codec, which we did.  
most people saw improvements, but we still have a few that are complaining.

They go back to IPC and say things are better.

Anyone having any issues like this?  If so, how did you correct it?

TIA
Scott


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Re: [cisco-voip] [EXTERNAL] Re: PSTN Calls Incorrectly Flagged as "Potential SPAM"

[Pawlowski, Adam]

My understanding of how it would work, would be that the carrier would assert 
you originated the traffic and were doing so legitimately, not that you had to 
do it yourself.

However, that of course leads to a whole other world of fraud attempts, if 
someone can bust in and use your system like a hat or use call forwarding.

My experience with CNAM troubles in the past has been that 95% of the time the 
customer misread it, it was generic with the name of a locale, or they were 
using a device with a contact list that had the wrong information in it. The 
other 5% is impossible as I’ve never had our carrier do anything other than say 
it is the other customer’s problem.  The carriers and I guess people in the 
know or connected in this business can work their contacts back channel but 
otherwise the customer seeing the bad information usually has to complain. 
Maybe others have better success.

I have had other people report names and various things have appeared on their 
mobile devices recently, which should a whole other mix of terrible if it’s not 
CNAM and populated from account data or something bizarre.

Adam

From: cisco-voip  On Behalf Of Lelio 
Fulgenzi
Sent: Friday, April 3, 2020 2:05 PM
To: Matthew Loraditch ; Ryan Huff 
; JASON BURWELL 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] [EXTERNAL] Re: PSTN Calls Incorrectly Flagged as 
"Potential SPAM"

I specifically do this so we can do TEHO from our campuses. Some are serviced 
by other carriers.

We signed a waiver to do so.



From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Matthew Loraditch
Sent: Friday, April 3, 2020 1:56 PM
To: Ryan Huff mailto:ryanh...@outlook.com>>; JASON 
BURWELL mailto:jason.burw...@foundersfcu.com>>
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] [EXTERNAL] Re: PSTN Calls Incorrectly Flagged as 
"Potential SPAM"

I’m wondering how this is all going work once shaken/stir are fully 
implemented. Are we going to have to prove we own numbers to other carriers? 
I’m sure many of us have DIDs outpulsed of alternative circuits at times.



Matthew Loraditch​

Sr. Network Engineer


p: 443.541.1518



w: www.heliontechnologies.com

 |

e: mloradi...@heliontechnologies.com


[Helion Technologies]


[Facebook]


[Twitter]


[LinkedIn]







From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Ryan Huff
Sent: Friday, April 3, 2020 1:52 PM
To: JASON BURWELL 
mailto:jason.burw...@foundersfcu.com>>
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] [EXTERNAL] Re: PSTN Calls Incorrectly Flagged as 
"Potential SPAM"

[EXTERNAL]

You need to become a thorn in the side of the AM for your upstream carrier. 
It’s a carrier -2- carrier fight at that point.
Sent from my iPhone

On Apr 3, 2020, at 13:49, JASON BURWELL 
mailto:jason.burw...@foundersfcu.com>> wrote:

Thanks for all the replies thus far. To answer a couple of the questions that 
have come up, we are using valid, working DID numbers that we own for all 
outbound Calling Number Masks. And none of the DIDs forward to other carriers, 
they are all pointed from the PSTN to our various gateways.

One thing that was mentioned is that a SPAM autodialer bot has at some point 
spoofed some of our numbers causing them to be flagged as SPAM which is 
certainly a possibility and nothing we can do about that. I regularly get calls 
even on my cell phone with the whole “hey I missed a call form you” from the 
caller and they get irritated when I tell them, sorry I did not call you.

I know there is nothing we can do from a configuration perspective. I was just 
hoping there was some managed whitelist these carriers used that I was unaware 
of. I know there are various 3rd party apps that do this but its definitely 
something being done at the carrier level as well because I frequently get 
these messages as well on a Verizon phone and I do not have and SPAM apps or 
subscriptions.

As more and more numbers are spoofed for SPAM calls I imagine at some point all 
numbers will be flagged at potential SPAM at this rate.

So unless I missed something, it sounds like there is really nothing we can do 
about it?

Jason



From: Ryan Huff mailto:ryanh...@outlook.com>>
Sent: Friday, April 3, 2020 12:30 PM
To: JASON BURWELL 
mailto:jason.burw...@foundersfcu.com>>
Cc: cisco-voip@puck.nether.net
Subject: [EXTERNAL] Re: [cisco-voip] PSTN Calls Incorrectly Flagged as 
"Potential SPAM"

CAUTION: This email originated outside of Founders Federal Credit Union. Do not 
click links or open attachments unless you recognize the sender and know the 
content is safe.

I’ve 

Re: [cisco-voip] jabber searching only user in AD but not local cucm database

[Pawlowski, Adam]

Hi,

Jabber is going to connect directly to an LDAP source unless you use UDS.

You'll have to create records in AD to match those sources, or go through the 
fun of UDS (avatar hosting, make sure records from your source are right).

You can alternatively supply some other sort of custom lookup tab, contacts, 
etc, but that is not really sustainable.

Best,

Adam Pawlowski
SUNY Buffalo



From: cisco-voip  On Behalf Of naresh 
rathore
Sent: Wednesday, April 1, 2020 8:22 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] jabber searching only user in AD but not local cucm 
database

hi,


I have the requirement to search both users configured in AD and its working 
fine. I have some local users on cucm as well, i am not able to search them, on 
CUCM i have same field populated as AD user, specially mailid.

any suggestions

System version: 12.5.1.11900-146.
Jabber version: 12.8.0
Regards

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

[Pawlowski, Adam]

We are running this in production with UCM 11.5, CCX 12.0, and Expressway 
X12.5.6 without any issues, CCM cores, or other problems.

If that core out bug is impacting in your version of the UCM then sure maybe 
avoid this, but at least per my own case we have not had any trouble with this 
path headers being enabled.



From: cisco-voip  On Behalf Of Lelio 
Fulgenzi
Sent: Monday, March 30, 2020 2:19 PM
To: Aman Chugh 
Cc: voyp list, cisco-voip 
Subject: Re: [cisco-voip] [External] Re: CCX phone agent over MRA?

Thanks guys.

I should have read that section more closely. Darn.

From: Aman Chugh mailto:aman.ch...@gmail.com>>
Sent: Monday, March 30, 2020 2:18 PM
To: Lelio Fulgenzi mailto:le...@uoguelph.ca>>
Cc: voyp list, cisco-voip 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] [External] Re: CCX phone agent over MRA?


It should affect all clients which are registered over MRA with multiple lines.

On Mon, Mar 30, 2020 at 2:06 PM Hunter Fuller 
mailto:hf0...@uah.edu>> wrote:
If you can do a tcpdump between expressway and call manager you will see a 404 
and it would verify that. We didn’t see it on Jabber because we just hadn’t 
tried it - only needed multiple lines on hard phones.

Or you can just change the setting on expressway, reregister Jabber, and hope 
for the best. Depends on what kind of lifestyle you subscribe to, I suppose.

On Mon, Mar 30, 2020 at 13:01 Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:
Um, ok. I just got a call from someone saying secondary lines are not working 
on Jabber desktop via MRA.

I was under the impression from our discussion below that this only affects the 
8800?

But it affects Jabber too?

From: Lelio Fulgenzi mailto:le...@uoguelph.ca>>
Sent: Monday, March 23, 2020 9:52 PM
To: Lelio Fulgenzi mailto:le...@uoguelph.ca>>; Wakelin, 
Frank mailto:fwake...@richmond.ca>>
Cc: voyp list, cisco-voip 
mailto:cisco-voip@puck.nether.net>>
Subject: RE: [cisco-voip] CCX phone agent over MRA?

Yeah. It exists! (on 8.10)

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Lelio Fulgenzi
Sent: Monday, March 23, 2020 9:41 PM
To: Wakelin, Frank mailto:fwake...@richmond.ca>>
Cc: voyp list, cisco-voip 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] CCX phone agent over MRA?

Wow. Interesting. Will have to read the X10.4 (*cough*) guide to see if a 
similar feature exists.

Thanks for this.
Sent from my iPhone

On Mar 23, 2020, at 9:29 PM, Wakelin, Frank 
mailto:fwake...@richmond.ca>> wrote:
Absolutely, here’s what Aman sent me:

Anther thing of note is use of SIP Path header on Expressway C. This may be 
needed to turned on if you multiple lines on 88xx phone. I have seen an issue 
when were we not able to ring second line on the phone when this was turned off 
on Expressway C.

You will need that turned up on Expressway C under unified communication - 
configuration.

I ran into this in testing in my environment few days back.

There are certain version requirements to have this turned on with CUCM.  Page 
33

https://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/expressway/config_guide/X8-11/Mobile-Remote-Access-via-Expressway-Deployment-Guide-X8-11-4.pdf


---
Frank Wakelin – Senior Network Analyst
Information Technology | City of Richmond

Office +16042764190
Mobile +17788394693
fwake...@richmond.ca

From: Lelio Fulgenzi mailto:le...@uoguelph.ca>>
Sent: March 23, 2020 6:23 PM
To: Wakelin, Frank mailto:fwake...@richmond.ca>>
Cc: voyp list, cisco-voip 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] CCX phone agent over MRA?


Um, so, outside of ccx, getting multiline 8800s working over MRA requires extra 
config?

Can you share any tech notes ?


Sent from my iPhone

On Mar 23, 2020, at 8:53 PM, Wakelin, Frank 
mailto:fwake...@richmond.ca>> wrote:
Yeah, once again as it turns out it was the Cisco TAC engineer not really 
knowing the product they are apparently supporting – which is fine, but they 
also never escalated the call to someone who does when the question was over 
their head either. – heavy sigh –

As it turns out we generally configure all of our agents contact centre lines 
as their second line.  After my post Aman reached out to me with a note about 
the use of the SIP Path header on Expressway C as this is needed to support 
multiple lines on 88xx phone. The lack of multiline support was what was 
killing the call to the agent extension (on the second line of the phone) when 
it was presented by CCX.  I had a chance to enable the SIP path header today 
and successfully tested CCX.

So thanks all for your assistance and more so your insistence that this is 
supported/working in your environments.  Thanks Aman for the mention of the SIP 
path header!

---
Frank Wakelin – Senior Network Analyst
Information Technology | City of Richmond

Office +16042764190
Mobile +17788394693
fwake...@richmond.ca

From: cisco-voip 

Re: [cisco-voip] Java and custom UCCX

[Pawlowski, Adam]

I shunted in a new oracle jdbc jar to replace the existing one (and thus losing 
the entire database connection pooling framework that CCX has) because the in 
built one is archaic and doesn’t work with oracle encryption or modern rev, 
then wrote some java code to handle simple queries.

It works until it doesn’t want to – SU installs or system reboots sometimes the 
jar doesn’t load for … no given reason at all, then you have to reboot to bring 
it up hopefully.

As bad as it sounds if you’re stuck on that system would it be more reasonable 
to write a broker/proxy that can shuttle these requests for you?

Adam

From: cisco-voip  On Behalf Of Anthony 
Holloway
Sent: Thursday, March 26, 2020 2:32 AM
To: Tim Smith 
Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net) 

Subject: Re: [cisco-voip] Java and custom UCCX

I did a Java 
thingoncenever
 again man.  never again.

I would recommend posting in the UCCX community 
forum,
 as there are at least 2 or 3 regulars who I believe could help with this type 
of thing.

And, it's less active, but the developer section for 
UCCX
 might be worth a shot too.

Last but not least, the dude's over at Cloverhound seem to love challenges, as 
they've competed in like a dozen Engineering Death match episodes.  Maybe ping 
them on Twitter.

Good luck.  I look forward to hearing the outcome.

On Wed, Mar 25, 2020 at 6:09 PM Tim Smith 
mailto:tim.sm...@enject.com.au>> wrote:
Big guys, I should be more specific :)

Exact challenge

  *UCCX 10.0 with JRE 1.6
  *   Does not support TLS 1.2
  *   We need to make a REST call with OAUTH and TLS 1.2
  *   Using bouncy castle to try and get around the JRE TLS limitation
Should also note we have working on 11.x with JRE 7 and no bouncy castle. 
(Which sounds less fun, but it works better :)

Cheers,

Tim

Get Outlook for iOS

From: Tim Smith
Sent: Thursday, March 26, 2020 10:00:07 AM
To: voyp list, cisco-voip 
(cisco-voip@puck.nether.net) 
mailto:cisco-voip@puck.nether.net>>
Subject: Java and custom UCCX

Hi guys,

I’m not sure if we are supposed to do this.
I am stuck on a Java and UCCX issue.

We are trying to run a custom class, and we are having issues with the loading 
of the class.
It seems very Cisco specific as we can run it anywhere else compiled and ran on 
same JDK/JRE etc.

We’ve put a lot of hours in so far and tried a lot of things, but at a bit of a 
roadblock.

I know there are some great UCCX people on here. Wondering if anyone might be 
up for a quick chat on Webex Teams.

Cheers,

Tim

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Re: [cisco-voip] Jabber client missing Advanced Settings

[Pawlowski, Adam]

I use Orca to modify our package to turn off that UPN discovery. In retrospect, 
I think this did more damage than not. It was slightly more important with the 
CLICK2X.

Many of the groups we sent it to with the flags and msiexec instructions 
couldn’t figure that out or be bothered, so we went to Orca, but, then as it 
invalidates the signature, SmartScreen and apparently Symantec Endpoint 
Protection both claim it’s corrupt or hacked.

Even on a home machine signed in with a Microsoft account, where it picks up on 
that, it works fine with UPN discovery enabled, and the number of cases where 
the logged in user is not the same as the one signing in to Jabber was pretty 
low.

From: cisco-voip  On Behalf Of Lelio 
Fulgenzi
Sent: Monday, March 23, 2020 7:03 PM
To: Anthony Holloway 
Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net) 

Subject: Re: [cisco-voip] Jabber client missing Advanced Settings

Ah. Gotcha. I just used iexpress to package it. Not sure if that’s the wrong 
way to go about it, but I had to do something in a hurry.

The signed cert made it pretty.

From: Anthony Holloway 
mailto:avholloway+cisco-v...@gmail.com>>
Sent: Monday, March 23, 2020 6:30 PM
To: Lelio Fulgenzi mailto:le...@uoguelph.ca>>
Cc: Terry Oakley mailto:terry.oak...@rdc.ab.ca>>; voyp 
list, cisco-voip 
(cisco-voip@puck.nether.net) 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Jabber client missing Advanced Settings

Orca is a MS tool for modifying MSI installer packages.

On Mon, Mar 23, 2020 at 5:21 PM Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:
I’m not sure what ORCA is. Is that a package builder? If so, you might want to 
try iexpress. That’s what I used to build my package with the switches I 
wanted, ie CLEAR=1 and UPN_DISCOVERY_ENABLED=false.

I went so far to get an EV Signing Cert from Digicert so that way people 
wouldn’t get a warning message when they downloaded it from our website.

Works like  charm.

Would probably have to spend quite a bit of time trying to replicate the cert 
signing part, but I’d open a ticket or start a chat.



From: Terry Oakley mailto:terry.oak...@rdc.ab.ca>>
Sent: Monday, March 23, 2020 4:37 PM
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: Jabber client missing Advanced Settings

From my research the same thing but we are just using a USB stick and installed 
the msi. Looks like we may have to do the Microsoft ORCA msi thing which I 
am totally unfamiliar with.

Lelio how is everything in Guelph?  Alberta is reeling for sure under this and 
everything else.

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



From: Lelio Fulgenzi mailto:le...@uoguelph.ca>>
Sent: Monday, March 23, 2020 2:22 PM
To: Terry Oakley mailto:terry.oak...@rdc.ab.ca>>
Cc: voyp list, cisco-voip 
(cisco-voip@puck.nether.net) 
mailto:cisco-voip@puck.nether.net>>
Subject: RE: Jabber client missing Advanced Settings

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 believe if you use a provisioning URL it removes the option. Could be wrong.


From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Terry Oakley via cisco-voip
Sent: Monday, March 23, 2020 4:06 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Jabber client missing Advanced Settings

Have been installing Jabber 12 for the past week and had no issues.   Installed 
today and now we cannot see the Advanced Settings to setup where the client 
should be directed. Has anyone seen this and have a resolution?

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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Re: [cisco-voip] Jabber profile with voicemail service but user without voicemail (issues?)

[Pawlowski, Adam]

In my experience it will try and authenticate and you'd get authentication 
failed alerts

They will also see an error in the client but usually only if they click on 
voicemail.

I think 12.7 or 12..8 added a param to stop all auth if any of it failed which 
could present as aproblem

From: cisco-voip  On Behalf Of Lelio 
Fulgenzi
Sent: Sunday, March 22, 2020 2:53 PM
To: voyp list, cisco-voip (cisco-voip@puck.nether.net) 

Subject: [cisco-voip] Jabber profile with voicemail service but user without 
voicemail (issues?)


Hi everyone,

What happens if I use a service profile that has voicemail as a service but 
I've assigned that to a user that does not have voicemail?

I'm not seeing any syslogs or alerts, but that could be because I don't have 
that alerting setup properly (and likely won't for a while).

Should I be ensuring that those without voicemail get a no voicemail service 
profile?

Lelio

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Re: [cisco-voip] Jabber and Finesse

[Pawlowski, Adam]

It has not had to be the second line for CCX in my experience, seems to  work 
just fine.

From: Terry Oakley 
Sent: Thursday, March 19, 2020 12:19 AM
To: NateCCIE 
Cc: Jose Colon II ; Pawlowski, Adam ; 
Terry Oakley via cisco-voip 
Subject: Re: [cisco-voip] Jabber and Finesse

I have the agent on the second line.   Will check that in the morning,  thank 
you
And will check to see if version of expressway is the issue,

Terry

Sent from my BlackBerry — the most secure mobile device — via the TELUS Network
From: natec...@gmail.com<mailto:natec...@gmail.com>
Sent: March 18, 2020 10:05 PM
To: terry.oak...@rdc.ab.ca<mailto:terry.oak...@rdc.ab.ca>
Cc: jcolon...@gmail.com<mailto:jcolon...@gmail.com>; 
aj...@buffalo.edu<mailto:aj...@buffalo.edu>; 
cisco-voip@puck.nether.net<mailto: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.

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 
mailto:cisco-voip@puck.nether.net>> 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 mailto:jcolon...@gmail.com>>
Sent: Wednesday, March 18, 2020 4:38:26 PM
To: Pawlowski, Adam mailto:aj...@buffalo.edu>>
Cc: Terry Oakley mailto:terry.oak...@rdc.ab.ca>>; 
cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> 
mailto: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 
mailto:aj...@buffalo.edu>> 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 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Terry Oakley via cisco-voip
Sent: Wednesday, March 18, 2020 6:29 PM
To: cisco-voip@puck.nether.net<mailto: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

Re: [cisco-voip] Jabber and Finesse

[Pawlowski, Adam]

Hi Terry - there's one other element to the setup that needs to be set in the 
VCS if it's not.

Maybe this is it?

Step 1

Go to VCS-C.

Step 2

Select VSC-C configuration > Unified Communication > Configuration > SIP Path 
headers and set it to On.


There's not really anything too substantial to configure otherwise. The CSF 
needs to be in the rmjtapi or rmcm or whatever application user's associated 
devices list, and then it "just works"

If you're getting that call cannot be completed message then it almost sounds 
like that line isn't registered and it has no other actions like VM or 
forwarding, which it shouldn't.

Adam

From: Terry Oakley 
Sent: Wednesday, March 18, 2020 10:25 PM
To: jcolon...@gmail.com; Pawlowski, Adam 
Cc: cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] Jabber and Finesse

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 mailto:jcolon...@gmail.com>>
Sent: Wednesday, March 18, 2020 4:38:26 PM
To: Pawlowski, Adam mailto:aj...@buffalo.edu>>
Cc: Terry Oakley mailto:terry.oak...@rdc.ab.ca>>; 
cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> 
mailto: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 
mailto:aj...@buffalo.edu>> 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 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Terry Oakley via cisco-voip
Sent: Wednesday, March 18, 2020 6:29 PM
To: cisco-voip@puck.nether.net<mailto: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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Re: [cisco-voip] Jabber and Finesse

[Pawlowski, Adam]

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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Re: [cisco-voip] gathering and offloading realtime stats (from RTMT)

[Pawlowski, Adam]

This would be cool and for sure something to work on when/if I get stuck in the 
home office. For now I've just been hastily adding code to my other scripts and 
libraries to bulk provision everyone for Jabber while trying not to implode the 
system, but that's about done.



> -Original Message-
> From: cisco-voip  On Behalf Of Lelio
> Fulgenzi
> Sent: Tuesday, March 17, 2020 7:27 AM
> To: Kent Roberts 
> Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net)  v...@puck.nether.net>
> Subject: Re: [cisco-voip] gathering and offloading realtime stats (from RTMT)
> 
> 
> Yes please! Thanks.
> 
> We are still using mrtg, but it should be fine for now.
> 
> Sent from my iPhone
> 
> > On Mar 16, 2020, at 8:21 PM, Kent Roberts  wrote:
> >
> > Yes there is   I can send you links and stuff I’ve done this for years and 
> > use
> grafania to graph it
> >
> >
> > Kent
> >
> >> On Mar 16, 2020, at 16:24, Lelio Fulgenzi  wrote:
> >>
> >> 
> >> Is there a way I can start dumping real-time stats from CUCM? We're
> moving heavily to remote working in light of what's going on and I'd like to
> keep an eye on things outside of SNMP polling.
> >> 
> >> ___
> >> cisco-voip mailing list
> >> cisco-voip@puck.nether.net
> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> ___
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Re: [cisco-voip] Expressway MRA registrations

[Pawlowski, Adam]

I don’t have a “view provisioning sessions” link on any of my Expressway

From: cisco-voip  On Behalf Of Brian Meade
Sent: Thursday, March 12, 2020 2:01 PM
To: SK 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Expressway MRA registrations

Status->Unified Communications and can look at View Provisioning Sessions link.

On Tue, Mar 10, 2020 at 11:33 PM SK 
mailto:cciecollab2...@gmail.com>> wrote:
Hello,

How can I find how many current registrations are active on the expressway for 
MRA?

Thank you .
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Re: [cisco-voip] is there a Jabber APN status page anywhere?

[Pawlowski, Adam]

We'd been getting push notification failures as well over the last week 
sporadically (and this morning as it happens)

Restarted XCP Config Manager and Cloud Onboarding services since we'd had some 
network trouble here and thought that maybe this would have been related.

Perhaps not?

Did you hear anything back? My success rate when opening cases about what 
appear to be cloud services like smart licensing has been pretty low.

Adam

From: cisco-voip  On Behalf Of Lelio 
Fulgenzi
Sent: Wednesday, February 26, 2020 3:55 PM
To: voyp list, cisco-voip (cisco-voip@puck.nether.net) 

Subject: [cisco-voip] is there a Jabber APN status page anywhere?


Is there a status page for Jabber's Apple Push Notification service? We started 
getting a number of errors which have seemingly stopped. I've got a TAC case 
open, but wanted to see if there's a status page to check.
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Re: [cisco-voip] Join Webex Button on TOuch 10

[Pawlowski, Adam]

I set up a dial pattern for this in our UCM, to match the dialing prefix set in 
Webex teams, which currently goes to our site.

It’s not that great from audio only devices because the meet IVR just answers 
with “enter your meeting number” and then is silent.

However – you can start a meeting this way very easily as you join as a video 
device as far as Webex is concerned.

From: cisco-voip  On Behalf Of Brian Meade
Sent: Wednesday, February 19, 2020 9:29 PM
To: JASON BURWELL 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Join Webex Button on TOuch 10

You can dial any 9-digit meeting number now like 
123456...@webex.com.  Webex changed to make these 
non-site specific.  So it's easy to make an Extron/Crestron interface to enter 
a 9-digit meeting number than have it just add @webex.com and 
dial.

The other thing you can do is just have a button for 
m...@webex.com which prompts you to enter the 9-digit 
meeting number via DTMF.

On Tue, Feb 18, 2020 at 4:12 PM JASON BURWELL via cisco-voip 
mailto:cisco-voip@puck.nether.net>> wrote:
I noticed with CE9.10 they have now added a “Join Webex” button that appears on 
the Touch 10 home screen that allows direct calling to a webex meeting without 
dialing the entire URI. This is a great feature but now I’m trying to figure 
out if anyone is having luck integrating this Quick connect to webex meeting 
feature in to room systems that are Crestron controlled and do not have a Touch 
10 present. Sure would be nice to be able to be able to have this functionality 
from the Crestron control as well. My A/V vendor says they “are working on it” 
so it could be a while. Anyone had luck with this?

Thanks Jason
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Re: [cisco-voip] CER Configuration - Guides?

[Pawlowski, Adam]

Oh one more interesting thing

The data import/export for switches, ports, etc are going to be the real 
lifesaver in maintaining your data. I’ve found that the import will sometimes 
allow different character lengths, or will provide more useful information 
about something (such as how it refers to an interface on a switch, which may 
not be how you’d expect for TE versus GE, FA, etc).

I think they posted a bug for it but at some point someone here imported a 
switch in this valid addressing format: 192.168.1 . See Wikipedia’s comment on 
that:

When fewer than four numbers are specified in the address in dotted notation, 
the last value is treated as an integer of as many bytes as are required to 
fill out the address to four octets. Thus, the address 127.65530 is equivalent 
to 127.0.255.250.

CER happily imports this and the tracking engine expands this forever cursing 
you with it appearing in the things-not-tracked email until you spot it later 
in tables. Maybe it at least shows up in the UI now.

Adam




From: cisco-voip  On Behalf Of Johnson, Tim
Sent: Wednesday, February 19, 2020 1:34 PM
To: Matthew Loraditch ; Matt Taber (mtaber) 

Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] CER Configuration - Guides?

I just worked my way through the official configuration guide the other day. It 
was a challenge, but I got it pieced together. The call flow diagram on this 
TAC document would have been super helpful to have!

From: cisco-voip  On Behalf Of Matthew 
Loraditch
Sent: Wednesday, February 19, 2020 1:07 PM
To: Matt Taber (mtaber) 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] CER Configuration - Guides?

Thank you, that is super useful



Matthew Loraditch​

Sr. Network Engineer


p: 443.541.1518



w: www.heliontechnologies.com

 |

e: mloradi...@heliontechnologies.com


[Helion Technologies]


[Facebook]


[Twitter]


[LinkedIn]







From: Matt Taber (mtaber) mailto:mta...@cisco.com>>
Sent: Wednesday, February 19, 2020 1:01 PM
To: Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>>
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] CER Configuration - Guides?

[EXTERNAL]

This guide is a bit more streamlined, with configuration screenshots taken from 
an example deployment:
https://www.cisco.com/c/en/us/support/docs/unified-communications/emergency-responder/211453-Cisco-Emergency-Responder-Integration-wi.html

-Matt

On Feb 19, 2020, at 12:53 PM, Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>> 
wrote:

Any good guides or blogs that go through this in a more real world way than the 
documentation?

I haven’t found anything super helpful yet.


Matthew Loraditch​

Sr. Network Engineer


p: 443.541.1518



w: www.heliontechnologies.com

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e: mloradi...@heliontechnologies.com



















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Re: [cisco-voip] CER Configuration - Guides?

[Pawlowski, Adam]

Its really not so bad once you go through it once. The system is super super 
simplistic in what it does. The hard part is keeping your data up to date.

The guide makes some assumptions about your system and dialplan, but I found 
that I needed to lab it once and go through it before I started making changes, 
as a lot of things in the guide or elsewhere seem to make the assumption that 
you’re going to run everything configured as 911/912/913XX with a 
911CSS and that, and you just have to keep your head on straight to set it up 
otherwise.

I’ve not found any need to have explicit .911 route patterns as digit 
manipulation works just fine in our case. Similarly, I have partitions for 
internal and external service translations, and have ELINs sitting out there, 
external trigger mapping back in there (for elevators, other non on-system 
devices which we route through CER), and in internal the 911, 9.911, and other 
security triggers.

We no longer use CER to maintain NENA records for PS-ALI, and edit it directly 
ourselves, but it worked reasonably okay when we did use it in the past.

That guide is pretty close to what I wrote internally on how we set it up. I 
should try my own blog post or video on it but I’m sure I’ve been beaten to the 
punch about a hundred times over by now.

Adam

From: cisco-voip  On Behalf Of Johnson, Tim
Sent: Wednesday, February 19, 2020 1:34 PM
To: Matthew Loraditch ; Matt Taber (mtaber) 

Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] CER Configuration - Guides?

I just worked my way through the official configuration guide the other day. It 
was a challenge, but I got it pieced together. The call flow diagram on this 
TAC document would have been super helpful to have!

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Matthew Loraditch
Sent: Wednesday, February 19, 2020 1:07 PM
To: Matt Taber (mtaber) mailto:mta...@cisco.com>>
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] CER Configuration - Guides?

Thank you, that is super useful



Matthew Loraditch​

Sr. Network Engineer


p: 443.541.1518



w: www.heliontechnologies.com

 |

e: mloradi...@heliontechnologies.com


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[Facebook]


[Twitter]


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From: Matt Taber (mtaber) mailto:mta...@cisco.com>>
Sent: Wednesday, February 19, 2020 1:01 PM
To: Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>>
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] CER Configuration - Guides?

[EXTERNAL]

This guide is a bit more streamlined, with configuration screenshots taken from 
an example deployment:
https://www.cisco.com/c/en/us/support/docs/unified-communications/emergency-responder/211453-Cisco-Emergency-Responder-Integration-wi.html

-Matt

On Feb 19, 2020, at 12:53 PM, Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>> 
wrote:

Any good guides or blogs that go through this in a more real world way than the 
documentation?

I haven’t found anything super helpful yet.


Matthew Loraditch​

Sr. Network Engineer


p: 443.541.1518



w: www.heliontechnologies.com

 |

e: mloradi...@heliontechnologies.com



















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Re: [cisco-voip] 7832 out of the box - needs to be reset to register

[Pawlowski, Adam]

We used to get the 7900s out of the box with four digit extensions on them 
sometimes that I assumed was from some sort of QA or testing



From: Lelio Fulgenzi 
Sent: Tuesday, February 18, 2020 4:18 PM
To: Pawlowski, Adam ; voyp list, cisco-voip 
(cisco-voip@puck.nether.net) 
Subject: RE: 7832 out of the box - needs to be reset to register

Interesting. Well, it was in a Cisco box with tape that I had to cut open. Do 
you think it could have been a refurb? Yikes.

From: Pawlowski, Adam mailto:aj...@buffalo.edu>>
Sent: Tuesday, February 18, 2020 4:16 PM
To: Lelio Fulgenzi mailto:le...@uoguelph.ca>>; voyp list, 
cisco-voip (cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Subject: RE: 7832 out of the box - needs to be reset to register

Did it come out of the box not actually brand new with an ITL / CTL on it?

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Lelio Fulgenzi
Sent: Tuesday, February 18, 2020 4:09 PM
To: voyp list, cisco-voip 
(cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Subject: [cisco-voip] 7832 out of the box - needs to be reset to register


Any particular reason that I have to reset all settings on a 7832 before it 
registers?

Never seen that before.

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Re: [cisco-voip] 7832 out of the box - needs to be reset to register

[Pawlowski, Adam]

Did it come out of the box not actually brand new with an ITL / CTL on it?

From: cisco-voip  On Behalf Of Lelio 
Fulgenzi
Sent: Tuesday, February 18, 2020 4:09 PM
To: voyp list, cisco-voip (cisco-voip@puck.nether.net) 

Subject: [cisco-voip] 7832 out of the box - needs to be reset to register


Any particular reason that I have to reset all settings on a 7832 before it 
registers?

Never seen that before.

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Re: [cisco-voip] Official Cisco E911 Guidance

[Pawlowski, Adam]

Yes, I noted that language which was probably carefully vetted for liability.

They also were careful about recommendations, suggestions, etc to avoid “Cisco 
said”.

There’s some interpretation that seems to be around “improvements”, but, I 
believe some of that is taking that word at face value and not referring back 
to the FCC’s clarification on it.

Certainly going to be a hot topic of discussion for a minute (and was in some 
circles leading up) depending on liability.

Still careful too to say sure we can monitor Jabber (*on wifi)

From: cisco-voip  On Behalf Of Anthony 
Holloway
Sent: Tuesday, February 18, 2020 3:54 PM
To: Matthew Loraditch 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Official Cisco E911 Guidance

I found this interesting:

Q: Do I need to upgrade UCM to be compliant?
A: No! Existing UCM deployments are configurable for Kari’s Law.

...

We strongly suggest that Cisco UCM customers evaluate other emergency safety 
add-ons through RedSky, Intrado, and Singlewire

On Tue, Feb 18, 2020 at 2:04 PM Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>> 
wrote:
This was just published:
https://blogs.cisco.com/collaboration/saying-yes-to-workplace-safety-how-ucm-customers-become-compliant-with-karis-law-and-ray-baums-act

https://www.cisco.com/c/dam/en/us/products/collateral/unified-communications/unified-communications-manager-callmanager/q-and-a-c67-743415.pdf








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Re: [cisco-voip] CUC 6 Second Intervals of Silence

[Pawlowski, Adam]

Interesting. We've never had much of a complaint about this and I don't recall 
hearing much of anything when listening to customer recordings. Some of them 
seem like they record their messages on their cell phone in a shower stall 
anyways and silence would be welcome.


If I didn't have other fires I'd be interested to see if that's present in our 
environment, as we're on 12.5 latest as well. Since Jabber doesn't use an RTP 
stream I wouldn't have heard it from there - I'll try and take a cap off a 
stream to a phone if I can.


As far as Reddit, there can be some interesting or useful discussions there, 
and it is more visible to the general public than this mailing list. That being 
said, I find it tends to attract noise and unhelpful replies because of that, 
and can be cumbersome to read. Since I'm buried in email all day I appreciate 
this forum over having to be on Twitter and Reddit in addition to e-mail, IRC, 
teams, the other teams, etc.



From: cisco-voip  on behalf of Joshua 
Lamont 
Sent: Sunday, February 16, 2020 12:38 PM
To: Lelio Fulgenzi
Cc: Cisco VoIP Group
Subject: Re: [cisco-voip] CUC 6 Second Intervals of Silence

Signing up is definitely recommend since it is required for commenting and 
replying.

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

On Feb 16, 2020, at 10:40 AM, Lelio Fulgenzi  wrote:

?

Looks like a good forum with interesting posts. Will have to consider following.

Do you have to sign up to get post notifications?

Sent from my iPhone

On Feb 15, 2020, at 12:08 AM, Anthony Holloway 
mailto:avholloway+cisco-v...@gmail.com>> wrote:

There's post on the ciscoUC subreddit about CUC greetings sounding "garbled" 
and it turns out there is a small bit of silence being inserted on the wire by 
CUC.  Has anyone here experienced this before?  This has been confirmed on 
11.0, 11.5, 12.0 and 12.5 so far.

https://www.reddit.com/r/ciscoUC/comments/f3xi5m/unity_connection_inserts_silence_every_6_seconds/


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Re: [cisco-voip] Webex calling / flex sku’s

[Pawlowski, Adam]

So, yes, in doing this research flex is a bit interesting. 

Early EAs had different restrictions on license quantity, which can impact a 
flex purchase. 1:1 may not be possible based on the license allowance ratios in 
Flex for anonymous phones or general mailboxes. 

It's also all subscription so your licenses end with the sub , not that anyone 
in enterprise should be floating support but that's not a game that seems that 
you can play with Flex. That just means processing budgets to re-up sooner, or, 
if you're looking at other solutions, getting that process rolling faster since 
you lose entitlement when you stop paying.

And yeah we have A-SPK -EDU for Webex/Teams meetings which is different than 
the public sector enterprise for sure. 

What I will say with Flex is that if you can meet minimums for service, you can 
up quantities within the term of the agreement, so you should be able to add 
"more" of something you already have without re-ordering the whole thing. You 
can't back down though I believe. I'm going to have misspoken about a lot of 
this since I'm an end user, we don't work with a partner, but our Cisco account 
team is very helpful in getting these answers when I come up with crazy 
questions - I hope yours is as well.

Adam

> -Original Message-
> From: Lelio Fulgenzi 
> Sent: Friday, February 14, 2020 8:44 AM
> To: Pawlowski, Adam 
> Cc: cisco-voip voyp list 
> Subject: Re: Webex calling / flex sku’s
> 
> Thanks! We only have a short time frame, so, based on my quick research, I
> think I may postpone the webex calling portion and only go for making our 
> trial
> webex site full fledged, matching our production site.
> 
> I’ll have some time to work out what our production webex calling and cc agent
> solution would look like, then get that going on trial first.
> 
> I’m hoping we can get them to give us spk-edu for a short period of time, co-
> term with our production site.
> 
> Interesting thing about the public sector SKUs, they don’t include the “how
> many student licenses do you need?” Question like the edu sku does. Not sure
> how they work that in.
> 
> Our EA comes due next year. I’m sure there will be a lot of stuff that changes
> (again). I wish we had licensed the trial site earlier.
> 
> One thing I know, is that more than likely we will move to flex.
> 
> Lelio
> 
> 
> Sent from my iPhone
> 
> > On Feb 14, 2020, at 8:37 AM, Pawlowski, Adam  wrote:
> >
> > We're actually looking to do something similar here. While obv a Cisco
> account contact should be able to help, I also like to know what I'm dealing
> with. The Webex Calling datasheets describe different tiers, including
> basic/enterprise, but under the EA the basic doesn't exist I guess, or at 
> least I
> got a warning from CCW playing with it.
> >
> > List pricing puts Webex Calling at almost 100% over premise calling, and
> UCM Cloud is like 80% more - both of those numbers are hard to reconcile.
> >
> > Based on CCW the enterprise calling is A-FLEX-EACL2  if you're doing this
> under A-FLEX-PUBLICSECT . I know you mentioned A-SPK-EDU but I don't know
> if that part is still available or not, they've made a lot of changes in the 
> last few
> years.
> >
> >
> >
> >> -Original Message-
> >> From: cisco-voip  On Behalf Of
> >> Lelio Fulgenzi
> >> Sent: Thursday, February 13, 2020 5:39 PM
> >> To: cisco-voip voyp list 
> >> Subject: [cisco-voip] Webex calling / flex sku’s
> >>
> >>
> >> I’m trying to do a last minute pitch for a full fledged lab
> >> environment for webex cloud : mc/ec/tc/sc, Teams, webex calling w/pstn,
> vm (, and call center).
> >>
> >> I know it’s not always as simple as top level sku’s but I’m wondering
> >> if anyone can shed some light.
> >>
> >> I’m gonna use A-SPK-EDU to start, it’s still the right one in our
> >> case I believe. I want to make sure I’m not getting Spark Call, but the
> broadsoft webex calling.
> >>
> >> Any pointers?
> >>
> >> Sent from my iPhone
> >> ___
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> >> https://puck.nether.net/mailman/listinfo/cisco-voip
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Re: [cisco-voip] Webex calling / flex sku’s

[Pawlowski, Adam]

We're actually looking to do something similar here. While obv a Cisco account 
contact should be able to help, I also like to know what I'm dealing with. The 
Webex Calling datasheets describe different tiers, including basic/enterprise, 
but under the EA the basic doesn't exist I guess, or at least I got a warning 
from CCW playing with it.

List pricing puts Webex Calling at almost 100% over premise calling, and UCM 
Cloud is like 80% more - both of those numbers are hard to reconcile. 

Based on CCW the enterprise calling is A-FLEX-EACL2  if you're doing this under 
A-FLEX-PUBLICSECT . I know you mentioned A-SPK-EDU but I don't know if that 
part is still available or not, they've made a lot of changes in the last few 
years.



> -Original Message-
> From: cisco-voip  On Behalf Of Lelio
> Fulgenzi
> Sent: Thursday, February 13, 2020 5:39 PM
> To: cisco-voip voyp list 
> Subject: [cisco-voip] Webex calling / flex sku’s
> 
> 
> I’m trying to do a last minute pitch for a full fledged lab environment for 
> webex
> cloud : mc/ec/tc/sc, Teams, webex calling w/pstn, vm (, and call center).
> 
> I know it’s not always as simple as top level sku’s but I’m wondering if 
> anyone
> can shed some light.
> 
> I’m gonna use A-SPK-EDU to start, it’s still the right one in our case I 
> believe. I
> want to make sure I’m not getting Spark Call, but the broadsoft webex calling.
> 
> Any pointers?
> 
> Sent from my iPhone
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Re: [cisco-voip] network roaming and Jabber - does it work?

[Pawlowski, Adam]

It does not in my experience.

If you roam and change IP addresses Jabber dies or the call is lost. If you 
connect through a VPN client and that reconnects fast enough then it maybe 
works.

I believe that's one of the things that maybe comes in the future is some sort 
of call anchoring.

The documentation I think recommends mobility and handing the call to your 
mobile number when you plan to go off net. If you skip the complication of SNR, 
you will run into people perhaps who have it configured and are using Jabber 
mobile so it will ring twice and they won't get either call.

The "Transition" to dual mode is a delete and re-do from the self care portal. 
Not the smoothest, but serviceable.

Adam

From: cisco-voip  On Behalf Of Lelio 
Fulgenzi
Sent: Monday, February 10, 2020 12:07 PM
To: voyp list, cisco-voip (cisco-voip@puck.nether.net) 

Subject: [cisco-voip] network roaming and Jabber - does it work?


One of the things we sort of expected to work was Jabber's ability to roam from 
wifi to cellular data and vice versa, whether it was going to/from on-prem to 
off-prem, or off-prem to off-prem.

It's something we still have to test and document accordingly for our clients, 
but I was hoping to get a running start with information from the group.

Of course, things are complicate when you insert SNR, but I'm going to leave 
that out for now.

Does this work? Or is this not a supported roaming feature?
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Re: [cisco-voip] 8851 - power draw

[Pawlowski, Adam]

Sorry for the delay, been a busy morning.

Hardware revs changed at some point including the device’s PoE class to go with 
it.

This is from my notes:

Phone   # of KEM  PoE? (802.3at)
8851   1 BEKEM or AKEM   Yes
8851   2 AKEMNo
8851   2 BEKEM  Yes* Only with v08 or later 
hardware - see back of phone
8861   1 BEKEM or AKEM   Yes
8861   2 BEKEM or AKEM   Yes, but USB Fast Charging is disabled on 
rear port
8861   3 BEKEM or AKEM   Yes, but USB Fast Charging is disabled on 
rear port

Anyways here are some power stats:

Hardware V01 with a BEKEM

Interface: Gi2/0/23
Inline Power Mode: auto
Operational status: on
Device Detected: yes
Device Type: Cisco IP Phone 8851
IEEE Class: 3
Discovery mechanism used/configured: Ieee and Cisco
Police: off

Power Allocated
Admin Value: 60.0
Power drawn from the source: 12.6
Power available to the device: 12.6

Actual consumption
Measured at the port: 7.4
Maximum Power drawn by the device since powered on: 8.3

Hardware V01 with no KEM

Interface: Te4/0/46
Inline Power Mode: auto
Operational status: on
Device Detected: yes
Device Type: Cisco IP Phone 8851
IEEE Class: 3
Discovery mechanism used/configured: Ieee and Cisco
Police: off

Power Allocated
Admin Value: 60.0
Power drawn from the source: 9.2
Power available to the device: 9.2

Actual consumption
Measured at the port: 3.4
Maximum Power drawn by the device since powered on: 5.4

Hardware V31 with a BEKEM

Interface: Gi2/0/20
Inline Power Mode: auto
Operational status: on
Device Detected: yes
Device Type: Cisco IP Phone 8851
IEEE Class: 4
Discovery mechanism used/configured: Ieee and Cisco
Police: off

Power Allocated
Admin Value: 60.0
Power drawn from the source: 12.3
Power available to the device: 12.3

Actual consumption
Measured at the port: 7.8
Maximum Power drawn by the device since powered on: 8.4

Hardware V31 with an A-KEM

Interface: Gi1/0/4
Inline Power Mode: auto
Operational status: on
Device Detected: yes
Device Type: Cisco IP Phone 8851
IEEE Class: 4
Discovery mechanism used/configured: Ieee and Cisco
Police: off

Power Allocated
Admin Value: 60.0
Power drawn from the source: 12.3
Power available to the device: 12.3

Actual consumption
Measured at the port: 7.0
Maximum Power drawn by the device since powered on: 7.4

Hardware V31 with no KEM

Interface: Gi1/0/15
Inline Power Mode: auto
Operational status: on
Device Detected: yes
Device Type: Cisco IP Phone 8851
IEEE Class: 4
Discovery mechanism used/configured: Ieee and Cisco
Police: off

Power Allocated
Admin Value: 60.0
Power drawn from the source: 8.9
Power available to the device: 8.9

Actual consumption
Measured at the port: 3.7
Maximum Power drawn by the device since powered on: 4.7






From: cisco-voip  On Behalf Of Terry Oakley
Sent: Friday, February 7, 2020 10:39 AM
To: Lelio Fulgenzi ; Biffle, Gerrad 

Cc: Cisco-VOIP 
Subject: Re: [cisco-voip] 8851 - power draw

On a quick test our 8851 draws 15.4 on boot and settles back to 9.8 at rest.

Terry

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Lelio Fulgenzi
Sent: Friday, February 7, 2020 7:47 AM
To: Biffle, Gerrad 
mailto:gerrad.bif...@greensboro-nc.gov>>
Cc: Cisco-VOIP mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] 8851 - power draw

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.

Hey Thanks! This does help.

Do you know if the ones drawing more than 9.4 have a KEM?

From: Biffle, Gerrad 
mailto:gerrad.bif...@greensboro-nc.gov>>
Sent: Friday, February 7, 2020 9:44 AM
To: Lelio Fulgenzi mailto:le...@uoguelph.ca>>
Cc: Cisco-VOIP mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] 8851 - power draw

Don’t know if you still need it or not.  Here is a quick grab from one of our 
switches:

Interface Admin  OperPower(Watts) Device  Class
From PSTo Device
- -- -- -- -- --- -

Gi2/17auto   on 14.5   13.8   IP Phone 8851   3
Gi2/18auto   on 9.48.9IP Phone 8851   4
Gi2/19auto   on 12.9   12.3   IP Phone 8851   4
Gi2/20auto   on 9.48.9IP Phone 8851   4
Gi2/21auto   off0.00.0n/a n/a
Gi2/22auto   off0.00.0n/a n/a
Gi2/23auto   on 9.48.9IP Phone 8851   4
Gi2/24auto   on 10.3   9.8IP Phone 8851   3
Gi2/25auto   on 10.3   9.8IP Phone 8851   3
Gi2/26auto   on 9.48.9IP Phone 8851   4
Gi2/27auto   on 6.66.3IP Phone 7962   2
Gi2/28auto   on 9.48.9IP Phone 8851  

Re: [cisco-voip] 8851 - power draw

[Pawlowski, Adam]

Sure, I can do that for you in the morning.


Check the datasheet on the KEM as well - those draw different power. The A-KEM 
is supported with 1 per, BEKEM with 2 (big easy is BE, code name for the model).


If you're running gear with a low budget, "show power inline  
detail" tells you what it's actually drawn, in case you need to dip into manual 
policing. The LLDP-MED/CDP power request values can be higher than what it 
really needs.


From: cisco-voip  on behalf of Lelio 
Fulgenzi 
Sent: Thursday, February 6, 2020 6:11 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] 12.5 Upgrade files posted on CCO

[Pawlowski, Adam]

As far as I know, ordering from PUT creates a sales order number that is used 
for entitlement when migrating your licenses. It also gives you a bootable ISO 
or should.



From: cisco-voip  On Behalf Of Nick via 
cisco-voip
Sent: Thursday, February 6, 2020 11:20 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] 12.5 Upgrade files posted on CCO

The upgrade files for CUCM 12.5 both SU1 and SU2 both state the following

For upgrades from 12.x only. Upgrades from 11.x or earlier are requested via PUT

Is this just incorrect wording as far as i am aware there has never been any 
different files for upgrades from 11 or 12, I have ordered 12.5.1 SU1 from PUT 
and the upgrade file is identical to the one you can download from CCO 
supposedly for 12 only?

Anyone able to confirm these are the same files?
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Re: [cisco-voip] CUCM requirements for AD account import - anything else other than SN=* (non-empty) ?

[Pawlowski, Adam]

We also map mail to directoryURI , and enable IM automatically, so I added a 
concat (&(mail=*)) to ensure we don’t sync people without a URI.


From: cisco-voip  On Behalf Of Lelio 
Fulgenzi
Sent: Thursday, January 30, 2020 6:45 PM
To: Hunter Fuller 
Cc: Charles Goldsmith ; voyp list, cisco-voip 
(cisco-voip@puck.nether.net) 
Subject: Re: [cisco-voip] CUCM requirements for AD account import - anything 
else other than SN=* (non-empty) ?

Using the ipPhone field is an option. However, it’s a bit of a chicken and egg 
thing. They don’t have an extension until we assign it. And we can’t import 
them until they have the number in that field.

So we’d have to assign them an extension, modify the attribute, then sync 
manually, then import, etc.

There are some other scenarios which still wouldn’t help though.

It’s a drag. ☹

But I’m still not defeated!

From: Hunter Fuller mailto:hf0...@uah.edu>>
Sent: Thursday, January 30, 2020 6:26 PM
To: Lelio Fulgenzi mailto:le...@uoguelph.ca>>
Cc: Charles Goldsmith mailto:w...@woka.us>>; voyp list, 
cisco-voip (cisco-voip@puck.nether.net) 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] CUCM requirements for AD account import - anything 
else other than SN=* (non-empty) ?

On Thu, Jan 30, 2020 at 4:51 PM Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:
We can’t filter on anything telephone number based. Sounds silly, but the 
information in the directory doesn’t always jive with what someone wants, 
extension wise.

This was true for us. Our solution was to populate the ipPhone field with their 
real phone number and populate telephoneNumber with what they wanted to appear 
in the directory (e.g., their admin assistant, or for some people in IT it's 
the help desk, etc. etc.). Then UCM/Unity use the ipPhone attribute for 
provisioning and all is well in the world.
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[cisco-voip] Webex Assistant

[Pawlowski, Adam]

Is anyone successfully using the Webex Assistant for all the cool features?

I've been testing this on a Kit Plus, and haven't had that much luck. It does 
offer you to say "join the meeting" for a OBTP scheduled meeting, but, then I 
get asked on-screen if I'm the host and the assistant responds to yes/no as 
though the question is still "Do you want to join the meeting", so we can't get 
by it.

Asking it to call anyone results in "Something went wrong", and asking it what 
it can do gives us an on screen display with an arrow to page over that isn't 
represented with any control to be able to scroll it.

This doesn't seem like it is supposed to behave this way, but, if it is going 
to then I'd rather turn it off than have a customer yelling into the box and 
getting frustrated.

Adam


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Re: [cisco-voip] Expressway Cluster failover for MRA...

[Pawlowski, Adam]

Is that A record needed for some function or another of Jabber to operate? 

We have two equally weighted SRV to our E and it will fail over to the other 
host when it's not available, depending on the service, if it chooses to 
operate. The client periodically re-looks up the SRV record anyways in my 
experience. The failover scenario isn't instant/clean anyways with Jabber but 
if it does improve timing on something that would be good to know.

Adam



> -Original Message-
> From: cisco-voip  On Behalf Of Ryan
> Huff
> Sent: Tuesday, January 28, 2020 9:04 PM
> To: Jonathan Charles 
> Cc: cisco-voip@puck.nether.net
> Subject: Re: [cisco-voip] Expressway Cluster failover for MRA...
> 
> 1.) It used to be in previous versions that all cluster nodes could 
> technically be
> active at any time and SRV weights and priorities could influence the path
> selection but not guarantee it end-to-end when all cluster nodes are up and
> running.
> 
> I believe this behavior has changed/improved and I think you are supposed to
> be able to control that now with SRV weights and priorities, but I could be
> wrong. I haven’t played with Expressway clustering in a bit.
> 
> 2.) As far as the Jabber registration goes; what I’ve done before in the edge 
> is
> have the collab-edge SRV point to the edge cluster FQDN as the target. Then I
> create round robin A records for the cluster FQDN (one resolving your each
> edge server). The for the edge certs, just make sure the edge cluster fqdn is 
> in
> the SAN.
> 
> This way if one of the edge server goes down, the Jabber client is ultimately
> still trying to resolve the same MRA FQDN via SRV lookup (this a key to Jabber
> client failover for MRA).
> 
> Thanks,
> 
> Ryan
> 
> > On Jan 28, 2020, at 20:50, Jonathan Charles  wrote:
> >
> > 
> > We have two pairs of Expressway clusters (C/E) at two different locations
> (primary and DR)...
> >
> > The cluster is up, however, we want to make sure that we are in
> Active/Standby.
> >
> > Currently, we have one of our SRV records for collab-edge set at 5 (the
> backup is at 10) with the same weight.
> >
> > The clustering guide says we should set the priority and weight on both SRV
> records the same, which will cause half of the registrations to go to the DR 
> site.
> It is far away and has less capability.
> >
> > How do we:
> >
> > 1 - Make sure the primary site handles all MRA registrations and the DR site
> is only used when the primary is down.
> > 2 = Make sure failover occurs automatically... currently Jabber users have 
> > to
> log out and back in to connect to the DR site.
> >
> >
> > Thanks!
> >
> >
> > Jonathan
> >
> > ___
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> > cisco-voip@puck.nether.net
> >
> https://nam04.safelinks.protection.outlook.com/?url=https%3A%2F%2Fpuck.net
> her.net%2Fmailman%2Flistinfo%2Fcisco-
> voipdata=02%7C01%7C%7C2f536d8162984707853908d7a45d8e24%7C8
> 4df9e7fe9f640afb435%7C1%7C0%7C637158594035084563
> p;sdata=atRtIR8sWZ60Ja8akD6GjzBIgBNC8GSJjaOmu%2BTxmWw%3Dres
> erved=0
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Re: [cisco-voip] CCX phone agent over MRA?

[Pawlowski, Adam]

From experience with xml services, if uses a post to push the phone to open XML 
objects, then, no that won’t work. You’ll either see it as the Expressway C’s 
IP like you said which doesn’t work as it can’t forward anything back, or if 
you chase the X-Forwarded-For header you get the device’s IP which also 
probably doesn’t help you, unless you’re using the Expressway to secure 
telephony or proxy it on-net somewhere you have direct access to.



From: cisco-voip  On Behalf Of Anthony 
Holloway
Sent: Thursday, January 23, 2020 3:30 PM
To: Lelio Fulgenzi 
Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net) 

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

Are you talking Finesse IP Phone Agent (FIPPA)?

If so, the below enhancement defect requesting that these types of details be 
documented (I mean should we even have to request that?) states that they 
tested FIPPA via MRA and it worked.

https://bst.cloudapps.cisco.com/bugsearch/bug/CSCvi51697

Just know that you'll have to add your UCCX server addresses to the HTTP Allow 
list on Expressway-C.

And this makes sense to me, since FIPPA is stateless and all needed information 
is included in the URL to perform the actions like Login, Logout, Reason Codes, 
Ready, Not Ready, etc.   The actual ringing of the phone and answering etc., 
are just phone functions, which we know works over MRA.  That's kind of the 
point.  ;)

What I am not sure of is whether the FIPPA push to phone works, if you're even 
using that; wherein, upon a new call, UCCX attempts to push content to the 
Agent's phone using the Phone API, but I would think, though I cannot confirm, 
that this would fail, since the phone IP is actually like 192.168.1.1 or 
something, and UCCX wont know to contact Expressway-C about it, nor would 
Expressway-C forward the API call on to the phone, etc.

Finesse itself, the web app on port 8445, would not be available over MRA, as 
the document states, and would require a VPN or other networking solution to be 
available to the Agent.  Brian Meade 
commented on a 
previous conversation to a similar topic that a reverse proxy would help in 
this scenario.

On Thu, Jan 23, 2020 at 2:07 PM Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:

Can anyone say whether or not a CCX phone agent (or finesse agent in the 
future) is supported over MRA?

The MRA guides say:

The Expressway does not support some Cisco Unified Contact Center Express 
(Unified CCX) features for contact center agents or other users who connect 
over MRA. Jabber for Mac and Jabber for Windows cannot provide deskphone 
control over MRA, because the Expressway pair does not traverse the CTI-QBE 
protocol. However, if these Jabber applications, or other CTI applications, can 
connect to Unified CM CTIManager (directly or through the VPN) they can provide 
deskphone control of MRA-connected clients.

We're looking at a simple phone agent setup, no desktop agent/control, etc.

Thoughts?
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Re: [cisco-voip] Jabber and locked screens - it works! but should it?

[Pawlowski, Adam]

The biggest issue with this that came up here during piloting with screen lock, 
is that a lock policy went into place during the pilot rollout. We'd been 
migrating test users with Plantronics headsets with the DA70, that has no 
buttons, and they found if they'd not been jiggling the computer mouse after a 
few minutes they can't answer their phone without logging back into the 
workstation. 

The DA80 has the buttons so you can answer it, but you can't see who's calling 
anymore. 

I'd always hoped that there'd be a tie in to a lock screen widget so you could 
see something but, alas. There's not as much of a coordination between the need 
for lock policy, user training, system idle, smartcard/key/windows hello etc 
and it represents a change in workflow that is still not accepted by most users 
when it comes to "telephone" because they've been able to pick up and talk into 
the blower for years without any extra steps.

Adam

> -Original Message-
> From: cisco-voip  On Behalf Of Bill
> Talley
> Sent: Thursday, January 23, 2020 10:12 AM
> To: Lelio Fulgenzi 
> Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net)  v...@puck.nether.net>
> Subject: Re: [cisco-voip] Jabber and locked screens - it works! but should it?
> 
> Might be handy for calling emergency services or security, but other than 
> that,
> it’s probably not something I would want the cleaning crew or mischievous co-
> workers being able to do.
> 
> Sent from an iPhone mobile device with very tiny touchscreen input keys.
> Please excude my typtos.
> 
> > On Jan 23, 2020, at 8:40 AM, Lelio Fulgenzi  wrote:
> >
> > 
> > OK. What do people think about Jabber working while a screen is locked? By
> this, I mean that if your headset has a "pickup" button, you can answer the 
> call.
> Now, I know the first thing you will say is, "what's different from the phone
> being picked up", and really, there isn't, but there is. I think people will 
> assume
> that if my computer is locked, it should not be allowed to work. OR at a
> minimum, allow it to be a configurable option.
> >
> > Then there's the possibility of using a handset that is Jabber compatible 
> > which
> could be used to dial numbers while the computer is locked.
> >
> > Thoughts?
> > 
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Re: [cisco-voip] Jabber 12.8 dropped

[Pawlowski, Adam]

I only see the part where it says for best results wait for it to load, then it 
turns into a grey screen, on anything except meetings.

I don’t remember any information about this becoming available from the EAP or 
other info.

Would be interesting to see, since pulling stats from XCP router counters and 
that is not all that great.



From: cisco-voip  On Behalf Of Lelio 
Fulgenzi
Sent: Wednesday, January 22, 2020 2:42 PM
To: Anthony Holloway 
Cc: cisco-voip voyp list 
Subject: Re: [cisco-voip] Jabber 12.8 dropped

Looks like the Jabber tab in control hub portal is available in rolling 
fashion. I still don’t see it. :(
-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

[University of Guelph Cornerstone with Improve Life tagline]

On Jan 21, 2020, at 11:48 PM, Anthony Holloway 
mailto:avholloway+cisco-v...@gmail.com>> wrote:
It's about time. Data analytics should be right up their in priorities with 
APIs.

On Tue, Jan 21, 2020, 10:02 PM Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:

Ok. I thought park was the coolest new feature. But I think it’s Jabber 
telemetry in control hub.

https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/jabber/12_8/cjab_b_feature-configuration-for-jabber-128/cjab_b_feature-configuration-for-jabber-128_chapter_0100.html#reference_B0BFF8B1E538B6FF6E1137CA4D28CE6D



-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

[University of Guelph Cornerstone with Improve Life tagline]

On Jan 21, 2020, at 10:06 PM, Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:

Looks like Jabber 12.8 for iPhone, iPad and windows has dropped.

Haven’t checked android (battery dead). Mac still not listed.

Call Park now supported on desktop. Woot!

Funny thing is... it’s Jan 21, but release day for Windows is listed as Jan 22. 
:O

-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

[University of Guelph Cornerstone with Improve Life tagline]
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Re: [cisco-voip] setting primary phone for Jabber users without a physical phone - licensing

[Pawlowski, Adam]

There is a getUniversalDeviceTemplate and addUniversalDeviceTemplate call you 
can use so you should be able to pull the record, manipulate the values, and 
put it back, with respect to how you’d do that with anything else via AXL.

I may try and play with this a bit myself later to see if it’s possible, or at 
least quickly “Report” on the UDTs and their settings for consistency, since 
the weird expanding interface isn’t great.



From: cisco-voip  On Behalf Of Tucci, Ben 
via cisco-voip
Sent: Tuesday, January 21, 2020 12:37 PM
To: voyp list, cisco-voip (cisco-voip@puck.nether.net) 

Subject: Re: [cisco-voip] setting primary phone for Jabber users without a 
physical phone - licensing

I was able to do a simplified version of that on a smaller table – we had to 
build an app internally for this due to 
CSCuz76525<https://bst.cloudapps.cisco.com/bugsearch/bug/CSCuz76525> (which has 
happened a few times to us). The AXL side requires that you send it all of the 
devices to add which isn’t ideal if we have many users trying to make changes. 
So we just call this insert from a separate web app for our admin users.

It’s a mix of data sources so it’s possible the syntax error is something in 
the quotes used or something else. I’m leaving the IDs in but they would need 
to be made specific.

insert into applicationuserdevicemap (pkid, fkdevice, fkapplicationuser, 
tkuserassociation) select newid(), pkid, 
'b11187e9-ba5d-aeeb-5afa-1950b50f99e9', 1 from device where name = 
'SEP204C9E6C71A0'

Those IDs are also from a lab environment we set up, and I would be very 
careful with these changes as you can insert a record ccmadmin can’t display 
and cause quite a panic. This was run on 10.5.2.


From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Anthony Holloway
Sent: Tuesday, January 21, 2020 12:05 PM
To: Pawlowski, Adam mailto:aj...@buffalo.edu>>
Cc: voyp list, cisco-voip 
(cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] setting primary phone for Jabber users without a 
physical phone - licensing

Disclaimer:  Please test in a lab thoroughly before playing cowboy/girl and 
just modifying the database willy-nilly.

I was able to pull the SQL insert from CCMAdmin tomcat logs when adding a new 
template from the GUI, and then use the syntax to insert my own records at the 
CLI.  However, using the below structure to copy an existing record fails, with 
a syntax error, which is a real shame.

admin:run sql insert into device (pkid, name, col1, col2, colN...) select 
newid(), concat(name, " - copy"), col1, col2, colN... where name = 'the name 
you want to copy'

Where col1, col2, and colN are the required columns needed to properly insert a 
new record (again, taken from CCMAdmin traces upon inserting a new template via 
the GUI).

The idea is to insert a new record, specifying all of the proper column names 
(e.g., pkid, name, fkdevicepool, etc.), while dynamically generating a new 
PKID, and appending " - copy" to the name of the record I want to copy.

According to some documentation, that command syntax should work in Informix.  
I tried a few other variations, but I can't figure it out.  Can anyone else 
figure it out?  Could be a useful construct for copying just about anything in 
the DB.

On Tue, Jan 21, 2020 at 10:16 AM Pawlowski, Adam 
mailto:aj...@buffalo.edu>> wrote:
They are in the database and you can pull them up as a phone if you nav to edit 
them with their record’s pkid as the key. I haven’t tried inserting one there 
but it’s in there.

I bet it would work as well as any of that does otherwise.




From: Lelio Fulgenzi mailto:le...@uoguelph.ca>>
Sent: Monday, January 20, 2020 1:58 PM
To: Lelio Fulgenzi mailto:le...@uoguelph.ca>>; Pawlowski, 
Adam mailto:aj...@buffalo.edu>>
Cc: voyp list, cisco-voip 
(cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Subject: RE: [cisco-voip] setting primary phone for Jabber users without a 
physical phone - licensing


Ok, so I’m finding a few more issues with these templates. I have a case open 
for more details. But, I’m likely going to have to live with them.

But what I don’t like? What I really don’t like?

There’s no way to copy universal templates. ARGH.

There’s not even a way to export them and then import them.

Why?!?!




From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Lelio Fulgenzi
Sent: Wednesday, January 15, 2020 8:56 PM
To: Pawlowski, Adam mailto:aj...@buffalo.edu>>
Cc: voyp list, cisco-voip 
(cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] setting primary phone for Jabber users without a 
physical phone - licensing

Sort of like how all the icons/graphics for ccmuser pages are brok

Re: [cisco-voip] setting primary phone for Jabber users without a physical phone - licensing

[Pawlowski, Adam]

They are in the database and you can pull them up as a phone if you nav to edit 
them with their record’s pkid as the key. I haven’t tried inserting one there 
but it’s in there.

I bet it would work as well as any of that does otherwise.




From: Lelio Fulgenzi 
Sent: Monday, January 20, 2020 1:58 PM
To: Lelio Fulgenzi ; Pawlowski, Adam 
Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net) 

Subject: RE: [cisco-voip] setting primary phone for Jabber users without a 
physical phone - licensing


Ok, so I’m finding a few more issues with these templates. I have a case open 
for more details. But, I’m likely going to have to live with them.

But what I don’t like? What I really don’t like?

There’s no way to copy universal templates. ARGH.

There’s not even a way to export them and then import them.

Why?!?!




From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Lelio Fulgenzi
Sent: Wednesday, January 15, 2020 8:56 PM
To: Pawlowski, Adam mailto:aj...@buffalo.edu>>
Cc: voyp list, cisco-voip 
(cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] setting primary phone for Jabber users without a 
physical phone - licensing

Sort of like how all the icons/graphics for ccmuser pages are broken for new 
phone types added with devpacks. Logs show nothing but errors. Opened up case, 
told it’s cosmetic.


-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<http://www.uoguelph.ca/ccs> | @UofGCCS on Instagram, 
Twitter and Facebook

[University of Guelph Cornerstone with Improve Life tagline]

On Jan 15, 2020, at 3:33 PM, Pawlowski, Adam 
mailto:aj...@buffalo.edu>> wrote:
The UI broke at some point and none of the proper localization fills in, so I 
guess get used to error.add .

It seems to use AXL or something similar in the background, and you can debug 
what the error is from the application log in RTMT, though half the time it’s a 
code and not useful.

Duplicate items will break it, as will things that are not valid for insert but 
are part of your universal templates, such as #DEPT# and someone has a 
department name with & in it.

As for the custom templates, they’re one of those once-a-year things I clean up 
with the database by assigning the phonetemplate record to the type, and then 
removing any of the phone templates that aren’t associated with anything.

I used to say they don’t hurt anything but if you invest in MigrationFX, it 
will leverage the phone button template as a selector to help you choose the 
correct migration. Unless you want it to clone SEPasdfghjkl-Invididual 
Template-8841-MFX all over, you’ll want these to be consistent.


From: Lelio Fulgenzi mailto:le...@uoguelph.ca>>
Sent: Wednesday, January 15, 2020 3:28 PM
To: Pawlowski, Adam mailto:aj...@buffalo.edu>>; Anthony 
Holloway 
mailto:avholloway+cisco-v...@gmail.com>>
Cc: voyp list, cisco-voip 
(cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Subject: RE: [cisco-voip] setting primary phone for Jabber users without a 
physical phone - licensing


If you really want to hear where it falls short, check this out:


  *   Create jabber phone with tool.
  *   It creates phone with custom phone button template.
  *   I don’t like it, so I change it to our standard.
  *   I want to test things again, so I delete phone and try to create again.
  *   It doesn’t work. It fails with the most descriptive error ever: 
“Error.add”
  *   TAC couldn’t figure it out.

Turns out, if you delete a phone with a custom phone button template, it will 
delete the template (if not in use elsewhere?). But because I changed the 
template to our standard template, the custom template remained.

When I tried to create the phone again using the tool, it tried to recreate the 
custom template, but because it already existed, it failed. ERROR.ADD

Ugh. UI 101….. better error messages!


From: Pawlowski, Adam mailto:aj...@buffalo.edu>>
Sent: Wednesday, January 15, 2020 3:03 PM
To: Lelio Fulgenzi mailto:le...@uoguelph.ca>>; Anthony 
Holloway 
mailto:avholloway+cisco-v...@gmail.com>>
Cc: voyp list, cisco-voip 
(cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Subject: RE: [cisco-voip] setting primary phone for Jabber users without a 
physical phone - licensing

Yes, it’s “almost” there, but falls short on those things you note.

But, it is at least consistent in what it does or doesn’t do, so you know what 
you have to fix or clean up.



From: Lelio Fulgenzi mailto:le...@uoguelph.ca>>
Sent: Wednesday, J

Re: [cisco-voip] Jabber mobile with shared primary extension

[Pawlowski, Adam]

You can log in and out it's just from all groups.

For a while the hunt group alerting names did not display on mobile, which it 
should now for android/ios. A recent change to the way those work in a recent 
SU seemed to break them again for some phones and required you to put the 
alerting name in " " for it to display on the device.

It does work but yes it will ring their extension wherever, if it's in the line 
group.

From: Lelio Fulgenzi 
Sent: Thursday, January 16, 2020 10:44 AM
To: Pawlowski, Adam ; voyp list, cisco-voip 
(cisco-voip@puck.nether.net) 
Subject: RE: Jabber mobile with shared primary extension

Yeah - that's my next thing to look at. What do you mean by labels? Do you mean 
the huntgroup login/logout buttons? The thing I know they're not gonna like is 
whether it rings their deskphone as well. I'd like to at least control that.

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Pawlowski, Adam
Sent: Thursday, January 16, 2020 10:41 AM
To: voyp list, cisco-voip 
(cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Jabber mobile with shared primary extension

I've used a hunt group for this which works on mobile now with the labels, but 
I hadn't assigned the same number to each station.

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Lelio Fulgenzi
Sent: Thursday, January 16, 2020 9:43 AM
To: voyp list, cisco-voip 
(cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Subject: [cisco-voip] Jabber mobile with shared primary extension


OK another question Anyone try a fleet of Jabber mobile devices all 
using the same shared primary extension, but, using different userIDs?


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Re: [cisco-voip] Jabber mobile with shared primary extension

[Pawlowski, Adam]

I've used a hunt group for this which works on mobile now with the labels, but 
I hadn't assigned the same number to each station.

From: cisco-voip  On Behalf Of Lelio 
Fulgenzi
Sent: Thursday, January 16, 2020 9:43 AM
To: voyp list, cisco-voip (cisco-voip@puck.nether.net) 

Subject: [cisco-voip] Jabber mobile with shared primary extension


OK another question Anyone try a fleet of Jabber mobile devices all 
using the same shared primary extension, but, using different userIDs?


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Re: [cisco-voip] setting primary phone for Jabber users without a physical phone - licensing

[Pawlowski, Adam]

The UI broke at some point and none of the proper localization fills in, so I 
guess get used to error.add .

It seems to use AXL or something similar in the background, and you can debug 
what the error is from the application log in RTMT, though half the time it’s a 
code and not useful.

Duplicate items will break it, as will things that are not valid for insert but 
are part of your universal templates, such as #DEPT# and someone has a 
department name with & in it.

As for the custom templates, they’re one of those once-a-year things I clean up 
with the database by assigning the phonetemplate record to the type, and then 
removing any of the phone templates that aren’t associated with anything.

I used to say they don’t hurt anything but if you invest in MigrationFX, it 
will leverage the phone button template as a selector to help you choose the 
correct migration. Unless you want it to clone SEPasdfghjkl-Invididual 
Template-8841-MFX all over, you’ll want these to be consistent.


From: Lelio Fulgenzi 
Sent: Wednesday, January 15, 2020 3:28 PM
To: Pawlowski, Adam ; Anthony Holloway 

Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net) 

Subject: RE: [cisco-voip] setting primary phone for Jabber users without a 
physical phone - licensing


If you really want to hear where it falls short, check this out:


  *   Create jabber phone with tool.
  *   It creates phone with custom phone button template.
  *   I don’t like it, so I change it to our standard.
  *   I want to test things again, so I delete phone and try to create again.
  *   It doesn’t work. It fails with the most descriptive error ever: 
“Error.add”
  *   TAC couldn’t figure it out.

Turns out, if you delete a phone with a custom phone button template, it will 
delete the template (if not in use elsewhere?). But because I changed the 
template to our standard template, the custom template remained.

When I tried to create the phone again using the tool, it tried to recreate the 
custom template, but because it already existed, it failed. ERROR.ADD

Ugh. UI 101….. better error messages!


From: Pawlowski, Adam mailto:aj...@buffalo.edu>>
Sent: Wednesday, January 15, 2020 3:03 PM
To: Lelio Fulgenzi mailto:le...@uoguelph.ca>>; Anthony 
Holloway 
mailto:avholloway+cisco-v...@gmail.com>>
Cc: voyp list, cisco-voip 
(cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Subject: RE: [cisco-voip] setting primary phone for Jabber users without a 
physical phone - licensing

Yes, it’s “almost” there, but falls short on those things you note.

But, it is at least consistent in what it does or doesn’t do, so you know what 
you have to fix or clean up.



From: Lelio Fulgenzi mailto:le...@uoguelph.ca>>
Sent: Wednesday, January 15, 2020 3:00 PM
To: Pawlowski, Adam mailto:aj...@buffalo.edu>>; Anthony 
Holloway 
mailto:avholloway+cisco-v...@gmail.com>>
Cc: voyp list, cisco-voip 
(cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Subject: RE: [cisco-voip] setting primary phone for Jabber users without a 
physical phone - licensing

Ok. I’ll have to keep that in mind.

I found a few “oddities” as I mentioned.


  *   No way to specify phone button templates. It creates a custom one for 
each user.
  *   It enables mobility by default which allows people to set up remote 
destinations for the jabber. We want to avoid that at launch so we will have to 
disable manually after the fact.
  *   It doesn’t populate the ASCII display fields.
  *   It doesn’t allow for the sip security profile and uses the universal 
template. Something we’ll have to change after.

They went so far with this tool, but missed out on a few things. I wish I had 
know about it sooner.

Lelio


From: Pawlowski, Adam mailto:aj...@buffalo.edu>>
Sent: Wednesday, January 15, 2020 2:54 PM
To: Lelio Fulgenzi mailto:le...@uoguelph.ca>>; Anthony 
Holloway 
mailto:avholloway+cisco-v...@gmail.com>>
Cc: voyp list, cisco-voip 
(cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Subject: RE: [cisco-voip] setting primary phone for Jabber users without a 
physical phone - licensing

Yes this works, just with the caveat about common phone profiles not working 
(again, at least for me, 11.5) though that’s not quick add’s fault.

Also, in my experience click the buttons, pressing enter has undesirable 
effects. I have been able to press “enter” when adding a new device and have 
had it double – insert which I didn’t think was possible.



From: Lelio Fulgenzi mailto:le...@uoguelph.ca>>
Sent: Wednesday, January 15, 2020 2:42 PM
To: Pawlowski, Adam mailto:aj...@buffalo.edu>>; Anthony 
Holloway 
mailto:avholloway+cisco-v...@gmail.com>>
Cc: voyp list, cisco-voip 
(cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:c

Re: [cisco-voip] setting primary phone for Jabber users without a physical phone - licensing

[Pawlowski, Adam]

Yes, it’s “almost” there, but falls short on those things you note.

But, it is at least consistent in what it does or doesn’t do, so you know what 
you have to fix or clean up.



From: Lelio Fulgenzi 
Sent: Wednesday, January 15, 2020 3:00 PM
To: Pawlowski, Adam ; Anthony Holloway 

Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net) 

Subject: RE: [cisco-voip] setting primary phone for Jabber users without a 
physical phone - licensing

Ok. I’ll have to keep that in mind.

I found a few “oddities” as I mentioned.


  *   No way to specify phone button templates. It creates a custom one for 
each user.
  *   It enables mobility by default which allows people to set up remote 
destinations for the jabber. We want to avoid that at launch so we will have to 
disable manually after the fact.
  *   It doesn’t populate the ASCII display fields.
  *   It doesn’t allow for the sip security profile and uses the universal 
template. Something we’ll have to change after.

They went so far with this tool, but missed out on a few things. I wish I had 
know about it sooner.

Lelio


From: Pawlowski, Adam mailto:aj...@buffalo.edu>>
Sent: Wednesday, January 15, 2020 2:54 PM
To: Lelio Fulgenzi mailto:le...@uoguelph.ca>>; Anthony 
Holloway 
mailto:avholloway+cisco-v...@gmail.com>>
Cc: voyp list, cisco-voip 
(cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Subject: RE: [cisco-voip] setting primary phone for Jabber users without a 
physical phone - licensing

Yes this works, just with the caveat about common phone profiles not working 
(again, at least for me, 11.5) though that’s not quick add’s fault.

Also, in my experience click the buttons, pressing enter has undesirable 
effects. I have been able to press “enter” when adding a new device and have 
had it double – insert which I didn’t think was possible.



From: Lelio Fulgenzi mailto:le...@uoguelph.ca>>
Sent: Wednesday, January 15, 2020 2:42 PM
To: Pawlowski, Adam mailto:aj...@buffalo.edu>>; Anthony 
Holloway 
mailto:avholloway+cisco-v...@gmail.com>>
Cc: voyp list, cisco-voip 
(cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Subject: RE: [cisco-voip] setting primary phone for Jabber users without a 
physical phone - licensing

Sorry. I’m mixing scenarios.

I think I have the process of adding Jabber devices for users with an existing 
phone down pat. With some oddities that we’ll have to address.

I’m concerned with creating Jabber devices for users without an existing phone 
-or- DN.

That being said, I believe I have that figured out, but will need to test it 
out.


  *   Go to quick user/phone add
  *   Select user (that has already been imported)
  *   Go to extension section
  *   Add a new extension using the appropriate Line template
  *   Click save (likely)
  *   Go to manage devices and add the appropriate device

We’ll have to see how that works out.

Lelio



From: Pawlowski, Adam mailto:aj...@buffalo.edu>>
Sent: Wednesday, January 15, 2020 10:50 AM
To: Lelio Fulgenzi mailto:le...@uoguelph.ca>>; Anthony 
Holloway 
mailto:avholloway+cisco-v...@gmail.com>>
Cc: voyp list, cisco-voip 
(cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Subject: RE: [cisco-voip] setting primary phone for Jabber users without a 
physical phone - licensing

I’m still a bit confused, unless you mean that with the quick add you’ve 
already set extensions for the user so that the templates will work, as you’ve 
already provisioned a phone?

We are sort of in the same boat. The sync will provision based on a mask if 
your LDAP data contains good numbers. You can also tell it to start assigning 
numbers from a pool on the same screen.  I have multiple exchanges I issue 
numbers from based on customer/location so this doesn’t do me any good 
pool-wise, and we don’t control the LDAP data so anything goes there.

You can configure the DN first, yes, as long as it is the primary extension it 
will appear in the quick add tool. I haven’t quite figured out where it stores 
the rest of the DNs you can add there or their order but the primary extension 
sure appears there.

I’m not sure if Prime Provisioning can assist with this task since it has some 
more capabilities, if it works for your installation.

From: Lelio Fulgenzi mailto:le...@uoguelph.ca>>
Sent: Wednesday, January 15, 2020 10:33 AM
To: Anthony Holloway 
mailto:avholloway+cisco-v...@gmail.com>>
Cc: Pawlowski, Adam mailto:aj...@buffalo.edu>>; voyp list, 
cisco-voip (cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Subject: RE: [cisco-voip] setting primary phone for Jabber users without a 
physical phone - licensing

Yeah – I was looking at that. I’m not sure we can take that approach because we 
have so

Re: [cisco-voip] setting primary phone for Jabber users without a physical phone - licensing

[Pawlowski, Adam]

Yes this works, just with the caveat about common phone profiles not working 
(again, at least for me, 11.5) though that’s not quick add’s fault.

Also, in my experience click the buttons, pressing enter has undesirable 
effects. I have been able to press “enter” when adding a new device and have 
had it double – insert which I didn’t think was possible.



From: Lelio Fulgenzi 
Sent: Wednesday, January 15, 2020 2:42 PM
To: Pawlowski, Adam ; Anthony Holloway 

Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net) 

Subject: RE: [cisco-voip] setting primary phone for Jabber users without a 
physical phone - licensing

Sorry. I’m mixing scenarios.

I think I have the process of adding Jabber devices for users with an existing 
phone down pat. With some oddities that we’ll have to address.

I’m concerned with creating Jabber devices for users without an existing phone 
-or- DN.

That being said, I believe I have that figured out, but will need to test it 
out.


  *   Go to quick user/phone add
  *   Select user (that has already been imported)
  *   Go to extension section
  *   Add a new extension using the appropriate Line template
  *   Click save (likely)
  *   Go to manage devices and add the appropriate device

We’ll have to see how that works out.

Lelio



From: Pawlowski, Adam mailto:aj...@buffalo.edu>>
Sent: Wednesday, January 15, 2020 10:50 AM
To: Lelio Fulgenzi mailto:le...@uoguelph.ca>>; Anthony 
Holloway 
mailto:avholloway+cisco-v...@gmail.com>>
Cc: voyp list, cisco-voip 
(cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Subject: RE: [cisco-voip] setting primary phone for Jabber users without a 
physical phone - licensing

I’m still a bit confused, unless you mean that with the quick add you’ve 
already set extensions for the user so that the templates will work, as you’ve 
already provisioned a phone?

We are sort of in the same boat. The sync will provision based on a mask if 
your LDAP data contains good numbers. You can also tell it to start assigning 
numbers from a pool on the same screen.  I have multiple exchanges I issue 
numbers from based on customer/location so this doesn’t do me any good 
pool-wise, and we don’t control the LDAP data so anything goes there.

You can configure the DN first, yes, as long as it is the primary extension it 
will appear in the quick add tool. I haven’t quite figured out where it stores 
the rest of the DNs you can add there or their order but the primary extension 
sure appears there.

I’m not sure if Prime Provisioning can assist with this task since it has some 
more capabilities, if it works for your installation.

From: Lelio Fulgenzi mailto:le...@uoguelph.ca>>
Sent: Wednesday, January 15, 2020 10:33 AM
To: Anthony Holloway 
mailto:avholloway+cisco-v...@gmail.com>>
Cc: Pawlowski, Adam mailto:aj...@buffalo.edu>>; voyp list, 
cisco-voip (cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Subject: RE: [cisco-voip] setting primary phone for Jabber users without a 
physical phone - licensing

Yeah – I was looking at that. I’m not sure we can take that approach because we 
have so many already imported. And extensions are assigned by us, not the AD 
team. It’s weird.

I will look through.

Question: Is there a “pool” that CCM will grab an extension from to assign 
magically?

From: Anthony Holloway 
mailto:avholloway+cisco-v...@gmail.com>>
Sent: Wednesday, January 15, 2020 10:28 AM
To: Lelio Fulgenzi mailto:le...@uoguelph.ca>>
Cc: Pawlowski, Adam mailto:aj...@buffalo.edu>>; voyp list, 
cisco-voip (cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] setting primary phone for Jabber users without a 
physical phone - licensing

The way I use self-provisioning, is to allow CUCM to create the lines for me at 
user synchronization time.

System > LDAP > Dir > Enterprise Users* > Feature Group Template > Common 
Features* > User Profile > Common User* > Universal Line Template > Common 
Line* > All kinds of template settings
System > LDAP > Dir > Enterprise Users* > Apply Mask to synced telephone 
numbers to create a new line for inserted users > Checked
System > LDAP > Dir > Enterprise Users* > Apply Mask to synced telephone 
numbers to create a new line for inserted users  > Mask > **

*These are just the names of my objects, your names might be different.
**That's 12 Xs to accommodate a NANP +E164 number from LDAP

On Tue, Jan 14, 2020 at 9:54 PM Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:

I guess I’m missing the piece of how is an extension assigned then?

With an existing phone, the extension is assigned automatically.

I was gonna do some testing, but any pointers would help.

I mean, can I create 

Re: [cisco-voip] setting primary phone for Jabber users without a physical phone - licensing

[Pawlowski, Adam]

I’m still a bit confused, unless you mean that with the quick add you’ve 
already set extensions for the user so that the templates will work, as you’ve 
already provisioned a phone?

We are sort of in the same boat. The sync will provision based on a mask if 
your LDAP data contains good numbers. You can also tell it to start assigning 
numbers from a pool on the same screen.  I have multiple exchanges I issue 
numbers from based on customer/location so this doesn’t do me any good 
pool-wise, and we don’t control the LDAP data so anything goes there.

You can configure the DN first, yes, as long as it is the primary extension it 
will appear in the quick add tool. I haven’t quite figured out where it stores 
the rest of the DNs you can add there or their order but the primary extension 
sure appears there.

I’m not sure if Prime Provisioning can assist with this task since it has some 
more capabilities, if it works for your installation.

From: Lelio Fulgenzi 
Sent: Wednesday, January 15, 2020 10:33 AM
To: Anthony Holloway 
Cc: Pawlowski, Adam ; voyp list, cisco-voip 
(cisco-voip@puck.nether.net) 
Subject: RE: [cisco-voip] setting primary phone for Jabber users without a 
physical phone - licensing

Yeah – I was looking at that. I’m not sure we can take that approach because we 
have so many already imported. And extensions are assigned by us, not the AD 
team. It’s weird.

I will look through.

Question: Is there a “pool” that CCM will grab an extension from to assign 
magically?

From: Anthony Holloway 
mailto:avholloway+cisco-v...@gmail.com>>
Sent: Wednesday, January 15, 2020 10:28 AM
To: Lelio Fulgenzi mailto:le...@uoguelph.ca>>
Cc: Pawlowski, Adam mailto:aj...@buffalo.edu>>; voyp list, 
cisco-voip (cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] setting primary phone for Jabber users without a 
physical phone - licensing

The way I use self-provisioning, is to allow CUCM to create the lines for me at 
user synchronization time.

System > LDAP > Dir > Enterprise Users* > Feature Group Template > Common 
Features* > User Profile > Common User* > Universal Line Template > Common 
Line* > All kinds of template settings
System > LDAP > Dir > Enterprise Users* > Apply Mask to synced telephone 
numbers to create a new line for inserted users > Checked
System > LDAP > Dir > Enterprise Users* > Apply Mask to synced telephone 
numbers to create a new line for inserted users  > Mask > **

*These are just the names of my objects, your names might be different.
**That's 12 Xs to accommodate a NANP +E164 number from LDAP

On Tue, Jan 14, 2020 at 9:54 PM Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:

I guess I’m missing the piece of how is an extension assigned then?

With an existing phone, the extension is assigned automatically.

I was gonna do some testing, but any pointers would help.

I mean, can I create the DN first? And configure DN with userID?

I’m still getting used to this process.
-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<http://www.uoguelph.ca/ccs> | @UofGCCS on Instagram, 
Twitter and Facebook



On Jan 14, 2020, at 8:35 AM, Pawlowski, Adam 
mailto:aj...@buffalo.edu>> wrote:
The quick user/phone add should work as generically for Jabber devices as 
anything else. Just that I’ve found that the common phone profile will not 
populate the correct “Cisco Support Field” for mobile and I have to go into 
those and key in the configuration file that customer should have.

It ended up being easier to insert via AXL to avoid having to fool with that.

Adam

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Lelio Fulgenzi
Sent: Monday, January 13, 2020 1:13 PM
To: voyp list, cisco-voip 
(cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Subject: [cisco-voip] setting primary phone for Jabber users without a physical 
phone - licensing


I’m trying to setup an easy to replicate (troubleshoot/reconcile) system in 
place for our Jabber deployment.

We’ve decided to assign each Jabber device type to each user so there’s no 
end-user support issues when they want to use it on another device and/or move 
from one device type to another.

If I’m not mistaken, in this case, and, in any case where a primary phone is 
not set, more licenses are used then would need to be. The issue at hand here, 
is what happens when we deploy Jabber without a physical phone?

I’m already finding that the quick/add feature really only works when a phone 
is configured (although 

Re: [cisco-voip] setting primary phone for Jabber users without a physical phone - licensing

[Pawlowski, Adam]

The quick user/phone add should work as generically for Jabber devices as 
anything else. Just that I've found that the common phone profile will not 
populate the correct "Cisco Support Field" for mobile and I have to go into 
those and key in the configuration file that customer should have.

It ended up being easier to insert via AXL to avoid having to fool with that.

Adam

From: cisco-voip  On Behalf Of Lelio 
Fulgenzi
Sent: Monday, January 13, 2020 1:13 PM
To: voyp list, cisco-voip (cisco-voip@puck.nether.net) 

Subject: [cisco-voip] setting primary phone for Jabber users without a physical 
phone - licensing


I'm trying to setup an easy to replicate (troubleshoot/reconcile) system in 
place for our Jabber deployment.

We've decided to assign each Jabber device type to each user so there's no 
end-user support issues when they want to use it on another device and/or move 
from one device type to another.

If I'm not mistaken, in this case, and, in any case where a primary phone is 
not set, more licenses are used then would need to be. The issue at hand here, 
is what happens when we deploy Jabber without a physical phone?

I'm already finding that the quick/add feature really only works when a phone 
is configured (although I have to work through that). So, I'm wondering, is it 
worthwhile to configure a "dummy" phone for Jabber users and set the primary 
phone for all Jabber clients to this dummy phone? Or should I ask them to pick 
the CSF jabber device as the primary phone for example?

What have others done?

Lelio

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[cisco-voip] Offline Analog Call Recording

[Pawlowski, Adam]

Hello all,

I'm trying to find a solution that will support not only BiB call recording for 
IP Phones and Jabber, of which there are many, but, also analog lines. However, 
we're not looking to record them from a VG. These lines are used as a backup 
during maintenance to the VoIP system, or when the facility is isolated from 
the network or VoIP is otherwise down. We'd like to record the audio from them 
and either spool it to be sent to the recording server later, or, less ideally, 
the recording server would be a physical server located at that site so that 
our network interruptions aren't their network interruptions.

I can find any number of solutions that will work for the BiB devices, but, not 
this analog scenario. I've seen some options from Revcord and HigherGround 
which sort of can accomplish this but are larger software suites.

Anyone aware of anything that can do this?

Regards,



Adam Pawlowski

SUNYAB

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Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

[Pawlowski, Adam]

We went from 15.4(3)M9 to 15.7(3)M4b

Noted also in this version that we previously allowed video/media to flow to 
the CUBE just in case we were ever to use it for routing media calls, and this 
version (15.7) drops calls from Jabber, etc when we place them on hold.

TAC said it was something to do with the number of options in the SDP changing 
and it being a protocol violation. The IEC error and Q code did sort of 
indicate this, but, as to why Cisco's systems don't want to play nice protocol 
wise with each other I don't know.



From: Ryan Huff 
Sent: Monday, December 2, 2019 10:25 AM
To: Pawlowski, Adam ; cisco-voip@puck.nether.net
Subject: Re: Native Call Queuing on UCM 12.5 with SIP Trunks

Adam,

I've not encountered the streaming issue with a SIP trunk, just H.323.. 
interesting.

Out of curiosity, what code level in the ISR G2 did you go to when you 
encountered this? The behavior of simplex streaming is to send a null/fake IP 
address in the logic channel multimedia control message, which for anything 
required to check incoming RTP packets against the IP/port would fail (creating 
the silent MoH condition) which I was under the impression was only the G3 ISRs.

Thanks,

Ryan

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
on behalf of Pawlowski, Adam mailto:aj...@buffalo.edu>>
Sent: Monday, December 2, 2019 8:50 AM
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks


After an upgrade on our CUBE to a later IOS (ISR G2) we needed to have the 
duplex streaming parameter enabled for MOH/Ringback to work properly.



I believe there is a note about that being needed for the 4400 series gateways 
as well, but, I know we didn't have it before but needed it at some point to 
get media to be heard properly by the far end in a network hold kind of 
situation.



Adam







From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Mark H. Turpin
Sent: Monday, December 2, 2019 8:20 AM
To: Dana Tong mailto:dana.t...@yellit.com.au>>; 
cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks



Does the problem go away if you force an MTP?



Can you provide a debug ccsip messages of an external call?



Can you share your sanitized CUBE config?



Curious if your call/media is changing after the initial announcement and your 
ITSP doesn't care for the way you're attempting to change it.





From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
on behalf of Dana Tong mailto:dana.t...@yellit.com.au>>
Sent: Monday, December 2, 2019 12:36 AM
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> 
mailto:cisco-voip@puck.nether.net>>
Subject: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks



*** EXTERNAL EMAIL - DO NOT CLICK LINKS ***

Hi all,



This being back on the tools is doing my head in. I think I've been away too 
long.



So I have configured native call queuing on UCM 12.5.x

Internal calls queue fine. The initial announcement plays. Period announcements 
work and the MOH is fine in between.



External calls hear the initial announcement.

There is no MOH after the announcement and the external user has ring-back tone.

There is no periodic announcement.



Any thoughts on why internal is okay and external is not working? Is it 
relating to SIP supplementary services?



Cheers

Dana


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Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

[Pawlowski, Adam]

After an upgrade on our CUBE to a later IOS (ISR G2) we needed to have the 
duplex streaming parameter enabled for MOH/Ringback to work properly.

I believe there is a note about that being needed for the 4400 series gateways 
as well, but, I know we didn't have it before but needed it at some point to 
get media to be heard properly by the far end in a network hold kind of 
situation.

Adam



From: cisco-voip  On Behalf Of Mark H. 
Turpin
Sent: Monday, December 2, 2019 8:20 AM
To: Dana Tong ; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

Does the problem go away if you force an MTP?

Can you provide a debug ccsip messages of an external call?

Can you share your sanitized CUBE config?

Curious if your call/media is changing after the initial announcement and your 
ITSP doesn't care for the way you're attempting to change it.


From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
on behalf of Dana Tong mailto:dana.t...@yellit.com.au>>
Sent: Monday, December 2, 2019 12:36 AM
To: cisco-voip@puck.nether.net 
mailto:cisco-voip@puck.nether.net>>
Subject: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

*** EXTERNAL EMAIL - DO NOT CLICK LINKS ***

Hi all,



This being back on the tools is doing my head in. I think I've been away too 
long.



So I have configured native call queuing on UCM 12.5.x

Internal calls queue fine. The initial announcement plays. Period announcements 
work and the MOH is fine in between.



External calls hear the initial announcement.

There is no MOH after the announcement and the external user has ring-back tone.

There is no periodic announcement.



Any thoughts on why internal is okay and external is not working? Is it 
relating to SIP supplementary services?



Cheers

Dana


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Re: [cisco-voip] using the quick user/phone add - some artifacts

[Pawlowski, Adam]

I haven't found a solution for that in the UCM  yet, I'm hoping it is fixed 
better in 12.5 .


It may be one of those things that we have to get a case open for and then get 
a defect going on, but, this takes up time.


The generic error is awful to deal with as well. & in the department name 
breaking a template, something being too long that you've keyed in but no 
explanation. I found the best one the other day so far. If you interact with 
the dialog boxes that come up for a new phone/DN, and use the keyboard to 
submit it, it double submits. The phone gives you an "Error" but then it is 
added. When I did it with a DN, it actually inserted the DN twice (same pattern 
and partition) so I have no idea what happened there.


Regarding the Universal templates, they have a lot of blank space and are 
annoying to use that's for sure.  I've only seen it add the custom template for 
Jabber for whatever reason but for the phones it will use the Universal from 
the template.


More or less some day I'd like to get some better process going to review the 
devices in the UCM and tweak and touch these things up. I have a lot of that 
working in Ruby anyways for spot checks and work. Easy enough to grab the 
device and make sure the displayname and asciidisplayname values are the same 
and write it back. I have no time for cool things like this but the ultimate 
goal is to allow the technicians to use the template without straying too far 
or worrying about minutae, that we can pick up over night. We'd also look for 
user name changes and re-write caller IDs and display names. This if course 
means if the directory source says your name is Ronald then we are going to put 
that there. Trivial enough with programming though to dip into another field 
somewhere for a preferred name but I'm getting way off course now.


Adam



From: cisco-voip  on behalf of Lelio 
Fulgenzi 
Sent: Monday, November 25, 2019 2:11 PM
To: voyp list, cisco-voip (cisco-voip@puck.nether.net)
Subject: [cisco-voip] using the quick user/phone add - some artifacts


So, tried out the quick user/phone add. I like it, although I had and am still 
having some issues. (See notes below for first problem I had, with solution)

The issue/concern I’m having now is that the process does the following:


  *   Creates a custom phone button template with device name in the name of 
template and uses that instead of the device type phone button template used 
when creating manually.
  *   It uses the “Universal device template – model independent security 
profile” instead of the device type security profile when creating manually.
  *   The process automatically enables mobility for the user and associates 
their userID to the mobility ID.
  *   ASCII Display (Caller ID) is not filled in like the other field.

Has anyone found a way around any of these issues?

I checked the universal device template and there’s not options that I can see 
to pick different phone button template.

I tried creating a copy of the default universal phone button template and that 
is available in the drop down of the universal device template, but with the 
same results.

This will mean a lot of going back to change things to match our current SOPs.






Original Issue. Fixed.

For some historical archiving, I ran into the following problem: After deleting 
a phone I created using the tool, I was not able to (re)create it again. It 
gave me an ominous “error.add” pop-up.

Turns out, what I had done, was notice that the create process created a custom 
phone button template, removed this and used the standard device phone button 
template and then delete the phone after some testing. This left the custom 
phone button template (with the userID in the name) in the system. Subsequent 
(re)creates tried to create that same custom phone button template and was 
failing. “error.add” was referring to the fact it couldn’t create the template 
since it already existed.

---
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] Server-groups and failover...

[Pawlowski, Adam]

Sure

Again some time ago but what we’d run into was that PSTN calls in would fail on 
the network side if we didn’t signal progress back within a period of time. 
With stock timers, it never was able to move through all of the available call 
processors (4) before this happened, so you’d not really had effective 
redundancy, nor could it fall through for handling if you wanted to do that. It 
was based on watching the debugging to see how the flow went when we ACL’d off 
various UCMs from the CUBE itself.

As you can see form other postings in the thread, I did not know enough about 
this at the time to adjust other timers to account for either mid-call or call 
teardown signaling. I’m still learning, as always.

On the outbound end to the ITSP we also had a BGP relationship setup with them 
to tear down routing if the physical circuit failed somewhere outbound. That 
allowed us two disparate CUBEs that would still be available for call routing, 
resources, etc even if we lost an outbound circuit – we could still reach both 
peers for LB/HA across the surviving one at another structure from either CUBE 
in the arrangement.

Adam



From: Anthony Holloway 
Sent: Tuesday, November 19, 2019 9:37 AM
To: Pawlowski, Adam 
Cc: Jonathan Charles ; Cisco VoIP Group 

Subject: Re: [cisco-voip] Server-groups and failover...

I'm curious as to how you arrived at those settings. Would you be able to share 
your path to arriving at those specific settings?

I typically leave the 500ms alone but do lower the retries to so that the total 
worst case delay would be ~3sec. For no other reason than that was always the 
for h323 tcp timeout we all used to use.

I do also use OPTIONS though, because in my opinion, tweaking/enabling both 
provides the best user experience. Though I don't typically adjust the default 
OPTIONS timers and retries. Do you?

On Tue, Nov 19, 2019, 7:06 AM Pawlowski, Adam 
mailto:aj...@buffalo.edu>> wrote:
I think I had to make some adjustments to our timers as well to get this to 
work before network timeout or similar:

sip-ua
retry invite 2
timers trying 100
!

I know I also goofed this up between dial peer group and server group, one of 
the two will retry within the group, the other sure doesn’t.


From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Jonathan Charles
Sent: Monday, November 18, 2019 11:56 PM
To: Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>>
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] Server-groups and failover...

I pasted the wrong part of the script (to manually change it)...

Here is the actual config:



voice class server-group 1
 ipv4 172.31.120.43
 ipv4 172.31.125.43 preference 2
 description Verizon SIP
!

Jonathan

On Mon, Nov 18, 2019 at 10:22 PM Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>> 
wrote:
First off, I'm wondering why it says "no ipv4" in front of your two addresses.  
That might be your problem right there.

Secondly, I'd recommend putting an explicit preference on your entries, it's 
just better for everyone, and you don't get a credit back from Cisco for saving 
on a few ascii characters by implicitly using the default.  Plus, if the 
default is 0, which it is, then your next preference should be technically 1.  
But then having nothing and 1 seems silly, because if pref 1 is actually pref 
2, then well, might as well call them pref nothing and pref 8.  I digress.

You might not have failed over, because you might not have provided the system 
with the correction conditions to failover...E.g., you didn't wait long enough.

No seriously, by default SIP failover occurs after 30 seconds.  Unless, did you 
lower the retry count under sip-ua?  Or did you enable SIP options?  If you 
enabled SIP options, have your confirmed that it's turned on correctly?

Can you share the output of the following commands:

show run | section sip-ua|sip.options-keepalive

show dial-peer voice summary

Feel free to redact what you need to, in terms of IPs or usernames/passwords.  
I am only looking for the features and settings for retries and keepalives.


On Mon, Nov 18, 2019 at 9:26 PM Jonathan Charles 
mailto:jonv...@gmail.com>> wrote:
Using session server groups on outbound dial-peers and it does not appear to be 
failing over:


voice class server-group 1
 no ipv4 172.31.125.43  preference 2
 no ipv4 172.31.120.43
 description Verizon SIP
!

We had the 172.31.20.43 go down (no response to invites) and we did NOT 
failover to the second (.125.43)...

What is needed to force a failover to the next configured SBC?


Jonathan
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Re: [cisco-voip] Server-groups and failover...

[Pawlowski, Adam]

In this router we actually tried something different (and gave up on this 
implementation thus I think the dial-peer groups maybe not being effective here 
for redundancy, and we’d just not fixed it, it’s been a couple of years – it 
may have been the options ping either not working or marking the whole peer out 
of service despite other available targets).

I also like to complicate things as you can see below:

dial-peer voice 200 voip
description PSTN ITSP SIP Trunk
translation-profile outgoing ITSP-PSTN-Out
session protocol sipv2
session transport udp
session server-group 200
destination dpg 400
destination e164-pattern-map 2200
incoming called e164-pattern-map 200
voice-class codec 200
voice-class sip profiles 200
voice-class sip options-keepalive
dtmf-relay rtp-nte sip-kpml
no vad
dial-peer voice 400 voip
description Production UCM Dial Peer
translation-profile outgoing Windstream-PSTN-In
session protocol sipv2
session target dns:prodall.cmgroup.srv.domain
session transport tcp
destination dpg 200
destination e164-pattern-map 400
voice-class codec 400
voice-class sip options-keepalive
dtmf-relay rtp-nte sip-kpml
no vad
dial-peer voice 500 voip
description Non-Working Number 404 Code Fallthrough
service non_working_number out-bound
destination-pattern +.*T
session target loopback:rtp


voice class dpg 400
description Production UCM Dial Peer Group
dial-peer 400
dial-peer 500 preference 1
!
voice class dpg 200
description ITSP SIP Dial Peer Group
dial-peer 200
!
voice class server-group 200
ipv4 10.250.0.5
ipv4 10.250.0.6 preference 1
description ITSP Server Group
!

From: Johnson, Tim 
Sent: Tuesday, November 19, 2019 8:23 AM
To: Pawlowski, Adam ; 'Jonathan Charles' 
; Anthony Holloway 
Cc: cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] Server-groups and failover...

What’s your dial peer configuration look like? Curious if you have ‘huntstop’ 
configured.

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Pawlowski, Adam
Sent: Tuesday, November 19, 2019 8:07 AM
To: 'Jonathan Charles' mailto:jonv...@gmail.com>>; Anthony 
Holloway 
mailto:avholloway+cisco-v...@gmail.com>>
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] Server-groups and failover...

I think I had to make some adjustments to our timers as well to get this to 
work before network timeout or similar:

sip-ua
retry invite 2
timers trying 100
!

I know I also goofed this up between dial peer group and server group, one of 
the two will retry within the group, the other sure doesn’t.


From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Jonathan Charles
Sent: Monday, November 18, 2019 11:56 PM
To: Anthony Holloway 
mailto:avholloway+cisco-v...@gmail.com>>
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] Server-groups and failover...

I pasted the wrong part of the script (to manually change it)...

Here is the actual config:



voice class server-group 1
 ipv4 172.31.120.43
 ipv4 172.31.125.43 preference 2
 description Verizon SIP
!

Jonathan

On Mon, Nov 18, 2019 at 10:22 PM Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>> 
wrote:
First off, I'm wondering why it says "no ipv4" in front of your two addresses.  
That might be your problem right there.

Secondly, I'd recommend putting an explicit preference on your entries, it's 
just better for everyone, and you don't get a credit back from Cisco for saving 
on a few ascii characters by implicitly using the default.  Plus, if the 
default is 0, which it is, then your next preference should be technically 1.  
But then having nothing and 1 seems silly, because if pref 1 is actually pref 
2, then well, might as well call them pref nothing and pref 8.  I digress.

You might not have failed over, because you might not have provided the system 
with the correction conditions to failover...E.g., you didn't wait long enough.

No seriously, by default SIP failover occurs after 30 seconds.  Unless, did you 
lower the retry count under sip-ua?  Or did you enable SIP options?  If you 
enabled SIP options, have your confirmed that it's turned on correctly?

Can you share the output of the following commands:

show run | section sip-ua|sip.options-keepalive

show dial-peer voice summary

Feel free to redact what you need to, in terms of IPs or usernames/passwords.  
I am only looking for the features and settings for retries and keepalives.


On Mon, Nov 18, 2019 at 9:26 PM Jonathan Charles 
mailto:jonv...@gmail.com>> wrote:
Using session server groups on outbound dial-peers and it does not appear to be 
failing over:


voice class server-group 1
 no ipv4 172.31.125.43  preference 2
 no ipv4 172.31.120.43
 description Verizon SIP
!

We had the 172.31.20.43 go down (no response to invites) and we did NOT 
failover to the second (.125.43)...

What is n

Re: [cisco-voip] Server-groups and failover...

[Pawlowski, Adam]

I think I had to make some adjustments to our timers as well to get this to 
work before network timeout or similar:

sip-ua
retry invite 2
timers trying 100
!

I know I also goofed this up between dial peer group and server group, one of 
the two will retry within the group, the other sure doesn’t.


From: cisco-voip  On Behalf Of Jonathan 
Charles
Sent: Monday, November 18, 2019 11:56 PM
To: Anthony Holloway 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Server-groups and failover...

I pasted the wrong part of the script (to manually change it)...

Here is the actual config:



voice class server-group 1
 ipv4 172.31.120.43
 ipv4 172.31.125.43 preference 2
 description Verizon SIP
!

Jonathan

On Mon, Nov 18, 2019 at 10:22 PM Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>> 
wrote:
First off, I'm wondering why it says "no ipv4" in front of your two addresses.  
That might be your problem right there.

Secondly, I'd recommend putting an explicit preference on your entries, it's 
just better for everyone, and you don't get a credit back from Cisco for saving 
on a few ascii characters by implicitly using the default.  Plus, if the 
default is 0, which it is, then your next preference should be technically 1.  
But then having nothing and 1 seems silly, because if pref 1 is actually pref 
2, then well, might as well call them pref nothing and pref 8.  I digress.

You might not have failed over, because you might not have provided the system 
with the correction conditions to failover...E.g., you didn't wait long enough.

No seriously, by default SIP failover occurs after 30 seconds.  Unless, did you 
lower the retry count under sip-ua?  Or did you enable SIP options?  If you 
enabled SIP options, have your confirmed that it's turned on correctly?

Can you share the output of the following commands:

show run | section sip-ua|sip.options-keepalive

show dial-peer voice summary

Feel free to redact what you need to, in terms of IPs or usernames/passwords.  
I am only looking for the features and settings for retries and keepalives.


On Mon, Nov 18, 2019 at 9:26 PM Jonathan Charles 
mailto:jonv...@gmail.com>> wrote:
Using session server groups on outbound dial-peers and it does not appear to be 
failing over:


voice class server-group 1
 no ipv4 172.31.125.43  preference 2
 no ipv4 172.31.120.43
 description Verizon SIP
!

We had the 172.31.20.43 go down (no response to invites) and we did NOT 
failover to the second (.125.43)...

What is needed to force a failover to the next configured SBC?


Jonathan
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Re: [cisco-voip] VCS Expressway upgrade, 8.7 to 12.5

[Pawlowski, Adam]

If anything like this farts up on us then I usually remove it from the 
Expressway and put it back. It doesn’t really cause a ton of harm to do so and 
seems to clear a lot of issues.

Troubleshooting certificate issues ends up being kind of obnoxious as I’ve yet 
to find any debugging that really expands on the fault errors that you can see 
in the event log.

If this were me and I ran into “Decryption Error” and not a validation error, 
I’d be looking at say the default TLS level and ciphers as they may have 
changed between 8.7 -> 12.5. I know it’s default TLS 1.2 at this point. I’d 
actually just changed the s_channel string or what have you to try and 
eliminate the ciphers flagged for no forward secrecy and it broke MRA for me as 
well. I got “Internal Server Error” back in Jabber.

I don’t know if it enforces key lengths on the certificates at this point 
either but that’s where I’d go with this one personally.

Adam



From: cisco-voip  On Behalf Of Jonathan 
Charles
Sent: Monday, November 18, 2019 12:24 AM
To: Ryan Huff 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] VCS Expressway upgrade, 8.7 to 12.5

So, just an FYI, with MRA down, we rolled back to 8.7.3... going to try this 
again next Friday...

Should we try going to an intermediate version first, say, 8.11 or something?


Jonathan

On Sun, Nov 17, 2019 at 6:15 PM Jonathan Charles 
mailto:jonv...@gmail.com>> wrote:
Loaded the local certs... no joy...


Jonathan

On Sun, Nov 17, 2019 at 6:07 PM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
Have you tried adding the IMP identity cert into the Expressway trust? It 
shouldn’t have to work that way, but if it does, might point to an issue with 
how the CA chain is being recognized in the trust.

Also, make sure to do a full reboot of the Expressway node after adding certs 
into the truststore (again, you shouldn’t have to do that but I’ve seen this 
work before).
Sent from my iPhone


On Nov 17, 2019, at 18:58, Jonathan Charles 
mailto:jonv...@gmail.com>> wrote:

When I try to refresh the IMP nodes, I get Failed: Unable to communicate with 
[[IMPNODE] CryptoError: Decryption failure.

On Sun, Nov 17, 2019 at 5:54 PM Jonathan Charles 
mailto:jonv...@gmail.com>> wrote:
I re-uploaded the root and intermediate CA certificate... still get the same 
error...

I also tried adding a new AXL user... same error...


Jonathan

On Sun, Nov 17, 2019 at 5:48 PM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
Likely certificate / trust issues..
Sent from my iPhone


On Nov 17, 2019, at 18:36, Jonathan Charles 
mailto:jonv...@gmail.com>> wrote:

Yep, we are running into clustering issues...

Getting Inactive: (Remote host is reachable but connection is not established. 
Either refresh this page, or check the credentials.)

For IMP connection, so MRA is down...

Still looking for a fix...


Jonathan

On Fri, Nov 15, 2019 at 7:17 PM Erick Bergquist 
mailto:erick...@gmail.com>> wrote:
I’ve done 2 8.11.x to 12.5.5 fine (clustered setup, 4). There is a bug with 
clustering to watch out for but I did not encounter it. The 12.5 Cisco download 
page has a note and link about this.

Currently working on jabberd process high memory consumption issue on one node 
that has been present since 8.11.x which 12.5 had memory leak fix for but still 
an issue. Slow memory increase over time just on one of the edge nodes.

Going to look over 12.5.6 release notes now

Erick



On Fri, Nov 15, 2019 at 3:28 PM Matt Jacobson 
mailto:m4ttjacob...@gmail.com>> wrote:
If that is the case, then I would double check that it is supported. In the 
release notes there is a chart for supported platforms based on serial numbers. 
If it is a legacy Tandberg box, then I suspect 12.x may not work out for you.

On Fri, Nov 15, 2019 at 14:30 Jonathan Charles 
mailto:jonv...@gmail.com>> wrote:
This is a legacy Tandberg VCS for video only... no MRA, no remote phones... 
just inbound and outbound sip video...


Jonathan

On Fri, Nov 15, 2019 at 12:44 PM Pawlowski, Adam 
mailto:aj...@buffalo.edu>> wrote:
We’re at 12.5.3 and probably moving to 12.5.5/12.5.6 somewhere in the Holiday 
timeframe when everything quiets down a bit.

There hasn’t been really any significant issue upgrading from 8 -> 12, but 
there have been a couple of bugs that largely are all resolved by deleting and 
rebuilding whatever the thing is that is misbehaving.

The requirement for the _cup_login and _cisco-uds SRVs went away though it 
still endlessly logs a warning about not finding them, but it will work.

You do also gain the ability to play with the openssl cipher strings but in my 
limited experience trying to change those to bump them up a notch, it ends up 
breaking XMPP or something.

Adam

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Jonathan Charles
Sent: Friday, November 15, 2019 11:59 AM
To: Ryan Huff mailto:ryanh...@outlook.com>>
Cc: cisco-vo

Re: [cisco-voip] VCS Expressway upgrade, 8.7 to 12.5

[Pawlowski, Adam]

We’re at 12.5.3 and probably moving to 12.5.5/12.5.6 somewhere in the Holiday 
timeframe when everything quiets down a bit.

There hasn’t been really any significant issue upgrading from 8 -> 12, but 
there have been a couple of bugs that largely are all resolved by deleting and 
rebuilding whatever the thing is that is misbehaving.

The requirement for the _cup_login and _cisco-uds SRVs went away though it 
still endlessly logs a warning about not finding them, but it will work.

You do also gain the ability to play with the openssl cipher strings but in my 
limited experience trying to change those to bump them up a notch, it ends up 
breaking XMPP or something.

Adam

From: cisco-voip  On Behalf Of Jonathan 
Charles
Sent: Friday, November 15, 2019 11:59 AM
To: Ryan Huff 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] VCS Expressway upgrade, 8.7 to 12.5

Thanks, the latest is 12.5.6, released last week, I am avoiding it like the 
plague...and the bug fix doesn't apply to us.

I am going with 12.5.5 (released in August).

I already have release keys (Cisco AM sent them over)...

Hybrid services are on a separate VCS-C that is already 12.5.

My plan is to get new certs if we have any issues


Thanks!


Jonathan

On Fri, Nov 15, 2019 at 10:46 AM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
A couple of thoughts for you...


  *   Get the software release key for 12.x now (you'll be asked to enter it 
during the upgrade in the GUI). You'll need to work with TAC > GLO for this if 
(and I assume this would be your case) the existing 8.7 serial is active in 
Cisco's licensing system. The caveat to trying to do this with Cisco's 
self-service license re-host tool is that while the 8.7 serial is active, it 
won't allow you to assign the new 12.x software release PAK to the serial 
because the serial is already assigned to another software release key.

 *   Take a backup first, your only roll back option is to re-install 8.7 
and restore the backup.

  *   Your VMware Hypervisor needs to be 6.0/5/7.

  *   If you have Hybrid Services configured, make sure the management 
connector is up to date first.

  *   SSL Certificate validation changed a bit in 8.8+

 *   Verify proper forward / reverse DNS for all the relevant touch points
 *   Make sure the Expressway certificate trust is up-to-date with all the 
current CUCM,CUC,IMP identity certificates (self-signed) or CA certificates 
(public CA signed certificates).
 *   no duplicate certificates in the Expressway trusts
Beyond that, just pay attention to the caveats list in the upgrade doc for your 
version of 12.5.x (12.5.4 is the latest I think).

Thanks,

Ryan


From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
on behalf of Jonathan Charles mailto:jonv...@gmail.com>>
Sent: Friday, November 15, 2019 10:57 AM
To: cisco-voip@puck.nether.net 
mailto:cisco-voip@puck.nether.net>>
Subject: [cisco-voip] VCS Expressway upgrade, 8.7 to 12.5

Can we just upgrade directly or do we need to go to an intermediary version 
first?

Also, any gotchas besides new certificates?


Jonathan
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Re: [cisco-voip] Webex room devices with 3rd party meetings

[Pawlowski, Adam]

I don't know if that whole mess is necessary to join a Zoom call. If the party 
who sent you the Zoom invite has the connector service for interop, you can 
usually call the meeting by @zoomcrc.com as  SIP call. If they 
don't, it prompts you to sign in on an app with your account (which is paying 
for a connector license I guess) and then you can join the meeting that way. 

In my experience with that, I've yet to come across a Zoom meeting I've been 
invited to where someone has the connector on their account, and we don't pay 
for one. In that case the Room Kit ends up being largely useless. Yes you can 
share in a PC but then making use of the camera is expensive.

And before anyone waves in with "Room Kit Mini", the framing on that is meant 
for some very limited room sizes and is not suitable to the type of space you'd 
be installing a proper codec to anyways. 

All of these disparate services work best if you can coax the other 
participant(s) into playing in your house with your services. 

We've unfortunately decided not to pursue the Webex systems given this, and 
have instead gone with external cameras and microphones that we can present to 
a PC as a USB device. Not for everyone and doesn't fit all cases.



> -Original Message-
> From: cisco-voip  On Behalf Of James
> Andrewartha
> Sent: Wednesday, November 13, 2019 10:43 PM
> To: cisco-voip@puck.nether.net
> Subject: Re: [cisco-voip] Webex room devices with 3rd party meetings
> 
> Zoom do sell a connector for Cisco room systems, but I don't know how well it
> works for joining third-party zoom meetings. The setup instructions are quite
> involved https://support.zoom.us/hc/en-us/articles/115003126346-Zoom-
> Connector-for-Cisco
> 
> I've just put in a Logitech Tap Microsoft Teams Room system and the
> announced compatibility with Webex and Zoom makes me happy we went with
> them. Someone noted there's been no public agreement between Webex and
> Zoom, only between them and MS Teams.
> 
> --
> James Andrewartha
> Network & Projects Engineer
> Christ Church Grammar School
> Claremont, Western Australia
> Ph. (08) 9442 1757
> Mob. 0424 160 877
> 
> On 14/11/19 7:13 am, Lelio Fulgenzi wrote:
> >
> > The question is whether or not the Zoom devices can dial other systems.
> >
> > If you're getting Zoom across the board, that's fine. But if you can't
> participate with other systems, then you've further pigeon hole'd yourself.
> >
> > If that makes sense.
> >
> > The fact that Webex/Microsoft signed a deal is even more weight for Webex
> devices.
> >
> > But I hear what you're saying. If your zoom, hard to say buy webex. ☹
> >
> > ---
> > 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
> >
> >
> >
> > -Original Message-
> > From: Myron Young 
> > Sent: Wednesday, November 13, 2019 5:52 PM
> > To: Lelio Fulgenzi 
> > Cc: cisco-voip voyp list 
> > Subject: Re: Webex room devices with 3rd party meetings
> >
> > Thanks for the insight. I will try with the cloud device to test this out 
> > and see if
> i can convince the powers that be. Otherwise we may be going full Zoom which
> I’m not adverse to, but would be very unfamiliar territories.
> >
> >> On Nov 13, 2019, at 4:40 PM, Lelio Fulgenzi  wrote:
> >>
> >> 
> >> I've not had any production experience, only testing.
> >>
> >> From a cloud registered endpoint perspective, you'll likely be using the
> "Call" button and have to enter the video address of the meeting. You 
> shouldn't
> have any issues.
> >>
> >> What I don't like is the fact that forwarded meeting invites do not appear 
> >> on
> the screen and/or show up with a join button.
> >>
> >> Scenario: Your org is exclusively Webex, but your vendor sends you a Zoom
> invite. They can't book your room. You need to book it yourself. The easiest 
> way
> is to setup your O365 to allow rooms to auto-accept forwarded invites (and
> include all meeting notes). However, the meeting information never shows up
> on the device. I have tried replicating the meeting information via copy and
> paste and I believe that works (now I can't remember). My beef was that even
> forwarded Webex invites don't work!
> >>
> >> I'm hoping they resolve that soon.
> >>
> >> You could also consider writing an applet/macro for the device for Zoom,
> etc that asks for a meeting number and tags the domain.
> >>
> >> The other thing is, if the far end supports video endpoints, they typically
> support call back as well.
> >>
> >> The other other thing is, if you download the far end's mobile app, you
> should be able to click on the meeting in your mobile calendar, it launch the
> mobile app, and that should have a "call my video device" option.
> >>
> >> All things worth trying...
> >>
> >> Let us know 

Re: [cisco-voip] big news - Microsoft and Cisco collaborating again!

[Pawlowski, Adam]

Reading into the blog there are two parts to this as far as I can tell.

One of them is some sort of interop, which I’m going to guess maybe something 
similar to CMS or a cloud equivalent. The experience on that who knows.

The other one appears to be using WebRTC based “guest join” applications on the 
room kit hardware itself. That’s a bit of a moving target if the vendors aren’t 
working together but it would be very much so needed to be able to join Zoom or 
GoToMeeting meetings that no one pays for the premium services for.

The call control strip in that ‘video’ on the blog post doesn’t appear to me to 
be a Cisco one so maybe that is the WebRTC mockup?





From: cisco-voip  On Behalf Of James 
Buchanan
Sent: Wednesday, November 6, 2019 8:55 AM
To: Lelio Fulgenzi 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] big news - Microsoft and Cisco collaborating again!

I didn't see anything in the announcement about CUCM period. I thought it was 
really the ability to join each other's meetings from each other's room-based 
video systems. Issues such as federation, etc. don't seem to be addressed in 
the announcement.

On Wed, Nov 6, 2019 at 1:49 PM Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:
It will be interesting to see how much CUCM integration there is. From what I 
gather, it's going to be Webex room kits that are cloud registered that will 
have access. But, who knows, could be more!

---
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



-Original Message-
From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of James Andrewartha
Sent: Tuesday, November 5, 2019 10:41 PM
To: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] big news - Microsoft and Cisco collaborating again!

On 5/11/19 8:21 pm, Palmer, Brian wrote:
> I always figured that Microsoft would utilize Office365 to push more
> away from Cisco products and into the Microsoft Realm.  Microsoft will
> offer enterprise products for free in order to pull them into the fold.
> I figure all Microsoft needs now is to just buy a company that can
> fill the void they lack in the collaboration space and then they can
> attempt to offer an even fuller offering with office included.

Microsoft is trying to build out their cloud PBX product, but it's nowhere near 
enterprise. My suspicion is that some big Microsoft O365 customers with 
complicated CUCM environments wanted to connect it to MS Teams, and Cisco 
didn't want people to rip out Webex room systems for MS Teams Room systems.

Personally I'm quite happy since we fall into the first category, we're heavily 
into the Microsoft ecosystem but have CUCM, and were looking at ripping it out 
so we could get Teams integration, but concluded that Teams Phones and Cloud 
PBX wasn't mature enough for us (yet).

--
James Andrewartha
Network & Projects Engineer
Christ Church Grammar School
Claremont, Western Australia
Ph. (08) 9442 1757
Mob. 0424 160 877
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[cisco-voip] Jabber and the Manager/Assistant Scenario

[Pawlowski, Adam]

Hello all,

I think I've asked this before but my memory gets worse by the minute nowadays.

Is there an answer to the manager/assistant scenario when using Jabber? I don't 
mean IPMA, but, rather, that a VIP or similar has their directory information 
updated to show their telephone number as a main departmental, or that of their 
office/reception area. This works as they'd expect for the first person, but, 
when multiple persons have the same telephone number listed, calls from that 
person seem to show up in Jabber as a randomly selected contact, as Jabber does 
contact resolution based on the number. I can come up with scenarios that work 
for the IP Phones since they don't have this contact resolution concept, but, 
it doesn't carry over to Jabber.

Long and short we don't control the directory and when people don't want to be 
called or a department wants all calls to funnel through an attendant, the 
results are not great.

I suppose this would also apply if someone shares an extension, since again 
display names would be ignored by Jabber you'd never know for sure who you were 
talking with until you answered the call.

Is there some way around this?

Adam


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Re: [cisco-voip] [CAUTION! EXTERNAL] SIP Plar to 8841 phones

[Pawlowski, Adam]

I've gotten this to work without defining the calling line, however, this 
results in all of the lines on the phone attempting to ring down which is lame.

I can see your comment in the blog post about this and I actually have a 
discussion about the use of this in about 40 minutes so --- rad ! 

Nice post thanks for putting that up !

Adam


> -Original Message-
> From: cisco-voip  On Behalf Of
> Mergenthal, Chase
> Sent: Thursday, October 24, 2019 9:12 AM
> To: Myron Young ; cisco-voip voyp list  v...@puck.nether.net>
> Subject: Re: [cisco-voip] [CAUTION! EXTERNAL] SIP Plar to 8841 phones
> 
> Running CUCM 12.5 and a 8851's and it works fine.
> 
> Maybe try to create a new CSS / PT for your null translation pattern, and make
> sure its applied to the phone...
> It might also help if you create a new SIP Dial rule with a single line for 
> the
> PLAR for testing...
> 
> --
> Chase Mergenthal
> 
> -Original Message-
> From: cisco-voip  On Behalf Of Myron
> Young
> Sent: Thursday, October 24, 2019 7:38 AM
> To: cisco-voip voyp list 
> Subject: [CAUTION! EXTERNAL] [cisco-voip] SIP Plar to 8841 phones
> 
> 
> ⚠ The sender was not validated and could be a phish. ⚠ Slow down, read
> carefully and look for signs that it may be a phish. Click the report phish 
> button
> or forward this email to phishing@bestbuy. com if you think it's malicious.
> 
> 
> 
> Hello,
> 
> Has anyone ever gotten the plar to work for SIP 8841 phones? I’m setting it up
> now with one 6901 skinny phone and two 8841 sip phones on CUCM 9.1 but the
> 8841 phones don’t dial the 6901 phone. I’m able to get it working from skinny
> 6901 phone to 8841 phone no problem but can’t get it to work opposite way
> around. I have all the partitions, CSS, translation patterns and sip dial 
> rules
> configured, but when i pick up handset of 8841 phone I just get dial tone
> instead of ring back from calling the 6901 phone.
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Re: [cisco-voip] Webex Proximity Callback to Video Endpoints with CUCM

[Pawlowski, Adam]

I can’t sign in to this but I would agree, I ended up making a pattern-specific 
allow rule in the Expressway’s CPL to let this in but it is not great for 
scaling.



From: cisco-voip  On Behalf Of Brian Meade
Sent: Monday, October 21, 2019 2:01 PM
To: cisco-voip voyp list 
Subject: [cisco-voip] Webex Proximity Callback to Video Endpoints with CUCM

Hey everyone,

With Webex Meetings video callback to a CUCM-registered video endpoint via 
proximity, Webex always tries the d...@domain.com 
format rather than the Primary URI configured.

We've confirmed that the video endpoints advertise the 
d...@domain.com as well as the Primary URI but Webex 
Meetings is setup to prefer d...@domain.com.

In our organization and many customer organizations, the B2B Expressways do not 
allow d...@domain.com calling.

This issue has become much worse with the new Webex join experience launched 
recently.

At ePlus, my coworker opened a Collaboration Idea for resolving this- 
https://ciscocollaboration.ideas.aha.io/ideas/COLLAB-I-3255

Webex BU is using this to prioritize fixes and new features.

Can you vote for this issue if it is applicable to yourself or a customer of 
yours?  We're really hoping to drive some visibility of this issue.

Thanks,
Brian Meade
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Re: [cisco-voip] UCCx and Jabber - compatible or not?

[Pawlowski, Adam]

Only in Phone Only or it doesn’t work, or if you are assigning a custom 
configuration or phone.

In our case (love to toot my own horn), jabber-config.xml turns off phone and 
voicemail. If you want a person to cover a group VM box, or have special 
features (pChat etc) the only way to tell the client what to do is to build the 
CSF device to steer the configuration.

Regarding this, there’s a community forum post that says if you sign in to the 
phone, then open Jabber, then sign out of the phone, that it “works” because 
CTI still can find the station. I assume this has something to do with the 
order in which the devices are retrieved when you ask for a terminal list with 
CTI but I don’t know.

It was followed up with a long post of “yeah maybe it works maybe it doesn’t 
but it is not supported”.

Someone should write some nice wrapped code to move your DN around like a fake 
extension mobility. If you have no other considerations, with AXL it’s saving 
the phone back with that DN out of the line list, and saving it back on the CSF 
device. Probably break a bunch of things but if not I’m sure there’s a few 
bucks to be made off of that.

Since it is a Friday I don’t intend to touch any of that and demo is since I’ll 
probably end up deleting everyone’s phone line or something.

Adam

From: cisco-voip  On Behalf Of Lelio 
Fulgenzi
Sent: Friday, October 18, 2019 3:36 PM
To: Matthew Loraditch ; Johnson, Tim 
; voyp list, cisco-voip (cisco-voip@puck.nether.net) 

Subject: Re: [cisco-voip] UCCx and Jabber - compatible or not?

Ok, now you’re really freaking me out.

You don’t need a CSF device to log in with Jabber?


---
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]

From: Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>>
Sent: Friday, October 18, 2019 3:32 PM
To: Johnson, Tim mailto:johns...@cmich.edu>>; Lelio 
Fulgenzi mailto:le...@uoguelph.ca>>; voyp list, cisco-voip 
(cisco-voip@puck.nether.net) 
mailto:cisco-voip@puck.nether.net>>
Subject: RE: UCCx and Jabber - compatible or not?

Yeah just don’t create one if you don’t like having them blank. You don’t need 
a CSF for deskphone control.



Matthew Loraditch​

Sr. Network Engineer


p: 443.541.1518



w: www.heliontechnologies.com

 |

e: mloradi...@heliontechnologies.com


[Helion Technologies]


[Facebook]



[Twitter]


[LinkedIn]







From: Johnson, Tim mailto:johns...@cmich.edu>>
Sent: Friday, October 18, 2019 3:30 PM
To: Lelio Fulgenzi mailto:le...@uoguelph.ca>>; Matthew 
Loraditch 
mailto:mloradi...@heliontechnologies.com>>; 
voyp list, cisco-voip 
(cisco-voip@puck.nether.net) 
mailto:cisco-voip@puck.nether.net>>
Subject: RE: UCCx and Jabber - compatible or not?

Yeah, just don’t have a CSF device assigned to that user. Or, if you do have a 
CSF device for that user, make sure their IPCC extension is not assigned to it.


From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Lelio Fulgenzi
Sent: Friday, October 18, 2019 3:30 PM
To: Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>>; 
voyp list, cisco-voip 
(cisco-voip@puck.nether.net) 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] UCCx and Jabber - compatible or not?

Interesting. Devices without DNs just freak me out!

---
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]

From: Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>>
Sent: Friday, October 18, 2019 3:28 PM
To: Lelio Fulgenzi mailto:le...@uoguelph.ca>>; voyp list, 
cisco-voip (cisco-voip@puck.nether.net) 
mailto:cisco-voip@puck.nether.net>>
Subject: RE: UCCx and Jabber - compatible or not?

Correct. If you have agents who want to  use Jabber and have 

Re: [cisco-voip] tools to parse CDR quickly on Windows (windows grep?)

[Pawlowski, Adam]

We have an in house tool that parses the CDR records being loaded by the loader 
into a database.

The data is indexed so searching with a front end is pretty easy .

We're actually eliminating this tool as we do not want to store call detail 
records for liability and data storage reasons . We do not provide user 
reporting on call detail and this has yet to significantly impact anyone's 
business that I'm aware of .

Adam



From: cisco-voip  On Behalf Of Lelio 
Fulgenzi
Sent: Thursday, October 17, 2019 9:55 AM
To: voyp list, cisco-voip (cisco-voip@puck.nether.net) 

Subject: [cisco-voip] tools to parse CDR quickly on Windows (windows grep?)


What tools have others used to parse CDRs for reports for users/departments?

I've written a script, but would like to be able to Windows-a-fy this so others 
can do this easily.

Is there a good windows grep out there or something else?

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] Super Duper Improved Jabber on it's way to you...

[Pawlowski, Adam]

Yeah the feature parity is getting better with regard to a Webex Meetings 
meeting and joining from Jabber, but if your users play around with the 
in-meeting content, whiteboard (who actually uses these), etc then it becomes 
painful to explain which way to join what meeting and how.

I would pre-deploy the Webex Meetings app with the installer flags set to not 
add desktop icons and pop up because it certainly will do that do you once 
you’ve joined a modern meeting by launching it from Jabber. I am still very 
salty about that.

Otherwise if you do have the luxury of forcing it to launch in Jabber only and 
that works for your customers and their needs/workflow then that is certainly 
less complex to deal with.

Other than that we’ve not really had any issues with Jabber 12.7 specifically 
at this point.



From: cisco-voip  On Behalf Of Brian Meade
Sent: Friday, October 11, 2019 10:22 AM
To: Lelio Fulgenzi 
Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net) 

Subject: Re: [cisco-voip] Super Duper Improved Jabber on it's way to you...

Heads up, major issue is having Jabber 12.7 and Webex Meetings app installed.

By default, Jabber 12.7 tries to launch Webex meetings in Jabber as a SIP 
client which obviously isn't the full Webex experience.

Users can change if they join with Webex Meetings app or Jabber app in dropdown 
on meeting reminder or in the dropdown at the bottom of the Meetings tab.

You can change this in the jabber-config.xml under Options section with a 
parameter called ConfMediaType- 
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/jabber/12_7/cjab_b_parameter-reference-guide-jabber-127/cjab_b_parameter-reference-guide-jabber-127_chapter_0100.html#CJAB_RF_C2BEDA3A_00

You can also change the default layout back to the old style via the UXModel 
parameter- 
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/jabber/12_7/cjab_b_parameter-reference-guide-jabber-127/cjab_b_parameter-reference-guide-jabber-127_chapter_011.html#reference_B422E797A575DD26F869FE1DAFD0CB96





On Fri, Oct 11, 2019 at 9:36 AM Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:

Are you excited? Cisco sure is For Jabber 12.7 !

At my Cisco Live recap, I talked about how Jabber would be seeing a major 
cosmetic overhaul to match it's other collaboration clients *cough* Webex Teams 
*cough* was Spark *cough*. It was released mid-September, and aside from the 
visual changes, I haven't heard anything bad about it. There will be a dot one 
version released in the next month or so, which we will likely use as the roll 
out version, so we want to get some experience with 12.7 as soon as possible.

I'll be testing on my machine today and if there's nothing glaring, I'll be 
pushing it out to the rest of the team as well.

Let me know if you have any issues.

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] Jabber Softphone over WiFi

[Pawlowski, Adam]

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) 

Subject: Re: [cisco-voip] Jabber Softphone over WiFi


As long as you have all the components in place, Jabber as a desktop 
replacement is doable. The issue comes down to ... do you have all the 
components in place and/or are you ready to live without the feature a missing 
component gives you?

Some components are not an option, say, split view DNS and Expressway for 
on-prem and off-prem delivery. There's no way I've found to do that without 
split view DNS. We ended up having to deploy a set of delegated DNS servers for 
a specific discovery domain in order to deliver the off-premise and on-premise 
functionality.

Some components are ok to put aside... i.e. quality of service.

The other issue is features and functionality. Jabber is close, but not 100% 
feature parity. For example, no park on Jabber for Windows. I also just 
noticed, is there a way to deploy speed dials on Jabber? I'm not sure.

Just some general thoughts about that.

---
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]

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Casper, Steven via cisco-voip
Sent: Thursday, October 10, 2019 12:55 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Jabber Softphone over WiFi

Has anybody tried this as an IP phone replacement strategy for a large office 
location?

Thanks,
Steve



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Re: [cisco-voip] best way to change device and user related information

[Pawlowski, Adam]

You for sure would have to touch UCCX if you change the end users around in the 
UCM at all in my experience.

Assuming everything was built properly with everything tied together, you 
should be able to get information from the end users about their associated 
devices and extensions.

I like pain so I would take an option one type approach where I'd:


-  Query for user IDs using LDAP since you can do that, or get a list 
of them of the users to remove

-  Pull the UCM data using AXL (or bat or whatever it doesn't matter) 
to get the associated devices, primary extensions, and ipcc extensions of the 
users you don't want

-  Update the sync to drop the users and wait for garbage collection or 
delete with dbquery

-  I would use AXL to delete the devices and directory numbers, but I 
don't know how you would do that with BAT if it does anything resembling a 
delete.


You could also try a dbquery to delete all devices where you drop all devices 
of class 562, 575, 503, 652 and fkenduser is null. Phones are a bit trickier 
but you'd know what they are from pulling that data earlier, so you can remove 
those with some method. At that point the DNs should show up on unassigned DNs 
and can be deleted from there.

Someone probably has a better way to do this. I know quick add will delete if 
you convert the user to local and then delete them, it prompts to remove the 
devices, but, that would be a pain for 1000 users.

Adam


From: cisco-voip  On Behalf Of naresh 
rathore
Sent: Wednesday, October 9, 2019 2:50 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] best way to change device and user related information

hi


last year we have installed cucm 11.5 and integrated with uccx 11.6. company 
which was hired, copied all organization user and device related information to 
this cucm 11.5 (1200 users and Devices). but uccx was integrated and only one 
branch users (around 150 users and devices)(For e.g branch Z) were configured 
in uccx.

now i have been told to keep Branch Z (around 150 users and 150 devices) 
related device/jabber/Iphone and delete remaining 1000 devices and 1000 user 
related information.

I have two options

Option 1

  1.  change the ldap search based to specific branch
  2.  get the user information and change those user to local users using BAT 
(if possible) and delete these users using BAT tool
  3.  get the device information and delete those devices using BAT
  4.  get the device profile information of other branch and delete these using 
BAT
  5.  this way Branch Z users will still be on CUCM and UCCX and all will work
Option 2
second option will be to delete all users and devices and add device and user 
related to Branch Z using BAT tool. in this case i have to configure IPCC 
extension and i might have to configure skill/resource side of things on UCCX

Pls suggest what you guys recommend?


Regards


Naray
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Re: [cisco-voip] SIP Domain substitution

[Pawlowski, Adam]

I am a glutton for punishment, so for our PoC deployment we have a hybrid 
connector cluster, and a B2B/MRA cluster that does everything else.

We have no non-numeric URIs, so I use CPL rules (going to move that to XML 
since managing them otherwise is not great) to accept calls formatted to my 
“collaboration endpoints” (ce-…) and customer URIs which all start with a 
letter. Calls targeted directly at the expressway, elsewise to the domains, or 
E164 calls with no domain are all dropped.

The Expressway doesn’t accept in anything that would be directly routable once 
it hits the UCM to try and avoid this problem.

While it is certainly possible I suppose to have an authenticated zone that can 
place dialable calls out via UCM, CUBE, SBC, etc that seems like not the best 
idea, but impossible as a Webex endpoint won’t get you there.

Out of the ~1000 calls in the Expressway E cluster’s history, spanning about 
the last two hours, 4 of them were legit from MRA clients and the rest were 
spam so I would not want to leave any opportunity there either.


From: cisco-voip  On Behalf Of Lelio 
Fulgenzi
Sent: Saturday, October 5, 2019 10:13 AM
To: Ryan Huff 
Cc: cisco-voip voyp list 
Subject: Re: [cisco-voip] SIP Domain substitution


Yeah. I hear ya. I’ll have to do more research. To see how much effort is 
actually required.

My biggest concern is enabling PSTN access and not opening up a security 
exposure.

I was at the collab techtorial where the presenter put the fear of Dog into us 
about spinning up a separate expressway cluster to ensure no pstn abuse.

I’ll try with Macros first. If I can get extension dialing easy enough, it buys 
me s proof of concept that can get me more support.
-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

[University of Guelph Cornerstone with Improve Life tagline]

On Oct 5, 2019, at 9:36 AM, Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
Webex Hybrid Calling definitely sounds like a good fit for you then; it’ll also 
give your cloud registered devices a way to dial the on-prem extensions.

Basically, when the cloud registered device is setup, you select Hybrid calling 
as the PSTN service (assuming Hybrid calling has already been setup) and then 
it sends signaling to CUCM via Expressway-C > CUCM. The effective media path is 
device <> cloud.

If you have Expressway B2B, you can also leverage that to allow your cloud 
devices to make B2B SIP calls via Cloud > Expressway-C > CUCM > Expressway-C > 
Expressway-E > Internet. The idea was to make Hybrid Calling for cloud devices 
“transparent” to the user over cloud calling in terms of PSTN capabilities, 
with the added feature of interacting with on-prem extensions as if the device 
was registered on-prem.

There are more than a few scenarios where Webex Hybrid calling will trombone 
the call, and it’s by design and due to the nature of the scenario. Under the 
hood, the call legs and SIP messages can get hairy from a troubleshooting 
perspective (TranslatorX is a beast for this), and Cisco has had more than a 
few complaints about it, but it is what it is.
Sent from my iPhone


On Oct 4, 2019, at 23:40, Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:


I’m trying to do a bit of everything, really.

In our case, I’d like to have a few cloud registered WebEx room devices still 
be able to call our extensions. It’s the one thing we loose vs. on-prem reg 
WebEx room devices.

I still have to get it working (I’m guessing there are IP address ranges I have 
to permit) but a cloud registered device can call @myphone.acme.com

If I can create a macro on cloud registered devices like you can on CE devices, 
then it gives me that functionality.

We don’t have Webex Teams deployed. We don’t have Webex pstn / calling enabled.

So it’s either a hybrid call setup or a macro.

I’ll have to investigate further.
-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

[University of Guelph Cornerstone with Improve Life tagline]

On Oct 4, 2019, at 11:25 PM, Anthony Holloway 
mailto:avholloway+cisco-v...@gmail.com>> wrote:
Wait, I thought this was for other businesses to call you.  Are